Threads.cpp revision 7c1ec5f038e63a5eb8b04434577c25bc23f5f410
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->stream->common.get_sample_rate(&mOutput->stream->common); 2085 mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common); 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 mHALFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 2096 mFormat = mHALFormat; 2097 if (!audio_is_valid_format(mFormat)) { 2098 LOG_ALWAYS_FATAL("HAL format %#x not valid for output", mFormat); 2099 } 2100 if ((mType == MIXER || mType == DUPLICATING) 2101 && !isValidPcmSinkFormat(mFormat)) { 2102 LOG_FATAL("HAL format %#x not supported for mixed output", 2103 mFormat); 2104 } 2105 mFrameSize = mOutput->getFrameSize(); 2106 mBufferSize = mOutput->stream->common.get_buffer_size(&mOutput->stream->common); 2107 mFrameCount = mBufferSize / mFrameSize; 2108 if (mFrameCount & 15) { 2109 ALOGW("HAL output buffer size is %u frames but AudioMixer requires multiples of 16 frames", 2110 mFrameCount); 2111 } 2112 2113 if ((mOutput->flags & AUDIO_OUTPUT_FLAG_NON_BLOCKING) && 2114 (mOutput->stream->set_callback != NULL)) { 2115 if (mOutput->stream->set_callback(mOutput->stream, 2116 AudioFlinger::PlaybackThread::asyncCallback, this) == 0) { 2117 mUseAsyncWrite = true; 2118 mCallbackThread = new AudioFlinger::AsyncCallbackThread(this); 2119 } 2120 } 2121 2122 mHwSupportsPause = false; 2123 if (mOutput->flags & AUDIO_OUTPUT_FLAG_DIRECT) { 2124 if (mOutput->stream->pause != NULL) { 2125 if (mOutput->stream->resume != NULL) { 2126 mHwSupportsPause = true; 2127 } else { 2128 ALOGW("direct output implements pause but not resume"); 2129 } 2130 } else if (mOutput->stream->resume != NULL) { 2131 ALOGW("direct output implements resume but not pause"); 2132 } 2133 } 2134 if (!mHwSupportsPause && mOutput->flags & AUDIO_OUTPUT_FLAG_HW_AV_SYNC) { 2135 LOG_ALWAYS_FATAL("HW_AV_SYNC requested but HAL does not implement pause and resume"); 2136 } 2137 2138 if (mType == DUPLICATING && mMixerBufferEnabled && mEffectBufferEnabled) { 2139 // For best precision, we use float instead of the associated output 2140 // device format (typically PCM 16 bit). 2141 2142 mFormat = AUDIO_FORMAT_PCM_FLOAT; 2143 mFrameSize = mChannelCount * audio_bytes_per_sample(mFormat); 2144 mBufferSize = mFrameSize * mFrameCount; 2145 2146 // TODO: We currently use the associated output device channel mask and sample rate. 2147 // (1) Perhaps use the ORed channel mask of all downstream MixerThreads 2148 // (if a valid mask) to avoid premature downmix. 2149 // (2) Perhaps use the maximum sample rate of all downstream MixerThreads 2150 // instead of the output device sample rate to avoid loss of high frequency information. 2151 // This may need to be updated as MixerThread/OutputTracks are added and not here. 2152 } 2153 2154 // Calculate size of normal sink buffer relative to the HAL output buffer size 2155 double multiplier = 1.0; 2156 if (mType == MIXER && (kUseFastMixer == FastMixer_Static || 2157 kUseFastMixer == FastMixer_Dynamic)) { 2158 size_t minNormalFrameCount = (kMinNormalSinkBufferSizeMs * mSampleRate) / 1000; 2159 size_t maxNormalFrameCount = (kMaxNormalSinkBufferSizeMs * mSampleRate) / 1000; 2160 // round up minimum and round down maximum to nearest 16 frames to satisfy AudioMixer 2161 minNormalFrameCount = (minNormalFrameCount + 15) & ~15; 2162 maxNormalFrameCount = maxNormalFrameCount & ~15; 2163 if (maxNormalFrameCount < minNormalFrameCount) { 2164 maxNormalFrameCount = minNormalFrameCount; 2165 } 2166 multiplier = (double) minNormalFrameCount / (double) mFrameCount; 2167 if (multiplier <= 1.0) { 2168 multiplier = 1.0; 2169 } else if (multiplier <= 2.0) { 2170 if (2 * mFrameCount <= maxNormalFrameCount) { 2171 multiplier = 2.0; 2172 } else { 2173 multiplier = (double) maxNormalFrameCount / (double) mFrameCount; 2174 } 2175 } else { 2176 // prefer an even multiplier, for compatibility with doubling of fast tracks due to HAL 2177 // SRC (it would be unusual for the normal sink buffer size to not be a multiple of fast 2178 // track, but we sometimes have to do this to satisfy the maximum frame count 2179 // constraint) 2180 // FIXME this rounding up should not be done if no HAL SRC 2181 uint32_t truncMult = (uint32_t) multiplier; 2182 if ((truncMult & 1)) { 2183 if ((truncMult + 1) * mFrameCount <= maxNormalFrameCount) { 2184 ++truncMult; 2185 } 2186 } 2187 multiplier = (double) truncMult; 2188 } 2189 } 2190 mNormalFrameCount = multiplier * mFrameCount; 2191 // round up to nearest 16 frames to satisfy AudioMixer 2192 if (mType == MIXER || mType == DUPLICATING) { 2193 mNormalFrameCount = (mNormalFrameCount + 15) & ~15; 2194 } 2195 ALOGI("HAL output buffer size %u frames, normal sink buffer size %u frames", mFrameCount, 2196 mNormalFrameCount); 2197 2198 // Check if we want to throttle the processing to no more than 2x normal rate 2199 mThreadThrottle = property_get_bool("af.thread.throttle", true /* default_value */); 2200 mThreadThrottleTimeMs = 0; 2201 mThreadThrottleEndMs = 0; 2202 mHalfBufferMs = mNormalFrameCount * 1000 / (2 * mSampleRate); 2203 2204 // mSinkBuffer is the sink buffer. Size is always multiple-of-16 frames. 2205 // Originally this was int16_t[] array, need to remove legacy implications. 2206 free(mSinkBuffer); 2207 mSinkBuffer = NULL; 2208 // For sink buffer size, we use the frame size from the downstream sink to avoid problems 2209 // with non PCM formats for compressed music, e.g. AAC, and Offload threads. 2210 const size_t sinkBufferSize = mNormalFrameCount * mFrameSize; 2211 (void)posix_memalign(&mSinkBuffer, 32, sinkBufferSize); 2212 2213 // We resize the mMixerBuffer according to the requirements of the sink buffer which 2214 // drives the output. 2215 free(mMixerBuffer); 2216 mMixerBuffer = NULL; 2217 if (mMixerBufferEnabled) { 2218 mMixerBufferFormat = AUDIO_FORMAT_PCM_FLOAT; // also valid: AUDIO_FORMAT_PCM_16_BIT. 2219 mMixerBufferSize = mNormalFrameCount * mChannelCount 2220 * audio_bytes_per_sample(mMixerBufferFormat); 2221 (void)posix_memalign(&mMixerBuffer, 32, mMixerBufferSize); 2222 } 2223 free(mEffectBuffer); 2224 mEffectBuffer = NULL; 2225 if (mEffectBufferEnabled) { 2226 mEffectBufferFormat = AUDIO_FORMAT_PCM_16_BIT; // Note: Effects support 16b only 2227 mEffectBufferSize = mNormalFrameCount * mChannelCount 2228 * audio_bytes_per_sample(mEffectBufferFormat); 2229 (void)posix_memalign(&mEffectBuffer, 32, mEffectBufferSize); 2230 } 2231 2232 // force reconfiguration of effect chains and engines to take new buffer size and audio 2233 // parameters into account 2234 // Note that mLock is not held when readOutputParameters_l() is called from the constructor 2235 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 2236 // matter. 2237 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 2238 Vector< sp<EffectChain> > effectChains = mEffectChains; 2239 for (size_t i = 0; i < effectChains.size(); i ++) { 2240 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 2241 } 2242} 2243 2244 2245status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames) 2246{ 2247 if (halFrames == NULL || dspFrames == NULL) { 2248 return BAD_VALUE; 2249 } 2250 Mutex::Autolock _l(mLock); 2251 if (initCheck() != NO_ERROR) { 2252 return INVALID_OPERATION; 2253 } 2254 size_t framesWritten = mBytesWritten / mFrameSize; 2255 *halFrames = framesWritten; 2256 2257 if (isSuspended()) { 2258 // return an estimation of rendered frames when the output is suspended 2259 size_t latencyFrames = (latency_l() * mSampleRate) / 1000; 2260 *dspFrames = framesWritten >= latencyFrames ? framesWritten - latencyFrames : 0; 2261 return NO_ERROR; 2262 } else { 2263 status_t status; 2264 uint32_t frames; 2265 status = mOutput->getRenderPosition(&frames); 2266 *dspFrames = (size_t)frames; 2267 return status; 2268 } 2269} 2270 2271uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) const 2272{ 2273 Mutex::Autolock _l(mLock); 2274 uint32_t result = 0; 2275 if (getEffectChain_l(sessionId) != 0) { 2276 result = EFFECT_SESSION; 2277 } 2278 2279 for (size_t i = 0; i < mTracks.size(); ++i) { 2280 sp<Track> track = mTracks[i]; 2281 if (sessionId == track->sessionId() && !track->isInvalid()) { 2282 result |= TRACK_SESSION; 2283 break; 2284 } 2285 } 2286 2287 return result; 2288} 2289 2290uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 2291{ 2292 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 2293 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 2294 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 2295 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 2296 } 2297 for (size_t i = 0; i < mTracks.size(); i++) { 2298 sp<Track> track = mTracks[i]; 2299 if (sessionId == track->sessionId() && !track->isInvalid()) { 2300 return AudioSystem::getStrategyForStream(track->streamType()); 2301 } 2302 } 2303 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 2304} 2305 2306 2307AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 2308{ 2309 Mutex::Autolock _l(mLock); 2310 return mOutput; 2311} 2312 2313AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 2314{ 2315 Mutex::Autolock _l(mLock); 2316 AudioStreamOut *output = mOutput; 2317 mOutput = NULL; 2318 // FIXME FastMixer might also have a raw ptr to mOutputSink; 2319 // must push a NULL and wait for ack 2320 mOutputSink.clear(); 2321 mPipeSink.clear(); 2322 mNormalSink.clear(); 2323 return output; 2324} 2325 2326// this method must always be called either with ThreadBase mLock held or inside the thread loop 2327audio_stream_t* AudioFlinger::PlaybackThread::stream() const 2328{ 2329 if (mOutput == NULL) { 2330 return NULL; 2331 } 2332 return &mOutput->stream->common; 2333} 2334 2335uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() const 2336{ 2337 return (uint32_t)((uint32_t)((mNormalFrameCount * 1000) / mSampleRate) * 1000); 2338} 2339 2340status_t AudioFlinger::PlaybackThread::setSyncEvent(const sp<SyncEvent>& event) 2341{ 2342 if (!isValidSyncEvent(event)) { 2343 return BAD_VALUE; 2344 } 2345 2346 Mutex::Autolock _l(mLock); 2347 2348 for (size_t i = 0; i < mTracks.size(); ++i) { 2349 sp<Track> track = mTracks[i]; 2350 if (event->triggerSession() == track->sessionId()) { 2351 (void) track->setSyncEvent(event); 2352 return NO_ERROR; 2353 } 2354 } 2355 2356 return NAME_NOT_FOUND; 2357} 2358 2359bool AudioFlinger::PlaybackThread::isValidSyncEvent(const sp<SyncEvent>& event) const 2360{ 2361 return event->type() == AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE; 2362} 2363 2364void AudioFlinger::PlaybackThread::threadLoop_removeTracks( 2365 const Vector< sp<Track> >& tracksToRemove) 2366{ 2367 size_t count = tracksToRemove.size(); 2368 if (count > 0) { 2369 for (size_t i = 0 ; i < count ; i++) { 2370 const sp<Track>& track = tracksToRemove.itemAt(i); 2371 if (track->isExternalTrack()) { 2372 AudioSystem::stopOutput(mId, track->streamType(), 2373 (audio_session_t)track->sessionId()); 2374#ifdef ADD_BATTERY_DATA 2375 // to track the speaker usage 2376 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 2377#endif 2378 if (track->isTerminated()) { 2379 AudioSystem::releaseOutput(mId, track->streamType(), 2380 (audio_session_t)track->sessionId()); 2381 } 2382 } 2383 } 2384 } 2385} 2386 2387void AudioFlinger::PlaybackThread::checkSilentMode_l() 2388{ 2389 if (!mMasterMute) { 2390 char value[PROPERTY_VALUE_MAX]; 2391 if (property_get("ro.audio.silent", value, "0") > 0) { 2392 char *endptr; 2393 unsigned long ul = strtoul(value, &endptr, 0); 2394 if (*endptr == '\0' && ul != 0) { 2395 ALOGD("Silence is golden"); 2396 // The setprop command will not allow a property to be changed after 2397 // the first time it is set, so we don't have to worry about un-muting. 2398 setMasterMute_l(true); 2399 } 2400 } 2401 } 2402} 2403 2404// shared by MIXER and DIRECT, overridden by DUPLICATING 2405ssize_t AudioFlinger::PlaybackThread::threadLoop_write() 2406{ 2407 // FIXME rewrite to reduce number of system calls 2408 mLastWriteTime = systemTime(); 2409 mInWrite = true; 2410 ssize_t bytesWritten; 2411 const size_t offset = mCurrentWriteLength - mBytesRemaining; 2412 2413 // If an NBAIO sink is present, use it to write the normal mixer's submix 2414 if (mNormalSink != 0) { 2415 2416 const size_t count = mBytesRemaining / mFrameSize; 2417 2418 ATRACE_BEGIN("write"); 2419 // update the setpoint when AudioFlinger::mScreenState changes 2420 uint32_t screenState = AudioFlinger::mScreenState; 2421 if (screenState != mScreenState) { 2422 mScreenState = screenState; 2423 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 2424 if (pipe != NULL) { 2425 pipe->setAvgFrames((mScreenState & 1) ? 2426 (pipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2427 } 2428 } 2429 ssize_t framesWritten = mNormalSink->write((char *)mSinkBuffer + offset, count); 2430 ATRACE_END(); 2431 if (framesWritten > 0) { 2432 bytesWritten = framesWritten * mFrameSize; 2433 } else { 2434 bytesWritten = framesWritten; 2435 } 2436 mLatchDValid = false; 2437 status_t status = mNormalSink->getTimestamp(mLatchD.mTimestamp); 2438 if (status == NO_ERROR) { 2439 size_t totalFramesWritten = mNormalSink->framesWritten(); 2440 if (totalFramesWritten >= mLatchD.mTimestamp.mPosition) { 2441 mLatchD.mUnpresentedFrames = totalFramesWritten - mLatchD.mTimestamp.mPosition; 2442 // mLatchD.mFramesReleased is set immediately before D is clocked into Q 2443 mLatchDValid = true; 2444 } 2445 } 2446 // otherwise use the HAL / AudioStreamOut directly 2447 } else { 2448 // Direct output and offload threads 2449 2450 if (mUseAsyncWrite) { 2451 ALOGW_IF(mWriteAckSequence & 1, "threadLoop_write(): out of sequence write request"); 2452 mWriteAckSequence += 2; 2453 mWriteAckSequence |= 1; 2454 ALOG_ASSERT(mCallbackThread != 0); 2455 mCallbackThread->setWriteBlocked(mWriteAckSequence); 2456 } 2457 // FIXME We should have an implementation of timestamps for direct output threads. 2458 // They are used e.g for multichannel PCM playback over HDMI. 2459 bytesWritten = mOutput->write((char *)mSinkBuffer + offset, mBytesRemaining); 2460 if (mUseAsyncWrite && 2461 ((bytesWritten < 0) || (bytesWritten == (ssize_t)mBytesRemaining))) { 2462 // do not wait for async callback in case of error of full write 2463 mWriteAckSequence &= ~1; 2464 ALOG_ASSERT(mCallbackThread != 0); 2465 mCallbackThread->setWriteBlocked(mWriteAckSequence); 2466 } 2467 } 2468 2469 mNumWrites++; 2470 mInWrite = false; 2471 mStandby = false; 2472 return bytesWritten; 2473} 2474 2475void AudioFlinger::PlaybackThread::threadLoop_drain() 2476{ 2477 if (mOutput->stream->drain) { 2478 ALOGV("draining %s", (mMixerStatus == MIXER_DRAIN_TRACK) ? "early" : "full"); 2479 if (mUseAsyncWrite) { 2480 ALOGW_IF(mDrainSequence & 1, "threadLoop_drain(): out of sequence drain request"); 2481 mDrainSequence |= 1; 2482 ALOG_ASSERT(mCallbackThread != 0); 2483 mCallbackThread->setDraining(mDrainSequence); 2484 } 2485 mOutput->stream->drain(mOutput->stream, 2486 (mMixerStatus == MIXER_DRAIN_TRACK) ? AUDIO_DRAIN_EARLY_NOTIFY 2487 : AUDIO_DRAIN_ALL); 2488 } 2489} 2490 2491void AudioFlinger::PlaybackThread::threadLoop_exit() 2492{ 2493 { 2494 Mutex::Autolock _l(mLock); 2495 for (size_t i = 0; i < mTracks.size(); i++) { 2496 sp<Track> track = mTracks[i]; 2497 track->invalidate(); 2498 } 2499 } 2500} 2501 2502/* 2503The derived values that are cached: 2504 - mSinkBufferSize from frame count * frame size 2505 - mActiveSleepTimeUs from activeSleepTimeUs() 2506 - mIdleSleepTimeUs from idleSleepTimeUs() 2507 - mStandbyDelayNs from mActiveSleepTimeUs (DIRECT only) 2508 - maxPeriod from frame count and sample rate (MIXER only) 2509 2510The parameters that affect these derived values are: 2511 - frame count 2512 - frame size 2513 - sample rate 2514 - device type: A2DP or not 2515 - device latency 2516 - format: PCM or not 2517 - active sleep time 2518 - idle sleep time 2519*/ 2520 2521void AudioFlinger::PlaybackThread::cacheParameters_l() 2522{ 2523 mSinkBufferSize = mNormalFrameCount * mFrameSize; 2524 mActiveSleepTimeUs = activeSleepTimeUs(); 2525 mIdleSleepTimeUs = idleSleepTimeUs(); 2526} 2527 2528void AudioFlinger::PlaybackThread::invalidateTracks(audio_stream_type_t streamType) 2529{ 2530 ALOGV("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 2531 this, streamType, mTracks.size()); 2532 Mutex::Autolock _l(mLock); 2533 2534 size_t size = mTracks.size(); 2535 for (size_t i = 0; i < size; i++) { 2536 sp<Track> t = mTracks[i]; 2537 if (t->streamType() == streamType) { 2538 t->invalidate(); 2539 } 2540 } 2541} 2542 2543status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 2544{ 2545 int session = chain->sessionId(); 2546 int16_t* buffer = reinterpret_cast<int16_t*>(mEffectBufferEnabled 2547 ? mEffectBuffer : mSinkBuffer); 2548 bool ownsBuffer = false; 2549 2550 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 2551 if (session > 0) { 2552 // Only one effect chain can be present in direct output thread and it uses 2553 // the sink buffer as input 2554 if (mType != DIRECT) { 2555 size_t numSamples = mNormalFrameCount * mChannelCount; 2556 buffer = new int16_t[numSamples]; 2557 memset(buffer, 0, numSamples * sizeof(int16_t)); 2558 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 2559 ownsBuffer = true; 2560 } 2561 2562 // Attach all tracks with same session ID to this chain. 2563 for (size_t i = 0; i < mTracks.size(); ++i) { 2564 sp<Track> track = mTracks[i]; 2565 if (session == track->sessionId()) { 2566 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), 2567 buffer); 2568 track->setMainBuffer(buffer); 2569 chain->incTrackCnt(); 2570 } 2571 } 2572 2573 // indicate all active tracks in the chain 2574 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 2575 sp<Track> track = mActiveTracks[i].promote(); 2576 if (track == 0) { 2577 continue; 2578 } 2579 if (session == track->sessionId()) { 2580 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 2581 chain->incActiveTrackCnt(); 2582 } 2583 } 2584 } 2585 chain->setThread(this); 2586 chain->setInBuffer(buffer, ownsBuffer); 2587 chain->setOutBuffer(reinterpret_cast<int16_t*>(mEffectBufferEnabled 2588 ? mEffectBuffer : mSinkBuffer)); 2589 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 2590 // chains list in order to be processed last as it contains output stage effects 2591 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 2592 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 2593 // after track specific effects and before output stage 2594 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 2595 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 2596 // Effect chain for other sessions are inserted at beginning of effect 2597 // chains list to be processed before output mix effects. Relative order between other 2598 // sessions is not important 2599 size_t size = mEffectChains.size(); 2600 size_t i = 0; 2601 for (i = 0; i < size; i++) { 2602 if (mEffectChains[i]->sessionId() < session) { 2603 break; 2604 } 2605 } 2606 mEffectChains.insertAt(chain, i); 2607 checkSuspendOnAddEffectChain_l(chain); 2608 2609 return NO_ERROR; 2610} 2611 2612size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 2613{ 2614 int session = chain->sessionId(); 2615 2616 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 2617 2618 for (size_t i = 0; i < mEffectChains.size(); i++) { 2619 if (chain == mEffectChains[i]) { 2620 mEffectChains.removeAt(i); 2621 // detach all active tracks from the chain 2622 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 2623 sp<Track> track = mActiveTracks[i].promote(); 2624 if (track == 0) { 2625 continue; 2626 } 2627 if (session == track->sessionId()) { 2628 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 2629 chain.get(), session); 2630 chain->decActiveTrackCnt(); 2631 } 2632 } 2633 2634 // detach all tracks with same session ID from this chain 2635 for (size_t i = 0; i < mTracks.size(); ++i) { 2636 sp<Track> track = mTracks[i]; 2637 if (session == track->sessionId()) { 2638 track->setMainBuffer(reinterpret_cast<int16_t*>(mSinkBuffer)); 2639 chain->decTrackCnt(); 2640 } 2641 } 2642 break; 2643 } 2644 } 2645 return mEffectChains.size(); 2646} 2647 2648status_t AudioFlinger::PlaybackThread::attachAuxEffect( 2649 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2650{ 2651 Mutex::Autolock _l(mLock); 2652 return attachAuxEffect_l(track, EffectId); 2653} 2654 2655status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 2656 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2657{ 2658 status_t status = NO_ERROR; 2659 2660 if (EffectId == 0) { 2661 track->setAuxBuffer(0, NULL); 2662 } else { 2663 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 2664 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 2665 if (effect != 0) { 2666 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 2667 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 2668 } else { 2669 status = INVALID_OPERATION; 2670 } 2671 } else { 2672 status = BAD_VALUE; 2673 } 2674 } 2675 return status; 2676} 2677 2678void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 2679{ 2680 for (size_t i = 0; i < mTracks.size(); ++i) { 2681 sp<Track> track = mTracks[i]; 2682 if (track->auxEffectId() == effectId) { 2683 attachAuxEffect_l(track, 0); 2684 } 2685 } 2686} 2687 2688bool AudioFlinger::PlaybackThread::threadLoop() 2689{ 2690 Vector< sp<Track> > tracksToRemove; 2691 2692 mStandbyTimeNs = systemTime(); 2693 2694 // MIXER 2695 nsecs_t lastWarning = 0; 2696 2697 // DUPLICATING 2698 // FIXME could this be made local to while loop? 2699 writeFrames = 0; 2700 2701 int lastGeneration = 0; 2702 2703 cacheParameters_l(); 2704 mSleepTimeUs = mIdleSleepTimeUs; 2705 2706 if (mType == MIXER) { 2707 sleepTimeShift = 0; 2708 } 2709 2710 CpuStats cpuStats; 2711 const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid())); 2712 2713 acquireWakeLock(); 2714 2715 // mNBLogWriter->log can only be called while thread mutex mLock is held. 2716 // So if you need to log when mutex is unlocked, set logString to a non-NULL string, 2717 // and then that string will be logged at the next convenient opportunity. 