Threads.cpp revision ca5e6143740299c877d69e97f7968cd04476d32c
1/* 2** 3** Copyright 2012, The Android Open Source Project 4** 5** Licensed under the Apache License, Version 2.0 (the "License"); 6** you may not use this file except in compliance with the License. 7** You may obtain a copy of the License at 8** 9** http://www.apache.org/licenses/LICENSE-2.0 10** 11** Unless required by applicable law or agreed to in writing, software 12** distributed under the License is distributed on an "AS IS" BASIS, 13** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. 14** See the License for the specific language governing permissions and 15** limitations under the License. 16*/ 17 18 19#define LOG_TAG "AudioFlinger" 20//#define LOG_NDEBUG 0 21#define ATRACE_TAG ATRACE_TAG_AUDIO 22 23#include "Configuration.h" 24#include <math.h> 25#include <fcntl.h> 26#include <linux/futex.h> 27#include <sys/stat.h> 28#include <sys/syscall.h> 29#include <cutils/properties.h> 30#include <media/AudioParameter.h> 31#include <media/AudioResamplerPublic.h> 32#include <utils/Log.h> 33#include <utils/Trace.h> 34 35#include <private/media/AudioTrackShared.h> 36#include <hardware/audio.h> 37#include <audio_effects/effect_ns.h> 38#include <audio_effects/effect_aec.h> 39#include <audio_utils/primitives.h> 40#include <audio_utils/format.h> 41#include <audio_utils/minifloat.h> 42 43// NBAIO implementations 44#include <media/nbaio/AudioStreamInSource.h> 45#include <media/nbaio/AudioStreamOutSink.h> 46#include <media/nbaio/MonoPipe.h> 47#include <media/nbaio/MonoPipeReader.h> 48#include <media/nbaio/Pipe.h> 49#include <media/nbaio/PipeReader.h> 50#include <media/nbaio/SourceAudioBufferProvider.h> 51 52#include <powermanager/PowerManager.h> 53 54#include <common_time/cc_helper.h> 55#include <common_time/local_clock.h> 56 57#include "AudioFlinger.h" 58#include "AudioMixer.h" 59#include "BufferProviders.h" 60#include "FastMixer.h" 61#include "FastCapture.h" 62#include "ServiceUtilities.h" 63#include "SchedulingPolicyService.h" 64 65#ifdef ADD_BATTERY_DATA 66#include <media/IMediaPlayerService.h> 67#include <media/IMediaDeathNotifier.h> 68#endif 69 70#ifdef DEBUG_CPU_USAGE 71#include <cpustats/CentralTendencyStatistics.h> 72#include <cpustats/ThreadCpuUsage.h> 73#endif 74 75// ---------------------------------------------------------------------------- 76 77// Note: the following macro is used for extremely verbose logging message. In 78// order to run with ALOG_ASSERT turned on, we need to have LOG_NDEBUG set to 79// 0; but one side effect of this is to turn all LOGV's as well. Some messages 80// are so verbose that we want to suppress them even when we have ALOG_ASSERT 81// turned on. Do not uncomment the #def below unless you really know what you 82// are doing and want to see all of the extremely verbose messages. 83//#define VERY_VERY_VERBOSE_LOGGING 84#ifdef VERY_VERY_VERBOSE_LOGGING 85#define ALOGVV ALOGV 86#else 87#define ALOGVV(a...) do { } while(0) 88#endif 89 90// TODO: Move these macro/inlines to a header file. 91#define max(a, b) ((a) > (b) ? (a) : (b)) 92template <typename T> 93static inline T min(const T& a, const T& b) 94{ 95 return a < b ? a : b; 96} 97 98#ifndef ARRAY_SIZE 99#define ARRAY_SIZE(a) (sizeof(a) / sizeof(a[0])) 100#endif 101 102namespace android { 103 104// retry counts for buffer fill timeout 105// 50 * ~20msecs = 1 second 106static const int8_t kMaxTrackRetries = 50; 107static const int8_t kMaxTrackStartupRetries = 50; 108// allow less retry attempts on direct output thread. 109// direct outputs can be a scarce resource in audio hardware and should 110// be released as quickly as possible. 111static const int8_t kMaxTrackRetriesDirect = 2; 112 113// don't warn about blocked writes or record buffer overflows more often than this 114static const nsecs_t kWarningThrottleNs = seconds(5); 115 116// RecordThread loop sleep time upon application overrun or audio HAL read error 117static const int kRecordThreadSleepUs = 5000; 118 119// maximum time to wait in sendConfigEvent_l() for a status to be received 120static const nsecs_t kConfigEventTimeoutNs = seconds(2); 121 122// minimum sleep time for the mixer thread loop when tracks are active but in underrun 123static const uint32_t kMinThreadSleepTimeUs = 5000; 124// maximum divider applied to the active sleep time in the mixer thread loop 125static const uint32_t kMaxThreadSleepTimeShift = 2; 126 127// minimum normal sink buffer size, expressed in milliseconds rather than frames 128static const uint32_t kMinNormalSinkBufferSizeMs = 20; 129// maximum normal sink buffer size 130static const uint32_t kMaxNormalSinkBufferSizeMs = 24; 131 132// Offloaded output thread standby delay: allows track transition without going to standby 133static const nsecs_t kOffloadStandbyDelayNs = seconds(1); 134 135// Whether to use fast mixer 136static const enum { 137 FastMixer_Never, // never initialize or use: for debugging only 138 FastMixer_Always, // always initialize and use, even if not needed: for debugging only 139 // normal mixer multiplier is 1 140 FastMixer_Static, // initialize if needed, then use all the time if initialized, 141 // multiplier is calculated based on min & max normal mixer buffer size 142 FastMixer_Dynamic, // initialize if needed, then use dynamically depending on track load, 143 // multiplier is calculated based on min & max normal mixer buffer size 144 // FIXME for FastMixer_Dynamic: 145 // Supporting this option will require fixing HALs that can't handle large writes. 146 // For example, one HAL implementation returns an error from a large write, 147 // and another HAL implementation corrupts memory, possibly in the sample rate converter. 148 // We could either fix the HAL implementations, or provide a wrapper that breaks 149 // up large writes into smaller ones, and the wrapper would need to deal with scheduler. 150} kUseFastMixer = FastMixer_Static; 151 152// Whether to use fast capture 153static const enum { 154 FastCapture_Never, // never initialize or use: for debugging only 155 FastCapture_Always, // always initialize and use, even if not needed: for debugging only 156 FastCapture_Static, // initialize if needed, then use all the time if initialized 157} kUseFastCapture = FastCapture_Static; 158 159// Priorities for requestPriority 160static const int kPriorityAudioApp = 2; 161static const int kPriorityFastMixer = 3; 162static const int kPriorityFastCapture = 3; 163 164// IAudioFlinger::createTrack() reports back to client the total size of shared memory area 165// for the track. The client then sub-divides this into smaller buffers for its use. 166// Currently the client uses N-buffering by default, but doesn't tell us about the value of N. 167// So for now we just assume that client is double-buffered for fast tracks. 168// FIXME It would be better for client to tell AudioFlinger the value of N, 169// so AudioFlinger could allocate the right amount of memory. 170// See the client's minBufCount and mNotificationFramesAct calculations for details. 171 172// This is the default value, if not specified by property. 173static const int kFastTrackMultiplier = 2; 174 175// The minimum and maximum allowed values 176static const int kFastTrackMultiplierMin = 1; 177static const int kFastTrackMultiplierMax = 2; 178 179// The actual value to use, which can be specified per-device via property af.fast_track_multiplier. 180static int sFastTrackMultiplier = kFastTrackMultiplier; 181 182// See Thread::readOnlyHeap(). 183// Initially this heap is used to allocate client buffers for "fast" AudioRecord. 184// Eventually it will be the single buffer that FastCapture writes into via HAL read(), 185// and that all "fast" AudioRecord clients read from. In either case, the size can be small. 186static const size_t kRecordThreadReadOnlyHeapSize = 0x2000; 187 188// ---------------------------------------------------------------------------- 189 190static pthread_once_t sFastTrackMultiplierOnce = PTHREAD_ONCE_INIT; 191 192static void sFastTrackMultiplierInit() 193{ 194 char value[PROPERTY_VALUE_MAX]; 195 if (property_get("af.fast_track_multiplier", value, NULL) > 0) { 196 char *endptr; 197 unsigned long ul = strtoul(value, &endptr, 0); 198 if (*endptr == '\0' && kFastTrackMultiplierMin <= ul && ul <= kFastTrackMultiplierMax) { 199 sFastTrackMultiplier = (int) ul; 200 } 201 } 202} 203 204// ---------------------------------------------------------------------------- 205 206#ifdef ADD_BATTERY_DATA 207// To collect the amplifier usage 208static void addBatteryData(uint32_t params) { 209 sp<IMediaPlayerService> service = IMediaDeathNotifier::getMediaPlayerService(); 210 if (service == NULL) { 211 // it already logged 212 return; 213 } 214 215 service->addBatteryData(params); 216} 217#endif 218 219 220// ---------------------------------------------------------------------------- 221// CPU Stats 222// ---------------------------------------------------------------------------- 223 224class CpuStats { 225public: 226 CpuStats(); 227 void sample(const String8 &title); 228#ifdef DEBUG_CPU_USAGE 229private: 230 ThreadCpuUsage mCpuUsage; // instantaneous thread CPU usage in wall clock ns 231 CentralTendencyStatistics mWcStats; // statistics on thread CPU usage in wall clock ns 232 233 CentralTendencyStatistics mHzStats; // statistics on thread CPU usage in cycles 234 235 int mCpuNum; // thread's current CPU number 236 int mCpukHz; // frequency of thread's current CPU in kHz 237#endif 238}; 239 240CpuStats::CpuStats() 241#ifdef DEBUG_CPU_USAGE 242 : mCpuNum(-1), mCpukHz(-1) 243#endif 244{ 245} 246 247void CpuStats::sample(const String8 &title 248#ifndef DEBUG_CPU_USAGE 249 __unused 250#endif 251 ) { 252#ifdef DEBUG_CPU_USAGE 253 // get current thread's delta CPU time in wall clock ns 254 double wcNs; 255 bool valid = mCpuUsage.sampleAndEnable(wcNs); 256 257 // record sample for wall clock statistics 258 if (valid) { 259 mWcStats.sample(wcNs); 260 } 261 262 // get the current CPU number 263 int cpuNum = sched_getcpu(); 264 265 // get the current CPU frequency in kHz 266 int cpukHz = mCpuUsage.getCpukHz(cpuNum); 267 268 // check if either CPU number or frequency changed 269 if (cpuNum != mCpuNum || cpukHz != mCpukHz) { 270 mCpuNum = cpuNum; 271 mCpukHz = cpukHz; 272 // ignore sample for purposes of cycles 273 valid = false; 274 } 275 276 // if no change in CPU number or frequency, then record sample for cycle statistics 277 if (valid && mCpukHz > 0) { 278 double cycles = wcNs * cpukHz * 0.000001; 279 mHzStats.sample(cycles); 280 } 281 282 unsigned n = mWcStats.n(); 283 // mCpuUsage.elapsed() is expensive, so don't call it every loop 284 if ((n & 127) == 1) { 285 long long elapsed = mCpuUsage.elapsed(); 286 if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) { 287 double perLoop = elapsed / (double) n; 288 double perLoop100 = perLoop * 0.01; 289 double perLoop1k = perLoop * 0.001; 290 double mean = mWcStats.mean(); 291 double stddev = mWcStats.stddev(); 292 double minimum = mWcStats.minimum(); 293 double maximum = mWcStats.maximum(); 294 double meanCycles = mHzStats.mean(); 295 double stddevCycles = mHzStats.stddev(); 296 double minCycles = mHzStats.minimum(); 297 double maxCycles = mHzStats.maximum(); 298 mCpuUsage.resetElapsed(); 299 mWcStats.reset(); 300 mHzStats.reset(); 301 ALOGD("CPU usage for %s over past %.1f secs\n" 302 " (%u mixer loops at %.1f mean ms per loop):\n" 303 " us per mix loop: mean=%.0f stddev=%.0f min=%.0f max=%.0f\n" 304 " %% of wall: mean=%.1f stddev=%.1f min=%.1f max=%.1f\n" 305 " MHz: mean=%.1f, stddev=%.1f, min=%.1f max=%.1f", 306 title.string(), 307 elapsed * .000000001, n, perLoop * .000001, 308 mean * .001, 309 stddev * .001, 310 minimum * .001, 311 maximum * .001, 312 mean / perLoop100, 313 stddev / perLoop100, 314 minimum / perLoop100, 315 maximum / perLoop100, 316 meanCycles / perLoop1k, 317 stddevCycles / perLoop1k, 318 minCycles / perLoop1k, 319 maxCycles / perLoop1k); 320 321 } 322 } 323#endif 324}; 325 326// ---------------------------------------------------------------------------- 327// ThreadBase 328// ---------------------------------------------------------------------------- 329 330// static 331const char *AudioFlinger::ThreadBase::threadTypeToString(AudioFlinger::ThreadBase::type_t type) 332{ 333 switch (type) { 334 case MIXER: 335 return "MIXER"; 336 case DIRECT: 337 return "DIRECT"; 338 case DUPLICATING: 339 return "DUPLICATING"; 340 case RECORD: 341 return "RECORD"; 342 case OFFLOAD: 343 return "OFFLOAD"; 344 default: 345 return "unknown"; 346 } 347} 348 349String8 devicesToString(audio_devices_t devices) 350{ 351 static const struct mapping { 352 audio_devices_t mDevices; 353 const char * mString; 354 } mappingsOut[] = { 355 AUDIO_DEVICE_OUT_EARPIECE, "EARPIECE", 356 AUDIO_DEVICE_OUT_SPEAKER, "SPEAKER", 357 AUDIO_DEVICE_OUT_WIRED_HEADSET, "WIRED_HEADSET", 358 AUDIO_DEVICE_OUT_WIRED_HEADPHONE, "WIRED_HEADPHONE", 359 AUDIO_DEVICE_OUT_BLUETOOTH_SCO, "BLUETOOTH_SCO", 360 AUDIO_DEVICE_OUT_BLUETOOTH_SCO_HEADSET, "BLUETOOTH_SCO_HEADSET", 361 AUDIO_DEVICE_OUT_BLUETOOTH_SCO_CARKIT, "BLUETOOTH_SCO_CARKIT", 362 AUDIO_DEVICE_OUT_BLUETOOTH_A2DP, "BLUETOOTH_A2DP", 363 AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES, "BLUETOOTH_A2DP_HEADPHONES", 364 AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER, "BLUETOOTH_A2DP_SPEAKER", 365 AUDIO_DEVICE_OUT_AUX_DIGITAL, "AUX_DIGITAL", 366 AUDIO_DEVICE_OUT_HDMI, "HDMI", 367 AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET, "ANLG_DOCK_HEADSET", 368 AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET, "DGTL_DOCK_HEADSET", 369 AUDIO_DEVICE_OUT_USB_ACCESSORY, "USB_ACCESSORY", 370 AUDIO_DEVICE_OUT_USB_DEVICE, "USB_DEVICE", 371 AUDIO_DEVICE_OUT_TELEPHONY_TX, "TELEPHONY_TX", 372 AUDIO_DEVICE_OUT_LINE, "LINE", 373 AUDIO_DEVICE_OUT_HDMI_ARC, "HDMI_ARC", 374 AUDIO_DEVICE_OUT_SPDIF, "SPDIF", 375 AUDIO_DEVICE_OUT_FM, "FM", 376 AUDIO_DEVICE_OUT_AUX_LINE, "AUX_LINE", 377 AUDIO_DEVICE_OUT_SPEAKER_SAFE, "SPEAKER_SAFE", 378 AUDIO_DEVICE_OUT_IP, "IP", 379 AUDIO_DEVICE_NONE, "NONE", // must be last 380 }, mappingsIn[] = { 381 AUDIO_DEVICE_IN_COMMUNICATION, "COMMUNICATION", 382 AUDIO_DEVICE_IN_AMBIENT, "AMBIENT", 383 AUDIO_DEVICE_IN_BUILTIN_MIC, "BUILTIN_MIC", 384 AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET, "BLUETOOTH_SCO_HEADSET", 385 AUDIO_DEVICE_IN_WIRED_HEADSET, "WIRED_HEADSET", 386 AUDIO_DEVICE_IN_AUX_DIGITAL, "AUX_DIGITAL", 387 AUDIO_DEVICE_IN_VOICE_CALL, "VOICE_CALL", 388 AUDIO_DEVICE_IN_TELEPHONY_RX, "TELEPHONY_RX", 389 AUDIO_DEVICE_IN_BACK_MIC, "BACK_MIC", 390 AUDIO_DEVICE_IN_REMOTE_SUBMIX, "REMOTE_SUBMIX", 391 AUDIO_DEVICE_IN_ANLG_DOCK_HEADSET, "ANLG_DOCK_HEADSET", 392 AUDIO_DEVICE_IN_DGTL_DOCK_HEADSET, "DGTL_DOCK_HEADSET", 393 AUDIO_DEVICE_IN_USB_ACCESSORY, "USB_ACCESSORY", 394 AUDIO_DEVICE_IN_USB_DEVICE, "USB_DEVICE", 395 AUDIO_DEVICE_IN_FM_TUNER, "FM_TUNER", 396 AUDIO_DEVICE_IN_TV_TUNER, "TV_TUNER", 397 AUDIO_DEVICE_IN_LINE, "LINE", 398 AUDIO_DEVICE_IN_SPDIF, "SPDIF", 399 AUDIO_DEVICE_IN_BLUETOOTH_A2DP, "BLUETOOTH_A2DP", 400 AUDIO_DEVICE_IN_LOOPBACK, "LOOPBACK", 401 AUDIO_DEVICE_IN_IP, "IP", 402 AUDIO_DEVICE_NONE, "NONE", // must be last 403 }; 404 String8 result; 405 audio_devices_t allDevices = AUDIO_DEVICE_NONE; 406 const mapping *entry; 407 if (devices & AUDIO_DEVICE_BIT_IN) { 408 devices &= ~AUDIO_DEVICE_BIT_IN; 409 entry = mappingsIn; 410 } else { 411 entry = mappingsOut; 412 } 413 for ( ; entry->mDevices != AUDIO_DEVICE_NONE; entry++) { 414 allDevices = (audio_devices_t) (allDevices | entry->mDevices); 415 if (devices & entry->mDevices) { 416 if (!result.isEmpty()) { 417 result.append("|"); 418 } 419 result.append(entry->mString); 420 } 421 } 422 if (devices & ~allDevices) { 423 if (!result.isEmpty()) { 424 result.append("|"); 425 } 426 result.appendFormat("0x%X", devices & ~allDevices); 427 } 428 if (result.isEmpty()) { 429 result.append(entry->mString); 430 } 431 return result; 432} 433 434String8 inputFlagsToString(audio_input_flags_t flags) 435{ 436 static const struct mapping { 437 audio_input_flags_t mFlag; 438 const char * mString; 439 } mappings[] = { 440 AUDIO_INPUT_FLAG_FAST, "FAST", 441 AUDIO_INPUT_FLAG_HW_HOTWORD, "HW_HOTWORD", 442 AUDIO_INPUT_FLAG_NONE, "NONE", // must be last 443 }; 444 String8 result; 445 audio_input_flags_t allFlags = AUDIO_INPUT_FLAG_NONE; 446 const mapping *entry; 447 for (entry = mappings; entry->mFlag != AUDIO_INPUT_FLAG_NONE; entry++) { 448 allFlags = (audio_input_flags_t) (allFlags | entry->mFlag); 449 if (flags & entry->mFlag) { 450 if (!result.isEmpty()) { 451 result.append("|"); 452 } 453 result.append(entry->mString); 454 } 455 } 456 if (flags & ~allFlags) { 457 if (!result.isEmpty()) { 458 result.append("|"); 459 } 460 result.appendFormat("0x%X", flags & ~allFlags); 461 } 462 if (result.isEmpty()) { 463 result.append(entry->mString); 464 } 465 return result; 466} 467 468String8 outputFlagsToString(audio_output_flags_t flags) 469{ 470 static const struct mapping { 471 audio_output_flags_t mFlag; 472 const char * mString; 473 } mappings[] = { 474 AUDIO_OUTPUT_FLAG_DIRECT, "DIRECT", 475 AUDIO_OUTPUT_FLAG_PRIMARY, "PRIMARY", 476 AUDIO_OUTPUT_FLAG_FAST, "FAST", 477 AUDIO_OUTPUT_FLAG_DEEP_BUFFER, "DEEP_BUFFER", 478 AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD, "COMPRESS_OFFLOAD", 479 AUDIO_OUTPUT_FLAG_NON_BLOCKING, "NON_BLOCKING", 480 AUDIO_OUTPUT_FLAG_HW_AV_SYNC, "HW_AV_SYNC", 481 AUDIO_OUTPUT_FLAG_NONE, "NONE", // must be last 482 }; 483 String8 result; 484 audio_output_flags_t allFlags = AUDIO_OUTPUT_FLAG_NONE; 485 const mapping *entry; 486 for (entry = mappings; entry->mFlag != AUDIO_OUTPUT_FLAG_NONE; entry++) { 487 allFlags = (audio_output_flags_t) (allFlags | entry->mFlag); 488 if (flags & entry->mFlag) { 489 if (!result.isEmpty()) { 490 result.append("|"); 491 } 492 result.append(entry->mString); 493 } 494 } 495 if (flags & ~allFlags) { 496 if (!result.isEmpty()) { 497 result.append("|"); 498 } 499 result.appendFormat("0x%X", flags & ~allFlags); 500 } 501 if (result.isEmpty()) { 502 result.append(entry->mString); 503 } 504 return result; 505} 506 507const char *sourceToString(audio_source_t source) 508{ 509 switch (source) { 510 case AUDIO_SOURCE_DEFAULT: return "default"; 511 case AUDIO_SOURCE_MIC: return "mic"; 512 case AUDIO_SOURCE_VOICE_UPLINK: return "voice uplink"; 513 case AUDIO_SOURCE_VOICE_DOWNLINK: return "voice downlink"; 514 case AUDIO_SOURCE_VOICE_CALL: return "voice call"; 515 case AUDIO_SOURCE_CAMCORDER: return "camcorder"; 516 case AUDIO_SOURCE_VOICE_RECOGNITION: return "voice recognition"; 517 case AUDIO_SOURCE_VOICE_COMMUNICATION: return "voice communication"; 518 case AUDIO_SOURCE_REMOTE_SUBMIX: return "remote submix"; 519 case AUDIO_SOURCE_FM_TUNER: return "FM tuner"; 520 case AUDIO_SOURCE_HOTWORD: return "hotword"; 521 default: return "unknown"; 522 } 523} 524 525AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, 526 audio_devices_t outDevice, audio_devices_t inDevice, type_t type, bool systemReady) 527 : Thread(false /*canCallJava*/), 528 mType(type), 529 mAudioFlinger(audioFlinger), 530 // mSampleRate, mFrameCount, mChannelMask, mChannelCount, mFrameSize, mFormat, mBufferSize 531 // are set by PlaybackThread::readOutputParameters_l() or 532 // RecordThread::readInputParameters_l() 533 //FIXME: mStandby should be true here. Is this some kind of hack? 534 mStandby(false), mOutDevice(outDevice), mInDevice(inDevice), 535 mPrevOutDevice(AUDIO_DEVICE_NONE), mPrevInDevice(AUDIO_DEVICE_NONE), 536 mAudioSource(AUDIO_SOURCE_DEFAULT), mId(id), 537 // mName will be set by concrete (non-virtual) subclass 538 mDeathRecipient(new PMDeathRecipient(this)), 539 mSystemReady(systemReady) 540{ 541 memset(&mPatch, 0, sizeof(struct audio_patch)); 542} 543 544AudioFlinger::ThreadBase::~ThreadBase() 545{ 546 // mConfigEvents should be empty, but just in case it isn't, free the memory it owns 547 mConfigEvents.clear(); 548 549 // do not lock the mutex in destructor 550 releaseWakeLock_l(); 551 if (mPowerManager != 0) { 552 sp<IBinder> binder = IInterface::asBinder(mPowerManager); 553 binder->unlinkToDeath(mDeathRecipient); 554 } 555} 556 557status_t AudioFlinger::ThreadBase::readyToRun() 558{ 559 status_t status = initCheck(); 560 if (status == NO_ERROR) { 561 ALOGI("AudioFlinger's thread %p ready to run", this); 562 } else { 563 ALOGE("No working audio driver found."); 564 } 565 return status; 566} 567 568void AudioFlinger::ThreadBase::exit() 569{ 570 ALOGV("ThreadBase::exit"); 571 // do any cleanup required for exit to succeed 572 preExit(); 573 { 574 // This lock prevents the following race in thread (uniprocessor for illustration): 575 // if (!exitPending()) { 576 // // context switch from here to exit() 577 // // exit() calls requestExit(), what exitPending() observes 578 // // exit() calls signal(), which is dropped since no waiters 579 // // context switch back from exit() to here 580 // mWaitWorkCV.wait(...); 581 // // now thread is hung 582 // } 583 AutoMutex lock(mLock); 584 requestExit(); 585 mWaitWorkCV.broadcast(); 586 } 587 // When Thread::requestExitAndWait is made virtual and this method is renamed to 588 // "virtual status_t requestExitAndWait()", replace by "return Thread::requestExitAndWait();" 589 requestExitAndWait(); 590} 591 592status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs) 593{ 594 status_t status; 595 596 ALOGV("ThreadBase::setParameters() %s", keyValuePairs.string()); 597 Mutex::Autolock _l(mLock); 598 599 return sendSetParameterConfigEvent_l(keyValuePairs); 600} 601 602// sendConfigEvent_l() must be called with ThreadBase::mLock held 603// Can temporarily release the lock if waiting for a reply from processConfigEvents_l(). 604status_t AudioFlinger::ThreadBase::sendConfigEvent_l(sp<ConfigEvent>& event) 605{ 606 status_t status = NO_ERROR; 607 608 if (event->mRequiresSystemReady && !mSystemReady) { 609 event->mWaitStatus = false; 610 mPendingConfigEvents.add(event); 611 return status; 612 } 613 mConfigEvents.add(event); 614 ALOGV("sendConfigEvent_l() num events %d event %d", mConfigEvents.size(), event->mType); 615 mWaitWorkCV.signal(); 616 mLock.unlock(); 617 { 618 Mutex::Autolock _l(event->mLock); 619 while (event->mWaitStatus) { 620 if (event->mCond.waitRelative(event->mLock, kConfigEventTimeoutNs) != NO_ERROR) { 621 event->mStatus = TIMED_OUT; 622 event->mWaitStatus = false; 623 } 624 } 625 status = event->mStatus; 626 } 627 mLock.lock(); 628 return status; 629} 630 631void AudioFlinger::ThreadBase::sendIoConfigEvent(audio_io_config_event event, pid_t pid) 632{ 633 Mutex::Autolock _l(mLock); 634 sendIoConfigEvent_l(event, pid); 635} 636 637// sendIoConfigEvent_l() must be called with ThreadBase::mLock held 638void AudioFlinger::ThreadBase::sendIoConfigEvent_l(audio_io_config_event event, pid_t pid) 639{ 640 sp<ConfigEvent> configEvent = (ConfigEvent *)new IoConfigEvent(event, pid); 641 sendConfigEvent_l(configEvent); 642} 643 644void AudioFlinger::ThreadBase::sendPrioConfigEvent(pid_t pid, pid_t tid, int32_t prio) 645{ 646 Mutex::Autolock _l(mLock); 647 sendPrioConfigEvent_l(pid, tid, prio); 648} 649 650// sendPrioConfigEvent_l() must be called with ThreadBase::mLock held 651void AudioFlinger::ThreadBase::sendPrioConfigEvent_l(pid_t pid, pid_t tid, int32_t prio) 652{ 653 sp<ConfigEvent> configEvent = (ConfigEvent *)new PrioConfigEvent(pid, tid, prio); 654 sendConfigEvent_l(configEvent); 655} 656 657// sendSetParameterConfigEvent_l() must be called with ThreadBase::mLock held 658status_t AudioFlinger::ThreadBase::sendSetParameterConfigEvent_l(const String8& keyValuePair) 659{ 660 sp<ConfigEvent> configEvent = (ConfigEvent *)new SetParameterConfigEvent(keyValuePair); 661 return sendConfigEvent_l(configEvent); 662} 663 664status_t AudioFlinger::ThreadBase::sendCreateAudioPatchConfigEvent( 665 const struct audio_patch *patch, 666 audio_patch_handle_t *handle) 667{ 668 Mutex::Autolock _l(mLock); 669 sp<ConfigEvent> configEvent = (ConfigEvent *)new CreateAudioPatchConfigEvent(*patch, *handle); 670 status_t status = sendConfigEvent_l(configEvent); 671 if (status == NO_ERROR) { 672 CreateAudioPatchConfigEventData *data = 673 (CreateAudioPatchConfigEventData *)configEvent->mData.get(); 674 *handle = data->mHandle; 675 } 676 return status; 677} 678 679status_t AudioFlinger::ThreadBase::sendReleaseAudioPatchConfigEvent( 680 const audio_patch_handle_t handle) 681{ 682 Mutex::Autolock _l(mLock); 683 sp<ConfigEvent> configEvent = (ConfigEvent *)new ReleaseAudioPatchConfigEvent(handle); 684 return sendConfigEvent_l(configEvent); 685} 686 687 688// post condition: mConfigEvents.isEmpty() 689void AudioFlinger::ThreadBase::processConfigEvents_l() 690{ 691 bool configChanged = false; 692 693 while (!mConfigEvents.isEmpty()) { 694 ALOGV("processConfigEvents_l() remaining events %d", mConfigEvents.size()); 695 sp<ConfigEvent> event = mConfigEvents[0]; 696 mConfigEvents.removeAt(0); 697 switch (event->mType) { 698 case CFG_EVENT_PRIO: { 699 PrioConfigEventData *data = (PrioConfigEventData *)event->mData.get(); 700 // FIXME Need to understand why this has to be done asynchronously 701 int err = requestPriority(data->mPid, data->mTid, data->mPrio, 702 true /*asynchronous*/); 703 if (err != 0) { 704 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 705 data->mPrio, data->mPid, data->mTid, err); 706 } 707 } break; 708 case CFG_EVENT_IO: { 709 IoConfigEventData *data = (IoConfigEventData *)event->mData.get(); 710 ioConfigChanged(data->mEvent, data->mPid); 711 } break; 712 case CFG_EVENT_SET_PARAMETER: { 713 SetParameterConfigEventData *data = (SetParameterConfigEventData *)event->mData.get(); 714 if (checkForNewParameter_l(data->mKeyValuePairs, event->mStatus)) { 715 configChanged = true; 716 } 717 } break; 718 case CFG_EVENT_CREATE_AUDIO_PATCH: { 719 CreateAudioPatchConfigEventData *data = 720 (CreateAudioPatchConfigEventData *)event->mData.get(); 721 event->mStatus = createAudioPatch_l(&data->mPatch, &data->mHandle); 722 } break; 723 case CFG_EVENT_RELEASE_AUDIO_PATCH: { 724 ReleaseAudioPatchConfigEventData *data = 725 (ReleaseAudioPatchConfigEventData *)event->mData.get(); 726 event->mStatus = releaseAudioPatch_l(data->mHandle); 727 } break; 728 default: 729 ALOG_ASSERT(false, "processConfigEvents_l() unknown event type %d", event->mType); 730 break; 731 } 732 { 733 Mutex::Autolock _l(event->mLock); 734 if (event->mWaitStatus) { 735 event->mWaitStatus = false; 736 event->mCond.signal(); 737 } 738 } 739 ALOGV_IF(mConfigEvents.isEmpty(), "processConfigEvents_l() DONE thread %p", this); 740 } 741 742 if (configChanged) { 743 cacheParameters_l(); 744 } 745} 746 747String8 channelMaskToString(audio_channel_mask_t mask, bool output) { 748 String8 s; 749 const audio_channel_representation_t representation = 750 audio_channel_mask_get_representation(mask); 751 752 switch (representation) { 753 case AUDIO_CHANNEL_REPRESENTATION_POSITION: { 754 if (output) { 755 if (mask & AUDIO_CHANNEL_OUT_FRONT_LEFT) s.append("front-left, "); 756 if (mask & AUDIO_CHANNEL_OUT_FRONT_RIGHT) s.append("front-right, "); 757 if (mask & AUDIO_CHANNEL_OUT_FRONT_CENTER) s.append("front-center, "); 758 if (mask & AUDIO_CHANNEL_OUT_LOW_FREQUENCY) s.append("low freq, "); 759 if (mask & AUDIO_CHANNEL_OUT_BACK_LEFT) s.append("back-left, "); 760 if (mask & AUDIO_CHANNEL_OUT_BACK_RIGHT) s.append("back-right, "); 761 if (mask & AUDIO_CHANNEL_OUT_FRONT_LEFT_OF_CENTER) s.append("front-left-of-center, "); 762 if (mask & AUDIO_CHANNEL_OUT_FRONT_RIGHT_OF_CENTER) s.append("front-right-of-center, "); 763 if (mask & AUDIO_CHANNEL_OUT_BACK_CENTER) s.append("back-center, "); 764 if (mask & AUDIO_CHANNEL_OUT_SIDE_LEFT) s.append("side-left, "); 765 if (mask & AUDIO_CHANNEL_OUT_SIDE_RIGHT) s.append("side-right, "); 766 if (mask & AUDIO_CHANNEL_OUT_TOP_CENTER) s.append("top-center ,"); 767 if (mask & AUDIO_CHANNEL_OUT_TOP_FRONT_LEFT) s.append("top-front-left, "); 768 if (mask & AUDIO_CHANNEL_OUT_TOP_FRONT_CENTER) s.append("top-front-center, "); 769 if (mask & AUDIO_CHANNEL_OUT_TOP_FRONT_RIGHT) s.append("top-front-right, "); 770 if (mask & AUDIO_CHANNEL_OUT_TOP_BACK_LEFT) s.append("top-back-left, "); 771 if (mask & AUDIO_CHANNEL_OUT_TOP_BACK_CENTER) s.append("top-back-center, " ); 772 if (mask & AUDIO_CHANNEL_OUT_TOP_BACK_RIGHT) s.append("top-back-right, " ); 773 if (mask & ~AUDIO_CHANNEL_OUT_ALL) s.append("unknown, "); 774 } else { 775 if (mask & AUDIO_CHANNEL_IN_LEFT) s.append("left, "); 776 if (mask & AUDIO_CHANNEL_IN_RIGHT) s.append("right, "); 777 if (mask & AUDIO_CHANNEL_IN_FRONT) s.append("front, "); 778 if (mask & AUDIO_CHANNEL_IN_BACK) s.append("back, "); 779 if (mask & AUDIO_CHANNEL_IN_LEFT_PROCESSED) s.append("left-processed, "); 780 if (mask & AUDIO_CHANNEL_IN_RIGHT_PROCESSED) s.append("right-processed, "); 781 if (mask & AUDIO_CHANNEL_IN_FRONT_PROCESSED) s.append("front-processed, "); 782 if (mask & AUDIO_CHANNEL_IN_BACK_PROCESSED) s.append("back-processed, "); 783 if (mask & AUDIO_CHANNEL_IN_PRESSURE) s.append("pressure, "); 784 if (mask & AUDIO_CHANNEL_IN_X_AXIS) s.append("X, "); 785 if (mask & AUDIO_CHANNEL_IN_Y_AXIS) s.append("Y, "); 786 if (mask & AUDIO_CHANNEL_IN_Z_AXIS) s.append("Z, "); 787 if (mask & AUDIO_CHANNEL_IN_VOICE_UPLINK) s.append("voice-uplink, "); 788 if (mask & AUDIO_CHANNEL_IN_VOICE_DNLINK) s.append("voice-dnlink, "); 789 if (mask & ~AUDIO_CHANNEL_IN_ALL) s.append("unknown, "); 790 } 791 const int len = s.length(); 792 if (len > 2) { 793 char *str = s.lockBuffer(len); // needed? 