Threads.cpp revision 3f273d10817ddb2f792ae043de692efcdf1988ae
1/* 2** 3** Copyright 2012, The Android Open Source Project 4** 5** Licensed under the Apache License, Version 2.0 (the "License"); 6** you may not use this file except in compliance with the License. 7** You may obtain a copy of the License at 8** 9** http://www.apache.org/licenses/LICENSE-2.0 10** 11** Unless required by applicable law or agreed to in writing, software 12** distributed under the License is distributed on an "AS IS" BASIS, 13** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. 14** See the License for the specific language governing permissions and 15** limitations under the License. 16*/ 17 18 19#define LOG_TAG "AudioFlinger" 20//#define LOG_NDEBUG 0 21#define ATRACE_TAG ATRACE_TAG_AUDIO 22 23#include "Configuration.h" 24#include <math.h> 25#include <fcntl.h> 26#include <linux/futex.h> 27#include <sys/stat.h> 28#include <sys/syscall.h> 29#include <cutils/properties.h> 30#include <media/AudioParameter.h> 31#include <media/AudioResamplerPublic.h> 32#include <utils/Log.h> 33#include <utils/Trace.h> 34 35#include <private/media/AudioTrackShared.h> 36#include <hardware/audio.h> 37#include <audio_effects/effect_ns.h> 38#include <audio_effects/effect_aec.h> 39#include <audio_utils/primitives.h> 40#include <audio_utils/format.h> 41#include <audio_utils/minifloat.h> 42 43// NBAIO implementations 44#include <media/nbaio/AudioStreamInSource.h> 45#include <media/nbaio/AudioStreamOutSink.h> 46#include <media/nbaio/MonoPipe.h> 47#include <media/nbaio/MonoPipeReader.h> 48#include <media/nbaio/Pipe.h> 49#include <media/nbaio/PipeReader.h> 50#include <media/nbaio/SourceAudioBufferProvider.h> 51#include <mediautils/BatteryNotifier.h> 52 53#include <powermanager/PowerManager.h> 54 55#include <common_time/cc_helper.h> 56#include <common_time/local_clock.h> 57 58#include "AudioFlinger.h" 59#include "AudioMixer.h" 60#include "BufferProviders.h" 61#include "FastMixer.h" 62#include "FastCapture.h" 63#include "ServiceUtilities.h" 64#include "mediautils/SchedulingPolicyService.h" 65 66#ifdef ADD_BATTERY_DATA 67#include <media/IMediaPlayerService.h> 68#include <media/IMediaDeathNotifier.h> 69#endif 70 71#ifdef DEBUG_CPU_USAGE 72#include <cpustats/CentralTendencyStatistics.h> 73#include <cpustats/ThreadCpuUsage.h> 74#endif 75 76// ---------------------------------------------------------------------------- 77 78// Note: the following macro is used for extremely verbose logging message. In 79// order to run with ALOG_ASSERT turned on, we need to have LOG_NDEBUG set to 80// 0; but one side effect of this is to turn all LOGV's as well. Some messages 81// are so verbose that we want to suppress them even when we have ALOG_ASSERT 82// turned on. Do not uncomment the #def below unless you really know what you 83// are doing and want to see all of the extremely verbose messages. 84//#define VERY_VERY_VERBOSE_LOGGING 85#ifdef VERY_VERY_VERBOSE_LOGGING 86#define ALOGVV ALOGV 87#else 88#define ALOGVV(a...) do { } while(0) 89#endif 90 91// TODO: Move these macro/inlines to a header file. 92#define max(a, b) ((a) > (b) ? (a) : (b)) 93template <typename T> 94static inline T min(const T& a, const T& b) 95{ 96 return a < b ? a : b; 97} 98 99#ifndef ARRAY_SIZE 100#define ARRAY_SIZE(a) (sizeof(a) / sizeof(a[0])) 101#endif 102 103namespace android { 104 105// retry counts for buffer fill timeout 106// 50 * ~20msecs = 1 second 107static const int8_t kMaxTrackRetries = 50; 108static const int8_t kMaxTrackStartupRetries = 50; 109// allow less retry attempts on direct output thread. 110// direct outputs can be a scarce resource in audio hardware and should 111// be released as quickly as possible. 112static const int8_t kMaxTrackRetriesDirect = 2; 113 114// don't warn about blocked writes or record buffer overflows more often than this 115static const nsecs_t kWarningThrottleNs = seconds(5); 116 117// RecordThread loop sleep time upon application overrun or audio HAL read error 118static const int kRecordThreadSleepUs = 5000; 119 120// maximum time to wait in sendConfigEvent_l() for a status to be received 121static const nsecs_t kConfigEventTimeoutNs = seconds(2); 122 123// minimum sleep time for the mixer thread loop when tracks are active but in underrun 124static const uint32_t kMinThreadSleepTimeUs = 5000; 125// maximum divider applied to the active sleep time in the mixer thread loop 126static const uint32_t kMaxThreadSleepTimeShift = 2; 127 128// minimum normal sink buffer size, expressed in milliseconds rather than frames 129// FIXME This should be based on experimentally observed scheduling jitter 130static const uint32_t kMinNormalSinkBufferSizeMs = 20; 131// maximum normal sink buffer size 132static const uint32_t kMaxNormalSinkBufferSizeMs = 24; 133 134// minimum capture buffer size in milliseconds to _not_ need a fast capture thread 135// FIXME This should be based on experimentally observed scheduling jitter 136static const uint32_t kMinNormalCaptureBufferSizeMs = 12; 137 138// Offloaded output thread standby delay: allows track transition without going to standby 139static const nsecs_t kOffloadStandbyDelayNs = seconds(1); 140 141// Whether to use fast mixer 142static const enum { 143 FastMixer_Never, // never initialize or use: for debugging only 144 FastMixer_Always, // always initialize and use, even if not needed: for debugging only 145 // normal mixer multiplier is 1 146 FastMixer_Static, // initialize if needed, then use all the time if initialized, 147 // multiplier is calculated based on min & max normal mixer buffer size 148 FastMixer_Dynamic, // initialize if needed, then use dynamically depending on track load, 149 // multiplier is calculated based on min & max normal mixer buffer size 150 // FIXME for FastMixer_Dynamic: 151 // Supporting this option will require fixing HALs that can't handle large writes. 152 // For example, one HAL implementation returns an error from a large write, 153 // and another HAL implementation corrupts memory, possibly in the sample rate converter. 154 // We could either fix the HAL implementations, or provide a wrapper that breaks 155 // up large writes into smaller ones, and the wrapper would need to deal with scheduler. 156} kUseFastMixer = FastMixer_Static; 157 158// Whether to use fast capture 159static const enum { 160 FastCapture_Never, // never initialize or use: for debugging only 161 FastCapture_Always, // always initialize and use, even if not needed: for debugging only 162 FastCapture_Static, // initialize if needed, then use all the time if initialized 163} kUseFastCapture = FastCapture_Static; 164 165// Priorities for requestPriority 166static const int kPriorityAudioApp = 2; 167static const int kPriorityFastMixer = 3; 168static const int kPriorityFastCapture = 3; 169 170// IAudioFlinger::createTrack() reports back to client the total size of shared memory area 171// for the track. The client then sub-divides this into smaller buffers for its use. 172// Currently the client uses N-buffering by default, but doesn't tell us about the value of N. 173// So for now we just assume that client is double-buffered for fast tracks. 174// FIXME It would be better for client to tell AudioFlinger the value of N, 175// so AudioFlinger could allocate the right amount of memory. 176// See the client's minBufCount and mNotificationFramesAct calculations for details. 177 178// This is the default value, if not specified by property. 179static const int kFastTrackMultiplier = 2; 180 181// The minimum and maximum allowed values 182static const int kFastTrackMultiplierMin = 1; 183static const int kFastTrackMultiplierMax = 2; 184 185// The actual value to use, which can be specified per-device via property af.fast_track_multiplier. 186static int sFastTrackMultiplier = kFastTrackMultiplier; 187 188// See Thread::readOnlyHeap(). 189// Initially this heap is used to allocate client buffers for "fast" AudioRecord. 190// Eventually it will be the single buffer that FastCapture writes into via HAL read(), 191// and that all "fast" AudioRecord clients read from. In either case, the size can be small. 192static const size_t kRecordThreadReadOnlyHeapSize = 0x2000; 193 194// ---------------------------------------------------------------------------- 195 196static pthread_once_t sFastTrackMultiplierOnce = PTHREAD_ONCE_INIT; 197 198static void sFastTrackMultiplierInit() 199{ 200 char value[PROPERTY_VALUE_MAX]; 201 if (property_get("af.fast_track_multiplier", value, NULL) > 0) { 202 char *endptr; 203 unsigned long ul = strtoul(value, &endptr, 0); 204 if (*endptr == '\0' && kFastTrackMultiplierMin <= ul && ul <= kFastTrackMultiplierMax) { 205 sFastTrackMultiplier = (int) ul; 206 } 207 } 208} 209 210// ---------------------------------------------------------------------------- 211 212#ifdef ADD_BATTERY_DATA 213// To collect the amplifier usage 214static void addBatteryData(uint32_t params) { 215 sp<IMediaPlayerService> service = IMediaDeathNotifier::getMediaPlayerService(); 216 if (service == NULL) { 217 // it already logged 218 return; 219 } 220 221 service->addBatteryData(params); 222} 223#endif 224 225 226// ---------------------------------------------------------------------------- 227// CPU Stats 228// ---------------------------------------------------------------------------- 229 230class CpuStats { 231public: 232 CpuStats(); 233 void sample(const String8 &title); 234#ifdef DEBUG_CPU_USAGE 235private: 236 ThreadCpuUsage mCpuUsage; // instantaneous thread CPU usage in wall clock ns 237 CentralTendencyStatistics mWcStats; // statistics on thread CPU usage in wall clock ns 238 239 CentralTendencyStatistics mHzStats; // statistics on thread CPU usage in cycles 240 241 int mCpuNum; // thread's current CPU number 242 int mCpukHz; // frequency of thread's current CPU in kHz 243#endif 244}; 245 246CpuStats::CpuStats() 247#ifdef DEBUG_CPU_USAGE 248 : mCpuNum(-1), mCpukHz(-1) 249#endif 250{ 251} 252 253void CpuStats::sample(const String8 &title 254#ifndef DEBUG_CPU_USAGE 255 __unused 256#endif 257 ) { 258#ifdef DEBUG_CPU_USAGE 259 // get current thread's delta CPU time in wall clock ns 260 double wcNs; 261 bool valid = mCpuUsage.sampleAndEnable(wcNs); 262 263 // record sample for wall clock statistics 264 if (valid) { 265 mWcStats.sample(wcNs); 266 } 267 268 // get the current CPU number 269 int cpuNum = sched_getcpu(); 270 271 // get the current CPU frequency in kHz 272 int cpukHz = mCpuUsage.getCpukHz(cpuNum); 273 274 // check if either CPU number or frequency changed 275 if (cpuNum != mCpuNum || cpukHz != mCpukHz) { 276 mCpuNum = cpuNum; 277 mCpukHz = cpukHz; 278 // ignore sample for purposes of cycles 279 valid = false; 280 } 281 282 // if no change in CPU number or frequency, then record sample for cycle statistics 283 if (valid && mCpukHz > 0) { 284 double cycles = wcNs * cpukHz * 0.000001; 285 mHzStats.sample(cycles); 286 } 287 288 unsigned n = mWcStats.n(); 289 // mCpuUsage.elapsed() is expensive, so don't call it every loop 290 if ((n & 127) == 1) { 291 long long elapsed = mCpuUsage.elapsed(); 292 if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) { 293 double perLoop = elapsed / (double) n; 294 double perLoop100 = perLoop * 0.01; 295 double perLoop1k = perLoop * 0.001; 296 double mean = mWcStats.mean(); 297 double stddev = mWcStats.stddev(); 298 double minimum = mWcStats.minimum(); 299 double maximum = mWcStats.maximum(); 300 double meanCycles = mHzStats.mean(); 301 double stddevCycles = mHzStats.stddev(); 302 double minCycles = mHzStats.minimum(); 303 double maxCycles = mHzStats.maximum(); 304 mCpuUsage.resetElapsed(); 305 mWcStats.reset(); 306 mHzStats.reset(); 307 ALOGD("CPU usage for %s over past %.1f secs\n" 308 " (%u mixer loops at %.1f mean ms per loop):\n" 309 " us per mix loop: mean=%.0f stddev=%.0f min=%.0f max=%.0f\n" 310 " %% of wall: mean=%.1f stddev=%.1f min=%.1f max=%.1f\n" 311 " MHz: mean=%.1f, stddev=%.1f, min=%.1f max=%.1f", 312 title.string(), 313 elapsed * .000000001, n, perLoop * .000001, 314 mean * .001, 315 stddev * .001, 316 minimum * .001, 317 maximum * .001, 318 mean / perLoop100, 319 stddev / perLoop100, 320 minimum / perLoop100, 321 maximum / perLoop100, 322 meanCycles / perLoop1k, 323 stddevCycles / perLoop1k, 324 minCycles / perLoop1k, 325 maxCycles / perLoop1k); 326 327 } 328 } 329#endif 330}; 331 332// ---------------------------------------------------------------------------- 333// ThreadBase 334// ---------------------------------------------------------------------------- 335 336// static 337const char *AudioFlinger::ThreadBase::threadTypeToString(AudioFlinger::ThreadBase::type_t type) 338{ 339 switch (type) { 340 case MIXER: 341 return "MIXER"; 342 case DIRECT: 343 return "DIRECT"; 344 case DUPLICATING: 345 return "DUPLICATING"; 346 case RECORD: 347 return "RECORD"; 348 case OFFLOAD: 349 return "OFFLOAD"; 350 default: 351 return "unknown"; 352 } 353} 354 355String8 devicesToString(audio_devices_t devices) 356{ 357 static const struct mapping { 358 audio_devices_t mDevices; 359 const char * mString; 360 } mappingsOut[] = { 361 AUDIO_DEVICE_OUT_EARPIECE, "EARPIECE", 362 AUDIO_DEVICE_OUT_SPEAKER, "SPEAKER", 363 AUDIO_DEVICE_OUT_WIRED_HEADSET, "WIRED_HEADSET", 364 AUDIO_DEVICE_OUT_WIRED_HEADPHONE, "WIRED_HEADPHONE", 365 AUDIO_DEVICE_OUT_BLUETOOTH_SCO, "BLUETOOTH_SCO", 366 AUDIO_DEVICE_OUT_BLUETOOTH_SCO_HEADSET, "BLUETOOTH_SCO_HEADSET", 367 AUDIO_DEVICE_OUT_BLUETOOTH_SCO_CARKIT, "BLUETOOTH_SCO_CARKIT", 368 AUDIO_DEVICE_OUT_BLUETOOTH_A2DP, "BLUETOOTH_A2DP", 369 AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES, "BLUETOOTH_A2DP_HEADPHONES", 370 AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER, "BLUETOOTH_A2DP_SPEAKER", 371 AUDIO_DEVICE_OUT_AUX_DIGITAL, "AUX_DIGITAL", 372 AUDIO_DEVICE_OUT_HDMI, "HDMI", 373 AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET, "ANLG_DOCK_HEADSET", 374 AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET, "DGTL_DOCK_HEADSET", 375 AUDIO_DEVICE_OUT_USB_ACCESSORY, "USB_ACCESSORY", 376 AUDIO_DEVICE_OUT_USB_DEVICE, "USB_DEVICE", 377 AUDIO_DEVICE_OUT_TELEPHONY_TX, "TELEPHONY_TX", 378 AUDIO_DEVICE_OUT_LINE, "LINE", 379 AUDIO_DEVICE_OUT_HDMI_ARC, "HDMI_ARC", 380 AUDIO_DEVICE_OUT_SPDIF, "SPDIF", 381 AUDIO_DEVICE_OUT_FM, "FM", 382 AUDIO_DEVICE_OUT_AUX_LINE, "AUX_LINE", 383 AUDIO_DEVICE_OUT_SPEAKER_SAFE, "SPEAKER_SAFE", 384 AUDIO_DEVICE_OUT_IP, "IP", 385 AUDIO_DEVICE_NONE, "NONE", // must be last 386 }, mappingsIn[] = { 387 AUDIO_DEVICE_IN_COMMUNICATION, "COMMUNICATION", 388 AUDIO_DEVICE_IN_AMBIENT, "AMBIENT", 389 AUDIO_DEVICE_IN_BUILTIN_MIC, "BUILTIN_MIC", 390 AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET, "BLUETOOTH_SCO_HEADSET", 391 AUDIO_DEVICE_IN_WIRED_HEADSET, "WIRED_HEADSET", 392 AUDIO_DEVICE_IN_AUX_DIGITAL, "AUX_DIGITAL", 393 AUDIO_DEVICE_IN_VOICE_CALL, "VOICE_CALL", 394 AUDIO_DEVICE_IN_TELEPHONY_RX, "TELEPHONY_RX", 395 AUDIO_DEVICE_IN_BACK_MIC, "BACK_MIC", 396 AUDIO_DEVICE_IN_REMOTE_SUBMIX, "REMOTE_SUBMIX", 397 AUDIO_DEVICE_IN_ANLG_DOCK_HEADSET, "ANLG_DOCK_HEADSET", 398 AUDIO_DEVICE_IN_DGTL_DOCK_HEADSET, "DGTL_DOCK_HEADSET", 399 AUDIO_DEVICE_IN_USB_ACCESSORY, "USB_ACCESSORY", 400 AUDIO_DEVICE_IN_USB_DEVICE, "USB_DEVICE", 401 AUDIO_DEVICE_IN_FM_TUNER, "FM_TUNER", 402 AUDIO_DEVICE_IN_TV_TUNER, "TV_TUNER", 403 AUDIO_DEVICE_IN_LINE, "LINE", 404 AUDIO_DEVICE_IN_SPDIF, "SPDIF", 405 AUDIO_DEVICE_IN_BLUETOOTH_A2DP, "BLUETOOTH_A2DP", 406 AUDIO_DEVICE_IN_LOOPBACK, "LOOPBACK", 407 AUDIO_DEVICE_IN_IP, "IP", 408 AUDIO_DEVICE_NONE, "NONE", // must be last 409 }; 410 String8 result; 411 audio_devices_t allDevices = AUDIO_DEVICE_NONE; 412 const mapping *entry; 413 if (devices & AUDIO_DEVICE_BIT_IN) { 414 devices &= ~AUDIO_DEVICE_BIT_IN; 415 entry = mappingsIn; 416 } else { 417 entry = mappingsOut; 418 } 419 for ( ; entry->mDevices != AUDIO_DEVICE_NONE; entry++) { 420 allDevices = (audio_devices_t) (allDevices | entry->mDevices); 421 if (devices & entry->mDevices) { 422 if (!result.isEmpty()) { 423 result.append("|"); 424 } 425 result.append(entry->mString); 426 } 427 } 428 if (devices & ~allDevices) { 429 if (!result.isEmpty()) { 430 result.append("|"); 431 } 432 result.appendFormat("0x%X", devices & ~allDevices); 433 } 434 if (result.isEmpty()) { 435 result.append(entry->mString); 436 } 437 return result; 438} 439 440String8 inputFlagsToString(audio_input_flags_t flags) 441{ 442 static const struct mapping { 443 audio_input_flags_t mFlag; 444 const char * mString; 445 } mappings[] = { 446 AUDIO_INPUT_FLAG_FAST, "FAST", 447 AUDIO_INPUT_FLAG_HW_HOTWORD, "HW_HOTWORD", 448 AUDIO_INPUT_FLAG_RAW, "RAW", 449 AUDIO_INPUT_FLAG_SYNC, "SYNC", 450 AUDIO_INPUT_FLAG_NONE, "NONE", // must be last 451 }; 452 String8 result; 453 audio_input_flags_t allFlags = AUDIO_INPUT_FLAG_NONE; 454 const mapping *entry; 455 for (entry = mappings; entry->mFlag != AUDIO_INPUT_FLAG_NONE; entry++) { 456 allFlags = (audio_input_flags_t) (allFlags | entry->mFlag); 457 if (flags & entry->mFlag) { 458 if (!result.isEmpty()) { 459 result.append("|"); 460 } 461 result.append(entry->mString); 462 } 463 } 464 if (flags & ~allFlags) { 465 if (!result.isEmpty()) { 466 result.append("|"); 467 } 468 result.appendFormat("0x%X", flags & ~allFlags); 469 } 470 if (result.isEmpty()) { 471 result.append(entry->mString); 472 } 473 return result; 474} 475 476String8 outputFlagsToString(audio_output_flags_t flags) 477{ 478 static const struct mapping { 479 audio_output_flags_t mFlag; 480 const char * mString; 481 } mappings[] = { 482 AUDIO_OUTPUT_FLAG_DIRECT, "DIRECT", 483 AUDIO_OUTPUT_FLAG_PRIMARY, "PRIMARY", 484 AUDIO_OUTPUT_FLAG_FAST, "FAST", 485 AUDIO_OUTPUT_FLAG_DEEP_BUFFER, "DEEP_BUFFER", 486 AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD, "COMPRESS_OFFLOAD", 487 AUDIO_OUTPUT_FLAG_NON_BLOCKING, "NON_BLOCKING", 488 AUDIO_OUTPUT_FLAG_HW_AV_SYNC, "HW_AV_SYNC", 489 AUDIO_OUTPUT_FLAG_RAW, "RAW", 490 AUDIO_OUTPUT_FLAG_SYNC, "SYNC", 491 AUDIO_OUTPUT_FLAG_IEC958_NONAUDIO, "IEC958_NONAUDIO", 492 AUDIO_OUTPUT_FLAG_NONE, "NONE", // must be last 493 }; 494 String8 result; 495 audio_output_flags_t allFlags = AUDIO_OUTPUT_FLAG_NONE; 496 const mapping *entry; 497 for (entry = mappings; entry->mFlag != AUDIO_OUTPUT_FLAG_NONE; entry++) { 498 allFlags = (audio_output_flags_t) (allFlags | entry->mFlag); 499 if (flags & entry->mFlag) { 500 if (!result.isEmpty()) { 501 result.append("|"); 502 } 503 result.append(entry->mString); 504 } 505 } 506 if (flags & ~allFlags) { 507 if (!result.isEmpty()) { 508 result.append("|"); 509 } 510 result.appendFormat("0x%X", flags & ~allFlags); 511 } 512 if (result.isEmpty()) { 513 result.append(entry->mString); 514 } 515 return result; 516} 517 518const char *sourceToString(audio_source_t source) 519{ 520 switch (source) { 521 case AUDIO_SOURCE_DEFAULT: return "default"; 522 case AUDIO_SOURCE_MIC: return "mic"; 523 case AUDIO_SOURCE_VOICE_UPLINK: return "voice uplink"; 524 case AUDIO_SOURCE_VOICE_DOWNLINK: return "voice downlink"; 525 case AUDIO_SOURCE_VOICE_CALL: return "voice call"; 526 case AUDIO_SOURCE_CAMCORDER: return "camcorder"; 527 case AUDIO_SOURCE_VOICE_RECOGNITION: return "voice recognition"; 528 case AUDIO_SOURCE_VOICE_COMMUNICATION: return "voice communication"; 529 case AUDIO_SOURCE_REMOTE_SUBMIX: return "remote submix"; 530 case AUDIO_SOURCE_FM_TUNER: return "FM tuner"; 531 case AUDIO_SOURCE_HOTWORD: return "hotword"; 532 default: return "unknown"; 533 } 534} 535 536AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, 537 audio_devices_t outDevice, audio_devices_t inDevice, type_t type, bool systemReady) 538 : Thread(false /*canCallJava*/), 539 mType(type), 540 mAudioFlinger(audioFlinger), 541 // mSampleRate, mFrameCount, mChannelMask, mChannelCount, mFrameSize, mFormat, mBufferSize 542 // are set by PlaybackThread::readOutputParameters_l() or 543 // RecordThread::readInputParameters_l() 544 //FIXME: mStandby should be true here. Is this some kind of hack? 545 mStandby(false), mOutDevice(outDevice), mInDevice(inDevice), 546 mPrevOutDevice(AUDIO_DEVICE_NONE), mPrevInDevice(AUDIO_DEVICE_NONE), 547 mAudioSource(AUDIO_SOURCE_DEFAULT), mId(id), 548 // mName will be set by concrete (non-virtual) subclass 549 mDeathRecipient(new PMDeathRecipient(this)), 550 mSystemReady(systemReady), 551 mNotifiedBatteryStart(false) 552{ 553 memset(&mPatch, 0, sizeof(struct audio_patch)); 554} 555 556AudioFlinger::ThreadBase::~ThreadBase() 557{ 558 // mConfigEvents should be empty, but just in case it isn't, free the memory it owns 559 mConfigEvents.clear(); 560 561 // do not lock the mutex in destructor 562 releaseWakeLock_l(); 563 if (mPowerManager != 0) { 564 sp<IBinder> binder = IInterface::asBinder(mPowerManager); 565 binder->unlinkToDeath(mDeathRecipient); 566 } 567} 568 569status_t AudioFlinger::ThreadBase::readyToRun() 570{ 571 status_t status = initCheck(); 572 if (status == NO_ERROR) { 573 ALOGI("AudioFlinger's thread %p ready to run", this); 574 } else { 575 ALOGE("No working audio driver found."); 576 } 577 return status; 578} 579 580void AudioFlinger::ThreadBase::exit() 581{ 582 ALOGV("ThreadBase::exit"); 583 // do any cleanup required for exit to succeed 584 preExit(); 585 { 586 // This lock prevents the following race in thread (uniprocessor for illustration): 587 // if (!exitPending()) { 588 // // context switch from here to exit() 589 // // exit() calls requestExit(), what exitPending() observes 590 // // exit() calls signal(), which is dropped since no waiters 591 // // context switch back from exit() to here 592 // mWaitWorkCV.wait(...); 593 // // now thread is hung 594 // } 595 AutoMutex lock(mLock); 596 requestExit(); 597 mWaitWorkCV.broadcast(); 598 } 599 // When Thread::requestExitAndWait is made virtual and this method is renamed to 600 // "virtual status_t requestExitAndWait()", replace by "return Thread::requestExitAndWait();" 601 requestExitAndWait(); 602} 603 604status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs) 605{ 606 status_t status; 607 608 ALOGV("ThreadBase::setParameters() %s", keyValuePairs.string()); 609 Mutex::Autolock _l(mLock); 610 611 return sendSetParameterConfigEvent_l(keyValuePairs); 612} 613 614// sendConfigEvent_l() must be called with ThreadBase::mLock held 615// Can temporarily release the lock if waiting for a reply from processConfigEvents_l(). 