2718 const char *logString = NULL; 2719 2720 checkSilentMode_l(); 2721 2722 while (!exitPending()) 2723 { 2724 cpuStats.sample(myName); 2725 2726 Vector< sp<EffectChain> > effectChains; 2727 2728 { // scope for mLock 2729 2730 Mutex::Autolock _l(mLock); 2731 2732 processConfigEvents_l(); 2733 2734 if (logString != NULL) { 2735 mNBLogWriter->logTimestamp(); 2736 mNBLogWriter->log(logString); 2737 logString = NULL; 2738 } 2739 2740 // Gather the framesReleased counters for all active tracks, 2741 // and latch them atomically with the timestamp. 2742 // FIXME We're using raw pointers as indices. A unique track ID would be a better index. 2743 mLatchD.mFramesReleased.clear(); 2744 size_t size = mActiveTracks.size(); 2745 for (size_t i = 0; i < size; i++) { 2746 sp<Track> t = mActiveTracks[i].promote(); 2747 if (t != 0) { 2748 mLatchD.mFramesReleased.add(t.get(), 2749 t->mAudioTrackServerProxy->framesReleased()); 2750 } 2751 } 2752 if (mLatchDValid) { 2753 mLatchQ = mLatchD; 2754 mLatchDValid = false; 2755 mLatchQValid = true; 2756 } 2757 2758 saveOutputTracks(); 2759 if (mSignalPending) { 2760 // A signal was raised while we were unlocked 2761 mSignalPending = false; 2762 } else if (waitingAsyncCallback_l()) { 2763 if (exitPending()) { 2764 break; 2765 } 2766 bool released = false; 2767 // The following works around a bug in the offload driver. Ideally we would release 2768 // the wake lock every time, but that causes the last offload buffer(s) to be 2769 // dropped while the device is on battery, so we need to hold a wake lock during 2770 // the drain phase. 2771 if (mBytesRemaining && !(mDrainSequence & 1)) { 2772 releaseWakeLock_l(); 2773 released = true; 2774 } 2775 mWakeLockUids.clear(); 2776 mActiveTracksGeneration++; 2777 ALOGV("wait async completion"); 2778 mWaitWorkCV.wait(mLock); 2779 ALOGV("async completion/wake"); 2780 if (released) { 2781 acquireWakeLock_l(); 2782 } 2783 mStandbyTimeNs = systemTime() + mStandbyDelayNs; 2784 mSleepTimeUs = 0; 2785 2786 continue; 2787 } 2788 if ((!mActiveTracks.size() && systemTime() > mStandbyTimeNs) || 2789 isSuspended()) { 2790 // put audio hardware into standby after short delay 2791 if (shouldStandby_l()) { 2792 2793 threadLoop_standby(); 2794 2795 mStandby = true; 2796 } 2797 2798 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2799 // we're about to wait, flush the binder command buffer 2800 IPCThreadState::self()->flushCommands(); 2801 2802 clearOutputTracks(); 2803 2804 if (exitPending()) { 2805 break; 2806 } 2807 2808 releaseWakeLock_l(); 2809 mWakeLockUids.clear(); 2810 mActiveTracksGeneration++; 2811 // wait until we have something to do... 2812 ALOGV("%s going to sleep", myName.string()); 2813 mWaitWorkCV.wait(mLock); 2814 ALOGV("%s waking up", myName.string()); 2815 acquireWakeLock_l(); 2816 2817 mMixerStatus = MIXER_IDLE; 2818 mMixerStatusIgnoringFastTracks = MIXER_IDLE; 2819 mBytesWritten = 0; 2820 mBytesRemaining = 0; 2821 checkSilentMode_l(); 2822 2823 mStandbyTimeNs = systemTime() + mStandbyDelayNs; 2824 mSleepTimeUs = mIdleSleepTimeUs; 2825 if (mType == MIXER) { 2826 sleepTimeShift = 0; 2827 } 2828 2829 continue; 2830 } 2831 } 2832 // mMixerStatusIgnoringFastTracks is also updated internally 2833 mMixerStatus = prepareTracks_l(&tracksToRemove); 2834 2835 // compare with previously applied list 2836 if (lastGeneration != mActiveTracksGeneration) { 2837 // update wakelock 2838 updateWakeLockUids_l(mWakeLockUids); 2839 lastGeneration = mActiveTracksGeneration; 2840 } 2841 2842 // prevent any changes in effect chain list and in each effect chain 2843 // during mixing and effect process as the audio buffers could be deleted 2844 // or modified if an effect is created or deleted 2845 lockEffectChains_l(effectChains); 2846 } // mLock scope ends 2847 2848 if (mBytesRemaining == 0) { 2849 mCurrentWriteLength = 0; 2850 if (mMixerStatus == MIXER_TRACKS_READY) { 2851 // threadLoop_mix() sets mCurrentWriteLength 2852 threadLoop_mix(); 2853 } else if ((mMixerStatus != MIXER_DRAIN_TRACK) 2854 && (mMixerStatus != MIXER_DRAIN_ALL)) { 2855 // threadLoop_sleepTime sets mSleepTimeUs to 0 if data 2856 // must be written to HAL 2857 threadLoop_sleepTime(); 2858 if (mSleepTimeUs == 0) { 2859 mCurrentWriteLength = mSinkBufferSize; 2860 } 2861 } 2862 // Either threadLoop_mix() or threadLoop_sleepTime() should have set 2863 // mMixerBuffer with data if mMixerBufferValid is true and mSleepTimeUs == 0. 2864 // Merge mMixerBuffer data into mEffectBuffer (if any effects are valid) 2865 // or mSinkBuffer (if there are no effects). 2866 // 2867 // This is done pre-effects computation; if effects change to 2868 // support higher precision, this needs to move. 2869 // 2870 // mMixerBufferValid is only set true by MixerThread::prepareTracks_l(). 2871 // TODO use mSleepTimeUs == 0 as an additional condition. 2872 if (mMixerBufferValid) { 2873 void *buffer = mEffectBufferValid ? mEffectBuffer : mSinkBuffer; 2874 audio_format_t format = mEffectBufferValid ? mEffectBufferFormat : mFormat; 2875 2876 memcpy_by_audio_format(buffer, format, mMixerBuffer, mMixerBufferFormat, 2877 mNormalFrameCount * mChannelCount); 2878 } 2879 2880 mBytesRemaining = mCurrentWriteLength; 2881 if (isSuspended()) { 2882 mSleepTimeUs = suspendSleepTimeUs(); 2883 // simulate write to HAL when suspended 2884 mBytesWritten += mSinkBufferSize; 2885 mBytesRemaining = 0; 2886 } 2887 2888 // only process effects if we're going to write 2889 if (mSleepTimeUs == 0 && mType != OFFLOAD) { 2890 for (size_t i = 0; i < effectChains.size(); i ++) { 2891 effectChains[i]->process_l(); 2892 } 2893 } 2894 } 2895 // Process effect chains for offloaded thread even if no audio 2896 // was read from audio track: process only updates effect state 2897 // and thus does have to be synchronized with audio writes but may have 2898 // to be called while waiting for async write callback 2899 if (mType == OFFLOAD) { 2900 for (size_t i = 0; i < effectChains.size(); i ++) { 2901 effectChains[i]->process_l(); 2902 } 2903 } 2904 2905 // Only if the Effects buffer is enabled and there is data in the 2906 // Effects buffer (buffer valid), we need to 2907 // copy into the sink buffer. 2908 // TODO use mSleepTimeUs == 0 as an additional condition. 2909 if (mEffectBufferValid) { 2910 //ALOGV("writing effect buffer to sink buffer format %#x", mFormat); 2911 memcpy_by_audio_format(mSinkBuffer, mFormat, mEffectBuffer, mEffectBufferFormat, 2912 mNormalFrameCount * mChannelCount); 2913 } 2914 2915 // enable changes in effect chain 2916 unlockEffectChains(effectChains); 2917 2918 if (!waitingAsyncCallback()) { 2919 // mSleepTimeUs == 0 means we must write to audio hardware 2920 if (mSleepTimeUs == 0) { 2921 ssize_t ret = 0; 2922 if (mBytesRemaining) { 2923 ret = threadLoop_write(); 2924 if (ret < 0) { 2925 mBytesRemaining = 0; 2926 } else { 2927 mBytesWritten += ret; 2928 mBytesRemaining -= ret; 2929 } 2930 } else if ((mMixerStatus == MIXER_DRAIN_TRACK) || 2931 (mMixerStatus == MIXER_DRAIN_ALL)) { 2932 threadLoop_drain(); 2933 } 2934 if (mType == MIXER && !mStandby) { 2935 // write blocked detection 2936 nsecs_t now = systemTime(); 2937 nsecs_t delta = now - mLastWriteTime; 2938 if (delta > maxPeriod) { 2939 mNumDelayedWrites++; 2940 if ((now - lastWarning) > kWarningThrottleNs) { 2941 ATRACE_NAME("underrun"); 2942 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2943 ns2ms(delta), mNumDelayedWrites, this); 2944 lastWarning = now; 2945 } 2946 } 2947 2948 if (mThreadThrottle 2949 && mMixerStatus == MIXER_TRACKS_READY // we are mixing (active tracks) 2950 && ret > 0) { // we wrote something 2951 // Limit MixerThread data processing to no more than twice the 2952 // expected processing rate. 2953 // 2954 // This helps prevent underruns with NuPlayer and other applications 2955 // which may set up buffers that are close to the minimum size, or use 2956 // deep buffers, and rely on a double-buffering sleep strategy to fill. 2957 // 2958 // The throttle smooths out sudden large data drains from the device, 2959 // e.g. when it comes out of standby, which often causes problems with 2960 // (1) mixer threads without a fast mixer (which has its own warm-up) 2961 // (2) minimum buffer sized tracks (even if the track is full, 2962 // the app won't fill fast enough to handle the sudden draw). 2963 2964 const int32_t deltaMs = delta / 1000000; 2965 const int32_t throttleMs = mHalfBufferMs - deltaMs; 2966 if ((signed)mHalfBufferMs >= throttleMs && throttleMs > 0) { 2967 usleep(throttleMs * 1000); 2968 // notify of throttle start on verbose log 2969 ALOGV_IF(mThreadThrottleEndMs == mThreadThrottleTimeMs, 2970 "mixer(%p) throttle begin:" 2971 " ret(%zd) deltaMs(%d) requires sleep %d ms", 2972 this, ret, deltaMs, throttleMs); 2973 mThreadThrottleTimeMs += throttleMs; 2974 } else { 2975 uint32_t diff = mThreadThrottleTimeMs - mThreadThrottleEndMs; 2976 if (diff > 0) { 2977 // notify of throttle end on debug log 2978 ALOGD("mixer(%p) throttle end: throttle time(%u)", this, diff); 2979 mThreadThrottleEndMs = mThreadThrottleTimeMs; 2980 } 2981 } 2982 } 2983 } 2984 2985 } else { 2986 ATRACE_BEGIN("sleep"); 2987 usleep(mSleepTimeUs); 2988 ATRACE_END(); 2989 } 2990 } 2991 2992 // Finally let go of removed track(s), without the lock held 2993 // since we can't guarantee the destructors won't acquire that 2994 // same lock. This will also mutate and push a new fast mixer state. 2995 threadLoop_removeTracks(tracksToRemove); 2996 tracksToRemove.clear(); 2997 2998 // FIXME I don't understand the need for this here; 2999 // it was in the original code but maybe the 3000 // assignment in saveOutputTracks() makes this unnecessary? 3001 clearOutputTracks(); 3002 3003 // Effect chains will be actually deleted here if they were removed from 3004 // mEffectChains list during mixing or effects processing 3005 effectChains.clear(); 3006 3007 // FIXME Note that the above .clear() is no longer necessary since effectChains 3008 // is now local to this block, but will keep it for now (at least until merge done). 3009 } 3010 3011 threadLoop_exit(); 3012 3013 if (!mStandby) { 3014 threadLoop_standby(); 3015 mStandby = true; 3016 } 3017 3018 releaseWakeLock(); 3019 mWakeLockUids.clear(); 3020 mActiveTracksGeneration++; 3021 3022 ALOGV("Thread %p type %d exiting", this, mType); 3023 return false; 3024} 3025 3026// removeTracks_l() must be called with ThreadBase::mLock held 3027void AudioFlinger::PlaybackThread::removeTracks_l(const Vector< sp<Track> >& tracksToRemove) 3028{ 3029 size_t count = tracksToRemove.size(); 3030 if (count > 0) { 3031 for (size_t i=0 ; i<count ; i++) { 3032 const sp<Track>& track = tracksToRemove.itemAt(i); 3033 mActiveTracks.remove(track); 3034 mWakeLockUids.remove(track->uid()); 3035 mActiveTracksGeneration++; 3036 ALOGV("removeTracks_l removing track on session %d", track->sessionId()); 3037 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 3038 if (chain != 0) { 3039 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), 3040 track->sessionId()); 3041 chain->decActiveTrackCnt(); 3042 } 3043 if (track->isTerminated()) { 3044 removeTrack_l(track); 3045 } 3046 } 3047 } 3048 3049} 3050 3051status_t AudioFlinger::PlaybackThread::getTimestamp_l(AudioTimestamp& timestamp) 3052{ 3053 if (mNormalSink != 0) { 3054 return mNormalSink->getTimestamp(timestamp); 3055 } 3056 if ((mType == OFFLOAD || mType == DIRECT) 3057 && mOutput != NULL && mOutput->stream->get_presentation_position) { 3058 uint64_t position64; 3059 int ret = mOutput->getPresentationPosition(&position64, ×tamp.mTime); 3060 if (ret == 0) { 3061 timestamp.mPosition = (uint32_t)position64; 3062 return NO_ERROR; 3063 } 3064 } 3065 return INVALID_OPERATION; 3066} 3067 3068status_t AudioFlinger::MixerThread::createAudioPatch_l(const struct audio_patch *patch, 3069 audio_patch_handle_t *handle) 3070{ 3071 // if !&IDLE, holds the FastMixer state to restore after new parameters processed 3072 FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE; 3073 if (mFastMixer != 0) { 3074 FastMixerStateQueue *sq = mFastMixer->sq(); 3075 FastMixerState *state = sq->begin(); 3076 if (!(state->mCommand & FastMixerState::IDLE)) { 3077 previousCommand = state->mCommand; 3078 state->mCommand = FastMixerState::HOT_IDLE; 3079 sq->end(); 3080 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3081 } else { 3082 sq->end(false /*didModify*/); 3083 } 3084 } 3085 status_t status = PlaybackThread::createAudioPatch_l(patch, handle); 3086 3087 if (!(previousCommand & FastMixerState::IDLE)) { 3088 ALOG_ASSERT(mFastMixer != 0); 3089 FastMixerStateQueue *sq = mFastMixer->sq(); 3090 FastMixerState *state = sq->begin(); 3091 ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE); 3092 state->mCommand = previousCommand; 3093 sq->end(); 3094 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3095 } 3096 3097 return status; 3098} 3099 3100status_t AudioFlinger::PlaybackThread::createAudioPatch_l(const struct audio_patch *patch, 3101 audio_patch_handle_t *handle) 3102{ 3103 status_t status = NO_ERROR; 3104 3105 // store new device and send to effects 3106 audio_devices_t type = AUDIO_DEVICE_NONE; 3107 for (unsigned int i = 0; i < patch->num_sinks; i++) { 3108 type |= patch->sinks[i].ext.device.type; 3109 } 3110 3111#ifdef ADD_BATTERY_DATA 3112 // when changing the audio output device, call addBatteryData to notify 3113 // the change 3114 if (mOutDevice != type) { 3115 uint32_t params = 0; 3116 // check whether speaker is on 3117 if (type & AUDIO_DEVICE_OUT_SPEAKER) { 3118 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 3119 } 3120 3121 audio_devices_t deviceWithoutSpeaker 3122 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 3123 // check if any other device (except speaker) is on 3124 if (type & deviceWithoutSpeaker) { 3125 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 3126 } 3127 3128 if (params != 0) { 3129 addBatteryData(params); 3130 } 3131 } 3132#endif 3133 3134 for (size_t i = 0; i < mEffectChains.size(); i++) { 3135 mEffectChains[i]->setDevice_l(type); 3136 } 3137 3138 // mPrevOutDevice is the latest device set by createAudioPatch_l(). It is not set when 3139 // the thread is created so that the first patch creation triggers an ioConfigChanged callback 3140 bool configChanged = mPrevOutDevice != type; 3141 mOutDevice = type; 3142 mPatch = *patch; 3143 3144 if (mOutput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 3145 audio_hw_device_t *hwDevice = mOutput->audioHwDev->hwDevice(); 3146 status = hwDevice->create_audio_patch(hwDevice, 3147 patch->num_sources, 3148 patch->sources, 3149 patch->num_sinks, 3150 patch->sinks, 3151 handle); 3152 } else { 3153 char *address; 3154 if (strcmp(patch->sinks[0].ext.device.address, "") != 0) { 3155 //FIXME: we only support address on first sink with HAL version < 3.0 3156 address = audio_device_address_to_parameter( 3157 patch->sinks[0].ext.device.type, 3158 patch->sinks[0].ext.device.address); 3159 } else { 3160 address = (char *)calloc(1, 1); 3161 } 3162 AudioParameter param = AudioParameter(String8(address)); 3163 free(address); 3164 param.addInt(String8(AUDIO_PARAMETER_STREAM_ROUTING), (int)type); 3165 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3166 param.toString().string()); 3167 *handle = AUDIO_PATCH_HANDLE_NONE; 3168 } 3169 if (configChanged) { 3170 mPrevOutDevice = type; 3171 sendIoConfigEvent_l(AUDIO_OUTPUT_CONFIG_CHANGED); 3172 } 3173 return status; 3174} 3175 3176status_t AudioFlinger::MixerThread::releaseAudioPatch_l(const audio_patch_handle_t handle) 3177{ 3178 // if !&IDLE, holds the FastMixer state to restore after new parameters processed 3179 FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE; 3180 if (mFastMixer != 0) { 3181 FastMixerStateQueue *sq = mFastMixer->sq(); 3182 FastMixerState *state = sq->begin(); 3183 if (!(state->mCommand & FastMixerState::IDLE)) { 3184 previousCommand = state->mCommand; 3185 state->mCommand = FastMixerState::HOT_IDLE; 3186 sq->end(); 3187 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3188 } else { 3189 sq->end(false /*didModify*/); 3190 } 3191 } 3192 3193 status_t status = PlaybackThread::releaseAudioPatch_l(handle); 3194 3195 if (!(previousCommand & FastMixerState::IDLE)) { 3196 ALOG_ASSERT(mFastMixer != 0); 3197 FastMixerStateQueue *sq = mFastMixer->sq(); 3198 FastMixerState *state = sq->begin(); 3199 ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE); 3200 state->mCommand = previousCommand; 3201 sq->end(); 3202 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3203 } 3204 3205 return status; 3206} 3207 3208status_t AudioFlinger::PlaybackThread::releaseAudioPatch_l(const audio_patch_handle_t handle) 3209{ 3210 status_t status = NO_ERROR; 3211 3212 mOutDevice = AUDIO_DEVICE_NONE; 3213 3214 if (mOutput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 3215 audio_hw_device_t *hwDevice = mOutput->audioHwDev->hwDevice(); 3216 status = hwDevice->release_audio_patch(hwDevice, handle); 3217 } else { 3218 AudioParameter param; 3219 param.addInt(String8(AUDIO_PARAMETER_STREAM_ROUTING), 0); 3220 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3221 param.toString().string()); 3222 } 3223 return status; 3224} 3225 3226void AudioFlinger::PlaybackThread::addPatchTrack(const sp<PatchTrack>& track) 3227{ 3228 Mutex::Autolock _l(mLock); 3229 mTracks.add(track); 3230} 3231 3232void AudioFlinger::PlaybackThread::deletePatchTrack(const sp<PatchTrack>& track) 3233{ 3234 Mutex::Autolock _l(mLock); 3235 destroyTrack_l(track); 3236} 3237 3238void AudioFlinger::PlaybackThread::getAudioPortConfig(struct audio_port_config *config) 3239{ 3240 ThreadBase::getAudioPortConfig(config); 3241 config->role = AUDIO_PORT_ROLE_SOURCE; 3242 config->ext.mix.hw_module = mOutput->audioHwDev->handle(); 3243 config->ext.mix.usecase.stream = AUDIO_STREAM_DEFAULT; 3244} 3245 3246// ---------------------------------------------------------------------------- 3247 3248AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 3249 audio_io_handle_t id, audio_devices_t device, bool systemReady, type_t type) 3250 : PlaybackThread(audioFlinger, output, id, device, type, systemReady), 3251 // mAudioMixer below 3252 // mFastMixer below 3253 mFastMixerFutex(0) 3254 // mOutputSink below 3255 // mPipeSink below 3256 // mNormalSink below 3257{ 3258 ALOGV("MixerThread() id=%d device=%#x type=%d", id, device, type); 3259 ALOGV("mSampleRate=%u, mChannelMask=%#x, mChannelCount=%u, mFormat=%d, mFrameSize=%u, " 3260 "mFrameCount=%d, mNormalFrameCount=%d", 3261 mSampleRate, mChannelMask, mChannelCount, mFormat, mFrameSize, mFrameCount, 3262 mNormalFrameCount); 3263 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 3264 3265 if (type == DUPLICATING) { 3266 // The Duplicating thread uses the AudioMixer and delivers data to OutputTracks 3267 // (downstream MixerThreads) in DuplicatingThread::threadLoop_write(). 3268 // Do not create or use mFastMixer, mOutputSink, mPipeSink, or mNormalSink. 3269 return; 3270 } 3271 // create an NBAIO sink for the HAL output stream, and negotiate 3272 mOutputSink = new AudioStreamOutSink(output->stream); 3273 size_t numCounterOffers = 0; 3274 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount, mFormat)}; 3275 ssize_t index = mOutputSink->negotiate(offers, 1, NULL, numCounterOffers); 3276 ALOG_ASSERT(index == 0); 3277 3278 // initialize fast mixer depending on configuration 3279 bool initFastMixer; 3280 switch (kUseFastMixer) { 3281 case FastMixer_Never: 3282 initFastMixer = false; 3283 break; 3284 case FastMixer_Always: 3285 initFastMixer = true; 3286 break; 3287 case FastMixer_Static: 3288 case FastMixer_Dynamic: 3289 initFastMixer = mFrameCount < mNormalFrameCount; 3290 break; 3291 } 3292 if (initFastMixer) { 3293 audio_format_t fastMixerFormat; 3294 if (mMixerBufferEnabled && mEffectBufferEnabled) { 3295 fastMixerFormat = AUDIO_FORMAT_PCM_FLOAT; 3296 } else { 3297 fastMixerFormat = AUDIO_FORMAT_PCM_16_BIT; 3298 } 3299 if (mFormat != fastMixerFormat) { 3300 // change our Sink format to accept our intermediate precision 3301 mFormat = fastMixerFormat; 3302 free(mSinkBuffer); 3303 mFrameSize = mChannelCount * audio_bytes_per_sample(mFormat); 3304 const size_t sinkBufferSize = mNormalFrameCount * mFrameSize; 3305 (void)posix_memalign(&mSinkBuffer, 32, sinkBufferSize); 3306 } 3307 3308 // create a MonoPipe to connect our submix to FastMixer 3309 NBAIO_Format format = mOutputSink->format(); 3310 NBAIO_Format origformat = format; 3311 // adjust format to match that of the Fast Mixer 3312 ALOGV("format changed from %d to %d", format.mFormat, fastMixerFormat); 3313 format.mFormat = fastMixerFormat; 3314 format.mFrameSize = audio_bytes_per_sample(format.mFormat) * format.mChannelCount; 3315 3316 // This pipe depth compensates for scheduling latency of the normal mixer thread. 3317 // When it wakes up after a maximum latency, it runs a few cycles quickly before 3318 // finally blocking. Note the pipe implementation rounds up the request to a power of 2. 3319 MonoPipe *monoPipe = new MonoPipe(mNormalFrameCount * 4, format, true /*writeCanBlock*/); 3320 const NBAIO_Format offers[1] = {format}; 3321 size_t numCounterOffers = 0; 3322 ssize_t index = monoPipe->negotiate(offers, 1, NULL, numCounterOffers); 3323 ALOG_ASSERT(index == 0); 3324 monoPipe->setAvgFrames((mScreenState & 1) ? 