794 s.unlockBuffer(len - 2); // remove trailing ", " 795 } 796 return s; 797 } 798 case AUDIO_CHANNEL_REPRESENTATION_INDEX: 799 s.appendFormat("index mask, bits:%#x", audio_channel_mask_get_bits(mask)); 800 return s; 801 default: 802 s.appendFormat("unknown mask, representation:%d bits:%#x", 803 representation, audio_channel_mask_get_bits(mask)); 804 return s; 805 } 806} 807 808void AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args __unused) 809{ 810 const size_t SIZE = 256; 811 char buffer[SIZE]; 812 String8 result; 813 814 bool locked = AudioFlinger::dumpTryLock(mLock); 815 if (!locked) { 816 dprintf(fd, "thread %p may be deadlocked\n", this); 817 } 818 819 dprintf(fd, " Thread name: %s\n", mThreadName); 820 dprintf(fd, " I/O handle: %d\n", mId); 821 dprintf(fd, " TID: %d\n", getTid()); 822 dprintf(fd, " Standby: %s\n", mStandby ? "yes" : "no"); 823 dprintf(fd, " Sample rate: %u Hz\n", mSampleRate); 824 dprintf(fd, " HAL frame count: %zu\n", mFrameCount); 825 dprintf(fd, " HAL format: 0x%x (%s)\n", mHALFormat, formatToString(mHALFormat)); 826 dprintf(fd, " HAL buffer size: %u bytes\n", mBufferSize); 827 dprintf(fd, " Channel count: %u\n", mChannelCount); 828 dprintf(fd, " Channel mask: 0x%08x (%s)\n", mChannelMask, 829 channelMaskToString(mChannelMask, mType != RECORD).string()); 830 dprintf(fd, " Format: 0x%x (%s)\n", mFormat, formatToString(mFormat)); 831 dprintf(fd, " Frame size: %zu bytes\n", mFrameSize); 832 dprintf(fd, " Pending config events:"); 833 size_t numConfig = mConfigEvents.size(); 834 if (numConfig) { 835 for (size_t i = 0; i < numConfig; i++) { 836 mConfigEvents[i]->dump(buffer, SIZE); 837 dprintf(fd, "\n %s", buffer); 838 } 839 dprintf(fd, "\n"); 840 } else { 841 dprintf(fd, " none\n"); 842 } 843 dprintf(fd, " Output device: %#x (%s)\n", mOutDevice, devicesToString(mOutDevice).string()); 844 dprintf(fd, " Input device: %#x (%s)\n", mInDevice, devicesToString(mInDevice).string()); 845 dprintf(fd, " Audio source: %d (%s)\n", mAudioSource, sourceToString(mAudioSource)); 846 847 if (locked) { 848 mLock.unlock(); 849 } 850} 851 852void AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args) 853{ 854 const size_t SIZE = 256; 855 char buffer[SIZE]; 856 String8 result; 857 858 size_t numEffectChains = mEffectChains.size(); 859 snprintf(buffer, SIZE, " %zu Effect Chains\n", numEffectChains); 860 write(fd, buffer, strlen(buffer)); 861 862 for (size_t i = 0; i < numEffectChains; ++i) { 863 sp<EffectChain> chain = mEffectChains[i]; 864 if (chain != 0) { 865 chain->dump(fd, args); 866 } 867 } 868} 869 870void AudioFlinger::ThreadBase::acquireWakeLock(int uid) 871{ 872 Mutex::Autolock _l(mLock); 873 acquireWakeLock_l(uid); 874} 875 876String16 AudioFlinger::ThreadBase::getWakeLockTag() 877{ 878 switch (mType) { 879 case MIXER: 880 return String16("AudioMix"); 881 case DIRECT: 882 return String16("AudioDirectOut"); 883 case DUPLICATING: 884 return String16("AudioDup"); 885 case RECORD: 886 return String16("AudioIn"); 887 case OFFLOAD: 888 return String16("AudioOffload"); 889 default: 890 ALOG_ASSERT(false); 891 return String16("AudioUnknown"); 892 } 893} 894 895void AudioFlinger::ThreadBase::acquireWakeLock_l(int uid) 896{ 897 getPowerManager_l(); 898 if (mPowerManager != 0) { 899 sp<IBinder> binder = new BBinder(); 900 status_t status; 901 if (uid >= 0) { 902 status = mPowerManager->acquireWakeLockWithUid(POWERMANAGER_PARTIAL_WAKE_LOCK, 903 binder, 904 getWakeLockTag(), 905 String16("media"), 906 uid, 907 true /* FIXME force oneway contrary to .aidl */); 908 } else { 909 status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK, 910 binder, 911 getWakeLockTag(), 912 String16("media"), 913 true /* FIXME force oneway contrary to .aidl */); 914 } 915 if (status == NO_ERROR) { 916 mWakeLockToken = binder; 917 } 918 ALOGV("acquireWakeLock_l() %s status %d", mThreadName, status); 919 } 920} 921 922void AudioFlinger::ThreadBase::releaseWakeLock() 923{ 924 Mutex::Autolock _l(mLock); 925 releaseWakeLock_l(); 926} 927 928void AudioFlinger::ThreadBase::releaseWakeLock_l() 929{ 930 if (mWakeLockToken != 0) { 931 ALOGV("releaseWakeLock_l() %s", mThreadName); 932 if (mPowerManager != 0) { 933 mPowerManager->releaseWakeLock(mWakeLockToken, 0, 934 true /* FIXME force oneway contrary to .aidl */); 935 } 936 mWakeLockToken.clear(); 937 } 938} 939 940void AudioFlinger::ThreadBase::updateWakeLockUids(const SortedVector<int> &uids) { 941 Mutex::Autolock _l(mLock); 942 updateWakeLockUids_l(uids); 943} 944 945void AudioFlinger::ThreadBase::getPowerManager_l() { 946 if (mSystemReady && mPowerManager == 0) { 947 // use checkService() to avoid blocking if power service is not up yet 948 sp<IBinder> binder = 949 defaultServiceManager()->checkService(String16("power")); 950 if (binder == 0) { 951 ALOGW("Thread %s cannot connect to the power manager service", mThreadName); 952 } else { 953 mPowerManager = interface_cast<IPowerManager>(binder); 954 binder->linkToDeath(mDeathRecipient); 955 } 956 } 957} 958 959void AudioFlinger::ThreadBase::updateWakeLockUids_l(const SortedVector<int> &uids) { 960 getPowerManager_l(); 961 if (mWakeLockToken == NULL) { 962 ALOGE("no wake lock to update!"); 963 return; 964 } 965 if (mPowerManager != 0) { 966 sp<IBinder> binder = new BBinder(); 967 status_t status; 968 status = mPowerManager->updateWakeLockUids(mWakeLockToken, uids.size(), uids.array(), 969 true /* FIXME force oneway contrary to .aidl */); 970 ALOGV("acquireWakeLock_l() %s status %d", mThreadName, status); 971 } 972} 973 974void AudioFlinger::ThreadBase::clearPowerManager() 975{ 976 Mutex::Autolock _l(mLock); 977 releaseWakeLock_l(); 978 mPowerManager.clear(); 979} 980 981void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who __unused) 982{ 983 sp<ThreadBase> thread = mThread.promote(); 984 if (thread != 0) { 985 thread->clearPowerManager(); 986 } 987 ALOGW("power manager service died !!!"); 988} 989 990void AudioFlinger::ThreadBase::setEffectSuspended( 991 const effect_uuid_t *type, bool suspend, int sessionId) 992{ 993 Mutex::Autolock _l(mLock); 994 setEffectSuspended_l(type, suspend, sessionId); 995} 996 997void AudioFlinger::ThreadBase::setEffectSuspended_l( 998 const effect_uuid_t *type, bool suspend, int sessionId) 999{ 1000 sp<EffectChain> chain = getEffectChain_l(sessionId); 1001 if (chain != 0) { 1002 if (type != NULL) { 1003 chain->setEffectSuspended_l(type, suspend); 1004 } else { 1005 chain->setEffectSuspendedAll_l(suspend); 1006 } 1007 } 1008 1009 updateSuspendedSessions_l(type, suspend, sessionId); 1010} 1011 1012void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain) 1013{ 1014 ssize_t index = mSuspendedSessions.indexOfKey(chain->sessionId()); 1015 if (index < 0) { 1016 return; 1017 } 1018 1019 const KeyedVector <int, sp<SuspendedSessionDesc> >& sessionEffects = 1020 mSuspendedSessions.valueAt(index); 1021 1022 for (size_t i = 0; i < sessionEffects.size(); i++) { 1023 sp<SuspendedSessionDesc> desc = sessionEffects.valueAt(i); 1024 for (int j = 0; j < desc->mRefCount; j++) { 1025 if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) { 1026 chain->setEffectSuspendedAll_l(true); 1027 } else { 1028 ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x", 1029 desc->mType.timeLow); 1030 chain->setEffectSuspended_l(&desc->mType, true); 1031 } 1032 } 1033 } 1034} 1035 1036void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type, 1037 bool suspend, 1038 int sessionId) 1039{ 1040 ssize_t index = mSuspendedSessions.indexOfKey(sessionId); 1041 1042 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects; 1043 1044 if (suspend) { 1045 if (index >= 0) { 1046 sessionEffects = mSuspendedSessions.valueAt(index); 1047 } else { 1048 mSuspendedSessions.add(sessionId, sessionEffects); 1049 } 1050 } else { 1051 if (index < 0) { 1052 return; 1053 } 1054 sessionEffects = mSuspendedSessions.valueAt(index); 1055 } 1056 1057 1058 int key = EffectChain::kKeyForSuspendAll; 1059 if (type != NULL) { 1060 key = type->timeLow; 1061 } 1062 index = sessionEffects.indexOfKey(key); 1063 1064 sp<SuspendedSessionDesc> desc; 1065 if (suspend) { 1066 if (index >= 0) { 1067 desc = sessionEffects.valueAt(index); 1068 } else { 1069 desc = new SuspendedSessionDesc(); 1070 if (type != NULL) { 1071 desc->mType = *type; 1072 } 1073 sessionEffects.add(key, desc); 1074 ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key); 1075 } 1076 desc->mRefCount++; 1077 } else { 1078 if (index < 0) { 1079 return; 1080 } 1081 desc = sessionEffects.valueAt(index); 1082 if (--desc->mRefCount == 0) { 1083 ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key); 1084 sessionEffects.removeItemsAt(index); 1085 if (sessionEffects.isEmpty()) { 1086 ALOGV("updateSuspendedSessions_l() restore removing session %d", 1087 sessionId); 1088 mSuspendedSessions.removeItem(sessionId); 1089 } 1090 } 1091 } 1092 if (!sessionEffects.isEmpty()) { 1093 mSuspendedSessions.replaceValueFor(sessionId, sessionEffects); 1094 } 1095} 1096 1097void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 1098 bool enabled, 1099 int sessionId) 1100{ 1101 Mutex::Autolock _l(mLock); 1102 checkSuspendOnEffectEnabled_l(effect, enabled, sessionId); 1103} 1104 1105void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 1106 bool enabled, 1107 int sessionId) 1108{ 1109 if (mType != RECORD) { 1110 // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on 1111 // another session. This gives the priority to well behaved effect control panels 1112 // and applications not using global effects. 1113 // Enabling post processing in AUDIO_SESSION_OUTPUT_STAGE session does not affect 1114 // global effects 1115 if ((sessionId != AUDIO_SESSION_OUTPUT_MIX) && (sessionId != AUDIO_SESSION_OUTPUT_STAGE)) { 1116 setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX); 1117 } 1118 } 1119 1120 sp<EffectChain> chain = getEffectChain_l(sessionId); 1121 if (chain != 0) { 1122 chain->checkSuspendOnEffectEnabled(effect, enabled); 1123 } 1124} 1125 1126// ThreadBase::createEffect_l() must be called with AudioFlinger::mLock held 1127sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 1128 const sp<AudioFlinger::Client>& client, 1129 const sp<IEffectClient>& effectClient, 1130 int32_t priority, 1131 int sessionId, 1132 effect_descriptor_t *desc, 1133 int *enabled, 1134 status_t *status) 1135{ 1136 sp<EffectModule> effect; 1137 sp<EffectHandle> handle; 1138 status_t lStatus; 1139 sp<EffectChain> chain; 1140 bool chainCreated = false; 1141 bool effectCreated = false; 1142 bool effectRegistered = false; 1143 1144 lStatus = initCheck(); 1145 if (lStatus != NO_ERROR) { 1146 ALOGW("createEffect_l() Audio driver not initialized."); 1147 goto Exit; 1148 } 1149 1150 // Reject any effect on Direct output threads for now, since the format of 1151 // mSinkBuffer is not guaranteed to be compatible with effect processing (PCM 16 stereo). 1152 if (mType == DIRECT) { 1153 ALOGW("createEffect_l() Cannot add effect %s on Direct output type thread %s", 1154 desc->name, mThreadName); 1155 lStatus = BAD_VALUE; 1156 goto Exit; 1157 } 1158 1159 // Reject any effect on mixer or duplicating multichannel sinks. 1160 // TODO: fix both format and multichannel issues with effects. 1161 if ((mType == MIXER || mType == DUPLICATING) && mChannelCount != FCC_2) { 1162 ALOGW("createEffect_l() Cannot add effect %s for multichannel(%d) %s threads", 1163 desc->name, mChannelCount, mType == MIXER ? "MIXER" : "DUPLICATING"); 1164 lStatus = BAD_VALUE; 1165 goto Exit; 1166 } 1167 1168 // Allow global effects only on offloaded and mixer threads 1169 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 1170 switch (mType) { 1171 case MIXER: 1172 case OFFLOAD: 1173 break; 1174 case DIRECT: 1175 case DUPLICATING: 1176 case RECORD: 1177 default: 1178 ALOGW("createEffect_l() Cannot add global effect %s on thread %s", 1179 desc->name, mThreadName); 1180 lStatus = BAD_VALUE; 1181 goto Exit; 1182 } 1183 } 1184 1185 // Only Pre processor effects are allowed on input threads and only on input threads 1186 if ((mType == RECORD) != ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { 1187 ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d", 1188 desc->name, desc->flags, mType); 1189 lStatus = BAD_VALUE; 1190 goto Exit; 1191 } 1192 1193 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 1194 1195 { // scope for mLock 1196 Mutex::Autolock _l(mLock); 1197 1198 // check for existing effect chain with the requested audio session 1199 chain = getEffectChain_l(sessionId); 1200 if (chain == 0) { 1201 // create a new chain for this session 1202 ALOGV("createEffect_l() new effect chain for session %d", sessionId); 1203 chain = new EffectChain(this, sessionId); 1204 addEffectChain_l(chain); 1205 chain->setStrategy(getStrategyForSession_l(sessionId)); 1206 chainCreated = true; 1207 } else { 1208 effect = chain->getEffectFromDesc_l(desc); 1209 } 1210 1211 ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get()); 1212 1213 if (effect == 0) { 1214 int id = mAudioFlinger->nextUniqueId(); 1215 // Check CPU and memory usage 1216 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 1217 if (lStatus != NO_ERROR) { 1218 goto Exit; 1219 } 1220 effectRegistered = true; 1221 // create a new effect module if none present in the chain 1222 effect = new EffectModule(this, chain, desc, id, sessionId); 1223 lStatus = effect->status(); 1224 if (lStatus != NO_ERROR) { 1225 goto Exit; 1226 } 1227 effect->setOffloaded(mType == OFFLOAD, mId); 1228 1229 lStatus = chain->addEffect_l(effect); 1230 if (lStatus != NO_ERROR) { 1231 goto Exit; 1232 } 1233 effectCreated = true; 1234 1235 effect->setDevice(mOutDevice); 1236 effect->setDevice(mInDevice); 1237 effect->setMode(mAudioFlinger->getMode()); 1238 effect->setAudioSource(mAudioSource); 1239 } 1240 // create effect handle and connect it to effect module 1241 handle = new EffectHandle(effect, client, effectClient, priority); 1242 lStatus = handle->initCheck(); 1243 if (lStatus == OK) { 1244 lStatus = effect->addHandle(handle.get()); 1245 } 1246 if (enabled != NULL) { 1247 *enabled = (int)effect->isEnabled(); 1248 } 1249 } 1250 1251Exit: 1252 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 1253 Mutex::Autolock _l(mLock); 1254 if (effectCreated) { 1255 chain->removeEffect_l(effect); 1256 } 1257 if (effectRegistered) { 1258 AudioSystem::unregisterEffect(effect->id()); 1259 } 1260 if (chainCreated) { 1261 removeEffectChain_l(chain); 1262 } 1263 handle.clear(); 1264 } 1265 1266 *status = lStatus; 1267 return handle; 1268} 1269 1270sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect(int sessionId, int effectId) 1271{ 1272 Mutex::Autolock _l(mLock); 1273 return getEffect_l(sessionId, effectId); 1274} 1275 1276sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId) 1277{ 1278 sp<EffectChain> chain = getEffectChain_l(sessionId); 1279 return chain != 0 ? chain->getEffectFromId_l(effectId) : 0; 1280} 1281 1282// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 1283// PlaybackThread::mLock held 1284status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 1285{ 1286 // check for existing effect chain with the requested audio session 1287 int sessionId = effect->sessionId(); 1288 sp<EffectChain> chain = getEffectChain_l(sessionId); 1289 bool chainCreated = false; 1290 1291 ALOGD_IF((mType == OFFLOAD) && !effect->isOffloadable(), 1292 "addEffect_l() on offloaded thread %p: effect %s does not support offload flags %x", 1293 this, effect->desc().name, effect->desc().flags); 1294 1295 if (chain == 0) { 1296 // create a new chain for this session 1297 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 1298 chain = new EffectChain(this, sessionId); 1299 addEffectChain_l(chain); 1300 chain->setStrategy(getStrategyForSession_l(sessionId)); 1301 chainCreated = true; 1302 } 1303 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 1304 1305 if (chain->getEffectFromId_l(effect->id()) != 0) { 1306 ALOGW("addEffect_l() %p effect %s already present in chain %p", 1307 this, effect->desc().name, chain.get()); 1308 return BAD_VALUE; 1309 } 1310 1311 effect->setOffloaded(mType == OFFLOAD, mId); 1312 1313 status_t status = chain->addEffect_l(effect); 1314 if (status != NO_ERROR) { 1315 if (chainCreated) { 1316 removeEffectChain_l(chain); 1317 } 1318 return status; 1319 } 1320 1321 effect->setDevice(mOutDevice); 1322 effect->setDevice(mInDevice); 1323 effect->setMode(mAudioFlinger->getMode()); 1324 effect->setAudioSource(mAudioSource); 1325 return NO_ERROR; 1326} 1327 1328void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 1329 1330 ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 1331 effect_descriptor_t desc = effect->desc(); 1332 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 1333 detachAuxEffect_l(effect->id()); 1334 } 1335 1336 sp<EffectChain> chain = effect->chain().promote(); 1337 if (chain != 0) { 1338 // remove effect chain if removing last effect 1339 if (chain->removeEffect_l(effect) == 0) { 1340 removeEffectChain_l(chain); 1341 } 1342 } else { 1343 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 1344 } 1345} 1346 1347void AudioFlinger::ThreadBase::lockEffectChains_l( 1348 Vector< sp<AudioFlinger::EffectChain> >& effectChains) 1349{ 1350 effectChains = mEffectChains; 1351 for (size_t i = 0; i < mEffectChains.size(); i++) { 1352 mEffectChains[i]->lock(); 1353 } 1354} 1355 1356void AudioFlinger::ThreadBase::unlockEffectChains( 1357 const Vector< sp<AudioFlinger::EffectChain> >& effectChains) 1358{ 1359 for (size_t i = 0; i < effectChains.size(); i++) { 1360 effectChains[i]->unlock(); 1361 } 1362} 1363 1364sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId) 1365{ 1366 Mutex::Autolock _l(mLock); 1367 return getEffectChain_l(sessionId); 1368} 1369 1370sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) const 1371{ 1372 size_t size = mEffectChains.size(); 1373 for (size_t i = 0; i < size; i++) { 1374 if (mEffectChains[i]->sessionId() == sessionId) { 1375 return mEffectChains[i]; 1376 } 1377 } 1378 return 0; 1379} 1380 1381void AudioFlinger::ThreadBase::setMode(audio_mode_t mode) 1382{ 1383 Mutex::Autolock _l(mLock); 1384 size_t size = mEffectChains.size(); 1385 for (size_t i = 0; i < size; i++) { 1386 mEffectChains[i]->setMode_l(mode); 1387 } 1388} 1389 1390void AudioFlinger::ThreadBase::getAudioPortConfig(struct audio_port_config *config) 1391{ 1392 config->type = AUDIO_PORT_TYPE_MIX; 1393 config->ext.mix.handle = mId; 1394 config->sample_rate = mSampleRate; 1395 config->format = mFormat; 1396 config->channel_mask = mChannelMask; 1397 config->config_mask = AUDIO_PORT_CONFIG_SAMPLE_RATE|AUDIO_PORT_CONFIG_CHANNEL_MASK| 1398 AUDIO_PORT_CONFIG_FORMAT; 1399} 1400 1401void AudioFlinger::ThreadBase::systemReady() 1402{ 1403 Mutex::Autolock _l(mLock); 1404 if (mSystemReady) { 1405 return; 1406 } 1407 mSystemReady = true; 1408 1409 for (size_t i = 0; i < mPendingConfigEvents.size(); i++) { 1410 sendConfigEvent_l(mPendingConfigEvents.editItemAt(i)); 1411 } 1412 mPendingConfigEvents.clear(); 1413} 1414 1415 1416// ---------------------------------------------------------------------------- 1417// Playback 1418// ---------------------------------------------------------------------------- 1419 1420AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, 1421 AudioStreamOut* output, 1422 audio_io_handle_t id, 1423 audio_devices_t device, 1424 type_t type, 1425 bool systemReady) 1426 : ThreadBase(audioFlinger, id, device, AUDIO_DEVICE_NONE, type, systemReady), 1427 mNormalFrameCount(0), mSinkBuffer(NULL), 1428 mMixerBufferEnabled(AudioFlinger::kEnableExtendedPrecision), 1429 mMixerBuffer(NULL), 1430 mMixerBufferSize(0), 1431 mMixerBufferFormat(AUDIO_FORMAT_INVALID), 1432 mMixerBufferValid(false), 1433 mEffectBufferEnabled(AudioFlinger::kEnableExtendedPrecision), 1434 mEffectBuffer(NULL), 1435 mEffectBufferSize(0), 1436 mEffectBufferFormat(AUDIO_FORMAT_INVALID), 1437 mEffectBufferValid(false), 1438 mSuspended(0), mBytesWritten(0), 1439 mActiveTracksGeneration(0), 1440 // mStreamTypes[] initialized in constructor body 1441 mOutput(output), 1442 mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false), 1443 mMixerStatus(MIXER_IDLE), 1444 mMixerStatusIgnoringFastTracks(MIXER_IDLE), 1445 mStandbyDelayNs(AudioFlinger::mStandbyTimeInNsecs), 1446 mBytesRemaining(0), 1447 mCurrentWriteLength(0), 1448 mUseAsyncWrite(false), 1449 mWriteAckSequence(0), 1450 mDrainSequence(0), 1451 mSignalPending(false), 1452 mScreenState(AudioFlinger::mScreenState), 1453 // index 0 is reserved for normal mixer's submix 1454 mFastTrackAvailMask(((1 << FastMixerState::kMaxFastTracks) - 1) & ~1), 1455 mHwSupportsPause(false), mHwPaused(false), mFlushPending(false), 1456 // mLatchD, mLatchQ, 1457 mLatchDValid(false), mLatchQValid(false) 1458{ 1459 snprintf(mThreadName, kThreadNameLength, "AudioOut_%X", id); 1460 mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mThreadName); 1461 1462 // Assumes constructor is called by AudioFlinger with it's mLock held, but 1463 // it would be safer to explicitly pass initial masterVolume/masterMute as 1464 // parameter. 1465 // 1466 // If the HAL we are using has support for master volume or master mute, 1467 // then do not attenuate or mute during mixing (just leave the volume at 1.0 1468 // and the mute set to false). 1469 mMasterVolume = audioFlinger->masterVolume_l(); 1470 mMasterMute = audioFlinger->masterMute_l(); 1471 if (mOutput && mOutput->audioHwDev) { 1472 if (mOutput->audioHwDev->canSetMasterVolume()) { 1473 mMasterVolume = 1.0; 1474 } 1475 1476 if (mOutput->audioHwDev->canSetMasterMute()) { 1477 mMasterMute = false; 1478 } 1479 } 1480 1481 readOutputParameters_l(); 1482 1483 // ++ operator does not compile 1484 for (audio_stream_type_t stream = AUDIO_STREAM_MIN; stream < AUDIO_STREAM_CNT; 1485 stream = (audio_stream_type_t) (stream + 1)) { 1486 mStreamTypes[stream].volume = mAudioFlinger->streamVolume_l(stream); 1487 mStreamTypes[stream].mute = mAudioFlinger->streamMute_l(stream); 1488 } 1489} 1490 1491AudioFlinger::PlaybackThread::~PlaybackThread() 1492{ 1493 mAudioFlinger->unregisterWriter(mNBLogWriter); 1494 free(mSinkBuffer); 1495 free(mMixerBuffer); 1496 free(mEffectBuffer); 1497} 1498 1499void AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) 1500{ 1501 dumpInternals(fd, args); 1502 dumpTracks(fd, args); 1503 dumpEffectChains(fd, args); 1504} 1505 1506void AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args __unused) 1507{ 1508 const size_t SIZE = 256; 1509 char buffer[SIZE]; 1510 String8 result; 1511 1512 result.appendFormat(" Stream volumes in dB: "); 1513 for (int i = 0; i < AUDIO_STREAM_CNT; ++i) { 1514 const stream_type_t *st = &mStreamTypes[i]; 1515 if (i > 0) { 1516 result.appendFormat(", "); 1517 } 1518 result.appendFormat("%d:%.2g", i, 20.0 * log10(st->volume)); 1519 if (st->mute) { 1520 result.append("M"); 1521 } 1522 } 1523 result.append("\n"); 1524 write(fd, result.string(), result.length()); 1525 result.clear(); 1526 1527 // These values are "raw"; they will wrap around. See prepareTracks_l() for a better way. 1528 FastTrackUnderruns underruns = getFastTrackUnderruns(0); 1529 dprintf(fd, " Normal mixer raw underrun counters: partial=%u empty=%u\n", 1530 underruns.mBitFields.mPartial, underruns.mBitFields.mEmpty); 1531 1532 size_t numtracks = mTracks.size(); 1533 size_t numactive = mActiveTracks.size(); 1534 dprintf(fd, " %d Tracks", numtracks); 1535 size_t numactiveseen = 0; 1536 if (numtracks) { 1537 dprintf(fd, " of which %d are active\n", numactive); 1538 Track::appendDumpHeader(result); 1539 for (size_t i = 0; i < numtracks; ++i) { 1540 sp<Track> track = mTracks[i]; 1541 if (track != 0) { 1542 bool active = mActiveTracks.indexOf(track) >= 0; 1543 if (active) { 1544 numactiveseen++; 1545 } 1546 track->dump(buffer, SIZE, active); 1547 result.append(buffer); 1548 } 1549 } 1550 } else { 1551 result.append("\n"); 1552 } 1553 if (numactiveseen != numactive) { 1554 // some tracks in the active list were not in the tracks list 1555 snprintf(buffer, SIZE, " The following tracks are in the active list but" 1556 " not in the track list\n"); 1557 result.append(buffer); 1558 Track::appendDumpHeader(result); 1559 for (size_t i = 0; i < numactive; ++i) { 1560 sp<Track> track = mActiveTracks[i].promote(); 1561 if (track != 0 && mTracks.indexOf(track) < 0) { 1562 track->dump(buffer, SIZE, true); 1563 result.append(buffer); 1564 } 1565 } 1566 } 1567 1568 write(fd, result.string(), result.size()); 1569} 1570 1571void AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 1572{ 1573 dprintf(fd, "\nOutput thread %p type %d (%s):\n", this, type(), threadTypeToString(type())); 1574 1575 dumpBase(fd, args); 1576 1577 dprintf(fd, " Normal frame count: %zu\n", mNormalFrameCount); 1578 dprintf(fd, " Last write occurred (msecs): %llu\n", ns2ms(systemTime() - mLastWriteTime)); 1579 dprintf(fd, " Total writes: %d\n", mNumWrites); 1580 dprintf(fd, " Delayed writes: %d\n", mNumDelayedWrites); 1581 dprintf(fd, " Blocked in write: %s\n", mInWrite ? "yes" : "no"); 1582 dprintf(fd, " Suspend count: %d\n", mSuspended); 1583 dprintf(fd, " Sink buffer : %p\n", mSinkBuffer); 1584 dprintf(fd, " Mixer buffer: %p\n", mMixerBuffer); 1585 dprintf(fd, " Effect buffer: %p\n", mEffectBuffer); 1586 dprintf(fd, " Fast track availMask=%#x\n", mFastTrackAvailMask); 1587 AudioStreamOut *output = mOutput; 1588 audio_output_flags_t flags = output != NULL ? output->flags : AUDIO_OUTPUT_FLAG_NONE; 1589 String8 flagsAsString = outputFlagsToString(flags); 1590 dprintf(fd, " AudioStreamOut: %p flags %#x (%s)\n", output, flags, flagsAsString.string()); 1591} 1592 1593// Thread virtuals 1594 1595void AudioFlinger::PlaybackThread::onFirstRef() 1596{ 1597 run(mThreadName, ANDROID_PRIORITY_URGENT_AUDIO); 1598} 1599 1600// ThreadBase virtuals 1601void AudioFlinger::PlaybackThread::preExit() 1602{ 1603 ALOGV(" preExit()"); 1604 // FIXME this is using hard-coded strings but in the future, this functionality will be 1605 // converted to use audio HAL extensions required to support tunneling 1606 mOutput->stream->common.set_parameters(&mOutput->stream->common, "exiting=1"); 1607} 1608 1609// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 1610sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 1611 const sp<AudioFlinger::Client>& client, 1612 audio_stream_type_t streamType, 1613 uint32_t sampleRate, 1614 audio_format_t format, 1615 audio_channel_mask_t channelMask, 1616 size_t *pFrameCount, 1617 const sp<IMemory>& sharedBuffer, 1618 int sessionId, 1619 IAudioFlinger::track_flags_t *flags, 1620 pid_t tid, 1621 int uid, 1622 status_t *status) 1623{ 1624 size_t frameCount = *pFrameCount; 1625 sp<Track> track; 1626 status_t lStatus; 1627 1628 bool isTimed = (*flags & IAudioFlinger::TRACK_TIMED) != 0; 1629 1630 // client expresses a preference for FAST, but we get the final say 1631 if (*flags & IAudioFlinger::TRACK_FAST) { 1632 if ( 1633 // not timed 1634 (!isTimed) && 1635 // either of these use cases: 1636 ( 1637 // use case 1: shared buffer with any frame count 1638 ( 1639 (sharedBuffer != 0) 1640 ) || 1641 // use case 2: frame count is default or at least as large as HAL 1642 ( 1643 // we formerly checked for a callback handler (non-0 tid), 1644 // but that is no longer required for TRANSFER_OBTAIN mode 1645 ((frameCount == 0) || 1646 (frameCount >= mFrameCount)) 1647 ) 1648 ) && 1649 // PCM data 1650 audio_is_linear_pcm(format) && 1651 // TODO: extract as a data library function that checks that a computationally 1652 // expensive downmixer is not required: isFastOutputChannelConversion() 1653 (channelMask == mChannelMask || 1654 mChannelMask != AUDIO_CHANNEL_OUT_STEREO || 1655 (channelMask == AUDIO_CHANNEL_OUT_MONO 1656 /* && mChannelMask == AUDIO_CHANNEL_OUT_STEREO */)) && 1657 // hardware sample rate 1658 (sampleRate == mSampleRate) && 1659 // normal mixer has an associated fast mixer 1660 hasFastMixer() && 1661 // there are sufficient fast track slots available 1662 (mFastTrackAvailMask != 0) 1663 // FIXME test that MixerThread for this fast track has a capable output HAL 1664 // FIXME add a permission test also? 1665 ) { 1666 // if frameCount not specified, then it defaults to fast mixer (HAL) frame count 1667 if (frameCount == 0) { 1668 // read the fast track multiplier property the first time it is needed 1669 int ok = pthread_once(&sFastTrackMultiplierOnce, sFastTrackMultiplierInit); 1670 if (ok != 0) { 1671 ALOGE("%s pthread_once failed: %d", __func__, ok); 1672 } 1673 frameCount = mFrameCount * sFastTrackMultiplier; 1674 } 1675 ALOGV("AUDIO_OUTPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d", 1676 frameCount, mFrameCount); 1677 } else { 1678 ALOGV("AUDIO_OUTPUT_FLAG_FAST denied: isTimed=%d sharedBuffer=%p frameCount=%d " 1679 "mFrameCount=%d format=%#x mFormat=%#x isLinear=%d channelMask=%#x " 1680 "sampleRate=%u mSampleRate=%u " 1681 "hasFastMixer=%d tid=%d fastTrackAvailMask=%#x", 1682 isTimed, sharedBuffer.get(), frameCount, mFrameCount, format, mFormat, 1683 audio_is_linear_pcm(format), 1684 channelMask, sampleRate, mSampleRate, hasFastMixer(), tid, mFastTrackAvailMask); 1685 *flags &= ~IAudioFlinger::TRACK_FAST; 1686 } 1687 } 1688 // For normal PCM streaming tracks, update minimum frame count. 1689 // For compatibility with AudioTrack calculation, buffer depth is forced 1690 // to be at least 2 x the normal mixer frame count and cover audio hardware latency. 1691 // This is probably too conservative, but legacy application code may depend on it. 1692 // If you change this calculation, also review the start threshold which is related. 1693 if (!(*flags & IAudioFlinger::TRACK_FAST) 1694 && audio_is_linear_pcm(format) && sharedBuffer == 0) { 1695 // this must match AudioTrack.cpp calculateMinFrameCount(). 1696 // TODO: Move to a common library 1697 uint32_t latencyMs = mOutput->stream->get_latency(mOutput->stream); 1698 uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate); 1699 if (minBufCount < 2) { 1700 minBufCount = 2; 1701 } 1702 // For normal mixing tracks, if speed is > 1.0f (normal), AudioTrack 1703 // or the client should compute and pass in a larger buffer request. 