616status_t AudioFlinger::ThreadBase::sendConfigEvent_l(sp<ConfigEvent>& event) 617{ 618 status_t status = NO_ERROR; 619 620 if (event->mRequiresSystemReady && !mSystemReady) { 621 event->mWaitStatus = false; 622 mPendingConfigEvents.add(event); 623 return status; 624 } 625 mConfigEvents.add(event); 626 ALOGV("sendConfigEvent_l() num events %d event %d", mConfigEvents.size(), event->mType); 627 mWaitWorkCV.signal(); 628 mLock.unlock(); 629 { 630 Mutex::Autolock _l(event->mLock); 631 while (event->mWaitStatus) { 632 if (event->mCond.waitRelative(event->mLock, kConfigEventTimeoutNs) != NO_ERROR) { 633 event->mStatus = TIMED_OUT; 634 event->mWaitStatus = false; 635 } 636 } 637 status = event->mStatus; 638 } 639 mLock.lock(); 640 return status; 641} 642 643void AudioFlinger::ThreadBase::sendIoConfigEvent(audio_io_config_event event, pid_t pid) 644{ 645 Mutex::Autolock _l(mLock); 646 sendIoConfigEvent_l(event, pid); 647} 648 649// sendIoConfigEvent_l() must be called with ThreadBase::mLock held 650void AudioFlinger::ThreadBase::sendIoConfigEvent_l(audio_io_config_event event, pid_t pid) 651{ 652 sp<ConfigEvent> configEvent = (ConfigEvent *)new IoConfigEvent(event, pid); 653 sendConfigEvent_l(configEvent); 654} 655 656void AudioFlinger::ThreadBase::sendPrioConfigEvent(pid_t pid, pid_t tid, int32_t prio) 657{ 658 Mutex::Autolock _l(mLock); 659 sendPrioConfigEvent_l(pid, tid, prio); 660} 661 662// sendPrioConfigEvent_l() must be called with ThreadBase::mLock held 663void AudioFlinger::ThreadBase::sendPrioConfigEvent_l(pid_t pid, pid_t tid, int32_t prio) 664{ 665 sp<ConfigEvent> configEvent = (ConfigEvent *)new PrioConfigEvent(pid, tid, prio); 666 sendConfigEvent_l(configEvent); 667} 668 669// sendSetParameterConfigEvent_l() must be called with ThreadBase::mLock held 670status_t AudioFlinger::ThreadBase::sendSetParameterConfigEvent_l(const String8& keyValuePair) 671{ 672 sp<ConfigEvent> configEvent = (ConfigEvent *)new SetParameterConfigEvent(keyValuePair); 673 return sendConfigEvent_l(configEvent); 674} 675 676status_t AudioFlinger::ThreadBase::sendCreateAudioPatchConfigEvent( 677 const struct audio_patch *patch, 678 audio_patch_handle_t *handle) 679{ 680 Mutex::Autolock _l(mLock); 681 sp<ConfigEvent> configEvent = (ConfigEvent *)new CreateAudioPatchConfigEvent(*patch, *handle); 682 status_t status = sendConfigEvent_l(configEvent); 683 if (status == NO_ERROR) { 684 CreateAudioPatchConfigEventData *data = 685 (CreateAudioPatchConfigEventData *)configEvent->mData.get(); 686 *handle = data->mHandle; 687 } 688 return status; 689} 690 691status_t AudioFlinger::ThreadBase::sendReleaseAudioPatchConfigEvent( 692 const audio_patch_handle_t handle) 693{ 694 Mutex::Autolock _l(mLock); 695 sp<ConfigEvent> configEvent = (ConfigEvent *)new ReleaseAudioPatchConfigEvent(handle); 696 return sendConfigEvent_l(configEvent); 697} 698 699 700// post condition: mConfigEvents.isEmpty() 701void AudioFlinger::ThreadBase::processConfigEvents_l() 702{ 703 bool configChanged = false; 704 705 while (!mConfigEvents.isEmpty()) { 706 ALOGV("processConfigEvents_l() remaining events %d", mConfigEvents.size()); 707 sp<ConfigEvent> event = mConfigEvents[0]; 708 mConfigEvents.removeAt(0); 709 switch (event->mType) { 710 case CFG_EVENT_PRIO: { 711 PrioConfigEventData *data = (PrioConfigEventData *)event->mData.get(); 712 // FIXME Need to understand why this has to be done asynchronously 713 int err = requestPriority(data->mPid, data->mTid, data->mPrio, 714 true /*asynchronous*/); 715 if (err != 0) { 716 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 717 data->mPrio, data->mPid, data->mTid, err); 718 } 719 } break; 720 case CFG_EVENT_IO: { 721 IoConfigEventData *data = (IoConfigEventData *)event->mData.get(); 722 ioConfigChanged(data->mEvent, data->mPid); 723 } break; 724 case CFG_EVENT_SET_PARAMETER: { 725 SetParameterConfigEventData *data = (SetParameterConfigEventData *)event->mData.get(); 726 if (checkForNewParameter_l(data->mKeyValuePairs, event->mStatus)) { 727 configChanged = true; 728 } 729 } break; 730 case CFG_EVENT_CREATE_AUDIO_PATCH: { 731 CreateAudioPatchConfigEventData *data = 732 (CreateAudioPatchConfigEventData *)event->mData.get(); 733 event->mStatus = createAudioPatch_l(&data->mPatch, &data->mHandle); 734 } break; 735 case CFG_EVENT_RELEASE_AUDIO_PATCH: { 736 ReleaseAudioPatchConfigEventData *data = 737 (ReleaseAudioPatchConfigEventData *)event->mData.get(); 738 event->mStatus = releaseAudioPatch_l(data->mHandle); 739 } break; 740 default: 741 ALOG_ASSERT(false, "processConfigEvents_l() unknown event type %d", event->mType); 742 break; 743 } 744 { 745 Mutex::Autolock _l(event->mLock); 746 if (event->mWaitStatus) { 747 event->mWaitStatus = false; 748 event->mCond.signal(); 749 } 750 } 751 ALOGV_IF(mConfigEvents.isEmpty(), "processConfigEvents_l() DONE thread %p", this); 752 } 753 754 if (configChanged) { 755 cacheParameters_l(); 756 } 757} 758 759String8 channelMaskToString(audio_channel_mask_t mask, bool output) { 760 String8 s; 761 const audio_channel_representation_t representation = 762 audio_channel_mask_get_representation(mask); 763 764 switch (representation) { 765 case AUDIO_CHANNEL_REPRESENTATION_POSITION: { 766 if (output) { 767 if (mask & AUDIO_CHANNEL_OUT_FRONT_LEFT) s.append("front-left, "); 768 if (mask & AUDIO_CHANNEL_OUT_FRONT_RIGHT) s.append("front-right, "); 769 if (mask & AUDIO_CHANNEL_OUT_FRONT_CENTER) s.append("front-center, "); 770 if (mask & AUDIO_CHANNEL_OUT_LOW_FREQUENCY) s.append("low freq, "); 771 if (mask & AUDIO_CHANNEL_OUT_BACK_LEFT) s.append("back-left, "); 772 if (mask & AUDIO_CHANNEL_OUT_BACK_RIGHT) s.append("back-right, "); 773 if (mask & AUDIO_CHANNEL_OUT_FRONT_LEFT_OF_CENTER) s.append("front-left-of-center, "); 774 if (mask & AUDIO_CHANNEL_OUT_FRONT_RIGHT_OF_CENTER) s.append("front-right-of-center, "); 775 if (mask & AUDIO_CHANNEL_OUT_BACK_CENTER) s.append("back-center, "); 776 if (mask & AUDIO_CHANNEL_OUT_SIDE_LEFT) s.append("side-left, "); 777 if (mask & AUDIO_CHANNEL_OUT_SIDE_RIGHT) s.append("side-right, "); 778 if (mask & AUDIO_CHANNEL_OUT_TOP_CENTER) s.append("top-center ,"); 779 if (mask & AUDIO_CHANNEL_OUT_TOP_FRONT_LEFT) s.append("top-front-left, "); 780 if (mask & AUDIO_CHANNEL_OUT_TOP_FRONT_CENTER) s.append("top-front-center, "); 781 if (mask & AUDIO_CHANNEL_OUT_TOP_FRONT_RIGHT) s.append("top-front-right, "); 782 if (mask & AUDIO_CHANNEL_OUT_TOP_BACK_LEFT) s.append("top-back-left, "); 783 if (mask & AUDIO_CHANNEL_OUT_TOP_BACK_CENTER) s.append("top-back-center, " ); 784 if (mask & AUDIO_CHANNEL_OUT_TOP_BACK_RIGHT) s.append("top-back-right, " ); 785 if (mask & ~AUDIO_CHANNEL_OUT_ALL) s.append("unknown, "); 786 } else { 787 if (mask & AUDIO_CHANNEL_IN_LEFT) s.append("left, "); 788 if (mask & AUDIO_CHANNEL_IN_RIGHT) s.append("right, "); 789 if (mask & AUDIO_CHANNEL_IN_FRONT) s.append("front, "); 790 if (mask & AUDIO_CHANNEL_IN_BACK) s.append("back, "); 791 if (mask & AUDIO_CHANNEL_IN_LEFT_PROCESSED) s.append("left-processed, "); 792 if (mask & AUDIO_CHANNEL_IN_RIGHT_PROCESSED) s.append("right-processed, "); 793 if (mask & AUDIO_CHANNEL_IN_FRONT_PROCESSED) s.append("front-processed, "); 794 if (mask & AUDIO_CHANNEL_IN_BACK_PROCESSED) s.append("back-processed, "); 795 if (mask & AUDIO_CHANNEL_IN_PRESSURE) s.append("pressure, "); 796 if (mask & AUDIO_CHANNEL_IN_X_AXIS) s.append("X, "); 797 if (mask & AUDIO_CHANNEL_IN_Y_AXIS) s.append("Y, "); 798 if (mask & AUDIO_CHANNEL_IN_Z_AXIS) s.append("Z, "); 799 if (mask & AUDIO_CHANNEL_IN_VOICE_UPLINK) s.append("voice-uplink, "); 800 if (mask & AUDIO_CHANNEL_IN_VOICE_DNLINK) s.append("voice-dnlink, "); 801 if (mask & ~AUDIO_CHANNEL_IN_ALL) s.append("unknown, "); 802 } 803 const int len = s.length(); 804 if (len > 2) { 805 char *str = s.lockBuffer(len); // needed? 806 s.unlockBuffer(len - 2); // remove trailing ", " 807 } 808 return s; 809 } 810 case AUDIO_CHANNEL_REPRESENTATION_INDEX: 811 s.appendFormat("index mask, bits:%#x", audio_channel_mask_get_bits(mask)); 812 return s; 813 default: 814 s.appendFormat("unknown mask, representation:%d bits:%#x", 815 representation, audio_channel_mask_get_bits(mask)); 816 return s; 817 } 818} 819 820void AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args __unused) 821{ 822 const size_t SIZE = 256; 823 char buffer[SIZE]; 824 String8 result; 825 826 bool locked = AudioFlinger::dumpTryLock(mLock); 827 if (!locked) { 828 dprintf(fd, "thread %p may be deadlocked\n", this); 829 } 830 831 dprintf(fd, " Thread name: %s\n", mThreadName); 832 dprintf(fd, " I/O handle: %d\n", mId); 833 dprintf(fd, " TID: %d\n", getTid()); 834 dprintf(fd, " Standby: %s\n", mStandby ? "yes" : "no"); 835 dprintf(fd, " Sample rate: %u Hz\n", mSampleRate); 836 dprintf(fd, " HAL frame count: %zu\n", mFrameCount); 837 dprintf(fd, " HAL format: 0x%x (%s)\n", mHALFormat, formatToString(mHALFormat)); 838 dprintf(fd, " HAL buffer size: %u bytes\n", mBufferSize); 839 dprintf(fd, " Channel count: %u\n", mChannelCount); 840 dprintf(fd, " Channel mask: 0x%08x (%s)\n", mChannelMask, 841 channelMaskToString(mChannelMask, mType != RECORD).string()); 842 dprintf(fd, " Processing format: 0x%x (%s)\n", mFormat, formatToString(mFormat)); 843 dprintf(fd, " Processing frame size: %zu bytes\n", mFrameSize); 844 dprintf(fd, " Pending config events:"); 845 size_t numConfig = mConfigEvents.size(); 846 if (numConfig) { 847 for (size_t i = 0; i < numConfig; i++) { 848 mConfigEvents[i]->dump(buffer, SIZE); 849 dprintf(fd, "\n %s", buffer); 850 } 851 dprintf(fd, "\n"); 852 } else { 853 dprintf(fd, " none\n"); 854 } 855 dprintf(fd, " Output device: %#x (%s)\n", mOutDevice, devicesToString(mOutDevice).string()); 856 dprintf(fd, " Input device: %#x (%s)\n", mInDevice, devicesToString(mInDevice).string()); 857 dprintf(fd, " Audio source: %d (%s)\n", mAudioSource, sourceToString(mAudioSource)); 858 859 if (locked) { 860 mLock.unlock(); 861 } 862} 863 864void AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args) 865{ 866 const size_t SIZE = 256; 867 char buffer[SIZE]; 868 String8 result; 869 870 size_t numEffectChains = mEffectChains.size(); 871 snprintf(buffer, SIZE, " %zu Effect Chains\n", numEffectChains); 872 write(fd, buffer, strlen(buffer)); 873 874 for (size_t i = 0; i < numEffectChains; ++i) { 875 sp<EffectChain> chain = mEffectChains[i]; 876 if (chain != 0) { 877 chain->dump(fd, args); 878 } 879 } 880} 881 882void AudioFlinger::ThreadBase::acquireWakeLock(int uid) 883{ 884 Mutex::Autolock _l(mLock); 885 acquireWakeLock_l(uid); 886} 887 888String16 AudioFlinger::ThreadBase::getWakeLockTag() 889{ 890 switch (mType) { 891 case MIXER: 892 return String16("AudioMix"); 893 case DIRECT: 894 return String16("AudioDirectOut"); 895 case DUPLICATING: 896 return String16("AudioDup"); 897 case RECORD: 898 return String16("AudioIn"); 899 case OFFLOAD: 900 return String16("AudioOffload"); 901 default: 902 ALOG_ASSERT(false); 903 return String16("AudioUnknown"); 904 } 905} 906 907void AudioFlinger::ThreadBase::acquireWakeLock_l(int uid) 908{ 909 getPowerManager_l(); 910 if (mPowerManager != 0) { 911 sp<IBinder> binder = new BBinder(); 912 status_t status; 913 if (uid >= 0) { 914 status = mPowerManager->acquireWakeLockWithUid(POWERMANAGER_PARTIAL_WAKE_LOCK, 915 binder, 916 getWakeLockTag(), 917 String16("media"), 918 uid, 919 true /* FIXME force oneway contrary to .aidl */); 920 } else { 921 status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK, 922 binder, 923 getWakeLockTag(), 924 String16("media"), 925 true /* FIXME force oneway contrary to .aidl */); 926 } 927 if (status == NO_ERROR) { 928 mWakeLockToken = binder; 929 } 930 ALOGV("acquireWakeLock_l() %s status %d", mThreadName, status); 931 } 932 933 if (!mNotifiedBatteryStart) { 934 BatteryNotifier::getInstance().noteStartAudio(); 935 mNotifiedBatteryStart = true; 936 } 937} 938 939void AudioFlinger::ThreadBase::releaseWakeLock() 940{ 941 Mutex::Autolock _l(mLock); 942 releaseWakeLock_l(); 943} 944 945void AudioFlinger::ThreadBase::releaseWakeLock_l() 946{ 947 if (mWakeLockToken != 0) { 948 ALOGV("releaseWakeLock_l() %s", mThreadName); 949 if (mPowerManager != 0) { 950 mPowerManager->releaseWakeLock(mWakeLockToken, 0, 951 true /* FIXME force oneway contrary to .aidl */); 952 } 953 mWakeLockToken.clear(); 954 } 955 956 if (mNotifiedBatteryStart) { 957 BatteryNotifier::getInstance().noteStopAudio(); 958 mNotifiedBatteryStart = false; 959 } 960} 961 962void AudioFlinger::ThreadBase::updateWakeLockUids(const SortedVector<int> &uids) { 963 Mutex::Autolock _l(mLock); 964 updateWakeLockUids_l(uids); 965} 966 967void AudioFlinger::ThreadBase::getPowerManager_l() { 968 if (mSystemReady && mPowerManager == 0) { 969 // use checkService() to avoid blocking if power service is not up yet 970 sp<IBinder> binder = 971 defaultServiceManager()->checkService(String16("power")); 972 if (binder == 0) { 973 ALOGW("Thread %s cannot connect to the power manager service", mThreadName); 974 } else { 975 mPowerManager = interface_cast<IPowerManager>(binder); 976 binder->linkToDeath(mDeathRecipient); 977 } 978 } 979} 980 981void AudioFlinger::ThreadBase::updateWakeLockUids_l(const SortedVector<int> &uids) { 982 getPowerManager_l(); 983 if (mWakeLockToken == NULL) { 984 ALOGE("no wake lock to update!"); 985 return; 986 } 987 if (mPowerManager != 0) { 988 sp<IBinder> binder = new BBinder(); 989 status_t status; 990 status = mPowerManager->updateWakeLockUids(mWakeLockToken, uids.size(), uids.array(), 991 true /* FIXME force oneway contrary to .aidl */); 992 ALOGV("acquireWakeLock_l() %s status %d", mThreadName, status); 993 } 994} 995 996void AudioFlinger::ThreadBase::clearPowerManager() 997{ 998 Mutex::Autolock _l(mLock); 999 releaseWakeLock_l(); 1000 mPowerManager.clear(); 1001} 1002 1003void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who __unused) 1004{ 1005 sp<ThreadBase> thread = mThread.promote(); 1006 if (thread != 0) { 1007 thread->clearPowerManager(); 1008 } 1009 ALOGW("power manager service died !!!"); 1010} 1011 1012void AudioFlinger::ThreadBase::setEffectSuspended( 1013 const effect_uuid_t *type, bool suspend, int sessionId) 1014{ 1015 Mutex::Autolock _l(mLock); 1016 setEffectSuspended_l(type, suspend, sessionId); 1017} 1018 1019void AudioFlinger::ThreadBase::setEffectSuspended_l( 1020 const effect_uuid_t *type, bool suspend, int sessionId) 1021{ 1022 sp<EffectChain> chain = getEffectChain_l(sessionId); 1023 if (chain != 0) { 1024 if (type != NULL) { 1025 chain->setEffectSuspended_l(type, suspend); 1026 } else { 1027 chain->setEffectSuspendedAll_l(suspend); 1028 } 1029 } 1030 1031 updateSuspendedSessions_l(type, suspend, sessionId); 1032} 1033 1034void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain) 1035{ 1036 ssize_t index = mSuspendedSessions.indexOfKey(chain->sessionId()); 1037 if (index < 0) { 1038 return; 1039 } 1040 1041 const KeyedVector <int, sp<SuspendedSessionDesc> >& sessionEffects = 1042 mSuspendedSessions.valueAt(index); 1043 1044 for (size_t i = 0; i < sessionEffects.size(); i++) { 1045 sp<SuspendedSessionDesc> desc = sessionEffects.valueAt(i); 1046 for (int j = 0; j < desc->mRefCount; j++) { 1047 if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) { 1048 chain->setEffectSuspendedAll_l(true); 1049 } else { 1050 ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x", 1051 desc->mType.timeLow); 1052 chain->setEffectSuspended_l(&desc->mType, true); 1053 } 1054 } 1055 } 1056} 1057 1058void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type, 1059 bool suspend, 1060 int sessionId) 1061{ 1062 ssize_t index = mSuspendedSessions.indexOfKey(sessionId); 1063 1064 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects; 1065 1066 if (suspend) { 1067 if (index >= 0) { 1068 sessionEffects = mSuspendedSessions.valueAt(index); 1069 } else { 1070 mSuspendedSessions.add(sessionId, sessionEffects); 1071 } 1072 } else { 1073 if (index < 0) { 1074 return; 1075 } 1076 sessionEffects = mSuspendedSessions.valueAt(index); 1077 } 1078 1079 1080 int key = EffectChain::kKeyForSuspendAll; 1081 if (type != NULL) { 1082 key = type->timeLow; 1083 } 1084 index = sessionEffects.indexOfKey(key); 1085 1086 sp<SuspendedSessionDesc> desc; 1087 if (suspend) { 1088 if (index >= 0) { 1089 desc = sessionEffects.valueAt(index); 1090 } else { 1091 desc = new SuspendedSessionDesc(); 1092 if (type != NULL) { 1093 desc->mType = *type; 1094 } 1095 sessionEffects.add(key, desc); 1096 ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key); 1097 } 1098 desc->mRefCount++; 1099 } else { 1100 if (index < 0) { 1101 return; 1102 } 1103 desc = sessionEffects.valueAt(index); 1104 if (--desc->mRefCount == 0) { 1105 ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key); 1106 sessionEffects.removeItemsAt(index); 1107 if (sessionEffects.isEmpty()) { 1108 ALOGV("updateSuspendedSessions_l() restore removing session %d", 1109 sessionId); 1110 mSuspendedSessions.removeItem(sessionId); 1111 } 1112 } 1113 } 1114 if (!sessionEffects.isEmpty()) { 1115 mSuspendedSessions.replaceValueFor(sessionId, sessionEffects); 1116 } 1117} 1118 1119void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 1120 bool enabled, 1121 int sessionId) 1122{ 1123 Mutex::Autolock _l(mLock); 1124 checkSuspendOnEffectEnabled_l(effect, enabled, sessionId); 1125} 1126 1127void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 1128 bool enabled, 1129 int sessionId) 1130{ 1131 if (mType != RECORD) { 1132 // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on 1133 // another session. This gives the priority to well behaved effect control panels 1134 // and applications not using global effects. 1135 // Enabling post processing in AUDIO_SESSION_OUTPUT_STAGE session does not affect 1136 // global effects 1137 if ((sessionId != AUDIO_SESSION_OUTPUT_MIX) && (sessionId != AUDIO_SESSION_OUTPUT_STAGE)) { 1138 setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX); 1139 } 1140 } 1141 1142 sp<EffectChain> chain = getEffectChain_l(sessionId); 1143 if (chain != 0) { 1144 chain->checkSuspendOnEffectEnabled(effect, enabled); 1145 } 1146} 1147 1148// ThreadBase::createEffect_l() must be called with AudioFlinger::mLock held 1149sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 1150 const sp<AudioFlinger::Client>& client, 1151 const sp<IEffectClient>& effectClient, 1152 int32_t priority, 1153 int sessionId, 1154 effect_descriptor_t *desc, 1155 int *enabled, 1156 status_t *status) 1157{ 1158 sp<EffectModule> effect; 1159 sp<EffectHandle> handle; 1160 status_t lStatus; 1161 sp<EffectChain> chain; 1162 bool chainCreated = false; 1163 bool effectCreated = false; 1164 bool effectRegistered = false; 1165 1166 lStatus = initCheck(); 1167 if (lStatus != NO_ERROR) { 1168 ALOGW("createEffect_l() Audio driver not initialized."); 1169 goto Exit; 1170 } 1171 1172 // Reject any effect on Direct output threads for now, since the format of 1173 // mSinkBuffer is not guaranteed to be compatible with effect processing (PCM 16 stereo). 1174 if (mType == DIRECT) { 1175 ALOGW("createEffect_l() Cannot add effect %s on Direct output type thread %s", 1176 desc->name, mThreadName); 1177 lStatus = BAD_VALUE; 1178 goto Exit; 1179 } 1180 1181 // Reject any effect on mixer or duplicating multichannel sinks. 1182 // TODO: fix both format and multichannel issues with effects. 1183 if ((mType == MIXER || mType == DUPLICATING) && mChannelCount != FCC_2) { 1184 ALOGW("createEffect_l() Cannot add effect %s for multichannel(%d) %s threads", 1185 desc->name, mChannelCount, mType == MIXER ? "MIXER" : "DUPLICATING"); 1186 lStatus = BAD_VALUE; 1187 goto Exit; 1188 } 1189 1190 // Allow global effects only on offloaded and mixer threads 1191 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 1192 switch (mType) { 1193 case MIXER: 1194 case OFFLOAD: 1195 break; 1196 case DIRECT: 1197 case DUPLICATING: 1198 case RECORD: 1199 default: 1200 ALOGW("createEffect_l() Cannot add global effect %s on thread %s", 1201 desc->name, mThreadName); 1202 lStatus = BAD_VALUE; 1203 goto Exit; 1204 } 1205 } 1206 1207 // Only Pre processor effects are allowed on input threads and only on input threads 1208 if ((mType == RECORD) != ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { 1209 ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d", 1210 desc->name, desc->flags, mType); 1211 lStatus = BAD_VALUE; 1212 goto Exit; 1213 } 1214 1215 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 1216 1217 { // scope for mLock 1218 Mutex::Autolock _l(mLock); 1219 1220 // check for existing effect chain with the requested audio session 1221 chain = getEffectChain_l(sessionId); 1222 if (chain == 0) { 1223 // create a new chain for this session 1224 ALOGV("createEffect_l() new effect chain for session %d", sessionId); 1225 chain = new EffectChain(this, sessionId); 1226 addEffectChain_l(chain); 1227 chain->setStrategy(getStrategyForSession_l(sessionId)); 1228 chainCreated = true; 1229 } else { 1230 effect = chain->getEffectFromDesc_l(desc); 1231 } 1232 1233 ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get()); 1234 1235 if (effect == 0) { 1236 int id = mAudioFlinger->nextUniqueId(); 1237 // Check CPU and memory usage 1238 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 1239 if (lStatus != NO_ERROR) { 1240 goto Exit; 1241 } 1242 effectRegistered = true; 1243 // create a new effect module if none present in the chain 1244 effect = new EffectModule(this, chain, desc, id, sessionId); 1245 lStatus = effect->status(); 1246 if (lStatus != NO_ERROR) { 1247 goto Exit; 1248 } 1249 effect->setOffloaded(mType == OFFLOAD, mId); 1250 1251 lStatus = chain->addEffect_l(effect); 1252 if (lStatus != NO_ERROR) { 1253 goto Exit; 1254 } 1255 effectCreated = true; 1256 1257 effect->setDevice(mOutDevice); 1258 effect->setDevice(mInDevice); 1259 effect->setMode(mAudioFlinger->getMode()); 1260 effect->setAudioSource(mAudioSource); 1261 } 1262 // create effect handle and connect it to effect module 1263 handle = new EffectHandle(effect, client, effectClient, priority); 1264 lStatus = handle->initCheck(); 1265 if (lStatus == OK) { 1266 lStatus = effect->addHandle(handle.get()); 1267 } 1268 if (enabled != NULL) { 1269 *enabled = (int)effect->isEnabled(); 1270 } 1271 } 1272 1273Exit: 1274 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 1275 Mutex::Autolock _l(mLock); 1276 if (effectCreated) { 1277 chain->removeEffect_l(effect); 1278 } 1279 if (effectRegistered) { 1280 AudioSystem::unregisterEffect(effect->id()); 1281 } 1282 if (chainCreated) { 1283 removeEffectChain_l(chain); 1284 } 1285 handle.clear(); 1286 } 1287 1288 *status = lStatus; 1289 return handle; 1290} 1291 1292sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect(int sessionId, int effectId) 1293{ 1294 Mutex::Autolock _l(mLock); 1295 return getEffect_l(sessionId, effectId); 1296} 1297 1298sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId) 1299{ 1300 sp<EffectChain> chain = getEffectChain_l(sessionId); 1301 return chain != 0 ? chain->getEffectFromId_l(effectId) : 0; 1302} 1303 1304// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 1305// PlaybackThread::mLock held 1306status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 1307{ 1308 // check for existing effect chain with the requested audio session 1309 int sessionId = effect->sessionId(); 1310 sp<EffectChain> chain = getEffectChain_l(sessionId); 1311 bool chainCreated = false; 1312 1313 ALOGD_IF((mType == OFFLOAD) && !effect->isOffloadable(), 1314 "addEffect_l() on offloaded thread %p: effect %s does not support offload flags %x", 1315 this, effect->desc().name, effect->desc().flags); 1316 1317 if (chain == 0) { 1318 // create a new chain for this session 1319 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 1320 chain = new EffectChain(this, sessionId); 1321 addEffectChain_l(chain); 1322 chain->setStrategy(getStrategyForSession_l(sessionId)); 1323 chainCreated = true; 1324 } 1325 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 1326 1327 if (chain->getEffectFromId_l(effect->id()) != 0) { 1328 ALOGW("addEffect_l() %p effect %s already present in chain %p", 1329 this, effect->desc().name, chain.get()); 1330 return BAD_VALUE; 1331 } 1332 1333 effect->setOffloaded(mType == OFFLOAD, mId); 1334 1335 status_t status = chain->addEffect_l(effect); 1336 if (status != NO_ERROR) { 1337 if (chainCreated) { 1338 removeEffectChain_l(chain); 1339 } 1340 return status; 1341 } 1342 1343 effect->setDevice(mOutDevice); 1344 effect->setDevice(mInDevice); 1345 effect->setMode(mAudioFlinger->getMode()); 1346 effect->setAudioSource(mAudioSource); 1347 return NO_ERROR; 1348} 1349 1350void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 1351 1352 ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 1353 effect_descriptor_t desc = effect->desc(); 1354 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 1355 detachAuxEffect_l(effect->id()); 1356 } 1357 1358 sp<EffectChain> chain = effect->chain().promote(); 1359 if (chain != 0) { 1360 // remove effect chain if removing last effect 1361 if (chain->removeEffect_l(effect) == 0) { 1362 removeEffectChain_l(chain); 1363 } 1364 } else { 1365 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 1366 } 1367} 1368 1369void AudioFlinger::ThreadBase::lockEffectChains_l( 1370 Vector< sp<AudioFlinger::EffectChain> >& effectChains) 1371{ 1372 effectChains = mEffectChains; 1373 for (size_t i = 0; i < mEffectChains.size(); i++) { 1374 mEffectChains[i]->lock(); 1375 } 1376} 1377 1378void AudioFlinger::ThreadBase::unlockEffectChains( 1379 const Vector< sp<AudioFlinger::EffectChain> >& effectChains) 1380{ 1381 for (size_t i = 0; i < effectChains.size(); i++) { 1382 effectChains[i]->unlock(); 1383 } 1384} 1385 1386sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId) 1387{ 1388 Mutex::Autolock _l(mLock); 1389 return getEffectChain_l(sessionId); 1390} 1391 1392sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) const 1393{ 1394 size_t size = mEffectChains.size(); 1395 for (size_t i = 0; i < size; i++) { 1396 if (mEffectChains[i]->sessionId() == sessionId) { 1397 return mEffectChains[i]; 1398 } 1399 } 1400 return 0; 1401} 1402 1403void AudioFlinger::ThreadBase::setMode(audio_mode_t mode) 1404{ 1405 Mutex::Autolock _l(mLock); 1406 size_t size = mEffectChains.size(); 1407 for (size_t i = 0; i < size; i++) { 1408 mEffectChains[i]->setMode_l(mode); 1409 } 1410} 1411 1412void AudioFlinger::ThreadBase::getAudioPortConfig(struct audio_port_config *config) 1413{ 1414 config->type = AUDIO_PORT_TYPE_MIX; 1415 config->ext.mix.handle = mId; 1416 config->sample_rate = mSampleRate; 1417 config->format = mFormat; 1418 config->channel_mask = mChannelMask; 1419 config->config_mask = AUDIO_PORT_CONFIG_SAMPLE_RATE|AUDIO_PORT_CONFIG_CHANNEL_MASK| 1420 AUDIO_PORT_CONFIG_FORMAT; 1421} 1422 1423void AudioFlinger::ThreadBase::systemReady() 1424{ 1425 Mutex::Autolock _l(mLock); 1426 if (mSystemReady) { 1427 return; 1428 } 1429 mSystemReady = true; 1430 1431 for (size_t i = 0; i < mPendingConfigEvents.size(); i++) { 1432 sendConfigEvent_l(mPendingConfigEvents.editItemAt(i)); 1433 } 1434 mPendingConfigEvents.clear(); 1435} 1436 1437 1438// ---------------------------------------------------------------------------- 1439// Playback 1440// ---------------------------------------------------------------------------- 1441 1442AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, 1443 AudioStreamOut* output, 1444 audio_io_handle_t id, 1445 audio_devices_t device, 1446 type_t type, 1447 bool systemReady) 1448 : ThreadBase(audioFlinger, id, device, AUDIO_DEVICE_NONE, type, systemReady), 1449 mNormalFrameCount(0), mSinkBuffer(NULL), 1450 mMixerBufferEnabled(AudioFlinger::kEnableExtendedPrecision), 1451 mMixerBuffer(NULL), 1452 mMixerBufferSize(0), 1453 mMixerBufferFormat(AUDIO_FORMAT_INVALID), 1454 mMixerBufferValid(false), 1455 mEffectBufferEnabled(AudioFlinger::kEnableExtendedPrecision), 1456 mEffectBuffer(NULL), 1457 mEffectBufferSize(0), 1458 mEffectBufferFormat(AUDIO_FORMAT_INVALID), 1459 mEffectBufferValid(false), 1460 mSuspended(0), mBytesWritten(0), 1461 mActiveTracksGeneration(0), 1462 // mStreamTypes[] initialized in constructor body 1463 mOutput(output), 1464 mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false), 1465 mMixerStatus(MIXER_IDLE), 1466 mMixerStatusIgnoringFastTracks(MIXER_IDLE), 1467 mStandbyDelayNs(AudioFlinger::mStandbyTimeInNsecs), 1468 mBytesRemaining(0), 1469 mCurrentWriteLength(0), 1470 mUseAsyncWrite(false), 1471 mWriteAckSequence(0), 1472 mDrainSequence(0), 1473 mSignalPending(false), 1474 mScreenState(AudioFlinger::mScreenState), 1475 // index 0 is reserved for normal mixer's submix 1476 mFastTrackAvailMask(((1 << FastMixerState::kMaxFastTracks) - 1) & ~1), 1477 mHwSupportsPause(false), mHwPaused(false), mFlushPending(false), 1478 // mLatchD, mLatchQ, 1479 mLatchDValid(false), mLatchQValid(false) 1480{ 1481 snprintf(mThreadName, kThreadNameLength, "AudioOut_%X", id); 1482 mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mThreadName); 1483 1484 // Assumes constructor is called by AudioFlinger with it's mLock held, but 1485 // it would be safer to explicitly pass initial masterVolume/masterMute as 1486 // parameter. 