3325 (monoPipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 3326 mPipeSink = monoPipe; 3327 3328#ifdef TEE_SINK 3329 if (mTeeSinkOutputEnabled) { 3330 // create a Pipe to archive a copy of FastMixer's output for dumpsys 3331 Pipe *teeSink = new Pipe(mTeeSinkOutputFrames, origformat); 3332 const NBAIO_Format offers2[1] = {origformat}; 3333 numCounterOffers = 0; 3334 index = teeSink->negotiate(offers2, 1, NULL, numCounterOffers); 3335 ALOG_ASSERT(index == 0); 3336 mTeeSink = teeSink; 3337 PipeReader *teeSource = new PipeReader(*teeSink); 3338 numCounterOffers = 0; 3339 index = teeSource->negotiate(offers2, 1, NULL, numCounterOffers); 3340 ALOG_ASSERT(index == 0); 3341 mTeeSource = teeSource; 3342 } 3343#endif 3344 3345 // create fast mixer and configure it initially with just one fast track for our submix 3346 mFastMixer = new FastMixer(); 3347 FastMixerStateQueue *sq = mFastMixer->sq(); 3348#ifdef STATE_QUEUE_DUMP 3349 sq->setObserverDump(&mStateQueueObserverDump); 3350 sq->setMutatorDump(&mStateQueueMutatorDump); 3351#endif 3352 FastMixerState *state = sq->begin(); 3353 FastTrack *fastTrack = &state->mFastTracks[0]; 3354 // wrap the source side of the MonoPipe to make it an AudioBufferProvider 3355 fastTrack->mBufferProvider = new SourceAudioBufferProvider(new MonoPipeReader(monoPipe)); 3356 fastTrack->mVolumeProvider = NULL; 3357 fastTrack->mChannelMask = mChannelMask; // mPipeSink channel mask for audio to FastMixer 3358 fastTrack->mFormat = mFormat; // mPipeSink format for audio to FastMixer 3359 fastTrack->mGeneration++; 3360 state->mFastTracksGen++; 3361 state->mTrackMask = 1; 3362 // fast mixer will use the HAL output sink 3363 state->mOutputSink = mOutputSink.get(); 3364 state->mOutputSinkGen++; 3365 state->mFrameCount = mFrameCount; 3366 state->mCommand = FastMixerState::COLD_IDLE; 3367 // already done in constructor initialization list 3368 //mFastMixerFutex = 0; 3369 state->mColdFutexAddr = &mFastMixerFutex; 3370 state->mColdGen++; 3371 state->mDumpState = &mFastMixerDumpState; 3372#ifdef TEE_SINK 3373 state->mTeeSink = mTeeSink.get(); 3374#endif 3375 mFastMixerNBLogWriter = audioFlinger->newWriter_l(kFastMixerLogSize, "FastMixer"); 3376 state->mNBLogWriter = mFastMixerNBLogWriter.get(); 3377 sq->end(); 3378 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3379 3380 // start the fast mixer 3381 mFastMixer->run("FastMixer", PRIORITY_URGENT_AUDIO); 3382 pid_t tid = mFastMixer->getTid(); 3383 sendPrioConfigEvent(getpid_cached, tid, kPriorityFastMixer); 3384 3385#ifdef AUDIO_WATCHDOG 3386 // create and start the watchdog 3387 mAudioWatchdog = new AudioWatchdog(); 3388 mAudioWatchdog->setDump(&mAudioWatchdogDump); 3389 mAudioWatchdog->run("AudioWatchdog", PRIORITY_URGENT_AUDIO); 3390 tid = mAudioWatchdog->getTid(); 3391 sendPrioConfigEvent(getpid_cached, tid, kPriorityFastMixer); 3392#endif 3393 3394 } 3395 3396 switch (kUseFastMixer) { 3397 case FastMixer_Never: 3398 case FastMixer_Dynamic: 3399 mNormalSink = mOutputSink; 3400 break; 3401 case FastMixer_Always: 3402 mNormalSink = mPipeSink; 3403 break; 3404 case FastMixer_Static: 3405 mNormalSink = initFastMixer ? mPipeSink : mOutputSink; 3406 break; 3407 } 3408} 3409 3410AudioFlinger::MixerThread::~MixerThread() 3411{ 3412 if (mFastMixer != 0) { 3413 FastMixerStateQueue *sq = mFastMixer->sq(); 3414 FastMixerState *state = sq->begin(); 3415 if (state->mCommand == FastMixerState::COLD_IDLE) { 3416 int32_t old = android_atomic_inc(&mFastMixerFutex); 3417 if (old == -1) { 3418 (void) syscall(__NR_futex, &mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 3419 } 3420 } 3421 state->mCommand = FastMixerState::EXIT; 3422 sq->end(); 3423 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3424 mFastMixer->join(); 3425 // Though the fast mixer thread has exited, it's state queue is still valid. 3426 // We'll use that extract the final state which contains one remaining fast track 3427 // corresponding to our sub-mix. 3428 state = sq->begin(); 3429 ALOG_ASSERT(state->mTrackMask == 1); 3430 FastTrack *fastTrack = &state->mFastTracks[0]; 3431 ALOG_ASSERT(fastTrack->mBufferProvider != NULL); 3432 delete fastTrack->mBufferProvider; 3433 sq->end(false /*didModify*/); 3434 mFastMixer.clear(); 3435#ifdef AUDIO_WATCHDOG 3436 if (mAudioWatchdog != 0) { 3437 mAudioWatchdog->requestExit(); 3438 mAudioWatchdog->requestExitAndWait(); 3439 mAudioWatchdog.clear(); 3440 } 3441#endif 3442 } 3443 mAudioFlinger->unregisterWriter(mFastMixerNBLogWriter); 3444 delete mAudioMixer; 3445} 3446 3447 3448uint32_t AudioFlinger::MixerThread::correctLatency_l(uint32_t latency) const 3449{ 3450 if (mFastMixer != 0) { 3451 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 3452 latency += (pipe->getAvgFrames() * 1000) / mSampleRate; 3453 } 3454 return latency; 3455} 3456 3457 3458void AudioFlinger::MixerThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) 3459{ 3460 PlaybackThread::threadLoop_removeTracks(tracksToRemove); 3461} 3462 3463ssize_t AudioFlinger::MixerThread::threadLoop_write() 3464{ 3465 // FIXME we should only do one push per cycle; confirm this is true 3466 // Start the fast mixer if it's not already running 3467 if (mFastMixer != 0) { 3468 FastMixerStateQueue *sq = mFastMixer->sq(); 3469 FastMixerState *state = sq->begin(); 3470 if (state->mCommand != FastMixerState::MIX_WRITE && 3471 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)) { 3472 if (state->mCommand == FastMixerState::COLD_IDLE) { 3473 int32_t old = android_atomic_inc(&mFastMixerFutex); 3474 if (old == -1) { 3475 (void) syscall(__NR_futex, &mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 3476 } 3477#ifdef AUDIO_WATCHDOG 3478 if (mAudioWatchdog != 0) { 3479 mAudioWatchdog->resume(); 3480 } 3481#endif 3482 } 3483 state->mCommand = FastMixerState::MIX_WRITE; 3484#ifdef FAST_THREAD_STATISTICS 3485 mFastMixerDumpState.increaseSamplingN(mAudioFlinger->isLowRamDevice() ? 3486 FastThreadDumpState::kSamplingNforLowRamDevice : FastThreadDumpState::kSamplingN); 3487#endif 3488 sq->end(); 3489 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3490 if (kUseFastMixer == FastMixer_Dynamic) { 3491 mNormalSink = mPipeSink; 3492 } 3493 } else { 3494 sq->end(false /*didModify*/); 3495 } 3496 } 3497 return PlaybackThread::threadLoop_write(); 3498} 3499 3500void AudioFlinger::MixerThread::threadLoop_standby() 3501{ 3502 // Idle the fast mixer if it's currently running 3503 if (mFastMixer != 0) { 3504 FastMixerStateQueue *sq = mFastMixer->sq(); 3505 FastMixerState *state = sq->begin(); 3506 if (!(state->mCommand & FastMixerState::IDLE)) { 3507 state->mCommand = FastMixerState::COLD_IDLE; 3508 state->mColdFutexAddr = &mFastMixerFutex; 3509 state->mColdGen++; 3510 mFastMixerFutex = 0; 3511 sq->end(); 3512 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 3513 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3514 if (kUseFastMixer == FastMixer_Dynamic) { 3515 mNormalSink = mOutputSink; 3516 } 3517#ifdef AUDIO_WATCHDOG 3518 if (mAudioWatchdog != 0) { 3519 mAudioWatchdog->pause(); 3520 } 3521#endif 3522 } else { 3523 sq->end(false /*didModify*/); 3524 } 3525 } 3526 PlaybackThread::threadLoop_standby(); 3527} 3528 3529bool AudioFlinger::PlaybackThread::waitingAsyncCallback_l() 3530{ 3531 return false; 3532} 3533 3534bool AudioFlinger::PlaybackThread::shouldStandby_l() 3535{ 3536 return !mStandby; 3537} 3538 3539bool AudioFlinger::PlaybackThread::waitingAsyncCallback() 3540{ 3541 Mutex::Autolock _l(mLock); 3542 return waitingAsyncCallback_l(); 3543} 3544 3545// shared by MIXER and DIRECT, overridden by DUPLICATING 3546void AudioFlinger::PlaybackThread::threadLoop_standby() 3547{ 3548 ALOGV("Audio hardware entering standby, mixer %p, suspend count %d", this, mSuspended); 3549 mOutput->standby(); 3550 if (mUseAsyncWrite != 0) { 3551 // discard any pending drain or write ack by incrementing sequence 3552 mWriteAckSequence = (mWriteAckSequence + 2) & ~1; 3553 mDrainSequence = (mDrainSequence + 2) & ~1; 3554 ALOG_ASSERT(mCallbackThread != 0); 3555 mCallbackThread->setWriteBlocked(mWriteAckSequence); 3556 mCallbackThread->setDraining(mDrainSequence); 3557 } 3558 mHwPaused = false; 3559} 3560 3561void AudioFlinger::PlaybackThread::onAddNewTrack_l() 3562{ 3563 ALOGV("signal playback thread"); 3564 broadcast_l(); 3565} 3566 3567void AudioFlinger::MixerThread::threadLoop_mix() 3568{ 3569 // obtain the presentation timestamp of the next output buffer 3570 int64_t pts; 3571 status_t status = INVALID_OPERATION; 3572 3573 if (mNormalSink != 0) { 3574 status = mNormalSink->getNextWriteTimestamp(&pts); 3575 } else { 3576 status = mOutputSink->getNextWriteTimestamp(&pts); 3577 } 3578 3579 if (status != NO_ERROR) { 3580 pts = AudioBufferProvider::kInvalidPTS; 3581 } 3582 3583 // mix buffers... 3584 mAudioMixer->process(pts); 3585 mCurrentWriteLength = mSinkBufferSize; 3586 // increase sleep time progressively when application underrun condition clears. 3587 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 3588 // that a steady state of alternating ready/not ready conditions keeps the sleep time 3589 // such that we would underrun the audio HAL. 3590 if ((mSleepTimeUs == 0) && (sleepTimeShift > 0)) { 3591 sleepTimeShift--; 3592 } 3593 mSleepTimeUs = 0; 3594 mStandbyTimeNs = systemTime() + mStandbyDelayNs; 3595 //TODO: delay standby when effects have a tail 3596 3597} 3598 3599void AudioFlinger::MixerThread::threadLoop_sleepTime() 3600{ 3601 // If no tracks are ready, sleep once for the duration of an output 3602 // buffer size, then write 0s to the output 3603 if (mSleepTimeUs == 0) { 3604 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3605 mSleepTimeUs = mActiveSleepTimeUs >> sleepTimeShift; 3606 if (mSleepTimeUs < kMinThreadSleepTimeUs) { 3607 mSleepTimeUs = kMinThreadSleepTimeUs; 3608 } 3609 // reduce sleep time in case of consecutive application underruns to avoid 3610 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 3611 // duration we would end up writing less data than needed by the audio HAL if 3612 // the condition persists. 3613 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 3614 sleepTimeShift++; 3615 } 3616 } else { 3617 mSleepTimeUs = mIdleSleepTimeUs; 3618 } 3619 } else if (mBytesWritten != 0 || (mMixerStatus == MIXER_TRACKS_ENABLED)) { 3620 // clear out mMixerBuffer or mSinkBuffer, to ensure buffers are cleared 3621 // before effects processing or output. 3622 if (mMixerBufferValid) { 3623 memset(mMixerBuffer, 0, mMixerBufferSize); 3624 } else { 3625 memset(mSinkBuffer, 0, mSinkBufferSize); 3626 } 3627 mSleepTimeUs = 0; 3628 ALOGV_IF(mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED), 3629 "anticipated start"); 3630 } 3631 // TODO add standby time extension fct of effect tail 3632} 3633 3634// prepareTracks_l() must be called with ThreadBase::mLock held 3635AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 3636 Vector< sp<Track> > *tracksToRemove) 3637{ 3638 3639 mixer_state mixerStatus = MIXER_IDLE; 3640 // find out which tracks need to be processed 3641 size_t count = mActiveTracks.size(); 3642 size_t mixedTracks = 0; 3643 size_t tracksWithEffect = 0; 3644 // counts only _active_ fast tracks 3645 size_t fastTracks = 0; 3646 uint32_t resetMask = 0; // bit mask of fast tracks that need to be reset 3647 3648 float masterVolume = mMasterVolume; 3649 bool masterMute = mMasterMute; 3650 3651 if (masterMute) { 3652 masterVolume = 0; 3653 } 3654 // Delegate master volume control to effect in output mix effect chain if needed 3655 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 3656 if (chain != 0) { 3657 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 3658 chain->setVolume_l(&v, &v); 3659 masterVolume = (float)((v + (1 << 23)) >> 24); 3660 chain.clear(); 3661 } 3662 3663 // prepare a new state to push 3664 FastMixerStateQueue *sq = NULL; 3665 FastMixerState *state = NULL; 3666 bool didModify = false; 3667 FastMixerStateQueue::block_t block = FastMixerStateQueue::BLOCK_UNTIL_PUSHED; 3668 if (mFastMixer != 0) { 3669 sq = mFastMixer->sq(); 3670 state = sq->begin(); 3671 } 3672 3673 mMixerBufferValid = false; // mMixerBuffer has no valid data until appropriate tracks found. 3674 mEffectBufferValid = false; // mEffectBuffer has no valid data until tracks found. 3675 3676 for (size_t i=0 ; i<count ; i++) { 3677 const sp<Track> t = mActiveTracks[i].promote(); 3678 if (t == 0) { 3679 continue; 3680 } 3681 3682 // this const just means the local variable doesn't change 3683 Track* const track = t.get(); 3684 3685 // process fast tracks 3686 if (track->isFastTrack()) { 3687 3688 // It's theoretically possible (though unlikely) for a fast track to be created 3689 // and then removed within the same normal mix cycle. This is not a problem, as 3690 // the track never becomes active so it's fast mixer slot is never touched. 3691 // The converse, of removing an (active) track and then creating a new track 3692 // at the identical fast mixer slot within the same normal mix cycle, 3693 // is impossible because the slot isn't marked available until the end of each cycle. 3694 int j = track->mFastIndex; 3695 ALOG_ASSERT(0 < j && j < (int)FastMixerState::kMaxFastTracks); 3696 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << j))); 3697 FastTrack *fastTrack = &state->mFastTracks[j]; 3698 3699 // Determine whether the track is currently in underrun condition, 3700 // and whether it had a recent underrun. 3701 FastTrackDump *ftDump = &mFastMixerDumpState.mTracks[j]; 3702 FastTrackUnderruns underruns = ftDump->mUnderruns; 3703 uint32_t recentFull = (underruns.mBitFields.mFull - 3704 track->mObservedUnderruns.mBitFields.mFull) & UNDERRUN_MASK; 3705 uint32_t recentPartial = (underruns.mBitFields.mPartial - 3706 track->mObservedUnderruns.mBitFields.mPartial) & UNDERRUN_MASK; 3707 uint32_t recentEmpty = (underruns.mBitFields.mEmpty - 3708 track->mObservedUnderruns.mBitFields.mEmpty) & UNDERRUN_MASK; 3709 uint32_t recentUnderruns = recentPartial + recentEmpty; 3710 track->mObservedUnderruns = underruns; 3711 // don't count underruns that occur while stopping or pausing 3712 // or stopped which can occur when flush() is called while active 3713 if (!(track->isStopping() || track->isPausing() || track->isStopped()) && 3714 recentUnderruns > 0) { 3715 // FIXME fast mixer will pull & mix partial buffers, but we count as a full underrun 3716 track->mAudioTrackServerProxy->tallyUnderrunFrames(recentUnderruns * mFrameCount); 3717 } 3718 3719 // This is similar to the state machine for normal tracks, 3720 // with a few modifications for fast tracks. 3721 bool isActive = true; 3722 switch (track->mState) { 3723 case TrackBase::STOPPING_1: 3724 // track stays active in STOPPING_1 state until first underrun 3725 if (recentUnderruns > 0 || track->isTerminated()) { 3726 track->mState = TrackBase::STOPPING_2; 3727 } 3728 break; 3729 case TrackBase::PAUSING: 3730 // ramp down is not yet implemented 3731 track->setPaused(); 3732 break; 3733 case TrackBase::RESUMING: 3734 // ramp up is not yet implemented 3735 track->mState = TrackBase::ACTIVE; 3736 break; 3737 case TrackBase::ACTIVE: 3738 if (recentFull > 0 || recentPartial > 0) { 3739 // track has provided at least some frames recently: reset retry count 3740 track->mRetryCount = kMaxTrackRetries; 3741 } 3742 if (recentUnderruns == 0) { 3743 // no recent underruns: stay active 3744 break; 3745 } 3746 // there has recently been an underrun of some kind 3747 if (track->sharedBuffer() == 0) { 3748 // were any of the recent underruns "empty" (no frames available)? 3749 if (recentEmpty == 0) { 3750 // no, then ignore the partial underruns as they are allowed indefinitely 3751 break; 3752 } 3753 // there has recently been an "empty" underrun: decrement the retry counter 3754 if (--(track->mRetryCount) > 0) { 3755 break; 3756 } 3757 // indicate to client process that the track was disabled because of underrun; 3758 // it will then automatically call start() when data is available 3759 android_atomic_or(CBLK_DISABLED, &track->mCblk->mFlags); 3760 // remove from active list, but state remains ACTIVE [confusing but true] 3761 isActive = false; 3762 break; 3763 } 3764 // fall through 3765 case TrackBase::STOPPING_2: 3766 case TrackBase::PAUSED: 3767 case TrackBase::STOPPED: 3768 case TrackBase::FLUSHED: // flush() while active 3769 // Check for presentation complete if track is inactive 3770 // We have consumed all the buffers of this track. 3771 // This would be incomplete if we auto-paused on underrun 3772 { 3773 size_t audioHALFrames = 3774 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 3775 size_t framesWritten = mBytesWritten / mFrameSize; 3776 if (!(mStandby || track->presentationComplete(framesWritten, audioHALFrames))) { 3777 // track stays in active list until presentation is complete 3778 break; 3779 } 3780 } 3781 if (track->isStopping_2()) { 3782 track->mState = TrackBase::STOPPED; 3783 } 3784 if (track->isStopped()) { 3785 // Can't reset directly, as fast mixer is still polling this track 3786 // track->reset(); 3787 // So instead mark this track as needing to be reset after push with ack 3788 resetMask |= 1 << i; 3789 } 3790 isActive = false; 3791 break; 3792 case TrackBase::IDLE: 3793 default: 3794 LOG_ALWAYS_FATAL("unexpected track state %d", track->mState); 3795 } 3796 3797 if (isActive) { 3798 // was it previously inactive? 3799 if (!(state->mTrackMask & (1 << j))) { 3800 ExtendedAudioBufferProvider *eabp = track; 3801 VolumeProvider *vp = track; 3802 fastTrack->mBufferProvider = eabp; 3803 fastTrack->mVolumeProvider = vp; 3804 fastTrack->mChannelMask = track->mChannelMask; 3805 fastTrack->mFormat = track->mFormat; 3806 fastTrack->mGeneration++; 3807 state->mTrackMask |= 1 << j; 3808 didModify = true; 3809 // no acknowledgement required for newly active tracks 3810 } 3811 // cache the combined master volume and stream type volume for fast mixer; this 3812 // lacks any synchronization or barrier so VolumeProvider may read a stale value 3813 track->mCachedVolume = masterVolume * mStreamTypes[track->streamType()].volume; 3814 ++fastTracks; 3815 } else { 3816 // was it previously active? 3817 if (state->mTrackMask & (1 << j)) { 3818 fastTrack->mBufferProvider = NULL; 3819 fastTrack->mGeneration++; 3820 state->mTrackMask &= ~(1 << j); 3821 didModify = true; 3822 // If any fast tracks were removed, we must wait for acknowledgement 3823 // because we're about to decrement the last sp<> on those tracks. 3824 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3825 } else { 3826 LOG_ALWAYS_FATAL("fast track %d should have been active", j); 3827 } 3828 tracksToRemove->add(track); 3829 // Avoids a misleading display in dumpsys 3830 track->mObservedUnderruns.mBitFields.mMostRecent = UNDERRUN_FULL; 3831 } 3832 continue; 3833 } 3834 3835 { // local variable scope to avoid goto warning 3836 3837 audio_track_cblk_t* cblk = track->cblk(); 3838 3839 // The first time a track is added we wait 3840 // for all its buffers to be filled before processing it 3841 int name = track->name(); 3842 // make sure that we have enough frames to mix one full buffer. 3843 // enforce this condition only once to enable draining the buffer in case the client 3844 // app does not call stop() and relies on underrun to stop: 3845 // hence the test on (mMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 3846 // during last round 3847 size_t desiredFrames; 3848 const uint32_t sampleRate = track->mAudioTrackServerProxy->getSampleRate(); 3849 AudioPlaybackRate playbackRate = track->mAudioTrackServerProxy->getPlaybackRate(); 3850 3851 desiredFrames = sourceFramesNeededWithTimestretch( 3852 sampleRate, mNormalFrameCount, mSampleRate, playbackRate.mSpeed); 3853 // TODO: ONLY USED FOR LEGACY RESAMPLERS, remove when they are removed. 3854 // add frames already consumed but not yet released by the resampler 3855 // because mAudioTrackServerProxy->framesReady() will include these frames 3856 desiredFrames += mAudioMixer->getUnreleasedFrames(track->name()); 3857 3858 uint32_t minFrames = 1; 3859 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() && 3860 (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) { 3861 minFrames = desiredFrames; 3862 } 3863 3864 size_t framesReady = track->framesReady(); 3865 if (ATRACE_ENABLED()) { 3866 // I wish we had formatted trace names 3867 char traceName[16]; 3868 strcpy(traceName, "nRdy"); 3869 int name = track->name(); 3870 if (AudioMixer::TRACK0 <= name && 3871 name < (int) (AudioMixer::TRACK0 + AudioMixer::MAX_NUM_TRACKS)) { 3872 name -= AudioMixer::TRACK0; 3873 traceName[4] = (name / 10) + '0'; 3874 traceName[5] = (name % 10) + '0'; 3875 } else { 3876 traceName[4] = '?'; 3877 traceName[5] = '?'; 3878 } 3879 traceName[6] = '\0'; 3880 ATRACE_INT(traceName, framesReady); 3881 } 3882 if ((framesReady >= minFrames) && track->isReady() && 3883 !track->isPaused() && !track->isTerminated()) 3884 { 3885 ALOGVV("track %d s=%08x [OK] on thread %p", name, cblk->mServer, this); 3886 3887 mixedTracks++; 3888 3889 // track->mainBuffer() != mSinkBuffer or mMixerBuffer means 3890 // there is an effect chain connected to the track 3891 chain.clear(); 3892 if (track->mainBuffer() != mSinkBuffer && 3893 track->mainBuffer() != mMixerBuffer) { 3894 if (mEffectBufferEnabled) { 3895 mEffectBufferValid = true; // Later can set directly. 