1704 size_t minFrameCount = 1705 minBufCount * sourceFramesNeededWithTimestretch( 1706 sampleRate, mNormalFrameCount, 1707 mSampleRate, AUDIO_TIMESTRETCH_SPEED_NORMAL /*speed*/); 1708 if (frameCount < minFrameCount) { // including frameCount == 0 1709 frameCount = minFrameCount; 1710 } 1711 } 1712 *pFrameCount = frameCount; 1713 1714 switch (mType) { 1715 1716 case DIRECT: 1717 if (audio_is_linear_pcm(format)) { 1718 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1719 ALOGE("createTrack_l() Bad parameter: sampleRate %u format %#x, channelMask 0x%08x " 1720 "for output %p with format %#x", 1721 sampleRate, format, channelMask, mOutput, mFormat); 1722 lStatus = BAD_VALUE; 1723 goto Exit; 1724 } 1725 } 1726 break; 1727 1728 case OFFLOAD: 1729 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1730 ALOGE("createTrack_l() Bad parameter: sampleRate %d format %#x, channelMask 0x%08x \"" 1731 "for output %p with format %#x", 1732 sampleRate, format, channelMask, mOutput, mFormat); 1733 lStatus = BAD_VALUE; 1734 goto Exit; 1735 } 1736 break; 1737 1738 default: 1739 if (!audio_is_linear_pcm(format)) { 1740 ALOGE("createTrack_l() Bad parameter: format %#x \"" 1741 "for output %p with format %#x", 1742 format, mOutput, mFormat); 1743 lStatus = BAD_VALUE; 1744 goto Exit; 1745 } 1746 if (sampleRate > mSampleRate * AUDIO_RESAMPLER_DOWN_RATIO_MAX) { 1747 ALOGE("Sample rate out of range: %u mSampleRate %u", sampleRate, mSampleRate); 1748 lStatus = BAD_VALUE; 1749 goto Exit; 1750 } 1751 break; 1752 1753 } 1754 1755 lStatus = initCheck(); 1756 if (lStatus != NO_ERROR) { 1757 ALOGE("createTrack_l() audio driver not initialized"); 1758 goto Exit; 1759 } 1760 1761 { // scope for mLock 1762 Mutex::Autolock _l(mLock); 1763 1764 // all tracks in same audio session must share the same routing strategy otherwise 1765 // conflicts will happen when tracks are moved from one output to another by audio policy 1766 // manager 1767 uint32_t strategy = AudioSystem::getStrategyForStream(streamType); 1768 for (size_t i = 0; i < mTracks.size(); ++i) { 1769 sp<Track> t = mTracks[i]; 1770 if (t != 0 && t->isExternalTrack()) { 1771 uint32_t actual = AudioSystem::getStrategyForStream(t->streamType()); 1772 if (sessionId == t->sessionId() && strategy != actual) { 1773 ALOGE("createTrack_l() mismatched strategy; expected %u but found %u", 1774 strategy, actual); 1775 lStatus = BAD_VALUE; 1776 goto Exit; 1777 } 1778 } 1779 } 1780 1781 if (!isTimed) { 1782 track = new Track(this, client, streamType, sampleRate, format, 1783 channelMask, frameCount, NULL, sharedBuffer, 1784 sessionId, uid, *flags, TrackBase::TYPE_DEFAULT); 1785 } else { 1786 track = TimedTrack::create(this, client, streamType, sampleRate, format, 1787 channelMask, frameCount, sharedBuffer, sessionId, uid); 1788 } 1789 1790 // new Track always returns non-NULL, 1791 // but TimedTrack::create() is a factory that could fail by returning NULL 1792 lStatus = track != 0 ? track->initCheck() : (status_t) NO_MEMORY; 1793 if (lStatus != NO_ERROR) { 1794 ALOGE("createTrack_l() initCheck failed %d; no control block?", lStatus); 1795 // track must be cleared from the caller as the caller has the AF lock 1796 goto Exit; 1797 } 1798 mTracks.add(track); 1799 1800 sp<EffectChain> chain = getEffectChain_l(sessionId); 1801 if (chain != 0) { 1802 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1803 track->setMainBuffer(chain->inBuffer()); 1804 chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType())); 1805 chain->incTrackCnt(); 1806 } 1807 1808 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 1809 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 1810 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 1811 // so ask activity manager to do this on our behalf 1812 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 1813 } 1814 } 1815 1816 lStatus = NO_ERROR; 1817 1818Exit: 1819 *status = lStatus; 1820 return track; 1821} 1822 1823uint32_t AudioFlinger::PlaybackThread::correctLatency_l(uint32_t latency) const 1824{ 1825 return latency; 1826} 1827 1828uint32_t AudioFlinger::PlaybackThread::latency() const 1829{ 1830 Mutex::Autolock _l(mLock); 1831 return latency_l(); 1832} 1833uint32_t AudioFlinger::PlaybackThread::latency_l() const 1834{ 1835 if (initCheck() == NO_ERROR) { 1836 return correctLatency_l(mOutput->stream->get_latency(mOutput->stream)); 1837 } else { 1838 return 0; 1839 } 1840} 1841 1842void AudioFlinger::PlaybackThread::setMasterVolume(float value) 1843{ 1844 Mutex::Autolock _l(mLock); 1845 // Don't apply master volume in SW if our HAL can do it for us. 1846 if (mOutput && mOutput->audioHwDev && 1847 mOutput->audioHwDev->canSetMasterVolume()) { 1848 mMasterVolume = 1.0; 1849 } else { 1850 mMasterVolume = value; 1851 } 1852} 1853 1854void AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1855{ 1856 Mutex::Autolock _l(mLock); 1857 // Don't apply master mute in SW if our HAL can do it for us. 1858 if (mOutput && mOutput->audioHwDev && 1859 mOutput->audioHwDev->canSetMasterMute()) { 1860 mMasterMute = false; 1861 } else { 1862 mMasterMute = muted; 1863 } 1864} 1865 1866void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) 1867{ 1868 Mutex::Autolock _l(mLock); 1869 mStreamTypes[stream].volume = value; 1870 broadcast_l(); 1871} 1872 1873void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) 1874{ 1875 Mutex::Autolock _l(mLock); 1876 mStreamTypes[stream].mute = muted; 1877 broadcast_l(); 1878} 1879 1880float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const 1881{ 1882 Mutex::Autolock _l(mLock); 1883 return mStreamTypes[stream].volume; 1884} 1885 1886// addTrack_l() must be called with ThreadBase::mLock held 1887status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 1888{ 1889 status_t status = ALREADY_EXISTS; 1890 1891 // set retry count for buffer fill 1892 track->mRetryCount = kMaxTrackStartupRetries; 1893 if (mActiveTracks.indexOf(track) < 0) { 1894 // the track is newly added, make sure it fills up all its 1895 // buffers before playing. This is to ensure the client will 1896 // effectively get the latency it requested. 1897 if (track->isExternalTrack()) { 1898 TrackBase::track_state state = track->mState; 1899 mLock.unlock(); 1900 status = AudioSystem::startOutput(mId, track->streamType(), 1901 (audio_session_t)track->sessionId()); 1902 mLock.lock(); 1903 // abort track was stopped/paused while we released the lock 1904 if (state != track->mState) { 1905 if (status == NO_ERROR) { 1906 mLock.unlock(); 1907 AudioSystem::stopOutput(mId, track->streamType(), 1908 (audio_session_t)track->sessionId()); 1909 mLock.lock(); 1910 } 1911 return INVALID_OPERATION; 1912 } 1913 // abort if start is rejected by audio policy manager 1914 if (status != NO_ERROR) { 1915 return PERMISSION_DENIED; 1916 } 1917#ifdef ADD_BATTERY_DATA 1918 // to track the speaker usage 1919 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 1920#endif 1921 } 1922 1923 track->mFillingUpStatus = track->sharedBuffer() != 0 ? Track::FS_FILLED : Track::FS_FILLING; 1924 track->mResetDone = false; 1925 track->mPresentationCompleteFrames = 0; 1926 mActiveTracks.add(track); 1927 mWakeLockUids.add(track->uid()); 1928 mActiveTracksGeneration++; 1929 mLatestActiveTrack = track; 1930 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1931 if (chain != 0) { 1932 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), 1933 track->sessionId()); 1934 chain->incActiveTrackCnt(); 1935 } 1936 1937 status = NO_ERROR; 1938 } 1939 1940 onAddNewTrack_l(); 1941 return status; 1942} 1943 1944bool AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 1945{ 1946 track->terminate(); 1947 // active tracks are removed by threadLoop() 1948 bool trackActive = (mActiveTracks.indexOf(track) >= 0); 1949 track->mState = TrackBase::STOPPED; 1950 if (!trackActive) { 1951 removeTrack_l(track); 1952 } else if (track->isFastTrack() || track->isOffloaded() || track->isDirect()) { 1953 track->mState = TrackBase::STOPPING_1; 1954 } 1955 1956 return trackActive; 1957} 1958 1959void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 1960{ 1961 track->triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 1962 mTracks.remove(track); 1963 deleteTrackName_l(track->name()); 1964 // redundant as track is about to be destroyed, for dumpsys only 1965 track->mName = -1; 1966 if (track->isFastTrack()) { 1967 int index = track->mFastIndex; 1968 ALOG_ASSERT(0 < index && index < (int)FastMixerState::kMaxFastTracks); 1969 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << index))); 1970 mFastTrackAvailMask |= 1 << index; 1971 // redundant as track is about to be destroyed, for dumpsys only 1972 track->mFastIndex = -1; 1973 } 1974 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1975 if (chain != 0) { 1976 chain->decTrackCnt(); 1977 } 1978} 1979 1980void AudioFlinger::PlaybackThread::broadcast_l() 1981{ 1982 // Thread could be blocked waiting for async 1983 // so signal it to handle state changes immediately 1984 // If threadLoop is currently unlocked a signal of mWaitWorkCV will 1985 // be lost so we also flag to prevent it blocking on mWaitWorkCV 1986 mSignalPending = true; 1987 mWaitWorkCV.broadcast(); 1988} 1989 1990String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 1991{ 1992 Mutex::Autolock _l(mLock); 1993 if (initCheck() != NO_ERROR) { 1994 return String8(); 1995 } 1996 1997 char *s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 1998 const String8 out_s8(s); 1999 free(s); 2000 return out_s8; 2001} 2002 2003void AudioFlinger::PlaybackThread::ioConfigChanged(audio_io_config_event event, pid_t pid) { 2004 sp<AudioIoDescriptor> desc = new AudioIoDescriptor(); 2005 ALOGV("PlaybackThread::ioConfigChanged, thread %p, event %d", this, event); 2006 2007 desc->mIoHandle = mId; 2008 2009 switch (event) { 2010 case AUDIO_OUTPUT_OPENED: 2011 case AUDIO_OUTPUT_CONFIG_CHANGED: 2012 desc->mPatch = mPatch; 2013 desc->mChannelMask = mChannelMask; 2014 desc->mSamplingRate = mSampleRate; 2015 desc->mFormat = mFormat; 2016 desc->mFrameCount = mNormalFrameCount; // FIXME see 2017 // AudioFlinger::frameCount(audio_io_handle_t) 2018 desc->mLatency = latency_l(); 2019 break; 2020 2021 case AUDIO_OUTPUT_CLOSED: 2022 default: 2023 break; 2024 } 2025 mAudioFlinger->ioConfigChanged(event, desc, pid); 2026} 2027 2028void AudioFlinger::PlaybackThread::writeCallback() 2029{ 2030 ALOG_ASSERT(mCallbackThread != 0); 2031 mCallbackThread->resetWriteBlocked(); 2032} 2033 2034void AudioFlinger::PlaybackThread::drainCallback() 2035{ 2036 ALOG_ASSERT(mCallbackThread != 0); 2037 mCallbackThread->resetDraining(); 2038} 2039 2040void AudioFlinger::PlaybackThread::resetWriteBlocked(uint32_t sequence) 2041{ 2042 Mutex::Autolock _l(mLock); 2043 // reject out of sequence requests 2044 if ((mWriteAckSequence & 1) && (sequence == mWriteAckSequence)) { 2045 mWriteAckSequence &= ~1; 2046 mWaitWorkCV.signal(); 2047 } 2048} 2049 2050void AudioFlinger::PlaybackThread::resetDraining(uint32_t sequence) 2051{ 2052 Mutex::Autolock _l(mLock); 2053 // reject out of sequence requests 2054 if ((mDrainSequence & 1) && (sequence == mDrainSequence)) { 2055 mDrainSequence &= ~1; 2056 mWaitWorkCV.signal(); 2057 } 2058} 2059 2060// static 2061int AudioFlinger::PlaybackThread::asyncCallback(stream_callback_event_t event, 2062 void *param __unused, 2063 void *cookie) 2064{ 2065 AudioFlinger::PlaybackThread *me = (AudioFlinger::PlaybackThread *)cookie; 2066 ALOGV("asyncCallback() event %d", event); 2067 switch (event) { 2068 case STREAM_CBK_EVENT_WRITE_READY: 2069 me->writeCallback(); 2070 break; 2071 case STREAM_CBK_EVENT_DRAIN_READY: 2072 me->drainCallback(); 2073 break; 2074 default: 2075 ALOGW("asyncCallback() unknown event %d", event); 2076 break; 2077 } 2078 return 0; 2079} 2080 2081void AudioFlinger::PlaybackThread::readOutputParameters_l() 2082{ 2083 // unfortunately we have no way of recovering from errors here, hence the LOG_ALWAYS_FATAL 2084 mSampleRate = mOutput->getSampleRate(); 2085 mChannelMask = mOutput->getChannelMask(); 2086 if (!audio_is_output_channel(mChannelMask)) { 2087 LOG_ALWAYS_FATAL("HAL channel mask %#x not valid for output", mChannelMask); 2088 } 2089 if ((mType == MIXER || mType == DUPLICATING) 2090 && !isValidPcmSinkChannelMask(mChannelMask)) { 2091 LOG_ALWAYS_FATAL("HAL channel mask %#x not supported for mixed output", 2092 mChannelMask); 2093 } 2094 mChannelCount = audio_channel_count_from_out_mask(mChannelMask); 2095 2096 // Get actual HAL format. 2097 mHALFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 2098 // Get format from the shim, which will be different than the HAL format 2099 // if playing compressed audio over HDMI passthrough. 2100 mFormat = mOutput->getFormat(); 2101 if (!audio_is_valid_format(mFormat)) { 2102 LOG_ALWAYS_FATAL("HAL format %#x not valid for output", mFormat); 2103 } 2104 if ((mType == MIXER || mType == DUPLICATING) 2105 && !isValidPcmSinkFormat(mFormat)) { 2106 LOG_FATAL("HAL format %#x not supported for mixed output", 2107 mFormat); 2108 } 2109 mFrameSize = mOutput->getFrameSize(); 2110 mBufferSize = mOutput->stream->common.get_buffer_size(&mOutput->stream->common); 2111 mFrameCount = mBufferSize / mFrameSize; 2112 if (mFrameCount & 15) { 2113 ALOGW("HAL output buffer size is %u frames but AudioMixer requires multiples of 16 frames", 2114 mFrameCount); 2115 } 2116 2117 if ((mOutput->flags & AUDIO_OUTPUT_FLAG_NON_BLOCKING) && 2118 (mOutput->stream->set_callback != NULL)) { 2119 if (mOutput->stream->set_callback(mOutput->stream, 2120 AudioFlinger::PlaybackThread::asyncCallback, this) == 0) { 2121 mUseAsyncWrite = true; 2122 mCallbackThread = new AudioFlinger::AsyncCallbackThread(this); 2123 } 2124 } 2125 2126 mHwSupportsPause = false; 2127 if (mOutput->flags & AUDIO_OUTPUT_FLAG_DIRECT) { 2128 if (mOutput->stream->pause != NULL) { 2129 if (mOutput->stream->resume != NULL) { 2130 mHwSupportsPause = true; 2131 } else { 2132 ALOGW("direct output implements pause but not resume"); 2133 } 2134 } else if (mOutput->stream->resume != NULL) { 2135 ALOGW("direct output implements resume but not pause"); 2136 } 2137 } 2138 if (!mHwSupportsPause && mOutput->flags & AUDIO_OUTPUT_FLAG_HW_AV_SYNC) { 2139 LOG_ALWAYS_FATAL("HW_AV_SYNC requested but HAL does not implement pause and resume"); 2140 } 2141 2142 if (mType == DUPLICATING && mMixerBufferEnabled && mEffectBufferEnabled) { 2143 // For best precision, we use float instead of the associated output 2144 // device format (typically PCM 16 bit). 2145 2146 mFormat = AUDIO_FORMAT_PCM_FLOAT; 2147 mFrameSize = mChannelCount * audio_bytes_per_sample(mFormat); 2148 mBufferSize = mFrameSize * mFrameCount; 2149 2150 // TODO: We currently use the associated output device channel mask and sample rate. 2151 // (1) Perhaps use the ORed channel mask of all downstream MixerThreads 2152 // (if a valid mask) to avoid premature downmix. 2153 // (2) Perhaps use the maximum sample rate of all downstream MixerThreads 2154 // instead of the output device sample rate to avoid loss of high frequency information. 2155 // This may need to be updated as MixerThread/OutputTracks are added and not here. 2156 } 2157 2158 // Calculate size of normal sink buffer relative to the HAL output buffer size 2159 double multiplier = 1.0; 2160 if (mType == MIXER && (kUseFastMixer == FastMixer_Static || 2161 kUseFastMixer == FastMixer_Dynamic)) { 2162 size_t minNormalFrameCount = (kMinNormalSinkBufferSizeMs * mSampleRate) / 1000; 2163 size_t maxNormalFrameCount = (kMaxNormalSinkBufferSizeMs * mSampleRate) / 1000; 2164 // round up minimum and round down maximum to nearest 16 frames to satisfy AudioMixer 2165 minNormalFrameCount = (minNormalFrameCount + 15) & ~15; 2166 maxNormalFrameCount = maxNormalFrameCount & ~15; 2167 if (maxNormalFrameCount < minNormalFrameCount) { 2168 maxNormalFrameCount = minNormalFrameCount; 2169 } 2170 multiplier = (double) minNormalFrameCount / (double) mFrameCount; 2171 if (multiplier <= 1.0) { 2172 multiplier = 1.0; 2173 } else if (multiplier <= 2.0) { 2174 if (2 * mFrameCount <= maxNormalFrameCount) { 2175 multiplier = 2.0; 2176 } else { 2177 multiplier = (double) maxNormalFrameCount / (double) mFrameCount; 2178 } 2179 } else { 2180 // prefer an even multiplier, for compatibility with doubling of fast tracks due to HAL 2181 // SRC (it would be unusual for the normal sink buffer size to not be a multiple of fast 2182 // track, but we sometimes have to do this to satisfy the maximum frame count 2183 // constraint) 2184 // FIXME this rounding up should not be done if no HAL SRC 2185 uint32_t truncMult = (uint32_t) multiplier; 2186 if ((truncMult & 1)) { 2187 if ((truncMult + 1) * mFrameCount <= maxNormalFrameCount) { 2188 ++truncMult; 2189 } 2190 } 2191 multiplier = (double) truncMult; 2192 } 2193 } 2194 mNormalFrameCount = multiplier * mFrameCount; 2195 // round up to nearest 16 frames to satisfy AudioMixer 2196 if (mType == MIXER || mType == DUPLICATING) { 2197 mNormalFrameCount = (mNormalFrameCount + 15) & ~15; 2198 } 2199 ALOGI("HAL output buffer size %u frames, normal sink buffer size %u frames", mFrameCount, 2200 mNormalFrameCount); 2201 2202 // Check if we want to throttle the processing to no more than 2x normal rate 2203 mThreadThrottle = property_get_bool("af.thread.throttle", true /* default_value */); 2204 mThreadThrottleTimeMs = 0; 2205 mThreadThrottleEndMs = 0; 2206 mHalfBufferMs = mNormalFrameCount * 1000 / (2 * mSampleRate); 2207 2208 // mSinkBuffer is the sink buffer. Size is always multiple-of-16 frames. 2209 // Originally this was int16_t[] array, need to remove legacy implications. 2210 free(mSinkBuffer); 2211 mSinkBuffer = NULL; 2212 // For sink buffer size, we use the frame size from the downstream sink to avoid problems 2213 // with non PCM formats for compressed music, e.g. AAC, and Offload threads. 2214 const size_t sinkBufferSize = mNormalFrameCount * mFrameSize; 2215 (void)posix_memalign(&mSinkBuffer, 32, sinkBufferSize); 2216 2217 // We resize the mMixerBuffer according to the requirements of the sink buffer which 2218 // drives the output. 2219 free(mMixerBuffer); 2220 mMixerBuffer = NULL; 2221 if (mMixerBufferEnabled) { 2222 mMixerBufferFormat = AUDIO_FORMAT_PCM_FLOAT; // also valid: AUDIO_FORMAT_PCM_16_BIT. 2223 mMixerBufferSize = mNormalFrameCount * mChannelCount 2224 * audio_bytes_per_sample(mMixerBufferFormat); 2225 (void)posix_memalign(&mMixerBuffer, 32, mMixerBufferSize); 2226 } 2227 free(mEffectBuffer); 2228 mEffectBuffer = NULL; 2229 if (mEffectBufferEnabled) { 2230 mEffectBufferFormat = AUDIO_FORMAT_PCM_16_BIT; // Note: Effects support 16b only 2231 mEffectBufferSize = mNormalFrameCount * mChannelCount 2232 * audio_bytes_per_sample(mEffectBufferFormat); 2233 (void)posix_memalign(&mEffectBuffer, 32, mEffectBufferSize); 2234 } 2235 2236 // force reconfiguration of effect chains and engines to take new buffer size and audio 2237 // parameters into account 2238 // Note that mLock is not held when readOutputParameters_l() is called from the constructor 2239 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 2240 // matter. 2241 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 2242 Vector< sp<EffectChain> > effectChains = mEffectChains; 2243 for (size_t i = 0; i < effectChains.size(); i ++) { 2244 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 2245 } 2246} 2247 2248 2249status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames) 2250{ 2251 if (halFrames == NULL || dspFrames == NULL) { 2252 return BAD_VALUE; 2253 } 2254 Mutex::Autolock _l(mLock); 2255 if (initCheck() != NO_ERROR) { 2256 return INVALID_OPERATION; 2257 } 2258 size_t framesWritten = mBytesWritten / mFrameSize; 2259 *halFrames = framesWritten; 2260 2261 if (isSuspended()) { 2262 // return an estimation of rendered frames when the output is suspended 2263 size_t latencyFrames = (latency_l() * mSampleRate) / 1000; 2264 *dspFrames = framesWritten >= latencyFrames ? framesWritten - latencyFrames : 0; 2265 return NO_ERROR; 2266 } else { 2267 status_t status; 2268 uint32_t frames; 2269 status = mOutput->getRenderPosition(&frames); 2270 *dspFrames = (size_t)frames; 2271 return status; 2272 } 2273} 2274 2275uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) const 2276{ 2277 Mutex::Autolock _l(mLock); 2278 uint32_t result = 0; 2279 if (getEffectChain_l(sessionId) != 0) { 2280 result = EFFECT_SESSION; 2281 } 2282 2283 for (size_t i = 0; i < mTracks.size(); ++i) { 2284 sp<Track> track = mTracks[i]; 2285 if (sessionId == track->sessionId() && !track->isInvalid()) { 2286 result |= TRACK_SESSION; 2287 break; 2288 } 2289 } 2290 2291 return result; 2292} 2293 2294uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 2295{ 2296 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 2297 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 2298 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 2299 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 2300 } 2301 for (size_t i = 0; i < mTracks.size(); i++) { 2302 sp<Track> track = mTracks[i]; 2303 if (sessionId == track->sessionId() && !track->isInvalid()) { 2304 return AudioSystem::getStrategyForStream(track->streamType()); 2305 } 2306 } 2307 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 2308} 2309 2310 2311AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 2312{ 2313 Mutex::Autolock _l(mLock); 2314 return mOutput; 2315} 2316 2317AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 2318{ 2319 Mutex::Autolock _l(mLock); 2320 AudioStreamOut *output = mOutput; 2321 mOutput = NULL; 2322 // FIXME FastMixer might also have a raw ptr to mOutputSink; 2323 // must push a NULL and wait for ack 2324 mOutputSink.clear(); 2325 mPipeSink.clear(); 2326 mNormalSink.clear(); 2327 return output; 2328} 2329 2330// this method must always be called either with ThreadBase mLock held or inside the thread loop 2331audio_stream_t* AudioFlinger::PlaybackThread::stream() const 2332{ 2333 if (mOutput == NULL) { 2334 return NULL; 2335 } 2336 return &mOutput->stream->common; 2337} 2338 2339uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() const 2340{ 2341 return (uint32_t)((uint32_t)((mNormalFrameCount * 1000) / mSampleRate) * 1000); 2342} 2343 2344status_t AudioFlinger::PlaybackThread::setSyncEvent(const sp<SyncEvent>& event) 2345{ 2346 if (!isValidSyncEvent(event)) { 2347 return BAD_VALUE; 2348 } 2349 2350 Mutex::Autolock _l(mLock); 2351 2352 for (size_t i = 0; i < mTracks.size(); ++i) { 2353 sp<Track> track = mTracks[i]; 2354 if (event->triggerSession() == track->sessionId()) { 2355 (void) track->setSyncEvent(event); 2356 return NO_ERROR; 2357 } 2358 } 2359 2360 return NAME_NOT_FOUND; 2361} 2362 2363bool AudioFlinger::PlaybackThread::isValidSyncEvent(const sp<SyncEvent>& event) const 2364{ 2365 return event->type() == AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE; 2366} 2367 2368void AudioFlinger::PlaybackThread::threadLoop_removeTracks( 2369 const Vector< sp<Track> >& tracksToRemove) 2370{ 2371 size_t count = tracksToRemove.size(); 2372 if (count > 0) { 2373 for (size_t i = 0 ; i < count ; i++) { 2374 const sp<Track>& track = tracksToRemove.itemAt(i); 2375 if (track->isExternalTrack()) { 2376 AudioSystem::stopOutput(mId, track->streamType(), 2377 (audio_session_t)track->sessionId()); 2378#ifdef ADD_BATTERY_DATA 2379 // to track the speaker usage 2380 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 2381#endif 2382 if (track->isTerminated()) { 2383 AudioSystem::releaseOutput(mId, track->streamType(), 2384 (audio_session_t)track->sessionId()); 2385 } 2386 } 2387 } 2388 } 2389} 2390 2391void AudioFlinger::PlaybackThread::checkSilentMode_l() 2392{ 2393 if (!mMasterMute) { 2394 char value[PROPERTY_VALUE_MAX]; 2395 if (property_get("ro.audio.silent", value, "0") > 0) { 2396 char *endptr; 2397 unsigned long ul = strtoul(value, &endptr, 0); 2398 if (*endptr == '\0' && ul != 0) { 2399 ALOGD("Silence is golden"); 2400 // The setprop command will not allow a property to be changed after 2401 // the first time it is set, so we don't have to worry about un-muting. 2402 setMasterMute_l(true); 2403 } 2404 } 2405 } 2406} 2407 2408// shared by MIXER and DIRECT, overridden by DUPLICATING 2409ssize_t AudioFlinger::PlaybackThread::threadLoop_write() 2410{ 2411 // FIXME rewrite to reduce number of system calls 2412 mLastWriteTime = systemTime(); 2413 mInWrite = true; 2414 ssize_t bytesWritten; 2415 const size_t offset = mCurrentWriteLength - mBytesRemaining; 2416 2417 // If an NBAIO sink is present, use it to write the normal mixer's submix 2418 if (mNormalSink != 0) { 2419 2420 const size_t count = mBytesRemaining / mFrameSize; 2421 2422 ATRACE_BEGIN("write"); 2423 // update the setpoint when AudioFlinger::mScreenState changes 2424 uint32_t screenState = AudioFlinger::mScreenState; 2425 if (screenState != mScreenState) { 2426 mScreenState = screenState; 2427 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 2428 if (pipe != NULL) { 2429 pipe->setAvgFrames((mScreenState & 1) ? 2430 (pipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2431 } 2432 } 2433 ssize_t framesWritten = mNormalSink->write((char *)mSinkBuffer + offset, count); 2434 ATRACE_END(); 2435 if (framesWritten > 0) { 2436 bytesWritten = framesWritten * mFrameSize; 2437 } else { 2438 bytesWritten = framesWritten; 2439 } 2440 mLatchDValid = false; 2441 status_t status = mNormalSink->getTimestamp(mLatchD.mTimestamp); 2442 if (status == NO_ERROR) { 2443 size_t totalFramesWritten = mNormalSink->framesWritten(); 2444 if (totalFramesWritten >= mLatchD.mTimestamp.mPosition) { 2445 mLatchD.mUnpresentedFrames = totalFramesWritten - mLatchD.mTimestamp.mPosition; 2446 // mLatchD.mFramesReleased is set immediately before D is clocked into Q 2447 mLatchDValid = true; 2448 } 2449 } 2450 // otherwise use the HAL / AudioStreamOut directly 2451 } else { 2452 // Direct output and offload threads 2453 2454 if (mUseAsyncWrite) { 2455 ALOGW_IF(mWriteAckSequence & 1, "threadLoop_write(): out of sequence write request"); 2456 mWriteAckSequence += 2; 2457 mWriteAckSequence |= 1; 2458 ALOG_ASSERT(mCallbackThread != 0); 2459 mCallbackThread->setWriteBlocked(mWriteAckSequence); 2460 } 2461 // FIXME We should have an implementation of timestamps for direct output threads. 2462 // They are used e.g for multichannel PCM playback over HDMI. 2463 bytesWritten = mOutput->write((char *)mSinkBuffer + offset, mBytesRemaining); 2464 if (mUseAsyncWrite && 2465 ((bytesWritten < 0) || (bytesWritten == (ssize_t)mBytesRemaining))) { 2466 // do not wait for async callback in case of error of full write 2467 mWriteAckSequence &= ~1; 2468 ALOG_ASSERT(mCallbackThread != 0); 2469 mCallbackThread->setWriteBlocked(mWriteAckSequence); 2470 } 2471 } 2472 2473 mNumWrites++; 2474 mInWrite = false; 2475 mStandby = false; 2476 return bytesWritten; 2477} 2478 2479void AudioFlinger::PlaybackThread::threadLoop_drain() 2480{ 2481 if (mOutput->stream->drain) { 2482 ALOGV("draining %s", (mMixerStatus == MIXER_DRAIN_TRACK) ? "early" : "full"); 2483 if (mUseAsyncWrite) { 2484 ALOGW_IF(mDrainSequence & 1, "threadLoop_drain(): out of sequence drain request"); 2485 mDrainSequence |= 1; 2486 ALOG_ASSERT(mCallbackThread != 0); 2487 mCallbackThread->setDraining(mDrainSequence); 2488 } 2489 mOutput->stream->drain(mOutput->stream, 2490 (mMixerStatus == MIXER_DRAIN_TRACK) ? AUDIO_DRAIN_EARLY_NOTIFY 2491 : AUDIO_DRAIN_ALL); 2492 } 2493} 2494 2495void AudioFlinger::PlaybackThread::threadLoop_exit() 2496{ 2497 { 2498 Mutex::Autolock _l(mLock); 2499 for (size_t i = 0; i < mTracks.size(); i++) { 2500 sp<Track> track = mTracks[i]; 2501 track->invalidate(); 2502 } 2503 } 2504} 2505 2506/* 2507The derived values that are cached: 2508 - mSinkBufferSize from frame count * frame size 2509 - mActiveSleepTimeUs from activeSleepTimeUs() 2510 - mIdleSleepTimeUs from idleSleepTimeUs() 2511 - mStandbyDelayNs from mActiveSleepTimeUs (DIRECT only) 2512 - maxPeriod from frame count and sample rate (MIXER only) 2513 2514The parameters that affect these derived values are: 2515 - frame count 2516 - frame size 2517 - sample rate 2518 - device type: A2DP or not 2519 - device latency 2520 - format: PCM or not 2521 - active sleep time 2522 - idle sleep time 2523*/ 2524 2525void AudioFlinger::PlaybackThread::cacheParameters_l() 2526{ 2527 mSinkBufferSize = mNormalFrameCount * mFrameSize; 2528 mActiveSleepTimeUs = activeSleepTimeUs(); 2529 mIdleSleepTimeUs = idleSleepTimeUs(); 2530} 2531 2532void AudioFlinger::PlaybackThread::invalidateTracks(audio_stream_type_t streamType) 2533{ 2534 ALOGV("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 2535 this, streamType, mTracks.size()); 2536 Mutex::Autolock _l(mLock); 2537 2538 size_t size = mTracks.size(); 2539 for (size_t i = 0; i < size; i++) { 2540 sp<Track> t = mTracks[i]; 2541 if (t->streamType() == streamType) { 2542 t->invalidate(); 2543 } 2544 } 2545} 2546 2547status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 2548{ 2549 int session = chain->sessionId(); 2550 int16_t* buffer = reinterpret_cast<int16_t*>(mEffectBufferEnabled 2551 ? mEffectBuffer : mSinkBuffer); 2552 bool ownsBuffer = false; 2553 2554 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 2555 if (session > 0) { 2556 // Only one effect chain can be present in direct output thread and it uses 2557 // the sink buffer as input 2558 if (mType != DIRECT) { 2559 size_t numSamples = mNormalFrameCount * mChannelCount; 2560 buffer = new int16_t[numSamples]; 2561 memset(buffer, 0, numSamples * sizeof(int16_t)); 2562 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 2563 ownsBuffer = true; 2564 } 2565 2566 // Attach all tracks with same session ID to this chain. 2567 for (size_t i = 0; i < mTracks.size(); ++i) { 2568 sp<Track> track = mTracks[i]; 2569 if (session == track->sessionId()) { 2570 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), 2571 buffer); 2572 track->setMainBuffer(buffer); 2573 chain->incTrackCnt(); 2574 } 2575 } 2576 2577 // indicate all active tracks in the chain 2578 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 2579 sp<Track> track = mActiveTracks[i].promote(); 2580 if (track == 0) { 2581 continue; 2582 } 2583 if (session == track->sessionId()) { 2584 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 2585 chain->incActiveTrackCnt(); 2586 } 2587 } 2588 } 2589 chain->setThread(this); 2590 chain->setInBuffer(buffer, ownsBuffer); 2591 chain->setOutBuffer(reinterpret_cast<int16_t*>(mEffectBufferEnabled 2592 ? mEffectBuffer : mSinkBuffer)); 2593 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 2594 // chains list in order to be processed last as it contains output stage effects 2595 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 2596 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 2597 // after track specific effects and before output stage 2598 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 2599 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 2600 // Effect chain for other sessions are inserted at beginning of effect 2601 // chains list to be processed before output mix effects. Relative order between other 2602 // sessions is not important 2603 size_t size = mEffectChains.size(); 2604 size_t i = 0; 2605 for (i = 0; i < size; i++) { 2606 if (mEffectChains[i]->sessionId() < session) { 2607 break; 2608 } 2609 } 2610 mEffectChains.insertAt(chain, i); 2611 checkSuspendOnAddEffectChain_l(chain); 2612 2613 return NO_ERROR; 2614} 2615 2616size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 2617{ 2618 int session = chain->sessionId(); 2619 2620 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 2621 2622 for (size_t i = 0; i < mEffectChains.size(); i++) { 2623 if (chain == mEffectChains[i]) { 2624 mEffectChains.removeAt(i); 2625 // detach all active tracks from the chain 2626 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 2627 sp<Track> track = mActiveTracks[i].promote(); 2628 if (track == 0) { 2629 continue; 2630 } 2631 if (session == track->sessionId()) { 2632 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 2633 chain.get(), session); 2634 chain->decActiveTrackCnt(); 2635 } 2636 } 2637 2638 // detach all tracks with same session ID from this chain 2639 for (size_t i = 0; i < mTracks.size(); ++i) { 2640 sp<Track> track = mTracks[i]; 2641 if (session == track->sessionId()) { 2642 track->setMainBuffer(reinterpret_cast<int16_t*>(mSinkBuffer)); 2643 chain->decTrackCnt(); 2644 } 2645 } 2646 break; 2647 } 2648 } 2649 return mEffectChains.size(); 2650} 2651 2652status_t AudioFlinger::PlaybackThread::attachAuxEffect( 2653 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2654{ 2655 Mutex::Autolock _l(mLock); 2656 return attachAuxEffect_l(track, EffectId); 2657} 2658 2659status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 2660 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2661{ 2662 status_t status = NO_ERROR; 2663 2664 if (EffectId == 0) { 2665 track->setAuxBuffer(0, NULL); 2666 } else { 2667 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 2668 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 2669 if (effect != 0) { 2670 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 2671 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 2672 } else { 2673 status = INVALID_OPERATION; 2674 } 2675 } else { 2676 status = BAD_VALUE; 2677 } 2678 } 2679 return status; 2680} 2681 2682void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 2683{ 2684 for (size_t i = 0; i < mTracks.size(); ++i) { 2685 sp<Track> track = mTracks[i]; 2686 if (track->auxEffectId() == effectId) { 2687 attachAuxEffect_l(track, 0); 2688 } 2689 } 2690} 2691 2692bool AudioFlinger::PlaybackThread::threadLoop() 2693{ 2694 Vector< sp<Track> > tracksToRemove; 2695 2696 mStandbyTimeNs = systemTime(); 2697 2698 // MIXER 2699 nsecs_t lastWarning = 0; 2700 2701 // DUPLICATING 2702 // FIXME could this be made local to while loop? 2703 writeFrames = 0; 2704 2705 int lastGeneration = 0; 2706 2707 cacheParameters_l(); 2708 mSleepTimeUs = mIdleSleepTimeUs; 2709 2710 if (mType == MIXER) { 2711 sleepTimeShift = 0; 2712 } 2713 2714 CpuStats cpuStats; 2715 const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid())); 2716 2717 acquireWakeLock(); 2718 2719 // mNBLogWriter->log can only be called while thread mutex mLock is held. 2720 // So if you need to log when mutex is unlocked, set logString to a non-NULL string, 2721 // and then that string will be logged at the next convenient opportunity. 