1487 // 1488 // If the HAL we are using has support for master volume or master mute, 1489 // then do not attenuate or mute during mixing (just leave the volume at 1.0 1490 // and the mute set to false). 1491 mMasterVolume = audioFlinger->masterVolume_l(); 1492 mMasterMute = audioFlinger->masterMute_l(); 1493 if (mOutput && mOutput->audioHwDev) { 1494 if (mOutput->audioHwDev->canSetMasterVolume()) { 1495 mMasterVolume = 1.0; 1496 } 1497 1498 if (mOutput->audioHwDev->canSetMasterMute()) { 1499 mMasterMute = false; 1500 } 1501 } 1502 1503 readOutputParameters_l(); 1504 1505 // ++ operator does not compile 1506 for (audio_stream_type_t stream = AUDIO_STREAM_MIN; stream < AUDIO_STREAM_CNT; 1507 stream = (audio_stream_type_t) (stream + 1)) { 1508 mStreamTypes[stream].volume = mAudioFlinger->streamVolume_l(stream); 1509 mStreamTypes[stream].mute = mAudioFlinger->streamMute_l(stream); 1510 } 1511} 1512 1513AudioFlinger::PlaybackThread::~PlaybackThread() 1514{ 1515 mAudioFlinger->unregisterWriter(mNBLogWriter); 1516 free(mSinkBuffer); 1517 free(mMixerBuffer); 1518 free(mEffectBuffer); 1519} 1520 1521void AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) 1522{ 1523 dumpInternals(fd, args); 1524 dumpTracks(fd, args); 1525 dumpEffectChains(fd, args); 1526} 1527 1528void AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args __unused) 1529{ 1530 const size_t SIZE = 256; 1531 char buffer[SIZE]; 1532 String8 result; 1533 1534 result.appendFormat(" Stream volumes in dB: "); 1535 for (int i = 0; i < AUDIO_STREAM_CNT; ++i) { 1536 const stream_type_t *st = &mStreamTypes[i]; 1537 if (i > 0) { 1538 result.appendFormat(", "); 1539 } 1540 result.appendFormat("%d:%.2g", i, 20.0 * log10(st->volume)); 1541 if (st->mute) { 1542 result.append("M"); 1543 } 1544 } 1545 result.append("\n"); 1546 write(fd, result.string(), result.length()); 1547 result.clear(); 1548 1549 // These values are "raw"; they will wrap around. See prepareTracks_l() for a better way. 1550 FastTrackUnderruns underruns = getFastTrackUnderruns(0); 1551 dprintf(fd, " Normal mixer raw underrun counters: partial=%u empty=%u\n", 1552 underruns.mBitFields.mPartial, underruns.mBitFields.mEmpty); 1553 1554 size_t numtracks = mTracks.size(); 1555 size_t numactive = mActiveTracks.size(); 1556 dprintf(fd, " %d Tracks", numtracks); 1557 size_t numactiveseen = 0; 1558 if (numtracks) { 1559 dprintf(fd, " of which %d are active\n", numactive); 1560 Track::appendDumpHeader(result); 1561 for (size_t i = 0; i < numtracks; ++i) { 1562 sp<Track> track = mTracks[i]; 1563 if (track != 0) { 1564 bool active = mActiveTracks.indexOf(track) >= 0; 1565 if (active) { 1566 numactiveseen++; 1567 } 1568 track->dump(buffer, SIZE, active); 1569 result.append(buffer); 1570 } 1571 } 1572 } else { 1573 result.append("\n"); 1574 } 1575 if (numactiveseen != numactive) { 1576 // some tracks in the active list were not in the tracks list 1577 snprintf(buffer, SIZE, " The following tracks are in the active list but" 1578 " not in the track list\n"); 1579 result.append(buffer); 1580 Track::appendDumpHeader(result); 1581 for (size_t i = 0; i < numactive; ++i) { 1582 sp<Track> track = mActiveTracks[i].promote(); 1583 if (track != 0 && mTracks.indexOf(track) < 0) { 1584 track->dump(buffer, SIZE, true); 1585 result.append(buffer); 1586 } 1587 } 1588 } 1589 1590 write(fd, result.string(), result.size()); 1591} 1592 1593void AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 1594{ 1595 dprintf(fd, "\nOutput thread %p type %d (%s):\n", this, type(), threadTypeToString(type())); 1596 1597 dumpBase(fd, args); 1598 1599 dprintf(fd, " Normal frame count: %zu\n", mNormalFrameCount); 1600 dprintf(fd, " Last write occurred (msecs): %llu\n", ns2ms(systemTime() - mLastWriteTime)); 1601 dprintf(fd, " Total writes: %d\n", mNumWrites); 1602 dprintf(fd, " Delayed writes: %d\n", mNumDelayedWrites); 1603 dprintf(fd, " Blocked in write: %s\n", mInWrite ? "yes" : "no"); 1604 dprintf(fd, " Suspend count: %d\n", mSuspended); 1605 dprintf(fd, " Sink buffer : %p\n", mSinkBuffer); 1606 dprintf(fd, " Mixer buffer: %p\n", mMixerBuffer); 1607 dprintf(fd, " Effect buffer: %p\n", mEffectBuffer); 1608 dprintf(fd, " Fast track availMask=%#x\n", mFastTrackAvailMask); 1609 AudioStreamOut *output = mOutput; 1610 audio_output_flags_t flags = output != NULL ? output->flags : AUDIO_OUTPUT_FLAG_NONE; 1611 String8 flagsAsString = outputFlagsToString(flags); 1612 dprintf(fd, " AudioStreamOut: %p flags %#x (%s)\n", output, flags, flagsAsString.string()); 1613} 1614 1615// Thread virtuals 1616 1617void AudioFlinger::PlaybackThread::onFirstRef() 1618{ 1619 run(mThreadName, ANDROID_PRIORITY_URGENT_AUDIO); 1620} 1621 1622// ThreadBase virtuals 1623void AudioFlinger::PlaybackThread::preExit() 1624{ 1625 ALOGV(" preExit()"); 1626 // FIXME this is using hard-coded strings but in the future, this functionality will be 1627 // converted to use audio HAL extensions required to support tunneling 1628 mOutput->stream->common.set_parameters(&mOutput->stream->common, "exiting=1"); 1629} 1630 1631// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 1632sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 1633 const sp<AudioFlinger::Client>& client, 1634 audio_stream_type_t streamType, 1635 uint32_t sampleRate, 1636 audio_format_t format, 1637 audio_channel_mask_t channelMask, 1638 size_t *pFrameCount, 1639 const sp<IMemory>& sharedBuffer, 1640 int sessionId, 1641 IAudioFlinger::track_flags_t *flags, 1642 pid_t tid, 1643 int uid, 1644 status_t *status) 1645{ 1646 size_t frameCount = *pFrameCount; 1647 sp<Track> track; 1648 status_t lStatus; 1649 1650 bool isTimed = (*flags & IAudioFlinger::TRACK_TIMED) != 0; 1651 1652 // client expresses a preference for FAST, but we get the final say 1653 if (*flags & IAudioFlinger::TRACK_FAST) { 1654 if ( 1655 // not timed 1656 (!isTimed) && 1657 // either of these use cases: 1658 ( 1659 // use case 1: shared buffer with any frame count 1660 ( 1661 (sharedBuffer != 0) 1662 ) || 1663 // use case 2: frame count is default or at least as large as HAL 1664 ( 1665 // we formerly checked for a callback handler (non-0 tid), 1666 // but that is no longer required for TRANSFER_OBTAIN mode 1667 ((frameCount == 0) || 1668 (frameCount >= mFrameCount)) 1669 ) 1670 ) && 1671 // PCM data 1672 audio_is_linear_pcm(format) && 1673 // TODO: extract as a data library function that checks that a computationally 1674 // expensive downmixer is not required: isFastOutputChannelConversion() 1675 (channelMask == mChannelMask || 1676 mChannelMask != AUDIO_CHANNEL_OUT_STEREO || 1677 (channelMask == AUDIO_CHANNEL_OUT_MONO 1678 /* && mChannelMask == AUDIO_CHANNEL_OUT_STEREO */)) && 1679 // hardware sample rate 1680 (sampleRate == mSampleRate) && 1681 // normal mixer has an associated fast mixer 1682 hasFastMixer() && 1683 // there are sufficient fast track slots available 1684 (mFastTrackAvailMask != 0) 1685 // FIXME test that MixerThread for this fast track has a capable output HAL 1686 // FIXME add a permission test also? 1687 ) { 1688 // if frameCount not specified, then it defaults to fast mixer (HAL) frame count 1689 if (frameCount == 0) { 1690 // read the fast track multiplier property the first time it is needed 1691 int ok = pthread_once(&sFastTrackMultiplierOnce, sFastTrackMultiplierInit); 1692 if (ok != 0) { 1693 ALOGE("%s pthread_once failed: %d", __func__, ok); 1694 } 1695 frameCount = mFrameCount * sFastTrackMultiplier; 1696 } 1697 ALOGV("AUDIO_OUTPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d", 1698 frameCount, mFrameCount); 1699 } else { 1700 ALOGV("AUDIO_OUTPUT_FLAG_FAST denied: isTimed=%d sharedBuffer=%p frameCount=%d " 1701 "mFrameCount=%d format=%#x mFormat=%#x isLinear=%d channelMask=%#x " 1702 "sampleRate=%u mSampleRate=%u " 1703 "hasFastMixer=%d tid=%d fastTrackAvailMask=%#x", 1704 isTimed, sharedBuffer.get(), frameCount, mFrameCount, format, mFormat, 1705 audio_is_linear_pcm(format), 1706 channelMask, sampleRate, mSampleRate, hasFastMixer(), tid, mFastTrackAvailMask); 1707 *flags &= ~IAudioFlinger::TRACK_FAST; 1708 } 1709 } 1710 // For normal PCM streaming tracks, update minimum frame count. 1711 // For compatibility with AudioTrack calculation, buffer depth is forced 1712 // to be at least 2 x the normal mixer frame count and cover audio hardware latency. 1713 // This is probably too conservative, but legacy application code may depend on it. 1714 // If you change this calculation, also review the start threshold which is related. 1715 if (!(*flags & IAudioFlinger::TRACK_FAST) 1716 && audio_is_linear_pcm(format) && sharedBuffer == 0) { 1717 // this must match AudioTrack.cpp calculateMinFrameCount(). 1718 // TODO: Move to a common library 1719 uint32_t latencyMs = mOutput->stream->get_latency(mOutput->stream); 1720 uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate); 1721 if (minBufCount < 2) { 1722 minBufCount = 2; 1723 } 1724 // For normal mixing tracks, if speed is > 1.0f (normal), AudioTrack 1725 // or the client should compute and pass in a larger buffer request. 1726 size_t minFrameCount = 1727 minBufCount * sourceFramesNeededWithTimestretch( 1728 sampleRate, mNormalFrameCount, 1729 mSampleRate, AUDIO_TIMESTRETCH_SPEED_NORMAL /*speed*/); 1730 if (frameCount < minFrameCount) { // including frameCount == 0 1731 frameCount = minFrameCount; 1732 } 1733 } 1734 *pFrameCount = frameCount; 1735 1736 switch (mType) { 1737 1738 case DIRECT: 1739 if (audio_is_linear_pcm(format)) { 1740 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1741 ALOGE("createTrack_l() Bad parameter: sampleRate %u format %#x, channelMask 0x%08x " 1742 "for output %p with format %#x", 1743 sampleRate, format, channelMask, mOutput, mFormat); 1744 lStatus = BAD_VALUE; 1745 goto Exit; 1746 } 1747 } 1748 break; 1749 1750 case OFFLOAD: 1751 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1752 ALOGE("createTrack_l() Bad parameter: sampleRate %d format %#x, channelMask 0x%08x \"" 1753 "for output %p with format %#x", 1754 sampleRate, format, channelMask, mOutput, mFormat); 1755 lStatus = BAD_VALUE; 1756 goto Exit; 1757 } 1758 break; 1759 1760 default: 1761 if (!audio_is_linear_pcm(format)) { 1762 ALOGE("createTrack_l() Bad parameter: format %#x \"" 1763 "for output %p with format %#x", 1764 format, mOutput, mFormat); 1765 lStatus = BAD_VALUE; 1766 goto Exit; 1767 } 1768 if (sampleRate > mSampleRate * AUDIO_RESAMPLER_DOWN_RATIO_MAX) { 1769 ALOGE("Sample rate out of range: %u mSampleRate %u", sampleRate, mSampleRate); 1770 lStatus = BAD_VALUE; 1771 goto Exit; 1772 } 1773 break; 1774 1775 } 1776 1777 lStatus = initCheck(); 1778 if (lStatus != NO_ERROR) { 1779 ALOGE("createTrack_l() audio driver not initialized"); 1780 goto Exit; 1781 } 1782 1783 { // scope for mLock 1784 Mutex::Autolock _l(mLock); 1785 1786 // all tracks in same audio session must share the same routing strategy otherwise 1787 // conflicts will happen when tracks are moved from one output to another by audio policy 1788 // manager 1789 uint32_t strategy = AudioSystem::getStrategyForStream(streamType); 1790 for (size_t i = 0; i < mTracks.size(); ++i) { 1791 sp<Track> t = mTracks[i]; 1792 if (t != 0 && t->isExternalTrack()) { 1793 uint32_t actual = AudioSystem::getStrategyForStream(t->streamType()); 1794 if (sessionId == t->sessionId() && strategy != actual) { 1795 ALOGE("createTrack_l() mismatched strategy; expected %u but found %u", 1796 strategy, actual); 1797 lStatus = BAD_VALUE; 1798 goto Exit; 1799 } 1800 } 1801 } 1802 1803 if (!isTimed) { 1804 track = new Track(this, client, streamType, sampleRate, format, 1805 channelMask, frameCount, NULL, sharedBuffer, 1806 sessionId, uid, *flags, TrackBase::TYPE_DEFAULT); 1807 } else { 1808 track = TimedTrack::create(this, client, streamType, sampleRate, format, 1809 channelMask, frameCount, sharedBuffer, sessionId, uid); 1810 } 1811 1812 // new Track always returns non-NULL, 1813 // but TimedTrack::create() is a factory that could fail by returning NULL 1814 lStatus = track != 0 ? track->initCheck() : (status_t) NO_MEMORY; 1815 if (lStatus != NO_ERROR) { 1816 ALOGE("createTrack_l() initCheck failed %d; no control block?", lStatus); 1817 // track must be cleared from the caller as the caller has the AF lock 1818 goto Exit; 1819 } 1820 mTracks.add(track); 1821 1822 sp<EffectChain> chain = getEffectChain_l(sessionId); 1823 if (chain != 0) { 1824 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1825 track->setMainBuffer(chain->inBuffer()); 1826 chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType())); 1827 chain->incTrackCnt(); 1828 } 1829 1830 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 1831 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 1832 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 1833 // so ask activity manager to do this on our behalf 1834 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 1835 } 1836 } 1837 1838 lStatus = NO_ERROR; 1839 1840Exit: 1841 *status = lStatus; 1842 return track; 1843} 1844 1845uint32_t AudioFlinger::PlaybackThread::correctLatency_l(uint32_t latency) const 1846{ 1847 return latency; 1848} 1849 1850uint32_t AudioFlinger::PlaybackThread::latency() const 1851{ 1852 Mutex::Autolock _l(mLock); 1853 return latency_l(); 1854} 1855uint32_t AudioFlinger::PlaybackThread::latency_l() const 1856{ 1857 if (initCheck() == NO_ERROR) { 1858 return correctLatency_l(mOutput->stream->get_latency(mOutput->stream)); 1859 } else { 1860 return 0; 1861 } 1862} 1863 1864void AudioFlinger::PlaybackThread::setMasterVolume(float value) 1865{ 1866 Mutex::Autolock _l(mLock); 1867 // Don't apply master volume in SW if our HAL can do it for us. 1868 if (mOutput && mOutput->audioHwDev && 1869 mOutput->audioHwDev->canSetMasterVolume()) { 1870 mMasterVolume = 1.0; 1871 } else { 1872 mMasterVolume = value; 1873 } 1874} 1875 1876void AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1877{ 1878 Mutex::Autolock _l(mLock); 1879 // Don't apply master mute in SW if our HAL can do it for us. 1880 if (mOutput && mOutput->audioHwDev && 1881 mOutput->audioHwDev->canSetMasterMute()) { 1882 mMasterMute = false; 1883 } else { 1884 mMasterMute = muted; 1885 } 1886} 1887 1888void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) 1889{ 1890 Mutex::Autolock _l(mLock); 1891 mStreamTypes[stream].volume = value; 1892 broadcast_l(); 1893} 1894 1895void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) 1896{ 1897 Mutex::Autolock _l(mLock); 1898 mStreamTypes[stream].mute = muted; 1899 broadcast_l(); 1900} 1901 1902float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const 1903{ 1904 Mutex::Autolock _l(mLock); 1905 return mStreamTypes[stream].volume; 1906} 1907 1908// addTrack_l() must be called with ThreadBase::mLock held 1909status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 1910{ 1911 status_t status = ALREADY_EXISTS; 1912 1913 // set retry count for buffer fill 1914 track->mRetryCount = kMaxTrackStartupRetries; 1915 if (mActiveTracks.indexOf(track) < 0) { 1916 // the track is newly added, make sure it fills up all its 1917 // buffers before playing. This is to ensure the client will 1918 // effectively get the latency it requested. 1919 if (track->isExternalTrack()) { 1920 TrackBase::track_state state = track->mState; 1921 mLock.unlock(); 1922 status = AudioSystem::startOutput(mId, track->streamType(), 1923 (audio_session_t)track->sessionId()); 1924 mLock.lock(); 1925 // abort track was stopped/paused while we released the lock 1926 if (state != track->mState) { 1927 if (status == NO_ERROR) { 1928 mLock.unlock(); 1929 AudioSystem::stopOutput(mId, track->streamType(), 1930 (audio_session_t)track->sessionId()); 1931 mLock.lock(); 1932 } 1933 return INVALID_OPERATION; 1934 } 1935 // abort if start is rejected by audio policy manager 1936 if (status != NO_ERROR) { 1937 return PERMISSION_DENIED; 1938 } 1939#ifdef ADD_BATTERY_DATA 1940 // to track the speaker usage 1941 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 1942#endif 1943 } 1944 1945 track->mFillingUpStatus = track->sharedBuffer() != 0 ? Track::FS_FILLED : Track::FS_FILLING; 1946 track->mResetDone = false; 1947 track->mPresentationCompleteFrames = 0; 1948 mActiveTracks.add(track); 1949 mWakeLockUids.add(track->uid()); 1950 mActiveTracksGeneration++; 1951 mLatestActiveTrack = track; 1952 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1953 if (chain != 0) { 1954 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), 1955 track->sessionId()); 1956 chain->incActiveTrackCnt(); 1957 } 1958 1959 status = NO_ERROR; 1960 } 1961 1962 onAddNewTrack_l(); 1963 return status; 1964} 1965 1966bool AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 1967{ 1968 track->terminate(); 1969 // active tracks are removed by threadLoop() 1970 bool trackActive = (mActiveTracks.indexOf(track) >= 0); 1971 track->mState = TrackBase::STOPPED; 1972 if (!trackActive) { 1973 removeTrack_l(track); 1974 } else if (track->isFastTrack() || track->isOffloaded() || track->isDirect()) { 1975 track->mState = TrackBase::STOPPING_1; 1976 } 1977 1978 return trackActive; 1979} 1980 1981void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 1982{ 1983 track->triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 1984 mTracks.remove(track); 1985 deleteTrackName_l(track->name()); 1986 // redundant as track is about to be destroyed, for dumpsys only 1987 track->mName = -1; 1988 if (track->isFastTrack()) { 1989 int index = track->mFastIndex; 1990 ALOG_ASSERT(0 < index && index < (int)FastMixerState::kMaxFastTracks); 1991 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << index))); 1992 mFastTrackAvailMask |= 1 << index; 1993 // redundant as track is about to be destroyed, for dumpsys only 1994 track->mFastIndex = -1; 1995 } 1996 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1997 if (chain != 0) { 1998 chain->decTrackCnt(); 1999 } 2000} 2001 2002void AudioFlinger::PlaybackThread::broadcast_l() 2003{ 2004 // Thread could be blocked waiting for async 2005 // so signal it to handle state changes immediately 2006 // If threadLoop is currently unlocked a signal of mWaitWorkCV will 2007 // be lost so we also flag to prevent it blocking on mWaitWorkCV 2008 mSignalPending = true; 2009 mWaitWorkCV.broadcast(); 2010} 2011 2012String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 2013{ 2014 Mutex::Autolock _l(mLock); 2015 if (initCheck() != NO_ERROR) { 2016 return String8(); 2017 } 2018 2019 char *s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 2020 const String8 out_s8(s); 2021 free(s); 2022 return out_s8; 2023} 2024 2025void AudioFlinger::PlaybackThread::ioConfigChanged(audio_io_config_event event, pid_t pid) { 2026 sp<AudioIoDescriptor> desc = new AudioIoDescriptor(); 2027 ALOGV("PlaybackThread::ioConfigChanged, thread %p, event %d", this, event); 2028 2029 desc->mIoHandle = mId; 2030 2031 switch (event) { 2032 case AUDIO_OUTPUT_OPENED: 2033 case AUDIO_OUTPUT_CONFIG_CHANGED: 2034 desc->mPatch = mPatch; 2035 desc->mChannelMask = mChannelMask; 2036 desc->mSamplingRate = mSampleRate; 2037 desc->mFormat = mFormat; 2038 desc->mFrameCount = mNormalFrameCount; // FIXME see 2039 // AudioFlinger::frameCount(audio_io_handle_t) 2040 desc->mLatency = latency_l(); 2041 break; 2042 2043 case AUDIO_OUTPUT_CLOSED: 2044 default: 2045 break; 2046 } 2047 mAudioFlinger->ioConfigChanged(event, desc, pid); 2048} 2049 2050void AudioFlinger::PlaybackThread::writeCallback() 2051{ 2052 ALOG_ASSERT(mCallbackThread != 0); 2053 mCallbackThread->resetWriteBlocked(); 2054} 2055 2056void AudioFlinger::PlaybackThread::drainCallback() 2057{ 2058 ALOG_ASSERT(mCallbackThread != 0); 2059 mCallbackThread->resetDraining(); 2060} 2061 2062void AudioFlinger::PlaybackThread::resetWriteBlocked(uint32_t sequence) 2063{ 2064 Mutex::Autolock _l(mLock); 2065 // reject out of sequence requests 2066 if ((mWriteAckSequence & 1) && (sequence == mWriteAckSequence)) { 2067 mWriteAckSequence &= ~1; 2068 mWaitWorkCV.signal(); 2069 } 2070} 2071 2072void AudioFlinger::PlaybackThread::resetDraining(uint32_t sequence) 2073{ 2074 Mutex::Autolock _l(mLock); 2075 // reject out of sequence requests 2076 if ((mDrainSequence & 1) && (sequence == mDrainSequence)) { 2077 mDrainSequence &= ~1; 2078 mWaitWorkCV.signal(); 2079 } 2080} 2081 2082// static 2083int AudioFlinger::PlaybackThread::asyncCallback(stream_callback_event_t event, 2084 void *param __unused, 2085 void *cookie) 2086{ 2087 AudioFlinger::PlaybackThread *me = (AudioFlinger::PlaybackThread *)cookie; 2088 ALOGV("asyncCallback() event %d", event); 2089 switch (event) { 2090 case STREAM_CBK_EVENT_WRITE_READY: 2091 me->writeCallback(); 2092 break; 2093 case STREAM_CBK_EVENT_DRAIN_READY: 2094 me->drainCallback(); 2095 break; 2096 default: 2097 ALOGW("asyncCallback() unknown event %d", event); 2098 break; 2099 } 2100 return 0; 2101} 2102 2103void AudioFlinger::PlaybackThread::readOutputParameters_l() 2104{ 2105 // unfortunately we have no way of recovering from errors here, hence the LOG_ALWAYS_FATAL 2106 mSampleRate = mOutput->getSampleRate(); 2107 mChannelMask = mOutput->getChannelMask(); 2108 if (!audio_is_output_channel(mChannelMask)) { 2109 LOG_ALWAYS_FATAL("HAL channel mask %#x not valid for output", mChannelMask); 2110 } 2111 if ((mType == MIXER || mType == DUPLICATING) 2112 && !isValidPcmSinkChannelMask(mChannelMask)) { 2113 LOG_ALWAYS_FATAL("HAL channel mask %#x not supported for mixed output", 2114 mChannelMask); 2115 } 2116 mChannelCount = audio_channel_count_from_out_mask(mChannelMask); 2117 2118 // Get actual HAL format. 2119 mHALFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 2120 // Get format from the shim, which will be different than the HAL format 2121 // if playing compressed audio over HDMI passthrough. 2122 mFormat = mOutput->getFormat(); 2123 if (!audio_is_valid_format(mFormat)) { 2124 LOG_ALWAYS_FATAL("HAL format %#x not valid for output", mFormat); 2125 } 2126 if ((mType == MIXER || mType == DUPLICATING) 2127 && !isValidPcmSinkFormat(mFormat)) { 2128 LOG_FATAL("HAL format %#x not supported for mixed output", 2129 mFormat); 2130 } 2131 mFrameSize = mOutput->getFrameSize(); 2132 mBufferSize = mOutput->stream->common.get_buffer_size(&mOutput->stream->common); 2133 mFrameCount = mBufferSize / mFrameSize; 2134 if (mFrameCount & 15) { 2135 ALOGW("HAL output buffer size is %u frames but AudioMixer requires multiples of 16 frames", 2136 mFrameCount); 2137 } 2138 2139 if ((mOutput->flags & AUDIO_OUTPUT_FLAG_NON_BLOCKING) && 2140 (mOutput->stream->set_callback != NULL)) { 2141 if (mOutput->stream->set_callback(mOutput->stream, 2142 AudioFlinger::PlaybackThread::asyncCallback, this) == 0) { 2143 mUseAsyncWrite = true; 2144 mCallbackThread = new AudioFlinger::AsyncCallbackThread(this); 2145 } 2146 } 2147 2148 mHwSupportsPause = false; 2149 if (mOutput->flags & AUDIO_OUTPUT_FLAG_DIRECT) { 2150 if (mOutput->stream->pause != NULL) { 2151 if (mOutput->stream->resume != NULL) { 2152 mHwSupportsPause = true; 2153 } else { 2154 ALOGW("direct output implements pause but not resume"); 2155 } 2156 } else if (mOutput->stream->resume != NULL) { 2157 ALOGW("direct output implements resume but not pause"); 2158 } 2159 } 2160 if (!mHwSupportsPause && mOutput->flags & AUDIO_OUTPUT_FLAG_HW_AV_SYNC) { 2161 LOG_ALWAYS_FATAL("HW_AV_SYNC requested but HAL does not implement pause and resume"); 2162 } 2163 2164 if (mType == DUPLICATING && mMixerBufferEnabled && mEffectBufferEnabled) { 2165 // For best precision, we use float instead of the associated output 2166 // device format (typically PCM 16 bit). 