3896 } 3897 chain = getEffectChain_l(track->sessionId()); 3898 // Delegate volume control to effect in track effect chain if needed 3899 if (chain != 0) { 3900 tracksWithEffect++; 3901 } else { 3902 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on " 3903 "session %d", 3904 name, track->sessionId()); 3905 } 3906 } 3907 3908 3909 int param = AudioMixer::VOLUME; 3910 if (track->mFillingUpStatus == Track::FS_FILLED) { 3911 // no ramp for the first volume setting 3912 track->mFillingUpStatus = Track::FS_ACTIVE; 3913 if (track->mState == TrackBase::RESUMING) { 3914 track->mState = TrackBase::ACTIVE; 3915 param = AudioMixer::RAMP_VOLUME; 3916 } 3917 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 3918 // FIXME should not make a decision based on mServer 3919 } else if (cblk->mServer != 0) { 3920 // If the track is stopped before the first frame was mixed, 3921 // do not apply ramp 3922 param = AudioMixer::RAMP_VOLUME; 3923 } 3924 3925 // compute volume for this track 3926 uint32_t vl, vr; // in U8.24 integer format 3927 float vlf, vrf, vaf; // in [0.0, 1.0] float format 3928 if (track->isPausing() || mStreamTypes[track->streamType()].mute) { 3929 vl = vr = 0; 3930 vlf = vrf = vaf = 0.; 3931 if (track->isPausing()) { 3932 track->setPaused(); 3933 } 3934 } else { 3935 3936 // read original volumes with volume control 3937 float typeVolume = mStreamTypes[track->streamType()].volume; 3938 float v = masterVolume * typeVolume; 3939 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 3940 gain_minifloat_packed_t vlr = proxy->getVolumeLR(); 3941 vlf = float_from_gain(gain_minifloat_unpack_left(vlr)); 3942 vrf = float_from_gain(gain_minifloat_unpack_right(vlr)); 3943 // track volumes come from shared memory, so can't be trusted and must be clamped 3944 if (vlf > GAIN_FLOAT_UNITY) { 3945 ALOGV("Track left volume out of range: %.3g", vlf); 3946 vlf = GAIN_FLOAT_UNITY; 3947 } 3948 if (vrf > GAIN_FLOAT_UNITY) { 3949 ALOGV("Track right volume out of range: %.3g", vrf); 3950 vrf = GAIN_FLOAT_UNITY; 3951 } 3952 // now apply the master volume and stream type volume 3953 vlf *= v; 3954 vrf *= v; 3955 // assuming master volume and stream type volume each go up to 1.0, 3956 // then derive vl and vr as U8.24 versions for the effect chain 3957 const float scaleto8_24 = MAX_GAIN_INT * MAX_GAIN_INT; 3958 vl = (uint32_t) (scaleto8_24 * vlf); 3959 vr = (uint32_t) (scaleto8_24 * vrf); 3960 // vl and vr are now in U8.24 format 3961 uint16_t sendLevel = proxy->getSendLevel_U4_12(); 3962 // send level comes from shared memory and so may be corrupt 3963 if (sendLevel > MAX_GAIN_INT) { 3964 ALOGV("Track send level out of range: %04X", sendLevel); 3965 sendLevel = MAX_GAIN_INT; 3966 } 3967 // vaf is represented as [0.0, 1.0] float by rescaling sendLevel 3968 vaf = v * sendLevel * (1. / MAX_GAIN_INT); 3969 } 3970 3971 // Delegate volume control to effect in track effect chain if needed 3972 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 3973 // Do not ramp volume if volume is controlled by effect 3974 param = AudioMixer::VOLUME; 3975 // Update remaining floating point volume levels 3976 vlf = (float)vl / (1 << 24); 3977 vrf = (float)vr / (1 << 24); 3978 track->mHasVolumeController = true; 3979 } else { 3980 // force no volume ramp when volume controller was just disabled or removed 3981 // from effect chain to avoid volume spike 3982 if (track->mHasVolumeController) { 3983 param = AudioMixer::VOLUME; 3984 } 3985 track->mHasVolumeController = false; 3986 } 3987 3988 // XXX: these things DON'T need to be done each time 3989 mAudioMixer->setBufferProvider(name, track); 3990 mAudioMixer->enable(name); 3991 3992 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, &vlf); 3993 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, &vrf); 3994 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, &vaf); 3995 mAudioMixer->setParameter( 3996 name, 3997 AudioMixer::TRACK, 3998 AudioMixer::FORMAT, (void *)track->format()); 3999 mAudioMixer->setParameter( 4000 name, 4001 AudioMixer::TRACK, 4002 AudioMixer::CHANNEL_MASK, (void *)(uintptr_t)track->channelMask()); 4003 mAudioMixer->setParameter( 4004 name, 4005 AudioMixer::TRACK, 4006 AudioMixer::MIXER_CHANNEL_MASK, (void *)(uintptr_t)mChannelMask); 4007 // limit track sample rate to 2 x output sample rate, which changes at re-configuration 4008 uint32_t maxSampleRate = mSampleRate * AUDIO_RESAMPLER_DOWN_RATIO_MAX; 4009 uint32_t reqSampleRate = track->mAudioTrackServerProxy->getSampleRate(); 4010 if (reqSampleRate == 0) { 4011 reqSampleRate = mSampleRate; 4012 } else if (reqSampleRate > maxSampleRate) { 4013 reqSampleRate = maxSampleRate; 4014 } 4015 mAudioMixer->setParameter( 4016 name, 4017 AudioMixer::RESAMPLE, 4018 AudioMixer::SAMPLE_RATE, 4019 (void *)(uintptr_t)reqSampleRate); 4020 4021 AudioPlaybackRate playbackRate = track->mAudioTrackServerProxy->getPlaybackRate(); 4022 mAudioMixer->setParameter( 4023 name, 4024 AudioMixer::TIMESTRETCH, 4025 AudioMixer::PLAYBACK_RATE, 4026 &playbackRate); 4027 4028 /* 4029 * Select the appropriate output buffer for the track. 4030 * 4031 * Tracks with effects go into their own effects chain buffer 4032 * and from there into either mEffectBuffer or mSinkBuffer. 4033 * 4034 * Other tracks can use mMixerBuffer for higher precision 4035 * channel accumulation. If this buffer is enabled 4036 * (mMixerBufferEnabled true), then selected tracks will accumulate 4037 * into it. 4038 * 4039 */ 4040 if (mMixerBufferEnabled 4041 && (track->mainBuffer() == mSinkBuffer 4042 || track->mainBuffer() == mMixerBuffer)) { 4043 mAudioMixer->setParameter( 4044 name, 4045 AudioMixer::TRACK, 4046 AudioMixer::MIXER_FORMAT, (void *)mMixerBufferFormat); 4047 mAudioMixer->setParameter( 4048 name, 4049 AudioMixer::TRACK, 4050 AudioMixer::MAIN_BUFFER, (void *)mMixerBuffer); 4051 // TODO: override track->mainBuffer()? 4052 mMixerBufferValid = true; 4053 } else { 4054 mAudioMixer->setParameter( 4055 name, 4056 AudioMixer::TRACK, 4057 AudioMixer::MIXER_FORMAT, (void *)AUDIO_FORMAT_PCM_16_BIT); 4058 mAudioMixer->setParameter( 4059 name, 4060 AudioMixer::TRACK, 4061 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 4062 } 4063 mAudioMixer->setParameter( 4064 name, 4065 AudioMixer::TRACK, 4066 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 4067 4068 // reset retry count 4069 track->mRetryCount = kMaxTrackRetries; 4070 4071 // If one track is ready, set the mixer ready if: 4072 // - the mixer was not ready during previous round OR 4073 // - no other track is not ready 4074 if (mMixerStatusIgnoringFastTracks != MIXER_TRACKS_READY || 4075 mixerStatus != MIXER_TRACKS_ENABLED) { 4076 mixerStatus = MIXER_TRACKS_READY; 4077 } 4078 } else { 4079 if (framesReady < desiredFrames && !track->isStopped() && !track->isPaused()) { 4080 ALOGV("track(%p) underrun, framesReady(%zu) < framesDesired(%zd)", 4081 track, framesReady, desiredFrames); 4082 track->mAudioTrackServerProxy->tallyUnderrunFrames(desiredFrames); 4083 } 4084 // clear effect chain input buffer if an active track underruns to avoid sending 4085 // previous audio buffer again to effects 4086 chain = getEffectChain_l(track->sessionId()); 4087 if (chain != 0) { 4088 chain->clearInputBuffer(); 4089 } 4090 4091 ALOGVV("track %d s=%08x [NOT READY] on thread %p", name, cblk->mServer, this); 4092 if ((track->sharedBuffer() != 0) || track->isTerminated() || 4093 track->isStopped() || track->isPaused()) { 4094 // We have consumed all the buffers of this track. 4095 // Remove it from the list of active tracks. 4096 // TODO: use actual buffer filling status instead of latency when available from 4097 // audio HAL 4098 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 4099 size_t framesWritten = mBytesWritten / mFrameSize; 4100 if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) { 4101 if (track->isStopped()) { 4102 track->reset(); 4103 } 4104 tracksToRemove->add(track); 4105 } 4106 } else { 4107 // No buffers for this track. Give it a few chances to 4108 // fill a buffer, then remove it from active list. 4109 if (--(track->mRetryCount) <= 0) { 4110 ALOGI("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 4111 tracksToRemove->add(track); 4112 // indicate to client process that the track was disabled because of underrun; 4113 // it will then automatically call start() when data is available 4114 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 4115 // If one track is not ready, mark the mixer also not ready if: 4116 // - the mixer was ready during previous round OR 4117 // - no other track is ready 4118 } else if (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY || 4119 mixerStatus != MIXER_TRACKS_READY) { 4120 mixerStatus = MIXER_TRACKS_ENABLED; 4121 } 4122 } 4123 mAudioMixer->disable(name); 4124 } 4125 4126 } // local variable scope to avoid goto warning 4127track_is_ready: ; 4128 4129 } 4130 4131 // Push the new FastMixer state if necessary 4132 bool pauseAudioWatchdog = false; 4133 if (didModify) { 4134 state->mFastTracksGen++; 4135 // if the fast mixer was active, but now there are no fast tracks, then put it in cold idle 4136 if (kUseFastMixer == FastMixer_Dynamic && 4137 state->mCommand == FastMixerState::MIX_WRITE && state->mTrackMask <= 1) { 4138 state->mCommand = FastMixerState::COLD_IDLE; 4139 state->mColdFutexAddr = &mFastMixerFutex; 4140 state->mColdGen++; 4141 mFastMixerFutex = 0; 4142 if (kUseFastMixer == FastMixer_Dynamic) { 4143 mNormalSink = mOutputSink; 4144 } 4145 // If we go into cold idle, need to wait for acknowledgement 4146 // so that fast mixer stops doing I/O. 4147 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 4148 pauseAudioWatchdog = true; 4149 } 4150 } 4151 if (sq != NULL) { 4152 sq->end(didModify); 4153 sq->push(block); 4154 } 4155#ifdef AUDIO_WATCHDOG 4156 if (pauseAudioWatchdog && mAudioWatchdog != 0) { 4157 mAudioWatchdog->pause(); 4158 } 4159#endif 4160 4161 // Now perform the deferred reset on fast tracks that have stopped 4162 while (resetMask != 0) { 4163 size_t i = __builtin_ctz(resetMask); 4164 ALOG_ASSERT(i < count); 4165 resetMask &= ~(1 << i); 4166 sp<Track> t = mActiveTracks[i].promote(); 4167 if (t == 0) { 4168 continue; 4169 } 4170 Track* track = t.get(); 4171 ALOG_ASSERT(track->isFastTrack() && track->isStopped()); 4172 track->reset(); 4173 } 4174 4175 // remove all the tracks that need to be... 4176 removeTracks_l(*tracksToRemove); 4177 4178 if (getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX) != 0) { 4179 mEffectBufferValid = true; 4180 } 4181 4182 if (mEffectBufferValid) { 4183 // as long as there are effects we should clear the effects buffer, to avoid 4184 // passing a non-clean buffer to the effect chain 4185 memset(mEffectBuffer, 0, mEffectBufferSize); 4186 } 4187 // sink or mix buffer must be cleared if all tracks are connected to an 4188 // effect chain as in this case the mixer will not write to the sink or mix buffer 4189 // and track effects will accumulate into it 4190 if ((mBytesRemaining == 0) && ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 4191 (mixedTracks == 0 && fastTracks > 0))) { 4192 // FIXME as a performance optimization, should remember previous zero status 4193 if (mMixerBufferValid) { 4194 memset(mMixerBuffer, 0, mMixerBufferSize); 4195 // TODO: In testing, mSinkBuffer below need not be cleared because 4196 // the PlaybackThread::threadLoop() copies mMixerBuffer into mSinkBuffer 4197 // after mixing. 4198 // 4199 // To enforce this guarantee: 4200 // ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 4201 // (mixedTracks == 0 && fastTracks > 0)) 4202 // must imply MIXER_TRACKS_READY. 4203 // Later, we may clear buffers regardless, and skip much of this logic. 4204 } 4205 // FIXME as a performance optimization, should remember previous zero status 4206 memset(mSinkBuffer, 0, mNormalFrameCount * mFrameSize); 4207 } 4208 4209 // if any fast tracks, then status is ready 4210 mMixerStatusIgnoringFastTracks = mixerStatus; 4211 if (fastTracks > 0) { 4212 mixerStatus = MIXER_TRACKS_READY; 4213 } 4214 return mixerStatus; 4215} 4216 4217// getTrackName_l() must be called with ThreadBase::mLock held 4218int AudioFlinger::MixerThread::getTrackName_l(audio_channel_mask_t channelMask, 4219 audio_format_t format, int sessionId) 4220{ 4221 return mAudioMixer->getTrackName(channelMask, format, sessionId); 4222} 4223 4224// deleteTrackName_l() must be called with ThreadBase::mLock held 4225void AudioFlinger::MixerThread::deleteTrackName_l(int name) 4226{ 4227 ALOGV("remove track (%d) and delete from mixer", name); 4228 mAudioMixer->deleteTrackName(name); 4229} 4230 4231// checkForNewParameter_l() must be called with ThreadBase::mLock held 4232bool AudioFlinger::MixerThread::checkForNewParameter_l(const String8& keyValuePair, 4233 status_t& status) 4234{ 4235 bool reconfig = false; 4236 4237 status = NO_ERROR; 4238 4239 // if !&IDLE, holds the FastMixer state to restore after new parameters processed 4240 FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE; 4241 if (mFastMixer != 0) { 4242 FastMixerStateQueue *sq = mFastMixer->sq(); 4243 FastMixerState *state = sq->begin(); 4244 if (!(state->mCommand & FastMixerState::IDLE)) { 4245 previousCommand = state->mCommand; 4246 state->mCommand = FastMixerState::HOT_IDLE; 4247 sq->end(); 4248 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 4249 } else { 4250 sq->end(false /*didModify*/); 4251 } 4252 } 4253 4254 AudioParameter param = AudioParameter(keyValuePair); 4255 int value; 4256 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 4257 reconfig = true; 4258 } 4259 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 4260 if (!isValidPcmSinkFormat((audio_format_t) value)) { 4261 status = BAD_VALUE; 4262 } else { 4263 // no need to save value, since it's constant 4264 reconfig = true; 4265 } 4266 } 4267 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 4268 if (!isValidPcmSinkChannelMask((audio_channel_mask_t) value)) { 4269 status = BAD_VALUE; 4270 } else { 4271 // no need to save value, since it's constant 4272 reconfig = true; 4273 } 4274 } 4275 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 4276 // do not accept frame count changes if tracks are open as the track buffer 4277 // size depends on frame count and correct behavior would not be guaranteed 4278 // if frame count is changed after track creation 4279 if (!mTracks.isEmpty()) { 4280 status = INVALID_OPERATION; 4281 } else { 4282 reconfig = true; 4283 } 4284 } 4285 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 4286#ifdef ADD_BATTERY_DATA 4287 // when changing the audio output device, call addBatteryData to notify 4288 // the change 4289 if (mOutDevice != value) { 4290 uint32_t params = 0; 4291 // check whether speaker is on 4292 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 4293 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 4294 } 4295 4296 audio_devices_t deviceWithoutSpeaker 4297 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 4298 // check if any other device (except speaker) is on 4299 if (value & deviceWithoutSpeaker) { 4300 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 4301 } 4302 4303 if (params != 0) { 4304 addBatteryData(params); 4305 } 4306 } 4307#endif 4308 4309 // forward device change to effects that have requested to be 4310 // aware of attached audio device. 4311 if (value != AUDIO_DEVICE_NONE) { 4312 mOutDevice = value; 4313 for (size_t i = 0; i < mEffectChains.size(); i++) { 4314 mEffectChains[i]->setDevice_l(mOutDevice); 4315 } 4316 } 4317 } 4318 4319 if (status == NO_ERROR) { 4320 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 4321 keyValuePair.string()); 4322 if (!mStandby && status == INVALID_OPERATION) { 4323 mOutput->standby(); 4324 mStandby = true; 4325 mBytesWritten = 0; 4326 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 4327 keyValuePair.string()); 4328 } 4329 if (status == NO_ERROR && reconfig) { 4330 readOutputParameters_l(); 4331 delete mAudioMixer; 4332 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 4333 for (size_t i = 0; i < mTracks.size() ; i++) { 4334 int name = getTrackName_l(mTracks[i]->mChannelMask, 4335 mTracks[i]->mFormat, mTracks[i]->mSessionId); 4336 if (name < 0) { 4337 break; 4338 } 4339 mTracks[i]->mName = name; 4340 } 4341 sendIoConfigEvent_l(AUDIO_OUTPUT_CONFIG_CHANGED); 4342 } 4343 } 4344 4345 if (!(previousCommand & FastMixerState::IDLE)) { 4346 ALOG_ASSERT(mFastMixer != 0); 4347 FastMixerStateQueue *sq = mFastMixer->sq(); 4348 FastMixerState *state = sq->begin(); 4349 ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE); 4350 state->mCommand = previousCommand; 4351 sq->end(); 4352 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 4353 } 4354 4355 return reconfig; 4356} 4357 4358 4359void AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 4360{ 4361 const size_t SIZE = 256; 4362 char buffer[SIZE]; 4363 String8 result; 4364 4365 PlaybackThread::dumpInternals(fd, args); 4366 dprintf(fd, " Thread throttle time (msecs): %u\n", mThreadThrottleTimeMs); 4367 dprintf(fd, " AudioMixer tracks: 0x%08x\n", mAudioMixer->trackNames()); 4368 4369 // Make a non-atomic copy of fast mixer dump state so it won't change underneath us 4370 const FastMixerDumpState copy(mFastMixerDumpState); 4371 copy.dump(fd); 4372 4373#ifdef STATE_QUEUE_DUMP 4374 // Similar for state queue 4375 StateQueueObserverDump observerCopy = mStateQueueObserverDump; 4376 observerCopy.dump(fd); 4377 StateQueueMutatorDump mutatorCopy = mStateQueueMutatorDump; 4378 mutatorCopy.dump(fd); 4379#endif 4380 4381#ifdef TEE_SINK 4382 // Write the tee output to a .wav file 4383 dumpTee(fd, mTeeSource, mId); 4384#endif 4385 4386#ifdef AUDIO_WATCHDOG 4387 if (mAudioWatchdog != 0) { 4388 // Make a non-atomic copy of audio watchdog dump so it won't change underneath us 4389 AudioWatchdogDump wdCopy = mAudioWatchdogDump; 4390 wdCopy.dump(fd); 4391 } 4392#endif 4393} 4394 4395uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const 4396{ 4397 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000) / 2; 4398} 4399 4400uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() const 4401{ 4402 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000); 4403} 4404 4405void AudioFlinger::MixerThread::cacheParameters_l() 4406{ 4407 PlaybackThread::cacheParameters_l(); 4408 4409 // FIXME: Relaxed timing because of a certain device that can't meet latency 4410 // Should be reduced to 2x after the vendor fixes the driver issue 4411 // increase threshold again due to low power audio mode. The way this warning 4412 // threshold is calculated and its usefulness should be reconsidered anyway. 4413 maxPeriod = seconds(mNormalFrameCount) / mSampleRate * 15; 4414} 4415 4416// ---------------------------------------------------------------------------- 4417 4418AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 4419 AudioStreamOut* output, audio_io_handle_t id, audio_devices_t device, bool systemReady) 4420 : PlaybackThread(audioFlinger, output, id, device, DIRECT, systemReady) 4421 // mLeftVolFloat, mRightVolFloat 4422{ 4423} 4424 4425AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 4426 AudioStreamOut* output, audio_io_handle_t id, uint32_t device, 4427 ThreadBase::type_t type, bool systemReady) 4428 : PlaybackThread(audioFlinger, output, id, device, type, systemReady) 4429 // mLeftVolFloat, mRightVolFloat 4430{ 4431} 4432 4433AudioFlinger::DirectOutputThread::~DirectOutputThread() 4434{ 4435} 4436 4437void AudioFlinger::DirectOutputThread::processVolume_l(Track *track, bool lastTrack) 4438{ 4439 audio_track_cblk_t* cblk = track->cblk(); 4440 float left, right; 4441 4442 if (mMasterMute || mStreamTypes[track->streamType()].mute) { 4443 left = right = 0; 4444 } else { 4445 float typeVolume = mStreamTypes[track->streamType()].volume; 4446 float v = mMasterVolume * typeVolume; 4447 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 4448 gain_minifloat_packed_t vlr = proxy->getVolumeLR(); 4449 left = float_from_gain(gain_minifloat_unpack_left(vlr)); 4450 if (left > GAIN_FLOAT_UNITY) { 4451 left = GAIN_FLOAT_UNITY; 4452 } 4453 left *= v; 4454 right = float_from_gain(gain_minifloat_unpack_right(vlr)); 4455 if (right > GAIN_FLOAT_UNITY) { 4456 right = GAIN_FLOAT_UNITY; 4457 } 4458 right *= v; 4459 } 4460 4461 if (lastTrack) { 4462 if (left != mLeftVolFloat || right != mRightVolFloat) { 4463 mLeftVolFloat = left; 4464 mRightVolFloat = right; 4465 4466 // Convert volumes from float to 8.24 4467 uint32_t vl = (uint32_t)(left * (1 << 24)); 4468 uint32_t vr = (uint32_t)(right * (1 << 24)); 4469 4470 // Delegate volume control to effect in track effect chain if needed 4471 // only one effect chain can be present on DirectOutputThread, so if 4472 // there is one, the track is connected to it 4473 if (!mEffectChains.isEmpty()) { 4474 mEffectChains[0]->setVolume_l(&vl, &vr); 4475 left = (float)vl / (1 << 24); 4476 right = (float)vr / (1 << 24); 4477 } 4478 if (mOutput->stream->set_volume) { 4479 mOutput->stream->set_volume(mOutput->stream, left, right); 4480 } 4481 } 4482 } 4483} 4484 4485void AudioFlinger::DirectOutputThread::onAddNewTrack_l() 4486{ 4487 sp<Track> previousTrack = mPreviousTrack.promote(); 4488 sp<Track> latestTrack = mLatestActiveTrack.promote(); 4489 4490 if (previousTrack != 0 && latestTrack != 0 && 4491 (previousTrack->sessionId() != latestTrack->sessionId())) { 4492 mFlushPending = true; 4493 } 4494 PlaybackThread::onAddNewTrack_l(); 4495} 4496 4497AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l( 4498 Vector< sp<Track> > *tracksToRemove 4499) 4500{ 4501 size_t count = mActiveTracks.size(); 4502 mixer_state mixerStatus = MIXER_IDLE; 4503 bool doHwPause = false; 4504 bool doHwResume = false; 4505 4506 // find out which tracks need to be processed 4507 for (size_t i = 0; i < count; i++) { 4508 sp<Track> t = mActiveTracks[i].promote(); 4509 // The track died recently 4510 if (t == 0) { 4511 continue; 4512 } 4513 4514 if (t->isInvalid()) { 4515 ALOGW("An invalidated track shouldn't be in active list"); 4516 tracksToRemove->add(t); 4517 continue; 4518 } 4519 4520 Track* const track = t.get(); 4521 audio_track_cblk_t* cblk = track->cblk(); 4522 // Only consider last track started for volume and mixer state control. 4523 // In theory an older track could underrun and restart after the new one starts 4524 // but as we only care about the transition phase between two tracks on a 4525 // direct output, it is not a problem to ignore the underrun case. 4526 sp<Track> l = mLatestActiveTrack.promote(); 4527 bool last = l.get() == track; 4528 4529 if (track->isPausing()) { 4530 track->setPaused(); 4531 if (mHwSupportsPause && last && !mHwPaused) { 4532 doHwPause = true; 4533 mHwPaused = true; 4534 } 4535 tracksToRemove->add(track); 4536 } else if (track->isFlushPending()) { 4537 track->flushAck(); 4538 if (last) { 4539 mFlushPending = true; 4540 } 4541 } else if (track->isResumePending()) { 4542 track->resumeAck(); 4543 if (last && mHwPaused) { 4544 doHwResume = true; 4545 mHwPaused = false; 4546 } 4547 } 4548 4549 // The first time a track is added we wait 4550 // for all its buffers to be filled before processing it. 4551 // Allow draining the buffer in case the client 4552 // app does not call stop() and relies on underrun to stop: 4553 // hence the test on (track->mRetryCount > 1). 