2722 const char *logString = NULL; 2723 2724 checkSilentMode_l(); 2725 2726 while (!exitPending()) 2727 { 2728 cpuStats.sample(myName); 2729 2730 Vector< sp<EffectChain> > effectChains; 2731 2732 { // scope for mLock 2733 2734 Mutex::Autolock _l(mLock); 2735 2736 processConfigEvents_l(); 2737 2738 if (logString != NULL) { 2739 mNBLogWriter->logTimestamp(); 2740 mNBLogWriter->log(logString); 2741 logString = NULL; 2742 } 2743 2744 // Gather the framesReleased counters for all active tracks, 2745 // and latch them atomically with the timestamp. 2746 // FIXME We're using raw pointers as indices. A unique track ID would be a better index. 2747 mLatchD.mFramesReleased.clear(); 2748 size_t size = mActiveTracks.size(); 2749 for (size_t i = 0; i < size; i++) { 2750 sp<Track> t = mActiveTracks[i].promote(); 2751 if (t != 0) { 2752 mLatchD.mFramesReleased.add(t.get(), 2753 t->mAudioTrackServerProxy->framesReleased()); 2754 } 2755 } 2756 if (mLatchDValid) { 2757 mLatchQ = mLatchD; 2758 mLatchDValid = false; 2759 mLatchQValid = true; 2760 } 2761 2762 saveOutputTracks(); 2763 if (mSignalPending) { 2764 // A signal was raised while we were unlocked 2765 mSignalPending = false; 2766 } else if (waitingAsyncCallback_l()) { 2767 if (exitPending()) { 2768 break; 2769 } 2770 bool released = false; 2771 // The following works around a bug in the offload driver. Ideally we would release 2772 // the wake lock every time, but that causes the last offload buffer(s) to be 2773 // dropped while the device is on battery, so we need to hold a wake lock during 2774 // the drain phase. 2775 if (mBytesRemaining && !(mDrainSequence & 1)) { 2776 releaseWakeLock_l(); 2777 released = true; 2778 } 2779 mWakeLockUids.clear(); 2780 mActiveTracksGeneration++; 2781 ALOGV("wait async completion"); 2782 mWaitWorkCV.wait(mLock); 2783 ALOGV("async completion/wake"); 2784 if (released) { 2785 acquireWakeLock_l(); 2786 } 2787 mStandbyTimeNs = systemTime() + mStandbyDelayNs; 2788 mSleepTimeUs = 0; 2789 2790 continue; 2791 } 2792 if ((!mActiveTracks.size() && systemTime() > mStandbyTimeNs) || 2793 isSuspended()) { 2794 // put audio hardware into standby after short delay 2795 if (shouldStandby_l()) { 2796 2797 threadLoop_standby(); 2798 2799 mStandby = true; 2800 } 2801 2802 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2803 // we're about to wait, flush the binder command buffer 2804 IPCThreadState::self()->flushCommands(); 2805 2806 clearOutputTracks(); 2807 2808 if (exitPending()) { 2809 break; 2810 } 2811 2812 releaseWakeLock_l(); 2813 mWakeLockUids.clear(); 2814 mActiveTracksGeneration++; 2815 // wait until we have something to do... 2816 ALOGV("%s going to sleep", myName.string()); 2817 mWaitWorkCV.wait(mLock); 2818 ALOGV("%s waking up", myName.string()); 2819 acquireWakeLock_l(); 2820 2821 mMixerStatus = MIXER_IDLE; 2822 mMixerStatusIgnoringFastTracks = MIXER_IDLE; 2823 mBytesWritten = 0; 2824 mBytesRemaining = 0; 2825 checkSilentMode_l(); 2826 2827 mStandbyTimeNs = systemTime() + mStandbyDelayNs; 2828 mSleepTimeUs = mIdleSleepTimeUs; 2829 if (mType == MIXER) { 2830 sleepTimeShift = 0; 2831 } 2832 2833 continue; 2834 } 2835 } 2836 // mMixerStatusIgnoringFastTracks is also updated internally 2837 mMixerStatus = prepareTracks_l(&tracksToRemove); 2838 2839 // compare with previously applied list 2840 if (lastGeneration != mActiveTracksGeneration) { 2841 // update wakelock 2842 updateWakeLockUids_l(mWakeLockUids); 2843 lastGeneration = mActiveTracksGeneration; 2844 } 2845 2846 // prevent any changes in effect chain list and in each effect chain 2847 // during mixing and effect process as the audio buffers could be deleted 2848 // or modified if an effect is created or deleted 2849 lockEffectChains_l(effectChains); 2850 } // mLock scope ends 2851 2852 if (mBytesRemaining == 0) { 2853 mCurrentWriteLength = 0; 2854 if (mMixerStatus == MIXER_TRACKS_READY) { 2855 // threadLoop_mix() sets mCurrentWriteLength 2856 threadLoop_mix(); 2857 } else if ((mMixerStatus != MIXER_DRAIN_TRACK) 2858 && (mMixerStatus != MIXER_DRAIN_ALL)) { 2859 // threadLoop_sleepTime sets mSleepTimeUs to 0 if data 2860 // must be written to HAL 2861 threadLoop_sleepTime(); 2862 if (mSleepTimeUs == 0) { 2863 mCurrentWriteLength = mSinkBufferSize; 2864 } 2865 } 2866 // Either threadLoop_mix() or threadLoop_sleepTime() should have set 2867 // mMixerBuffer with data if mMixerBufferValid is true and mSleepTimeUs == 0. 2868 // Merge mMixerBuffer data into mEffectBuffer (if any effects are valid) 2869 // or mSinkBuffer (if there are no effects). 2870 // 2871 // This is done pre-effects computation; if effects change to 2872 // support higher precision, this needs to move. 2873 // 2874 // mMixerBufferValid is only set true by MixerThread::prepareTracks_l(). 2875 // TODO use mSleepTimeUs == 0 as an additional condition. 2876 if (mMixerBufferValid) { 2877 void *buffer = mEffectBufferValid ? mEffectBuffer : mSinkBuffer; 2878 audio_format_t format = mEffectBufferValid ? mEffectBufferFormat : mFormat; 2879 2880 memcpy_by_audio_format(buffer, format, mMixerBuffer, mMixerBufferFormat, 2881 mNormalFrameCount * mChannelCount); 2882 } 2883 2884 mBytesRemaining = mCurrentWriteLength; 2885 if (isSuspended()) { 2886 mSleepTimeUs = suspendSleepTimeUs(); 2887 // simulate write to HAL when suspended 2888 mBytesWritten += mSinkBufferSize; 2889 mBytesRemaining = 0; 2890 } 2891 2892 // only process effects if we're going to write 2893 if (mSleepTimeUs == 0 && mType != OFFLOAD) { 2894 for (size_t i = 0; i < effectChains.size(); i ++) { 2895 effectChains[i]->process_l(); 2896 } 2897 } 2898 } 2899 // Process effect chains for offloaded thread even if no audio 2900 // was read from audio track: process only updates effect state 2901 // and thus does have to be synchronized with audio writes but may have 2902 // to be called while waiting for async write callback 2903 if (mType == OFFLOAD) { 2904 for (size_t i = 0; i < effectChains.size(); i ++) { 2905 effectChains[i]->process_l(); 2906 } 2907 } 2908 2909 // Only if the Effects buffer is enabled and there is data in the 2910 // Effects buffer (buffer valid), we need to 2911 // copy into the sink buffer. 2912 // TODO use mSleepTimeUs == 0 as an additional condition. 2913 if (mEffectBufferValid) { 2914 //ALOGV("writing effect buffer to sink buffer format %#x", mFormat); 2915 memcpy_by_audio_format(mSinkBuffer, mFormat, mEffectBuffer, mEffectBufferFormat, 2916 mNormalFrameCount * mChannelCount); 2917 } 2918 2919 // enable changes in effect chain 2920 unlockEffectChains(effectChains); 2921 2922 if (!waitingAsyncCallback()) { 2923 // mSleepTimeUs == 0 means we must write to audio hardware 2924 if (mSleepTimeUs == 0) { 2925 ssize_t ret = 0; 2926 if (mBytesRemaining) { 2927 ret = threadLoop_write(); 2928 if (ret < 0) { 2929 mBytesRemaining = 0; 2930 } else { 2931 mBytesWritten += ret; 2932 mBytesRemaining -= ret; 2933 } 2934 } else if ((mMixerStatus == MIXER_DRAIN_TRACK) || 2935 (mMixerStatus == MIXER_DRAIN_ALL)) { 2936 threadLoop_drain(); 2937 } 2938 if (mType == MIXER && !mStandby) { 2939 // write blocked detection 2940 nsecs_t now = systemTime(); 2941 nsecs_t delta = now - mLastWriteTime; 2942 if (delta > maxPeriod) { 2943 mNumDelayedWrites++; 2944 if ((now - lastWarning) > kWarningThrottleNs) { 2945 ATRACE_NAME("underrun"); 2946 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2947 ns2ms(delta), mNumDelayedWrites, this); 2948 lastWarning = now; 2949 } 2950 } 2951 2952 if (mThreadThrottle 2953 && mMixerStatus == MIXER_TRACKS_READY // we are mixing (active tracks) 2954 && ret > 0) { // we wrote something 2955 // Limit MixerThread data processing to no more than twice the 2956 // expected processing rate. 2957 // 2958 // This helps prevent underruns with NuPlayer and other applications 2959 // which may set up buffers that are close to the minimum size, or use 2960 // deep buffers, and rely on a double-buffering sleep strategy to fill. 2961 // 2962 // The throttle smooths out sudden large data drains from the device, 2963 // e.g. when it comes out of standby, which often causes problems with 2964 // (1) mixer threads without a fast mixer (which has its own warm-up) 2965 // (2) minimum buffer sized tracks (even if the track is full, 2966 // the app won't fill fast enough to handle the sudden draw). 2967 2968 const int32_t deltaMs = delta / 1000000; 2969 const int32_t throttleMs = mHalfBufferMs - deltaMs; 2970 if ((signed)mHalfBufferMs >= throttleMs && throttleMs > 0) { 2971 usleep(throttleMs * 1000); 2972 // notify of throttle start on verbose log 2973 ALOGV_IF(mThreadThrottleEndMs == mThreadThrottleTimeMs, 2974 "mixer(%p) throttle begin:" 2975 " ret(%zd) deltaMs(%d) requires sleep %d ms", 2976 this, ret, deltaMs, throttleMs); 2977 mThreadThrottleTimeMs += throttleMs; 2978 } else { 2979 uint32_t diff = mThreadThrottleTimeMs - mThreadThrottleEndMs; 2980 if (diff > 0) { 2981 // notify of throttle end on debug log 2982 ALOGD("mixer(%p) throttle end: throttle time(%u)", this, diff); 2983 mThreadThrottleEndMs = mThreadThrottleTimeMs; 2984 } 2985 } 2986 } 2987 } 2988 2989 } else { 2990 ATRACE_BEGIN("sleep"); 2991 usleep(mSleepTimeUs); 2992 ATRACE_END(); 2993 } 2994 } 2995 2996 // Finally let go of removed track(s), without the lock held 2997 // since we can't guarantee the destructors won't acquire that 2998 // same lock. This will also mutate and push a new fast mixer state. 2999 threadLoop_removeTracks(tracksToRemove); 3000 tracksToRemove.clear(); 3001 3002 // FIXME I don't understand the need for this here; 3003 // it was in the original code but maybe the 3004 // assignment in saveOutputTracks() makes this unnecessary? 3005 clearOutputTracks(); 3006 3007 // Effect chains will be actually deleted here if they were removed from 3008 // mEffectChains list during mixing or effects processing 3009 effectChains.clear(); 3010 3011 // FIXME Note that the above .clear() is no longer necessary since effectChains 3012 // is now local to this block, but will keep it for now (at least until merge done). 3013 } 3014 3015 threadLoop_exit(); 3016 3017 if (!mStandby) { 3018 threadLoop_standby(); 3019 mStandby = true; 3020 } 3021 3022 releaseWakeLock(); 3023 mWakeLockUids.clear(); 3024 mActiveTracksGeneration++; 3025 3026 ALOGV("Thread %p type %d exiting", this, mType); 3027 return false; 3028} 3029 3030// removeTracks_l() must be called with ThreadBase::mLock held 3031void AudioFlinger::PlaybackThread::removeTracks_l(const Vector< sp<Track> >& tracksToRemove) 3032{ 3033 size_t count = tracksToRemove.size(); 3034 if (count > 0) { 3035 for (size_t i=0 ; i<count ; i++) { 3036 const sp<Track>& track = tracksToRemove.itemAt(i); 3037 mActiveTracks.remove(track); 3038 mWakeLockUids.remove(track->uid()); 3039 mActiveTracksGeneration++; 3040 ALOGV("removeTracks_l removing track on session %d", track->sessionId()); 3041 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 3042 if (chain != 0) { 3043 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), 3044 track->sessionId()); 3045 chain->decActiveTrackCnt(); 3046 } 3047 if (track->isTerminated()) { 3048 removeTrack_l(track); 3049 } 3050 } 3051 } 3052 3053} 3054 3055status_t AudioFlinger::PlaybackThread::getTimestamp_l(AudioTimestamp& timestamp) 3056{ 3057 if (mNormalSink != 0) { 3058 return mNormalSink->getTimestamp(timestamp); 3059 } 3060 if ((mType == OFFLOAD || mType == DIRECT) 3061 && mOutput != NULL && mOutput->stream->get_presentation_position) { 3062 uint64_t position64; 3063 int ret = mOutput->getPresentationPosition(&position64, ×tamp.mTime); 3064 if (ret == 0) { 3065 timestamp.mPosition = (uint32_t)position64; 3066 return NO_ERROR; 3067 } 3068 } 3069 return INVALID_OPERATION; 3070} 3071 3072status_t AudioFlinger::MixerThread::createAudioPatch_l(const struct audio_patch *patch, 3073 audio_patch_handle_t *handle) 3074{ 3075 // if !&IDLE, holds the FastMixer state to restore after new parameters processed 3076 FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE; 3077 if (mFastMixer != 0) { 3078 FastMixerStateQueue *sq = mFastMixer->sq(); 3079 FastMixerState *state = sq->begin(); 3080 if (!(state->mCommand & FastMixerState::IDLE)) { 3081 previousCommand = state->mCommand; 3082 state->mCommand = FastMixerState::HOT_IDLE; 3083 sq->end(); 3084 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3085 } else { 3086 sq->end(false /*didModify*/); 3087 } 3088 } 3089 status_t status = PlaybackThread::createAudioPatch_l(patch, handle); 3090 3091 if (!(previousCommand & FastMixerState::IDLE)) { 3092 ALOG_ASSERT(mFastMixer != 0); 3093 FastMixerStateQueue *sq = mFastMixer->sq(); 3094 FastMixerState *state = sq->begin(); 3095 ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE); 3096 state->mCommand = previousCommand; 3097 sq->end(); 3098 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3099 } 3100 3101 return status; 3102} 3103 3104status_t AudioFlinger::PlaybackThread::createAudioPatch_l(const struct audio_patch *patch, 3105 audio_patch_handle_t *handle) 3106{ 3107 status_t status = NO_ERROR; 3108 3109 // store new device and send to effects 3110 audio_devices_t type = AUDIO_DEVICE_NONE; 3111 for (unsigned int i = 0; i < patch->num_sinks; i++) { 3112 type |= patch->sinks[i].ext.device.type; 3113 } 3114 3115#ifdef ADD_BATTERY_DATA 3116 // when changing the audio output device, call addBatteryData to notify 3117 // the change 3118 if (mOutDevice != type) { 3119 uint32_t params = 0; 3120 // check whether speaker is on 3121 if (type & AUDIO_DEVICE_OUT_SPEAKER) { 3122 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 3123 } 3124 3125 audio_devices_t deviceWithoutSpeaker 3126 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 3127 // check if any other device (except speaker) is on 3128 if (type & deviceWithoutSpeaker) { 3129 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 3130 } 3131 3132 if (params != 0) { 3133 addBatteryData(params); 3134 } 3135 } 3136#endif 3137 3138 for (size_t i = 0; i < mEffectChains.size(); i++) { 3139 mEffectChains[i]->setDevice_l(type); 3140 } 3141 3142 // mPrevOutDevice is the latest device set by createAudioPatch_l(). It is not set when 3143 // the thread is created so that the first patch creation triggers an ioConfigChanged callback 3144 bool configChanged = mPrevOutDevice != type; 3145 mOutDevice = type; 3146 mPatch = *patch; 3147 3148 if (mOutput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 3149 audio_hw_device_t *hwDevice = mOutput->audioHwDev->hwDevice(); 3150 status = hwDevice->create_audio_patch(hwDevice, 3151 patch->num_sources, 3152 patch->sources, 3153 patch->num_sinks, 3154 patch->sinks, 3155 handle); 3156 } else { 3157 char *address; 3158 if (strcmp(patch->sinks[0].ext.device.address, "") != 0) { 3159 //FIXME: we only support address on first sink with HAL version < 3.0 3160 address = audio_device_address_to_parameter( 3161 patch->sinks[0].ext.device.type, 3162 patch->sinks[0].ext.device.address); 3163 } else { 3164 address = (char *)calloc(1, 1); 3165 } 3166 AudioParameter param = AudioParameter(String8(address)); 3167 free(address); 3168 param.addInt(String8(AUDIO_PARAMETER_STREAM_ROUTING), (int)type); 3169 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3170 param.toString().string()); 3171 *handle = AUDIO_PATCH_HANDLE_NONE; 3172 } 3173 if (configChanged) { 3174 mPrevOutDevice = type; 3175 sendIoConfigEvent_l(AUDIO_OUTPUT_CONFIG_CHANGED); 3176 } 3177 return status; 3178} 3179 3180status_t AudioFlinger::MixerThread::releaseAudioPatch_l(const audio_patch_handle_t handle) 3181{ 3182 // if !&IDLE, holds the FastMixer state to restore after new parameters processed 3183 FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE; 3184 if (mFastMixer != 0) { 3185 FastMixerStateQueue *sq = mFastMixer->sq(); 3186 FastMixerState *state = sq->begin(); 3187 if (!(state->mCommand & FastMixerState::IDLE)) { 3188 previousCommand = state->mCommand; 3189 state->mCommand = FastMixerState::HOT_IDLE; 3190 sq->end(); 3191 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3192 } else { 3193 sq->end(false /*didModify*/); 3194 } 3195 } 3196 3197 status_t status = PlaybackThread::releaseAudioPatch_l(handle); 3198 3199 if (!(previousCommand & FastMixerState::IDLE)) { 3200 ALOG_ASSERT(mFastMixer != 0); 3201 FastMixerStateQueue *sq = mFastMixer->sq(); 3202 FastMixerState *state = sq->begin(); 3203 ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE); 3204 state->mCommand = previousCommand; 3205 sq->end(); 3206 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3207 } 3208 3209 return status; 3210} 3211 3212status_t AudioFlinger::PlaybackThread::releaseAudioPatch_l(const audio_patch_handle_t handle) 3213{ 3214 status_t status = NO_ERROR; 3215 3216 mOutDevice = AUDIO_DEVICE_NONE; 3217 3218 if (mOutput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 3219 audio_hw_device_t *hwDevice = mOutput->audioHwDev->hwDevice(); 3220 status = hwDevice->release_audio_patch(hwDevice, handle); 3221 } else { 3222 AudioParameter param; 3223 param.addInt(String8(AUDIO_PARAMETER_STREAM_ROUTING), 0); 3224 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3225 param.toString().string()); 3226 } 3227 return status; 3228} 3229 3230void AudioFlinger::PlaybackThread::addPatchTrack(const sp<PatchTrack>& track) 3231{ 3232 Mutex::Autolock _l(mLock); 3233 mTracks.add(track); 3234} 3235 3236void AudioFlinger::PlaybackThread::deletePatchTrack(const sp<PatchTrack>& track) 3237{ 3238 Mutex::Autolock _l(mLock); 3239 destroyTrack_l(track); 3240} 3241 3242void AudioFlinger::PlaybackThread::getAudioPortConfig(struct audio_port_config *config) 3243{ 3244 ThreadBase::getAudioPortConfig(config); 3245 config->role = AUDIO_PORT_ROLE_SOURCE; 3246 config->ext.mix.hw_module = mOutput->audioHwDev->handle(); 3247 config->ext.mix.usecase.stream = AUDIO_STREAM_DEFAULT; 3248} 3249 3250// ---------------------------------------------------------------------------- 3251 3252AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 3253 audio_io_handle_t id, audio_devices_t device, bool systemReady, type_t type) 3254 : PlaybackThread(audioFlinger, output, id, device, type, systemReady), 3255 // mAudioMixer below 3256 // mFastMixer below 3257 mFastMixerFutex(0) 3258 // mOutputSink below 3259 // mPipeSink below 3260 // mNormalSink below 3261{ 3262 ALOGV("MixerThread() id=%d device=%#x type=%d", id, device, type); 3263 ALOGV("mSampleRate=%u, mChannelMask=%#x, mChannelCount=%u, mFormat=%d, mFrameSize=%u, " 3264 "mFrameCount=%d, mNormalFrameCount=%d", 3265 mSampleRate, mChannelMask, mChannelCount, mFormat, mFrameSize, mFrameCount, 3266 mNormalFrameCount); 3267 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 3268 3269 if (type == DUPLICATING) { 3270 // The Duplicating thread uses the AudioMixer and delivers data to OutputTracks 3271 // (downstream MixerThreads) in DuplicatingThread::threadLoop_write(). 3272 // Do not create or use mFastMixer, mOutputSink, mPipeSink, or mNormalSink. 3273 return; 3274 } 3275 // create an NBAIO sink for the HAL output stream, and negotiate 3276 mOutputSink = new AudioStreamOutSink(output->stream); 3277 size_t numCounterOffers = 0; 3278 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount, mFormat)}; 3279 ssize_t index = mOutputSink->negotiate(offers, 1, NULL, numCounterOffers); 3280 ALOG_ASSERT(index == 0); 3281 3282 // initialize fast mixer depending on configuration 3283 bool initFastMixer; 3284 switch (kUseFastMixer) { 3285 case FastMixer_Never: 3286 initFastMixer = false; 3287 break; 3288 case FastMixer_Always: 3289 initFastMixer = true; 3290 break; 3291 case FastMixer_Static: 3292 case FastMixer_Dynamic: 3293 initFastMixer = mFrameCount < mNormalFrameCount; 3294 break; 3295 } 3296 if (initFastMixer) { 3297 audio_format_t fastMixerFormat; 3298 if (mMixerBufferEnabled && mEffectBufferEnabled) { 3299 fastMixerFormat = AUDIO_FORMAT_PCM_FLOAT; 3300 } else { 3301 fastMixerFormat = AUDIO_FORMAT_PCM_16_BIT; 3302 } 3303 if (mFormat != fastMixerFormat) { 3304 // change our Sink format to accept our intermediate precision 3305 mFormat = fastMixerFormat; 3306 free(mSinkBuffer); 3307 mFrameSize = mChannelCount * audio_bytes_per_sample(mFormat); 3308 const size_t sinkBufferSize = mNormalFrameCount * mFrameSize; 3309 (void)posix_memalign(&mSinkBuffer, 32, sinkBufferSize); 3310 } 3311 3312 // create a MonoPipe to connect our submix to FastMixer 3313 NBAIO_Format format = mOutputSink->format(); 3314 NBAIO_Format origformat = format; 3315 // adjust format to match that of the Fast Mixer 3316 ALOGV("format changed from %d to %d", format.mFormat, fastMixerFormat); 3317 format.mFormat = fastMixerFormat; 3318 format.mFrameSize = audio_bytes_per_sample(format.mFormat) * format.mChannelCount; 3319 3320 // This pipe depth compensates for scheduling latency of the normal mixer thread. 3321 // When it wakes up after a maximum latency, it runs a few cycles quickly before 3322 // finally blocking. Note the pipe implementation rounds up the request to a power of 2. 3323 MonoPipe *monoPipe = new MonoPipe(mNormalFrameCount * 4, format, true /*writeCanBlock*/); 3324 const NBAIO_Format offers[1] = {format}; 3325 size_t numCounterOffers = 0; 3326 ssize_t index = monoPipe->negotiate(offers, 1, NULL, numCounterOffers); 3327 ALOG_ASSERT(index == 0); 3328 monoPipe->setAvgFrames((mScreenState & 1) ? 3329 (monoPipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 3330 mPipeSink = monoPipe; 3331 3332#ifdef TEE_SINK 3333 if (mTeeSinkOutputEnabled) { 3334 // create a Pipe to archive a copy of FastMixer's output for dumpsys 3335 Pipe *teeSink = new Pipe(mTeeSinkOutputFrames, origformat); 3336 const NBAIO_Format offers2[1] = {origformat}; 3337 numCounterOffers = 0; 3338 index = teeSink->negotiate(offers2, 1, NULL, numCounterOffers); 3339 ALOG_ASSERT(index == 0); 3340 mTeeSink = teeSink; 3341 PipeReader *teeSource = new PipeReader(*teeSink); 3342 numCounterOffers = 0; 3343 index = teeSource->negotiate(offers2, 1, NULL, numCounterOffers); 3344 ALOG_ASSERT(index == 0); 3345 mTeeSource = teeSource; 3346 } 3347#endif 3348 3349 // create fast mixer and configure it initially with just one fast track for our submix 3350 mFastMixer = new FastMixer(); 3351 FastMixerStateQueue *sq = mFastMixer->sq(); 3352#ifdef STATE_QUEUE_DUMP 3353 sq->setObserverDump(&mStateQueueObserverDump); 3354 sq->setMutatorDump(&mStateQueueMutatorDump); 3355#endif 3356 FastMixerState *state = sq->begin(); 3357 FastTrack *fastTrack = &state->mFastTracks[0]; 3358 // wrap the source side of the MonoPipe to make it an AudioBufferProvider 3359 fastTrack->mBufferProvider = new SourceAudioBufferProvider(new MonoPipeReader(monoPipe)); 3360 fastTrack->mVolumeProvider = NULL; 3361 fastTrack->mChannelMask = mChannelMask; // mPipeSink channel mask for audio to FastMixer 3362 fastTrack->mFormat = mFormat; // mPipeSink format for audio to FastMixer 3363 fastTrack->mGeneration++; 3364 state->mFastTracksGen++; 3365 state->mTrackMask = 1; 3366 // fast mixer will use the HAL output sink 3367 state->mOutputSink = mOutputSink.get(); 3368 state->mOutputSinkGen++; 3369 state->mFrameCount = mFrameCount; 3370 state->mCommand = FastMixerState::COLD_IDLE; 3371 // already done in constructor initialization list 3372 //mFastMixerFutex = 0; 3373 state->mColdFutexAddr = &mFastMixerFutex; 3374 state->mColdGen++; 3375 state->mDumpState = &mFastMixerDumpState; 3376#ifdef TEE_SINK 3377 state->mTeeSink = mTeeSink.get(); 3378#endif 3379 mFastMixerNBLogWriter = audioFlinger->newWriter_l(kFastMixerLogSize, "FastMixer"); 3380 state->mNBLogWriter = mFastMixerNBLogWriter.get(); 3381 sq->end(); 3382 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3383 3384 // start the fast mixer 3385 mFastMixer->run("FastMixer", PRIORITY_URGENT_AUDIO); 3386 pid_t tid = mFastMixer->getTid(); 3387 sendPrioConfigEvent(getpid_cached, tid, kPriorityFastMixer); 3388 3389#ifdef AUDIO_WATCHDOG 3390 // create and start the watchdog 3391 mAudioWatchdog = new AudioWatchdog(); 3392 mAudioWatchdog->setDump(&mAudioWatchdogDump); 3393 mAudioWatchdog->run("AudioWatchdog", PRIORITY_URGENT_AUDIO); 3394 tid = mAudioWatchdog->getTid(); 3395 sendPrioConfigEvent(getpid_cached, tid, kPriorityFastMixer); 3396#endif 3397 3398 } 3399 3400 switch (kUseFastMixer) { 3401 case FastMixer_Never: 3402 case FastMixer_Dynamic: 3403 mNormalSink = mOutputSink; 3404 break; 3405 case FastMixer_Always: 3406 mNormalSink = mPipeSink; 3407 break; 3408 case FastMixer_Static: 3409 mNormalSink = initFastMixer ? mPipeSink : mOutputSink; 3410 break; 3411 } 3412} 3413 3414AudioFlinger::MixerThread::~MixerThread() 3415{ 3416 if (mFastMixer != 0) { 3417 FastMixerStateQueue *sq = mFastMixer->sq(); 3418 FastMixerState *state = sq->begin(); 3419 if (state->mCommand == FastMixerState::COLD_IDLE) { 3420 int32_t old = android_atomic_inc(&mFastMixerFutex); 3421 if (old == -1) { 3422 (void) syscall(__NR_futex, &mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 3423 } 3424 } 3425 state->mCommand = FastMixerState::EXIT; 3426 sq->end(); 3427 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3428 mFastMixer->join(); 3429 // Though the fast mixer thread has exited, it's state queue is still valid. 3430 // We'll use that extract the final state which contains one remaining fast track 3431 // corresponding to our sub-mix. 3432 state = sq->begin(); 3433 ALOG_ASSERT(state->mTrackMask == 1); 3434 FastTrack *fastTrack = &state->mFastTracks[0]; 3435 ALOG_ASSERT(fastTrack->mBufferProvider != NULL); 3436 delete fastTrack->mBufferProvider; 3437 sq->end(false /*didModify*/); 3438 mFastMixer.clear(); 3439#ifdef AUDIO_WATCHDOG 3440 if (mAudioWatchdog != 0) { 3441 mAudioWatchdog->requestExit(); 3442 mAudioWatchdog->requestExitAndWait(); 3443 mAudioWatchdog.clear(); 3444 } 3445#endif 3446 } 3447 mAudioFlinger->unregisterWriter(mFastMixerNBLogWriter); 3448 delete mAudioMixer; 3449} 3450 3451 3452uint32_t AudioFlinger::MixerThread::correctLatency_l(uint32_t latency) const 3453{ 3454 if (mFastMixer != 0) { 3455 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 3456 latency += (pipe->getAvgFrames() * 1000) / mSampleRate; 3457 } 3458 return latency; 3459} 3460 3461 3462void AudioFlinger::MixerThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) 3463{ 3464 PlaybackThread::threadLoop_removeTracks(tracksToRemove); 3465} 3466 3467ssize_t AudioFlinger::MixerThread::threadLoop_write() 3468{ 3469 // FIXME we should only do one push per cycle; confirm this is true 3470 // Start the fast mixer if it's not already running 3471 if (mFastMixer != 0) { 3472 FastMixerStateQueue *sq = mFastMixer->sq(); 3473 FastMixerState *state = sq->begin(); 3474 if (state->mCommand != FastMixerState::MIX_WRITE && 3475 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)) { 3476 if (state->mCommand == FastMixerState::COLD_IDLE) { 3477 int32_t old = android_atomic_inc(&mFastMixerFutex); 3478 if (old == -1) { 3479 (void) syscall(__NR_futex, &mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 3480 } 3481#ifdef AUDIO_WATCHDOG 3482 if (mAudioWatchdog != 0) { 3483 mAudioWatchdog->resume(); 3484 } 3485#endif 3486 } 3487 state->mCommand = FastMixerState::MIX_WRITE; 3488#ifdef FAST_THREAD_STATISTICS 3489 mFastMixerDumpState.increaseSamplingN(mAudioFlinger->isLowRamDevice() ? 3490 FastThreadDumpState::kSamplingNforLowRamDevice : FastThreadDumpState::kSamplingN); 3491#endif 3492 sq->end(); 3493 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3494 if (kUseFastMixer == FastMixer_Dynamic) { 3495 mNormalSink = mPipeSink; 3496 } 3497 } else { 3498 sq->end(false /*didModify*/); 3499 } 3500 } 3501 return PlaybackThread::threadLoop_write(); 3502} 3503 3504void AudioFlinger::MixerThread::threadLoop_standby() 3505{ 3506 // Idle the fast mixer if it's currently running 3507 if (mFastMixer != 0) { 3508 FastMixerStateQueue *sq = mFastMixer->sq(); 3509 FastMixerState *state = sq->begin(); 3510 if (!(state->mCommand & FastMixerState::IDLE)) { 3511 state->mCommand = FastMixerState::COLD_IDLE; 3512 state->mColdFutexAddr = &mFastMixerFutex; 3513 state->mColdGen++; 3514 mFastMixerFutex = 0; 3515 sq->end(); 3516 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 3517 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3518 if (kUseFastMixer == FastMixer_Dynamic) { 3519 mNormalSink = mOutputSink; 3520 } 3521#ifdef AUDIO_WATCHDOG 3522 if (mAudioWatchdog != 0) { 3523 mAudioWatchdog->pause(); 3524 } 3525#endif 3526 } else { 3527 sq->end(false /*didModify*/); 3528 } 3529 } 3530 PlaybackThread::threadLoop_standby(); 3531} 3532 3533bool AudioFlinger::PlaybackThread::waitingAsyncCallback_l() 3534{ 3535 return false; 3536} 3537 3538bool AudioFlinger::PlaybackThread::shouldStandby_l() 3539{ 3540 return !mStandby; 3541} 3542 3543bool AudioFlinger::PlaybackThread::waitingAsyncCallback() 3544{ 3545 Mutex::Autolock _l(mLock); 3546 return waitingAsyncCallback_l(); 3547} 3548 3549// shared by MIXER and DIRECT, overridden by DUPLICATING 3550void AudioFlinger::PlaybackThread::threadLoop_standby() 3551{ 3552 ALOGV("Audio hardware entering standby, mixer %p, suspend count %d", this, mSuspended); 3553 mOutput->standby(); 3554 if (mUseAsyncWrite != 0) { 3555 // discard any pending drain or write ack by incrementing sequence 3556 mWriteAckSequence = (mWriteAckSequence + 2) & ~1; 3557 mDrainSequence = (mDrainSequence + 2) & ~1; 3558 ALOG_ASSERT(mCallbackThread != 0); 3559 mCallbackThread->setWriteBlocked(mWriteAckSequence); 3560 mCallbackThread->setDraining(mDrainSequence); 3561 } 3562 mHwPaused = false; 3563} 3564 3565void AudioFlinger::PlaybackThread::onAddNewTrack_l() 3566{ 3567 ALOGV("signal playback thread"); 3568 broadcast_l(); 3569} 3570 3571void AudioFlinger::MixerThread::threadLoop_mix() 3572{ 3573 // obtain the presentation timestamp of the next output buffer 3574 int64_t pts; 3575 status_t status = INVALID_OPERATION; 3576 3577 if (mNormalSink != 0) { 3578 status = mNormalSink->getNextWriteTimestamp(&pts); 3579 } else { 3580 status = mOutputSink->getNextWriteTimestamp(&pts); 3581 } 3582 3583 if (status != NO_ERROR) { 3584 pts = AudioBufferProvider::kInvalidPTS; 3585 } 3586 3587 // mix buffers... 