2167 2168 mFormat = AUDIO_FORMAT_PCM_FLOAT; 2169 mFrameSize = mChannelCount * audio_bytes_per_sample(mFormat); 2170 mBufferSize = mFrameSize * mFrameCount; 2171 2172 // TODO: We currently use the associated output device channel mask and sample rate. 2173 // (1) Perhaps use the ORed channel mask of all downstream MixerThreads 2174 // (if a valid mask) to avoid premature downmix. 2175 // (2) Perhaps use the maximum sample rate of all downstream MixerThreads 2176 // instead of the output device sample rate to avoid loss of high frequency information. 2177 // This may need to be updated as MixerThread/OutputTracks are added and not here. 2178 } 2179 2180 // Calculate size of normal sink buffer relative to the HAL output buffer size 2181 double multiplier = 1.0; 2182 if (mType == MIXER && (kUseFastMixer == FastMixer_Static || 2183 kUseFastMixer == FastMixer_Dynamic)) { 2184 size_t minNormalFrameCount = (kMinNormalSinkBufferSizeMs * mSampleRate) / 1000; 2185 size_t maxNormalFrameCount = (kMaxNormalSinkBufferSizeMs * mSampleRate) / 1000; 2186 // round up minimum and round down maximum to nearest 16 frames to satisfy AudioMixer 2187 minNormalFrameCount = (minNormalFrameCount + 15) & ~15; 2188 maxNormalFrameCount = maxNormalFrameCount & ~15; 2189 if (maxNormalFrameCount < minNormalFrameCount) { 2190 maxNormalFrameCount = minNormalFrameCount; 2191 } 2192 multiplier = (double) minNormalFrameCount / (double) mFrameCount; 2193 if (multiplier <= 1.0) { 2194 multiplier = 1.0; 2195 } else if (multiplier <= 2.0) { 2196 if (2 * mFrameCount <= maxNormalFrameCount) { 2197 multiplier = 2.0; 2198 } else { 2199 multiplier = (double) maxNormalFrameCount / (double) mFrameCount; 2200 } 2201 } else { 2202 // prefer an even multiplier, for compatibility with doubling of fast tracks due to HAL 2203 // SRC (it would be unusual for the normal sink buffer size to not be a multiple of fast 2204 // track, but we sometimes have to do this to satisfy the maximum frame count 2205 // constraint) 2206 // FIXME this rounding up should not be done if no HAL SRC 2207 uint32_t truncMult = (uint32_t) multiplier; 2208 if ((truncMult & 1)) { 2209 if ((truncMult + 1) * mFrameCount <= maxNormalFrameCount) { 2210 ++truncMult; 2211 } 2212 } 2213 multiplier = (double) truncMult; 2214 } 2215 } 2216 mNormalFrameCount = multiplier * mFrameCount; 2217 // round up to nearest 16 frames to satisfy AudioMixer 2218 if (mType == MIXER || mType == DUPLICATING) { 2219 mNormalFrameCount = (mNormalFrameCount + 15) & ~15; 2220 } 2221 ALOGI("HAL output buffer size %u frames, normal sink buffer size %u frames", mFrameCount, 2222 mNormalFrameCount); 2223 2224 // Check if we want to throttle the processing to no more than 2x normal rate 2225 mThreadThrottle = property_get_bool("af.thread.throttle", true /* default_value */); 2226 mThreadThrottleTimeMs = 0; 2227 mThreadThrottleEndMs = 0; 2228 mHalfBufferMs = mNormalFrameCount * 1000 / (2 * mSampleRate); 2229 2230 // mSinkBuffer is the sink buffer. Size is always multiple-of-16 frames. 2231 // Originally this was int16_t[] array, need to remove legacy implications. 2232 free(mSinkBuffer); 2233 mSinkBuffer = NULL; 2234 // For sink buffer size, we use the frame size from the downstream sink to avoid problems 2235 // with non PCM formats for compressed music, e.g. AAC, and Offload threads. 2236 const size_t sinkBufferSize = mNormalFrameCount * mFrameSize; 2237 (void)posix_memalign(&mSinkBuffer, 32, sinkBufferSize); 2238 2239 // We resize the mMixerBuffer according to the requirements of the sink buffer which 2240 // drives the output. 2241 free(mMixerBuffer); 2242 mMixerBuffer = NULL; 2243 if (mMixerBufferEnabled) { 2244 mMixerBufferFormat = AUDIO_FORMAT_PCM_FLOAT; // also valid: AUDIO_FORMAT_PCM_16_BIT. 2245 mMixerBufferSize = mNormalFrameCount * mChannelCount 2246 * audio_bytes_per_sample(mMixerBufferFormat); 2247 (void)posix_memalign(&mMixerBuffer, 32, mMixerBufferSize); 2248 } 2249 free(mEffectBuffer); 2250 mEffectBuffer = NULL; 2251 if (mEffectBufferEnabled) { 2252 mEffectBufferFormat = AUDIO_FORMAT_PCM_16_BIT; // Note: Effects support 16b only 2253 mEffectBufferSize = mNormalFrameCount * mChannelCount 2254 * audio_bytes_per_sample(mEffectBufferFormat); 2255 (void)posix_memalign(&mEffectBuffer, 32, mEffectBufferSize); 2256 } 2257 2258 // force reconfiguration of effect chains and engines to take new buffer size and audio 2259 // parameters into account 2260 // Note that mLock is not held when readOutputParameters_l() is called from the constructor 2261 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 2262 // matter. 2263 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 2264 Vector< sp<EffectChain> > effectChains = mEffectChains; 2265 for (size_t i = 0; i < effectChains.size(); i ++) { 2266 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 2267 } 2268} 2269 2270 2271status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames) 2272{ 2273 if (halFrames == NULL || dspFrames == NULL) { 2274 return BAD_VALUE; 2275 } 2276 Mutex::Autolock _l(mLock); 2277 if (initCheck() != NO_ERROR) { 2278 return INVALID_OPERATION; 2279 } 2280 size_t framesWritten = mBytesWritten / mFrameSize; 2281 *halFrames = framesWritten; 2282 2283 if (isSuspended()) { 2284 // return an estimation of rendered frames when the output is suspended 2285 size_t latencyFrames = (latency_l() * mSampleRate) / 1000; 2286 *dspFrames = framesWritten >= latencyFrames ? framesWritten - latencyFrames : 0; 2287 return NO_ERROR; 2288 } else { 2289 status_t status; 2290 uint32_t frames; 2291 status = mOutput->getRenderPosition(&frames); 2292 *dspFrames = (size_t)frames; 2293 return status; 2294 } 2295} 2296 2297uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) const 2298{ 2299 Mutex::Autolock _l(mLock); 2300 uint32_t result = 0; 2301 if (getEffectChain_l(sessionId) != 0) { 2302 result = EFFECT_SESSION; 2303 } 2304 2305 for (size_t i = 0; i < mTracks.size(); ++i) { 2306 sp<Track> track = mTracks[i]; 2307 if (sessionId == track->sessionId() && !track->isInvalid()) { 2308 result |= TRACK_SESSION; 2309 break; 2310 } 2311 } 2312 2313 return result; 2314} 2315 2316uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 2317{ 2318 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 2319 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 2320 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 2321 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 2322 } 2323 for (size_t i = 0; i < mTracks.size(); i++) { 2324 sp<Track> track = mTracks[i]; 2325 if (sessionId == track->sessionId() && !track->isInvalid()) { 2326 return AudioSystem::getStrategyForStream(track->streamType()); 2327 } 2328 } 2329 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 2330} 2331 2332 2333AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 2334{ 2335 Mutex::Autolock _l(mLock); 2336 return mOutput; 2337} 2338 2339AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 2340{ 2341 Mutex::Autolock _l(mLock); 2342 AudioStreamOut *output = mOutput; 2343 mOutput = NULL; 2344 // FIXME FastMixer might also have a raw ptr to mOutputSink; 2345 // must push a NULL and wait for ack 2346 mOutputSink.clear(); 2347 mPipeSink.clear(); 2348 mNormalSink.clear(); 2349 return output; 2350} 2351 2352// this method must always be called either with ThreadBase mLock held or inside the thread loop 2353audio_stream_t* AudioFlinger::PlaybackThread::stream() const 2354{ 2355 if (mOutput == NULL) { 2356 return NULL; 2357 } 2358 return &mOutput->stream->common; 2359} 2360 2361uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() const 2362{ 2363 return (uint32_t)((uint32_t)((mNormalFrameCount * 1000) / mSampleRate) * 1000); 2364} 2365 2366status_t AudioFlinger::PlaybackThread::setSyncEvent(const sp<SyncEvent>& event) 2367{ 2368 if (!isValidSyncEvent(event)) { 2369 return BAD_VALUE; 2370 } 2371 2372 Mutex::Autolock _l(mLock); 2373 2374 for (size_t i = 0; i < mTracks.size(); ++i) { 2375 sp<Track> track = mTracks[i]; 2376 if (event->triggerSession() == track->sessionId()) { 2377 (void) track->setSyncEvent(event); 2378 return NO_ERROR; 2379 } 2380 } 2381 2382 return NAME_NOT_FOUND; 2383} 2384 2385bool AudioFlinger::PlaybackThread::isValidSyncEvent(const sp<SyncEvent>& event) const 2386{ 2387 return event->type() == AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE; 2388} 2389 2390void AudioFlinger::PlaybackThread::threadLoop_removeTracks( 2391 const Vector< sp<Track> >& tracksToRemove) 2392{ 2393 size_t count = tracksToRemove.size(); 2394 if (count > 0) { 2395 for (size_t i = 0 ; i < count ; i++) { 2396 const sp<Track>& track = tracksToRemove.itemAt(i); 2397 if (track->isExternalTrack()) { 2398 AudioSystem::stopOutput(mId, track->streamType(), 2399 (audio_session_t)track->sessionId()); 2400#ifdef ADD_BATTERY_DATA 2401 // to track the speaker usage 2402 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 2403#endif 2404 if (track->isTerminated()) { 2405 AudioSystem::releaseOutput(mId, track->streamType(), 2406 (audio_session_t)track->sessionId()); 2407 } 2408 } 2409 } 2410 } 2411} 2412 2413void AudioFlinger::PlaybackThread::checkSilentMode_l() 2414{ 2415 if (!mMasterMute) { 2416 char value[PROPERTY_VALUE_MAX]; 2417 if (property_get("ro.audio.silent", value, "0") > 0) { 2418 char *endptr; 2419 unsigned long ul = strtoul(value, &endptr, 0); 2420 if (*endptr == '\0' && ul != 0) { 2421 ALOGD("Silence is golden"); 2422 // The setprop command will not allow a property to be changed after 2423 // the first time it is set, so we don't have to worry about un-muting. 2424 setMasterMute_l(true); 2425 } 2426 } 2427 } 2428} 2429 2430// shared by MIXER and DIRECT, overridden by DUPLICATING 2431ssize_t AudioFlinger::PlaybackThread::threadLoop_write() 2432{ 2433 // FIXME rewrite to reduce number of system calls 2434 mLastWriteTime = systemTime(); 2435 mInWrite = true; 2436 ssize_t bytesWritten; 2437 const size_t offset = mCurrentWriteLength - mBytesRemaining; 2438 2439 // If an NBAIO sink is present, use it to write the normal mixer's submix 2440 if (mNormalSink != 0) { 2441 2442 const size_t count = mBytesRemaining / mFrameSize; 2443 2444 ATRACE_BEGIN("write"); 2445 // update the setpoint when AudioFlinger::mScreenState changes 2446 uint32_t screenState = AudioFlinger::mScreenState; 2447 if (screenState != mScreenState) { 2448 mScreenState = screenState; 2449 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 2450 if (pipe != NULL) { 2451 pipe->setAvgFrames((mScreenState & 1) ? 2452 (pipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2453 } 2454 } 2455 ssize_t framesWritten = mNormalSink->write((char *)mSinkBuffer + offset, count); 2456 ATRACE_END(); 2457 if (framesWritten > 0) { 2458 bytesWritten = framesWritten * mFrameSize; 2459 } else { 2460 bytesWritten = framesWritten; 2461 } 2462 mLatchDValid = false; 2463 status_t status = mNormalSink->getTimestamp(mLatchD.mTimestamp); 2464 if (status == NO_ERROR) { 2465 size_t totalFramesWritten = mNormalSink->framesWritten(); 2466 if (totalFramesWritten >= mLatchD.mTimestamp.mPosition) { 2467 mLatchD.mUnpresentedFrames = totalFramesWritten - mLatchD.mTimestamp.mPosition; 2468 // mLatchD.mFramesReleased is set immediately before D is clocked into Q 2469 mLatchDValid = true; 2470 } 2471 } 2472 // otherwise use the HAL / AudioStreamOut directly 2473 } else { 2474 // Direct output and offload threads 2475 2476 if (mUseAsyncWrite) { 2477 ALOGW_IF(mWriteAckSequence & 1, "threadLoop_write(): out of sequence write request"); 2478 mWriteAckSequence += 2; 2479 mWriteAckSequence |= 1; 2480 ALOG_ASSERT(mCallbackThread != 0); 2481 mCallbackThread->setWriteBlocked(mWriteAckSequence); 2482 } 2483 // FIXME We should have an implementation of timestamps for direct output threads. 2484 // They are used e.g for multichannel PCM playback over HDMI. 2485 bytesWritten = mOutput->write((char *)mSinkBuffer + offset, mBytesRemaining); 2486 if (mUseAsyncWrite && 2487 ((bytesWritten < 0) || (bytesWritten == (ssize_t)mBytesRemaining))) { 2488 // do not wait for async callback in case of error of full write 2489 mWriteAckSequence &= ~1; 2490 ALOG_ASSERT(mCallbackThread != 0); 2491 mCallbackThread->setWriteBlocked(mWriteAckSequence); 2492 } 2493 } 2494 2495 mNumWrites++; 2496 mInWrite = false; 2497 mStandby = false; 2498 return bytesWritten; 2499} 2500 2501void AudioFlinger::PlaybackThread::threadLoop_drain() 2502{ 2503 if (mOutput->stream->drain) { 2504 ALOGV("draining %s", (mMixerStatus == MIXER_DRAIN_TRACK) ? "early" : "full"); 2505 if (mUseAsyncWrite) { 2506 ALOGW_IF(mDrainSequence & 1, "threadLoop_drain(): out of sequence drain request"); 2507 mDrainSequence |= 1; 2508 ALOG_ASSERT(mCallbackThread != 0); 2509 mCallbackThread->setDraining(mDrainSequence); 2510 } 2511 mOutput->stream->drain(mOutput->stream, 2512 (mMixerStatus == MIXER_DRAIN_TRACK) ? AUDIO_DRAIN_EARLY_NOTIFY 2513 : AUDIO_DRAIN_ALL); 2514 } 2515} 2516 2517void AudioFlinger::PlaybackThread::threadLoop_exit() 2518{ 2519 { 2520 Mutex::Autolock _l(mLock); 2521 for (size_t i = 0; i < mTracks.size(); i++) { 2522 sp<Track> track = mTracks[i]; 2523 track->invalidate(); 2524 } 2525 } 2526} 2527 2528/* 2529The derived values that are cached: 2530 - mSinkBufferSize from frame count * frame size 2531 - mActiveSleepTimeUs from activeSleepTimeUs() 2532 - mIdleSleepTimeUs from idleSleepTimeUs() 2533 - mStandbyDelayNs from mActiveSleepTimeUs (DIRECT only) 2534 - maxPeriod from frame count and sample rate (MIXER only) 2535 2536The parameters that affect these derived values are: 2537 - frame count 2538 - frame size 2539 - sample rate 2540 - device type: A2DP or not 2541 - device latency 2542 - format: PCM or not 2543 - active sleep time 2544 - idle sleep time 2545*/ 2546 2547void AudioFlinger::PlaybackThread::cacheParameters_l() 2548{ 2549 mSinkBufferSize = mNormalFrameCount * mFrameSize; 2550 mActiveSleepTimeUs = activeSleepTimeUs(); 2551 mIdleSleepTimeUs = idleSleepTimeUs(); 2552} 2553 2554void AudioFlinger::PlaybackThread::invalidateTracks(audio_stream_type_t streamType) 2555{ 2556 ALOGV("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 2557 this, streamType, mTracks.size()); 2558 Mutex::Autolock _l(mLock); 2559 2560 size_t size = mTracks.size(); 2561 for (size_t i = 0; i < size; i++) { 2562 sp<Track> t = mTracks[i]; 2563 if (t->streamType() == streamType && t->isExternalTrack()) { 2564 t->invalidate(); 2565 } 2566 } 2567} 2568 2569status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 2570{ 2571 int session = chain->sessionId(); 2572 int16_t* buffer = reinterpret_cast<int16_t*>(mEffectBufferEnabled 2573 ? mEffectBuffer : mSinkBuffer); 2574 bool ownsBuffer = false; 2575 2576 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 2577 if (session > 0) { 2578 // Only one effect chain can be present in direct output thread and it uses 2579 // the sink buffer as input 2580 if (mType != DIRECT) { 2581 size_t numSamples = mNormalFrameCount * mChannelCount; 2582 buffer = new int16_t[numSamples]; 2583 memset(buffer, 0, numSamples * sizeof(int16_t)); 2584 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 2585 ownsBuffer = true; 2586 } 2587 2588 // Attach all tracks with same session ID to this chain. 2589 for (size_t i = 0; i < mTracks.size(); ++i) { 2590 sp<Track> track = mTracks[i]; 2591 if (session == track->sessionId()) { 2592 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), 2593 buffer); 2594 track->setMainBuffer(buffer); 2595 chain->incTrackCnt(); 2596 } 2597 } 2598 2599 // indicate all active tracks in the chain 2600 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 2601 sp<Track> track = mActiveTracks[i].promote(); 2602 if (track == 0) { 2603 continue; 2604 } 2605 if (session == track->sessionId()) { 2606 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 2607 chain->incActiveTrackCnt(); 2608 } 2609 } 2610 } 2611 chain->setThread(this); 2612 chain->setInBuffer(buffer, ownsBuffer); 2613 chain->setOutBuffer(reinterpret_cast<int16_t*>(mEffectBufferEnabled 2614 ? mEffectBuffer : mSinkBuffer)); 2615 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 2616 // chains list in order to be processed last as it contains output stage effects 2617 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 2618 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 2619 // after track specific effects and before output stage 2620 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 2621 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 2622 // Effect chain for other sessions are inserted at beginning of effect 2623 // chains list to be processed before output mix effects. Relative order between other 2624 // sessions is not important 2625 size_t size = mEffectChains.size(); 2626 size_t i = 0; 2627 for (i = 0; i < size; i++) { 2628 if (mEffectChains[i]->sessionId() < session) { 2629 break; 2630 } 2631 } 2632 mEffectChains.insertAt(chain, i); 2633 checkSuspendOnAddEffectChain_l(chain); 2634 2635 return NO_ERROR; 2636} 2637 2638size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 2639{ 2640 int session = chain->sessionId(); 2641 2642 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 2643 2644 for (size_t i = 0; i < mEffectChains.size(); i++) { 2645 if (chain == mEffectChains[i]) { 2646 mEffectChains.removeAt(i); 2647 // detach all active tracks from the chain 2648 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 2649 sp<Track> track = mActiveTracks[i].promote(); 2650 if (track == 0) { 2651 continue; 2652 } 2653 if (session == track->sessionId()) { 2654 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 2655 chain.get(), session); 2656 chain->decActiveTrackCnt(); 2657 } 2658 } 2659 2660 // detach all tracks with same session ID from this chain 2661 for (size_t i = 0; i < mTracks.size(); ++i) { 2662 sp<Track> track = mTracks[i]; 2663 if (session == track->sessionId()) { 2664 track->setMainBuffer(reinterpret_cast<int16_t*>(mSinkBuffer)); 2665 chain->decTrackCnt(); 2666 } 2667 } 2668 break; 2669 } 2670 } 2671 return mEffectChains.size(); 2672} 2673 2674status_t AudioFlinger::PlaybackThread::attachAuxEffect( 2675 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2676{ 2677 Mutex::Autolock _l(mLock); 2678 return attachAuxEffect_l(track, EffectId); 2679} 2680 2681status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 2682 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2683{ 2684 status_t status = NO_ERROR; 2685 2686 if (EffectId == 0) { 2687 track->setAuxBuffer(0, NULL); 2688 } else { 2689 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 2690 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 2691 if (effect != 0) { 2692 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 2693 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 2694 } else { 2695 status = INVALID_OPERATION; 2696 } 2697 } else { 2698 status = BAD_VALUE; 2699 } 2700 } 2701 return status; 2702} 2703 2704void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 2705{ 2706 for (size_t i = 0; i < mTracks.size(); ++i) { 2707 sp<Track> track = mTracks[i]; 2708 if (track->auxEffectId() == effectId) { 2709 attachAuxEffect_l(track, 0); 2710 } 2711 } 2712} 2713 2714bool AudioFlinger::PlaybackThread::threadLoop() 2715{ 2716 Vector< sp<Track> > tracksToRemove; 2717 2718 mStandbyTimeNs = systemTime(); 2719 2720 // MIXER 2721 nsecs_t lastWarning = 0; 2722 2723 // DUPLICATING 2724 // FIXME could this be made local to while loop? 2725 writeFrames = 0; 2726 2727 int lastGeneration = 0; 2728 2729 cacheParameters_l(); 2730 mSleepTimeUs = mIdleSleepTimeUs; 2731 2732 if (mType == MIXER) { 2733 sleepTimeShift = 0; 2734 } 2735 2736 CpuStats cpuStats; 2737 const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid())); 2738 2739 acquireWakeLock(); 2740 2741 // mNBLogWriter->log can only be called while thread mutex mLock is held. 2742 // So if you need to log when mutex is unlocked, set logString to a non-NULL string, 2743 // and then that string will be logged at the next convenient opportunity. 2744 const char *logString = NULL; 2745 2746 checkSilentMode_l(); 2747 2748 while (!exitPending()) 2749 { 2750 cpuStats.sample(myName); 2751 2752 Vector< sp<EffectChain> > effectChains; 2753 2754 { // scope for mLock 2755 2756 Mutex::Autolock _l(mLock); 2757 2758 processConfigEvents_l(); 2759 2760 if (logString != NULL) { 2761 mNBLogWriter->logTimestamp(); 2762 mNBLogWriter->log(logString); 2763 logString = NULL; 2764 } 2765 2766 // Gather the framesReleased counters for all active tracks, 2767 // and latch them atomically with the timestamp. 2768 // FIXME We're using raw pointers as indices. A unique track ID would be a better index. 2769 mLatchD.mFramesReleased.clear(); 2770 size_t size = mActiveTracks.size(); 2771 for (size_t i = 0; i < size; i++) { 2772 sp<Track> t = mActiveTracks[i].promote(); 2773 if (t != 0) { 2774 mLatchD.mFramesReleased.add(t.get(), 2775 t->mAudioTrackServerProxy->framesReleased()); 2776 } 2777 } 2778 if (mLatchDValid) { 2779 mLatchQ = mLatchD; 2780 mLatchDValid = false; 2781 mLatchQValid = true; 2782 } 2783 2784 saveOutputTracks(); 2785 if (mSignalPending) { 2786 // A signal was raised while we were unlocked 2787 mSignalPending = false; 2788 } else if (waitingAsyncCallback_l()) { 2789 if (exitPending()) { 2790 break; 2791 } 2792 bool released = false; 2793 // The following works around a bug in the offload driver. Ideally we would release 2794 // the wake lock every time, but that causes the last offload buffer(s) to be 2795 // dropped while the device is on battery, so we need to hold a wake lock during 2796 // the drain phase. 2797 if (mBytesRemaining && !(mDrainSequence & 1)) { 2798 releaseWakeLock_l(); 2799 released = true; 2800 } 2801 mWakeLockUids.clear(); 2802 mActiveTracksGeneration++; 2803 ALOGV("wait async completion"); 2804 mWaitWorkCV.wait(mLock); 2805 ALOGV("async completion/wake"); 2806 if (released) { 2807 acquireWakeLock_l(); 2808 } 2809 mStandbyTimeNs = systemTime() + mStandbyDelayNs; 2810 mSleepTimeUs = 0; 2811 2812 continue; 2813 } 2814 if ((!mActiveTracks.size() && systemTime() > mStandbyTimeNs) || 2815 isSuspended()) { 2816 // put audio hardware into standby after short delay 2817 if (shouldStandby_l()) { 2818 2819 threadLoop_standby(); 2820 2821 mStandby = true; 2822 } 2823 2824 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2825 // we're about to wait, flush the binder command buffer 2826 IPCThreadState::self()->flushCommands(); 2827 2828 clearOutputTracks(); 2829 2830 if (exitPending()) { 2831 break; 2832 } 2833 2834 releaseWakeLock_l(); 2835 mWakeLockUids.clear(); 2836 mActiveTracksGeneration++; 2837 // wait until we have something to do... 2838 ALOGV("%s going to sleep", myName.string()); 2839 mWaitWorkCV.wait(mLock); 2840 ALOGV("%s waking up", myName.string()); 2841 acquireWakeLock_l(); 2842 2843 mMixerStatus = MIXER_IDLE; 2844 mMixerStatusIgnoringFastTracks = MIXER_IDLE; 2845 mBytesWritten = 0; 2846 mBytesRemaining = 0; 2847 checkSilentMode_l(); 2848 2849 mStandbyTimeNs = systemTime() + mStandbyDelayNs; 2850 mSleepTimeUs = mIdleSleepTimeUs; 2851 if (mType == MIXER) { 2852 sleepTimeShift = 0; 2853 } 2854 2855 continue; 2856 } 2857 } 2858 // mMixerStatusIgnoringFastTracks is also updated internally 2859 mMixerStatus = prepareTracks_l(&tracksToRemove); 2860 2861 // compare with previously applied list 2862 if (lastGeneration != mActiveTracksGeneration) { 2863 // update wakelock 2864 updateWakeLockUids_l(mWakeLockUids); 2865 lastGeneration = mActiveTracksGeneration; 2866 } 2867 2868 // prevent any changes in effect chain list and in each effect chain 2869 // during mixing and effect process as the audio buffers could be deleted 2870 // or modified if an effect is created or deleted 2871 lockEffectChains_l(effectChains); 2872 } // mLock scope ends 2873 2874 if (mBytesRemaining == 0) { 2875 mCurrentWriteLength = 0; 2876 if (mMixerStatus == MIXER_TRACKS_READY) { 2877 // threadLoop_mix() sets mCurrentWriteLength 2878 threadLoop_mix(); 2879 } else if ((mMixerStatus != MIXER_DRAIN_TRACK) 2880 && (mMixerStatus != MIXER_DRAIN_ALL)) { 2881 // threadLoop_sleepTime sets mSleepTimeUs to 0 if data 2882 // must be written to HAL 2883 threadLoop_sleepTime(); 2884 if (mSleepTimeUs == 0) { 2885 mCurrentWriteLength = mSinkBufferSize; 2886 } 2887 } 2888 // Either threadLoop_mix() or threadLoop_sleepTime() should have set 2889 // mMixerBuffer with data if mMixerBufferValid is true and mSleepTimeUs == 0. 2890 // Merge mMixerBuffer data into mEffectBuffer (if any effects are valid) 2891 // or mSinkBuffer (if there are no effects). 2892 // 2893 // This is done pre-effects computation; if effects change to 2894 // support higher precision, this needs to move. 2895 // 2896 // mMixerBufferValid is only set true by MixerThread::prepareTracks_l(). 2897 // TODO use mSleepTimeUs == 0 as an additional condition. 