4554 // If retryCount<=1 then track is about to underrun and be removed. 4555 uint32_t minFrames; 4556 if ((track->sharedBuffer() == 0) && !track->isStopping_1() && !track->isPausing() 4557 && (track->mRetryCount > 1)) { 4558 minFrames = mNormalFrameCount; 4559 } else { 4560 minFrames = 1; 4561 } 4562 4563 if ((track->framesReady() >= minFrames) && track->isReady() && !track->isPaused() && 4564 !track->isStopping_2() && !track->isStopped()) 4565 { 4566 ALOGVV("track %d s=%08x [OK]", track->name(), cblk->mServer); 4567 4568 if (track->mFillingUpStatus == Track::FS_FILLED) { 4569 track->mFillingUpStatus = Track::FS_ACTIVE; 4570 // make sure processVolume_l() will apply new volume even if 0 4571 mLeftVolFloat = mRightVolFloat = -1.0; 4572 if (!mHwSupportsPause) { 4573 track->resumeAck(); 4574 } 4575 } 4576 4577 // compute volume for this track 4578 processVolume_l(track, last); 4579 if (last) { 4580 sp<Track> previousTrack = mPreviousTrack.promote(); 4581 if (previousTrack != 0) { 4582 if (track != previousTrack.get()) { 4583 // Flush any data still being written from last track 4584 mBytesRemaining = 0; 4585 // flush data already sent if changing audio session as audio 4586 // comes from a different source. Also invalidate previous track to force a 4587 // seek when resuming. 4588 if (previousTrack->sessionId() != track->sessionId()) { 4589 previousTrack->invalidate(); 4590 } 4591 } 4592 } 4593 mPreviousTrack = track; 4594 4595 // reset retry count 4596 track->mRetryCount = kMaxTrackRetriesDirect; 4597 mActiveTrack = t; 4598 mixerStatus = MIXER_TRACKS_READY; 4599 if (mHwPaused) { 4600 doHwResume = true; 4601 mHwPaused = false; 4602 } 4603 } 4604 } else { 4605 // clear effect chain input buffer if the last active track started underruns 4606 // to avoid sending previous audio buffer again to effects 4607 if (!mEffectChains.isEmpty() && last) { 4608 mEffectChains[0]->clearInputBuffer(); 4609 } 4610 if (track->isStopping_1()) { 4611 track->mState = TrackBase::STOPPING_2; 4612 if (last && mHwPaused) { 4613 doHwResume = true; 4614 mHwPaused = false; 4615 } 4616 } 4617 if ((track->sharedBuffer() != 0) || track->isStopped() || 4618 track->isStopping_2() || track->isPaused()) { 4619 // We have consumed all the buffers of this track. 4620 // Remove it from the list of active tracks. 4621 size_t audioHALFrames; 4622 if (audio_is_linear_pcm(mFormat)) { 4623 audioHALFrames = (latency_l() * mSampleRate) / 1000; 4624 } else { 4625 audioHALFrames = 0; 4626 } 4627 4628 size_t framesWritten = mBytesWritten / mFrameSize; 4629 if (mStandby || !last || 4630 track->presentationComplete(framesWritten, audioHALFrames)) { 4631 if (track->isStopping_2()) { 4632 track->mState = TrackBase::STOPPED; 4633 } 4634 if (track->isStopped()) { 4635 track->reset(); 4636 } 4637 tracksToRemove->add(track); 4638 } 4639 } else { 4640 // No buffers for this track. Give it a few chances to 4641 // fill a buffer, then remove it from active list. 4642 // Only consider last track started for mixer state control 4643 if (--(track->mRetryCount) <= 0) { 4644 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 4645 tracksToRemove->add(track); 4646 // indicate to client process that the track was disabled because of underrun; 4647 // it will then automatically call start() when data is available 4648 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 4649 } else if (last) { 4650 mixerStatus = MIXER_TRACKS_ENABLED; 4651 if (mHwSupportsPause && !mHwPaused && !mStandby) { 4652 doHwPause = true; 4653 mHwPaused = true; 4654 } 4655 } 4656 } 4657 } 4658 } 4659 4660 // if an active track did not command a flush, check for pending flush on stopped tracks 4661 if (!mFlushPending) { 4662 for (size_t i = 0; i < mTracks.size(); i++) { 4663 if (mTracks[i]->isFlushPending()) { 4664 mTracks[i]->flushAck(); 4665 mFlushPending = true; 4666 } 4667 } 4668 } 4669 4670 // make sure the pause/flush/resume sequence is executed in the right order. 4671 // If a flush is pending and a track is active but the HW is not paused, force a HW pause 4672 // before flush and then resume HW. This can happen in case of pause/flush/resume 4673 // if resume is received before pause is executed. 4674 if (mHwSupportsPause && !mStandby && 4675 (doHwPause || (mFlushPending && !mHwPaused && (count != 0)))) { 4676 mOutput->stream->pause(mOutput->stream); 4677 } 4678 if (mFlushPending) { 4679 flushHw_l(); 4680 } 4681 if (mHwSupportsPause && !mStandby && doHwResume) { 4682 mOutput->stream->resume(mOutput->stream); 4683 } 4684 // remove all the tracks that need to be... 4685 removeTracks_l(*tracksToRemove); 4686 4687 return mixerStatus; 4688} 4689 4690void AudioFlinger::DirectOutputThread::threadLoop_mix() 4691{ 4692 size_t frameCount = mFrameCount; 4693 int8_t *curBuf = (int8_t *)mSinkBuffer; 4694 // output audio to hardware 4695 while (frameCount) { 4696 AudioBufferProvider::Buffer buffer; 4697 buffer.frameCount = frameCount; 4698 status_t status = mActiveTrack->getNextBuffer(&buffer); 4699 if (status != NO_ERROR || buffer.raw == NULL) { 4700 memset(curBuf, 0, frameCount * mFrameSize); 4701 break; 4702 } 4703 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 4704 frameCount -= buffer.frameCount; 4705 curBuf += buffer.frameCount * mFrameSize; 4706 mActiveTrack->releaseBuffer(&buffer); 4707 } 4708 mCurrentWriteLength = curBuf - (int8_t *)mSinkBuffer; 4709 mSleepTimeUs = 0; 4710 mStandbyTimeNs = systemTime() + mStandbyDelayNs; 4711 mActiveTrack.clear(); 4712} 4713 4714void AudioFlinger::DirectOutputThread::threadLoop_sleepTime() 4715{ 4716 // do not write to HAL when paused 4717 if (mHwPaused || (usesHwAvSync() && mStandby)) { 4718 mSleepTimeUs = mIdleSleepTimeUs; 4719 return; 4720 } 4721 if (mSleepTimeUs == 0) { 4722 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4723 mSleepTimeUs = mActiveSleepTimeUs; 4724 } else { 4725 mSleepTimeUs = mIdleSleepTimeUs; 4726 } 4727 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 4728 memset(mSinkBuffer, 0, mFrameCount * mFrameSize); 4729 mSleepTimeUs = 0; 4730 } 4731} 4732 4733void AudioFlinger::DirectOutputThread::threadLoop_exit() 4734{ 4735 { 4736 Mutex::Autolock _l(mLock); 4737 for (size_t i = 0; i < mTracks.size(); i++) { 4738 if (mTracks[i]->isFlushPending()) { 4739 mTracks[i]->flushAck(); 4740 mFlushPending = true; 4741 } 4742 } 4743 if (mFlushPending) { 4744 flushHw_l(); 4745 } 4746 } 4747 PlaybackThread::threadLoop_exit(); 4748} 4749 4750// must be called with thread mutex locked 4751bool AudioFlinger::DirectOutputThread::shouldStandby_l() 4752{ 4753 bool trackPaused = false; 4754 bool trackStopped = false; 4755 4756 // do not put the HAL in standby when paused. AwesomePlayer clear the offloaded AudioTrack 4757 // after a timeout and we will enter standby then. 4758 if (mTracks.size() > 0) { 4759 trackPaused = mTracks[mTracks.size() - 1]->isPaused(); 4760 trackStopped = mTracks[mTracks.size() - 1]->isStopped() || 4761 mTracks[mTracks.size() - 1]->mState == TrackBase::IDLE; 4762 } 4763 4764 return !mStandby && !(trackPaused || (mHwPaused && !trackStopped)); 4765} 4766 4767// getTrackName_l() must be called with ThreadBase::mLock held 4768int AudioFlinger::DirectOutputThread::getTrackName_l(audio_channel_mask_t channelMask __unused, 4769 audio_format_t format __unused, int sessionId __unused) 4770{ 4771 return 0; 4772} 4773 4774// deleteTrackName_l() must be called with ThreadBase::mLock held 4775void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name __unused) 4776{ 4777} 4778 4779// checkForNewParameter_l() must be called with ThreadBase::mLock held 4780bool AudioFlinger::DirectOutputThread::checkForNewParameter_l(const String8& keyValuePair, 4781 status_t& status) 4782{ 4783 bool reconfig = false; 4784 4785 status = NO_ERROR; 4786 4787 AudioParameter param = AudioParameter(keyValuePair); 4788 int value; 4789 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 4790 // forward device change to effects that have requested to be 4791 // aware of attached audio device. 4792 if (value != AUDIO_DEVICE_NONE) { 4793 mOutDevice = value; 4794 for (size_t i = 0; i < mEffectChains.size(); i++) { 4795 mEffectChains[i]->setDevice_l(mOutDevice); 4796 } 4797 } 4798 } 4799 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 4800 // do not accept frame count changes if tracks are open as the track buffer 4801 // size depends on frame count and correct behavior would not be garantied 4802 // if frame count is changed after track creation 4803 if (!mTracks.isEmpty()) { 4804 status = INVALID_OPERATION; 4805 } else { 4806 reconfig = true; 4807 } 4808 } 4809 if (status == NO_ERROR) { 4810 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 4811 keyValuePair.string()); 4812 if (!mStandby && status == INVALID_OPERATION) { 4813 mOutput->standby(); 4814 mStandby = true; 4815 mBytesWritten = 0; 4816 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 4817 keyValuePair.string()); 4818 } 4819 if (status == NO_ERROR && reconfig) { 4820 readOutputParameters_l(); 4821 sendIoConfigEvent_l(AUDIO_OUTPUT_CONFIG_CHANGED); 4822 } 4823 } 4824 4825 return reconfig; 4826} 4827 4828uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() const 4829{ 4830 uint32_t time; 4831 if (audio_is_linear_pcm(mFormat)) { 4832 time = PlaybackThread::activeSleepTimeUs(); 4833 } else { 4834 time = 10000; 4835 } 4836 return time; 4837} 4838 4839uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() const 4840{ 4841 uint32_t time; 4842 if (audio_is_linear_pcm(mFormat)) { 4843 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 4844 } else { 4845 time = 10000; 4846 } 4847 return time; 4848} 4849 4850uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() const 4851{ 4852 uint32_t time; 4853 if (audio_is_linear_pcm(mFormat)) { 4854 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 4855 } else { 4856 time = 10000; 4857 } 4858 return time; 4859} 4860 4861void AudioFlinger::DirectOutputThread::cacheParameters_l() 4862{ 4863 PlaybackThread::cacheParameters_l(); 4864 4865 // use shorter standby delay as on normal output to release 4866 // hardware resources as soon as possible 4867 // no delay on outputs with HW A/V sync 4868 if (usesHwAvSync()) { 4869 mStandbyDelayNs = 0; 4870 } else if ((mType == OFFLOAD) && !audio_is_linear_pcm(mFormat)) { 4871 mStandbyDelayNs = kOffloadStandbyDelayNs; 4872 } else { 4873 mStandbyDelayNs = microseconds(mActiveSleepTimeUs*2); 4874 } 4875} 4876 4877void AudioFlinger::DirectOutputThread::flushHw_l() 4878{ 4879 mOutput->flush(); 4880 mHwPaused = false; 4881 mFlushPending = false; 4882} 4883 4884// ---------------------------------------------------------------------------- 4885 4886AudioFlinger::AsyncCallbackThread::AsyncCallbackThread( 4887 const wp<AudioFlinger::PlaybackThread>& playbackThread) 4888 : Thread(false /*canCallJava*/), 4889 mPlaybackThread(playbackThread), 4890 mWriteAckSequence(0), 4891 mDrainSequence(0) 4892{ 4893} 4894 4895AudioFlinger::AsyncCallbackThread::~AsyncCallbackThread() 4896{ 4897} 4898 4899void AudioFlinger::AsyncCallbackThread::onFirstRef() 4900{ 4901 run("Offload Cbk", ANDROID_PRIORITY_URGENT_AUDIO); 4902} 4903 4904bool AudioFlinger::AsyncCallbackThread::threadLoop() 4905{ 4906 while (!exitPending()) { 4907 uint32_t writeAckSequence; 4908 uint32_t drainSequence; 4909 4910 { 4911 Mutex::Autolock _l(mLock); 4912 while (!((mWriteAckSequence & 1) || 4913 (mDrainSequence & 1) || 4914 exitPending())) { 4915 mWaitWorkCV.wait(mLock); 4916 } 4917 4918 if (exitPending()) { 4919 break; 4920 } 4921 ALOGV("AsyncCallbackThread mWriteAckSequence %d mDrainSequence %d", 4922 mWriteAckSequence, mDrainSequence); 4923 writeAckSequence = mWriteAckSequence; 4924 mWriteAckSequence &= ~1; 4925 drainSequence = mDrainSequence; 4926 mDrainSequence &= ~1; 4927 } 4928 { 4929 sp<AudioFlinger::PlaybackThread> playbackThread = mPlaybackThread.promote(); 4930 if (playbackThread != 0) { 4931 if (writeAckSequence & 1) { 4932 playbackThread->resetWriteBlocked(writeAckSequence >> 1); 4933 } 4934 if (drainSequence & 1) { 4935 playbackThread->resetDraining(drainSequence >> 1); 4936 } 4937 } 4938 } 4939 } 4940 return false; 4941} 4942 4943void AudioFlinger::AsyncCallbackThread::exit() 4944{ 4945 ALOGV("AsyncCallbackThread::exit"); 4946 Mutex::Autolock _l(mLock); 4947 requestExit(); 4948 mWaitWorkCV.broadcast(); 4949} 4950 4951void AudioFlinger::AsyncCallbackThread::setWriteBlocked(uint32_t sequence) 4952{ 4953 Mutex::Autolock _l(mLock); 4954 // bit 0 is cleared 4955 mWriteAckSequence = sequence << 1; 4956} 4957 4958void AudioFlinger::AsyncCallbackThread::resetWriteBlocked() 4959{ 4960 Mutex::Autolock _l(mLock); 4961 // ignore unexpected callbacks 4962 if (mWriteAckSequence & 2) { 4963 mWriteAckSequence |= 1; 4964 mWaitWorkCV.signal(); 4965 } 4966} 4967 4968void AudioFlinger::AsyncCallbackThread::setDraining(uint32_t sequence) 4969{ 4970 Mutex::Autolock _l(mLock); 4971 // bit 0 is cleared 4972 mDrainSequence = sequence << 1; 4973} 4974 4975void AudioFlinger::AsyncCallbackThread::resetDraining() 4976{ 4977 Mutex::Autolock _l(mLock); 4978 // ignore unexpected callbacks 4979 if (mDrainSequence & 2) { 4980 mDrainSequence |= 1; 4981 mWaitWorkCV.signal(); 4982 } 4983} 4984 4985 4986// ---------------------------------------------------------------------------- 4987AudioFlinger::OffloadThread::OffloadThread(const sp<AudioFlinger>& audioFlinger, 4988 AudioStreamOut* output, audio_io_handle_t id, uint32_t device, bool systemReady) 4989 : DirectOutputThread(audioFlinger, output, id, device, OFFLOAD, systemReady), 4990 mPausedBytesRemaining(0) 4991{ 4992 //FIXME: mStandby should be set to true by ThreadBase constructor 4993 mStandby = true; 4994} 4995 4996void AudioFlinger::OffloadThread::threadLoop_exit() 4997{ 4998 if (mFlushPending || mHwPaused) { 4999 // If a flush is pending or track was paused, just discard buffered data 5000 flushHw_l(); 5001 } else { 5002 mMixerStatus = MIXER_DRAIN_ALL; 5003 threadLoop_drain(); 5004 } 5005 if (mUseAsyncWrite) { 5006 ALOG_ASSERT(mCallbackThread != 0); 5007 mCallbackThread->exit(); 5008 } 5009 PlaybackThread::threadLoop_exit(); 5010} 5011 5012AudioFlinger::PlaybackThread::mixer_state AudioFlinger::OffloadThread::prepareTracks_l( 5013 Vector< sp<Track> > *tracksToRemove 5014) 5015{ 5016 size_t count = mActiveTracks.size(); 5017 5018 mixer_state mixerStatus = MIXER_IDLE; 5019 bool doHwPause = false; 5020 bool doHwResume = false; 5021 5022 ALOGV("OffloadThread::prepareTracks_l active tracks %d", count); 5023 5024 // find out which tracks need to be processed 5025 for (size_t i = 0; i < count; i++) { 5026 sp<Track> t = mActiveTracks[i].promote(); 5027 // The track died recently 5028 if (t == 0) { 5029 continue; 5030 } 5031 Track* const track = t.get(); 5032 audio_track_cblk_t* cblk = track->cblk(); 5033 // Only consider last track started for volume and mixer state control. 5034 // In theory an older track could underrun and restart after the new one starts 5035 // but as we only care about the transition phase between two tracks on a 5036 // direct output, it is not a problem to ignore the underrun case. 5037 sp<Track> l = mLatestActiveTrack.promote(); 5038 bool last = l.get() == track; 5039 5040 if (track->isInvalid()) { 5041 ALOGW("An invalidated track shouldn't be in active list"); 5042 tracksToRemove->add(track); 5043 continue; 5044 } 5045 5046 if (track->mState == TrackBase::IDLE) { 5047 ALOGW("An idle track shouldn't be in active list"); 5048 continue; 5049 } 5050 5051 if (track->isPausing()) { 5052 track->setPaused(); 5053 if (last) { 5054 if (mHwSupportsPause && !mHwPaused) { 5055 doHwPause = true; 5056 mHwPaused = true; 5057 } 5058 // If we were part way through writing the mixbuffer to 5059 // the HAL we must save this until we resume 5060 // BUG - this will be wrong if a different track is made active, 5061 // in that case we want to discard the pending data in the 5062 // mixbuffer and tell the client to present it again when the 5063 // track is resumed 5064 mPausedWriteLength = mCurrentWriteLength; 5065 mPausedBytesRemaining = mBytesRemaining; 5066 mBytesRemaining = 0; // stop writing 5067 } 5068 tracksToRemove->add(track); 5069 } else if (track->isFlushPending()) { 5070 track->flushAck(); 5071 if (last) { 5072 mFlushPending = true; 5073 } 5074 } else if (track->isResumePending()){ 5075 track->resumeAck(); 5076 if (last) { 5077 if (mPausedBytesRemaining) { 5078 // Need to continue write that was interrupted 5079 mCurrentWriteLength = mPausedWriteLength; 5080 mBytesRemaining = mPausedBytesRemaining; 5081 mPausedBytesRemaining = 0; 5082 } 5083 if (mHwPaused) { 5084 doHwResume = true; 5085 mHwPaused = false; 5086 // threadLoop_mix() will handle the case that we need to 5087 // resume an interrupted write 5088 } 5089 // enable write to audio HAL 5090 mSleepTimeUs = 0; 5091 5092 // Do not handle new data in this iteration even if track->framesReady() 5093 mixerStatus = MIXER_TRACKS_ENABLED; 5094 } 5095 } else if (track->framesReady() && track->isReady() && 5096 !track->isPaused() && !track->isTerminated() && !track->isStopping_2()) { 5097 ALOGVV("OffloadThread: track %d s=%08x [OK]", track->name(), cblk->mServer); 5098 if (track->mFillingUpStatus == Track::FS_FILLED) { 5099 track->mFillingUpStatus = Track::FS_ACTIVE; 5100 // make sure processVolume_l() will apply new volume even if 0 5101 mLeftVolFloat = mRightVolFloat = -1.0; 5102 } 5103 5104 if (last) { 5105 sp<Track> previousTrack = mPreviousTrack.promote(); 5106 if (previousTrack != 0) { 5107 if (track != previousTrack.get()) { 5108 // Flush any data still being written from last track 5109 mBytesRemaining = 0; 5110 if (mPausedBytesRemaining) { 5111 // Last track was paused so we also need to flush saved 5112 // mixbuffer state and invalidate track so that it will 5113 // re-submit that unwritten data when it is next resumed 5114 mPausedBytesRemaining = 0; 5115 // Invalidate is a bit drastic - would be more efficient 5116 // to have a flag to tell client that some of the 5117 // previously written data was lost 5118 previousTrack->invalidate(); 5119 } 5120 // flush data already sent to the DSP if changing audio session as audio 5121 // comes from a different source. Also invalidate previous track to force a 5122 // seek when resuming. 5123 if (previousTrack->sessionId() != track->sessionId()) { 5124 previousTrack->invalidate(); 5125 } 5126 } 5127 } 5128 mPreviousTrack = track; 5129 // reset retry count 5130 track->mRetryCount = kMaxTrackRetriesOffload; 5131 mActiveTrack = t; 5132 mixerStatus = MIXER_TRACKS_READY; 5133 } 5134 } else { 5135 ALOGVV("OffloadThread: track %d s=%08x [NOT READY]", track->name(), cblk->mServer); 5136 if (track->isStopping_1()) { 5137 // Hardware buffer can hold a large amount of audio so we must 5138 // wait for all current track's data to drain before we say 5139 // that the track is stopped. 5140 if (mBytesRemaining == 0) { 5141 // Only start draining when all data in mixbuffer 5142 // has been written 5143 ALOGV("OffloadThread: underrun and STOPPING_1 -> draining, STOPPING_2"); 5144 track->mState = TrackBase::STOPPING_2; // so presentation completes after drain 5145 // do not drain if no data was ever sent to HAL (mStandby == true) 5146 if (last && !mStandby) { 5147 // do not modify drain sequence if we are already draining. This happens 5148 // when resuming from pause after drain. 5149 if ((mDrainSequence & 1) == 0) { 5150 mSleepTimeUs = 0; 5151 mStandbyTimeNs = systemTime() + mStandbyDelayNs; 5152 mixerStatus = MIXER_DRAIN_TRACK; 5153 mDrainSequence += 2; 5154 } 5155 if (mHwPaused) { 5156 // It is possible to move from PAUSED to STOPPING_1 without 5157 // a resume so we must ensure hardware is running 5158 doHwResume = true; 5159 mHwPaused = false; 5160 } 5161 } 5162 } 5163 } else if (track->isStopping_2()) { 5164 // Drain has completed or we are in standby, signal presentation complete 5165 if (!(mDrainSequence & 1) || !last || mStandby) { 5166 track->mState = TrackBase::STOPPED; 5167 size_t audioHALFrames = 5168 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 5169 size_t framesWritten = 5170 mBytesWritten / mOutput->getFrameSize(); 5171 track->presentationComplete(framesWritten, audioHALFrames); 5172 track->reset(); 5173 tracksToRemove->add(track); 5174 } 5175 } else { 5176 // No buffers for this track. Give it a few chances to 5177 // fill a buffer, then remove it from active list. 5178 if (--(track->mRetryCount) <= 0) { 5179 ALOGV("OffloadThread: BUFFER TIMEOUT: remove(%d) from active list", 5180 track->name()); 5181 tracksToRemove->add(track); 5182 // indicate to client process that the track was disabled because of underrun; 5183 // it will then automatically call start() when data is available 5184 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 5185 } else if (last){ 5186 mixerStatus = MIXER_TRACKS_ENABLED; 5187 } 5188 } 5189 } 5190 // compute volume for this track 5191 processVolume_l(track, last); 5192 } 5193 5194 // make sure the pause/flush/resume sequence is executed in the right order. 5195 // If a flush is pending and a track is active but the HW is not paused, force a HW pause 5196 // before flush and then resume HW. This can happen in case of pause/flush/resume 5197 // if resume is received before pause is executed. 