3588 mAudioMixer->process(pts); 3589 mCurrentWriteLength = mSinkBufferSize; 3590 // increase sleep time progressively when application underrun condition clears. 3591 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 3592 // that a steady state of alternating ready/not ready conditions keeps the sleep time 3593 // such that we would underrun the audio HAL. 3594 if ((mSleepTimeUs == 0) && (sleepTimeShift > 0)) { 3595 sleepTimeShift--; 3596 } 3597 mSleepTimeUs = 0; 3598 mStandbyTimeNs = systemTime() + mStandbyDelayNs; 3599 //TODO: delay standby when effects have a tail 3600 3601} 3602 3603void AudioFlinger::MixerThread::threadLoop_sleepTime() 3604{ 3605 // If no tracks are ready, sleep once for the duration of an output 3606 // buffer size, then write 0s to the output 3607 if (mSleepTimeUs == 0) { 3608 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3609 mSleepTimeUs = mActiveSleepTimeUs >> sleepTimeShift; 3610 if (mSleepTimeUs < kMinThreadSleepTimeUs) { 3611 mSleepTimeUs = kMinThreadSleepTimeUs; 3612 } 3613 // reduce sleep time in case of consecutive application underruns to avoid 3614 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 3615 // duration we would end up writing less data than needed by the audio HAL if 3616 // the condition persists. 3617 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 3618 sleepTimeShift++; 3619 } 3620 } else { 3621 mSleepTimeUs = mIdleSleepTimeUs; 3622 } 3623 } else if (mBytesWritten != 0 || (mMixerStatus == MIXER_TRACKS_ENABLED)) { 3624 // clear out mMixerBuffer or mSinkBuffer, to ensure buffers are cleared 3625 // before effects processing or output. 3626 if (mMixerBufferValid) { 3627 memset(mMixerBuffer, 0, mMixerBufferSize); 3628 } else { 3629 memset(mSinkBuffer, 0, mSinkBufferSize); 3630 } 3631 mSleepTimeUs = 0; 3632 ALOGV_IF(mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED), 3633 "anticipated start"); 3634 } 3635 // TODO add standby time extension fct of effect tail 3636} 3637 3638// prepareTracks_l() must be called with ThreadBase::mLock held 3639AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 3640 Vector< sp<Track> > *tracksToRemove) 3641{ 3642 3643 mixer_state mixerStatus = MIXER_IDLE; 3644 // find out which tracks need to be processed 3645 size_t count = mActiveTracks.size(); 3646 size_t mixedTracks = 0; 3647 size_t tracksWithEffect = 0; 3648 // counts only _active_ fast tracks 3649 size_t fastTracks = 0; 3650 uint32_t resetMask = 0; // bit mask of fast tracks that need to be reset 3651 3652 float masterVolume = mMasterVolume; 3653 bool masterMute = mMasterMute; 3654 3655 if (masterMute) { 3656 masterVolume = 0; 3657 } 3658 // Delegate master volume control to effect in output mix effect chain if needed 3659 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 3660 if (chain != 0) { 3661 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 3662 chain->setVolume_l(&v, &v); 3663 masterVolume = (float)((v + (1 << 23)) >> 24); 3664 chain.clear(); 3665 } 3666 3667 // prepare a new state to push 3668 FastMixerStateQueue *sq = NULL; 3669 FastMixerState *state = NULL; 3670 bool didModify = false; 3671 FastMixerStateQueue::block_t block = FastMixerStateQueue::BLOCK_UNTIL_PUSHED; 3672 if (mFastMixer != 0) { 3673 sq = mFastMixer->sq(); 3674 state = sq->begin(); 3675 } 3676 3677 mMixerBufferValid = false; // mMixerBuffer has no valid data until appropriate tracks found. 3678 mEffectBufferValid = false; // mEffectBuffer has no valid data until tracks found. 3679 3680 for (size_t i=0 ; i<count ; i++) { 3681 const sp<Track> t = mActiveTracks[i].promote(); 3682 if (t == 0) { 3683 continue; 3684 } 3685 3686 // this const just means the local variable doesn't change 3687 Track* const track = t.get(); 3688 3689 // process fast tracks 3690 if (track->isFastTrack()) { 3691 3692 // It's theoretically possible (though unlikely) for a fast track to be created 3693 // and then removed within the same normal mix cycle. This is not a problem, as 3694 // the track never becomes active so it's fast mixer slot is never touched. 3695 // The converse, of removing an (active) track and then creating a new track 3696 // at the identical fast mixer slot within the same normal mix cycle, 3697 // is impossible because the slot isn't marked available until the end of each cycle. 3698 int j = track->mFastIndex; 3699 ALOG_ASSERT(0 < j && j < (int)FastMixerState::kMaxFastTracks); 3700 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << j))); 3701 FastTrack *fastTrack = &state->mFastTracks[j]; 3702 3703 // Determine whether the track is currently in underrun condition, 3704 // and whether it had a recent underrun. 3705 FastTrackDump *ftDump = &mFastMixerDumpState.mTracks[j]; 3706 FastTrackUnderruns underruns = ftDump->mUnderruns; 3707 uint32_t recentFull = (underruns.mBitFields.mFull - 3708 track->mObservedUnderruns.mBitFields.mFull) & UNDERRUN_MASK; 3709 uint32_t recentPartial = (underruns.mBitFields.mPartial - 3710 track->mObservedUnderruns.mBitFields.mPartial) & UNDERRUN_MASK; 3711 uint32_t recentEmpty = (underruns.mBitFields.mEmpty - 3712 track->mObservedUnderruns.mBitFields.mEmpty) & UNDERRUN_MASK; 3713 uint32_t recentUnderruns = recentPartial + recentEmpty; 3714 track->mObservedUnderruns = underruns; 3715 // don't count underruns that occur while stopping or pausing 3716 // or stopped which can occur when flush() is called while active 3717 if (!(track->isStopping() || track->isPausing() || track->isStopped()) && 3718 recentUnderruns > 0) { 3719 // FIXME fast mixer will pull & mix partial buffers, but we count as a full underrun 3720 track->mAudioTrackServerProxy->tallyUnderrunFrames(recentUnderruns * mFrameCount); 3721 } 3722 3723 // This is similar to the state machine for normal tracks, 3724 // with a few modifications for fast tracks. 3725 bool isActive = true; 3726 switch (track->mState) { 3727 case TrackBase::STOPPING_1: 3728 // track stays active in STOPPING_1 state until first underrun 3729 if (recentUnderruns > 0 || track->isTerminated()) { 3730 track->mState = TrackBase::STOPPING_2; 3731 } 3732 break; 3733 case TrackBase::PAUSING: 3734 // ramp down is not yet implemented 3735 track->setPaused(); 3736 break; 3737 case TrackBase::RESUMING: 3738 // ramp up is not yet implemented 3739 track->mState = TrackBase::ACTIVE; 3740 break; 3741 case TrackBase::ACTIVE: 3742 if (recentFull > 0 || recentPartial > 0) { 3743 // track has provided at least some frames recently: reset retry count 3744 track->mRetryCount = kMaxTrackRetries; 3745 } 3746 if (recentUnderruns == 0) { 3747 // no recent underruns: stay active 3748 break; 3749 } 3750 // there has recently been an underrun of some kind 3751 if (track->sharedBuffer() == 0) { 3752 // were any of the recent underruns "empty" (no frames available)? 3753 if (recentEmpty == 0) { 3754 // no, then ignore the partial underruns as they are allowed indefinitely 3755 break; 3756 } 3757 // there has recently been an "empty" underrun: decrement the retry counter 3758 if (--(track->mRetryCount) > 0) { 3759 break; 3760 } 3761 // indicate to client process that the track was disabled because of underrun; 3762 // it will then automatically call start() when data is available 3763 android_atomic_or(CBLK_DISABLED, &track->mCblk->mFlags); 3764 // remove from active list, but state remains ACTIVE [confusing but true] 3765 isActive = false; 3766 break; 3767 } 3768 // fall through 3769 case TrackBase::STOPPING_2: 3770 case TrackBase::PAUSED: 3771 case TrackBase::STOPPED: 3772 case TrackBase::FLUSHED: // flush() while active 3773 // Check for presentation complete if track is inactive 3774 // We have consumed all the buffers of this track. 3775 // This would be incomplete if we auto-paused on underrun 3776 { 3777 size_t audioHALFrames = 3778 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 3779 size_t framesWritten = mBytesWritten / mFrameSize; 3780 if (!(mStandby || track->presentationComplete(framesWritten, audioHALFrames))) { 3781 // track stays in active list until presentation is complete 3782 break; 3783 } 3784 } 3785 if (track->isStopping_2()) { 3786 track->mState = TrackBase::STOPPED; 3787 } 3788 if (track->isStopped()) { 3789 // Can't reset directly, as fast mixer is still polling this track 3790 // track->reset(); 3791 // So instead mark this track as needing to be reset after push with ack 3792 resetMask |= 1 << i; 3793 } 3794 isActive = false; 3795 break; 3796 case TrackBase::IDLE: 3797 default: 3798 LOG_ALWAYS_FATAL("unexpected track state %d", track->mState); 3799 } 3800 3801 if (isActive) { 3802 // was it previously inactive? 3803 if (!(state->mTrackMask & (1 << j))) { 3804 ExtendedAudioBufferProvider *eabp = track; 3805 VolumeProvider *vp = track; 3806 fastTrack->mBufferProvider = eabp; 3807 fastTrack->mVolumeProvider = vp; 3808 fastTrack->mChannelMask = track->mChannelMask; 3809 fastTrack->mFormat = track->mFormat; 3810 fastTrack->mGeneration++; 3811 state->mTrackMask |= 1 << j; 3812 didModify = true; 3813 // no acknowledgement required for newly active tracks 3814 } 3815 // cache the combined master volume and stream type volume for fast mixer; this 3816 // lacks any synchronization or barrier so VolumeProvider may read a stale value 3817 track->mCachedVolume = masterVolume * mStreamTypes[track->streamType()].volume; 3818 ++fastTracks; 3819 } else { 3820 // was it previously active? 3821 if (state->mTrackMask & (1 << j)) { 3822 fastTrack->mBufferProvider = NULL; 3823 fastTrack->mGeneration++; 3824 state->mTrackMask &= ~(1 << j); 3825 didModify = true; 3826 // If any fast tracks were removed, we must wait for acknowledgement 3827 // because we're about to decrement the last sp<> on those tracks. 3828 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3829 } else { 3830 LOG_ALWAYS_FATAL("fast track %d should have been active", j); 3831 } 3832 tracksToRemove->add(track); 3833 // Avoids a misleading display in dumpsys 3834 track->mObservedUnderruns.mBitFields.mMostRecent = UNDERRUN_FULL; 3835 } 3836 continue; 3837 } 3838 3839 { // local variable scope to avoid goto warning 3840 3841 audio_track_cblk_t* cblk = track->cblk(); 3842 3843 // The first time a track is added we wait 3844 // for all its buffers to be filled before processing it 3845 int name = track->name(); 3846 // make sure that we have enough frames to mix one full buffer. 3847 // enforce this condition only once to enable draining the buffer in case the client 3848 // app does not call stop() and relies on underrun to stop: 3849 // hence the test on (mMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 3850 // during last round 3851 size_t desiredFrames; 3852 const uint32_t sampleRate = track->mAudioTrackServerProxy->getSampleRate(); 3853 AudioPlaybackRate playbackRate = track->mAudioTrackServerProxy->getPlaybackRate(); 3854 3855 desiredFrames = sourceFramesNeededWithTimestretch( 3856 sampleRate, mNormalFrameCount, mSampleRate, playbackRate.mSpeed); 3857 // TODO: ONLY USED FOR LEGACY RESAMPLERS, remove when they are removed. 3858 // add frames already consumed but not yet released by the resampler 3859 // because mAudioTrackServerProxy->framesReady() will include these frames 3860 desiredFrames += mAudioMixer->getUnreleasedFrames(track->name()); 3861 3862 uint32_t minFrames = 1; 3863 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() && 3864 (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) { 3865 minFrames = desiredFrames; 3866 } 3867 3868 size_t framesReady = track->framesReady(); 3869 if (ATRACE_ENABLED()) { 3870 // I wish we had formatted trace names 3871 char traceName[16]; 3872 strcpy(traceName, "nRdy"); 3873 int name = track->name(); 3874 if (AudioMixer::TRACK0 <= name && 3875 name < (int) (AudioMixer::TRACK0 + AudioMixer::MAX_NUM_TRACKS)) { 3876 name -= AudioMixer::TRACK0; 3877 traceName[4] = (name / 10) + '0'; 3878 traceName[5] = (name % 10) + '0'; 3879 } else { 3880 traceName[4] = '?'; 3881 traceName[5] = '?'; 3882 } 3883 traceName[6] = '\0'; 3884 ATRACE_INT(traceName, framesReady); 3885 } 3886 if ((framesReady >= minFrames) && track->isReady() && 3887 !track->isPaused() && !track->isTerminated()) 3888 { 3889 ALOGVV("track %d s=%08x [OK] on thread %p", name, cblk->mServer, this); 3890 3891 mixedTracks++; 3892 3893 // track->mainBuffer() != mSinkBuffer or mMixerBuffer means 3894 // there is an effect chain connected to the track 3895 chain.clear(); 3896 if (track->mainBuffer() != mSinkBuffer && 3897 track->mainBuffer() != mMixerBuffer) { 3898 if (mEffectBufferEnabled) { 3899 mEffectBufferValid = true; // Later can set directly. 3900 } 3901 chain = getEffectChain_l(track->sessionId()); 3902 // Delegate volume control to effect in track effect chain if needed 3903 if (chain != 0) { 3904 tracksWithEffect++; 3905 } else { 3906 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on " 3907 "session %d", 3908 name, track->sessionId()); 3909 } 3910 } 3911 3912 3913 int param = AudioMixer::VOLUME; 3914 if (track->mFillingUpStatus == Track::FS_FILLED) { 3915 // no ramp for the first volume setting 3916 track->mFillingUpStatus = Track::FS_ACTIVE; 3917 if (track->mState == TrackBase::RESUMING) { 3918 track->mState = TrackBase::ACTIVE; 3919 param = AudioMixer::RAMP_VOLUME; 3920 } 3921 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 3922 // FIXME should not make a decision based on mServer 3923 } else if (cblk->mServer != 0) { 3924 // If the track is stopped before the first frame was mixed, 3925 // do not apply ramp 3926 param = AudioMixer::RAMP_VOLUME; 3927 } 3928 3929 // compute volume for this track 3930 uint32_t vl, vr; // in U8.24 integer format 3931 float vlf, vrf, vaf; // in [0.0, 1.0] float format 3932 if (track->isPausing() || mStreamTypes[track->streamType()].mute) { 3933 vl = vr = 0; 3934 vlf = vrf = vaf = 0.; 3935 if (track->isPausing()) { 3936 track->setPaused(); 3937 } 3938 } else { 3939 3940 // read original volumes with volume control 3941 float typeVolume = mStreamTypes[track->streamType()].volume; 3942 float v = masterVolume * typeVolume; 3943 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 3944 gain_minifloat_packed_t vlr = proxy->getVolumeLR(); 3945 vlf = float_from_gain(gain_minifloat_unpack_left(vlr)); 3946 vrf = float_from_gain(gain_minifloat_unpack_right(vlr)); 3947 // track volumes come from shared memory, so can't be trusted and must be clamped 3948 if (vlf > GAIN_FLOAT_UNITY) { 3949 ALOGV("Track left volume out of range: %.3g", vlf); 3950 vlf = GAIN_FLOAT_UNITY; 3951 } 3952 if (vrf > GAIN_FLOAT_UNITY) { 3953 ALOGV("Track right volume out of range: %.3g", vrf); 3954 vrf = GAIN_FLOAT_UNITY; 3955 } 3956 // now apply the master volume and stream type volume 3957 vlf *= v; 3958 vrf *= v; 3959 // assuming master volume and stream type volume each go up to 1.0, 3960 // then derive vl and vr as U8.24 versions for the effect chain 3961 const float scaleto8_24 = MAX_GAIN_INT * MAX_GAIN_INT; 3962 vl = (uint32_t) (scaleto8_24 * vlf); 3963 vr = (uint32_t) (scaleto8_24 * vrf); 3964 // vl and vr are now in U8.24 format 3965 uint16_t sendLevel = proxy->getSendLevel_U4_12(); 3966 // send level comes from shared memory and so may be corrupt 3967 if (sendLevel > MAX_GAIN_INT) { 3968 ALOGV("Track send level out of range: %04X", sendLevel); 3969 sendLevel = MAX_GAIN_INT; 3970 } 3971 // vaf is represented as [0.0, 1.0] float by rescaling sendLevel 3972 vaf = v * sendLevel * (1. / MAX_GAIN_INT); 3973 } 3974 3975 // Delegate volume control to effect in track effect chain if needed 3976 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 3977 // Do not ramp volume if volume is controlled by effect 3978 param = AudioMixer::VOLUME; 3979 // Update remaining floating point volume levels 3980 vlf = (float)vl / (1 << 24); 3981 vrf = (float)vr / (1 << 24); 3982 track->mHasVolumeController = true; 3983 } else { 3984 // force no volume ramp when volume controller was just disabled or removed 3985 // from effect chain to avoid volume spike 3986 if (track->mHasVolumeController) { 3987 param = AudioMixer::VOLUME; 3988 } 3989 track->mHasVolumeController = false; 3990 } 3991 3992 // XXX: these things DON'T need to be done each time 3993 mAudioMixer->setBufferProvider(name, track); 3994 mAudioMixer->enable(name); 3995 3996 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, &vlf); 3997 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, &vrf); 3998 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, &vaf); 3999 mAudioMixer->setParameter( 4000 name, 4001 AudioMixer::TRACK, 4002 AudioMixer::FORMAT, (void *)track->format()); 4003 mAudioMixer->setParameter( 4004 name, 4005 AudioMixer::TRACK, 4006 AudioMixer::CHANNEL_MASK, (void *)(uintptr_t)track->channelMask()); 4007 mAudioMixer->setParameter( 4008 name, 4009 AudioMixer::TRACK, 4010 AudioMixer::MIXER_CHANNEL_MASK, (void *)(uintptr_t)mChannelMask); 4011 // limit track sample rate to 2 x output sample rate, which changes at re-configuration 4012 uint32_t maxSampleRate = mSampleRate * AUDIO_RESAMPLER_DOWN_RATIO_MAX; 4013 uint32_t reqSampleRate = track->mAudioTrackServerProxy->getSampleRate(); 4014 if (reqSampleRate == 0) { 4015 reqSampleRate = mSampleRate; 4016 } else if (reqSampleRate > maxSampleRate) { 4017 reqSampleRate = maxSampleRate; 4018 } 4019 mAudioMixer->setParameter( 4020 name, 4021 AudioMixer::RESAMPLE, 4022 AudioMixer::SAMPLE_RATE, 4023 (void *)(uintptr_t)reqSampleRate); 4024 4025 AudioPlaybackRate playbackRate = track->mAudioTrackServerProxy->getPlaybackRate(); 4026 mAudioMixer->setParameter( 4027 name, 4028 AudioMixer::TIMESTRETCH, 4029 AudioMixer::PLAYBACK_RATE, 4030 &playbackRate); 4031 4032 /* 4033 * Select the appropriate output buffer for the track. 4034 * 4035 * Tracks with effects go into their own effects chain buffer 4036 * and from there into either mEffectBuffer or mSinkBuffer. 4037 * 4038 * Other tracks can use mMixerBuffer for higher precision 4039 * channel accumulation. If this buffer is enabled 4040 * (mMixerBufferEnabled true), then selected tracks will accumulate 4041 * into it. 4042 * 4043 */ 4044 if (mMixerBufferEnabled 4045 && (track->mainBuffer() == mSinkBuffer 4046 || track->mainBuffer() == mMixerBuffer)) { 4047 mAudioMixer->setParameter( 4048 name, 4049 AudioMixer::TRACK, 4050 AudioMixer::MIXER_FORMAT, (void *)mMixerBufferFormat); 4051 mAudioMixer->setParameter( 4052 name, 4053 AudioMixer::TRACK, 4054 AudioMixer::MAIN_BUFFER, (void *)mMixerBuffer); 4055 // TODO: override track->mainBuffer()? 4056 mMixerBufferValid = true; 4057 } else { 4058 mAudioMixer->setParameter( 4059 name, 4060 AudioMixer::TRACK, 4061 AudioMixer::MIXER_FORMAT, (void *)AUDIO_FORMAT_PCM_16_BIT); 4062 mAudioMixer->setParameter( 4063 name, 4064 AudioMixer::TRACK, 4065 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 4066 } 4067 mAudioMixer->setParameter( 4068 name, 4069 AudioMixer::TRACK, 4070 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 4071 4072 // reset retry count 4073 track->mRetryCount = kMaxTrackRetries; 4074 4075 // If one track is ready, set the mixer ready if: 4076 // - the mixer was not ready during previous round OR 4077 // - no other track is not ready 4078 if (mMixerStatusIgnoringFastTracks != MIXER_TRACKS_READY || 4079 mixerStatus != MIXER_TRACKS_ENABLED) { 4080 mixerStatus = MIXER_TRACKS_READY; 4081 } 4082 } else { 4083 if (framesReady < desiredFrames && !track->isStopped() && !track->isPaused()) { 4084 ALOGV("track(%p) underrun, framesReady(%zu) < framesDesired(%zd)", 4085 track, framesReady, desiredFrames); 4086 track->mAudioTrackServerProxy->tallyUnderrunFrames(desiredFrames); 4087 } 4088 // clear effect chain input buffer if an active track underruns to avoid sending 4089 // previous audio buffer again to effects 4090 chain = getEffectChain_l(track->sessionId()); 4091 if (chain != 0) { 4092 chain->clearInputBuffer(); 4093 } 4094 4095 ALOGVV("track %d s=%08x [NOT READY] on thread %p", name, cblk->mServer, this); 4096 if ((track->sharedBuffer() != 0) || track->isTerminated() || 4097 track->isStopped() || track->isPaused()) { 4098 // We have consumed all the buffers of this track. 4099 // Remove it from the list of active tracks. 4100 // TODO: use actual buffer filling status instead of latency when available from 4101 // audio HAL 4102 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 4103 size_t framesWritten = mBytesWritten / mFrameSize; 4104 if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) { 4105 if (track->isStopped()) { 4106 track->reset(); 4107 } 4108 tracksToRemove->add(track); 4109 } 4110 } else { 4111 // No buffers for this track. Give it a few chances to 4112 // fill a buffer, then remove it from active list. 4113 if (--(track->mRetryCount) <= 0) { 4114 ALOGI("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 4115 tracksToRemove->add(track); 4116 // indicate to client process that the track was disabled because of underrun; 4117 // it will then automatically call start() when data is available 4118 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 4119 // If one track is not ready, mark the mixer also not ready if: 4120 // - the mixer was ready during previous round OR 4121 // - no other track is ready 4122 } else if (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY || 4123 mixerStatus != MIXER_TRACKS_READY) { 4124 mixerStatus = MIXER_TRACKS_ENABLED; 4125 } 4126 } 4127 mAudioMixer->disable(name); 4128 } 4129 4130 } // local variable scope to avoid goto warning 4131track_is_ready: ; 4132 4133 } 4134 4135 // Push the new FastMixer state if necessary 4136 bool pauseAudioWatchdog = false; 4137 if (didModify) { 4138 state->mFastTracksGen++; 4139 // if the fast mixer was active, but now there are no fast tracks, then put it in cold idle 4140 if (kUseFastMixer == FastMixer_Dynamic && 4141 state->mCommand == FastMixerState::MIX_WRITE && state->mTrackMask <= 1) { 4142 state->mCommand = FastMixerState::COLD_IDLE; 4143 state->mColdFutexAddr = &mFastMixerFutex; 4144 state->mColdGen++; 4145 mFastMixerFutex = 0; 4146 if (kUseFastMixer == FastMixer_Dynamic) { 4147 mNormalSink = mOutputSink; 4148 } 4149 // If we go into cold idle, need to wait for acknowledgement 4150 // so that fast mixer stops doing I/O. 4151 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 4152 pauseAudioWatchdog = true; 4153 } 4154 } 4155 if (sq != NULL) { 4156 sq->end(didModify); 4157 sq->push(block); 4158 } 4159#ifdef AUDIO_WATCHDOG 4160 if (pauseAudioWatchdog && mAudioWatchdog != 0) { 4161 mAudioWatchdog->pause(); 4162 } 4163#endif 4164 4165 // Now perform the deferred reset on fast tracks that have stopped 4166 while (resetMask != 0) { 4167 size_t i = __builtin_ctz(resetMask); 4168 ALOG_ASSERT(i < count); 4169 resetMask &= ~(1 << i); 4170 sp<Track> t = mActiveTracks[i].promote(); 4171 if (t == 0) { 4172 continue; 4173 } 4174 Track* track = t.get(); 4175 ALOG_ASSERT(track->isFastTrack() && track->isStopped()); 4176 track->reset(); 4177 } 4178 4179 // remove all the tracks that need to be... 4180 removeTracks_l(*tracksToRemove); 4181 4182 if (getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX) != 0) { 4183 mEffectBufferValid = true; 4184 } 4185 4186 if (mEffectBufferValid) { 4187 // as long as there are effects we should clear the effects buffer, to avoid 4188 // passing a non-clean buffer to the effect chain 4189 memset(mEffectBuffer, 0, mEffectBufferSize); 4190 } 4191 // sink or mix buffer must be cleared if all tracks are connected to an 4192 // effect chain as in this case the mixer will not write to the sink or mix buffer 4193 // and track effects will accumulate into it 4194 if ((mBytesRemaining == 0) && ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 4195 (mixedTracks == 0 && fastTracks > 0))) { 4196 // FIXME as a performance optimization, should remember previous zero status 4197 if (mMixerBufferValid) { 4198 memset(mMixerBuffer, 0, mMixerBufferSize); 4199 // TODO: In testing, mSinkBuffer below need not be cleared because 4200 // the PlaybackThread::threadLoop() copies mMixerBuffer into mSinkBuffer 4201 // after mixing. 4202 // 4203 // To enforce this guarantee: 4204 // ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 4205 // (mixedTracks == 0 && fastTracks > 0)) 4206 // must imply MIXER_TRACKS_READY. 4207 // Later, we may clear buffers regardless, and skip much of this logic. 4208 } 4209 // FIXME as a performance optimization, should remember previous zero status 4210 memset(mSinkBuffer, 0, mNormalFrameCount * mFrameSize); 4211 } 4212 4213 // if any fast tracks, then status is ready 4214 mMixerStatusIgnoringFastTracks = mixerStatus; 4215 if (fastTracks > 0) { 4216 mixerStatus = MIXER_TRACKS_READY; 4217 } 4218 return mixerStatus; 4219} 4220 4221// getTrackName_l() must be called with ThreadBase::mLock held 4222int AudioFlinger::MixerThread::getTrackName_l(audio_channel_mask_t channelMask, 4223 audio_format_t format, int sessionId) 4224{ 4225 return mAudioMixer->getTrackName(channelMask, format, sessionId); 4226} 4227 4228// deleteTrackName_l() must be called with ThreadBase::mLock held 4229void AudioFlinger::MixerThread::deleteTrackName_l(int name) 4230{ 4231 ALOGV("remove track (%d) and delete from mixer", name); 4232 mAudioMixer->deleteTrackName(name); 4233} 4234 4235// checkForNewParameter_l() must be called with ThreadBase::mLock held 4236bool AudioFlinger::MixerThread::checkForNewParameter_l(const String8& keyValuePair, 4237 status_t& status) 4238{ 4239 bool reconfig = false; 4240 4241 status = NO_ERROR; 4242 4243 // if !&IDLE, holds the FastMixer state to restore after new parameters processed 4244 FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE; 4245 if (mFastMixer != 0) { 4246 FastMixerStateQueue *sq = mFastMixer->sq(); 4247 FastMixerState *state = sq->begin(); 4248 if (!(state->mCommand & FastMixerState::IDLE)) { 4249 previousCommand = state->mCommand; 4250 state->mCommand = FastMixerState::HOT_IDLE; 4251 sq->end(); 4252 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 4253 } else { 4254 sq->end(false /*didModify*/); 4255 } 4256 } 4257 4258 AudioParameter param = AudioParameter(keyValuePair); 4259 int value; 4260 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 4261 reconfig = true; 4262 } 4263 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 4264 if (!isValidPcmSinkFormat((audio_format_t) value)) { 4265 status = BAD_VALUE; 4266 } else { 4267 // no need to save value, since it's constant 4268 reconfig = true; 4269 } 4270 } 4271 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 4272 if (!isValidPcmSinkChannelMask((audio_channel_mask_t) value)) { 4273 status = BAD_VALUE; 4274 } else { 4275 // no need to save value, since it's constant 4276 reconfig = true; 4277 } 4278 } 4279 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 4280 // do not accept frame count changes if tracks are open as the track buffer 4281 // size depends on frame count and correct behavior would not be guaranteed 4282 // if frame count is changed after track creation 4283 if (!mTracks.isEmpty()) { 4284 status = INVALID_OPERATION; 4285 } else { 4286 reconfig = true; 4287 } 4288 } 4289 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 4290#ifdef ADD_BATTERY_DATA 4291 // when changing the audio output device, call addBatteryData to notify 4292 // the change 4293 if (mOutDevice != value) { 4294 uint32_t params = 0; 4295 // check whether speaker is on 4296 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 4297 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 4298 } 4299 4300 audio_devices_t deviceWithoutSpeaker 4301 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 4302 // check if any other device (except speaker) is on 4303 if (value & deviceWithoutSpeaker) { 4304 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 4305 } 4306 4307 if (params != 0) { 4308 addBatteryData(params); 4309 } 4310 } 4311#endif 4312 4313 // forward device change to effects that have requested to be 4314 // aware of attached audio device. 4315 if (value != AUDIO_DEVICE_NONE) { 4316 mOutDevice = value; 4317 for (size_t i = 0; i < mEffectChains.size(); i++) { 4318 mEffectChains[i]->setDevice_l(mOutDevice); 4319 } 4320 } 4321 } 4322 4323 if (status == NO_ERROR) { 4324 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 4325 keyValuePair.string()); 4326 if (!mStandby && status == INVALID_OPERATION) { 4327 mOutput->standby(); 4328 mStandby = true; 4329 mBytesWritten = 0; 4330 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 4331 keyValuePair.string()); 4332 } 4333 if (status == NO_ERROR && reconfig) { 4334 readOutputParameters_l(); 4335 delete mAudioMixer; 4336 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 4337 for (size_t i = 0; i < mTracks.size() ; i++) { 4338 int name = getTrackName_l(mTracks[i]->mChannelMask, 4339 mTracks[i]->mFormat, mTracks[i]->mSessionId); 4340 if (name < 0) { 4341 break; 4342 } 4343 mTracks[i]->mName = name; 4344 } 4345 sendIoConfigEvent_l(AUDIO_OUTPUT_CONFIG_CHANGED); 4346 } 4347 } 4348 4349 if (!(previousCommand & FastMixerState::IDLE)) { 4350 ALOG_ASSERT(mFastMixer != 0); 4351 FastMixerStateQueue *sq = mFastMixer->sq(); 4352 FastMixerState *state = sq->begin(); 4353 ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE); 4354 state->mCommand = previousCommand; 4355 sq->end(); 4356 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 4357 } 4358 4359 return reconfig; 4360} 4361 4362 4363void AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 4364{ 4365 const size_t SIZE = 256; 4366 char buffer[SIZE]; 4367 String8 result; 4368 4369 PlaybackThread::dumpInternals(fd, args); 4370 dprintf(fd, " Thread throttle time (msecs): %u\n", mThreadThrottleTimeMs); 4371 dprintf(fd, " AudioMixer tracks: 0x%08x\n", mAudioMixer->trackNames()); 4372 4373 // Make a non-atomic copy of fast mixer dump state so it won't change underneath us 4374 const FastMixerDumpState copy(mFastMixerDumpState); 4375 copy.dump(fd); 4376 4377#ifdef STATE_QUEUE_DUMP 4378 // Similar for state queue 4379 StateQueueObserverDump observerCopy = mStateQueueObserverDump; 4380 observerCopy.dump(fd); 4381 StateQueueMutatorDump mutatorCopy = mStateQueueMutatorDump; 4382 mutatorCopy.dump(fd); 4383#endif 4384 4385#ifdef TEE_SINK 4386 // Write the tee output to a .wav file 4387 dumpTee(fd, mTeeSource, mId); 4388#endif 4389 4390#ifdef AUDIO_WATCHDOG 4391 if (mAudioWatchdog != 0) { 4392 // Make a non-atomic copy of audio watchdog dump so it won't change underneath us 4393 AudioWatchdogDump wdCopy = mAudioWatchdogDump; 4394 wdCopy.dump(fd); 4395 } 4396#endif 4397} 4398 4399uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const 4400{ 4401 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000) / 2; 4402} 4403 4404uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() const 4405{ 4406 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000); 4407} 4408 4409void AudioFlinger::MixerThread::cacheParameters_l() 4410{ 4411 PlaybackThread::cacheParameters_l(); 4412 4413 // FIXME: Relaxed timing because of a certain device that can't meet latency 4414 // Should be reduced to 2x after the vendor fixes the driver issue 4415 // increase threshold again due to low power audio mode. The way this warning 4416 // threshold is calculated and its usefulness should be reconsidered anyway. 