2898 if (mMixerBufferValid) { 2899 void *buffer = mEffectBufferValid ? mEffectBuffer : mSinkBuffer; 2900 audio_format_t format = mEffectBufferValid ? mEffectBufferFormat : mFormat; 2901 2902 memcpy_by_audio_format(buffer, format, mMixerBuffer, mMixerBufferFormat, 2903 mNormalFrameCount * mChannelCount); 2904 } 2905 2906 mBytesRemaining = mCurrentWriteLength; 2907 if (isSuspended()) { 2908 mSleepTimeUs = suspendSleepTimeUs(); 2909 // simulate write to HAL when suspended 2910 mBytesWritten += mSinkBufferSize; 2911 mBytesRemaining = 0; 2912 } 2913 2914 // only process effects if we're going to write 2915 if (mSleepTimeUs == 0 && mType != OFFLOAD) { 2916 for (size_t i = 0; i < effectChains.size(); i ++) { 2917 effectChains[i]->process_l(); 2918 } 2919 } 2920 } 2921 // Process effect chains for offloaded thread even if no audio 2922 // was read from audio track: process only updates effect state 2923 // and thus does have to be synchronized with audio writes but may have 2924 // to be called while waiting for async write callback 2925 if (mType == OFFLOAD) { 2926 for (size_t i = 0; i < effectChains.size(); i ++) { 2927 effectChains[i]->process_l(); 2928 } 2929 } 2930 2931 // Only if the Effects buffer is enabled and there is data in the 2932 // Effects buffer (buffer valid), we need to 2933 // copy into the sink buffer. 2934 // TODO use mSleepTimeUs == 0 as an additional condition. 2935 if (mEffectBufferValid) { 2936 //ALOGV("writing effect buffer to sink buffer format %#x", mFormat); 2937 memcpy_by_audio_format(mSinkBuffer, mFormat, mEffectBuffer, mEffectBufferFormat, 2938 mNormalFrameCount * mChannelCount); 2939 } 2940 2941 // enable changes in effect chain 2942 unlockEffectChains(effectChains); 2943 2944 if (!waitingAsyncCallback()) { 2945 // mSleepTimeUs == 0 means we must write to audio hardware 2946 if (mSleepTimeUs == 0) { 2947 ssize_t ret = 0; 2948 if (mBytesRemaining) { 2949 ret = threadLoop_write(); 2950 if (ret < 0) { 2951 mBytesRemaining = 0; 2952 } else { 2953 mBytesWritten += ret; 2954 mBytesRemaining -= ret; 2955 } 2956 } else if ((mMixerStatus == MIXER_DRAIN_TRACK) || 2957 (mMixerStatus == MIXER_DRAIN_ALL)) { 2958 threadLoop_drain(); 2959 } 2960 if (mType == MIXER && !mStandby) { 2961 // write blocked detection 2962 nsecs_t now = systemTime(); 2963 nsecs_t delta = now - mLastWriteTime; 2964 if (delta > maxPeriod) { 2965 mNumDelayedWrites++; 2966 if ((now - lastWarning) > kWarningThrottleNs) { 2967 ATRACE_NAME("underrun"); 2968 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2969 ns2ms(delta), mNumDelayedWrites, this); 2970 lastWarning = now; 2971 } 2972 } 2973 2974 if (mThreadThrottle 2975 && mMixerStatus == MIXER_TRACKS_READY // we are mixing (active tracks) 2976 && ret > 0) { // we wrote something 2977 // Limit MixerThread data processing to no more than twice the 2978 // expected processing rate. 2979 // 2980 // This helps prevent underruns with NuPlayer and other applications 2981 // which may set up buffers that are close to the minimum size, or use 2982 // deep buffers, and rely on a double-buffering sleep strategy to fill. 2983 // 2984 // The throttle smooths out sudden large data drains from the device, 2985 // e.g. when it comes out of standby, which often causes problems with 2986 // (1) mixer threads without a fast mixer (which has its own warm-up) 2987 // (2) minimum buffer sized tracks (even if the track is full, 2988 // the app won't fill fast enough to handle the sudden draw). 2989 2990 const int32_t deltaMs = delta / 1000000; 2991 const int32_t throttleMs = mHalfBufferMs - deltaMs; 2992 if ((signed)mHalfBufferMs >= throttleMs && throttleMs > 0) { 2993 usleep(throttleMs * 1000); 2994 // notify of throttle start on verbose log 2995 ALOGV_IF(mThreadThrottleEndMs == mThreadThrottleTimeMs, 2996 "mixer(%p) throttle begin:" 2997 " ret(%zd) deltaMs(%d) requires sleep %d ms", 2998 this, ret, deltaMs, throttleMs); 2999 mThreadThrottleTimeMs += throttleMs; 3000 } else { 3001 uint32_t diff = mThreadThrottleTimeMs - mThreadThrottleEndMs; 3002 if (diff > 0) { 3003 // notify of throttle end on debug log 3004 ALOGD("mixer(%p) throttle end: throttle time(%u)", this, diff); 3005 mThreadThrottleEndMs = mThreadThrottleTimeMs; 3006 } 3007 } 3008 } 3009 } 3010 3011 } else { 3012 ATRACE_BEGIN("sleep"); 3013 usleep(mSleepTimeUs); 3014 ATRACE_END(); 3015 } 3016 } 3017 3018 // Finally let go of removed track(s), without the lock held 3019 // since we can't guarantee the destructors won't acquire that 3020 // same lock. This will also mutate and push a new fast mixer state. 3021 threadLoop_removeTracks(tracksToRemove); 3022 tracksToRemove.clear(); 3023 3024 // FIXME I don't understand the need for this here; 3025 // it was in the original code but maybe the 3026 // assignment in saveOutputTracks() makes this unnecessary? 3027 clearOutputTracks(); 3028 3029 // Effect chains will be actually deleted here if they were removed from 3030 // mEffectChains list during mixing or effects processing 3031 effectChains.clear(); 3032 3033 // FIXME Note that the above .clear() is no longer necessary since effectChains 3034 // is now local to this block, but will keep it for now (at least until merge done). 3035 } 3036 3037 threadLoop_exit(); 3038 3039 if (!mStandby) { 3040 threadLoop_standby(); 3041 mStandby = true; 3042 } 3043 3044 releaseWakeLock(); 3045 mWakeLockUids.clear(); 3046 mActiveTracksGeneration++; 3047 3048 ALOGV("Thread %p type %d exiting", this, mType); 3049 return false; 3050} 3051 3052// removeTracks_l() must be called with ThreadBase::mLock held 3053void AudioFlinger::PlaybackThread::removeTracks_l(const Vector< sp<Track> >& tracksToRemove) 3054{ 3055 size_t count = tracksToRemove.size(); 3056 if (count > 0) { 3057 for (size_t i=0 ; i<count ; i++) { 3058 const sp<Track>& track = tracksToRemove.itemAt(i); 3059 mActiveTracks.remove(track); 3060 mWakeLockUids.remove(track->uid()); 3061 mActiveTracksGeneration++; 3062 ALOGV("removeTracks_l removing track on session %d", track->sessionId()); 3063 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 3064 if (chain != 0) { 3065 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), 3066 track->sessionId()); 3067 chain->decActiveTrackCnt(); 3068 } 3069 if (track->isTerminated()) { 3070 removeTrack_l(track); 3071 } 3072 } 3073 } 3074 3075} 3076 3077status_t AudioFlinger::PlaybackThread::getTimestamp_l(AudioTimestamp& timestamp) 3078{ 3079 if (mNormalSink != 0) { 3080 return mNormalSink->getTimestamp(timestamp); 3081 } 3082 if ((mType == OFFLOAD || mType == DIRECT) 3083 && mOutput != NULL && mOutput->stream->get_presentation_position) { 3084 uint64_t position64; 3085 int ret = mOutput->getPresentationPosition(&position64, ×tamp.mTime); 3086 if (ret == 0) { 3087 timestamp.mPosition = (uint32_t)position64; 3088 return NO_ERROR; 3089 } 3090 } 3091 return INVALID_OPERATION; 3092} 3093 3094status_t AudioFlinger::MixerThread::createAudioPatch_l(const struct audio_patch *patch, 3095 audio_patch_handle_t *handle) 3096{ 3097 // if !&IDLE, holds the FastMixer state to restore after new parameters processed 3098 FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE; 3099 if (mFastMixer != 0) { 3100 FastMixerStateQueue *sq = mFastMixer->sq(); 3101 FastMixerState *state = sq->begin(); 3102 if (!(state->mCommand & FastMixerState::IDLE)) { 3103 previousCommand = state->mCommand; 3104 state->mCommand = FastMixerState::HOT_IDLE; 3105 sq->end(); 3106 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3107 } else { 3108 sq->end(false /*didModify*/); 3109 } 3110 } 3111 status_t status = PlaybackThread::createAudioPatch_l(patch, handle); 3112 3113 if (!(previousCommand & FastMixerState::IDLE)) { 3114 ALOG_ASSERT(mFastMixer != 0); 3115 FastMixerStateQueue *sq = mFastMixer->sq(); 3116 FastMixerState *state = sq->begin(); 3117 ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE); 3118 state->mCommand = previousCommand; 3119 sq->end(); 3120 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3121 } 3122 3123 return status; 3124} 3125 3126status_t AudioFlinger::PlaybackThread::createAudioPatch_l(const struct audio_patch *patch, 3127 audio_patch_handle_t *handle) 3128{ 3129 status_t status = NO_ERROR; 3130 3131 // store new device and send to effects 3132 audio_devices_t type = AUDIO_DEVICE_NONE; 3133 for (unsigned int i = 0; i < patch->num_sinks; i++) { 3134 type |= patch->sinks[i].ext.device.type; 3135 } 3136 3137#ifdef ADD_BATTERY_DATA 3138 // when changing the audio output device, call addBatteryData to notify 3139 // the change 3140 if (mOutDevice != type) { 3141 uint32_t params = 0; 3142 // check whether speaker is on 3143 if (type & AUDIO_DEVICE_OUT_SPEAKER) { 3144 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 3145 } 3146 3147 audio_devices_t deviceWithoutSpeaker 3148 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 3149 // check if any other device (except speaker) is on 3150 if (type & deviceWithoutSpeaker) { 3151 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 3152 } 3153 3154 if (params != 0) { 3155 addBatteryData(params); 3156 } 3157 } 3158#endif 3159 3160 for (size_t i = 0; i < mEffectChains.size(); i++) { 3161 mEffectChains[i]->setDevice_l(type); 3162 } 3163 3164 // mPrevOutDevice is the latest device set by createAudioPatch_l(). It is not set when 3165 // the thread is created so that the first patch creation triggers an ioConfigChanged callback 3166 bool configChanged = mPrevOutDevice != type; 3167 mOutDevice = type; 3168 mPatch = *patch; 3169 3170 if (mOutput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 3171 audio_hw_device_t *hwDevice = mOutput->audioHwDev->hwDevice(); 3172 status = hwDevice->create_audio_patch(hwDevice, 3173 patch->num_sources, 3174 patch->sources, 3175 patch->num_sinks, 3176 patch->sinks, 3177 handle); 3178 } else { 3179 char *address; 3180 if (strcmp(patch->sinks[0].ext.device.address, "") != 0) { 3181 //FIXME: we only support address on first sink with HAL version < 3.0 3182 address = audio_device_address_to_parameter( 3183 patch->sinks[0].ext.device.type, 3184 patch->sinks[0].ext.device.address); 3185 } else { 3186 address = (char *)calloc(1, 1); 3187 } 3188 AudioParameter param = AudioParameter(String8(address)); 3189 free(address); 3190 param.addInt(String8(AUDIO_PARAMETER_STREAM_ROUTING), (int)type); 3191 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3192 param.toString().string()); 3193 *handle = AUDIO_PATCH_HANDLE_NONE; 3194 } 3195 if (configChanged) { 3196 mPrevOutDevice = type; 3197 sendIoConfigEvent_l(AUDIO_OUTPUT_CONFIG_CHANGED); 3198 } 3199 return status; 3200} 3201 3202status_t AudioFlinger::MixerThread::releaseAudioPatch_l(const audio_patch_handle_t handle) 3203{ 3204 // if !&IDLE, holds the FastMixer state to restore after new parameters processed 3205 FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE; 3206 if (mFastMixer != 0) { 3207 FastMixerStateQueue *sq = mFastMixer->sq(); 3208 FastMixerState *state = sq->begin(); 3209 if (!(state->mCommand & FastMixerState::IDLE)) { 3210 previousCommand = state->mCommand; 3211 state->mCommand = FastMixerState::HOT_IDLE; 3212 sq->end(); 3213 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3214 } else { 3215 sq->end(false /*didModify*/); 3216 } 3217 } 3218 3219 status_t status = PlaybackThread::releaseAudioPatch_l(handle); 3220 3221 if (!(previousCommand & FastMixerState::IDLE)) { 3222 ALOG_ASSERT(mFastMixer != 0); 3223 FastMixerStateQueue *sq = mFastMixer->sq(); 3224 FastMixerState *state = sq->begin(); 3225 ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE); 3226 state->mCommand = previousCommand; 3227 sq->end(); 3228 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3229 } 3230 3231 return status; 3232} 3233 3234status_t AudioFlinger::PlaybackThread::releaseAudioPatch_l(const audio_patch_handle_t handle) 3235{ 3236 status_t status = NO_ERROR; 3237 3238 mOutDevice = AUDIO_DEVICE_NONE; 3239 3240 if (mOutput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 3241 audio_hw_device_t *hwDevice = mOutput->audioHwDev->hwDevice(); 3242 status = hwDevice->release_audio_patch(hwDevice, handle); 3243 } else { 3244 AudioParameter param; 3245 param.addInt(String8(AUDIO_PARAMETER_STREAM_ROUTING), 0); 3246 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3247 param.toString().string()); 3248 } 3249 return status; 3250} 3251 3252void AudioFlinger::PlaybackThread::addPatchTrack(const sp<PatchTrack>& track) 3253{ 3254 Mutex::Autolock _l(mLock); 3255 mTracks.add(track); 3256} 3257 3258void AudioFlinger::PlaybackThread::deletePatchTrack(const sp<PatchTrack>& track) 3259{ 3260 Mutex::Autolock _l(mLock); 3261 destroyTrack_l(track); 3262} 3263 3264void AudioFlinger::PlaybackThread::getAudioPortConfig(struct audio_port_config *config) 3265{ 3266 ThreadBase::getAudioPortConfig(config); 3267 config->role = AUDIO_PORT_ROLE_SOURCE; 3268 config->ext.mix.hw_module = mOutput->audioHwDev->handle(); 3269 config->ext.mix.usecase.stream = AUDIO_STREAM_DEFAULT; 3270} 3271 3272// ---------------------------------------------------------------------------- 3273 3274AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 3275 audio_io_handle_t id, audio_devices_t device, bool systemReady, type_t type) 3276 : PlaybackThread(audioFlinger, output, id, device, type, systemReady), 3277 // mAudioMixer below 3278 // mFastMixer below 3279 mFastMixerFutex(0) 3280 // mOutputSink below 3281 // mPipeSink below 3282 // mNormalSink below 3283{ 3284 ALOGV("MixerThread() id=%d device=%#x type=%d", id, device, type); 3285 ALOGV("mSampleRate=%u, mChannelMask=%#x, mChannelCount=%u, mFormat=%d, mFrameSize=%u, " 3286 "mFrameCount=%d, mNormalFrameCount=%d", 3287 mSampleRate, mChannelMask, mChannelCount, mFormat, mFrameSize, mFrameCount, 3288 mNormalFrameCount); 3289 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 3290 3291 if (type == DUPLICATING) { 3292 // The Duplicating thread uses the AudioMixer and delivers data to OutputTracks 3293 // (downstream MixerThreads) in DuplicatingThread::threadLoop_write(). 3294 // Do not create or use mFastMixer, mOutputSink, mPipeSink, or mNormalSink. 3295 return; 3296 } 3297 // create an NBAIO sink for the HAL output stream, and negotiate 3298 mOutputSink = new AudioStreamOutSink(output->stream); 3299 size_t numCounterOffers = 0; 3300 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount, mFormat)}; 3301 ssize_t index = mOutputSink->negotiate(offers, 1, NULL, numCounterOffers); 3302 ALOG_ASSERT(index == 0); 3303 3304 // initialize fast mixer depending on configuration 3305 bool initFastMixer; 3306 switch (kUseFastMixer) { 3307 case FastMixer_Never: 3308 initFastMixer = false; 3309 break; 3310 case FastMixer_Always: 3311 initFastMixer = true; 3312 break; 3313 case FastMixer_Static: 3314 case FastMixer_Dynamic: 3315 initFastMixer = mFrameCount < mNormalFrameCount; 3316 break; 3317 } 3318 if (initFastMixer) { 3319 audio_format_t fastMixerFormat; 3320 if (mMixerBufferEnabled && mEffectBufferEnabled) { 3321 fastMixerFormat = AUDIO_FORMAT_PCM_FLOAT; 3322 } else { 3323 fastMixerFormat = AUDIO_FORMAT_PCM_16_BIT; 3324 } 3325 if (mFormat != fastMixerFormat) { 3326 // change our Sink format to accept our intermediate precision 3327 mFormat = fastMixerFormat; 3328 free(mSinkBuffer); 3329 mFrameSize = mChannelCount * audio_bytes_per_sample(mFormat); 3330 const size_t sinkBufferSize = mNormalFrameCount * mFrameSize; 3331 (void)posix_memalign(&mSinkBuffer, 32, sinkBufferSize); 3332 } 3333 3334 // create a MonoPipe to connect our submix to FastMixer 3335 NBAIO_Format format = mOutputSink->format(); 3336 NBAIO_Format origformat = format; 3337 // adjust format to match that of the Fast Mixer 3338 ALOGV("format changed from %d to %d", format.mFormat, fastMixerFormat); 3339 format.mFormat = fastMixerFormat; 3340 format.mFrameSize = audio_bytes_per_sample(format.mFormat) * format.mChannelCount; 3341 3342 // This pipe depth compensates for scheduling latency of the normal mixer thread. 3343 // When it wakes up after a maximum latency, it runs a few cycles quickly before 3344 // finally blocking. Note the pipe implementation rounds up the request to a power of 2. 3345 MonoPipe *monoPipe = new MonoPipe(mNormalFrameCount * 4, format, true /*writeCanBlock*/); 3346 const NBAIO_Format offers[1] = {format}; 3347 size_t numCounterOffers = 0; 3348 ssize_t index = monoPipe->negotiate(offers, 1, NULL, numCounterOffers); 3349 ALOG_ASSERT(index == 0); 3350 monoPipe->setAvgFrames((mScreenState & 1) ? 3351 (monoPipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 3352 mPipeSink = monoPipe; 3353 3354#ifdef TEE_SINK 3355 if (mTeeSinkOutputEnabled) { 3356 // create a Pipe to archive a copy of FastMixer's output for dumpsys 3357 Pipe *teeSink = new Pipe(mTeeSinkOutputFrames, origformat); 3358 const NBAIO_Format offers2[1] = {origformat}; 3359 numCounterOffers = 0; 3360 index = teeSink->negotiate(offers2, 1, NULL, numCounterOffers); 3361 ALOG_ASSERT(index == 0); 3362 mTeeSink = teeSink; 3363 PipeReader *teeSource = new PipeReader(*teeSink); 3364 numCounterOffers = 0; 3365 index = teeSource->negotiate(offers2, 1, NULL, numCounterOffers); 3366 ALOG_ASSERT(index == 0); 3367 mTeeSource = teeSource; 3368 } 3369#endif 3370 3371 // create fast mixer and configure it initially with just one fast track for our submix 3372 mFastMixer = new FastMixer(); 3373 FastMixerStateQueue *sq = mFastMixer->sq(); 3374#ifdef STATE_QUEUE_DUMP 3375 sq->setObserverDump(&mStateQueueObserverDump); 3376 sq->setMutatorDump(&mStateQueueMutatorDump); 3377#endif 3378 FastMixerState *state = sq->begin(); 3379 FastTrack *fastTrack = &state->mFastTracks[0]; 3380 // wrap the source side of the MonoPipe to make it an AudioBufferProvider 3381 fastTrack->mBufferProvider = new SourceAudioBufferProvider(new MonoPipeReader(monoPipe)); 3382 fastTrack->mVolumeProvider = NULL; 3383 fastTrack->mChannelMask = mChannelMask; // mPipeSink channel mask for audio to FastMixer 3384 fastTrack->mFormat = mFormat; // mPipeSink format for audio to FastMixer 3385 fastTrack->mGeneration++; 3386 state->mFastTracksGen++; 3387 state->mTrackMask = 1; 3388 // fast mixer will use the HAL output sink 3389 state->mOutputSink = mOutputSink.get(); 3390 state->mOutputSinkGen++; 3391 state->mFrameCount = mFrameCount; 3392 state->mCommand = FastMixerState::COLD_IDLE; 3393 // already done in constructor initialization list 3394 //mFastMixerFutex = 0; 3395 state->mColdFutexAddr = &mFastMixerFutex; 3396 state->mColdGen++; 3397 state->mDumpState = &mFastMixerDumpState; 3398#ifdef TEE_SINK 3399 state->mTeeSink = mTeeSink.get(); 3400#endif 3401 mFastMixerNBLogWriter = audioFlinger->newWriter_l(kFastMixerLogSize, "FastMixer"); 3402 state->mNBLogWriter = mFastMixerNBLogWriter.get(); 3403 sq->end(); 3404 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3405 3406 // start the fast mixer 3407 mFastMixer->run("FastMixer", PRIORITY_URGENT_AUDIO); 3408 pid_t tid = mFastMixer->getTid(); 3409 sendPrioConfigEvent(getpid_cached, tid, kPriorityFastMixer); 3410 3411#ifdef AUDIO_WATCHDOG 3412 // create and start the watchdog 3413 mAudioWatchdog = new AudioWatchdog(); 3414 mAudioWatchdog->setDump(&mAudioWatchdogDump); 3415 mAudioWatchdog->run("AudioWatchdog", PRIORITY_URGENT_AUDIO); 3416 tid = mAudioWatchdog->getTid(); 3417 sendPrioConfigEvent(getpid_cached, tid, kPriorityFastMixer); 3418#endif 3419 3420 } 3421 3422 switch (kUseFastMixer) { 3423 case FastMixer_Never: 3424 case FastMixer_Dynamic: 3425 mNormalSink = mOutputSink; 3426 break; 3427 case FastMixer_Always: 3428 mNormalSink = mPipeSink; 3429 break; 3430 case FastMixer_Static: 3431 mNormalSink = initFastMixer ? mPipeSink : mOutputSink; 3432 break; 3433 } 3434} 3435 3436AudioFlinger::MixerThread::~MixerThread() 3437{ 3438 if (mFastMixer != 0) { 3439 FastMixerStateQueue *sq = mFastMixer->sq(); 3440 FastMixerState *state = sq->begin(); 3441 if (state->mCommand == FastMixerState::COLD_IDLE) { 3442 int32_t old = android_atomic_inc(&mFastMixerFutex); 3443 if (old == -1) { 3444 (void) syscall(__NR_futex, &mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 3445 } 3446 } 3447 state->mCommand = FastMixerState::EXIT; 3448 sq->end(); 3449 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3450 mFastMixer->join(); 3451 // Though the fast mixer thread has exited, it's state queue is still valid. 3452 // We'll use that extract the final state which contains one remaining fast track 3453 // corresponding to our sub-mix. 3454 state = sq->begin(); 3455 ALOG_ASSERT(state->mTrackMask == 1); 3456 FastTrack *fastTrack = &state->mFastTracks[0]; 3457 ALOG_ASSERT(fastTrack->mBufferProvider != NULL); 3458 delete fastTrack->mBufferProvider; 3459 sq->end(false /*didModify*/); 3460 mFastMixer.clear(); 3461#ifdef AUDIO_WATCHDOG 3462 if (mAudioWatchdog != 0) { 3463 mAudioWatchdog->requestExit(); 3464 mAudioWatchdog->requestExitAndWait(); 3465 mAudioWatchdog.clear(); 3466 } 3467#endif 3468 } 3469 mAudioFlinger->unregisterWriter(mFastMixerNBLogWriter); 3470 delete mAudioMixer; 3471} 3472 3473 3474uint32_t AudioFlinger::MixerThread::correctLatency_l(uint32_t latency) const 3475{ 3476 if (mFastMixer != 0) { 3477 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 3478 latency += (pipe->getAvgFrames() * 1000) / mSampleRate; 3479 } 3480 return latency; 3481} 3482 3483 3484void AudioFlinger::MixerThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) 3485{ 3486 PlaybackThread::threadLoop_removeTracks(tracksToRemove); 3487} 3488 3489ssize_t AudioFlinger::MixerThread::threadLoop_write() 3490{ 3491 // FIXME we should only do one push per cycle; confirm this is true 3492 // Start the fast mixer if it's not already running 3493 if (mFastMixer != 0) { 3494 FastMixerStateQueue *sq = mFastMixer->sq(); 3495 FastMixerState *state = sq->begin(); 3496 if (state->mCommand != FastMixerState::MIX_WRITE && 3497 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)) { 3498 if (state->mCommand == FastMixerState::COLD_IDLE) { 3499 3500 // FIXME workaround for first HAL write being CPU bound on some devices 3501 ATRACE_BEGIN("write"); 3502 mOutput->write((char *)mSinkBuffer, 0); 3503 ATRACE_END(); 3504 3505 int32_t old = android_atomic_inc(&mFastMixerFutex); 3506 if (old == -1) { 3507 (void) syscall(__NR_futex, &mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 3508 } 3509#ifdef AUDIO_WATCHDOG 3510 if (mAudioWatchdog != 0) { 3511 mAudioWatchdog->resume(); 3512 } 3513#endif 3514 } 3515 state->mCommand = FastMixerState::MIX_WRITE; 3516#ifdef FAST_THREAD_STATISTICS 3517 mFastMixerDumpState.increaseSamplingN(mAudioFlinger->isLowRamDevice() ? 3518 FastThreadDumpState::kSamplingNforLowRamDevice : FastThreadDumpState::kSamplingN); 3519#endif 3520 sq->end(); 3521 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3522 if (kUseFastMixer == FastMixer_Dynamic) { 3523 mNormalSink = mPipeSink; 3524 } 3525 } else { 3526 sq->end(false /*didModify*/); 3527 } 3528 } 3529 return PlaybackThread::threadLoop_write(); 3530} 3531 3532void AudioFlinger::MixerThread::threadLoop_standby() 3533{ 3534 // Idle the fast mixer if it's currently running 3535 if (mFastMixer != 0) { 3536 FastMixerStateQueue *sq = mFastMixer->sq(); 3537 FastMixerState *state = sq->begin(); 3538 if (!(state->mCommand & FastMixerState::IDLE)) { 3539 state->mCommand = FastMixerState::COLD_IDLE; 3540 state->mColdFutexAddr = &mFastMixerFutex; 3541 state->mColdGen++; 3542 mFastMixerFutex = 0; 3543 sq->end(); 3544 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 3545 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3546 if (kUseFastMixer == FastMixer_Dynamic) { 3547 mNormalSink = mOutputSink; 3548 } 3549#ifdef AUDIO_WATCHDOG 3550 if (mAudioWatchdog != 0) { 3551 mAudioWatchdog->pause(); 3552 } 3553#endif 3554 } else { 3555 sq->end(false /*didModify*/); 3556 } 3557 } 3558 PlaybackThread::threadLoop_standby(); 3559} 3560 3561bool AudioFlinger::PlaybackThread::waitingAsyncCallback_l() 3562{ 3563 return false; 3564} 3565 3566bool AudioFlinger::PlaybackThread::shouldStandby_l() 3567{ 3568 return !mStandby; 3569} 3570 3571bool AudioFlinger::PlaybackThread::waitingAsyncCallback() 3572{ 3573 Mutex::Autolock _l(mLock); 3574 return waitingAsyncCallback_l(); 3575} 3576 3577// shared by MIXER and DIRECT, overridden by DUPLICATING 3578void AudioFlinger::PlaybackThread::threadLoop_standby() 3579{ 3580 ALOGV("Audio hardware entering standby, mixer %p, suspend count %d", this, mSuspended); 3581 mOutput->standby(); 3582 if (mUseAsyncWrite != 0) { 3583 // discard any pending drain or write ack by incrementing sequence 3584 mWriteAckSequence = (mWriteAckSequence + 2) & ~1; 3585 mDrainSequence = (mDrainSequence + 2) & ~1; 3586 ALOG_ASSERT(mCallbackThread != 0); 3587 mCallbackThread->setWriteBlocked(mWriteAckSequence); 3588 mCallbackThread->setDraining(mDrainSequence); 3589 } 3590 mHwPaused = false; 3591} 3592 3593void AudioFlinger::PlaybackThread::onAddNewTrack_l() 3594{ 3595 ALOGV("signal playback thread"); 3596 broadcast_l(); 3597} 3598 3599void AudioFlinger::MixerThread::threadLoop_mix() 3600{ 3601 // obtain the presentation timestamp of the next output buffer 3602 int64_t pts; 3603 status_t status = INVALID_OPERATION; 3604 3605 if (mNormalSink != 0) { 3606 status = mNormalSink->getNextWriteTimestamp(&pts); 3607 } else { 3608 status = mOutputSink->getNextWriteTimestamp(&pts); 3609 } 3610 3611 if (status != NO_ERROR) { 3612 pts = AudioBufferProvider::kInvalidPTS; 3613 } 3614 3615 // mix buffers... 