5198 if (!mStandby && (doHwPause || (mFlushPending && !mHwPaused && (count != 0)))) { 5199 mOutput->stream->pause(mOutput->stream); 5200 } 5201 if (mFlushPending) { 5202 flushHw_l(); 5203 } 5204 if (!mStandby && doHwResume) { 5205 mOutput->stream->resume(mOutput->stream); 5206 } 5207 5208 // remove all the tracks that need to be... 5209 removeTracks_l(*tracksToRemove); 5210 5211 return mixerStatus; 5212} 5213 5214// must be called with thread mutex locked 5215bool AudioFlinger::OffloadThread::waitingAsyncCallback_l() 5216{ 5217 ALOGVV("waitingAsyncCallback_l mWriteAckSequence %d mDrainSequence %d", 5218 mWriteAckSequence, mDrainSequence); 5219 if (mUseAsyncWrite && ((mWriteAckSequence & 1) || (mDrainSequence & 1))) { 5220 return true; 5221 } 5222 return false; 5223} 5224 5225bool AudioFlinger::OffloadThread::waitingAsyncCallback() 5226{ 5227 Mutex::Autolock _l(mLock); 5228 return waitingAsyncCallback_l(); 5229} 5230 5231void AudioFlinger::OffloadThread::flushHw_l() 5232{ 5233 DirectOutputThread::flushHw_l(); 5234 // Flush anything still waiting in the mixbuffer 5235 mCurrentWriteLength = 0; 5236 mBytesRemaining = 0; 5237 mPausedWriteLength = 0; 5238 mPausedBytesRemaining = 0; 5239 5240 if (mUseAsyncWrite) { 5241 // discard any pending drain or write ack by incrementing sequence 5242 mWriteAckSequence = (mWriteAckSequence + 2) & ~1; 5243 mDrainSequence = (mDrainSequence + 2) & ~1; 5244 ALOG_ASSERT(mCallbackThread != 0); 5245 mCallbackThread->setWriteBlocked(mWriteAckSequence); 5246 mCallbackThread->setDraining(mDrainSequence); 5247 } 5248} 5249 5250// ---------------------------------------------------------------------------- 5251 5252AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 5253 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id, bool systemReady) 5254 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->outDevice(), 5255 systemReady, DUPLICATING), 5256 mWaitTimeMs(UINT_MAX) 5257{ 5258 addOutputTrack(mainThread); 5259} 5260 5261AudioFlinger::DuplicatingThread::~DuplicatingThread() 5262{ 5263 for (size_t i = 0; i < mOutputTracks.size(); i++) { 5264 mOutputTracks[i]->destroy(); 5265 } 5266} 5267 5268void AudioFlinger::DuplicatingThread::threadLoop_mix() 5269{ 5270 // mix buffers... 5271 if (outputsReady(outputTracks)) { 5272 mAudioMixer->process(AudioBufferProvider::kInvalidPTS); 5273 } else { 5274 if (mMixerBufferValid) { 5275 memset(mMixerBuffer, 0, mMixerBufferSize); 5276 } else { 5277 memset(mSinkBuffer, 0, mSinkBufferSize); 5278 } 5279 } 5280 mSleepTimeUs = 0; 5281 writeFrames = mNormalFrameCount; 5282 mCurrentWriteLength = mSinkBufferSize; 5283 mStandbyTimeNs = systemTime() + mStandbyDelayNs; 5284} 5285 5286void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() 5287{ 5288 if (mSleepTimeUs == 0) { 5289 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 5290 mSleepTimeUs = mActiveSleepTimeUs; 5291 } else { 5292 mSleepTimeUs = mIdleSleepTimeUs; 5293 } 5294 } else if (mBytesWritten != 0) { 5295 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 5296 writeFrames = mNormalFrameCount; 5297 memset(mSinkBuffer, 0, mSinkBufferSize); 5298 } else { 5299 // flush remaining overflow buffers in output tracks 5300 writeFrames = 0; 5301 } 5302 mSleepTimeUs = 0; 5303 } 5304} 5305 5306ssize_t AudioFlinger::DuplicatingThread::threadLoop_write() 5307{ 5308 for (size_t i = 0; i < outputTracks.size(); i++) { 5309 outputTracks[i]->write(mSinkBuffer, writeFrames); 5310 } 5311 mStandby = false; 5312 return (ssize_t)mSinkBufferSize; 5313} 5314 5315void AudioFlinger::DuplicatingThread::threadLoop_standby() 5316{ 5317 // DuplicatingThread implements standby by stopping all tracks 5318 for (size_t i = 0; i < outputTracks.size(); i++) { 5319 outputTracks[i]->stop(); 5320 } 5321} 5322 5323void AudioFlinger::DuplicatingThread::saveOutputTracks() 5324{ 5325 outputTracks = mOutputTracks; 5326} 5327 5328void AudioFlinger::DuplicatingThread::clearOutputTracks() 5329{ 5330 outputTracks.clear(); 5331} 5332 5333void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 5334{ 5335 Mutex::Autolock _l(mLock); 5336 // The downstream MixerThread consumes thread->frameCount() amount of frames per mix pass. 5337 // Adjust for thread->sampleRate() to determine minimum buffer frame count. 5338 // Then triple buffer because Threads do not run synchronously and may not be clock locked. 5339 const size_t frameCount = 5340 3 * sourceFramesNeeded(mSampleRate, thread->frameCount(), thread->sampleRate()); 5341 // TODO: Consider asynchronous sample rate conversion to handle clock disparity 5342 // from different OutputTracks and their associated MixerThreads (e.g. one may 5343 // nearly empty and the other may be dropping data). 5344 5345 sp<OutputTrack> outputTrack = new OutputTrack(thread, 5346 this, 5347 mSampleRate, 5348 mFormat, 5349 mChannelMask, 5350 frameCount, 5351 IPCThreadState::self()->getCallingUid()); 5352 if (outputTrack->cblk() != NULL) { 5353 thread->setStreamVolume(AUDIO_STREAM_PATCH, 1.0f); 5354 mOutputTracks.add(outputTrack); 5355 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack.get(), thread); 5356 updateWaitTime_l(); 5357 } 5358} 5359 5360void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 5361{ 5362 Mutex::Autolock _l(mLock); 5363 for (size_t i = 0; i < mOutputTracks.size(); i++) { 5364 if (mOutputTracks[i]->thread() == thread) { 5365 mOutputTracks[i]->destroy(); 5366 mOutputTracks.removeAt(i); 5367 updateWaitTime_l(); 5368 if (thread->getOutput() == mOutput) { 5369 mOutput = NULL; 5370 } 5371 return; 5372 } 5373 } 5374 ALOGV("removeOutputTrack(): unknown thread: %p", thread); 5375} 5376 5377// caller must hold mLock 5378void AudioFlinger::DuplicatingThread::updateWaitTime_l() 5379{ 5380 mWaitTimeMs = UINT_MAX; 5381 for (size_t i = 0; i < mOutputTracks.size(); i++) { 5382 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 5383 if (strong != 0) { 5384 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 5385 if (waitTimeMs < mWaitTimeMs) { 5386 mWaitTimeMs = waitTimeMs; 5387 } 5388 } 5389 } 5390} 5391 5392 5393bool AudioFlinger::DuplicatingThread::outputsReady( 5394 const SortedVector< sp<OutputTrack> > &outputTracks) 5395{ 5396 for (size_t i = 0; i < outputTracks.size(); i++) { 5397 sp<ThreadBase> thread = outputTracks[i]->thread().promote(); 5398 if (thread == 0) { 5399 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", 5400 outputTracks[i].get()); 5401 return false; 5402 } 5403 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 5404 // see note at standby() declaration 5405 if (playbackThread->standby() && !playbackThread->isSuspended()) { 5406 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), 5407 thread.get()); 5408 return false; 5409 } 5410 } 5411 return true; 5412} 5413 5414uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() const 5415{ 5416 return (mWaitTimeMs * 1000) / 2; 5417} 5418 5419void AudioFlinger::DuplicatingThread::cacheParameters_l() 5420{ 5421 // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first 5422 updateWaitTime_l(); 5423 5424 MixerThread::cacheParameters_l(); 5425} 5426 5427// ---------------------------------------------------------------------------- 5428// Record 5429// ---------------------------------------------------------------------------- 5430 5431AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 5432 AudioStreamIn *input, 5433 audio_io_handle_t id, 5434 audio_devices_t outDevice, 5435 audio_devices_t inDevice, 5436 bool systemReady 5437#ifdef TEE_SINK 5438 , const sp<NBAIO_Sink>& teeSink 5439#endif 5440 ) : 5441 ThreadBase(audioFlinger, id, outDevice, inDevice, RECORD, systemReady), 5442 mInput(input), mActiveTracksGen(0), mRsmpInBuffer(NULL), 5443 // mRsmpInFrames and mRsmpInFramesP2 are set by readInputParameters_l() 5444 mRsmpInRear(0) 5445#ifdef TEE_SINK 5446 , mTeeSink(teeSink) 5447#endif 5448 , mReadOnlyHeap(new MemoryDealer(kRecordThreadReadOnlyHeapSize, 5449 "RecordThreadRO", MemoryHeapBase::READ_ONLY)) 5450 // mFastCapture below 5451 , mFastCaptureFutex(0) 5452 // mInputSource 5453 // mPipeSink 5454 // mPipeSource 5455 , mPipeFramesP2(0) 5456 // mPipeMemory 5457 // mFastCaptureNBLogWriter 5458 , mFastTrackAvail(false) 5459{ 5460 snprintf(mThreadName, kThreadNameLength, "AudioIn_%X", id); 5461 mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mThreadName); 5462 5463 readInputParameters_l(); 5464 5465 // create an NBAIO source for the HAL input stream, and negotiate 5466 mInputSource = new AudioStreamInSource(input->stream); 5467 size_t numCounterOffers = 0; 5468 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount, mFormat)}; 5469 ssize_t index = mInputSource->negotiate(offers, 1, NULL, numCounterOffers); 5470 ALOG_ASSERT(index == 0); 5471 5472 // initialize fast capture depending on configuration 5473 bool initFastCapture; 5474 switch (kUseFastCapture) { 5475 case FastCapture_Never: 5476 initFastCapture = false; 5477 break; 5478 case FastCapture_Always: 5479 initFastCapture = true; 5480 break; 5481 case FastCapture_Static: 5482 uint32_t primaryOutputSampleRate; 5483 { 5484 AutoMutex _l(audioFlinger->mHardwareLock); 5485 primaryOutputSampleRate = audioFlinger->mPrimaryOutputSampleRate; 5486 } 5487 initFastCapture = 5488 // either capture sample rate is same as (a reasonable) primary output sample rate 5489 ((isMusicRate(primaryOutputSampleRate) && 5490 (mSampleRate == primaryOutputSampleRate)) || 5491 // or primary output sample rate is unknown, and capture sample rate is reasonable 5492 ((primaryOutputSampleRate == 0) && 5493 isMusicRate(mSampleRate))) && 5494 // and the buffer size is < 12 ms 5495 (mFrameCount * 1000) / mSampleRate < 12; 5496 break; 5497 // case FastCapture_Dynamic: 5498 } 5499 5500 if (initFastCapture) { 5501 // create a Pipe for FastCapture to write to, and for us and fast tracks to read from 5502 NBAIO_Format format = mInputSource->format(); 5503 size_t pipeFramesP2 = roundup(mSampleRate / 25); // double-buffering of 20 ms each 5504 size_t pipeSize = pipeFramesP2 * Format_frameSize(format); 5505 void *pipeBuffer; 5506 const sp<MemoryDealer> roHeap(readOnlyHeap()); 5507 sp<IMemory> pipeMemory; 5508 if ((roHeap == 0) || 5509 (pipeMemory = roHeap->allocate(pipeSize)) == 0 || 5510 (pipeBuffer = pipeMemory->pointer()) == NULL) { 5511 ALOGE("not enough memory for pipe buffer size=%zu", pipeSize); 5512 goto failed; 5513 } 5514 // pipe will be shared directly with fast clients, so clear to avoid leaking old information 5515 memset(pipeBuffer, 0, pipeSize); 5516 Pipe *pipe = new Pipe(pipeFramesP2, format, pipeBuffer); 5517 const NBAIO_Format offers[1] = {format}; 5518 size_t numCounterOffers = 0; 5519 ssize_t index = pipe->negotiate(offers, 1, NULL, numCounterOffers); 5520 ALOG_ASSERT(index == 0); 5521 mPipeSink = pipe; 5522 PipeReader *pipeReader = new PipeReader(*pipe); 5523 numCounterOffers = 0; 5524 index = pipeReader->negotiate(offers, 1, NULL, numCounterOffers); 5525 ALOG_ASSERT(index == 0); 5526 mPipeSource = pipeReader; 5527 mPipeFramesP2 = pipeFramesP2; 5528 mPipeMemory = pipeMemory; 5529 5530 // create fast capture 5531 mFastCapture = new FastCapture(); 5532 FastCaptureStateQueue *sq = mFastCapture->sq(); 5533#ifdef STATE_QUEUE_DUMP 5534 // FIXME 5535#endif 5536 FastCaptureState *state = sq->begin(); 5537 state->mCblk = NULL; 5538 state->mInputSource = mInputSource.get(); 5539 state->mInputSourceGen++; 5540 state->mPipeSink = pipe; 5541 state->mPipeSinkGen++; 5542 state->mFrameCount = mFrameCount; 5543 state->mCommand = FastCaptureState::COLD_IDLE; 5544 // already done in constructor initialization list 5545 //mFastCaptureFutex = 0; 5546 state->mColdFutexAddr = &mFastCaptureFutex; 5547 state->mColdGen++; 5548 state->mDumpState = &mFastCaptureDumpState; 5549#ifdef TEE_SINK 5550 // FIXME 5551#endif 5552 mFastCaptureNBLogWriter = audioFlinger->newWriter_l(kFastCaptureLogSize, "FastCapture"); 5553 state->mNBLogWriter = mFastCaptureNBLogWriter.get(); 5554 sq->end(); 5555 sq->push(FastCaptureStateQueue::BLOCK_UNTIL_PUSHED); 5556 5557 // start the fast capture 5558 mFastCapture->run("FastCapture", ANDROID_PRIORITY_URGENT_AUDIO); 5559 pid_t tid = mFastCapture->getTid(); 5560 sendPrioConfigEvent(getpid_cached, tid, kPriorityFastMixer); 5561#ifdef AUDIO_WATCHDOG 5562 // FIXME 5563#endif 5564 5565 mFastTrackAvail = true; 5566 } 5567failed: ; 5568 5569 // FIXME mNormalSource 5570} 5571 5572AudioFlinger::RecordThread::~RecordThread() 5573{ 5574 if (mFastCapture != 0) { 5575 FastCaptureStateQueue *sq = mFastCapture->sq(); 5576 FastCaptureState *state = sq->begin(); 5577 if (state->mCommand == FastCaptureState::COLD_IDLE) { 5578 int32_t old = android_atomic_inc(&mFastCaptureFutex); 5579 if (old == -1) { 5580 (void) syscall(__NR_futex, &mFastCaptureFutex, FUTEX_WAKE_PRIVATE, 1); 5581 } 5582 } 5583 state->mCommand = FastCaptureState::EXIT; 5584 sq->end(); 5585 sq->push(FastCaptureStateQueue::BLOCK_UNTIL_PUSHED); 5586 mFastCapture->join(); 5587 mFastCapture.clear(); 5588 } 5589 mAudioFlinger->unregisterWriter(mFastCaptureNBLogWriter); 5590 mAudioFlinger->unregisterWriter(mNBLogWriter); 5591 free(mRsmpInBuffer); 5592} 5593 5594void AudioFlinger::RecordThread::onFirstRef() 5595{ 5596 run(mThreadName, PRIORITY_URGENT_AUDIO); 5597} 5598 5599bool AudioFlinger::RecordThread::threadLoop() 5600{ 5601 nsecs_t lastWarning = 0; 5602 5603 inputStandBy(); 5604 5605reacquire_wakelock: 5606 sp<RecordTrack> activeTrack; 5607 int activeTracksGen; 5608 { 5609 Mutex::Autolock _l(mLock); 5610 size_t size = mActiveTracks.size(); 5611 activeTracksGen = mActiveTracksGen; 5612 if (size > 0) { 5613 // FIXME an arbitrary choice 5614 activeTrack = mActiveTracks[0]; 5615 acquireWakeLock_l(activeTrack->uid()); 5616 if (size > 1) { 5617 SortedVector<int> tmp; 5618 for (size_t i = 0; i < size; i++) { 5619 tmp.add(mActiveTracks[i]->uid()); 5620 } 5621 updateWakeLockUids_l(tmp); 5622 } 5623 } else { 5624 acquireWakeLock_l(-1); 5625 } 5626 } 5627 5628 // used to request a deferred sleep, to be executed later while mutex is unlocked 5629 uint32_t sleepUs = 0; 5630 5631 // loop while there is work to do 5632 for (;;) { 5633 Vector< sp<EffectChain> > effectChains; 5634 5635 // sleep with mutex unlocked 5636 if (sleepUs > 0) { 5637 ATRACE_BEGIN("sleep"); 5638 usleep(sleepUs); 5639 ATRACE_END(); 5640 sleepUs = 0; 5641 } 5642 5643 // activeTracks accumulates a copy of a subset of mActiveTracks 5644 Vector< sp<RecordTrack> > activeTracks; 5645 5646 // reference to the (first and only) active fast track 5647 sp<RecordTrack> fastTrack; 5648 5649 // reference to a fast track which is about to be removed 5650 sp<RecordTrack> fastTrackToRemove; 5651 5652 { // scope for mLock 5653 Mutex::Autolock _l(mLock); 5654 5655 processConfigEvents_l(); 5656 5657 // check exitPending here because checkForNewParameters_l() and 5658 // checkForNewParameters_l() can temporarily release mLock 5659 if (exitPending()) { 5660 break; 5661 } 5662 5663 // if no active track(s), then standby and release wakelock 5664 size_t size = mActiveTracks.size(); 5665 if (size == 0) { 5666 standbyIfNotAlreadyInStandby(); 5667 // exitPending() can't become true here 5668 releaseWakeLock_l(); 5669 ALOGV("RecordThread: loop stopping"); 5670 // go to sleep 5671 mWaitWorkCV.wait(mLock); 5672 ALOGV("RecordThread: loop starting"); 5673 goto reacquire_wakelock; 5674 } 5675 5676 if (mActiveTracksGen != activeTracksGen) { 5677 activeTracksGen = mActiveTracksGen; 5678 SortedVector<int> tmp; 5679 for (size_t i = 0; i < size; i++) { 5680 tmp.add(mActiveTracks[i]->uid()); 5681 } 5682 updateWakeLockUids_l(tmp); 5683 } 5684 5685 bool doBroadcast = false; 5686 for (size_t i = 0; i < size; ) { 5687 5688 activeTrack = mActiveTracks[i]; 5689 if (activeTrack->isTerminated()) { 5690 if (activeTrack->isFastTrack()) { 5691 ALOG_ASSERT(fastTrackToRemove == 0); 5692 fastTrackToRemove = activeTrack; 5693 } 5694 removeTrack_l(activeTrack); 5695 mActiveTracks.remove(activeTrack); 5696 mActiveTracksGen++; 5697 size--; 5698 continue; 5699 } 5700 5701 TrackBase::track_state activeTrackState = activeTrack->mState; 5702 switch (activeTrackState) { 5703 5704 case TrackBase::PAUSING: 5705 mActiveTracks.remove(activeTrack); 5706 mActiveTracksGen++; 5707 doBroadcast = true; 5708 size--; 5709 continue; 5710 5711 case TrackBase::STARTING_1: 5712 sleepUs = 10000; 5713 i++; 5714 continue; 5715 5716 case TrackBase::STARTING_2: 5717 doBroadcast = true; 5718 mStandby = false; 5719 activeTrack->mState = TrackBase::ACTIVE; 5720 break; 5721 5722 case TrackBase::ACTIVE: 5723 break; 5724 5725 case TrackBase::IDLE: 5726 i++; 5727 continue; 5728 5729 default: 5730 LOG_ALWAYS_FATAL("Unexpected activeTrackState %d", activeTrackState); 5731 } 5732 5733 activeTracks.add(activeTrack); 5734 i++; 5735 5736 if (activeTrack->isFastTrack()) { 5737 ALOG_ASSERT(!mFastTrackAvail); 5738 ALOG_ASSERT(fastTrack == 0); 5739 fastTrack = activeTrack; 5740 } 5741 } 5742 if (doBroadcast) { 5743 mStartStopCond.broadcast(); 5744 } 5745 5746 // sleep if there are no active tracks to process 5747 if (activeTracks.size() == 0) { 5748 if (sleepUs == 0) { 5749 sleepUs = kRecordThreadSleepUs; 5750 } 5751 continue; 5752 } 5753 sleepUs = 0; 5754 5755 lockEffectChains_l(effectChains); 5756 } 5757 5758 // thread mutex is now unlocked, mActiveTracks unknown, activeTracks.size() > 0 5759 5760 size_t size = effectChains.size(); 5761 for (size_t i = 0; i < size; i++) { 5762 // thread mutex is not locked, but effect chain is locked 5763 effectChains[i]->process_l(); 5764 } 5765 5766 // Push a new fast capture state if fast capture is not already running, or cblk change 5767 if (mFastCapture != 0) { 5768 FastCaptureStateQueue *sq = mFastCapture->sq(); 5769 FastCaptureState *state = sq->begin(); 5770 bool didModify = false; 5771 FastCaptureStateQueue::block_t block = FastCaptureStateQueue::BLOCK_UNTIL_PUSHED; 5772 if (state->mCommand != FastCaptureState::READ_WRITE /* FIXME && 5773 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)*/) { 5774 if (state->mCommand == FastCaptureState::COLD_IDLE) { 5775 int32_t old = android_atomic_inc(&mFastCaptureFutex); 5776 if (old == -1) { 5777 (void) syscall(__NR_futex, &mFastCaptureFutex, FUTEX_WAKE_PRIVATE, 1); 5778 } 5779 } 5780 state->mCommand = FastCaptureState::READ_WRITE; 5781#if 0 // FIXME 5782 mFastCaptureDumpState.increaseSamplingN(mAudioFlinger->isLowRamDevice() ? 5783 FastThreadDumpState::kSamplingNforLowRamDevice : 5784 FastThreadDumpState::kSamplingN); 5785#endif 5786 didModify = true; 5787 } 5788 audio_track_cblk_t *cblkOld = state->mCblk; 5789 audio_track_cblk_t *cblkNew = fastTrack != 0 ? fastTrack->cblk() : NULL; 5790 if (cblkNew != cblkOld) { 5791 state->mCblk = cblkNew; 5792 // block until acked if removing a fast track 5793 if (cblkOld != NULL) { 5794 block = FastCaptureStateQueue::BLOCK_UNTIL_ACKED; 5795 } 5796 didModify = true; 5797 } 5798 sq->end(didModify); 5799 if (didModify) { 5800 sq->push(block); 5801#if 0 5802 if (kUseFastCapture == FastCapture_Dynamic) { 5803 mNormalSource = mPipeSource; 5804 } 5805#endif 5806 } 5807 } 5808 5809 // now run the fast track destructor with thread mutex unlocked 5810 fastTrackToRemove.clear(); 5811 5812 // Read from HAL to keep up with fastest client if multiple active tracks, not slowest one. 5813 // Only the client(s) that are too slow will overrun. But if even the fastest client is too 5814 // slow, then this RecordThread will overrun by not calling HAL read often enough. 5815 // If destination is non-contiguous, first read past the nominal end of buffer, then 5816 // copy to the right place. Permitted because mRsmpInBuffer was over-allocated. 5817 5818 int32_t rear = mRsmpInRear & (mRsmpInFramesP2 - 1); 5819 ssize_t framesRead; 5820 5821 // If an NBAIO source is present, use it to read the normal capture's data 5822 if (mPipeSource != 0) { 5823 size_t framesToRead = mBufferSize / mFrameSize; 5824 framesRead = mPipeSource->read((uint8_t*)mRsmpInBuffer + rear * mFrameSize, 5825 framesToRead, AudioBufferProvider::kInvalidPTS); 5826 if (framesRead == 0) { 5827 // since pipe is non-blocking, simulate blocking input 5828 sleepUs = (framesToRead * 1000000LL) / mSampleRate; 5829 } 5830 // otherwise use the HAL / AudioStreamIn directly 5831 } else { 5832 ssize_t bytesRead = mInput->stream->read(mInput->stream, 5833 (uint8_t*)mRsmpInBuffer + rear * mFrameSize, mBufferSize); 5834 if (bytesRead < 0) { 5835 framesRead = bytesRead; 5836 } else { 5837 framesRead = bytesRead / mFrameSize; 5838 } 5839 } 5840 5841 if (framesRead < 0 || (framesRead == 0 && mPipeSource == 0)) { 5842 ALOGE("read failed: framesRead=%d", framesRead); 5843 // Force input into standby so that it tries to recover at next read attempt 5844 inputStandBy(); 5845 sleepUs = kRecordThreadSleepUs; 5846 } 5847 if (framesRead <= 0) { 5848 goto unlock; 5849 } 5850 ALOG_ASSERT(framesRead > 0); 5851 5852 if (mTeeSink != 0) { 5853 (void) mTeeSink->write((uint8_t*)mRsmpInBuffer + rear * mFrameSize, framesRead); 5854 } 5855 // If destination is non-contiguous, we now correct for reading past end of buffer. 5856 { 5857 size_t part1 = mRsmpInFramesP2 - rear; 5858 if ((size_t) framesRead > part1) { 5859 memcpy(mRsmpInBuffer, (uint8_t*)mRsmpInBuffer + mRsmpInFramesP2 * mFrameSize, 5860 (framesRead - part1) * mFrameSize); 5861 } 5862 } 5863 rear = mRsmpInRear += framesRead; 5864 5865 size = activeTracks.size(); 5866 // loop over each active track 5867 for (size_t i = 0; i < size; i++) { 5868 activeTrack = activeTracks[i]; 5869 5870 // skip fast tracks, as those are handled directly by FastCapture 5871 if (activeTrack->isFastTrack()) { 5872 continue; 5873 } 5874 5875 // TODO: This code probably should be moved to RecordTrack. 5876 // TODO: Update the activeTrack buffer converter in case of reconfigure. 5877 5878 enum { 5879 OVERRUN_UNKNOWN, 5880 OVERRUN_TRUE, 5881 OVERRUN_FALSE 5882 } overrun = OVERRUN_UNKNOWN; 5883 5884 // loop over getNextBuffer to handle circular sink 5885 for (;;) { 5886 5887 activeTrack->mSink.frameCount = ~0; 5888 status_t status = activeTrack->getNextBuffer(&activeTrack->mSink); 5889 size_t framesOut = activeTrack->mSink.frameCount; 5890 LOG_ALWAYS_FATAL_IF((status == OK) != (framesOut > 0)); 5891 5892 // check available frames and handle overrun conditions 5893 // if the record track isn't draining fast enough. 5894 bool hasOverrun; 5895 size_t framesIn; 5896 activeTrack->mResamplerBufferProvider->sync(&framesIn, &hasOverrun); 5897 if (hasOverrun) { 5898 overrun = OVERRUN_TRUE; 5899 } 5900 if (framesOut == 0 || framesIn == 0) { 5901 break; 5902 } 5903 5904 // Don't allow framesOut to be larger than what is possible with resampling 5905 // from framesIn. 