4417 maxPeriod = seconds(mNormalFrameCount) / mSampleRate * 15; 4418} 4419 4420// ---------------------------------------------------------------------------- 4421 4422AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 4423 AudioStreamOut* output, audio_io_handle_t id, audio_devices_t device, bool systemReady) 4424 : PlaybackThread(audioFlinger, output, id, device, DIRECT, systemReady) 4425 // mLeftVolFloat, mRightVolFloat 4426{ 4427} 4428 4429AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 4430 AudioStreamOut* output, audio_io_handle_t id, uint32_t device, 4431 ThreadBase::type_t type, bool systemReady) 4432 : PlaybackThread(audioFlinger, output, id, device, type, systemReady) 4433 // mLeftVolFloat, mRightVolFloat 4434{ 4435} 4436 4437AudioFlinger::DirectOutputThread::~DirectOutputThread() 4438{ 4439} 4440 4441void AudioFlinger::DirectOutputThread::processVolume_l(Track *track, bool lastTrack) 4442{ 4443 audio_track_cblk_t* cblk = track->cblk(); 4444 float left, right; 4445 4446 if (mMasterMute || mStreamTypes[track->streamType()].mute) { 4447 left = right = 0; 4448 } else { 4449 float typeVolume = mStreamTypes[track->streamType()].volume; 4450 float v = mMasterVolume * typeVolume; 4451 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 4452 gain_minifloat_packed_t vlr = proxy->getVolumeLR(); 4453 left = float_from_gain(gain_minifloat_unpack_left(vlr)); 4454 if (left > GAIN_FLOAT_UNITY) { 4455 left = GAIN_FLOAT_UNITY; 4456 } 4457 left *= v; 4458 right = float_from_gain(gain_minifloat_unpack_right(vlr)); 4459 if (right > GAIN_FLOAT_UNITY) { 4460 right = GAIN_FLOAT_UNITY; 4461 } 4462 right *= v; 4463 } 4464 4465 if (lastTrack) { 4466 if (left != mLeftVolFloat || right != mRightVolFloat) { 4467 mLeftVolFloat = left; 4468 mRightVolFloat = right; 4469 4470 // Convert volumes from float to 8.24 4471 uint32_t vl = (uint32_t)(left * (1 << 24)); 4472 uint32_t vr = (uint32_t)(right * (1 << 24)); 4473 4474 // Delegate volume control to effect in track effect chain if needed 4475 // only one effect chain can be present on DirectOutputThread, so if 4476 // there is one, the track is connected to it 4477 if (!mEffectChains.isEmpty()) { 4478 mEffectChains[0]->setVolume_l(&vl, &vr); 4479 left = (float)vl / (1 << 24); 4480 right = (float)vr / (1 << 24); 4481 } 4482 if (mOutput->stream->set_volume) { 4483 mOutput->stream->set_volume(mOutput->stream, left, right); 4484 } 4485 } 4486 } 4487} 4488 4489void AudioFlinger::DirectOutputThread::onAddNewTrack_l() 4490{ 4491 sp<Track> previousTrack = mPreviousTrack.promote(); 4492 sp<Track> latestTrack = mLatestActiveTrack.promote(); 4493 4494 if (previousTrack != 0 && latestTrack != 0) { 4495 if (mType == DIRECT) { 4496 if (previousTrack.get() != latestTrack.get()) { 4497 mFlushPending = true; 4498 } 4499 } else /* mType == OFFLOAD */ { 4500 if (previousTrack->sessionId() != latestTrack->sessionId()) { 4501 mFlushPending = true; 4502 } 4503 } 4504 } 4505 PlaybackThread::onAddNewTrack_l(); 4506} 4507 4508AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l( 4509 Vector< sp<Track> > *tracksToRemove 4510) 4511{ 4512 size_t count = mActiveTracks.size(); 4513 mixer_state mixerStatus = MIXER_IDLE; 4514 bool doHwPause = false; 4515 bool doHwResume = false; 4516 4517 // find out which tracks need to be processed 4518 for (size_t i = 0; i < count; i++) { 4519 sp<Track> t = mActiveTracks[i].promote(); 4520 // The track died recently 4521 if (t == 0) { 4522 continue; 4523 } 4524 4525 if (t->isInvalid()) { 4526 ALOGW("An invalidated track shouldn't be in active list"); 4527 tracksToRemove->add(t); 4528 continue; 4529 } 4530 4531 Track* const track = t.get(); 4532 audio_track_cblk_t* cblk = track->cblk(); 4533 // Only consider last track started for volume and mixer state control. 4534 // In theory an older track could underrun and restart after the new one starts 4535 // but as we only care about the transition phase between two tracks on a 4536 // direct output, it is not a problem to ignore the underrun case. 4537 sp<Track> l = mLatestActiveTrack.promote(); 4538 bool last = l.get() == track; 4539 4540 if (track->isPausing()) { 4541 track->setPaused(); 4542 if (mHwSupportsPause && last && !mHwPaused) { 4543 doHwPause = true; 4544 mHwPaused = true; 4545 } 4546 tracksToRemove->add(track); 4547 } else if (track->isFlushPending()) { 4548 track->flushAck(); 4549 if (last) { 4550 mFlushPending = true; 4551 } 4552 } else if (track->isResumePending()) { 4553 track->resumeAck(); 4554 if (last && mHwPaused) { 4555 doHwResume = true; 4556 mHwPaused = false; 4557 } 4558 } 4559 4560 // The first time a track is added we wait 4561 // for all its buffers to be filled before processing it. 4562 // Allow draining the buffer in case the client 4563 // app does not call stop() and relies on underrun to stop: 4564 // hence the test on (track->mRetryCount > 1). 4565 // If retryCount<=1 then track is about to underrun and be removed. 4566 // Do not use a high threshold for compressed audio. 4567 uint32_t minFrames; 4568 if ((track->sharedBuffer() == 0) && !track->isStopping_1() && !track->isPausing() 4569 && (track->mRetryCount > 1) && audio_is_linear_pcm(mFormat)) { 4570 minFrames = mNormalFrameCount; 4571 } else { 4572 minFrames = 1; 4573 } 4574 4575 if ((track->framesReady() >= minFrames) && track->isReady() && !track->isPaused() && 4576 !track->isStopping_2() && !track->isStopped()) 4577 { 4578 ALOGVV("track %d s=%08x [OK]", track->name(), cblk->mServer); 4579 4580 if (track->mFillingUpStatus == Track::FS_FILLED) { 4581 track->mFillingUpStatus = Track::FS_ACTIVE; 4582 // make sure processVolume_l() will apply new volume even if 0 4583 mLeftVolFloat = mRightVolFloat = -1.0; 4584 if (!mHwSupportsPause) { 4585 track->resumeAck(); 4586 } 4587 } 4588 4589 // compute volume for this track 4590 processVolume_l(track, last); 4591 if (last) { 4592 sp<Track> previousTrack = mPreviousTrack.promote(); 4593 if (previousTrack != 0) { 4594 if (track != previousTrack.get()) { 4595 // Flush any data still being written from last track 4596 mBytesRemaining = 0; 4597 // Invalidate previous track to force a seek when resuming. 4598 previousTrack->invalidate(); 4599 } 4600 } 4601 mPreviousTrack = track; 4602 4603 // reset retry count 4604 track->mRetryCount = kMaxTrackRetriesDirect; 4605 mActiveTrack = t; 4606 mixerStatus = MIXER_TRACKS_READY; 4607 if (mHwPaused) { 4608 doHwResume = true; 4609 mHwPaused = false; 4610 } 4611 } 4612 } else { 4613 // clear effect chain input buffer if the last active track started underruns 4614 // to avoid sending previous audio buffer again to effects 4615 if (!mEffectChains.isEmpty() && last) { 4616 mEffectChains[0]->clearInputBuffer(); 4617 } 4618 if (track->isStopping_1()) { 4619 track->mState = TrackBase::STOPPING_2; 4620 if (last && mHwPaused) { 4621 doHwResume = true; 4622 mHwPaused = false; 4623 } 4624 } 4625 if ((track->sharedBuffer() != 0) || track->isStopped() || 4626 track->isStopping_2() || track->isPaused()) { 4627 // We have consumed all the buffers of this track. 4628 // Remove it from the list of active tracks. 4629 size_t audioHALFrames; 4630 if (audio_is_linear_pcm(mFormat)) { 4631 audioHALFrames = (latency_l() * mSampleRate) / 1000; 4632 } else { 4633 audioHALFrames = 0; 4634 } 4635 4636 size_t framesWritten = mBytesWritten / mFrameSize; 4637 if (mStandby || !last || 4638 track->presentationComplete(framesWritten, audioHALFrames)) { 4639 if (track->isStopping_2()) { 4640 track->mState = TrackBase::STOPPED; 4641 } 4642 if (track->isStopped()) { 4643 track->reset(); 4644 } 4645 tracksToRemove->add(track); 4646 } 4647 } else { 4648 // No buffers for this track. Give it a few chances to 4649 // fill a buffer, then remove it from active list. 4650 // Only consider last track started for mixer state control 4651 if (--(track->mRetryCount) <= 0) { 4652 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 4653 tracksToRemove->add(track); 4654 // indicate to client process that the track was disabled because of underrun; 4655 // it will then automatically call start() when data is available 4656 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 4657 } else if (last) { 4658 ALOGW("pause because of UNDERRUN, framesReady = %zu," 4659 "minFrames = %u, mFormat = %#x", 4660 track->framesReady(), minFrames, mFormat); 4661 mixerStatus = MIXER_TRACKS_ENABLED; 4662 if (mHwSupportsPause && !mHwPaused && !mStandby) { 4663 doHwPause = true; 4664 mHwPaused = true; 4665 } 4666 } 4667 } 4668 } 4669 } 4670 4671 // if an active track did not command a flush, check for pending flush on stopped tracks 4672 if (!mFlushPending) { 4673 for (size_t i = 0; i < mTracks.size(); i++) { 4674 if (mTracks[i]->isFlushPending()) { 4675 mTracks[i]->flushAck(); 4676 mFlushPending = true; 4677 } 4678 } 4679 } 4680 4681 // make sure the pause/flush/resume sequence is executed in the right order. 4682 // If a flush is pending and a track is active but the HW is not paused, force a HW pause 4683 // before flush and then resume HW. This can happen in case of pause/flush/resume 4684 // if resume is received before pause is executed. 4685 if (mHwSupportsPause && !mStandby && 4686 (doHwPause || (mFlushPending && !mHwPaused && (count != 0)))) { 4687 mOutput->stream->pause(mOutput->stream); 4688 } 4689 if (mFlushPending) { 4690 flushHw_l(); 4691 } 4692 if (mHwSupportsPause && !mStandby && doHwResume) { 4693 mOutput->stream->resume(mOutput->stream); 4694 } 4695 // remove all the tracks that need to be... 4696 removeTracks_l(*tracksToRemove); 4697 4698 return mixerStatus; 4699} 4700 4701void AudioFlinger::DirectOutputThread::threadLoop_mix() 4702{ 4703 size_t frameCount = mFrameCount; 4704 int8_t *curBuf = (int8_t *)mSinkBuffer; 4705 // output audio to hardware 4706 while (frameCount) { 4707 AudioBufferProvider::Buffer buffer; 4708 buffer.frameCount = frameCount; 4709 status_t status = mActiveTrack->getNextBuffer(&buffer); 4710 if (status != NO_ERROR || buffer.raw == NULL) { 4711 memset(curBuf, 0, frameCount * mFrameSize); 4712 break; 4713 } 4714 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 4715 frameCount -= buffer.frameCount; 4716 curBuf += buffer.frameCount * mFrameSize; 4717 mActiveTrack->releaseBuffer(&buffer); 4718 } 4719 mCurrentWriteLength = curBuf - (int8_t *)mSinkBuffer; 4720 mSleepTimeUs = 0; 4721 mStandbyTimeNs = systemTime() + mStandbyDelayNs; 4722 mActiveTrack.clear(); 4723} 4724 4725void AudioFlinger::DirectOutputThread::threadLoop_sleepTime() 4726{ 4727 // do not write to HAL when paused 4728 if (mHwPaused || (usesHwAvSync() && mStandby)) { 4729 mSleepTimeUs = mIdleSleepTimeUs; 4730 return; 4731 } 4732 if (mSleepTimeUs == 0) { 4733 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4734 mSleepTimeUs = mActiveSleepTimeUs; 4735 } else { 4736 mSleepTimeUs = mIdleSleepTimeUs; 4737 } 4738 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 4739 memset(mSinkBuffer, 0, mFrameCount * mFrameSize); 4740 mSleepTimeUs = 0; 4741 } 4742} 4743 4744void AudioFlinger::DirectOutputThread::threadLoop_exit() 4745{ 4746 { 4747 Mutex::Autolock _l(mLock); 4748 for (size_t i = 0; i < mTracks.size(); i++) { 4749 if (mTracks[i]->isFlushPending()) { 4750 mTracks[i]->flushAck(); 4751 mFlushPending = true; 4752 } 4753 } 4754 if (mFlushPending) { 4755 flushHw_l(); 4756 } 4757 } 4758 PlaybackThread::threadLoop_exit(); 4759} 4760 4761// must be called with thread mutex locked 4762bool AudioFlinger::DirectOutputThread::shouldStandby_l() 4763{ 4764 bool trackPaused = false; 4765 bool trackStopped = false; 4766 4767 // do not put the HAL in standby when paused. AwesomePlayer clear the offloaded AudioTrack 4768 // after a timeout and we will enter standby then. 4769 if (mTracks.size() > 0) { 4770 trackPaused = mTracks[mTracks.size() - 1]->isPaused(); 4771 trackStopped = mTracks[mTracks.size() - 1]->isStopped() || 4772 mTracks[mTracks.size() - 1]->mState == TrackBase::IDLE; 4773 } 4774 4775 return !mStandby && !(trackPaused || (mHwPaused && !trackStopped)); 4776} 4777 4778// getTrackName_l() must be called with ThreadBase::mLock held 4779int AudioFlinger::DirectOutputThread::getTrackName_l(audio_channel_mask_t channelMask __unused, 4780 audio_format_t format __unused, int sessionId __unused) 4781{ 4782 return 0; 4783} 4784 4785// deleteTrackName_l() must be called with ThreadBase::mLock held 4786void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name __unused) 4787{ 4788} 4789 4790// checkForNewParameter_l() must be called with ThreadBase::mLock held 4791bool AudioFlinger::DirectOutputThread::checkForNewParameter_l(const String8& keyValuePair, 4792 status_t& status) 4793{ 4794 bool reconfig = false; 4795 4796 status = NO_ERROR; 4797 4798 AudioParameter param = AudioParameter(keyValuePair); 4799 int value; 4800 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 4801 // forward device change to effects that have requested to be 4802 // aware of attached audio device. 4803 if (value != AUDIO_DEVICE_NONE) { 4804 mOutDevice = value; 4805 for (size_t i = 0; i < mEffectChains.size(); i++) { 4806 mEffectChains[i]->setDevice_l(mOutDevice); 4807 } 4808 } 4809 } 4810 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 4811 // do not accept frame count changes if tracks are open as the track buffer 4812 // size depends on frame count and correct behavior would not be garantied 4813 // if frame count is changed after track creation 4814 if (!mTracks.isEmpty()) { 4815 status = INVALID_OPERATION; 4816 } else { 4817 reconfig = true; 4818 } 4819 } 4820 if (status == NO_ERROR) { 4821 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 4822 keyValuePair.string()); 4823 if (!mStandby && status == INVALID_OPERATION) { 4824 mOutput->standby(); 4825 mStandby = true; 4826 mBytesWritten = 0; 4827 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 4828 keyValuePair.string()); 4829 } 4830 if (status == NO_ERROR && reconfig) { 4831 readOutputParameters_l(); 4832 sendIoConfigEvent_l(AUDIO_OUTPUT_CONFIG_CHANGED); 4833 } 4834 } 4835 4836 return reconfig; 4837} 4838 4839uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() const 4840{ 4841 uint32_t time; 4842 if (audio_is_linear_pcm(mFormat)) { 4843 time = PlaybackThread::activeSleepTimeUs(); 4844 } else { 4845 time = 10000; 4846 } 4847 return time; 4848} 4849 4850uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() const 4851{ 4852 uint32_t time; 4853 if (audio_is_linear_pcm(mFormat)) { 4854 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 4855 } else { 4856 time = 10000; 4857 } 4858 return time; 4859} 4860 4861uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() const 4862{ 4863 uint32_t time; 4864 if (audio_is_linear_pcm(mFormat)) { 4865 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 4866 } else { 4867 time = 10000; 4868 } 4869 return time; 4870} 4871 4872void AudioFlinger::DirectOutputThread::cacheParameters_l() 4873{ 4874 PlaybackThread::cacheParameters_l(); 4875 4876 // use shorter standby delay as on normal output to release 4877 // hardware resources as soon as possible 4878 // no delay on outputs with HW A/V sync 4879 if (usesHwAvSync()) { 4880 mStandbyDelayNs = 0; 4881 } else if ((mType == OFFLOAD) && !audio_is_linear_pcm(mFormat)) { 4882 mStandbyDelayNs = kOffloadStandbyDelayNs; 4883 } else { 4884 mStandbyDelayNs = microseconds(mActiveSleepTimeUs*2); 4885 } 4886} 4887 4888void AudioFlinger::DirectOutputThread::flushHw_l() 4889{ 4890 mOutput->flush(); 4891 mHwPaused = false; 4892 mFlushPending = false; 4893} 4894 4895// ---------------------------------------------------------------------------- 4896 4897AudioFlinger::AsyncCallbackThread::AsyncCallbackThread( 4898 const wp<AudioFlinger::PlaybackThread>& playbackThread) 4899 : Thread(false /*canCallJava*/), 4900 mPlaybackThread(playbackThread), 4901 mWriteAckSequence(0), 4902 mDrainSequence(0) 4903{ 4904} 4905 4906AudioFlinger::AsyncCallbackThread::~AsyncCallbackThread() 4907{ 4908} 4909 4910void AudioFlinger::AsyncCallbackThread::onFirstRef() 4911{ 4912 run("Offload Cbk", ANDROID_PRIORITY_URGENT_AUDIO); 4913} 4914 4915bool AudioFlinger::AsyncCallbackThread::threadLoop() 4916{ 4917 while (!exitPending()) { 4918 uint32_t writeAckSequence; 4919 uint32_t drainSequence; 4920 4921 { 4922 Mutex::Autolock _l(mLock); 4923 while (!((mWriteAckSequence & 1) || 4924 (mDrainSequence & 1) || 4925 exitPending())) { 4926 mWaitWorkCV.wait(mLock); 4927 } 4928 4929 if (exitPending()) { 4930 break; 4931 } 4932 ALOGV("AsyncCallbackThread mWriteAckSequence %d mDrainSequence %d", 4933 mWriteAckSequence, mDrainSequence); 4934 writeAckSequence = mWriteAckSequence; 4935 mWriteAckSequence &= ~1; 4936 drainSequence = mDrainSequence; 4937 mDrainSequence &= ~1; 4938 } 4939 { 4940 sp<AudioFlinger::PlaybackThread> playbackThread = mPlaybackThread.promote(); 4941 if (playbackThread != 0) { 4942 if (writeAckSequence & 1) { 4943 playbackThread->resetWriteBlocked(writeAckSequence >> 1); 4944 } 4945 if (drainSequence & 1) { 4946 playbackThread->resetDraining(drainSequence >> 1); 4947 } 4948 } 4949 } 4950 } 4951 return false; 4952} 4953 4954void AudioFlinger::AsyncCallbackThread::exit() 4955{ 4956 ALOGV("AsyncCallbackThread::exit"); 4957 Mutex::Autolock _l(mLock); 4958 requestExit(); 4959 mWaitWorkCV.broadcast(); 4960} 4961 4962void AudioFlinger::AsyncCallbackThread::setWriteBlocked(uint32_t sequence) 4963{ 4964 Mutex::Autolock _l(mLock); 4965 // bit 0 is cleared 4966 mWriteAckSequence = sequence << 1; 4967} 4968 4969void AudioFlinger::AsyncCallbackThread::resetWriteBlocked() 4970{ 4971 Mutex::Autolock _l(mLock); 4972 // ignore unexpected callbacks 4973 if (mWriteAckSequence & 2) { 4974 mWriteAckSequence |= 1; 4975 mWaitWorkCV.signal(); 4976 } 4977} 4978 4979void AudioFlinger::AsyncCallbackThread::setDraining(uint32_t sequence) 4980{ 4981 Mutex::Autolock _l(mLock); 4982 // bit 0 is cleared 4983 mDrainSequence = sequence << 1; 4984} 4985 4986void AudioFlinger::AsyncCallbackThread::resetDraining() 4987{ 4988 Mutex::Autolock _l(mLock); 4989 // ignore unexpected callbacks 4990 if (mDrainSequence & 2) { 4991 mDrainSequence |= 1; 4992 mWaitWorkCV.signal(); 4993 } 4994} 4995 4996 4997// ---------------------------------------------------------------------------- 4998AudioFlinger::OffloadThread::OffloadThread(const sp<AudioFlinger>& audioFlinger, 4999 AudioStreamOut* output, audio_io_handle_t id, uint32_t device, bool systemReady) 5000 : DirectOutputThread(audioFlinger, output, id, device, OFFLOAD, systemReady), 5001 mPausedBytesRemaining(0) 5002{ 5003 //FIXME: mStandby should be set to true by ThreadBase constructor 5004 mStandby = true; 5005} 5006 5007void AudioFlinger::OffloadThread::threadLoop_exit() 5008{ 5009 if (mFlushPending || mHwPaused) { 5010 // If a flush is pending or track was paused, just discard buffered data 5011 flushHw_l(); 5012 } else { 5013 mMixerStatus = MIXER_DRAIN_ALL; 5014 threadLoop_drain(); 5015 } 5016 if (mUseAsyncWrite) { 5017 ALOG_ASSERT(mCallbackThread != 0); 5018 mCallbackThread->exit(); 5019 } 5020 PlaybackThread::threadLoop_exit(); 5021} 5022 5023AudioFlinger::PlaybackThread::mixer_state AudioFlinger::OffloadThread::prepareTracks_l( 5024 Vector< sp<Track> > *tracksToRemove 5025) 5026{ 5027 size_t count = mActiveTracks.size(); 5028 5029 mixer_state mixerStatus = MIXER_IDLE; 5030 bool doHwPause = false; 5031 bool doHwResume = false; 5032 5033 ALOGV("OffloadThread::prepareTracks_l active tracks %d", count); 5034 5035 // find out which tracks need to be processed 5036 for (size_t i = 0; i < count; i++) { 5037 sp<Track> t = mActiveTracks[i].promote(); 5038 // The track died recently 5039 if (t == 0) { 5040 continue; 5041 } 5042 Track* const track = t.get(); 5043 audio_track_cblk_t* cblk = track->cblk(); 5044 // Only consider last track started for volume and mixer state control. 5045 // In theory an older track could underrun and restart after the new one starts 5046 // but as we only care about the transition phase between two tracks on a 5047 // direct output, it is not a problem to ignore the underrun case. 5048 sp<Track> l = mLatestActiveTrack.promote(); 5049 bool last = l.get() == track; 5050 5051 if (track->isInvalid()) { 5052 ALOGW("An invalidated track shouldn't be in active list"); 5053 tracksToRemove->add(track); 5054 continue; 5055 } 5056 5057 if (track->mState == TrackBase::IDLE) { 5058 ALOGW("An idle track shouldn't be in active list"); 5059 continue; 5060 } 5061 5062 if (track->isPausing()) { 5063 track->setPaused(); 5064 if (last) { 5065 if (mHwSupportsPause && !mHwPaused) { 5066 doHwPause = true; 5067 mHwPaused = true; 5068 } 5069 // If we were part way through writing the mixbuffer to 5070 // the HAL we must save this until we resume 5071 // BUG - this will be wrong if a different track is made active, 5072 // in that case we want to discard the pending data in the 5073 // mixbuffer and tell the client to present it again when the 5074 // track is resumed 5075 mPausedWriteLength = mCurrentWriteLength; 5076 mPausedBytesRemaining = mBytesRemaining; 5077 mBytesRemaining = 0; // stop writing 5078 } 5079 tracksToRemove->add(track); 5080 } else if (track->isFlushPending()) { 5081 track->flushAck(); 5082 if (last) { 5083 mFlushPending = true; 5084 } 5085 } else if (track->isResumePending()){ 5086 track->resumeAck(); 5087 if (last) { 5088 if (mPausedBytesRemaining) { 5089 // Need to continue write that was interrupted 5090 mCurrentWriteLength = mPausedWriteLength; 5091 mBytesRemaining = mPausedBytesRemaining; 5092 mPausedBytesRemaining = 0; 5093 } 5094 if (mHwPaused) { 5095 doHwResume = true; 5096 mHwPaused = false; 5097 // threadLoop_mix() will handle the case that we need to 5098 // resume an interrupted write 5099 } 5100 // enable write to audio HAL 5101 mSleepTimeUs = 0; 5102 5103 // Do not handle new data in this iteration even if track->framesReady() 5104 mixerStatus = MIXER_TRACKS_ENABLED; 5105 } 5106 } else if (track->framesReady() && track->isReady() && 5107 !track->isPaused() && !track->isTerminated() && !track->isStopping_2()) { 5108 ALOGVV("OffloadThread: track %d s=%08x [OK]", track->name(), cblk->mServer); 5109 if (track->mFillingUpStatus == Track::FS_FILLED) { 5110 track->mFillingUpStatus = Track::FS_ACTIVE; 5111 // make sure processVolume_l() will apply new volume even if 0 5112 mLeftVolFloat = mRightVolFloat = -1.0; 5113 } 5114 5115 if (last) { 5116 sp<Track> previousTrack = mPreviousTrack.promote(); 5117 if (previousTrack != 0) { 5118 if (track != previousTrack.get()) { 5119 // Flush any data still being written from last track 5120 mBytesRemaining = 0; 5121 if (mPausedBytesRemaining) { 5122 // Last track was paused so we also need to flush saved 5123 // mixbuffer state and invalidate track so that it will 5124 // re-submit that unwritten data when it is next resumed 5125 mPausedBytesRemaining = 0; 5126 // Invalidate is a bit drastic - would be more efficient 5127 // to have a flag to tell client that some of the 5128 // previously written data was lost 5129 previousTrack->invalidate(); 5130 } 5131 // flush data already sent to the DSP if changing audio session as audio 5132 // comes from a different source. Also invalidate previous track to force a 5133 // seek when resuming. 5134 if (previousTrack->sessionId() != track->sessionId()) { 5135 previousTrack->invalidate(); 5136 } 5137 } 5138 } 5139 mPreviousTrack = track; 5140 // reset retry count 5141 track->mRetryCount = kMaxTrackRetriesOffload; 5142 mActiveTrack = t; 5143 mixerStatus = MIXER_TRACKS_READY; 5144 } 5145 } else { 5146 ALOGVV("OffloadThread: track %d s=%08x [NOT READY]", track->name(), cblk->mServer); 5147 if (track->isStopping_1()) { 5148 // Hardware buffer can hold a large amount of audio so we must 5149 // wait for all current track's data to drain before we say 5150 // that the track is stopped. 5151 if (mBytesRemaining == 0) { 5152 // Only start draining when all data in mixbuffer 5153 // has been written 5154 ALOGV("OffloadThread: underrun and STOPPING_1 -> draining, STOPPING_2"); 5155 track->mState = TrackBase::STOPPING_2; // so presentation completes after drain 5156 // do not drain if no data was ever sent to HAL (mStandby == true) 5157 if (last && !mStandby) { 5158 // do not modify drain sequence if we are already draining. This happens 5159 // when resuming from pause after drain. 5160 if ((mDrainSequence & 1) == 0) { 5161 mSleepTimeUs = 0; 5162 mStandbyTimeNs = systemTime() + mStandbyDelayNs; 5163 mixerStatus = MIXER_DRAIN_TRACK; 5164 mDrainSequence += 2; 5165 } 5166 if (mHwPaused) { 5167 // It is possible to move from PAUSED to STOPPING_1 without 5168 // a resume so we must ensure hardware is running 5169 doHwResume = true; 5170 mHwPaused = false; 5171 } 5172 } 5173 } 5174 } else if (track->isStopping_2()) { 5175 // Drain has completed or we are in standby, signal presentation complete 5176 if (!(mDrainSequence & 1) || !last || mStandby) { 5177 track->mState = TrackBase::STOPPED; 5178 size_t audioHALFrames = 5179 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 5180 size_t framesWritten = 5181 mBytesWritten / mOutput->getFrameSize(); 5182 track->presentationComplete(framesWritten, audioHALFrames); 5183 track->reset(); 5184 tracksToRemove->add(track); 5185 } 5186 } else { 5187 // No buffers for this track. Give it a few chances to 5188 // fill a buffer, then remove it from active list. 5189 if (--(track->mRetryCount) <= 0) { 5190 ALOGV("OffloadThread: BUFFER TIMEOUT: remove(%d) from active list", 5191 track->name()); 5192 tracksToRemove->add(track); 5193 // indicate to client process that the track was disabled because of underrun; 5194 // it will then automatically call start() when data is available 5195 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 5196 } else if (last){ 5197 mixerStatus = MIXER_TRACKS_ENABLED; 5198 } 5199 } 5200 } 5201 // compute volume for this track 5202 processVolume_l(track, last); 5203 } 5204 5205 // make sure the pause/flush/resume sequence is executed in the right order. 5206 // If a flush is pending and a track is active but the HW is not paused, force a HW pause 5207 // before flush and then resume HW. This can happen in case of pause/flush/resume 5208 // if resume is received before pause is executed. 5209 if (!mStandby && (doHwPause || (mFlushPending && !mHwPaused && (count != 0)))) { 5210 mOutput->stream->pause(mOutput->stream); 5211 } 5212 if (mFlushPending) { 5213 flushHw_l(); 5214 } 5215 if (!mStandby && doHwResume) { 5216 mOutput->stream->resume(mOutput->stream); 5217 } 5218 5219 // remove all the tracks that need to be... 5220 removeTracks_l(*tracksToRemove); 5221 5222 return mixerStatus; 5223} 5224 5225// must be called with thread mutex locked 5226bool AudioFlinger::OffloadThread::waitingAsyncCallback_l() 5227{ 5228 ALOGVV("waitingAsyncCallback_l mWriteAckSequence %d mDrainSequence %d", 5229 mWriteAckSequence, mDrainSequence); 5230 if (mUseAsyncWrite && ((mWriteAckSequence & 1) || (mDrainSequence & 1))) { 5231 return true; 5232 } 5233 return false; 5234} 5235 5236bool AudioFlinger::OffloadThread::waitingAsyncCallback() 5237{ 5238 Mutex::Autolock _l(mLock); 5239 return waitingAsyncCallback_l(); 5240} 5241 5242void AudioFlinger::OffloadThread::flushHw_l() 5243{ 5244 DirectOutputThread::flushHw_l(); 5245 // Flush anything still waiting in the mixbuffer 5246 mCurrentWriteLength = 0; 5247 mBytesRemaining = 0; 5248 mPausedWriteLength = 0; 5249 mPausedBytesRemaining = 0; 5250 5251 if (mUseAsyncWrite) { 5252 // discard any pending drain or write ack by incrementing sequence 5253 mWriteAckSequence = (mWriteAckSequence + 2) & ~1; 5254 mDrainSequence = (mDrainSequence + 2) & ~1; 5255 ALOG_ASSERT(mCallbackThread != 0); 5256 mCallbackThread->setWriteBlocked(mWriteAckSequence); 5257 mCallbackThread->setDraining(mDrainSequence); 5258 } 5259} 5260 5261// ---------------------------------------------------------------------------- 5262 5263AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 5264 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id, bool systemReady) 5265 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->outDevice(), 5266 systemReady, DUPLICATING), 5267 mWaitTimeMs(UINT_MAX) 5268{ 5269 addOutputTrack(mainThread); 5270} 5271 5272AudioFlinger::DuplicatingThread::~DuplicatingThread() 5273{ 5274 for (size_t i = 0; i < mOutputTracks.size(); i++) { 5275 mOutputTracks[i]->destroy(); 5276 } 5277} 5278 5279void AudioFlinger::DuplicatingThread::threadLoop_mix() 5280{ 5281 // mix buffers... 5282 if (outputsReady(outputTracks)) { 5283 mAudioMixer->process(AudioBufferProvider::kInvalidPTS); 5284 } else { 5285 if (mMixerBufferValid) { 5286 memset(mMixerBuffer, 0, mMixerBufferSize); 5287 } else { 5288 memset(mSinkBuffer, 0, mSinkBufferSize); 5289 } 5290 } 5291 mSleepTimeUs = 0; 5292 writeFrames = mNormalFrameCount; 5293 mCurrentWriteLength = mSinkBufferSize; 5294 mStandbyTimeNs = systemTime() + mStandbyDelayNs; 5295} 5296 5297void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() 5298{ 5299 if (mSleepTimeUs == 0) { 5300 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 5301 mSleepTimeUs = mActiveSleepTimeUs; 5302 } else { 5303 mSleepTimeUs = mIdleSleepTimeUs; 5304 } 5305 } else if (mBytesWritten != 0) { 5306 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 5307 writeFrames = mNormalFrameCount; 5308 memset(mSinkBuffer, 0, mSinkBufferSize); 5309 } else { 5310 // flush remaining overflow buffers in output tracks 5311 writeFrames = 0; 5312 } 5313 mSleepTimeUs = 0; 5314 } 5315} 5316 5317ssize_t AudioFlinger::DuplicatingThread::threadLoop_write() 5318{ 5319 for (size_t i = 0; i < outputTracks.size(); i++) { 5320 outputTracks[i]->write(mSinkBuffer, writeFrames); 5321 } 5322 mStandby = false; 5323 return (ssize_t)mSinkBufferSize; 5324} 5325 5326void AudioFlinger::DuplicatingThread::threadLoop_standby() 5327{ 5328 // DuplicatingThread implements standby by stopping all tracks 5329 for (size_t i = 0; i < outputTracks.size(); i++) { 5330 outputTracks[i]->stop(); 5331 } 5332} 5333 5334void AudioFlinger::DuplicatingThread::saveOutputTracks() 5335{ 5336 outputTracks = mOutputTracks; 5337} 5338 5339void AudioFlinger::DuplicatingThread::clearOutputTracks() 5340{ 5341 outputTracks.clear(); 5342} 5343 5344void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 5345{ 5346 Mutex::Autolock _l(mLock); 5347 // The downstream MixerThread consumes thread->frameCount() amount of frames per mix pass. 5348 // Adjust for thread->sampleRate() to determine minimum buffer frame count. 5349 // Then triple buffer because Threads do not run synchronously and may not be clock locked. 5350 const size_t frameCount = 5351 3 * sourceFramesNeeded(mSampleRate, thread->frameCount(), thread->sampleRate()); 5352 // TODO: Consider asynchronous sample rate conversion to handle clock disparity 5353 // from different OutputTracks and their associated MixerThreads (e.g. one may 5354 // nearly empty and the other may be dropping data). 