3616 mAudioMixer->process(pts); 3617 mCurrentWriteLength = mSinkBufferSize; 3618 // increase sleep time progressively when application underrun condition clears. 3619 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 3620 // that a steady state of alternating ready/not ready conditions keeps the sleep time 3621 // such that we would underrun the audio HAL. 3622 if ((mSleepTimeUs == 0) && (sleepTimeShift > 0)) { 3623 sleepTimeShift--; 3624 } 3625 mSleepTimeUs = 0; 3626 mStandbyTimeNs = systemTime() + mStandbyDelayNs; 3627 //TODO: delay standby when effects have a tail 3628 3629} 3630 3631void AudioFlinger::MixerThread::threadLoop_sleepTime() 3632{ 3633 // If no tracks are ready, sleep once for the duration of an output 3634 // buffer size, then write 0s to the output 3635 if (mSleepTimeUs == 0) { 3636 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3637 mSleepTimeUs = mActiveSleepTimeUs >> sleepTimeShift; 3638 if (mSleepTimeUs < kMinThreadSleepTimeUs) { 3639 mSleepTimeUs = kMinThreadSleepTimeUs; 3640 } 3641 // reduce sleep time in case of consecutive application underruns to avoid 3642 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 3643 // duration we would end up writing less data than needed by the audio HAL if 3644 // the condition persists. 3645 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 3646 sleepTimeShift++; 3647 } 3648 } else { 3649 mSleepTimeUs = mIdleSleepTimeUs; 3650 } 3651 } else if (mBytesWritten != 0 || (mMixerStatus == MIXER_TRACKS_ENABLED)) { 3652 // clear out mMixerBuffer or mSinkBuffer, to ensure buffers are cleared 3653 // before effects processing or output. 3654 if (mMixerBufferValid) { 3655 memset(mMixerBuffer, 0, mMixerBufferSize); 3656 } else { 3657 memset(mSinkBuffer, 0, mSinkBufferSize); 3658 } 3659 mSleepTimeUs = 0; 3660 ALOGV_IF(mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED), 3661 "anticipated start"); 3662 } 3663 // TODO add standby time extension fct of effect tail 3664} 3665 3666// prepareTracks_l() must be called with ThreadBase::mLock held 3667AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 3668 Vector< sp<Track> > *tracksToRemove) 3669{ 3670 3671 mixer_state mixerStatus = MIXER_IDLE; 3672 // find out which tracks need to be processed 3673 size_t count = mActiveTracks.size(); 3674 size_t mixedTracks = 0; 3675 size_t tracksWithEffect = 0; 3676 // counts only _active_ fast tracks 3677 size_t fastTracks = 0; 3678 uint32_t resetMask = 0; // bit mask of fast tracks that need to be reset 3679 3680 float masterVolume = mMasterVolume; 3681 bool masterMute = mMasterMute; 3682 3683 if (masterMute) { 3684 masterVolume = 0; 3685 } 3686 // Delegate master volume control to effect in output mix effect chain if needed 3687 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 3688 if (chain != 0) { 3689 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 3690 chain->setVolume_l(&v, &v); 3691 masterVolume = (float)((v + (1 << 23)) >> 24); 3692 chain.clear(); 3693 } 3694 3695 // prepare a new state to push 3696 FastMixerStateQueue *sq = NULL; 3697 FastMixerState *state = NULL; 3698 bool didModify = false; 3699 FastMixerStateQueue::block_t block = FastMixerStateQueue::BLOCK_UNTIL_PUSHED; 3700 if (mFastMixer != 0) { 3701 sq = mFastMixer->sq(); 3702 state = sq->begin(); 3703 } 3704 3705 mMixerBufferValid = false; // mMixerBuffer has no valid data until appropriate tracks found. 3706 mEffectBufferValid = false; // mEffectBuffer has no valid data until tracks found. 3707 3708 for (size_t i=0 ; i<count ; i++) { 3709 const sp<Track> t = mActiveTracks[i].promote(); 3710 if (t == 0) { 3711 continue; 3712 } 3713 3714 // this const just means the local variable doesn't change 3715 Track* const track = t.get(); 3716 3717 // process fast tracks 3718 if (track->isFastTrack()) { 3719 3720 // It's theoretically possible (though unlikely) for a fast track to be created 3721 // and then removed within the same normal mix cycle. This is not a problem, as 3722 // the track never becomes active so it's fast mixer slot is never touched. 3723 // The converse, of removing an (active) track and then creating a new track 3724 // at the identical fast mixer slot within the same normal mix cycle, 3725 // is impossible because the slot isn't marked available until the end of each cycle. 3726 int j = track->mFastIndex; 3727 ALOG_ASSERT(0 < j && j < (int)FastMixerState::kMaxFastTracks); 3728 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << j))); 3729 FastTrack *fastTrack = &state->mFastTracks[j]; 3730 3731 // Determine whether the track is currently in underrun condition, 3732 // and whether it had a recent underrun. 3733 FastTrackDump *ftDump = &mFastMixerDumpState.mTracks[j]; 3734 FastTrackUnderruns underruns = ftDump->mUnderruns; 3735 uint32_t recentFull = (underruns.mBitFields.mFull - 3736 track->mObservedUnderruns.mBitFields.mFull) & UNDERRUN_MASK; 3737 uint32_t recentPartial = (underruns.mBitFields.mPartial - 3738 track->mObservedUnderruns.mBitFields.mPartial) & UNDERRUN_MASK; 3739 uint32_t recentEmpty = (underruns.mBitFields.mEmpty - 3740 track->mObservedUnderruns.mBitFields.mEmpty) & UNDERRUN_MASK; 3741 uint32_t recentUnderruns = recentPartial + recentEmpty; 3742 track->mObservedUnderruns = underruns; 3743 // don't count underruns that occur while stopping or pausing 3744 // or stopped which can occur when flush() is called while active 3745 if (!(track->isStopping() || track->isPausing() || track->isStopped()) && 3746 recentUnderruns > 0) { 3747 // FIXME fast mixer will pull & mix partial buffers, but we count as a full underrun 3748 track->mAudioTrackServerProxy->tallyUnderrunFrames(recentUnderruns * mFrameCount); 3749 } 3750 3751 // This is similar to the state machine for normal tracks, 3752 // with a few modifications for fast tracks. 3753 bool isActive = true; 3754 switch (track->mState) { 3755 case TrackBase::STOPPING_1: 3756 // track stays active in STOPPING_1 state until first underrun 3757 if (recentUnderruns > 0 || track->isTerminated()) { 3758 track->mState = TrackBase::STOPPING_2; 3759 } 3760 break; 3761 case TrackBase::PAUSING: 3762 // ramp down is not yet implemented 3763 track->setPaused(); 3764 break; 3765 case TrackBase::RESUMING: 3766 // ramp up is not yet implemented 3767 track->mState = TrackBase::ACTIVE; 3768 break; 3769 case TrackBase::ACTIVE: 3770 if (recentFull > 0 || recentPartial > 0) { 3771 // track has provided at least some frames recently: reset retry count 3772 track->mRetryCount = kMaxTrackRetries; 3773 } 3774 if (recentUnderruns == 0) { 3775 // no recent underruns: stay active 3776 break; 3777 } 3778 // there has recently been an underrun of some kind 3779 if (track->sharedBuffer() == 0) { 3780 // were any of the recent underruns "empty" (no frames available)? 3781 if (recentEmpty == 0) { 3782 // no, then ignore the partial underruns as they are allowed indefinitely 3783 break; 3784 } 3785 // there has recently been an "empty" underrun: decrement the retry counter 3786 if (--(track->mRetryCount) > 0) { 3787 break; 3788 } 3789 // indicate to client process that the track was disabled because of underrun; 3790 // it will then automatically call start() when data is available 3791 android_atomic_or(CBLK_DISABLED, &track->mCblk->mFlags); 3792 // remove from active list, but state remains ACTIVE [confusing but true] 3793 isActive = false; 3794 break; 3795 } 3796 // fall through 3797 case TrackBase::STOPPING_2: 3798 case TrackBase::PAUSED: 3799 case TrackBase::STOPPED: 3800 case TrackBase::FLUSHED: // flush() while active 3801 // Check for presentation complete if track is inactive 3802 // We have consumed all the buffers of this track. 3803 // This would be incomplete if we auto-paused on underrun 3804 { 3805 size_t audioHALFrames = 3806 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 3807 size_t framesWritten = mBytesWritten / mFrameSize; 3808 if (!(mStandby || track->presentationComplete(framesWritten, audioHALFrames))) { 3809 // track stays in active list until presentation is complete 3810 break; 3811 } 3812 } 3813 if (track->isStopping_2()) { 3814 track->mState = TrackBase::STOPPED; 3815 } 3816 if (track->isStopped()) { 3817 // Can't reset directly, as fast mixer is still polling this track 3818 // track->reset(); 3819 // So instead mark this track as needing to be reset after push with ack 3820 resetMask |= 1 << i; 3821 } 3822 isActive = false; 3823 break; 3824 case TrackBase::IDLE: 3825 default: 3826 LOG_ALWAYS_FATAL("unexpected track state %d", track->mState); 3827 } 3828 3829 if (isActive) { 3830 // was it previously inactive? 3831 if (!(state->mTrackMask & (1 << j))) { 3832 ExtendedAudioBufferProvider *eabp = track; 3833 VolumeProvider *vp = track; 3834 fastTrack->mBufferProvider = eabp; 3835 fastTrack->mVolumeProvider = vp; 3836 fastTrack->mChannelMask = track->mChannelMask; 3837 fastTrack->mFormat = track->mFormat; 3838 fastTrack->mGeneration++; 3839 state->mTrackMask |= 1 << j; 3840 didModify = true; 3841 // no acknowledgement required for newly active tracks 3842 } 3843 // cache the combined master volume and stream type volume for fast mixer; this 3844 // lacks any synchronization or barrier so VolumeProvider may read a stale value 3845 track->mCachedVolume = masterVolume * mStreamTypes[track->streamType()].volume; 3846 ++fastTracks; 3847 } else { 3848 // was it previously active? 3849 if (state->mTrackMask & (1 << j)) { 3850 fastTrack->mBufferProvider = NULL; 3851 fastTrack->mGeneration++; 3852 state->mTrackMask &= ~(1 << j); 3853 didModify = true; 3854 // If any fast tracks were removed, we must wait for acknowledgement 3855 // because we're about to decrement the last sp<> on those tracks. 3856 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3857 } else { 3858 LOG_ALWAYS_FATAL("fast track %d should have been active", j); 3859 } 3860 tracksToRemove->add(track); 3861 // Avoids a misleading display in dumpsys 3862 track->mObservedUnderruns.mBitFields.mMostRecent = UNDERRUN_FULL; 3863 } 3864 continue; 3865 } 3866 3867 { // local variable scope to avoid goto warning 3868 3869 audio_track_cblk_t* cblk = track->cblk(); 3870 3871 // The first time a track is added we wait 3872 // for all its buffers to be filled before processing it 3873 int name = track->name(); 3874 // make sure that we have enough frames to mix one full buffer. 3875 // enforce this condition only once to enable draining the buffer in case the client 3876 // app does not call stop() and relies on underrun to stop: 3877 // hence the test on (mMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 3878 // during last round 3879 size_t desiredFrames; 3880 const uint32_t sampleRate = track->mAudioTrackServerProxy->getSampleRate(); 3881 AudioPlaybackRate playbackRate = track->mAudioTrackServerProxy->getPlaybackRate(); 3882 3883 desiredFrames = sourceFramesNeededWithTimestretch( 3884 sampleRate, mNormalFrameCount, mSampleRate, playbackRate.mSpeed); 3885 // TODO: ONLY USED FOR LEGACY RESAMPLERS, remove when they are removed. 3886 // add frames already consumed but not yet released by the resampler 3887 // because mAudioTrackServerProxy->framesReady() will include these frames 3888 desiredFrames += mAudioMixer->getUnreleasedFrames(track->name()); 3889 3890 uint32_t minFrames = 1; 3891 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() && 3892 (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) { 3893 minFrames = desiredFrames; 3894 } 3895 3896 size_t framesReady = track->framesReady(); 3897 if (ATRACE_ENABLED()) { 3898 // I wish we had formatted trace names 3899 char traceName[16]; 3900 strcpy(traceName, "nRdy"); 3901 int name = track->name(); 3902 if (AudioMixer::TRACK0 <= name && 3903 name < (int) (AudioMixer::TRACK0 + AudioMixer::MAX_NUM_TRACKS)) { 3904 name -= AudioMixer::TRACK0; 3905 traceName[4] = (name / 10) + '0'; 3906 traceName[5] = (name % 10) + '0'; 3907 } else { 3908 traceName[4] = '?'; 3909 traceName[5] = '?'; 3910 } 3911 traceName[6] = '\0'; 3912 ATRACE_INT(traceName, framesReady); 3913 } 3914 if ((framesReady >= minFrames) && track->isReady() && 3915 !track->isPaused() && !track->isTerminated()) 3916 { 3917 ALOGVV("track %d s=%08x [OK] on thread %p", name, cblk->mServer, this); 3918 3919 mixedTracks++; 3920 3921 // track->mainBuffer() != mSinkBuffer or mMixerBuffer means 3922 // there is an effect chain connected to the track 3923 chain.clear(); 3924 if (track->mainBuffer() != mSinkBuffer && 3925 track->mainBuffer() != mMixerBuffer) { 3926 if (mEffectBufferEnabled) { 3927 mEffectBufferValid = true; // Later can set directly. 3928 } 3929 chain = getEffectChain_l(track->sessionId()); 3930 // Delegate volume control to effect in track effect chain if needed 3931 if (chain != 0) { 3932 tracksWithEffect++; 3933 } else { 3934 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on " 3935 "session %d", 3936 name, track->sessionId()); 3937 } 3938 } 3939 3940 3941 int param = AudioMixer::VOLUME; 3942 if (track->mFillingUpStatus == Track::FS_FILLED) { 3943 // no ramp for the first volume setting 3944 track->mFillingUpStatus = Track::FS_ACTIVE; 3945 if (track->mState == TrackBase::RESUMING) { 3946 track->mState = TrackBase::ACTIVE; 3947 param = AudioMixer::RAMP_VOLUME; 3948 } 3949 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 3950 // FIXME should not make a decision based on mServer 3951 } else if (cblk->mServer != 0) { 3952 // If the track is stopped before the first frame was mixed, 3953 // do not apply ramp 3954 param = AudioMixer::RAMP_VOLUME; 3955 } 3956 3957 // compute volume for this track 3958 uint32_t vl, vr; // in U8.24 integer format 3959 float vlf, vrf, vaf; // in [0.0, 1.0] float format 3960 if (track->isPausing() || mStreamTypes[track->streamType()].mute) { 3961 vl = vr = 0; 3962 vlf = vrf = vaf = 0.; 3963 if (track->isPausing()) { 3964 track->setPaused(); 3965 } 3966 } else { 3967 3968 // read original volumes with volume control 3969 float typeVolume = mStreamTypes[track->streamType()].volume; 3970 float v = masterVolume * typeVolume; 3971 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 3972 gain_minifloat_packed_t vlr = proxy->getVolumeLR(); 3973 vlf = float_from_gain(gain_minifloat_unpack_left(vlr)); 3974 vrf = float_from_gain(gain_minifloat_unpack_right(vlr)); 3975 // track volumes come from shared memory, so can't be trusted and must be clamped 3976 if (vlf > GAIN_FLOAT_UNITY) { 3977 ALOGV("Track left volume out of range: %.3g", vlf); 3978 vlf = GAIN_FLOAT_UNITY; 3979 } 3980 if (vrf > GAIN_FLOAT_UNITY) { 3981 ALOGV("Track right volume out of range: %.3g", vrf); 3982 vrf = GAIN_FLOAT_UNITY; 3983 } 3984 // now apply the master volume and stream type volume 3985 vlf *= v; 3986 vrf *= v; 3987 // assuming master volume and stream type volume each go up to 1.0, 3988 // then derive vl and vr as U8.24 versions for the effect chain 3989 const float scaleto8_24 = MAX_GAIN_INT * MAX_GAIN_INT; 3990 vl = (uint32_t) (scaleto8_24 * vlf); 3991 vr = (uint32_t) (scaleto8_24 * vrf); 3992 // vl and vr are now in U8.24 format 3993 uint16_t sendLevel = proxy->getSendLevel_U4_12(); 3994 // send level comes from shared memory and so may be corrupt 3995 if (sendLevel > MAX_GAIN_INT) { 3996 ALOGV("Track send level out of range: %04X", sendLevel); 3997 sendLevel = MAX_GAIN_INT; 3998 } 3999 // vaf is represented as [0.0, 1.0] float by rescaling sendLevel 4000 vaf = v * sendLevel * (1. / MAX_GAIN_INT); 4001 } 4002 4003 // Delegate volume control to effect in track effect chain if needed 4004 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 4005 // Do not ramp volume if volume is controlled by effect 4006 param = AudioMixer::VOLUME; 4007 // Update remaining floating point volume levels 4008 vlf = (float)vl / (1 << 24); 4009 vrf = (float)vr / (1 << 24); 4010 track->mHasVolumeController = true; 4011 } else { 4012 // force no volume ramp when volume controller was just disabled or removed 4013 // from effect chain to avoid volume spike 4014 if (track->mHasVolumeController) { 4015 param = AudioMixer::VOLUME; 4016 } 4017 track->mHasVolumeController = false; 4018 } 4019 4020 // XXX: these things DON'T need to be done each time 4021 mAudioMixer->setBufferProvider(name, track); 4022 mAudioMixer->enable(name); 4023 4024 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, &vlf); 4025 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, &vrf); 4026 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, &vaf); 4027 mAudioMixer->setParameter( 4028 name, 4029 AudioMixer::TRACK, 4030 AudioMixer::FORMAT, (void *)track->format()); 4031 mAudioMixer->setParameter( 4032 name, 4033 AudioMixer::TRACK, 4034 AudioMixer::CHANNEL_MASK, (void *)(uintptr_t)track->channelMask()); 4035 mAudioMixer->setParameter( 4036 name, 4037 AudioMixer::TRACK, 4038 AudioMixer::MIXER_CHANNEL_MASK, (void *)(uintptr_t)mChannelMask); 4039 // limit track sample rate to 2 x output sample rate, which changes at re-configuration 4040 uint32_t maxSampleRate = mSampleRate * AUDIO_RESAMPLER_DOWN_RATIO_MAX; 4041 uint32_t reqSampleRate = track->mAudioTrackServerProxy->getSampleRate(); 4042 if (reqSampleRate == 0) { 4043 reqSampleRate = mSampleRate; 4044 } else if (reqSampleRate > maxSampleRate) { 4045 reqSampleRate = maxSampleRate; 4046 } 4047 mAudioMixer->setParameter( 4048 name, 4049 AudioMixer::RESAMPLE, 4050 AudioMixer::SAMPLE_RATE, 4051 (void *)(uintptr_t)reqSampleRate); 4052 4053 AudioPlaybackRate playbackRate = track->mAudioTrackServerProxy->getPlaybackRate(); 4054 mAudioMixer->setParameter( 4055 name, 4056 AudioMixer::TIMESTRETCH, 4057 AudioMixer::PLAYBACK_RATE, 4058 &playbackRate); 4059 4060 /* 4061 * Select the appropriate output buffer for the track. 4062 * 4063 * Tracks with effects go into their own effects chain buffer 4064 * and from there into either mEffectBuffer or mSinkBuffer. 4065 * 4066 * Other tracks can use mMixerBuffer for higher precision 4067 * channel accumulation. If this buffer is enabled 4068 * (mMixerBufferEnabled true), then selected tracks will accumulate 4069 * into it. 4070 * 4071 */ 4072 if (mMixerBufferEnabled 4073 && (track->mainBuffer() == mSinkBuffer 4074 || track->mainBuffer() == mMixerBuffer)) { 4075 mAudioMixer->setParameter( 4076 name, 4077 AudioMixer::TRACK, 4078 AudioMixer::MIXER_FORMAT, (void *)mMixerBufferFormat); 4079 mAudioMixer->setParameter( 4080 name, 4081 AudioMixer::TRACK, 4082 AudioMixer::MAIN_BUFFER, (void *)mMixerBuffer); 4083 // TODO: override track->mainBuffer()? 4084 mMixerBufferValid = true; 4085 } else { 4086 mAudioMixer->setParameter( 4087 name, 4088 AudioMixer::TRACK, 4089 AudioMixer::MIXER_FORMAT, (void *)AUDIO_FORMAT_PCM_16_BIT); 4090 mAudioMixer->setParameter( 4091 name, 4092 AudioMixer::TRACK, 4093 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 4094 } 4095 mAudioMixer->setParameter( 4096 name, 4097 AudioMixer::TRACK, 4098 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 4099 4100 // reset retry count 4101 track->mRetryCount = kMaxTrackRetries; 4102 4103 // If one track is ready, set the mixer ready if: 4104 // - the mixer was not ready during previous round OR 4105 // - no other track is not ready 4106 if (mMixerStatusIgnoringFastTracks != MIXER_TRACKS_READY || 4107 mixerStatus != MIXER_TRACKS_ENABLED) { 4108 mixerStatus = MIXER_TRACKS_READY; 4109 } 4110 } else { 4111 if (framesReady < desiredFrames && !track->isStopped() && !track->isPaused()) { 4112 ALOGV("track(%p) underrun, framesReady(%zu) < framesDesired(%zd)", 4113 track, framesReady, desiredFrames); 4114 track->mAudioTrackServerProxy->tallyUnderrunFrames(desiredFrames); 4115 } 4116 // clear effect chain input buffer if an active track underruns to avoid sending 4117 // previous audio buffer again to effects 4118 chain = getEffectChain_l(track->sessionId()); 4119 if (chain != 0) { 4120 chain->clearInputBuffer(); 4121 } 4122 4123 ALOGVV("track %d s=%08x [NOT READY] on thread %p", name, cblk->mServer, this); 4124 if ((track->sharedBuffer() != 0) || track->isTerminated() || 4125 track->isStopped() || track->isPaused()) { 4126 // We have consumed all the buffers of this track. 4127 // Remove it from the list of active tracks. 4128 // TODO: use actual buffer filling status instead of latency when available from 4129 // audio HAL 4130 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 4131 size_t framesWritten = mBytesWritten / mFrameSize; 4132 if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) { 4133 if (track->isStopped()) { 4134 track->reset(); 4135 } 4136 tracksToRemove->add(track); 4137 } 4138 } else { 4139 // No buffers for this track. Give it a few chances to 4140 // fill a buffer, then remove it from active list. 4141 if (--(track->mRetryCount) <= 0) { 4142 ALOGI("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 4143 tracksToRemove->add(track); 4144 // indicate to client process that the track was disabled because of underrun; 4145 // it will then automatically call start() when data is available 4146 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 4147 // If one track is not ready, mark the mixer also not ready if: 4148 // - the mixer was ready during previous round OR 4149 // - no other track is ready 4150 } else if (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY || 4151 mixerStatus != MIXER_TRACKS_READY) { 4152 mixerStatus = MIXER_TRACKS_ENABLED; 4153 } 4154 } 4155 mAudioMixer->disable(name); 4156 } 4157 4158 } // local variable scope to avoid goto warning 4159track_is_ready: ; 4160 4161 } 4162 4163 // Push the new FastMixer state if necessary 4164 bool pauseAudioWatchdog = false; 4165 if (didModify) { 4166 state->mFastTracksGen++; 4167 // if the fast mixer was active, but now there are no fast tracks, then put it in cold idle 4168 if (kUseFastMixer == FastMixer_Dynamic && 4169 state->mCommand == FastMixerState::MIX_WRITE && state->mTrackMask <= 1) { 4170 state->mCommand = FastMixerState::COLD_IDLE; 4171 state->mColdFutexAddr = &mFastMixerFutex; 4172 state->mColdGen++; 4173 mFastMixerFutex = 0; 4174 if (kUseFastMixer == FastMixer_Dynamic) { 4175 mNormalSink = mOutputSink; 4176 } 4177 // If we go into cold idle, need to wait for acknowledgement 4178 // so that fast mixer stops doing I/O. 4179 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 4180 pauseAudioWatchdog = true; 4181 } 4182 } 4183 if (sq != NULL) { 4184 sq->end(didModify); 4185 sq->push(block); 4186 } 4187#ifdef AUDIO_WATCHDOG 4188 if (pauseAudioWatchdog && mAudioWatchdog != 0) { 4189 mAudioWatchdog->pause(); 4190 } 4191#endif 4192 4193 // Now perform the deferred reset on fast tracks that have stopped 4194 while (resetMask != 0) { 4195 size_t i = __builtin_ctz(resetMask); 4196 ALOG_ASSERT(i < count); 4197 resetMask &= ~(1 << i); 4198 sp<Track> t = mActiveTracks[i].promote(); 4199 if (t == 0) { 4200 continue; 4201 } 4202 Track* track = t.get(); 4203 ALOG_ASSERT(track->isFastTrack() && track->isStopped()); 4204 track->reset(); 4205 } 4206 4207 // remove all the tracks that need to be... 4208 removeTracks_l(*tracksToRemove); 4209 4210 if (getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX) != 0) { 4211 mEffectBufferValid = true; 4212 } 4213 4214 if (mEffectBufferValid) { 4215 // as long as there are effects we should clear the effects buffer, to avoid 4216 // passing a non-clean buffer to the effect chain 4217 memset(mEffectBuffer, 0, mEffectBufferSize); 4218 } 4219 // sink or mix buffer must be cleared if all tracks are connected to an 4220 // effect chain as in this case the mixer will not write to the sink or mix buffer 4221 // and track effects will accumulate into it 4222 if ((mBytesRemaining == 0) && ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 4223 (mixedTracks == 0 && fastTracks > 0))) { 4224 // FIXME as a performance optimization, should remember previous zero status 4225 if (mMixerBufferValid) { 4226 memset(mMixerBuffer, 0, mMixerBufferSize); 4227 // TODO: In testing, mSinkBuffer below need not be cleared because 4228 // the PlaybackThread::threadLoop() copies mMixerBuffer into mSinkBuffer 4229 // after mixing. 4230 // 4231 // To enforce this guarantee: 4232 // ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 4233 // (mixedTracks == 0 && fastTracks > 0)) 4234 // must imply MIXER_TRACKS_READY. 4235 // Later, we may clear buffers regardless, and skip much of this logic. 