5906 // This isn't strictly necessary but helps limit buffer resizing in 5907 // RecordBufferConverter. TODO: remove when no longer needed. 5908 framesOut = min(framesOut, 5909 destinationFramesPossible( 5910 framesIn, mSampleRate, activeTrack->mSampleRate)); 5911 // process frames from the RecordThread buffer provider to the RecordTrack buffer 5912 framesOut = activeTrack->mRecordBufferConverter->convert( 5913 activeTrack->mSink.raw, activeTrack->mResamplerBufferProvider, framesOut); 5914 5915 if (framesOut > 0 && (overrun == OVERRUN_UNKNOWN)) { 5916 overrun = OVERRUN_FALSE; 5917 } 5918 5919 if (activeTrack->mFramesToDrop == 0) { 5920 if (framesOut > 0) { 5921 activeTrack->mSink.frameCount = framesOut; 5922 activeTrack->releaseBuffer(&activeTrack->mSink); 5923 } 5924 } else { 5925 // FIXME could do a partial drop of framesOut 5926 if (activeTrack->mFramesToDrop > 0) { 5927 activeTrack->mFramesToDrop -= framesOut; 5928 if (activeTrack->mFramesToDrop <= 0) { 5929 activeTrack->clearSyncStartEvent(); 5930 } 5931 } else { 5932 activeTrack->mFramesToDrop += framesOut; 5933 if (activeTrack->mFramesToDrop >= 0 || activeTrack->mSyncStartEvent == 0 || 5934 activeTrack->mSyncStartEvent->isCancelled()) { 5935 ALOGW("Synced record %s, session %d, trigger session %d", 5936 (activeTrack->mFramesToDrop >= 0) ? "timed out" : "cancelled", 5937 activeTrack->sessionId(), 5938 (activeTrack->mSyncStartEvent != 0) ? 5939 activeTrack->mSyncStartEvent->triggerSession() : 0); 5940 activeTrack->clearSyncStartEvent(); 5941 } 5942 } 5943 } 5944 5945 if (framesOut == 0) { 5946 break; 5947 } 5948 } 5949 5950 switch (overrun) { 5951 case OVERRUN_TRUE: 5952 // client isn't retrieving buffers fast enough 5953 if (!activeTrack->setOverflow()) { 5954 nsecs_t now = systemTime(); 5955 // FIXME should lastWarning per track? 5956 if ((now - lastWarning) > kWarningThrottleNs) { 5957 ALOGW("RecordThread: buffer overflow"); 5958 lastWarning = now; 5959 } 5960 } 5961 break; 5962 case OVERRUN_FALSE: 5963 activeTrack->clearOverflow(); 5964 break; 5965 case OVERRUN_UNKNOWN: 5966 break; 5967 } 5968 5969 } 5970 5971unlock: 5972 // enable changes in effect chain 5973 unlockEffectChains(effectChains); 5974 // effectChains doesn't need to be cleared, since it is cleared by destructor at scope end 5975 } 5976 5977 standbyIfNotAlreadyInStandby(); 5978 5979 { 5980 Mutex::Autolock _l(mLock); 5981 for (size_t i = 0; i < mTracks.size(); i++) { 5982 sp<RecordTrack> track = mTracks[i]; 5983 track->invalidate(); 5984 } 5985 mActiveTracks.clear(); 5986 mActiveTracksGen++; 5987 mStartStopCond.broadcast(); 5988 } 5989 5990 releaseWakeLock(); 5991 5992 ALOGV("RecordThread %p exiting", this); 5993 return false; 5994} 5995 5996void AudioFlinger::RecordThread::standbyIfNotAlreadyInStandby() 5997{ 5998 if (!mStandby) { 5999 inputStandBy(); 6000 mStandby = true; 6001 } 6002} 6003 6004void AudioFlinger::RecordThread::inputStandBy() 6005{ 6006 // Idle the fast capture if it's currently running 6007 if (mFastCapture != 0) { 6008 FastCaptureStateQueue *sq = mFastCapture->sq(); 6009 FastCaptureState *state = sq->begin(); 6010 if (!(state->mCommand & FastCaptureState::IDLE)) { 6011 state->mCommand = FastCaptureState::COLD_IDLE; 6012 state->mColdFutexAddr = &mFastCaptureFutex; 6013 state->mColdGen++; 6014 mFastCaptureFutex = 0; 6015 sq->end(); 6016 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 6017 sq->push(FastCaptureStateQueue::BLOCK_UNTIL_ACKED); 6018#if 0 6019 if (kUseFastCapture == FastCapture_Dynamic) { 6020 // FIXME 6021 } 6022#endif 6023#ifdef AUDIO_WATCHDOG 6024 // FIXME 6025#endif 6026 } else { 6027 sq->end(false /*didModify*/); 6028 } 6029 } 6030 mInput->stream->common.standby(&mInput->stream->common); 6031} 6032 6033// RecordThread::createRecordTrack_l() must be called with AudioFlinger::mLock held 6034sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 6035 const sp<AudioFlinger::Client>& client, 6036 uint32_t sampleRate, 6037 audio_format_t format, 6038 audio_channel_mask_t channelMask, 6039 size_t *pFrameCount, 6040 int sessionId, 6041 size_t *notificationFrames, 6042 int uid, 6043 IAudioFlinger::track_flags_t *flags, 6044 pid_t tid, 6045 status_t *status) 6046{ 6047 size_t frameCount = *pFrameCount; 6048 sp<RecordTrack> track; 6049 status_t lStatus; 6050 6051 // client expresses a preference for FAST, but we get the final say 6052 if (*flags & IAudioFlinger::TRACK_FAST) { 6053 if ( 6054 // we formerly checked for a callback handler (non-0 tid), 6055 // but that is no longer required for TRANSFER_OBTAIN mode 6056 // 6057 // frame count is not specified, or is exactly the pipe depth 6058 ((frameCount == 0) || (frameCount == mPipeFramesP2)) && 6059 // PCM data 6060 audio_is_linear_pcm(format) && 6061 // native format 6062 (format == mFormat) && 6063 // native channel mask 6064 (channelMask == mChannelMask) && 6065 // native hardware sample rate 6066 (sampleRate == mSampleRate) && 6067 // record thread has an associated fast capture 6068 hasFastCapture() && 6069 // there are sufficient fast track slots available 6070 mFastTrackAvail 6071 ) { 6072 ALOGV("AUDIO_INPUT_FLAG_FAST accepted: frameCount=%u mFrameCount=%u", 6073 frameCount, mFrameCount); 6074 } else { 6075 ALOGV("AUDIO_INPUT_FLAG_FAST denied: frameCount=%u mFrameCount=%u mPipeFramesP2=%u " 6076 "format=%#x isLinear=%d channelMask=%#x sampleRate=%u mSampleRate=%u " 6077 "hasFastCapture=%d tid=%d mFastTrackAvail=%d", 6078 frameCount, mFrameCount, mPipeFramesP2, 6079 format, audio_is_linear_pcm(format), channelMask, sampleRate, mSampleRate, 6080 hasFastCapture(), tid, mFastTrackAvail); 6081 *flags &= ~IAudioFlinger::TRACK_FAST; 6082 } 6083 } 6084 6085 // compute track buffer size in frames, and suggest the notification frame count 6086 if (*flags & IAudioFlinger::TRACK_FAST) { 6087 // fast track: frame count is exactly the pipe depth 6088 frameCount = mPipeFramesP2; 6089 // ignore requested notificationFrames, and always notify exactly once every HAL buffer 6090 *notificationFrames = mFrameCount; 6091 } else { 6092 // not fast track: max notification period is resampled equivalent of one HAL buffer time 6093 // or 20 ms if there is a fast capture 6094 // TODO This could be a roundupRatio inline, and const 6095 size_t maxNotificationFrames = ((int64_t) (hasFastCapture() ? mSampleRate/50 : mFrameCount) 6096 * sampleRate + mSampleRate - 1) / mSampleRate; 6097 // minimum number of notification periods is at least kMinNotifications, 6098 // and at least kMinMs rounded up to a whole notification period (minNotificationsByMs) 6099 static const size_t kMinNotifications = 3; 6100 static const uint32_t kMinMs = 30; 6101 // TODO This could be a roundupRatio inline 6102 const size_t minFramesByMs = (sampleRate * kMinMs + 1000 - 1) / 1000; 6103 // TODO This could be a roundupRatio inline 6104 const size_t minNotificationsByMs = (minFramesByMs + maxNotificationFrames - 1) / 6105 maxNotificationFrames; 6106 const size_t minFrameCount = maxNotificationFrames * 6107 max(kMinNotifications, minNotificationsByMs); 6108 frameCount = max(frameCount, minFrameCount); 6109 if (*notificationFrames == 0 || *notificationFrames > maxNotificationFrames) { 6110 *notificationFrames = maxNotificationFrames; 6111 } 6112 } 6113 *pFrameCount = frameCount; 6114 6115 lStatus = initCheck(); 6116 if (lStatus != NO_ERROR) { 6117 ALOGE("createRecordTrack_l() audio driver not initialized"); 6118 goto Exit; 6119 } 6120 6121 { // scope for mLock 6122 Mutex::Autolock _l(mLock); 6123 6124 track = new RecordTrack(this, client, sampleRate, 6125 format, channelMask, frameCount, NULL, sessionId, uid, 6126 *flags, TrackBase::TYPE_DEFAULT); 6127 6128 lStatus = track->initCheck(); 6129 if (lStatus != NO_ERROR) { 6130 ALOGE("createRecordTrack_l() initCheck failed %d; no control block?", lStatus); 6131 // track must be cleared from the caller as the caller has the AF lock 6132 goto Exit; 6133 } 6134 mTracks.add(track); 6135 6136 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 6137 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 6138 mAudioFlinger->btNrecIsOff(); 6139 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 6140 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 6141 6142 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 6143 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 6144 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 6145 // so ask activity manager to do this on our behalf 6146 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 6147 } 6148 } 6149 6150 lStatus = NO_ERROR; 6151 6152Exit: 6153 *status = lStatus; 6154 return track; 6155} 6156 6157status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, 6158 AudioSystem::sync_event_t event, 6159 int triggerSession) 6160{ 6161 ALOGV("RecordThread::start event %d, triggerSession %d", event, triggerSession); 6162 sp<ThreadBase> strongMe = this; 6163 status_t status = NO_ERROR; 6164 6165 if (event == AudioSystem::SYNC_EVENT_NONE) { 6166 recordTrack->clearSyncStartEvent(); 6167 } else if (event != AudioSystem::SYNC_EVENT_SAME) { 6168 recordTrack->mSyncStartEvent = mAudioFlinger->createSyncEvent(event, 6169 triggerSession, 6170 recordTrack->sessionId(), 6171 syncStartEventCallback, 6172 recordTrack); 6173 // Sync event can be cancelled by the trigger session if the track is not in a 6174 // compatible state in which case we start record immediately 6175 if (recordTrack->mSyncStartEvent->isCancelled()) { 6176 recordTrack->clearSyncStartEvent(); 6177 } else { 6178 // do not wait for the event for more than AudioSystem::kSyncRecordStartTimeOutMs 6179 recordTrack->mFramesToDrop = - 6180 ((AudioSystem::kSyncRecordStartTimeOutMs * recordTrack->mSampleRate) / 1000); 6181 } 6182 } 6183 6184 { 6185 // This section is a rendezvous between binder thread executing start() and RecordThread 6186 AutoMutex lock(mLock); 6187 if (mActiveTracks.indexOf(recordTrack) >= 0) { 6188 if (recordTrack->mState == TrackBase::PAUSING) { 6189 ALOGV("active record track PAUSING -> ACTIVE"); 6190 recordTrack->mState = TrackBase::ACTIVE; 6191 } else { 6192 ALOGV("active record track state %d", recordTrack->mState); 6193 } 6194 return status; 6195 } 6196 6197 // TODO consider other ways of handling this, such as changing the state to :STARTING and 6198 // adding the track to mActiveTracks after returning from AudioSystem::startInput(), 6199 // or using a separate command thread 6200 recordTrack->mState = TrackBase::STARTING_1; 6201 mActiveTracks.add(recordTrack); 6202 mActiveTracksGen++; 6203 status_t status = NO_ERROR; 6204 if (recordTrack->isExternalTrack()) { 6205 mLock.unlock(); 6206 status = AudioSystem::startInput(mId, (audio_session_t)recordTrack->sessionId()); 6207 mLock.lock(); 6208 // FIXME should verify that recordTrack is still in mActiveTracks 6209 if (status != NO_ERROR) { 6210 mActiveTracks.remove(recordTrack); 6211 mActiveTracksGen++; 6212 recordTrack->clearSyncStartEvent(); 6213 ALOGV("RecordThread::start error %d", status); 6214 return status; 6215 } 6216 } 6217 // Catch up with current buffer indices if thread is already running. 6218 // This is what makes a new client discard all buffered data. If the track's mRsmpInFront 6219 // was initialized to some value closer to the thread's mRsmpInFront, then the track could 6220 // see previously buffered data before it called start(), but with greater risk of overrun. 6221 6222 recordTrack->mResamplerBufferProvider->reset(); 6223 // clear any converter state as new data will be discontinuous 6224 recordTrack->mRecordBufferConverter->reset(); 6225 recordTrack->mState = TrackBase::STARTING_2; 6226 // signal thread to start 6227 mWaitWorkCV.broadcast(); 6228 if (mActiveTracks.indexOf(recordTrack) < 0) { 6229 ALOGV("Record failed to start"); 6230 status = BAD_VALUE; 6231 goto startError; 6232 } 6233 return status; 6234 } 6235 6236startError: 6237 if (recordTrack->isExternalTrack()) { 6238 AudioSystem::stopInput(mId, (audio_session_t)recordTrack->sessionId()); 6239 } 6240 recordTrack->clearSyncStartEvent(); 6241 // FIXME I wonder why we do not reset the state here? 6242 return status; 6243} 6244 6245void AudioFlinger::RecordThread::syncStartEventCallback(const wp<SyncEvent>& event) 6246{ 6247 sp<SyncEvent> strongEvent = event.promote(); 6248 6249 if (strongEvent != 0) { 6250 sp<RefBase> ptr = strongEvent->cookie().promote(); 6251 if (ptr != 0) { 6252 RecordTrack *recordTrack = (RecordTrack *)ptr.get(); 6253 recordTrack->handleSyncStartEvent(strongEvent); 6254 } 6255 } 6256} 6257 6258bool AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 6259 ALOGV("RecordThread::stop"); 6260 AutoMutex _l(mLock); 6261 if (mActiveTracks.indexOf(recordTrack) != 0 || recordTrack->mState == TrackBase::PAUSING) { 6262 return false; 6263 } 6264 // note that threadLoop may still be processing the track at this point [without lock] 6265 recordTrack->mState = TrackBase::PAUSING; 6266 // do not wait for mStartStopCond if exiting 6267 if (exitPending()) { 6268 return true; 6269 } 6270 // FIXME incorrect usage of wait: no explicit predicate or loop 6271 mStartStopCond.wait(mLock); 6272 // if we have been restarted, recordTrack is in mActiveTracks here 6273 if (exitPending() || mActiveTracks.indexOf(recordTrack) != 0) { 6274 ALOGV("Record stopped OK"); 6275 return true; 6276 } 6277 return false; 6278} 6279 6280bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event __unused) const 6281{ 6282 return false; 6283} 6284 6285status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event __unused) 6286{ 6287#if 0 // This branch is currently dead code, but is preserved in case it will be needed in future 6288 if (!isValidSyncEvent(event)) { 6289 return BAD_VALUE; 6290 } 6291 6292 int eventSession = event->triggerSession(); 6293 status_t ret = NAME_NOT_FOUND; 6294 6295 Mutex::Autolock _l(mLock); 6296 6297 for (size_t i = 0; i < mTracks.size(); i++) { 6298 sp<RecordTrack> track = mTracks[i]; 6299 if (eventSession == track->sessionId()) { 6300 (void) track->setSyncEvent(event); 6301 ret = NO_ERROR; 6302 } 6303 } 6304 return ret; 6305#else 6306 return BAD_VALUE; 6307#endif 6308} 6309 6310// destroyTrack_l() must be called with ThreadBase::mLock held 6311void AudioFlinger::RecordThread::destroyTrack_l(const sp<RecordTrack>& track) 6312{ 6313 track->terminate(); 6314 track->mState = TrackBase::STOPPED; 6315 // active tracks are removed by threadLoop() 6316 if (mActiveTracks.indexOf(track) < 0) { 6317 removeTrack_l(track); 6318 } 6319} 6320 6321void AudioFlinger::RecordThread::removeTrack_l(const sp<RecordTrack>& track) 6322{ 6323 mTracks.remove(track); 6324 // need anything related to effects here? 6325 if (track->isFastTrack()) { 6326 ALOG_ASSERT(!mFastTrackAvail); 6327 mFastTrackAvail = true; 6328 } 6329} 6330 6331void AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 6332{ 6333 dumpInternals(fd, args); 6334 dumpTracks(fd, args); 6335 dumpEffectChains(fd, args); 6336} 6337 6338void AudioFlinger::RecordThread::dumpInternals(int fd, const Vector<String16>& args) 6339{ 6340 dprintf(fd, "\nInput thread %p:\n", this); 6341 6342 dumpBase(fd, args); 6343 6344 if (mActiveTracks.size() == 0) { 6345 dprintf(fd, " No active record clients\n"); 6346 } 6347 dprintf(fd, " Fast capture thread: %s\n", hasFastCapture() ? "yes" : "no"); 6348 dprintf(fd, " Fast track available: %s\n", mFastTrackAvail ? "yes" : "no"); 6349 6350 // Make a non-atomic copy of fast capture dump state so it won't change underneath us 6351 const FastCaptureDumpState copy(mFastCaptureDumpState); 6352 copy.dump(fd); 6353} 6354 6355void AudioFlinger::RecordThread::dumpTracks(int fd, const Vector<String16>& args __unused) 6356{ 6357 const size_t SIZE = 256; 6358 char buffer[SIZE]; 6359 String8 result; 6360 6361 size_t numtracks = mTracks.size(); 6362 size_t numactive = mActiveTracks.size(); 6363 size_t numactiveseen = 0; 6364 dprintf(fd, " %d Tracks", numtracks); 6365 if (numtracks) { 6366 dprintf(fd, " of which %d are active\n", numactive); 6367 RecordTrack::appendDumpHeader(result); 6368 for (size_t i = 0; i < numtracks ; ++i) { 6369 sp<RecordTrack> track = mTracks[i]; 6370 if (track != 0) { 6371 bool active = mActiveTracks.indexOf(track) >= 0; 6372 if (active) { 6373 numactiveseen++; 6374 } 6375 track->dump(buffer, SIZE, active); 6376 result.append(buffer); 6377 } 6378 } 6379 } else { 6380 dprintf(fd, "\n"); 6381 } 6382 6383 if (numactiveseen != numactive) { 6384 snprintf(buffer, SIZE, " The following tracks are in the active list but" 6385 " not in the track list\n"); 6386 result.append(buffer); 6387 RecordTrack::appendDumpHeader(result); 6388 for (size_t i = 0; i < numactive; ++i) { 6389 sp<RecordTrack> track = mActiveTracks[i]; 6390 if (mTracks.indexOf(track) < 0) { 6391 track->dump(buffer, SIZE, true); 6392 result.append(buffer); 6393 } 6394 } 6395 6396 } 6397 write(fd, result.string(), result.size()); 6398} 6399 6400 6401void AudioFlinger::RecordThread::ResamplerBufferProvider::reset() 6402{ 6403 sp<ThreadBase> threadBase = mRecordTrack->mThread.promote(); 6404 RecordThread *recordThread = (RecordThread *) threadBase.get(); 6405 mRsmpInFront = recordThread->mRsmpInRear; 6406 mRsmpInUnrel = 0; 6407} 6408 6409void AudioFlinger::RecordThread::ResamplerBufferProvider::sync( 6410 size_t *framesAvailable, bool *hasOverrun) 6411{ 6412 sp<ThreadBase> threadBase = mRecordTrack->mThread.promote(); 6413 RecordThread *recordThread = (RecordThread *) threadBase.get(); 6414 const int32_t rear = recordThread->mRsmpInRear; 6415 const int32_t front = mRsmpInFront; 6416 const ssize_t filled = rear - front; 6417 6418 size_t framesIn; 6419 bool overrun = false; 6420 if (filled < 0) { 6421 // should not happen, but treat like a massive overrun and re-sync 6422 framesIn = 0; 6423 mRsmpInFront = rear; 6424 overrun = true; 6425 } else if ((size_t) filled <= recordThread->mRsmpInFrames) { 6426 framesIn = (size_t) filled; 6427 } else { 6428 // client is not keeping up with server, but give it latest data 6429 framesIn = recordThread->mRsmpInFrames; 6430 mRsmpInFront = /* front = */ rear - framesIn; 6431 overrun = true; 6432 } 6433 if (framesAvailable != NULL) { 6434 *framesAvailable = framesIn; 6435 } 6436 if (hasOverrun != NULL) { 6437 *hasOverrun = overrun; 6438 } 6439} 6440 6441// AudioBufferProvider interface 6442status_t AudioFlinger::RecordThread::ResamplerBufferProvider::getNextBuffer( 6443 AudioBufferProvider::Buffer* buffer, int64_t pts __unused) 6444{ 6445 sp<ThreadBase> threadBase = mRecordTrack->mThread.promote(); 6446 if (threadBase == 0) { 6447 buffer->frameCount = 0; 6448 buffer->raw = NULL; 6449 return NOT_ENOUGH_DATA; 6450 } 6451 RecordThread *recordThread = (RecordThread *) threadBase.get(); 6452 int32_t rear = recordThread->mRsmpInRear; 6453 int32_t front = mRsmpInFront; 6454 ssize_t filled = rear - front; 6455 // FIXME should not be P2 (don't want to increase latency) 6456 // FIXME if client not keeping up, discard 6457 LOG_ALWAYS_FATAL_IF(!(0 <= filled && (size_t) filled <= recordThread->mRsmpInFrames)); 6458 // 'filled' may be non-contiguous, so return only the first contiguous chunk 6459 front &= recordThread->mRsmpInFramesP2 - 1; 6460 size_t part1 = recordThread->mRsmpInFramesP2 - front; 6461 if (part1 > (size_t) filled) { 6462 part1 = filled; 6463 } 6464 size_t ask = buffer->frameCount; 6465 ALOG_ASSERT(ask > 0); 6466 if (part1 > ask) { 6467 part1 = ask; 6468 } 6469 if (part1 == 0) { 6470 // out of data is fine since the resampler will return a short-count. 6471 buffer->raw = NULL; 6472 buffer->frameCount = 0; 6473 mRsmpInUnrel = 0; 6474 return NOT_ENOUGH_DATA; 6475 } 6476 6477 buffer->raw = (uint8_t*)recordThread->mRsmpInBuffer + front * recordThread->mFrameSize; 6478 buffer->frameCount = part1; 6479 mRsmpInUnrel = part1; 6480 return NO_ERROR; 6481} 6482 6483// AudioBufferProvider interface 6484void AudioFlinger::RecordThread::ResamplerBufferProvider::releaseBuffer( 6485 AudioBufferProvider::Buffer* buffer) 6486{ 6487 size_t stepCount = buffer->frameCount; 6488 if (stepCount == 0) { 6489 return; 6490 } 6491 ALOG_ASSERT(stepCount <= mRsmpInUnrel); 6492 mRsmpInUnrel -= stepCount; 6493 mRsmpInFront += stepCount; 6494 buffer->raw = NULL; 6495 buffer->frameCount = 0; 6496} 6497 6498AudioFlinger::RecordThread::RecordBufferConverter::RecordBufferConverter( 6499 audio_channel_mask_t srcChannelMask, audio_format_t srcFormat, 6500 uint32_t srcSampleRate, 6501 audio_channel_mask_t dstChannelMask, audio_format_t dstFormat, 6502 uint32_t dstSampleRate) : 6503 mSrcChannelMask(AUDIO_CHANNEL_INVALID), // updateParameters will set following vars 6504 // mSrcFormat 6505 // mSrcSampleRate 6506 // mDstChannelMask 6507 // mDstFormat 6508 // mDstSampleRate 6509 // mSrcChannelCount 6510 // mDstChannelCount 6511 // mDstFrameSize 6512 mBuf(NULL), mBufFrames(0), mBufFrameSize(0), 6513 mResampler(NULL), 6514 mIsLegacyDownmix(false), 6515 mIsLegacyUpmix(false), 6516 mRequiresFloat(false), 6517 mInputConverterProvider(NULL) 6518{ 6519 (void)updateParameters(srcChannelMask, srcFormat, srcSampleRate, 6520 dstChannelMask, dstFormat, dstSampleRate); 6521} 6522 6523AudioFlinger::RecordThread::RecordBufferConverter::~RecordBufferConverter() { 6524 free(mBuf); 6525 delete mResampler; 6526 delete mInputConverterProvider; 6527} 6528 6529size_t AudioFlinger::RecordThread::RecordBufferConverter::convert(void *dst, 6530 AudioBufferProvider *provider, size_t frames) 6531{ 6532 if (mInputConverterProvider != NULL) { 6533 mInputConverterProvider->setBufferProvider(provider); 6534 provider = mInputConverterProvider; 6535 } 6536 6537 if (mResampler == NULL) { 6538 ALOGVV("NO RESAMPLING sampleRate:%u mSrcFormat:%#x mDstFormat:%#x", 6539 mSrcSampleRate, mSrcFormat, mDstFormat); 6540 6541 AudioBufferProvider::Buffer buffer; 6542 for (size_t i = frames; i > 0; ) { 6543 buffer.frameCount = i; 6544 status_t status = provider->getNextBuffer(&buffer, 0); 6545 if (status != OK || buffer.frameCount == 0) { 6546 frames -= i; // cannot fill request. 