5355 5356 sp<OutputTrack> outputTrack = new OutputTrack(thread, 5357 this, 5358 mSampleRate, 5359 mFormat, 5360 mChannelMask, 5361 frameCount, 5362 IPCThreadState::self()->getCallingUid()); 5363 if (outputTrack->cblk() != NULL) { 5364 thread->setStreamVolume(AUDIO_STREAM_PATCH, 1.0f); 5365 mOutputTracks.add(outputTrack); 5366 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack.get(), thread); 5367 updateWaitTime_l(); 5368 } 5369} 5370 5371void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 5372{ 5373 Mutex::Autolock _l(mLock); 5374 for (size_t i = 0; i < mOutputTracks.size(); i++) { 5375 if (mOutputTracks[i]->thread() == thread) { 5376 mOutputTracks[i]->destroy(); 5377 mOutputTracks.removeAt(i); 5378 updateWaitTime_l(); 5379 if (thread->getOutput() == mOutput) { 5380 mOutput = NULL; 5381 } 5382 return; 5383 } 5384 } 5385 ALOGV("removeOutputTrack(): unknown thread: %p", thread); 5386} 5387 5388// caller must hold mLock 5389void AudioFlinger::DuplicatingThread::updateWaitTime_l() 5390{ 5391 mWaitTimeMs = UINT_MAX; 5392 for (size_t i = 0; i < mOutputTracks.size(); i++) { 5393 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 5394 if (strong != 0) { 5395 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 5396 if (waitTimeMs < mWaitTimeMs) { 5397 mWaitTimeMs = waitTimeMs; 5398 } 5399 } 5400 } 5401} 5402 5403 5404bool AudioFlinger::DuplicatingThread::outputsReady( 5405 const SortedVector< sp<OutputTrack> > &outputTracks) 5406{ 5407 for (size_t i = 0; i < outputTracks.size(); i++) { 5408 sp<ThreadBase> thread = outputTracks[i]->thread().promote(); 5409 if (thread == 0) { 5410 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", 5411 outputTracks[i].get()); 5412 return false; 5413 } 5414 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 5415 // see note at standby() declaration 5416 if (playbackThread->standby() && !playbackThread->isSuspended()) { 5417 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), 5418 thread.get()); 5419 return false; 5420 } 5421 } 5422 return true; 5423} 5424 5425uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() const 5426{ 5427 return (mWaitTimeMs * 1000) / 2; 5428} 5429 5430void AudioFlinger::DuplicatingThread::cacheParameters_l() 5431{ 5432 // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first 5433 updateWaitTime_l(); 5434 5435 MixerThread::cacheParameters_l(); 5436} 5437 5438// ---------------------------------------------------------------------------- 5439// Record 5440// ---------------------------------------------------------------------------- 5441 5442AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 5443 AudioStreamIn *input, 5444 audio_io_handle_t id, 5445 audio_devices_t outDevice, 5446 audio_devices_t inDevice, 5447 bool systemReady 5448#ifdef TEE_SINK 5449 , const sp<NBAIO_Sink>& teeSink 5450#endif 5451 ) : 5452 ThreadBase(audioFlinger, id, outDevice, inDevice, RECORD, systemReady), 5453 mInput(input), mActiveTracksGen(0), mRsmpInBuffer(NULL), 5454 // mRsmpInFrames and mRsmpInFramesP2 are set by readInputParameters_l() 5455 mRsmpInRear(0) 5456#ifdef TEE_SINK 5457 , mTeeSink(teeSink) 5458#endif 5459 , mReadOnlyHeap(new MemoryDealer(kRecordThreadReadOnlyHeapSize, 5460 "RecordThreadRO", MemoryHeapBase::READ_ONLY)) 5461 // mFastCapture below 5462 , mFastCaptureFutex(0) 5463 // mInputSource 5464 // mPipeSink 5465 // mPipeSource 5466 , mPipeFramesP2(0) 5467 // mPipeMemory 5468 // mFastCaptureNBLogWriter 5469 , mFastTrackAvail(false) 5470{ 5471 snprintf(mThreadName, kThreadNameLength, "AudioIn_%X", id); 5472 mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mThreadName); 5473 5474 readInputParameters_l(); 5475 5476 // create an NBAIO source for the HAL input stream, and negotiate 5477 mInputSource = new AudioStreamInSource(input->stream); 5478 size_t numCounterOffers = 0; 5479 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount, mFormat)}; 5480 ssize_t index = mInputSource->negotiate(offers, 1, NULL, numCounterOffers); 5481 ALOG_ASSERT(index == 0); 5482 5483 // initialize fast capture depending on configuration 5484 bool initFastCapture; 5485 switch (kUseFastCapture) { 5486 case FastCapture_Never: 5487 initFastCapture = false; 5488 break; 5489 case FastCapture_Always: 5490 initFastCapture = true; 5491 break; 5492 case FastCapture_Static: 5493 uint32_t primaryOutputSampleRate; 5494 { 5495 AutoMutex _l(audioFlinger->mHardwareLock); 5496 primaryOutputSampleRate = audioFlinger->mPrimaryOutputSampleRate; 5497 } 5498 initFastCapture = 5499 // either capture sample rate is same as (a reasonable) primary output sample rate 5500 ((isMusicRate(primaryOutputSampleRate) && 5501 (mSampleRate == primaryOutputSampleRate)) || 5502 // or primary output sample rate is unknown, and capture sample rate is reasonable 5503 ((primaryOutputSampleRate == 0) && 5504 isMusicRate(mSampleRate))) && 5505 // and the buffer size is < 12 ms 5506 (mFrameCount * 1000) / mSampleRate < 12; 5507 break; 5508 // case FastCapture_Dynamic: 5509 } 5510 5511 if (initFastCapture) { 5512 // create a Pipe for FastCapture to write to, and for us and fast tracks to read from 5513 NBAIO_Format format = mInputSource->format(); 5514 size_t pipeFramesP2 = roundup(mSampleRate / 25); // double-buffering of 20 ms each 5515 size_t pipeSize = pipeFramesP2 * Format_frameSize(format); 5516 void *pipeBuffer; 5517 const sp<MemoryDealer> roHeap(readOnlyHeap()); 5518 sp<IMemory> pipeMemory; 5519 if ((roHeap == 0) || 5520 (pipeMemory = roHeap->allocate(pipeSize)) == 0 || 5521 (pipeBuffer = pipeMemory->pointer()) == NULL) { 5522 ALOGE("not enough memory for pipe buffer size=%zu", pipeSize); 5523 goto failed; 5524 } 5525 // pipe will be shared directly with fast clients, so clear to avoid leaking old information 5526 memset(pipeBuffer, 0, pipeSize); 5527 Pipe *pipe = new Pipe(pipeFramesP2, format, pipeBuffer); 5528 const NBAIO_Format offers[1] = {format}; 5529 size_t numCounterOffers = 0; 5530 ssize_t index = pipe->negotiate(offers, 1, NULL, numCounterOffers); 5531 ALOG_ASSERT(index == 0); 5532 mPipeSink = pipe; 5533 PipeReader *pipeReader = new PipeReader(*pipe); 5534 numCounterOffers = 0; 5535 index = pipeReader->negotiate(offers, 1, NULL, numCounterOffers); 5536 ALOG_ASSERT(index == 0); 5537 mPipeSource = pipeReader; 5538 mPipeFramesP2 = pipeFramesP2; 5539 mPipeMemory = pipeMemory; 5540 5541 // create fast capture 5542 mFastCapture = new FastCapture(); 5543 FastCaptureStateQueue *sq = mFastCapture->sq(); 5544#ifdef STATE_QUEUE_DUMP 5545 // FIXME 5546#endif 5547 FastCaptureState *state = sq->begin(); 5548 state->mCblk = NULL; 5549 state->mInputSource = mInputSource.get(); 5550 state->mInputSourceGen++; 5551 state->mPipeSink = pipe; 5552 state->mPipeSinkGen++; 5553 state->mFrameCount = mFrameCount; 5554 state->mCommand = FastCaptureState::COLD_IDLE; 5555 // already done in constructor initialization list 5556 //mFastCaptureFutex = 0; 5557 state->mColdFutexAddr = &mFastCaptureFutex; 5558 state->mColdGen++; 5559 state->mDumpState = &mFastCaptureDumpState; 5560#ifdef TEE_SINK 5561 // FIXME 5562#endif 5563 mFastCaptureNBLogWriter = audioFlinger->newWriter_l(kFastCaptureLogSize, "FastCapture"); 5564 state->mNBLogWriter = mFastCaptureNBLogWriter.get(); 5565 sq->end(); 5566 sq->push(FastCaptureStateQueue::BLOCK_UNTIL_PUSHED); 5567 5568 // start the fast capture 5569 mFastCapture->run("FastCapture", ANDROID_PRIORITY_URGENT_AUDIO); 5570 pid_t tid = mFastCapture->getTid(); 5571 sendPrioConfigEvent(getpid_cached, tid, kPriorityFastMixer); 5572#ifdef AUDIO_WATCHDOG 5573 // FIXME 5574#endif 5575 5576 mFastTrackAvail = true; 5577 } 5578failed: ; 5579 5580 // FIXME mNormalSource 5581} 5582 5583AudioFlinger::RecordThread::~RecordThread() 5584{ 5585 if (mFastCapture != 0) { 5586 FastCaptureStateQueue *sq = mFastCapture->sq(); 5587 FastCaptureState *state = sq->begin(); 5588 if (state->mCommand == FastCaptureState::COLD_IDLE) { 5589 int32_t old = android_atomic_inc(&mFastCaptureFutex); 5590 if (old == -1) { 5591 (void) syscall(__NR_futex, &mFastCaptureFutex, FUTEX_WAKE_PRIVATE, 1); 5592 } 5593 } 5594 state->mCommand = FastCaptureState::EXIT; 5595 sq->end(); 5596 sq->push(FastCaptureStateQueue::BLOCK_UNTIL_PUSHED); 5597 mFastCapture->join(); 5598 mFastCapture.clear(); 5599 } 5600 mAudioFlinger->unregisterWriter(mFastCaptureNBLogWriter); 5601 mAudioFlinger->unregisterWriter(mNBLogWriter); 5602 free(mRsmpInBuffer); 5603} 5604 5605void AudioFlinger::RecordThread::onFirstRef() 5606{ 5607 run(mThreadName, PRIORITY_URGENT_AUDIO); 5608} 5609 5610bool AudioFlinger::RecordThread::threadLoop() 5611{ 5612 nsecs_t lastWarning = 0; 5613 5614 inputStandBy(); 5615 5616reacquire_wakelock: 5617 sp<RecordTrack> activeTrack; 5618 int activeTracksGen; 5619 { 5620 Mutex::Autolock _l(mLock); 5621 size_t size = mActiveTracks.size(); 5622 activeTracksGen = mActiveTracksGen; 5623 if (size > 0) { 5624 // FIXME an arbitrary choice 5625 activeTrack = mActiveTracks[0]; 5626 acquireWakeLock_l(activeTrack->uid()); 5627 if (size > 1) { 5628 SortedVector<int> tmp; 5629 for (size_t i = 0; i < size; i++) { 5630 tmp.add(mActiveTracks[i]->uid()); 5631 } 5632 updateWakeLockUids_l(tmp); 5633 } 5634 } else { 5635 acquireWakeLock_l(-1); 5636 } 5637 } 5638 5639 // used to request a deferred sleep, to be executed later while mutex is unlocked 5640 uint32_t sleepUs = 0; 5641 5642 // loop while there is work to do 5643 for (;;) { 5644 Vector< sp<EffectChain> > effectChains; 5645 5646 // sleep with mutex unlocked 5647 if (sleepUs > 0) { 5648 ATRACE_BEGIN("sleep"); 5649 usleep(sleepUs); 5650 ATRACE_END(); 5651 sleepUs = 0; 5652 } 5653 5654 // activeTracks accumulates a copy of a subset of mActiveTracks 5655 Vector< sp<RecordTrack> > activeTracks; 5656 5657 // reference to the (first and only) active fast track 5658 sp<RecordTrack> fastTrack; 5659 5660 // reference to a fast track which is about to be removed 5661 sp<RecordTrack> fastTrackToRemove; 5662 5663 { // scope for mLock 5664 Mutex::Autolock _l(mLock); 5665 5666 processConfigEvents_l(); 5667 5668 // check exitPending here because checkForNewParameters_l() and 5669 // checkForNewParameters_l() can temporarily release mLock 5670 if (exitPending()) { 5671 break; 5672 } 5673 5674 // if no active track(s), then standby and release wakelock 5675 size_t size = mActiveTracks.size(); 5676 if (size == 0) { 5677 standbyIfNotAlreadyInStandby(); 5678 // exitPending() can't become true here 5679 releaseWakeLock_l(); 5680 ALOGV("RecordThread: loop stopping"); 5681 // go to sleep 5682 mWaitWorkCV.wait(mLock); 5683 ALOGV("RecordThread: loop starting"); 5684 goto reacquire_wakelock; 5685 } 5686 5687 if (mActiveTracksGen != activeTracksGen) { 5688 activeTracksGen = mActiveTracksGen; 5689 SortedVector<int> tmp; 5690 for (size_t i = 0; i < size; i++) { 5691 tmp.add(mActiveTracks[i]->uid()); 5692 } 5693 updateWakeLockUids_l(tmp); 5694 } 5695 5696 bool doBroadcast = false; 5697 for (size_t i = 0; i < size; ) { 5698 5699 activeTrack = mActiveTracks[i]; 5700 if (activeTrack->isTerminated()) { 5701 if (activeTrack->isFastTrack()) { 5702 ALOG_ASSERT(fastTrackToRemove == 0); 5703 fastTrackToRemove = activeTrack; 5704 } 5705 removeTrack_l(activeTrack); 5706 mActiveTracks.remove(activeTrack); 5707 mActiveTracksGen++; 5708 size--; 5709 continue; 5710 } 5711 5712 TrackBase::track_state activeTrackState = activeTrack->mState; 5713 switch (activeTrackState) { 5714 5715 case TrackBase::PAUSING: 5716 mActiveTracks.remove(activeTrack); 5717 mActiveTracksGen++; 5718 doBroadcast = true; 5719 size--; 5720 continue; 5721 5722 case TrackBase::STARTING_1: 5723 sleepUs = 10000; 5724 i++; 5725 continue; 5726 5727 case TrackBase::STARTING_2: 5728 doBroadcast = true; 5729 mStandby = false; 5730 activeTrack->mState = TrackBase::ACTIVE; 5731 break; 5732 5733 case TrackBase::ACTIVE: 5734 break; 5735 5736 case TrackBase::IDLE: 5737 i++; 5738 continue; 5739 5740 default: 5741 LOG_ALWAYS_FATAL("Unexpected activeTrackState %d", activeTrackState); 5742 } 5743 5744 activeTracks.add(activeTrack); 5745 i++; 5746 5747 if (activeTrack->isFastTrack()) { 5748 ALOG_ASSERT(!mFastTrackAvail); 5749 ALOG_ASSERT(fastTrack == 0); 5750 fastTrack = activeTrack; 5751 } 5752 } 5753 if (doBroadcast) { 5754 mStartStopCond.broadcast(); 5755 } 5756 5757 // sleep if there are no active tracks to process 5758 if (activeTracks.size() == 0) { 5759 if (sleepUs == 0) { 5760 sleepUs = kRecordThreadSleepUs; 5761 } 5762 continue; 5763 } 5764 sleepUs = 0; 5765 5766 lockEffectChains_l(effectChains); 5767 } 5768 5769 // thread mutex is now unlocked, mActiveTracks unknown, activeTracks.size() > 0 5770 5771 size_t size = effectChains.size(); 5772 for (size_t i = 0; i < size; i++) { 5773 // thread mutex is not locked, but effect chain is locked 5774 effectChains[i]->process_l(); 5775 } 5776 5777 // Push a new fast capture state if fast capture is not already running, or cblk change 5778 if (mFastCapture != 0) { 5779 FastCaptureStateQueue *sq = mFastCapture->sq(); 5780 FastCaptureState *state = sq->begin(); 5781 bool didModify = false; 5782 FastCaptureStateQueue::block_t block = FastCaptureStateQueue::BLOCK_UNTIL_PUSHED; 5783 if (state->mCommand != FastCaptureState::READ_WRITE /* FIXME && 5784 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)*/) { 5785 if (state->mCommand == FastCaptureState::COLD_IDLE) { 5786 int32_t old = android_atomic_inc(&mFastCaptureFutex); 5787 if (old == -1) { 5788 (void) syscall(__NR_futex, &mFastCaptureFutex, FUTEX_WAKE_PRIVATE, 1); 5789 } 5790 } 5791 state->mCommand = FastCaptureState::READ_WRITE; 5792#if 0 // FIXME 5793 mFastCaptureDumpState.increaseSamplingN(mAudioFlinger->isLowRamDevice() ? 5794 FastThreadDumpState::kSamplingNforLowRamDevice : 5795 FastThreadDumpState::kSamplingN); 5796#endif 5797 didModify = true; 5798 } 5799 audio_track_cblk_t *cblkOld = state->mCblk; 5800 audio_track_cblk_t *cblkNew = fastTrack != 0 ? fastTrack->cblk() : NULL; 5801 if (cblkNew != cblkOld) { 5802 state->mCblk = cblkNew; 5803 // block until acked if removing a fast track 5804 if (cblkOld != NULL) { 5805 block = FastCaptureStateQueue::BLOCK_UNTIL_ACKED; 5806 } 5807 didModify = true; 5808 } 5809 sq->end(didModify); 5810 if (didModify) { 5811 sq->push(block); 5812#if 0 5813 if (kUseFastCapture == FastCapture_Dynamic) { 5814 mNormalSource = mPipeSource; 5815 } 5816#endif 5817 } 5818 } 5819 5820 // now run the fast track destructor with thread mutex unlocked 5821 fastTrackToRemove.clear(); 5822 5823 // Read from HAL to keep up with fastest client if multiple active tracks, not slowest one. 5824 // Only the client(s) that are too slow will overrun. But if even the fastest client is too 5825 // slow, then this RecordThread will overrun by not calling HAL read often enough. 5826 // If destination is non-contiguous, first read past the nominal end of buffer, then 5827 // copy to the right place. Permitted because mRsmpInBuffer was over-allocated. 5828 5829 int32_t rear = mRsmpInRear & (mRsmpInFramesP2 - 1); 5830 ssize_t framesRead; 5831 5832 // If an NBAIO source is present, use it to read the normal capture's data 5833 if (mPipeSource != 0) { 5834 size_t framesToRead = mBufferSize / mFrameSize; 5835 framesRead = mPipeSource->read((uint8_t*)mRsmpInBuffer + rear * mFrameSize, 5836 framesToRead, AudioBufferProvider::kInvalidPTS); 5837 if (framesRead == 0) { 5838 // since pipe is non-blocking, simulate blocking input 5839 sleepUs = (framesToRead * 1000000LL) / mSampleRate; 5840 } 5841 // otherwise use the HAL / AudioStreamIn directly 5842 } else { 5843 ssize_t bytesRead = mInput->stream->read(mInput->stream, 5844 (uint8_t*)mRsmpInBuffer + rear * mFrameSize, mBufferSize); 5845 if (bytesRead < 0) { 5846 framesRead = bytesRead; 5847 } else { 5848 framesRead = bytesRead / mFrameSize; 5849 } 5850 } 5851 5852 if (framesRead < 0 || (framesRead == 0 && mPipeSource == 0)) { 5853 ALOGE("read failed: framesRead=%d", framesRead); 5854 // Force input into standby so that it tries to recover at next read attempt 5855 inputStandBy(); 5856 sleepUs = kRecordThreadSleepUs; 5857 } 5858 if (framesRead <= 0) { 5859 goto unlock; 5860 } 5861 ALOG_ASSERT(framesRead > 0); 5862 5863 if (mTeeSink != 0) { 5864 (void) mTeeSink->write((uint8_t*)mRsmpInBuffer + rear * mFrameSize, framesRead); 5865 } 5866 // If destination is non-contiguous, we now correct for reading past end of buffer. 5867 { 5868 size_t part1 = mRsmpInFramesP2 - rear; 5869 if ((size_t) framesRead > part1) { 5870 memcpy(mRsmpInBuffer, (uint8_t*)mRsmpInBuffer + mRsmpInFramesP2 * mFrameSize, 5871 (framesRead - part1) * mFrameSize); 5872 } 5873 } 5874 rear = mRsmpInRear += framesRead; 5875 5876 size = activeTracks.size(); 5877 // loop over each active track 5878 for (size_t i = 0; i < size; i++) { 5879 activeTrack = activeTracks[i]; 5880 5881 // skip fast tracks, as those are handled directly by FastCapture 5882 if (activeTrack->isFastTrack()) { 5883 continue; 5884 } 5885 5886 // TODO: This code probably should be moved to RecordTrack. 5887 // TODO: Update the activeTrack buffer converter in case of reconfigure. 5888 5889 enum { 5890 OVERRUN_UNKNOWN, 5891 OVERRUN_TRUE, 5892 OVERRUN_FALSE 5893 } overrun = OVERRUN_UNKNOWN; 5894 5895 // loop over getNextBuffer to handle circular sink 5896 for (;;) { 5897 5898 activeTrack->mSink.frameCount = ~0; 5899 status_t status = activeTrack->getNextBuffer(&activeTrack->mSink); 5900 size_t framesOut = activeTrack->mSink.frameCount; 5901 LOG_ALWAYS_FATAL_IF((status == OK) != (framesOut > 0)); 5902 5903 // check available frames and handle overrun conditions 5904 // if the record track isn't draining fast enough. 5905 bool hasOverrun; 5906 size_t framesIn; 5907 activeTrack->mResamplerBufferProvider->sync(&framesIn, &hasOverrun); 5908 if (hasOverrun) { 5909 overrun = OVERRUN_TRUE; 5910 } 5911 if (framesOut == 0 || framesIn == 0) { 5912 break; 5913 } 5914 5915 // Don't allow framesOut to be larger than what is possible with resampling 5916 // from framesIn. 5917 // This isn't strictly necessary but helps limit buffer resizing in 5918 // RecordBufferConverter. TODO: remove when no longer needed. 5919 framesOut = min(framesOut, 5920 destinationFramesPossible( 5921 framesIn, mSampleRate, activeTrack->mSampleRate)); 5922 // process frames from the RecordThread buffer provider to the RecordTrack buffer 5923 framesOut = activeTrack->mRecordBufferConverter->convert( 5924 activeTrack->mSink.raw, activeTrack->mResamplerBufferProvider, framesOut); 5925 5926 if (framesOut > 0 && (overrun == OVERRUN_UNKNOWN)) { 5927 overrun = OVERRUN_FALSE; 5928 } 5929 5930 if (activeTrack->mFramesToDrop == 0) { 5931 if (framesOut > 0) { 5932 activeTrack->mSink.frameCount = framesOut; 5933 activeTrack->releaseBuffer(&activeTrack->mSink); 5934 } 5935 } else { 5936 // FIXME could do a partial drop of framesOut 5937 if (activeTrack->mFramesToDrop > 0) { 5938 activeTrack->mFramesToDrop -= framesOut; 5939 if (activeTrack->mFramesToDrop <= 0) { 5940 activeTrack->clearSyncStartEvent(); 5941 } 5942 } else { 5943 activeTrack->mFramesToDrop += framesOut; 5944 if (activeTrack->mFramesToDrop >= 0 || activeTrack->mSyncStartEvent == 0 || 5945 activeTrack->mSyncStartEvent->isCancelled()) { 5946 ALOGW("Synced record %s, session %d, trigger session %d", 5947 (activeTrack->mFramesToDrop >= 0) ? "timed out" : "cancelled", 5948 activeTrack->sessionId(), 5949 (activeTrack->mSyncStartEvent != 0) ? 5950 activeTrack->mSyncStartEvent->triggerSession() : 0); 5951 activeTrack->clearSyncStartEvent(); 5952 } 5953 } 5954 } 5955 5956 if (framesOut == 0) { 5957 break; 5958 } 5959 } 5960 5961 switch (overrun) { 5962 case OVERRUN_TRUE: 5963 // client isn't retrieving buffers fast enough 5964 if (!activeTrack->setOverflow()) { 5965 nsecs_t now = systemTime(); 5966 // FIXME should lastWarning per track? 5967 if ((now - lastWarning) > kWarningThrottleNs) { 5968 ALOGW("RecordThread: buffer overflow"); 5969 lastWarning = now; 5970 } 5971 } 5972 break; 5973 case OVERRUN_FALSE: 5974 activeTrack->clearOverflow(); 5975 break; 5976 case OVERRUN_UNKNOWN: 5977 break; 5978 } 5979 5980 } 5981 5982unlock: 5983 // enable changes in effect chain 5984 unlockEffectChains(effectChains); 5985 // effectChains doesn't need to be cleared, since it is cleared by destructor at scope end 5986 } 5987 5988 standbyIfNotAlreadyInStandby(); 5989 5990 { 5991 Mutex::Autolock _l(mLock); 5992 for (size_t i = 0; i < mTracks.size(); i++) { 5993 sp<RecordTrack> track = mTracks[i]; 5994 track->invalidate(); 5995 } 5996 mActiveTracks.clear(); 5997 mActiveTracksGen++; 5998 mStartStopCond.broadcast(); 5999 } 6000 6001 releaseWakeLock(); 6002 6003 ALOGV("RecordThread %p exiting", this); 6004 return false; 6005} 6006 6007void AudioFlinger::RecordThread::standbyIfNotAlreadyInStandby() 6008{ 6009 if (!mStandby) { 6010 inputStandBy(); 6011 mStandby = true; 6012 } 6013} 6014 6015void AudioFlinger::RecordThread::inputStandBy() 6016{ 6017 // Idle the fast capture if it's currently running 6018 if (mFastCapture != 0) { 6019 FastCaptureStateQueue *sq = mFastCapture->sq(); 6020 FastCaptureState *state = sq->begin(); 6021 if (!(state->mCommand & FastCaptureState::IDLE)) { 6022 state->mCommand = FastCaptureState::COLD_IDLE; 6023 state->mColdFutexAddr = &mFastCaptureFutex; 6024 state->mColdGen++; 6025 mFastCaptureFutex = 0; 6026 sq->end(); 6027 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 6028 sq->push(FastCaptureStateQueue::BLOCK_UNTIL_ACKED); 6029#if 0 6030 if (kUseFastCapture == FastCapture_Dynamic) { 6031 // FIXME 6032 } 6033#endif 6034#ifdef AUDIO_WATCHDOG 6035 // FIXME 6036#endif 6037 } else { 6038 sq->end(false /*didModify*/); 6039 } 6040 } 6041 mInput->stream->common.standby(&mInput->stream->common); 6042} 6043 6044// RecordThread::createRecordTrack_l() must be called with AudioFlinger::mLock held 6045sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 6046 const sp<AudioFlinger::Client>& client, 6047 uint32_t sampleRate, 6048 audio_format_t format, 6049 audio_channel_mask_t channelMask, 6050 size_t *pFrameCount, 6051 int sessionId, 6052 size_t *notificationFrames, 6053 int uid, 6054 IAudioFlinger::track_flags_t *flags, 6055 pid_t tid, 6056 status_t *status) 6057{ 6058 size_t frameCount = *pFrameCount; 6059 sp<RecordTrack> track; 6060 status_t lStatus; 6061 6062 // client expresses a preference for FAST, but we get the final say 6063 if (*flags & IAudioFlinger::TRACK_FAST) { 6064 if ( 6065 // we formerly checked for a callback handler (non-0 tid), 6066 // but that is no longer required for TRANSFER_OBTAIN mode 6067 // 6068 // frame count is not specified, or is exactly the pipe depth 6069 ((frameCount == 0) || (frameCount == mPipeFramesP2)) && 6070 // PCM data 6071 audio_is_linear_pcm(format) && 6072 // native format 6073 (format == mFormat) && 6074 // native channel mask 6075 (channelMask == mChannelMask) && 6076 // native hardware sample rate 6077 (sampleRate == mSampleRate) && 6078 // record thread has an associated fast capture 6079 hasFastCapture() && 6080 // there are sufficient fast track slots available 6081 mFastTrackAvail 6082 ) { 6083 ALOGV("AUDIO_INPUT_FLAG_FAST accepted: frameCount=%u mFrameCount=%u", 6084 frameCount, mFrameCount); 6085 } else { 6086 ALOGV("AUDIO_INPUT_FLAG_FAST denied: frameCount=%u mFrameCount=%u mPipeFramesP2=%u " 6087 "format=%#x isLinear=%d channelMask=%#x sampleRate=%u mSampleRate=%u " 6088 "hasFastCapture=%d tid=%d mFastTrackAvail=%d", 6089 frameCount, mFrameCount, mPipeFramesP2, 6090 format, audio_is_linear_pcm(format), channelMask, sampleRate, mSampleRate, 6091 hasFastCapture(), tid, mFastTrackAvail); 6092 *flags &= ~IAudioFlinger::TRACK_FAST; 6093 } 6094 } 6095 6096 // compute track buffer size in frames, and suggest the notification frame count 6097 if (*flags & IAudioFlinger::TRACK_FAST) { 6098 // fast track: frame count is exactly the pipe depth 6099 frameCount = mPipeFramesP2; 6100 // ignore requested notificationFrames, and always notify exactly once every HAL buffer 6101 *notificationFrames = mFrameCount; 6102 } else { 6103 // not fast track: max notification period is resampled equivalent of one HAL buffer time 6104 // or 20 ms if there is a fast capture 6105 // TODO This could be a roundupRatio inline, and const 6106 size_t maxNotificationFrames = ((int64_t) (hasFastCapture() ? mSampleRate/50 : mFrameCount) 6107 * sampleRate + mSampleRate - 1) / mSampleRate; 6108 // minimum number of notification periods is at least kMinNotifications, 6109 // and at least kMinMs rounded up to a whole notification period (minNotificationsByMs) 6110 static const size_t kMinNotifications = 3; 6111 static const uint32_t kMinMs = 30; 6112 // TODO This could be a roundupRatio inline 6113 const size_t minFramesByMs = (sampleRate * kMinMs + 1000 - 1) / 1000; 6114 // TODO This could be a roundupRatio inline 6115 const size_t minNotificationsByMs = (minFramesByMs + maxNotificationFrames - 1) / 6116 maxNotificationFrames; 6117 const size_t minFrameCount = maxNotificationFrames * 6118 max(kMinNotifications, minNotificationsByMs); 6119 frameCount = max(frameCount, minFrameCount); 6120 if (*notificationFrames == 0 || *notificationFrames > maxNotificationFrames) { 6121 *notificationFrames = maxNotificationFrames; 6122 } 6123 } 6124 *pFrameCount = frameCount; 6125 6126 lStatus = initCheck(); 6127 if (lStatus != NO_ERROR) { 6128 ALOGE("createRecordTrack_l() audio driver not initialized"); 6129 goto Exit; 6130 } 6131 6132 { // scope for mLock 6133 Mutex::Autolock _l(mLock); 6134 6135 track = new RecordTrack(this, client, sampleRate, 6136 format, channelMask, frameCount, NULL, sessionId, uid, 6137 *flags, TrackBase::TYPE_DEFAULT); 6138 6139 lStatus = track->initCheck(); 6140 if (lStatus != NO_ERROR) { 6141 ALOGE("createRecordTrack_l() initCheck failed %d; no control block?", lStatus); 6142 // track must be cleared from the caller as the caller has the AF lock 6143 goto Exit; 6144 } 6145 mTracks.add(track); 6146 6147 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 6148 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 6149 mAudioFlinger->btNrecIsOff(); 6150 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 6151 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 6152 6153 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 6154 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 6155 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 6156 // so ask activity manager to do this on our behalf 6157 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 6158 } 6159 } 6160 6161 lStatus = NO_ERROR; 6162 6163Exit: 6164 *status = lStatus; 6165 return track; 6166} 6167 6168status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, 6169 AudioSystem::sync_event_t event, 6170 int triggerSession) 6171{ 6172 ALOGV("RecordThread::start event %d, triggerSession %d", event, triggerSession); 6173 sp<ThreadBase> strongMe = this; 6174 status_t status = NO_ERROR; 6175 6176 if (event == AudioSystem::SYNC_EVENT_NONE) { 6177 recordTrack->clearSyncStartEvent(); 6178 } else if (event != AudioSystem::SYNC_EVENT_SAME) { 6179 recordTrack->mSyncStartEvent = mAudioFlinger->createSyncEvent(event, 6180 triggerSession, 6181 recordTrack->sessionId(), 6182 syncStartEventCallback, 6183 recordTrack); 6184 // Sync event can be cancelled by the trigger session if the track is not in a 6185 // compatible state in which case we start record immediately 6186 if (recordTrack->mSyncStartEvent->isCancelled()) { 6187 recordTrack->clearSyncStartEvent(); 6188 } else { 6189 // do not wait for the event for more than AudioSystem::kSyncRecordStartTimeOutMs 6190 recordTrack->mFramesToDrop = - 6191 ((AudioSystem::kSyncRecordStartTimeOutMs * recordTrack->mSampleRate) / 1000); 6192 } 6193 } 6194 6195 { 6196 // This section is a rendezvous between binder thread executing start() and RecordThread 6197 AutoMutex lock(mLock); 6198 if (mActiveTracks.indexOf(recordTrack) >= 0) { 6199 if (recordTrack->mState == TrackBase::PAUSING) { 6200 ALOGV("active record track PAUSING -> ACTIVE"); 6201 recordTrack->mState = TrackBase::ACTIVE; 6202 } else { 6203 ALOGV("active record track state %d", recordTrack->mState); 6204 } 6205 return status; 6206 } 6207 6208 // TODO consider other ways of handling this, such as changing the state to :STARTING and 6209 // adding the track to mActiveTracks after returning from AudioSystem::startInput(), 6210 // or using a separate command thread 6211 recordTrack->mState = TrackBase::STARTING_1; 6212 mActiveTracks.add(recordTrack); 6213 mActiveTracksGen++; 6214 status_t status = NO_ERROR; 6215 if (recordTrack->isExternalTrack()) { 6216 mLock.unlock(); 6217 status = AudioSystem::startInput(mId, (audio_session_t)recordTrack->sessionId()); 6218 mLock.lock(); 6219 // FIXME should verify that recordTrack is still in mActiveTracks 6220 if (status != NO_ERROR) { 6221 mActiveTracks.remove(recordTrack); 6222 mActiveTracksGen++; 6223 recordTrack->clearSyncStartEvent(); 6224 ALOGV("RecordThread::start error %d", status); 6225 return status; 6226 } 6227 } 6228 // Catch up with current buffer indices if thread is already running. 6229 // This is what makes a new client discard all buffered data. If the track's mRsmpInFront 6230 // was initialized to some value closer to the thread's mRsmpInFront, then the track could 6231 // see previously buffered data before it called start(), but with greater risk of overrun. 