4236 } 4237 // FIXME as a performance optimization, should remember previous zero status 4238 memset(mSinkBuffer, 0, mNormalFrameCount * mFrameSize); 4239 } 4240 4241 // if any fast tracks, then status is ready 4242 mMixerStatusIgnoringFastTracks = mixerStatus; 4243 if (fastTracks > 0) { 4244 mixerStatus = MIXER_TRACKS_READY; 4245 } 4246 return mixerStatus; 4247} 4248 4249// getTrackName_l() must be called with ThreadBase::mLock held 4250int AudioFlinger::MixerThread::getTrackName_l(audio_channel_mask_t channelMask, 4251 audio_format_t format, int sessionId) 4252{ 4253 return mAudioMixer->getTrackName(channelMask, format, sessionId); 4254} 4255 4256// deleteTrackName_l() must be called with ThreadBase::mLock held 4257void AudioFlinger::MixerThread::deleteTrackName_l(int name) 4258{ 4259 ALOGV("remove track (%d) and delete from mixer", name); 4260 mAudioMixer->deleteTrackName(name); 4261} 4262 4263// checkForNewParameter_l() must be called with ThreadBase::mLock held 4264bool AudioFlinger::MixerThread::checkForNewParameter_l(const String8& keyValuePair, 4265 status_t& status) 4266{ 4267 bool reconfig = false; 4268 4269 status = NO_ERROR; 4270 4271 // if !&IDLE, holds the FastMixer state to restore after new parameters processed 4272 FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE; 4273 if (mFastMixer != 0) { 4274 FastMixerStateQueue *sq = mFastMixer->sq(); 4275 FastMixerState *state = sq->begin(); 4276 if (!(state->mCommand & FastMixerState::IDLE)) { 4277 previousCommand = state->mCommand; 4278 state->mCommand = FastMixerState::HOT_IDLE; 4279 sq->end(); 4280 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 4281 } else { 4282 sq->end(false /*didModify*/); 4283 } 4284 } 4285 4286 AudioParameter param = AudioParameter(keyValuePair); 4287 int value; 4288 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 4289 reconfig = true; 4290 } 4291 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 4292 if (!isValidPcmSinkFormat((audio_format_t) value)) { 4293 status = BAD_VALUE; 4294 } else { 4295 // no need to save value, since it's constant 4296 reconfig = true; 4297 } 4298 } 4299 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 4300 if (!isValidPcmSinkChannelMask((audio_channel_mask_t) value)) { 4301 status = BAD_VALUE; 4302 } else { 4303 // no need to save value, since it's constant 4304 reconfig = true; 4305 } 4306 } 4307 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 4308 // do not accept frame count changes if tracks are open as the track buffer 4309 // size depends on frame count and correct behavior would not be guaranteed 4310 // if frame count is changed after track creation 4311 if (!mTracks.isEmpty()) { 4312 status = INVALID_OPERATION; 4313 } else { 4314 reconfig = true; 4315 } 4316 } 4317 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 4318#ifdef ADD_BATTERY_DATA 4319 // when changing the audio output device, call addBatteryData to notify 4320 // the change 4321 if (mOutDevice != value) { 4322 uint32_t params = 0; 4323 // check whether speaker is on 4324 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 4325 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 4326 } 4327 4328 audio_devices_t deviceWithoutSpeaker 4329 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 4330 // check if any other device (except speaker) is on 4331 if (value & deviceWithoutSpeaker) { 4332 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 4333 } 4334 4335 if (params != 0) { 4336 addBatteryData(params); 4337 } 4338 } 4339#endif 4340 4341 // forward device change to effects that have requested to be 4342 // aware of attached audio device. 4343 if (value != AUDIO_DEVICE_NONE) { 4344 mOutDevice = value; 4345 for (size_t i = 0; i < mEffectChains.size(); i++) { 4346 mEffectChains[i]->setDevice_l(mOutDevice); 4347 } 4348 } 4349 } 4350 4351 if (status == NO_ERROR) { 4352 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 4353 keyValuePair.string()); 4354 if (!mStandby && status == INVALID_OPERATION) { 4355 mOutput->standby(); 4356 mStandby = true; 4357 mBytesWritten = 0; 4358 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 4359 keyValuePair.string()); 4360 } 4361 if (status == NO_ERROR && reconfig) { 4362 readOutputParameters_l(); 4363 delete mAudioMixer; 4364 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 4365 for (size_t i = 0; i < mTracks.size() ; i++) { 4366 int name = getTrackName_l(mTracks[i]->mChannelMask, 4367 mTracks[i]->mFormat, mTracks[i]->mSessionId); 4368 if (name < 0) { 4369 break; 4370 } 4371 mTracks[i]->mName = name; 4372 } 4373 sendIoConfigEvent_l(AUDIO_OUTPUT_CONFIG_CHANGED); 4374 } 4375 } 4376 4377 if (!(previousCommand & FastMixerState::IDLE)) { 4378 ALOG_ASSERT(mFastMixer != 0); 4379 FastMixerStateQueue *sq = mFastMixer->sq(); 4380 FastMixerState *state = sq->begin(); 4381 ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE); 4382 state->mCommand = previousCommand; 4383 sq->end(); 4384 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 4385 } 4386 4387 return reconfig; 4388} 4389 4390 4391void AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 4392{ 4393 const size_t SIZE = 256; 4394 char buffer[SIZE]; 4395 String8 result; 4396 4397 PlaybackThread::dumpInternals(fd, args); 4398 dprintf(fd, " Thread throttle time (msecs): %u\n", mThreadThrottleTimeMs); 4399 dprintf(fd, " AudioMixer tracks: 0x%08x\n", mAudioMixer->trackNames()); 4400 4401 // Make a non-atomic copy of fast mixer dump state so it won't change underneath us 4402 const FastMixerDumpState copy(mFastMixerDumpState); 4403 copy.dump(fd); 4404 4405#ifdef STATE_QUEUE_DUMP 4406 // Similar for state queue 4407 StateQueueObserverDump observerCopy = mStateQueueObserverDump; 4408 observerCopy.dump(fd); 4409 StateQueueMutatorDump mutatorCopy = mStateQueueMutatorDump; 4410 mutatorCopy.dump(fd); 4411#endif 4412 4413#ifdef TEE_SINK 4414 // Write the tee output to a .wav file 4415 dumpTee(fd, mTeeSource, mId); 4416#endif 4417 4418#ifdef AUDIO_WATCHDOG 4419 if (mAudioWatchdog != 0) { 4420 // Make a non-atomic copy of audio watchdog dump so it won't change underneath us 4421 AudioWatchdogDump wdCopy = mAudioWatchdogDump; 4422 wdCopy.dump(fd); 4423 } 4424#endif 4425} 4426 4427uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const 4428{ 4429 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000) / 2; 4430} 4431 4432uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() const 4433{ 4434 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000); 4435} 4436 4437void AudioFlinger::MixerThread::cacheParameters_l() 4438{ 4439 PlaybackThread::cacheParameters_l(); 4440 4441 // FIXME: Relaxed timing because of a certain device that can't meet latency 4442 // Should be reduced to 2x after the vendor fixes the driver issue 4443 // increase threshold again due to low power audio mode. The way this warning 4444 // threshold is calculated and its usefulness should be reconsidered anyway. 4445 maxPeriod = seconds(mNormalFrameCount) / mSampleRate * 15; 4446} 4447 4448// ---------------------------------------------------------------------------- 4449 4450AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 4451 AudioStreamOut* output, audio_io_handle_t id, audio_devices_t device, bool systemReady) 4452 : PlaybackThread(audioFlinger, output, id, device, DIRECT, systemReady) 4453 // mLeftVolFloat, mRightVolFloat 4454{ 4455} 4456 4457AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 4458 AudioStreamOut* output, audio_io_handle_t id, uint32_t device, 4459 ThreadBase::type_t type, bool systemReady) 4460 : PlaybackThread(audioFlinger, output, id, device, type, systemReady) 4461 // mLeftVolFloat, mRightVolFloat 4462{ 4463} 4464 4465AudioFlinger::DirectOutputThread::~DirectOutputThread() 4466{ 4467} 4468 4469void AudioFlinger::DirectOutputThread::processVolume_l(Track *track, bool lastTrack) 4470{ 4471 audio_track_cblk_t* cblk = track->cblk(); 4472 float left, right; 4473 4474 if (mMasterMute || mStreamTypes[track->streamType()].mute) { 4475 left = right = 0; 4476 } else { 4477 float typeVolume = mStreamTypes[track->streamType()].volume; 4478 float v = mMasterVolume * typeVolume; 4479 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 4480 gain_minifloat_packed_t vlr = proxy->getVolumeLR(); 4481 left = float_from_gain(gain_minifloat_unpack_left(vlr)); 4482 if (left > GAIN_FLOAT_UNITY) { 4483 left = GAIN_FLOAT_UNITY; 4484 } 4485 left *= v; 4486 right = float_from_gain(gain_minifloat_unpack_right(vlr)); 4487 if (right > GAIN_FLOAT_UNITY) { 4488 right = GAIN_FLOAT_UNITY; 4489 } 4490 right *= v; 4491 } 4492 4493 if (lastTrack) { 4494 if (left != mLeftVolFloat || right != mRightVolFloat) { 4495 mLeftVolFloat = left; 4496 mRightVolFloat = right; 4497 4498 // Convert volumes from float to 8.24 4499 uint32_t vl = (uint32_t)(left * (1 << 24)); 4500 uint32_t vr = (uint32_t)(right * (1 << 24)); 4501 4502 // Delegate volume control to effect in track effect chain if needed 4503 // only one effect chain can be present on DirectOutputThread, so if 4504 // there is one, the track is connected to it 4505 if (!mEffectChains.isEmpty()) { 4506 mEffectChains[0]->setVolume_l(&vl, &vr); 4507 left = (float)vl / (1 << 24); 4508 right = (float)vr / (1 << 24); 4509 } 4510 if (mOutput->stream->set_volume) { 4511 mOutput->stream->set_volume(mOutput->stream, left, right); 4512 } 4513 } 4514 } 4515} 4516 4517void AudioFlinger::DirectOutputThread::onAddNewTrack_l() 4518{ 4519 sp<Track> previousTrack = mPreviousTrack.promote(); 4520 sp<Track> latestTrack = mLatestActiveTrack.promote(); 4521 4522 if (previousTrack != 0 && latestTrack != 0) { 4523 if (mType == DIRECT) { 4524 if (previousTrack.get() != latestTrack.get()) { 4525 mFlushPending = true; 4526 } 4527 } else /* mType == OFFLOAD */ { 4528 if (previousTrack->sessionId() != latestTrack->sessionId()) { 4529 mFlushPending = true; 4530 } 4531 } 4532 } 4533 PlaybackThread::onAddNewTrack_l(); 4534} 4535 4536AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l( 4537 Vector< sp<Track> > *tracksToRemove 4538) 4539{ 4540 size_t count = mActiveTracks.size(); 4541 mixer_state mixerStatus = MIXER_IDLE; 4542 bool doHwPause = false; 4543 bool doHwResume = false; 4544 4545 // find out which tracks need to be processed 4546 for (size_t i = 0; i < count; i++) { 4547 sp<Track> t = mActiveTracks[i].promote(); 4548 // The track died recently 4549 if (t == 0) { 4550 continue; 4551 } 4552 4553 if (t->isInvalid()) { 4554 ALOGW("An invalidated track shouldn't be in active list"); 4555 tracksToRemove->add(t); 4556 continue; 4557 } 4558 4559 Track* const track = t.get(); 4560 audio_track_cblk_t* cblk = track->cblk(); 4561 // Only consider last track started for volume and mixer state control. 4562 // In theory an older track could underrun and restart after the new one starts 4563 // but as we only care about the transition phase between two tracks on a 4564 // direct output, it is not a problem to ignore the underrun case. 4565 sp<Track> l = mLatestActiveTrack.promote(); 4566 bool last = l.get() == track; 4567 4568 if (track->isPausing()) { 4569 track->setPaused(); 4570 if (mHwSupportsPause && last && !mHwPaused) { 4571 doHwPause = true; 4572 mHwPaused = true; 4573 } 4574 tracksToRemove->add(track); 4575 } else if (track->isFlushPending()) { 4576 track->flushAck(); 4577 if (last) { 4578 mFlushPending = true; 4579 } 4580 } else if (track->isResumePending()) { 4581 track->resumeAck(); 4582 if (last && mHwPaused) { 4583 doHwResume = true; 4584 mHwPaused = false; 4585 } 4586 } 4587 4588 // The first time a track is added we wait 4589 // for all its buffers to be filled before processing it. 4590 // Allow draining the buffer in case the client 4591 // app does not call stop() and relies on underrun to stop: 4592 // hence the test on (track->mRetryCount > 1). 4593 // If retryCount<=1 then track is about to underrun and be removed. 4594 // Do not use a high threshold for compressed audio. 4595 uint32_t minFrames; 4596 if ((track->sharedBuffer() == 0) && !track->isStopping_1() && !track->isPausing() 4597 && (track->mRetryCount > 1) && audio_is_linear_pcm(mFormat)) { 4598 minFrames = mNormalFrameCount; 4599 } else { 4600 minFrames = 1; 4601 } 4602 4603 if ((track->framesReady() >= minFrames) && track->isReady() && !track->isPaused() && 4604 !track->isStopping_2() && !track->isStopped()) 4605 { 4606 ALOGVV("track %d s=%08x [OK]", track->name(), cblk->mServer); 4607 4608 if (track->mFillingUpStatus == Track::FS_FILLED) { 4609 track->mFillingUpStatus = Track::FS_ACTIVE; 4610 // make sure processVolume_l() will apply new volume even if 0 4611 mLeftVolFloat = mRightVolFloat = -1.0; 4612 if (!mHwSupportsPause) { 4613 track->resumeAck(); 4614 } 4615 } 4616 4617 // compute volume for this track 4618 processVolume_l(track, last); 4619 if (last) { 4620 sp<Track> previousTrack = mPreviousTrack.promote(); 4621 if (previousTrack != 0) { 4622 if (track != previousTrack.get()) { 4623 // Flush any data still being written from last track 4624 mBytesRemaining = 0; 4625 // Invalidate previous track to force a seek when resuming. 4626 previousTrack->invalidate(); 4627 } 4628 } 4629 mPreviousTrack = track; 4630 4631 // reset retry count 4632 track->mRetryCount = kMaxTrackRetriesDirect; 4633 mActiveTrack = t; 4634 mixerStatus = MIXER_TRACKS_READY; 4635 if (mHwPaused) { 4636 doHwResume = true; 4637 mHwPaused = false; 4638 } 4639 } 4640 } else { 4641 // clear effect chain input buffer if the last active track started underruns 4642 // to avoid sending previous audio buffer again to effects 4643 if (!mEffectChains.isEmpty() && last) { 4644 mEffectChains[0]->clearInputBuffer(); 4645 } 4646 if (track->isStopping_1()) { 4647 track->mState = TrackBase::STOPPING_2; 4648 if (last && mHwPaused) { 4649 doHwResume = true; 4650 mHwPaused = false; 4651 } 4652 } 4653 if ((track->sharedBuffer() != 0) || track->isStopped() || 4654 track->isStopping_2() || track->isPaused()) { 4655 // We have consumed all the buffers of this track. 4656 // Remove it from the list of active tracks. 4657 size_t audioHALFrames; 4658 if (audio_is_linear_pcm(mFormat)) { 4659 audioHALFrames = (latency_l() * mSampleRate) / 1000; 4660 } else { 4661 audioHALFrames = 0; 4662 } 4663 4664 size_t framesWritten = mBytesWritten / mFrameSize; 4665 if (mStandby || !last || 4666 track->presentationComplete(framesWritten, audioHALFrames)) { 4667 if (track->isStopping_2()) { 4668 track->mState = TrackBase::STOPPED; 4669 } 4670 if (track->isStopped()) { 4671 track->reset(); 4672 } 4673 tracksToRemove->add(track); 4674 } 4675 } else { 4676 // No buffers for this track. Give it a few chances to 4677 // fill a buffer, then remove it from active list. 4678 // Only consider last track started for mixer state control 4679 if (--(track->mRetryCount) <= 0) { 4680 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 4681 tracksToRemove->add(track); 4682 // indicate to client process that the track was disabled because of underrun; 4683 // it will then automatically call start() when data is available 4684 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 4685 } else if (last) { 4686 ALOGW("pause because of UNDERRUN, framesReady = %zu," 4687 "minFrames = %u, mFormat = %#x", 4688 track->framesReady(), minFrames, mFormat); 4689 mixerStatus = MIXER_TRACKS_ENABLED; 4690 if (mHwSupportsPause && !mHwPaused && !mStandby) { 4691 doHwPause = true; 4692 mHwPaused = true; 4693 } 4694 } 4695 } 4696 } 4697 } 4698 4699 // if an active track did not command a flush, check for pending flush on stopped tracks 4700 if (!mFlushPending) { 4701 for (size_t i = 0; i < mTracks.size(); i++) { 4702 if (mTracks[i]->isFlushPending()) { 4703 mTracks[i]->flushAck(); 4704 mFlushPending = true; 4705 } 4706 } 4707 } 4708 4709 // make sure the pause/flush/resume sequence is executed in the right order. 4710 // If a flush is pending and a track is active but the HW is not paused, force a HW pause 4711 // before flush and then resume HW. This can happen in case of pause/flush/resume 4712 // if resume is received before pause is executed. 4713 if (mHwSupportsPause && !mStandby && 4714 (doHwPause || (mFlushPending && !mHwPaused && (count != 0)))) { 4715 mOutput->stream->pause(mOutput->stream); 4716 } 4717 if (mFlushPending) { 4718 flushHw_l(); 4719 } 4720 if (mHwSupportsPause && !mStandby && doHwResume) { 4721 mOutput->stream->resume(mOutput->stream); 4722 } 4723 // remove all the tracks that need to be... 4724 removeTracks_l(*tracksToRemove); 4725 4726 return mixerStatus; 4727} 4728 4729void AudioFlinger::DirectOutputThread::threadLoop_mix() 4730{ 4731 size_t frameCount = mFrameCount; 4732 int8_t *curBuf = (int8_t *)mSinkBuffer; 4733 // output audio to hardware 4734 while (frameCount) { 4735 AudioBufferProvider::Buffer buffer; 4736 buffer.frameCount = frameCount; 4737 status_t status = mActiveTrack->getNextBuffer(&buffer); 4738 if (status != NO_ERROR || buffer.raw == NULL) { 4739 memset(curBuf, 0, frameCount * mFrameSize); 4740 break; 4741 } 4742 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 4743 frameCount -= buffer.frameCount; 4744 curBuf += buffer.frameCount * mFrameSize; 4745 mActiveTrack->releaseBuffer(&buffer); 4746 } 4747 mCurrentWriteLength = curBuf - (int8_t *)mSinkBuffer; 4748 mSleepTimeUs = 0; 4749 mStandbyTimeNs = systemTime() + mStandbyDelayNs; 4750 mActiveTrack.clear(); 4751} 4752 4753void AudioFlinger::DirectOutputThread::threadLoop_sleepTime() 4754{ 4755 // do not write to HAL when paused 4756 if (mHwPaused || (usesHwAvSync() && mStandby)) { 4757 mSleepTimeUs = mIdleSleepTimeUs; 4758 return; 4759 } 4760 if (mSleepTimeUs == 0) { 4761 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4762 mSleepTimeUs = mActiveSleepTimeUs; 4763 } else { 4764 mSleepTimeUs = mIdleSleepTimeUs; 4765 } 4766 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 4767 memset(mSinkBuffer, 0, mFrameCount * mFrameSize); 4768 mSleepTimeUs = 0; 4769 } 4770} 4771 4772void AudioFlinger::DirectOutputThread::threadLoop_exit() 4773{ 4774 { 4775 Mutex::Autolock _l(mLock); 4776 for (size_t i = 0; i < mTracks.size(); i++) { 4777 if (mTracks[i]->isFlushPending()) { 4778 mTracks[i]->flushAck(); 4779 mFlushPending = true; 4780 } 4781 } 4782 if (mFlushPending) { 4783 flushHw_l(); 4784 } 4785 } 4786 PlaybackThread::threadLoop_exit(); 4787} 4788 4789// must be called with thread mutex locked 4790bool AudioFlinger::DirectOutputThread::shouldStandby_l() 4791{ 4792 bool trackPaused = false; 4793 bool trackStopped = false; 4794 4795 // do not put the HAL in standby when paused. AwesomePlayer clear the offloaded AudioTrack 4796 // after a timeout and we will enter standby then. 4797 if (mTracks.size() > 0) { 4798 trackPaused = mTracks[mTracks.size() - 1]->isPaused(); 4799 trackStopped = mTracks[mTracks.size() - 1]->isStopped() || 4800 mTracks[mTracks.size() - 1]->mState == TrackBase::IDLE; 4801 } 4802 4803 return !mStandby && !(trackPaused || (mHwPaused && !trackStopped)); 4804} 4805 4806// getTrackName_l() must be called with ThreadBase::mLock held 4807int AudioFlinger::DirectOutputThread::getTrackName_l(audio_channel_mask_t channelMask __unused, 4808 audio_format_t format __unused, int sessionId __unused) 4809{ 4810 return 0; 4811} 4812 4813// deleteTrackName_l() must be called with ThreadBase::mLock held 4814void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name __unused) 4815{ 4816} 4817 4818// checkForNewParameter_l() must be called with ThreadBase::mLock held 4819bool AudioFlinger::DirectOutputThread::checkForNewParameter_l(const String8& keyValuePair, 4820 status_t& status) 4821{ 4822 bool reconfig = false; 4823 4824 status = NO_ERROR; 4825 4826 AudioParameter param = AudioParameter(keyValuePair); 4827 int value; 4828 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 4829 // forward device change to effects that have requested to be 4830 // aware of attached audio device. 4831 if (value != AUDIO_DEVICE_NONE) { 4832 mOutDevice = value; 4833 for (size_t i = 0; i < mEffectChains.size(); i++) { 4834 mEffectChains[i]->setDevice_l(mOutDevice); 4835 } 4836 } 4837 } 4838 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 4839 // do not accept frame count changes if tracks are open as the track buffer 4840 // size depends on frame count and correct behavior would not be garantied 4841 // if frame count is changed after track creation 4842 if (!mTracks.isEmpty()) { 4843 status = INVALID_OPERATION; 4844 } else { 4845 reconfig = true; 4846 } 4847 } 4848 if (status == NO_ERROR) { 4849 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 4850 keyValuePair.string()); 4851 if (!mStandby && status == INVALID_OPERATION) { 4852 mOutput->standby(); 4853 mStandby = true; 4854 mBytesWritten = 0; 4855 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 4856 keyValuePair.string()); 4857 } 4858 if (status == NO_ERROR && reconfig) { 4859 readOutputParameters_l(); 4860 sendIoConfigEvent_l(AUDIO_OUTPUT_CONFIG_CHANGED); 4861 } 4862 } 4863 4864 return reconfig; 4865} 4866 4867uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() const 4868{ 4869 uint32_t time; 4870 if (audio_is_linear_pcm(mFormat)) { 4871 time = PlaybackThread::activeSleepTimeUs(); 4872 } else { 4873 time = 10000; 4874 } 4875 return time; 4876} 4877 4878uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() const 4879{ 4880 uint32_t time; 4881 if (audio_is_linear_pcm(mFormat)) { 4882 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 4883 } else { 4884 time = 10000; 4885 } 4886 return time; 4887} 4888 4889uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() const 4890{ 4891 uint32_t time; 4892 if (audio_is_linear_pcm(mFormat)) { 4893 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 4894 } else { 4895 time = 10000; 4896 } 4897 return time; 4898} 4899 4900void AudioFlinger::DirectOutputThread::cacheParameters_l() 4901{ 4902 PlaybackThread::cacheParameters_l(); 4903 4904 // use shorter standby delay as on normal output to release 4905 // hardware resources as soon as possible 4906 // no delay on outputs with HW A/V sync 4907 if (usesHwAvSync()) { 4908 mStandbyDelayNs = 0; 4909 } else if ((mType == OFFLOAD) && !audio_is_linear_pcm(mFormat)) { 4910 mStandbyDelayNs = kOffloadStandbyDelayNs; 4911 } else { 4912 mStandbyDelayNs = microseconds(mActiveSleepTimeUs*2); 4913 } 4914} 4915 4916void AudioFlinger::DirectOutputThread::flushHw_l() 4917{ 4918 mOutput->flush(); 4919 mHwPaused = false; 4920 mFlushPending = false; 4921} 4922 4923// ---------------------------------------------------------------------------- 4924 4925AudioFlinger::AsyncCallbackThread::AsyncCallbackThread( 4926 const wp<AudioFlinger::PlaybackThread>& playbackThread) 4927 : Thread(false /*canCallJava*/), 4928 mPlaybackThread(playbackThread), 4929 mWriteAckSequence(0), 4930 mDrainSequence(0) 4931{ 4932} 4933 4934AudioFlinger::AsyncCallbackThread::~AsyncCallbackThread() 4935{ 4936} 4937 4938void AudioFlinger::AsyncCallbackThread::onFirstRef() 4939{ 4940 run("Offload Cbk", ANDROID_PRIORITY_URGENT_AUDIO); 4941} 4942 4943bool AudioFlinger::AsyncCallbackThread::threadLoop() 4944{ 4945 while (!exitPending()) { 4946 uint32_t writeAckSequence; 4947 uint32_t drainSequence; 4948 4949 { 4950 Mutex::Autolock _l(mLock); 4951 while (!((mWriteAckSequence & 1) || 4952 (mDrainSequence & 1) || 4953 exitPending())) { 4954 mWaitWorkCV.wait(mLock); 4955 } 4956 4957 if (exitPending()) { 4958 break; 4959 } 4960 ALOGV("AsyncCallbackThread mWriteAckSequence %d mDrainSequence %d", 4961 mWriteAckSequence, mDrainSequence); 4962 writeAckSequence = mWriteAckSequence; 4963 mWriteAckSequence &= ~1; 4964 drainSequence = mDrainSequence; 4965 mDrainSequence &= ~1; 4966 } 4967 { 4968 sp<AudioFlinger::PlaybackThread> playbackThread = mPlaybackThread.promote(); 4969 if (playbackThread != 0) { 4970 if (writeAckSequence & 1) { 4971 playbackThread->resetWriteBlocked(writeAckSequence >> 1); 4972 } 4973 if (drainSequence & 1) { 4974 playbackThread->resetDraining(drainSequence >> 1); 4975 } 4976 } 4977 } 4978 } 4979 return false; 4980} 4981 4982void AudioFlinger::AsyncCallbackThread::exit() 4983{ 4984 ALOGV("AsyncCallbackThread::exit"); 4985 Mutex::Autolock _l(mLock); 4986 requestExit(); 4987 mWaitWorkCV.broadcast(); 4988} 4989 4990void AudioFlinger::AsyncCallbackThread::setWriteBlocked(uint32_t sequence) 4991{ 4992 Mutex::Autolock _l(mLock); 4993 // bit 0 is cleared 4994 mWriteAckSequence = sequence << 1; 4995} 4996 4997void AudioFlinger::AsyncCallbackThread::resetWriteBlocked() 4998{ 4999 Mutex::Autolock _l(mLock); 5000 // ignore unexpected callbacks 5001 if (mWriteAckSequence & 2) { 5002 mWriteAckSequence |= 1; 5003 mWaitWorkCV.signal(); 5004 } 5005} 5006 5007void AudioFlinger::AsyncCallbackThread::setDraining(uint32_t sequence) 5008{ 5009 Mutex::Autolock _l(mLock); 5010 // bit 0 is cleared 5011 mDrainSequence = sequence << 1; 5012} 5013 5014void AudioFlinger::AsyncCallbackThread::resetDraining() 5015{ 5016 Mutex::Autolock _l(mLock); 5017 // ignore unexpected callbacks 5018 if (mDrainSequence & 2) { 5019 mDrainSequence |= 1; 5020 mWaitWorkCV.signal(); 5021 } 5022} 5023 5024 5025// ---------------------------------------------------------------------------- 5026AudioFlinger::OffloadThread::OffloadThread(const sp<AudioFlinger>& audioFlinger, 5027 AudioStreamOut* output, audio_io_handle_t id, uint32_t device, bool systemReady) 5028 : DirectOutputThread(audioFlinger, output, id, device, OFFLOAD, systemReady), 5029 mPausedBytesRemaining(0) 5030{ 5031 //FIXME: mStandby should be set to true by ThreadBase constructor 5032 mStandby = true; 5033} 5034 5035void AudioFlinger::OffloadThread::threadLoop_exit() 5036{ 5037 if (mFlushPending || mHwPaused) { 5038 // If a flush is pending or track was paused, just discard buffered data 5039 flushHw_l(); 5040 } else { 5041 mMixerStatus = MIXER_DRAIN_ALL; 5042 threadLoop_drain(); 5043 } 5044 if (mUseAsyncWrite) { 5045 ALOG_ASSERT(mCallbackThread != 0); 5046 mCallbackThread->exit(); 5047 } 5048 PlaybackThread::threadLoop_exit(); 5049} 5050 5051AudioFlinger::PlaybackThread::mixer_state AudioFlinger::OffloadThread::prepareTracks_l( 5052 Vector< sp<Track> > *tracksToRemove 5053) 5054{ 5055 size_t count = mActiveTracks.size(); 5056 5057 mixer_state mixerStatus = MIXER_IDLE; 5058 bool doHwPause = false; 5059 bool doHwResume = false; 5060 5061 ALOGV("OffloadThread::prepareTracks_l active tracks %d", count); 5062 5063 // find out which tracks need to be processed 5064 for (size_t i = 0; i < count; i++) { 5065 sp<Track> t = mActiveTracks[i].promote(); 5066 // The track died recently 5067 if (t == 0) { 5068 continue; 5069 } 5070 Track* const track = t.get(); 5071 audio_track_cblk_t* cblk = track->cblk(); 5072 // Only consider last track started for volume and mixer state control. 