6547 break; 6548 } 6549 // format convert to destination buffer 6550 convertNoResampler(dst, buffer.raw, buffer.frameCount); 6551 6552 dst = (int8_t*)dst + buffer.frameCount * mDstFrameSize; 6553 i -= buffer.frameCount; 6554 provider->releaseBuffer(&buffer); 6555 } 6556 } else { 6557 ALOGVV("RESAMPLING mSrcSampleRate:%u mDstSampleRate:%u mSrcFormat:%#x mDstFormat:%#x", 6558 mSrcSampleRate, mDstSampleRate, mSrcFormat, mDstFormat); 6559 6560 // reallocate buffer if needed 6561 if (mBufFrameSize != 0 && mBufFrames < frames) { 6562 free(mBuf); 6563 mBufFrames = frames; 6564 (void)posix_memalign(&mBuf, 32, mBufFrames * mBufFrameSize); 6565 } 6566 // resampler accumulates, but we only have one source track 6567 memset(mBuf, 0, frames * mBufFrameSize); 6568 frames = mResampler->resample((int32_t*)mBuf, frames, provider); 6569 // format convert to destination buffer 6570 convertResampler(dst, mBuf, frames); 6571 } 6572 return frames; 6573} 6574 6575status_t AudioFlinger::RecordThread::RecordBufferConverter::updateParameters( 6576 audio_channel_mask_t srcChannelMask, audio_format_t srcFormat, 6577 uint32_t srcSampleRate, 6578 audio_channel_mask_t dstChannelMask, audio_format_t dstFormat, 6579 uint32_t dstSampleRate) 6580{ 6581 // quick evaluation if there is any change. 6582 if (mSrcFormat == srcFormat 6583 && mSrcChannelMask == srcChannelMask 6584 && mSrcSampleRate == srcSampleRate 6585 && mDstFormat == dstFormat 6586 && mDstChannelMask == dstChannelMask 6587 && mDstSampleRate == dstSampleRate) { 6588 return NO_ERROR; 6589 } 6590 6591 ALOGV("RecordBufferConverter updateParameters srcMask:%#x dstMask:%#x" 6592 " srcFormat:%#x dstFormat:%#x srcRate:%u dstRate:%u", 6593 srcChannelMask, dstChannelMask, srcFormat, dstFormat, srcSampleRate, dstSampleRate); 6594 const bool valid = 6595 audio_is_input_channel(srcChannelMask) 6596 && audio_is_input_channel(dstChannelMask) 6597 && audio_is_valid_format(srcFormat) && audio_is_linear_pcm(srcFormat) 6598 && audio_is_valid_format(dstFormat) && audio_is_linear_pcm(dstFormat) 6599 && (srcSampleRate <= dstSampleRate * AUDIO_RESAMPLER_DOWN_RATIO_MAX) 6600 ; // no upsampling checks for now 6601 if (!valid) { 6602 return BAD_VALUE; 6603 } 6604 6605 mSrcFormat = srcFormat; 6606 mSrcChannelMask = srcChannelMask; 6607 mSrcSampleRate = srcSampleRate; 6608 mDstFormat = dstFormat; 6609 mDstChannelMask = dstChannelMask; 6610 mDstSampleRate = dstSampleRate; 6611 6612 // compute derived parameters 6613 mSrcChannelCount = audio_channel_count_from_in_mask(srcChannelMask); 6614 mDstChannelCount = audio_channel_count_from_in_mask(dstChannelMask); 6615 mDstFrameSize = mDstChannelCount * audio_bytes_per_sample(mDstFormat); 6616 6617 // do we need to resample? 6618 delete mResampler; 6619 mResampler = NULL; 6620 if (mSrcSampleRate != mDstSampleRate) { 6621 mResampler = AudioResampler::create(AUDIO_FORMAT_PCM_FLOAT, 6622 mSrcChannelCount, mDstSampleRate); 6623 mResampler->setSampleRate(mSrcSampleRate); 6624 mResampler->setVolume(AudioMixer::UNITY_GAIN_FLOAT, AudioMixer::UNITY_GAIN_FLOAT); 6625 } 6626 6627 // are we running legacy channel conversion modes? 6628 mIsLegacyDownmix = (mSrcChannelMask == AUDIO_CHANNEL_IN_STEREO 6629 || mSrcChannelMask == AUDIO_CHANNEL_IN_FRONT_BACK) 6630 && mDstChannelMask == AUDIO_CHANNEL_IN_MONO; 6631 mIsLegacyUpmix = mSrcChannelMask == AUDIO_CHANNEL_IN_MONO 6632 && (mDstChannelMask == AUDIO_CHANNEL_IN_STEREO 6633 || mDstChannelMask == AUDIO_CHANNEL_IN_FRONT_BACK); 6634 6635 // do we need to process in float? 6636 mRequiresFloat = mResampler != NULL || mIsLegacyDownmix || mIsLegacyUpmix; 6637 6638 // do we need a staging buffer to convert for destination (we can still optimize this)? 6639 // we use mBufFrameSize > 0 to indicate both frame size as well as buffer necessity 6640 if (mResampler != NULL) { 6641 mBufFrameSize = max(mSrcChannelCount, FCC_2) 6642 * audio_bytes_per_sample(AUDIO_FORMAT_PCM_FLOAT); 6643 } else if ((mIsLegacyUpmix || mIsLegacyDownmix) && mDstFormat != AUDIO_FORMAT_PCM_FLOAT) { 6644 mBufFrameSize = mDstChannelCount * audio_bytes_per_sample(AUDIO_FORMAT_PCM_FLOAT); 6645 } else if (mSrcChannelMask != mDstChannelMask && mDstFormat != mSrcFormat) { 6646 mBufFrameSize = mDstChannelCount * audio_bytes_per_sample(mSrcFormat); 6647 } else { 6648 mBufFrameSize = 0; 6649 } 6650 mBufFrames = 0; // force the buffer to be resized. 6651 6652 // do we need an input converter buffer provider to give us float? 6653 delete mInputConverterProvider; 6654 mInputConverterProvider = NULL; 6655 if (mRequiresFloat && mSrcFormat != AUDIO_FORMAT_PCM_FLOAT) { 6656 mInputConverterProvider = new ReformatBufferProvider( 6657 audio_channel_count_from_in_mask(mSrcChannelMask), 6658 mSrcFormat, 6659 AUDIO_FORMAT_PCM_FLOAT, 6660 256 /* provider buffer frame count */); 6661 } 6662 6663 // do we need a remixer to do channel mask conversion 6664 if (!mIsLegacyDownmix && !mIsLegacyUpmix && mSrcChannelMask != mDstChannelMask) { 6665 (void) memcpy_by_index_array_initialization_from_channel_mask( 6666 mIdxAry, ARRAY_SIZE(mIdxAry), mDstChannelMask, mSrcChannelMask); 6667 } 6668 return NO_ERROR; 6669} 6670 6671void AudioFlinger::RecordThread::RecordBufferConverter::convertNoResampler( 6672 void *dst, const void *src, size_t frames) 6673{ 6674 // src is native type unless there is legacy upmix or downmix, whereupon it is float. 6675 if (mBufFrameSize != 0 && mBufFrames < frames) { 6676 free(mBuf); 6677 mBufFrames = frames; 6678 (void)posix_memalign(&mBuf, 32, mBufFrames * mBufFrameSize); 6679 } 6680 // do we need to do legacy upmix and downmix? 6681 if (mIsLegacyUpmix || mIsLegacyDownmix) { 6682 void *dstBuf = mBuf != NULL ? mBuf : dst; 6683 if (mIsLegacyUpmix) { 6684 upmix_to_stereo_float_from_mono_float((float *)dstBuf, 6685 (const float *)src, frames); 6686 } else /*mIsLegacyDownmix */ { 6687 downmix_to_mono_float_from_stereo_float((float *)dstBuf, 6688 (const float *)src, frames); 6689 } 6690 if (mBuf != NULL) { 6691 memcpy_by_audio_format(dst, mDstFormat, mBuf, AUDIO_FORMAT_PCM_FLOAT, 6692 frames * mDstChannelCount); 6693 } 6694 return; 6695 } 6696 // do we need to do channel mask conversion? 6697 if (mSrcChannelMask != mDstChannelMask) { 6698 void *dstBuf = mBuf != NULL ? mBuf : dst; 6699 memcpy_by_index_array(dstBuf, mDstChannelCount, 6700 src, mSrcChannelCount, mIdxAry, audio_bytes_per_sample(mSrcFormat), frames); 6701 if (dstBuf == dst) { 6702 return; // format is the same 6703 } 6704 } 6705 // convert to destination buffer 6706 const void *convertBuf = mBuf != NULL ? mBuf : src; 6707 memcpy_by_audio_format(dst, mDstFormat, convertBuf, mSrcFormat, 6708 frames * mDstChannelCount); 6709} 6710 6711void AudioFlinger::RecordThread::RecordBufferConverter::convertResampler( 6712 void *dst, /*not-a-const*/ void *src, size_t frames) 6713{ 6714 // src buffer format is ALWAYS float when entering this routine 6715 if (mIsLegacyUpmix) { 6716 ; // mono to stereo already handled by resampler 6717 } else if (mIsLegacyDownmix 6718 || (mSrcChannelMask == mDstChannelMask && mSrcChannelCount == 1)) { 6719 // the resampler outputs stereo for mono input channel (a feature?) 6720 // must convert to mono 6721 downmix_to_mono_float_from_stereo_float((float *)src, 6722 (const float *)src, frames); 6723 } else if (mSrcChannelMask != mDstChannelMask) { 6724 // convert to mono channel again for channel mask conversion (could be skipped 6725 // with further optimization). 6726 if (mSrcChannelCount == 1) { 6727 downmix_to_mono_float_from_stereo_float((float *)src, 6728 (const float *)src, frames); 6729 } 6730 // convert to destination format (in place, OK as float is larger than other types) 6731 if (mDstFormat != AUDIO_FORMAT_PCM_FLOAT) { 6732 memcpy_by_audio_format(src, mDstFormat, src, AUDIO_FORMAT_PCM_FLOAT, 6733 frames * mSrcChannelCount); 6734 } 6735 // channel convert and save to dst 6736 memcpy_by_index_array(dst, mDstChannelCount, 6737 src, mSrcChannelCount, mIdxAry, audio_bytes_per_sample(mDstFormat), frames); 6738 return; 6739 } 6740 // convert to destination format and save to dst 6741 memcpy_by_audio_format(dst, mDstFormat, src, AUDIO_FORMAT_PCM_FLOAT, 6742 frames * mDstChannelCount); 6743} 6744 6745bool AudioFlinger::RecordThread::checkForNewParameter_l(const String8& keyValuePair, 6746 status_t& status) 6747{ 6748 bool reconfig = false; 6749 6750 status = NO_ERROR; 6751 6752 audio_format_t reqFormat = mFormat; 6753 uint32_t samplingRate = mSampleRate; 6754 // TODO this may change if we want to support capture from HDMI PCM multi channel (e.g on TVs). 6755 audio_channel_mask_t channelMask = audio_channel_in_mask_from_count(mChannelCount); 6756 6757 AudioParameter param = AudioParameter(keyValuePair); 6758 int value; 6759 // TODO Investigate when this code runs. Check with audio policy when a sample rate and 6760 // channel count change can be requested. Do we mandate the first client defines the 6761 // HAL sampling rate and channel count or do we allow changes on the fly? 6762 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 6763 samplingRate = value; 6764 reconfig = true; 6765 } 6766 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 6767 if (!audio_is_linear_pcm((audio_format_t) value)) { 6768 status = BAD_VALUE; 6769 } else { 6770 reqFormat = (audio_format_t) value; 6771 reconfig = true; 6772 } 6773 } 6774 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 6775 audio_channel_mask_t mask = (audio_channel_mask_t) value; 6776 if (!audio_is_input_channel(mask) || 6777 audio_channel_count_from_in_mask(mask) > FCC_8) { 6778 status = BAD_VALUE; 6779 } else { 6780 channelMask = mask; 6781 reconfig = true; 6782 } 6783 } 6784 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 6785 // do not accept frame count changes if tracks are open as the track buffer 6786 // size depends on frame count and correct behavior would not be guaranteed 6787 // if frame count is changed after track creation 6788 if (mActiveTracks.size() > 0) { 6789 status = INVALID_OPERATION; 6790 } else { 6791 reconfig = true; 6792 } 6793 } 6794 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 6795 // forward device change to effects that have requested to be 6796 // aware of attached audio device. 6797 for (size_t i = 0; i < mEffectChains.size(); i++) { 6798 mEffectChains[i]->setDevice_l(value); 6799 } 6800 6801 // store input device and output device but do not forward output device to audio HAL. 6802 // Note that status is ignored by the caller for output device 6803 // (see AudioFlinger::setParameters() 6804 if (audio_is_output_devices(value)) { 6805 mOutDevice = value; 6806 status = BAD_VALUE; 6807 } else { 6808 mInDevice = value; 6809 if (value != AUDIO_DEVICE_NONE) { 6810 mPrevInDevice = value; 6811 } 6812 // disable AEC and NS if the device is a BT SCO headset supporting those 6813 // pre processings 6814 if (mTracks.size() > 0) { 6815 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 6816 mAudioFlinger->btNrecIsOff(); 6817 for (size_t i = 0; i < mTracks.size(); i++) { 6818 sp<RecordTrack> track = mTracks[i]; 6819 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId()); 6820 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId()); 6821 } 6822 } 6823 } 6824 } 6825 if (param.getInt(String8(AudioParameter::keyInputSource), value) == NO_ERROR && 6826 mAudioSource != (audio_source_t)value) { 6827 // forward device change to effects that have requested to be 6828 // aware of attached audio device. 6829 for (size_t i = 0; i < mEffectChains.size(); i++) { 6830 mEffectChains[i]->setAudioSource_l((audio_source_t)value); 6831 } 6832 mAudioSource = (audio_source_t)value; 6833 } 6834 6835 if (status == NO_ERROR) { 6836 status = mInput->stream->common.set_parameters(&mInput->stream->common, 6837 keyValuePair.string()); 6838 if (status == INVALID_OPERATION) { 6839 inputStandBy(); 6840 status = mInput->stream->common.set_parameters(&mInput->stream->common, 6841 keyValuePair.string()); 6842 } 6843 if (reconfig) { 6844 if (status == BAD_VALUE && 6845 audio_is_linear_pcm(mInput->stream->common.get_format(&mInput->stream->common)) && 6846 audio_is_linear_pcm(reqFormat) && 6847 (mInput->stream->common.get_sample_rate(&mInput->stream->common) 6848 <= (AUDIO_RESAMPLER_DOWN_RATIO_MAX * samplingRate)) && 6849 audio_channel_count_from_in_mask( 6850 mInput->stream->common.get_channels(&mInput->stream->common)) <= FCC_8) { 6851 status = NO_ERROR; 6852 } 6853 if (status == NO_ERROR) { 6854 readInputParameters_l(); 6855 sendIoConfigEvent_l(AUDIO_INPUT_CONFIG_CHANGED); 6856 } 6857 } 6858 } 6859 6860 return reconfig; 6861} 6862 6863String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 6864{ 6865 Mutex::Autolock _l(mLock); 6866 if (initCheck() != NO_ERROR) { 6867 return String8(); 6868 } 6869 6870 char *s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 6871 const String8 out_s8(s); 6872 free(s); 6873 return out_s8; 6874} 6875 6876void AudioFlinger::RecordThread::ioConfigChanged(audio_io_config_event event, pid_t pid) { 6877 sp<AudioIoDescriptor> desc = new AudioIoDescriptor(); 6878 6879 desc->mIoHandle = mId; 6880 6881 switch (event) { 6882 case AUDIO_INPUT_OPENED: 6883 case AUDIO_INPUT_CONFIG_CHANGED: 6884 desc->mPatch = mPatch; 6885 desc->mChannelMask = mChannelMask; 6886 desc->mSamplingRate = mSampleRate; 6887 desc->mFormat = mFormat; 6888 desc->mFrameCount = mFrameCount; 6889 desc->mLatency = 0; 6890 break; 6891 6892 case AUDIO_INPUT_CLOSED: 6893 default: 6894 break; 6895 } 6896 mAudioFlinger->ioConfigChanged(event, desc, pid); 6897} 6898 6899void AudioFlinger::RecordThread::readInputParameters_l() 6900{ 6901 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 6902 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 6903 mChannelCount = audio_channel_count_from_in_mask(mChannelMask); 6904 if (mChannelCount > FCC_8) { 6905 ALOGE("HAL channel count %d > %d", mChannelCount, FCC_8); 6906 } 6907 mHALFormat = mInput->stream->common.get_format(&mInput->stream->common); 6908 mFormat = mHALFormat; 6909 if (!audio_is_linear_pcm(mFormat)) { 6910 ALOGE("HAL format %#x is not linear pcm", mFormat); 6911 } 6912 mFrameSize = audio_stream_in_frame_size(mInput->stream); 6913 mBufferSize = mInput->stream->common.get_buffer_size(&mInput->stream->common); 6914 mFrameCount = mBufferSize / mFrameSize; 6915 // This is the formula for calculating the temporary buffer size. 6916 // With 7 HAL buffers, we can guarantee ability to down-sample the input by ratio of 6:1 to 6917 // 1 full output buffer, regardless of the alignment of the available input. 6918 // The value is somewhat arbitrary, and could probably be even larger. 6919 // A larger value should allow more old data to be read after a track calls start(), 6920 // without increasing latency. 6921 // 6922 // Note this is independent of the maximum downsampling ratio permitted for capture. 6923 mRsmpInFrames = mFrameCount * 7; 6924 mRsmpInFramesP2 = roundup(mRsmpInFrames); 6925 free(mRsmpInBuffer); 6926 6927 // TODO optimize audio capture buffer sizes ... 6928 // Here we calculate the size of the sliding buffer used as a source 6929 // for resampling. mRsmpInFramesP2 is currently roundup(mFrameCount * 7). 6930 // For current HAL frame counts, this is usually 2048 = 40 ms. It would 6931 // be better to have it derived from the pipe depth in the long term. 6932 // The current value is higher than necessary. However it should not add to latency. 6933 6934 // Over-allocate beyond mRsmpInFramesP2 to permit a HAL read past end of buffer 6935 (void)posix_memalign(&mRsmpInBuffer, 32, (mRsmpInFramesP2 + mFrameCount - 1) * mFrameSize); 6936 6937 // AudioRecord mSampleRate and mChannelCount are constant due to AudioRecord API constraints. 6938 // But if thread's mSampleRate or mChannelCount changes, how will that affect active tracks? 6939} 6940 6941uint32_t AudioFlinger::RecordThread::getInputFramesLost() 6942{ 6943 Mutex::Autolock _l(mLock); 6944 if (initCheck() != NO_ERROR) { 6945 return 0; 6946 } 6947 6948 return mInput->stream->get_input_frames_lost(mInput->stream); 6949} 6950 6951uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) const 6952{ 6953 Mutex::Autolock _l(mLock); 6954 uint32_t result = 0; 6955 if (getEffectChain_l(sessionId) != 0) { 6956 result = EFFECT_SESSION; 6957 } 6958 6959 for (size_t i = 0; i < mTracks.size(); ++i) { 6960 if (sessionId == mTracks[i]->sessionId()) { 6961 result |= TRACK_SESSION; 6962 break; 6963 } 6964 } 6965 6966 return result; 6967} 6968 6969KeyedVector<int, bool> AudioFlinger::RecordThread::sessionIds() const 6970{ 6971 KeyedVector<int, bool> ids; 6972 Mutex::Autolock _l(mLock); 6973 for (size_t j = 0; j < mTracks.size(); ++j) { 6974 sp<RecordThread::RecordTrack> track = mTracks[j]; 6975 int sessionId = track->sessionId(); 6976 if (ids.indexOfKey(sessionId) < 0) { 6977 ids.add(sessionId, true); 6978 } 6979 } 6980 return ids; 6981} 6982 6983AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 6984{ 6985 Mutex::Autolock _l(mLock); 6986 AudioStreamIn *input = mInput; 6987 mInput = NULL; 6988 return input; 6989} 6990 6991// this method must always be called either with ThreadBase mLock held or inside the thread loop 6992audio_stream_t* AudioFlinger::RecordThread::stream() const 6993{ 6994 if (mInput == NULL) { 6995 return NULL; 6996 } 6997 return &mInput->stream->common; 6998} 6999 7000status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 7001{ 7002 // only one chain per input thread 7003 if (mEffectChains.size() != 0) { 7004 ALOGW("addEffectChain_l() already one chain %p on thread %p", chain.get(), this); 7005 return INVALID_OPERATION; 7006 } 7007 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 7008 chain->setThread(this); 7009 chain->setInBuffer(NULL); 7010 chain->setOutBuffer(NULL); 7011 7012 checkSuspendOnAddEffectChain_l(chain); 7013 7014 // make sure enabled pre processing effects state is communicated to the HAL as we 7015 // just moved them to a new input stream. 7016 chain->syncHalEffectsState(); 7017 7018 mEffectChains.add(chain); 7019 7020 return NO_ERROR; 7021} 7022 7023size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 7024{ 7025 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 7026 ALOGW_IF(mEffectChains.size() != 1, 7027 "removeEffectChain_l() %p invalid chain size %d on thread %p", 7028 chain.get(), mEffectChains.size(), this); 7029 if (mEffectChains.size() == 1) { 7030 mEffectChains.removeAt(0); 7031 } 7032 return 0; 7033} 7034 7035status_t AudioFlinger::RecordThread::createAudioPatch_l(const struct audio_patch *patch, 7036 audio_patch_handle_t *handle) 7037{ 7038 status_t status = NO_ERROR; 7039 7040 // store new device and send to effects 7041 mInDevice = patch->sources[0].ext.device.type; 7042 mPatch = *patch; 7043 for (size_t i = 0; i < mEffectChains.size(); i++) { 7044 mEffectChains[i]->setDevice_l(mInDevice); 7045 } 7046 7047 // disable AEC and NS if the device is a BT SCO headset supporting those 7048 // pre processings 7049 if (mTracks.size() > 0) { 7050 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 7051 mAudioFlinger->btNrecIsOff(); 7052 for (size_t i = 0; i < mTracks.size(); i++) { 7053 sp<RecordTrack> track = mTracks[i]; 7054 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId()); 7055 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId()); 7056 } 7057 } 7058 7059 // store new source and send to effects 7060 if (mAudioSource != patch->sinks[0].ext.mix.usecase.source) { 7061 mAudioSource = patch->sinks[0].ext.mix.usecase.source; 7062 for (size_t i = 0; i < mEffectChains.size(); i++) { 7063 mEffectChains[i]->setAudioSource_l(mAudioSource); 7064 } 7065 } 7066 7067 if (mInput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 7068 audio_hw_device_t *hwDevice = mInput->audioHwDev->hwDevice(); 7069 status = hwDevice->create_audio_patch(hwDevice, 7070 patch->num_sources, 7071 patch->sources, 7072 patch->num_sinks, 7073 patch->sinks, 7074 handle); 7075 } else { 7076 char *address; 7077 if (strcmp(patch->sources[0].ext.device.address, "") != 0) { 7078 address = audio_device_address_to_parameter( 7079 patch->sources[0].ext.device.type, 7080 patch->sources[0].ext.device.address); 7081 } else { 7082 address = (char *)calloc(1, 1); 7083 } 7084 AudioParameter param = AudioParameter(String8(address)); 7085 free(address); 7086 param.addInt(String8(AUDIO_PARAMETER_STREAM_ROUTING), 7087 (int)patch->sources[0].ext.device.type); 7088 param.addInt(String8(AUDIO_PARAMETER_STREAM_INPUT_SOURCE), 7089 (int)patch->sinks[0].ext.mix.usecase.source); 7090 status = mInput->stream->common.set_parameters(&mInput->stream->common, 7091 param.toString().string()); 7092 *handle = AUDIO_PATCH_HANDLE_NONE; 7093 } 7094 7095 if (mInDevice != mPrevInDevice) { 7096 sendIoConfigEvent_l(AUDIO_INPUT_CONFIG_CHANGED); 7097 mPrevInDevice = mInDevice; 7098 } 7099 7100 return status; 7101} 7102 7103status_t AudioFlinger::RecordThread::releaseAudioPatch_l(const audio_patch_handle_t handle) 7104{ 7105 status_t status = NO_ERROR; 7106 7107 mInDevice = AUDIO_DEVICE_NONE; 7108 7109 if (mInput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 7110 audio_hw_device_t *hwDevice = mInput->audioHwDev->hwDevice(); 7111 status = hwDevice->release_audio_patch(hwDevice, handle); 7112 } else { 7113 AudioParameter param; 7114 param.addInt(String8(AUDIO_PARAMETER_STREAM_ROUTING), 0); 7115 status = mInput->stream->common.set_parameters(&mInput->stream->common, 7116 param.toString().string()); 7117 } 7118 return status; 7119} 7120 7121void AudioFlinger::RecordThread::addPatchRecord(const sp<PatchRecord>& record) 7122{ 7123 Mutex::Autolock _l(mLock); 7124 mTracks.add(record); 7125} 7126 7127void AudioFlinger::RecordThread::deletePatchRecord(const sp<PatchRecord>& record) 7128{ 7129 Mutex::Autolock _l(mLock); 7130 destroyTrack_l(record); 7131} 7132 7133void AudioFlinger::RecordThread::getAudioPortConfig(struct audio_port_config *config) 7134{ 7135 ThreadBase::getAudioPortConfig(config); 7136 config->role = AUDIO_PORT_ROLE_SINK; 7137 config->ext.mix.hw_module = mInput->audioHwDev->handle(); 7138 config->ext.mix.usecase.source = mAudioSource; 7139} 7140 7141} // namespace android 7142