6232 6233 recordTrack->mResamplerBufferProvider->reset(); 6234 // clear any converter state as new data will be discontinuous 6235 recordTrack->mRecordBufferConverter->reset(); 6236 recordTrack->mState = TrackBase::STARTING_2; 6237 // signal thread to start 6238 mWaitWorkCV.broadcast(); 6239 if (mActiveTracks.indexOf(recordTrack) < 0) { 6240 ALOGV("Record failed to start"); 6241 status = BAD_VALUE; 6242 goto startError; 6243 } 6244 return status; 6245 } 6246 6247startError: 6248 if (recordTrack->isExternalTrack()) { 6249 AudioSystem::stopInput(mId, (audio_session_t)recordTrack->sessionId()); 6250 } 6251 recordTrack->clearSyncStartEvent(); 6252 // FIXME I wonder why we do not reset the state here? 6253 return status; 6254} 6255 6256void AudioFlinger::RecordThread::syncStartEventCallback(const wp<SyncEvent>& event) 6257{ 6258 sp<SyncEvent> strongEvent = event.promote(); 6259 6260 if (strongEvent != 0) { 6261 sp<RefBase> ptr = strongEvent->cookie().promote(); 6262 if (ptr != 0) { 6263 RecordTrack *recordTrack = (RecordTrack *)ptr.get(); 6264 recordTrack->handleSyncStartEvent(strongEvent); 6265 } 6266 } 6267} 6268 6269bool AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 6270 ALOGV("RecordThread::stop"); 6271 AutoMutex _l(mLock); 6272 if (mActiveTracks.indexOf(recordTrack) != 0 || recordTrack->mState == TrackBase::PAUSING) { 6273 return false; 6274 } 6275 // note that threadLoop may still be processing the track at this point [without lock] 6276 recordTrack->mState = TrackBase::PAUSING; 6277 // do not wait for mStartStopCond if exiting 6278 if (exitPending()) { 6279 return true; 6280 } 6281 // FIXME incorrect usage of wait: no explicit predicate or loop 6282 mStartStopCond.wait(mLock); 6283 // if we have been restarted, recordTrack is in mActiveTracks here 6284 if (exitPending() || mActiveTracks.indexOf(recordTrack) != 0) { 6285 ALOGV("Record stopped OK"); 6286 return true; 6287 } 6288 return false; 6289} 6290 6291bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event __unused) const 6292{ 6293 return false; 6294} 6295 6296status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event __unused) 6297{ 6298#if 0 // This branch is currently dead code, but is preserved in case it will be needed in future 6299 if (!isValidSyncEvent(event)) { 6300 return BAD_VALUE; 6301 } 6302 6303 int eventSession = event->triggerSession(); 6304 status_t ret = NAME_NOT_FOUND; 6305 6306 Mutex::Autolock _l(mLock); 6307 6308 for (size_t i = 0; i < mTracks.size(); i++) { 6309 sp<RecordTrack> track = mTracks[i]; 6310 if (eventSession == track->sessionId()) { 6311 (void) track->setSyncEvent(event); 6312 ret = NO_ERROR; 6313 } 6314 } 6315 return ret; 6316#else 6317 return BAD_VALUE; 6318#endif 6319} 6320 6321// destroyTrack_l() must be called with ThreadBase::mLock held 6322void AudioFlinger::RecordThread::destroyTrack_l(const sp<RecordTrack>& track) 6323{ 6324 track->terminate(); 6325 track->mState = TrackBase::STOPPED; 6326 // active tracks are removed by threadLoop() 6327 if (mActiveTracks.indexOf(track) < 0) { 6328 removeTrack_l(track); 6329 } 6330} 6331 6332void AudioFlinger::RecordThread::removeTrack_l(const sp<RecordTrack>& track) 6333{ 6334 mTracks.remove(track); 6335 // need anything related to effects here? 6336 if (track->isFastTrack()) { 6337 ALOG_ASSERT(!mFastTrackAvail); 6338 mFastTrackAvail = true; 6339 } 6340} 6341 6342void AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 6343{ 6344 dumpInternals(fd, args); 6345 dumpTracks(fd, args); 6346 dumpEffectChains(fd, args); 6347} 6348 6349void AudioFlinger::RecordThread::dumpInternals(int fd, const Vector<String16>& args) 6350{ 6351 dprintf(fd, "\nInput thread %p:\n", this); 6352 6353 dumpBase(fd, args); 6354 6355 if (mActiveTracks.size() == 0) { 6356 dprintf(fd, " No active record clients\n"); 6357 } 6358 dprintf(fd, " Fast capture thread: %s\n", hasFastCapture() ? "yes" : "no"); 6359 dprintf(fd, " Fast track available: %s\n", mFastTrackAvail ? "yes" : "no"); 6360 6361 // Make a non-atomic copy of fast capture dump state so it won't change underneath us 6362 const FastCaptureDumpState copy(mFastCaptureDumpState); 6363 copy.dump(fd); 6364} 6365 6366void AudioFlinger::RecordThread::dumpTracks(int fd, const Vector<String16>& args __unused) 6367{ 6368 const size_t SIZE = 256; 6369 char buffer[SIZE]; 6370 String8 result; 6371 6372 size_t numtracks = mTracks.size(); 6373 size_t numactive = mActiveTracks.size(); 6374 size_t numactiveseen = 0; 6375 dprintf(fd, " %d Tracks", numtracks); 6376 if (numtracks) { 6377 dprintf(fd, " of which %d are active\n", numactive); 6378 RecordTrack::appendDumpHeader(result); 6379 for (size_t i = 0; i < numtracks ; ++i) { 6380 sp<RecordTrack> track = mTracks[i]; 6381 if (track != 0) { 6382 bool active = mActiveTracks.indexOf(track) >= 0; 6383 if (active) { 6384 numactiveseen++; 6385 } 6386 track->dump(buffer, SIZE, active); 6387 result.append(buffer); 6388 } 6389 } 6390 } else { 6391 dprintf(fd, "\n"); 6392 } 6393 6394 if (numactiveseen != numactive) { 6395 snprintf(buffer, SIZE, " The following tracks are in the active list but" 6396 " not in the track list\n"); 6397 result.append(buffer); 6398 RecordTrack::appendDumpHeader(result); 6399 for (size_t i = 0; i < numactive; ++i) { 6400 sp<RecordTrack> track = mActiveTracks[i]; 6401 if (mTracks.indexOf(track) < 0) { 6402 track->dump(buffer, SIZE, true); 6403 result.append(buffer); 6404 } 6405 } 6406 6407 } 6408 write(fd, result.string(), result.size()); 6409} 6410 6411 6412void AudioFlinger::RecordThread::ResamplerBufferProvider::reset() 6413{ 6414 sp<ThreadBase> threadBase = mRecordTrack->mThread.promote(); 6415 RecordThread *recordThread = (RecordThread *) threadBase.get(); 6416 mRsmpInFront = recordThread->mRsmpInRear; 6417 mRsmpInUnrel = 0; 6418} 6419 6420void AudioFlinger::RecordThread::ResamplerBufferProvider::sync( 6421 size_t *framesAvailable, bool *hasOverrun) 6422{ 6423 sp<ThreadBase> threadBase = mRecordTrack->mThread.promote(); 6424 RecordThread *recordThread = (RecordThread *) threadBase.get(); 6425 const int32_t rear = recordThread->mRsmpInRear; 6426 const int32_t front = mRsmpInFront; 6427 const ssize_t filled = rear - front; 6428 6429 size_t framesIn; 6430 bool overrun = false; 6431 if (filled < 0) { 6432 // should not happen, but treat like a massive overrun and re-sync 6433 framesIn = 0; 6434 mRsmpInFront = rear; 6435 overrun = true; 6436 } else if ((size_t) filled <= recordThread->mRsmpInFrames) { 6437 framesIn = (size_t) filled; 6438 } else { 6439 // client is not keeping up with server, but give it latest data 6440 framesIn = recordThread->mRsmpInFrames; 6441 mRsmpInFront = /* front = */ rear - framesIn; 6442 overrun = true; 6443 } 6444 if (framesAvailable != NULL) { 6445 *framesAvailable = framesIn; 6446 } 6447 if (hasOverrun != NULL) { 6448 *hasOverrun = overrun; 6449 } 6450} 6451 6452// AudioBufferProvider interface 6453status_t AudioFlinger::RecordThread::ResamplerBufferProvider::getNextBuffer( 6454 AudioBufferProvider::Buffer* buffer, int64_t pts __unused) 6455{ 6456 sp<ThreadBase> threadBase = mRecordTrack->mThread.promote(); 6457 if (threadBase == 0) { 6458 buffer->frameCount = 0; 6459 buffer->raw = NULL; 6460 return NOT_ENOUGH_DATA; 6461 } 6462 RecordThread *recordThread = (RecordThread *) threadBase.get(); 6463 int32_t rear = recordThread->mRsmpInRear; 6464 int32_t front = mRsmpInFront; 6465 ssize_t filled = rear - front; 6466 // FIXME should not be P2 (don't want to increase latency) 6467 // FIXME if client not keeping up, discard 6468 LOG_ALWAYS_FATAL_IF(!(0 <= filled && (size_t) filled <= recordThread->mRsmpInFrames)); 6469 // 'filled' may be non-contiguous, so return only the first contiguous chunk 6470 front &= recordThread->mRsmpInFramesP2 - 1; 6471 size_t part1 = recordThread->mRsmpInFramesP2 - front; 6472 if (part1 > (size_t) filled) { 6473 part1 = filled; 6474 } 6475 size_t ask = buffer->frameCount; 6476 ALOG_ASSERT(ask > 0); 6477 if (part1 > ask) { 6478 part1 = ask; 6479 } 6480 if (part1 == 0) { 6481 // out of data is fine since the resampler will return a short-count. 6482 buffer->raw = NULL; 6483 buffer->frameCount = 0; 6484 mRsmpInUnrel = 0; 6485 return NOT_ENOUGH_DATA; 6486 } 6487 6488 buffer->raw = (uint8_t*)recordThread->mRsmpInBuffer + front * recordThread->mFrameSize; 6489 buffer->frameCount = part1; 6490 mRsmpInUnrel = part1; 6491 return NO_ERROR; 6492} 6493 6494// AudioBufferProvider interface 6495void AudioFlinger::RecordThread::ResamplerBufferProvider::releaseBuffer( 6496 AudioBufferProvider::Buffer* buffer) 6497{ 6498 size_t stepCount = buffer->frameCount; 6499 if (stepCount == 0) { 6500 return; 6501 } 6502 ALOG_ASSERT(stepCount <= mRsmpInUnrel); 6503 mRsmpInUnrel -= stepCount; 6504 mRsmpInFront += stepCount; 6505 buffer->raw = NULL; 6506 buffer->frameCount = 0; 6507} 6508 6509AudioFlinger::RecordThread::RecordBufferConverter::RecordBufferConverter( 6510 audio_channel_mask_t srcChannelMask, audio_format_t srcFormat, 6511 uint32_t srcSampleRate, 6512 audio_channel_mask_t dstChannelMask, audio_format_t dstFormat, 6513 uint32_t dstSampleRate) : 6514 mSrcChannelMask(AUDIO_CHANNEL_INVALID), // updateParameters will set following vars 6515 // mSrcFormat 6516 // mSrcSampleRate 6517 // mDstChannelMask 6518 // mDstFormat 6519 // mDstSampleRate 6520 // mSrcChannelCount 6521 // mDstChannelCount 6522 // mDstFrameSize 6523 mBuf(NULL), mBufFrames(0), mBufFrameSize(0), 6524 mResampler(NULL), 6525 mIsLegacyDownmix(false), 6526 mIsLegacyUpmix(false), 6527 mRequiresFloat(false), 6528 mInputConverterProvider(NULL) 6529{ 6530 (void)updateParameters(srcChannelMask, srcFormat, srcSampleRate, 6531 dstChannelMask, dstFormat, dstSampleRate); 6532} 6533 6534AudioFlinger::RecordThread::RecordBufferConverter::~RecordBufferConverter() { 6535 free(mBuf); 6536 delete mResampler; 6537 delete mInputConverterProvider; 6538} 6539 6540size_t AudioFlinger::RecordThread::RecordBufferConverter::convert(void *dst, 6541 AudioBufferProvider *provider, size_t frames) 6542{ 6543 if (mInputConverterProvider != NULL) { 6544 mInputConverterProvider->setBufferProvider(provider); 6545 provider = mInputConverterProvider; 6546 } 6547 6548 if (mResampler == NULL) { 6549 ALOGVV("NO RESAMPLING sampleRate:%u mSrcFormat:%#x mDstFormat:%#x", 6550 mSrcSampleRate, mSrcFormat, mDstFormat); 6551 6552 AudioBufferProvider::Buffer buffer; 6553 for (size_t i = frames; i > 0; ) { 6554 buffer.frameCount = i; 6555 status_t status = provider->getNextBuffer(&buffer, 0); 6556 if (status != OK || buffer.frameCount == 0) { 6557 frames -= i; // cannot fill request. 6558 break; 6559 } 6560 // format convert to destination buffer 6561 convertNoResampler(dst, buffer.raw, buffer.frameCount); 6562 6563 dst = (int8_t*)dst + buffer.frameCount * mDstFrameSize; 6564 i -= buffer.frameCount; 6565 provider->releaseBuffer(&buffer); 6566 } 6567 } else { 6568 ALOGVV("RESAMPLING mSrcSampleRate:%u mDstSampleRate:%u mSrcFormat:%#x mDstFormat:%#x", 6569 mSrcSampleRate, mDstSampleRate, mSrcFormat, mDstFormat); 6570 6571 // reallocate buffer if needed 6572 if (mBufFrameSize != 0 && mBufFrames < frames) { 6573 free(mBuf); 6574 mBufFrames = frames; 6575 (void)posix_memalign(&mBuf, 32, mBufFrames * mBufFrameSize); 6576 } 6577 // resampler accumulates, but we only have one source track 6578 memset(mBuf, 0, frames * mBufFrameSize); 6579 frames = mResampler->resample((int32_t*)mBuf, frames, provider); 6580 // format convert to destination buffer 6581 convertResampler(dst, mBuf, frames); 6582 } 6583 return frames; 6584} 6585 6586status_t AudioFlinger::RecordThread::RecordBufferConverter::updateParameters( 6587 audio_channel_mask_t srcChannelMask, audio_format_t srcFormat, 6588 uint32_t srcSampleRate, 6589 audio_channel_mask_t dstChannelMask, audio_format_t dstFormat, 6590 uint32_t dstSampleRate) 6591{ 6592 // quick evaluation if there is any change. 6593 if (mSrcFormat == srcFormat 6594 && mSrcChannelMask == srcChannelMask 6595 && mSrcSampleRate == srcSampleRate 6596 && mDstFormat == dstFormat 6597 && mDstChannelMask == dstChannelMask 6598 && mDstSampleRate == dstSampleRate) { 6599 return NO_ERROR; 6600 } 6601 6602 ALOGV("RecordBufferConverter updateParameters srcMask:%#x dstMask:%#x" 6603 " srcFormat:%#x dstFormat:%#x srcRate:%u dstRate:%u", 6604 srcChannelMask, dstChannelMask, srcFormat, dstFormat, srcSampleRate, dstSampleRate); 6605 const bool valid = 6606 audio_is_input_channel(srcChannelMask) 6607 && audio_is_input_channel(dstChannelMask) 6608 && audio_is_valid_format(srcFormat) && audio_is_linear_pcm(srcFormat) 6609 && audio_is_valid_format(dstFormat) && audio_is_linear_pcm(dstFormat) 6610 && (srcSampleRate <= dstSampleRate * AUDIO_RESAMPLER_DOWN_RATIO_MAX) 6611 ; // no upsampling checks for now 6612 if (!valid) { 6613 return BAD_VALUE; 6614 } 6615 6616 mSrcFormat = srcFormat; 6617 mSrcChannelMask = srcChannelMask; 6618 mSrcSampleRate = srcSampleRate; 6619 mDstFormat = dstFormat; 6620 mDstChannelMask = dstChannelMask; 6621 mDstSampleRate = dstSampleRate; 6622 6623 // compute derived parameters 6624 mSrcChannelCount = audio_channel_count_from_in_mask(srcChannelMask); 6625 mDstChannelCount = audio_channel_count_from_in_mask(dstChannelMask); 6626 mDstFrameSize = mDstChannelCount * audio_bytes_per_sample(mDstFormat); 6627 6628 // do we need to resample? 6629 delete mResampler; 6630 mResampler = NULL; 6631 if (mSrcSampleRate != mDstSampleRate) { 6632 mResampler = AudioResampler::create(AUDIO_FORMAT_PCM_FLOAT, 6633 mSrcChannelCount, mDstSampleRate); 6634 mResampler->setSampleRate(mSrcSampleRate); 6635 mResampler->setVolume(AudioMixer::UNITY_GAIN_FLOAT, AudioMixer::UNITY_GAIN_FLOAT); 6636 } 6637 6638 // are we running legacy channel conversion modes? 6639 mIsLegacyDownmix = (mSrcChannelMask == AUDIO_CHANNEL_IN_STEREO 6640 || mSrcChannelMask == AUDIO_CHANNEL_IN_FRONT_BACK) 6641 && mDstChannelMask == AUDIO_CHANNEL_IN_MONO; 6642 mIsLegacyUpmix = mSrcChannelMask == AUDIO_CHANNEL_IN_MONO 6643 && (mDstChannelMask == AUDIO_CHANNEL_IN_STEREO 6644 || mDstChannelMask == AUDIO_CHANNEL_IN_FRONT_BACK); 6645 6646 // do we need to process in float? 6647 mRequiresFloat = mResampler != NULL || mIsLegacyDownmix || mIsLegacyUpmix; 6648 6649 // do we need a staging buffer to convert for destination (we can still optimize this)? 6650 // we use mBufFrameSize > 0 to indicate both frame size as well as buffer necessity 6651 if (mResampler != NULL) { 6652 mBufFrameSize = max(mSrcChannelCount, FCC_2) 6653 * audio_bytes_per_sample(AUDIO_FORMAT_PCM_FLOAT); 6654 } else if ((mIsLegacyUpmix || mIsLegacyDownmix) && mDstFormat != AUDIO_FORMAT_PCM_FLOAT) { 6655 mBufFrameSize = mDstChannelCount * audio_bytes_per_sample(AUDIO_FORMAT_PCM_FLOAT); 6656 } else if (mSrcChannelMask != mDstChannelMask && mDstFormat != mSrcFormat) { 6657 mBufFrameSize = mDstChannelCount * audio_bytes_per_sample(mSrcFormat); 6658 } else { 6659 mBufFrameSize = 0; 6660 } 6661 mBufFrames = 0; // force the buffer to be resized. 6662 6663 // do we need an input converter buffer provider to give us float? 6664 delete mInputConverterProvider; 6665 mInputConverterProvider = NULL; 6666 if (mRequiresFloat && mSrcFormat != AUDIO_FORMAT_PCM_FLOAT) { 6667 mInputConverterProvider = new ReformatBufferProvider( 6668 audio_channel_count_from_in_mask(mSrcChannelMask), 6669 mSrcFormat, 6670 AUDIO_FORMAT_PCM_FLOAT, 6671 256 /* provider buffer frame count */); 6672 } 6673 6674 // do we need a remixer to do channel mask conversion 6675 if (!mIsLegacyDownmix && !mIsLegacyUpmix && mSrcChannelMask != mDstChannelMask) { 6676 (void) memcpy_by_index_array_initialization_from_channel_mask( 6677 mIdxAry, ARRAY_SIZE(mIdxAry), mDstChannelMask, mSrcChannelMask); 6678 } 6679 return NO_ERROR; 6680} 6681 6682void AudioFlinger::RecordThread::RecordBufferConverter::convertNoResampler( 6683 void *dst, const void *src, size_t frames) 6684{ 6685 // src is native type unless there is legacy upmix or downmix, whereupon it is float. 6686 if (mBufFrameSize != 0 && mBufFrames < frames) { 6687 free(mBuf); 6688 mBufFrames = frames; 6689 (void)posix_memalign(&mBuf, 32, mBufFrames * mBufFrameSize); 6690 } 6691 // do we need to do legacy upmix and downmix? 6692 if (mIsLegacyUpmix || mIsLegacyDownmix) { 6693 void *dstBuf = mBuf != NULL ? mBuf : dst; 6694 if (mIsLegacyUpmix) { 6695 upmix_to_stereo_float_from_mono_float((float *)dstBuf, 6696 (const float *)src, frames); 6697 } else /*mIsLegacyDownmix */ { 6698 downmix_to_mono_float_from_stereo_float((float *)dstBuf, 6699 (const float *)src, frames); 6700 } 6701 if (mBuf != NULL) { 6702 memcpy_by_audio_format(dst, mDstFormat, mBuf, AUDIO_FORMAT_PCM_FLOAT, 6703 frames * mDstChannelCount); 6704 } 6705 return; 6706 } 6707 // do we need to do channel mask conversion? 6708 if (mSrcChannelMask != mDstChannelMask) { 6709 void *dstBuf = mBuf != NULL ? mBuf : dst; 6710 memcpy_by_index_array(dstBuf, mDstChannelCount, 6711 src, mSrcChannelCount, mIdxAry, audio_bytes_per_sample(mSrcFormat), frames); 6712 if (dstBuf == dst) { 6713 return; // format is the same 6714 } 6715 } 6716 // convert to destination buffer 6717 const void *convertBuf = mBuf != NULL ? mBuf : src; 6718 memcpy_by_audio_format(dst, mDstFormat, convertBuf, mSrcFormat, 6719 frames * mDstChannelCount); 6720} 6721 6722void AudioFlinger::RecordThread::RecordBufferConverter::convertResampler( 6723 void *dst, /*not-a-const*/ void *src, size_t frames) 6724{ 6725 // src buffer format is ALWAYS float when entering this routine 6726 if (mIsLegacyUpmix) { 6727 ; // mono to stereo already handled by resampler 6728 } else if (mIsLegacyDownmix 6729 || (mSrcChannelMask == mDstChannelMask && mSrcChannelCount == 1)) { 6730 // the resampler outputs stereo for mono input channel (a feature?) 6731 // must convert to mono 6732 downmix_to_mono_float_from_stereo_float((float *)src, 6733 (const float *)src, frames); 6734 } else if (mSrcChannelMask != mDstChannelMask) { 6735 // convert to mono channel again for channel mask conversion (could be skipped 6736 // with further optimization). 6737 if (mSrcChannelCount == 1) { 6738 downmix_to_mono_float_from_stereo_float((float *)src, 6739 (const float *)src, frames); 6740 } 6741 // convert to destination format (in place, OK as float is larger than other types) 6742 if (mDstFormat != AUDIO_FORMAT_PCM_FLOAT) { 6743 memcpy_by_audio_format(src, mDstFormat, src, AUDIO_FORMAT_PCM_FLOAT, 6744 frames * mSrcChannelCount); 6745 } 6746 // channel convert and save to dst 6747 memcpy_by_index_array(dst, mDstChannelCount, 6748 src, mSrcChannelCount, mIdxAry, audio_bytes_per_sample(mDstFormat), frames); 6749 return; 6750 } 6751 // convert to destination format and save to dst 6752 memcpy_by_audio_format(dst, mDstFormat, src, AUDIO_FORMAT_PCM_FLOAT, 6753 frames * mDstChannelCount); 6754} 6755 6756bool AudioFlinger::RecordThread::checkForNewParameter_l(const String8& keyValuePair, 6757 status_t& status) 6758{ 6759 bool reconfig = false; 6760 6761 status = NO_ERROR; 6762 6763 audio_format_t reqFormat = mFormat; 6764 uint32_t samplingRate = mSampleRate; 6765 // TODO this may change if we want to support capture from HDMI PCM multi channel (e.g on TVs). 6766 audio_channel_mask_t channelMask = audio_channel_in_mask_from_count(mChannelCount); 6767 6768 AudioParameter param = AudioParameter(keyValuePair); 6769 int value; 6770 // TODO Investigate when this code runs. Check with audio policy when a sample rate and 6771 // channel count change can be requested. Do we mandate the first client defines the 6772 // HAL sampling rate and channel count or do we allow changes on the fly? 6773 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 6774 samplingRate = value; 6775 reconfig = true; 6776 } 6777 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 6778 if (!audio_is_linear_pcm((audio_format_t) value)) { 6779 status = BAD_VALUE; 6780 } else { 6781 reqFormat = (audio_format_t) value; 6782 reconfig = true; 6783 } 6784 } 6785 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 6786 audio_channel_mask_t mask = (audio_channel_mask_t) value; 6787 if (!audio_is_input_channel(mask) || 6788 audio_channel_count_from_in_mask(mask) > FCC_8) { 6789 status = BAD_VALUE; 6790 } else { 6791 channelMask = mask; 6792 reconfig = true; 6793 } 6794 } 6795 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 6796 // do not accept frame count changes if tracks are open as the track buffer 6797 // size depends on frame count and correct behavior would not be guaranteed 6798 // if frame count is changed after track creation 6799 if (mActiveTracks.size() > 0) { 6800 status = INVALID_OPERATION; 6801 } else { 6802 reconfig = true; 6803 } 6804 } 6805 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 6806 // forward device change to effects that have requested to be 6807 // aware of attached audio device. 6808 for (size_t i = 0; i < mEffectChains.size(); i++) { 6809 mEffectChains[i]->setDevice_l(value); 6810 } 6811 6812 // store input device and output device but do not forward output device to audio HAL. 6813 // Note that status is ignored by the caller for output device 6814 // (see AudioFlinger::setParameters() 6815 if (audio_is_output_devices(value)) { 6816 mOutDevice = value; 6817 status = BAD_VALUE; 6818 } else { 6819 mInDevice = value; 6820 if (value != AUDIO_DEVICE_NONE) { 6821 mPrevInDevice = value; 6822 } 6823 // disable AEC and NS if the device is a BT SCO headset supporting those 6824 // pre processings 6825 if (mTracks.size() > 0) { 6826 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 6827 mAudioFlinger->btNrecIsOff(); 6828 for (size_t i = 0; i < mTracks.size(); i++) { 6829 sp<RecordTrack> track = mTracks[i]; 6830 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId()); 6831 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId()); 6832 } 6833 } 6834 } 6835 } 6836 if (param.getInt(String8(AudioParameter::keyInputSource), value) == NO_ERROR && 6837 mAudioSource != (audio_source_t)value) { 6838 // forward device change to effects that have requested to be 6839 // aware of attached audio device. 6840 for (size_t i = 0; i < mEffectChains.size(); i++) { 6841 mEffectChains[i]->setAudioSource_l((audio_source_t)value); 6842 } 6843 mAudioSource = (audio_source_t)value; 6844 } 6845 6846 if (status == NO_ERROR) { 6847 status = mInput->stream->common.set_parameters(&mInput->stream->common, 6848 keyValuePair.string()); 6849 if (status == INVALID_OPERATION) { 6850 inputStandBy(); 6851 status = mInput->stream->common.set_parameters(&mInput->stream->common, 6852 keyValuePair.string()); 6853 } 6854 if (reconfig) { 6855 if (status == BAD_VALUE && 6856 audio_is_linear_pcm(mInput->stream->common.get_format(&mInput->stream->common)) && 6857 audio_is_linear_pcm(reqFormat) && 6858 (mInput->stream->common.get_sample_rate(&mInput->stream->common) 6859 <= (AUDIO_RESAMPLER_DOWN_RATIO_MAX * samplingRate)) && 6860 audio_channel_count_from_in_mask( 6861 mInput->stream->common.get_channels(&mInput->stream->common)) <= FCC_8) { 6862 status = NO_ERROR; 6863 } 6864 if (status == NO_ERROR) { 6865 readInputParameters_l(); 6866 sendIoConfigEvent_l(AUDIO_INPUT_CONFIG_CHANGED); 6867 } 6868 } 6869 } 6870 6871 return reconfig; 6872} 6873 6874String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 6875{ 6876 Mutex::Autolock _l(mLock); 6877 if (initCheck() != NO_ERROR) { 6878 return String8(); 6879 } 6880 6881 char *s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 6882 const String8 out_s8(s); 6883 free(s); 6884 return out_s8; 6885} 6886 6887void AudioFlinger::RecordThread::ioConfigChanged(audio_io_config_event event, pid_t pid) { 6888 sp<AudioIoDescriptor> desc = new AudioIoDescriptor(); 6889 6890 desc->mIoHandle = mId; 6891 6892 switch (event) { 6893 case AUDIO_INPUT_OPENED: 6894 case AUDIO_INPUT_CONFIG_CHANGED: 6895 desc->mPatch = mPatch; 6896 desc->mChannelMask = mChannelMask; 6897 desc->mSamplingRate = mSampleRate; 6898 desc->mFormat = mFormat; 6899 desc->mFrameCount = mFrameCount; 6900 desc->mLatency = 0; 6901 break; 6902 6903 case AUDIO_INPUT_CLOSED: 6904 default: 6905 break; 6906 } 6907 mAudioFlinger->ioConfigChanged(event, desc, pid); 6908} 6909 6910void AudioFlinger::RecordThread::readInputParameters_l() 6911{ 6912 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 6913 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 6914 mChannelCount = audio_channel_count_from_in_mask(mChannelMask); 6915 if (mChannelCount > FCC_8) { 6916 ALOGE("HAL channel count %d > %d", mChannelCount, FCC_8); 6917 } 6918 mHALFormat = mInput->stream->common.get_format(&mInput->stream->common); 6919 mFormat = mHALFormat; 6920 if (!audio_is_linear_pcm(mFormat)) { 6921 ALOGE("HAL format %#x is not linear pcm", mFormat); 6922 } 6923 mFrameSize = audio_stream_in_frame_size(mInput->stream); 6924 mBufferSize = mInput->stream->common.get_buffer_size(&mInput->stream->common); 6925 mFrameCount = mBufferSize / mFrameSize; 6926 // This is the formula for calculating the temporary buffer size. 6927 // With 7 HAL buffers, we can guarantee ability to down-sample the input by ratio of 6:1 to 6928 // 1 full output buffer, regardless of the alignment of the available input. 6929 // The value is somewhat arbitrary, and could probably be even larger. 6930 // A larger value should allow more old data to be read after a track calls start(), 6931 // without increasing latency. 6932 // 6933 // Note this is independent of the maximum downsampling ratio permitted for capture. 6934 mRsmpInFrames = mFrameCount * 7; 6935 mRsmpInFramesP2 = roundup(mRsmpInFrames); 6936 free(mRsmpInBuffer); 6937 6938 // TODO optimize audio capture buffer sizes ... 6939 // Here we calculate the size of the sliding buffer used as a source 6940 // for resampling. mRsmpInFramesP2 is currently roundup(mFrameCount * 7). 6941 // For current HAL frame counts, this is usually 2048 = 40 ms. It would 6942 // be better to have it derived from the pipe depth in the long term. 6943 // The current value is higher than necessary. However it should not add to latency. 6944 6945 // Over-allocate beyond mRsmpInFramesP2 to permit a HAL read past end of buffer 6946 (void)posix_memalign(&mRsmpInBuffer, 32, (mRsmpInFramesP2 + mFrameCount - 1) * mFrameSize); 6947 6948 // AudioRecord mSampleRate and mChannelCount are constant due to AudioRecord API constraints. 6949 // But if thread's mSampleRate or mChannelCount changes, how will that affect active tracks? 6950} 6951 6952uint32_t AudioFlinger::RecordThread::getInputFramesLost() 6953{ 6954 Mutex::Autolock _l(mLock); 6955 if (initCheck() != NO_ERROR) { 6956 return 0; 6957 } 6958 6959 return mInput->stream->get_input_frames_lost(mInput->stream); 6960} 6961 6962uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) const 6963{ 6964 Mutex::Autolock _l(mLock); 6965 uint32_t result = 0; 6966 if (getEffectChain_l(sessionId) != 0) { 6967 result = EFFECT_SESSION; 6968 } 6969 6970 for (size_t i = 0; i < mTracks.size(); ++i) { 6971 if (sessionId == mTracks[i]->sessionId()) { 6972 result |= TRACK_SESSION; 6973 break; 6974 } 6975 } 6976 6977 return result; 6978} 6979 6980KeyedVector<int, bool> AudioFlinger::RecordThread::sessionIds() const 6981{ 6982 KeyedVector<int, bool> ids; 6983 Mutex::Autolock _l(mLock); 6984 for (size_t j = 0; j < mTracks.size(); ++j) { 6985 sp<RecordThread::RecordTrack> track = mTracks[j]; 6986 int sessionId = track->sessionId(); 6987 if (ids.indexOfKey(sessionId) < 0) { 6988 ids.add(sessionId, true); 6989 } 6990 } 6991 return ids; 6992} 6993 6994AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 6995{ 6996 Mutex::Autolock _l(mLock); 6997 AudioStreamIn *input = mInput; 6998 mInput = NULL; 6999 return input; 7000} 7001 7002// this method must always be called either with ThreadBase mLock held or inside the thread loop 7003audio_stream_t* AudioFlinger::RecordThread::stream() const 7004{ 7005 if (mInput == NULL) { 7006 return NULL; 7007 } 7008 return &mInput->stream->common; 7009} 7010 7011status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 7012{ 7013 // only one chain per input thread 7014 if (mEffectChains.size() != 0) { 7015 ALOGW("addEffectChain_l() already one chain %p on thread %p", chain.get(), this); 7016 return INVALID_OPERATION; 7017 } 7018 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 7019 chain->setThread(this); 7020 chain->setInBuffer(NULL); 7021 chain->setOutBuffer(NULL); 7022 7023 checkSuspendOnAddEffectChain_l(chain); 7024 7025 // make sure enabled pre processing effects state is communicated to the HAL as we 7026 // just moved them to a new input stream. 7027 chain->syncHalEffectsState(); 7028 7029 mEffectChains.add(chain); 7030 7031 return NO_ERROR; 7032} 7033 7034size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 7035{ 7036 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 7037 ALOGW_IF(mEffectChains.size() != 1, 7038 "removeEffectChain_l() %p invalid chain size %d on thread %p", 7039 chain.get(), mEffectChains.size(), this); 7040 if (mEffectChains.size() == 1) { 7041 mEffectChains.removeAt(0); 7042 } 7043 return 0; 7044} 7045 7046status_t AudioFlinger::RecordThread::createAudioPatch_l(const struct audio_patch *patch, 7047 audio_patch_handle_t *handle) 7048{ 7049 status_t status = NO_ERROR; 7050 7051 // store new device and send to effects 7052 mInDevice = patch->sources[0].ext.device.type; 7053 mPatch = *patch; 7054 for (size_t i = 0; i < mEffectChains.size(); i++) { 7055 mEffectChains[i]->setDevice_l(mInDevice); 7056 } 7057 7058 // disable AEC and NS if the device is a BT SCO headset supporting those 7059 // pre processings 7060 if (mTracks.size() > 0) { 7061 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 7062 mAudioFlinger->btNrecIsOff(); 7063 for (size_t i = 0; i < mTracks.size(); i++) { 7064 sp<RecordTrack> track = mTracks[i]; 7065 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId()); 7066 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId()); 7067 } 7068 } 7069 7070 // store new source and send to effects 7071 if (mAudioSource != patch->sinks[0].ext.mix.usecase.source) { 7072 mAudioSource = patch->sinks[0].ext.mix.usecase.source; 7073 for (size_t i = 0; i < mEffectChains.size(); i++) { 7074 mEffectChains[i]->setAudioSource_l(mAudioSource); 7075 } 7076 } 7077 7078 if (mInput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 7079 audio_hw_device_t *hwDevice = mInput->audioHwDev->hwDevice(); 7080 status = hwDevice->create_audio_patch(hwDevice, 7081 patch->num_sources, 7082 patch->sources, 7083 patch->num_sinks, 7084 patch->sinks, 7085 handle); 7086 } else { 7087 char *address; 7088 if (strcmp(patch->sources[0].ext.device.address, "") != 0) { 7089 address = audio_device_address_to_parameter( 7090 patch->sources[0].ext.device.type, 7091 patch->sources[0].ext.device.address); 7092 } else { 7093 address = (char *)calloc(1, 1); 7094 } 7095 AudioParameter param = AudioParameter(String8(address)); 7096 free(address); 7097 param.addInt(String8(AUDIO_PARAMETER_STREAM_ROUTING), 7098 (int)patch->sources[0].ext.device.type); 7099 param.addInt(String8(AUDIO_PARAMETER_STREAM_INPUT_SOURCE), 7100 (int)patch->sinks[0].ext.mix.usecase.source); 7101 status = mInput->stream->common.set_parameters(&mInput->stream->common, 7102 param.toString().string()); 7103 *handle = AUDIO_PATCH_HANDLE_NONE; 7104 } 7105 7106 if (mInDevice != mPrevInDevice) { 7107 sendIoConfigEvent_l(AUDIO_INPUT_CONFIG_CHANGED); 7108 mPrevInDevice = mInDevice; 7109 } 7110 7111 return status; 7112} 7113 7114status_t AudioFlinger::RecordThread::releaseAudioPatch_l(const audio_patch_handle_t handle) 7115{ 7116 status_t status = NO_ERROR; 7117 7118 mInDevice = AUDIO_DEVICE_NONE; 7119 7120 if (mInput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 7121 audio_hw_device_t *hwDevice = mInput->audioHwDev->hwDevice(); 7122 status = hwDevice->release_audio_patch(hwDevice, handle); 7123 } else { 7124 AudioParameter param; 7125 param.addInt(String8(AUDIO_PARAMETER_STREAM_ROUTING), 0); 7126 status = mInput->stream->common.set_parameters(&mInput->stream->common, 7127 param.toString().string()); 7128 } 7129 return status; 7130} 7131 7132void AudioFlinger::RecordThread::addPatchRecord(const sp<PatchRecord>& record) 7133{ 7134 Mutex::Autolock _l(mLock); 7135 mTracks.add(record); 7136} 7137 7138void AudioFlinger::RecordThread::deletePatchRecord(const sp<PatchRecord>& record) 7139{ 7140 Mutex::Autolock _l(mLock); 7141 destroyTrack_l(record); 7142} 7143 7144void AudioFlinger::RecordThread::getAudioPortConfig(struct audio_port_config *config) 7145{ 7146 ThreadBase::getAudioPortConfig(config); 7147 config->role = AUDIO_PORT_ROLE_SINK; 7148 config->ext.mix.hw_module = mInput->audioHwDev->handle(); 7149 config->ext.mix.usecase.source = mAudioSource; 7150} 7151 7152} // namespace android 7153