5073 // In theory an older track could underrun and restart after the new one starts 5074 // but as we only care about the transition phase between two tracks on a 5075 // direct output, it is not a problem to ignore the underrun case. 5076 sp<Track> l = mLatestActiveTrack.promote(); 5077 bool last = l.get() == track; 5078 5079 if (track->isInvalid()) { 5080 ALOGW("An invalidated track shouldn't be in active list"); 5081 tracksToRemove->add(track); 5082 continue; 5083 } 5084 5085 if (track->mState == TrackBase::IDLE) { 5086 ALOGW("An idle track shouldn't be in active list"); 5087 continue; 5088 } 5089 5090 if (track->isPausing()) { 5091 track->setPaused(); 5092 if (last) { 5093 if (mHwSupportsPause && !mHwPaused) { 5094 doHwPause = true; 5095 mHwPaused = true; 5096 } 5097 // If we were part way through writing the mixbuffer to 5098 // the HAL we must save this until we resume 5099 // BUG - this will be wrong if a different track is made active, 5100 // in that case we want to discard the pending data in the 5101 // mixbuffer and tell the client to present it again when the 5102 // track is resumed 5103 mPausedWriteLength = mCurrentWriteLength; 5104 mPausedBytesRemaining = mBytesRemaining; 5105 mBytesRemaining = 0; // stop writing 5106 } 5107 tracksToRemove->add(track); 5108 } else if (track->isFlushPending()) { 5109 track->flushAck(); 5110 if (last) { 5111 mFlushPending = true; 5112 } 5113 } else if (track->isResumePending()){ 5114 track->resumeAck(); 5115 if (last) { 5116 if (mPausedBytesRemaining) { 5117 // Need to continue write that was interrupted 5118 mCurrentWriteLength = mPausedWriteLength; 5119 mBytesRemaining = mPausedBytesRemaining; 5120 mPausedBytesRemaining = 0; 5121 } 5122 if (mHwPaused) { 5123 doHwResume = true; 5124 mHwPaused = false; 5125 // threadLoop_mix() will handle the case that we need to 5126 // resume an interrupted write 5127 } 5128 // enable write to audio HAL 5129 mSleepTimeUs = 0; 5130 5131 // Do not handle new data in this iteration even if track->framesReady() 5132 mixerStatus = MIXER_TRACKS_ENABLED; 5133 } 5134 } else if (track->framesReady() && track->isReady() && 5135 !track->isPaused() && !track->isTerminated() && !track->isStopping_2()) { 5136 ALOGVV("OffloadThread: track %d s=%08x [OK]", track->name(), cblk->mServer); 5137 if (track->mFillingUpStatus == Track::FS_FILLED) { 5138 track->mFillingUpStatus = Track::FS_ACTIVE; 5139 // make sure processVolume_l() will apply new volume even if 0 5140 mLeftVolFloat = mRightVolFloat = -1.0; 5141 } 5142 5143 if (last) { 5144 sp<Track> previousTrack = mPreviousTrack.promote(); 5145 if (previousTrack != 0) { 5146 if (track != previousTrack.get()) { 5147 // Flush any data still being written from last track 5148 mBytesRemaining = 0; 5149 if (mPausedBytesRemaining) { 5150 // Last track was paused so we also need to flush saved 5151 // mixbuffer state and invalidate track so that it will 5152 // re-submit that unwritten data when it is next resumed 5153 mPausedBytesRemaining = 0; 5154 // Invalidate is a bit drastic - would be more efficient 5155 // to have a flag to tell client that some of the 5156 // previously written data was lost 5157 previousTrack->invalidate(); 5158 } 5159 // flush data already sent to the DSP if changing audio session as audio 5160 // comes from a different source. Also invalidate previous track to force a 5161 // seek when resuming. 5162 if (previousTrack->sessionId() != track->sessionId()) { 5163 previousTrack->invalidate(); 5164 } 5165 } 5166 } 5167 mPreviousTrack = track; 5168 // reset retry count 5169 track->mRetryCount = kMaxTrackRetriesOffload; 5170 mActiveTrack = t; 5171 mixerStatus = MIXER_TRACKS_READY; 5172 } 5173 } else { 5174 ALOGVV("OffloadThread: track %d s=%08x [NOT READY]", track->name(), cblk->mServer); 5175 if (track->isStopping_1()) { 5176 // Hardware buffer can hold a large amount of audio so we must 5177 // wait for all current track's data to drain before we say 5178 // that the track is stopped. 5179 if (mBytesRemaining == 0) { 5180 // Only start draining when all data in mixbuffer 5181 // has been written 5182 ALOGV("OffloadThread: underrun and STOPPING_1 -> draining, STOPPING_2"); 5183 track->mState = TrackBase::STOPPING_2; // so presentation completes after drain 5184 // do not drain if no data was ever sent to HAL (mStandby == true) 5185 if (last && !mStandby) { 5186 // do not modify drain sequence if we are already draining. This happens 5187 // when resuming from pause after drain. 5188 if ((mDrainSequence & 1) == 0) { 5189 mSleepTimeUs = 0; 5190 mStandbyTimeNs = systemTime() + mStandbyDelayNs; 5191 mixerStatus = MIXER_DRAIN_TRACK; 5192 mDrainSequence += 2; 5193 } 5194 if (mHwPaused) { 5195 // It is possible to move from PAUSED to STOPPING_1 without 5196 // a resume so we must ensure hardware is running 5197 doHwResume = true; 5198 mHwPaused = false; 5199 } 5200 } 5201 } 5202 } else if (track->isStopping_2()) { 5203 // Drain has completed or we are in standby, signal presentation complete 5204 if (!(mDrainSequence & 1) || !last || mStandby) { 5205 track->mState = TrackBase::STOPPED; 5206 size_t audioHALFrames = 5207 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 5208 size_t framesWritten = 5209 mBytesWritten / mOutput->getFrameSize(); 5210 track->presentationComplete(framesWritten, audioHALFrames); 5211 track->reset(); 5212 tracksToRemove->add(track); 5213 } 5214 } else { 5215 // No buffers for this track. Give it a few chances to 5216 // fill a buffer, then remove it from active list. 5217 if (--(track->mRetryCount) <= 0) { 5218 ALOGV("OffloadThread: BUFFER TIMEOUT: remove(%d) from active list", 5219 track->name()); 5220 tracksToRemove->add(track); 5221 // indicate to client process that the track was disabled because of underrun; 5222 // it will then automatically call start() when data is available 5223 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 5224 } else if (last){ 5225 mixerStatus = MIXER_TRACKS_ENABLED; 5226 } 5227 } 5228 } 5229 // compute volume for this track 5230 processVolume_l(track, last); 5231 } 5232 5233 // make sure the pause/flush/resume sequence is executed in the right order. 5234 // If a flush is pending and a track is active but the HW is not paused, force a HW pause 5235 // before flush and then resume HW. This can happen in case of pause/flush/resume 5236 // if resume is received before pause is executed. 5237 if (!mStandby && (doHwPause || (mFlushPending && !mHwPaused && (count != 0)))) { 5238 mOutput->stream->pause(mOutput->stream); 5239 } 5240 if (mFlushPending) { 5241 flushHw_l(); 5242 } 5243 if (!mStandby && doHwResume) { 5244 mOutput->stream->resume(mOutput->stream); 5245 } 5246 5247 // remove all the tracks that need to be... 5248 removeTracks_l(*tracksToRemove); 5249 5250 return mixerStatus; 5251} 5252 5253// must be called with thread mutex locked 5254bool AudioFlinger::OffloadThread::waitingAsyncCallback_l() 5255{ 5256 ALOGVV("waitingAsyncCallback_l mWriteAckSequence %d mDrainSequence %d", 5257 mWriteAckSequence, mDrainSequence); 5258 if (mUseAsyncWrite && ((mWriteAckSequence & 1) || (mDrainSequence & 1))) { 5259 return true; 5260 } 5261 return false; 5262} 5263 5264bool AudioFlinger::OffloadThread::waitingAsyncCallback() 5265{ 5266 Mutex::Autolock _l(mLock); 5267 return waitingAsyncCallback_l(); 5268} 5269 5270void AudioFlinger::OffloadThread::flushHw_l() 5271{ 5272 DirectOutputThread::flushHw_l(); 5273 // Flush anything still waiting in the mixbuffer 5274 mCurrentWriteLength = 0; 5275 mBytesRemaining = 0; 5276 mPausedWriteLength = 0; 5277 mPausedBytesRemaining = 0; 5278 5279 if (mUseAsyncWrite) { 5280 // discard any pending drain or write ack by incrementing sequence 5281 mWriteAckSequence = (mWriteAckSequence + 2) & ~1; 5282 mDrainSequence = (mDrainSequence + 2) & ~1; 5283 ALOG_ASSERT(mCallbackThread != 0); 5284 mCallbackThread->setWriteBlocked(mWriteAckSequence); 5285 mCallbackThread->setDraining(mDrainSequence); 5286 } 5287} 5288 5289// ---------------------------------------------------------------------------- 5290 5291AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 5292 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id, bool systemReady) 5293 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->outDevice(), 5294 systemReady, DUPLICATING), 5295 mWaitTimeMs(UINT_MAX) 5296{ 5297 addOutputTrack(mainThread); 5298} 5299 5300AudioFlinger::DuplicatingThread::~DuplicatingThread() 5301{ 5302 for (size_t i = 0; i < mOutputTracks.size(); i++) { 5303 mOutputTracks[i]->destroy(); 5304 } 5305} 5306 5307void AudioFlinger::DuplicatingThread::threadLoop_mix() 5308{ 5309 // mix buffers... 5310 if (outputsReady(outputTracks)) { 5311 mAudioMixer->process(AudioBufferProvider::kInvalidPTS); 5312 } else { 5313 if (mMixerBufferValid) { 5314 memset(mMixerBuffer, 0, mMixerBufferSize); 5315 } else { 5316 memset(mSinkBuffer, 0, mSinkBufferSize); 5317 } 5318 } 5319 mSleepTimeUs = 0; 5320 writeFrames = mNormalFrameCount; 5321 mCurrentWriteLength = mSinkBufferSize; 5322 mStandbyTimeNs = systemTime() + mStandbyDelayNs; 5323} 5324 5325void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() 5326{ 5327 if (mSleepTimeUs == 0) { 5328 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 5329 mSleepTimeUs = mActiveSleepTimeUs; 5330 } else { 5331 mSleepTimeUs = mIdleSleepTimeUs; 5332 } 5333 } else if (mBytesWritten != 0) { 5334 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 5335 writeFrames = mNormalFrameCount; 5336 memset(mSinkBuffer, 0, mSinkBufferSize); 5337 } else { 5338 // flush remaining overflow buffers in output tracks 5339 writeFrames = 0; 5340 } 5341 mSleepTimeUs = 0; 5342 } 5343} 5344 5345ssize_t AudioFlinger::DuplicatingThread::threadLoop_write() 5346{ 5347 for (size_t i = 0; i < outputTracks.size(); i++) { 5348 outputTracks[i]->write(mSinkBuffer, writeFrames); 5349 } 5350 mStandby = false; 5351 return (ssize_t)mSinkBufferSize; 5352} 5353 5354void AudioFlinger::DuplicatingThread::threadLoop_standby() 5355{ 5356 // DuplicatingThread implements standby by stopping all tracks 5357 for (size_t i = 0; i < outputTracks.size(); i++) { 5358 outputTracks[i]->stop(); 5359 } 5360} 5361 5362void AudioFlinger::DuplicatingThread::saveOutputTracks() 5363{ 5364 outputTracks = mOutputTracks; 5365} 5366 5367void AudioFlinger::DuplicatingThread::clearOutputTracks() 5368{ 5369 outputTracks.clear(); 5370} 5371 5372void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 5373{ 5374 Mutex::Autolock _l(mLock); 5375 // The downstream MixerThread consumes thread->frameCount() amount of frames per mix pass. 5376 // Adjust for thread->sampleRate() to determine minimum buffer frame count. 5377 // Then triple buffer because Threads do not run synchronously and may not be clock locked. 5378 const size_t frameCount = 5379 3 * sourceFramesNeeded(mSampleRate, thread->frameCount(), thread->sampleRate()); 5380 // TODO: Consider asynchronous sample rate conversion to handle clock disparity 5381 // from different OutputTracks and their associated MixerThreads (e.g. one may 5382 // nearly empty and the other may be dropping data). 5383 5384 sp<OutputTrack> outputTrack = new OutputTrack(thread, 5385 this, 5386 mSampleRate, 5387 mFormat, 5388 mChannelMask, 5389 frameCount, 5390 IPCThreadState::self()->getCallingUid()); 5391 if (outputTrack->cblk() != NULL) { 5392 thread->setStreamVolume(AUDIO_STREAM_PATCH, 1.0f); 5393 mOutputTracks.add(outputTrack); 5394 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack.get(), thread); 5395 updateWaitTime_l(); 5396 } 5397} 5398 5399void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 5400{ 5401 Mutex::Autolock _l(mLock); 5402 for (size_t i = 0; i < mOutputTracks.size(); i++) { 5403 if (mOutputTracks[i]->thread() == thread) { 5404 mOutputTracks[i]->destroy(); 5405 mOutputTracks.removeAt(i); 5406 updateWaitTime_l(); 5407 if (thread->getOutput() == mOutput) { 5408 mOutput = NULL; 5409 } 5410 return; 5411 } 5412 } 5413 ALOGV("removeOutputTrack(): unknown thread: %p", thread); 5414} 5415 5416// caller must hold mLock 5417void AudioFlinger::DuplicatingThread::updateWaitTime_l() 5418{ 5419 mWaitTimeMs = UINT_MAX; 5420 for (size_t i = 0; i < mOutputTracks.size(); i++) { 5421 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 5422 if (strong != 0) { 5423 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 5424 if (waitTimeMs < mWaitTimeMs) { 5425 mWaitTimeMs = waitTimeMs; 5426 } 5427 } 5428 } 5429} 5430 5431 5432bool AudioFlinger::DuplicatingThread::outputsReady( 5433 const SortedVector< sp<OutputTrack> > &outputTracks) 5434{ 5435 for (size_t i = 0; i < outputTracks.size(); i++) { 5436 sp<ThreadBase> thread = outputTracks[i]->thread().promote(); 5437 if (thread == 0) { 5438 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", 5439 outputTracks[i].get()); 5440 return false; 5441 } 5442 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 5443 // see note at standby() declaration 5444 if (playbackThread->standby() && !playbackThread->isSuspended()) { 5445 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), 5446 thread.get()); 5447 return false; 5448 } 5449 } 5450 return true; 5451} 5452 5453uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() const 5454{ 5455 return (mWaitTimeMs * 1000) / 2; 5456} 5457 5458void AudioFlinger::DuplicatingThread::cacheParameters_l() 5459{ 5460 // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first 5461 updateWaitTime_l(); 5462 5463 MixerThread::cacheParameters_l(); 5464} 5465 5466// ---------------------------------------------------------------------------- 5467// Record 5468// ---------------------------------------------------------------------------- 5469 5470AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 5471 AudioStreamIn *input, 5472 audio_io_handle_t id, 5473 audio_devices_t outDevice, 5474 audio_devices_t inDevice, 5475 bool systemReady 5476#ifdef TEE_SINK 5477 , const sp<NBAIO_Sink>& teeSink 5478#endif 5479 ) : 5480 ThreadBase(audioFlinger, id, outDevice, inDevice, RECORD, systemReady), 5481 mInput(input), mActiveTracksGen(0), mRsmpInBuffer(NULL), 5482 // mRsmpInFrames and mRsmpInFramesP2 are set by readInputParameters_l() 5483 mRsmpInRear(0) 5484#ifdef TEE_SINK 5485 , mTeeSink(teeSink) 5486#endif 5487 , mReadOnlyHeap(new MemoryDealer(kRecordThreadReadOnlyHeapSize, 5488 "RecordThreadRO", MemoryHeapBase::READ_ONLY)) 5489 // mFastCapture below 5490 , mFastCaptureFutex(0) 5491 // mInputSource 5492 // mPipeSink 5493 // mPipeSource 5494 , mPipeFramesP2(0) 5495 // mPipeMemory 5496 // mFastCaptureNBLogWriter 5497 , mFastTrackAvail(false) 5498{ 5499 snprintf(mThreadName, kThreadNameLength, "AudioIn_%X", id); 5500 mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mThreadName); 5501 5502 readInputParameters_l(); 5503 5504 // create an NBAIO source for the HAL input stream, and negotiate 5505 mInputSource = new AudioStreamInSource(input->stream); 5506 size_t numCounterOffers = 0; 5507 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount, mFormat)}; 5508 ssize_t index = mInputSource->negotiate(offers, 1, NULL, numCounterOffers); 5509 ALOG_ASSERT(index == 0); 5510 5511 // initialize fast capture depending on configuration 5512 bool initFastCapture; 5513 switch (kUseFastCapture) { 5514 case FastCapture_Never: 5515 initFastCapture = false; 5516 break; 5517 case FastCapture_Always: 5518 initFastCapture = true; 5519 break; 5520 case FastCapture_Static: 5521 initFastCapture = (mFrameCount * 1000) / mSampleRate < kMinNormalCaptureBufferSizeMs; 5522 break; 5523 // case FastCapture_Dynamic: 5524 } 5525 5526 if (initFastCapture) { 5527 // create a Pipe for FastCapture to write to, and for us and fast tracks to read from 5528 NBAIO_Format format = mInputSource->format(); 5529 size_t pipeFramesP2 = roundup(mSampleRate / 25); // double-buffering of 20 ms each 5530 size_t pipeSize = pipeFramesP2 * Format_frameSize(format); 5531 void *pipeBuffer; 5532 const sp<MemoryDealer> roHeap(readOnlyHeap()); 5533 sp<IMemory> pipeMemory; 5534 if ((roHeap == 0) || 5535 (pipeMemory = roHeap->allocate(pipeSize)) == 0 || 5536 (pipeBuffer = pipeMemory->pointer()) == NULL) { 5537 ALOGE("not enough memory for pipe buffer size=%zu", pipeSize); 5538 goto failed; 5539 } 5540 // pipe will be shared directly with fast clients, so clear to avoid leaking old information 5541 memset(pipeBuffer, 0, pipeSize); 5542 Pipe *pipe = new Pipe(pipeFramesP2, format, pipeBuffer); 5543 const NBAIO_Format offers[1] = {format}; 5544 size_t numCounterOffers = 0; 5545 ssize_t index = pipe->negotiate(offers, 1, NULL, numCounterOffers); 5546 ALOG_ASSERT(index == 0); 5547 mPipeSink = pipe; 5548 PipeReader *pipeReader = new PipeReader(*pipe); 5549 numCounterOffers = 0; 5550 index = pipeReader->negotiate(offers, 1, NULL, numCounterOffers); 5551 ALOG_ASSERT(index == 0); 5552 mPipeSource = pipeReader; 5553 mPipeFramesP2 = pipeFramesP2; 5554 mPipeMemory = pipeMemory; 5555 5556 // create fast capture 5557 mFastCapture = new FastCapture(); 5558 FastCaptureStateQueue *sq = mFastCapture->sq(); 5559#ifdef STATE_QUEUE_DUMP 5560 // FIXME 5561#endif 5562 FastCaptureState *state = sq->begin(); 5563 state->mCblk = NULL; 5564 state->mInputSource = mInputSource.get(); 5565 state->mInputSourceGen++; 5566 state->mPipeSink = pipe; 5567 state->mPipeSinkGen++; 5568 state->mFrameCount = mFrameCount; 5569 state->mCommand = FastCaptureState::COLD_IDLE; 5570 // already done in constructor initialization list 5571 //mFastCaptureFutex = 0; 5572 state->mColdFutexAddr = &mFastCaptureFutex; 5573 state->mColdGen++; 5574 state->mDumpState = &mFastCaptureDumpState; 5575#ifdef TEE_SINK 5576 // FIXME 5577#endif 5578 mFastCaptureNBLogWriter = audioFlinger->newWriter_l(kFastCaptureLogSize, "FastCapture"); 5579 state->mNBLogWriter = mFastCaptureNBLogWriter.get(); 5580 sq->end(); 5581 sq->push(FastCaptureStateQueue::BLOCK_UNTIL_PUSHED); 5582 5583 // start the fast capture 5584 mFastCapture->run("FastCapture", ANDROID_PRIORITY_URGENT_AUDIO); 5585 pid_t tid = mFastCapture->getTid(); 5586 sendPrioConfigEvent(getpid_cached, tid, kPriorityFastMixer); 5587#ifdef AUDIO_WATCHDOG 5588 // FIXME 5589#endif 5590 5591 mFastTrackAvail = true; 5592 } 5593failed: ; 5594 5595 // FIXME mNormalSource 5596} 5597 5598AudioFlinger::RecordThread::~RecordThread() 5599{ 5600 if (mFastCapture != 0) { 5601 FastCaptureStateQueue *sq = mFastCapture->sq(); 5602 FastCaptureState *state = sq->begin(); 5603 if (state->mCommand == FastCaptureState::COLD_IDLE) { 5604 int32_t old = android_atomic_inc(&mFastCaptureFutex); 5605 if (old == -1) { 5606 (void) syscall(__NR_futex, &mFastCaptureFutex, FUTEX_WAKE_PRIVATE, 1); 5607 } 5608 } 5609 state->mCommand = FastCaptureState::EXIT; 5610 sq->end(); 5611 sq->push(FastCaptureStateQueue::BLOCK_UNTIL_PUSHED); 5612 mFastCapture->join(); 5613 mFastCapture.clear(); 5614 } 5615 mAudioFlinger->unregisterWriter(mFastCaptureNBLogWriter); 5616 mAudioFlinger->unregisterWriter(mNBLogWriter); 5617 free(mRsmpInBuffer); 5618} 5619 5620void AudioFlinger::RecordThread::onFirstRef() 5621{ 5622 run(mThreadName, PRIORITY_URGENT_AUDIO); 5623} 5624 5625bool AudioFlinger::RecordThread::threadLoop() 5626{ 5627 nsecs_t lastWarning = 0; 5628 5629 inputStandBy(); 5630 5631reacquire_wakelock: 5632 sp<RecordTrack> activeTrack; 5633 int activeTracksGen; 5634 { 5635 Mutex::Autolock _l(mLock); 5636 size_t size = mActiveTracks.size(); 5637 activeTracksGen = mActiveTracksGen; 5638 if (size > 0) { 5639 // FIXME an arbitrary choice 5640 activeTrack = mActiveTracks[0]; 5641 acquireWakeLock_l(activeTrack->uid()); 5642 if (size > 1) { 5643 SortedVector<int> tmp; 5644 for (size_t i = 0; i < size; i++) { 5645 tmp.add(mActiveTracks[i]->uid()); 5646 } 5647 updateWakeLockUids_l(tmp); 5648 } 5649 } else { 5650 acquireWakeLock_l(-1); 5651 } 5652 } 5653 5654 // used to request a deferred sleep, to be executed later while mutex is unlocked 5655 uint32_t sleepUs = 0; 5656 5657 // loop while there is work to do 5658 for (;;) { 5659 Vector< sp<EffectChain> > effectChains; 5660 5661 // sleep with mutex unlocked 5662 if (sleepUs > 0) { 5663 ATRACE_BEGIN("sleep"); 5664 usleep(sleepUs); 5665 ATRACE_END(); 5666 sleepUs = 0; 5667 } 5668 5669 // activeTracks accumulates a copy of a subset of mActiveTracks 5670 Vector< sp<RecordTrack> > activeTracks; 5671 5672 // reference to the (first and only) active fast track 5673 sp<RecordTrack> fastTrack; 5674 5675 // reference to a fast track which is about to be removed 5676 sp<RecordTrack> fastTrackToRemove; 5677 5678 { // scope for mLock 5679 Mutex::Autolock _l(mLock); 5680 5681 processConfigEvents_l(); 5682 5683 // check exitPending here because checkForNewParameters_l() and 5684 // checkForNewParameters_l() can temporarily release mLock 5685 if (exitPending()) { 5686 break; 5687 } 5688 5689 // if no active track(s), then standby and release wakelock 5690 size_t size = mActiveTracks.size(); 5691 if (size == 0) { 5692 standbyIfNotAlreadyInStandby(); 5693 // exitPending() can't become true here 5694 releaseWakeLock_l(); 5695 ALOGV("RecordThread: loop stopping"); 5696 // go to sleep 5697 mWaitWorkCV.wait(mLock); 5698 ALOGV("RecordThread: loop starting"); 5699 goto reacquire_wakelock; 5700 } 5701 5702 if (mActiveTracksGen != activeTracksGen) { 5703 activeTracksGen = mActiveTracksGen; 5704 SortedVector<int> tmp; 5705 for (size_t i = 0; i < size; i++) { 5706 tmp.add(mActiveTracks[i]->uid()); 5707 } 5708 updateWakeLockUids_l(tmp); 5709 } 5710 5711 bool doBroadcast = false; 5712 for (size_t i = 0; i < size; ) { 5713 5714 activeTrack = mActiveTracks[i]; 5715 if (activeTrack->isTerminated()) { 5716 if (activeTrack->isFastTrack()) { 5717 ALOG_ASSERT(fastTrackToRemove == 0); 5718 fastTrackToRemove = activeTrack; 5719 } 5720 removeTrack_l(activeTrack); 5721 mActiveTracks.remove(activeTrack); 5722 mActiveTracksGen++; 5723 size--; 5724 continue; 5725 } 5726 5727 TrackBase::track_state activeTrackState = activeTrack->mState; 5728 switch (activeTrackState) { 5729 5730 case TrackBase::PAUSING: 5731 mActiveTracks.remove(activeTrack); 5732 mActiveTracksGen++; 5733 doBroadcast = true; 5734 size--; 5735 continue; 5736 5737 case TrackBase::STARTING_1: 5738 sleepUs = 10000; 5739 i++; 5740 continue; 5741 5742 case TrackBase::STARTING_2: 5743 doBroadcast = true; 5744 mStandby = false; 5745 activeTrack->mState = TrackBase::ACTIVE; 5746 break; 5747 5748 case TrackBase::ACTIVE: 5749 break; 5750 5751 case TrackBase::IDLE: 5752 i++; 5753 continue; 5754 5755 default: 5756 LOG_ALWAYS_FATAL("Unexpected activeTrackState %d", activeTrackState); 5757 } 5758 5759 activeTracks.add(activeTrack); 5760 i++; 5761 5762 if (activeTrack->isFastTrack()) { 5763 ALOG_ASSERT(!mFastTrackAvail); 5764 ALOG_ASSERT(fastTrack == 0); 5765 fastTrack = activeTrack; 5766 } 5767 } 5768 if (doBroadcast) { 5769 mStartStopCond.broadcast(); 5770 } 5771 5772 // sleep if there are no active tracks to process 5773 if (activeTracks.size() == 0) { 5774 if (sleepUs == 0) { 5775 sleepUs = kRecordThreadSleepUs; 5776 } 5777 continue; 5778 } 5779 sleepUs = 0; 5780 5781 lockEffectChains_l(effectChains); 5782 } 5783 5784 // thread mutex is now unlocked, mActiveTracks unknown, activeTracks.size() > 0 5785 5786 size_t size = effectChains.size(); 5787 for (size_t i = 0; i < size; i++) { 5788 // thread mutex is not locked, but effect chain is locked 5789 effectChains[i]->process_l(); 5790 } 5791 5792 // Push a new fast capture state if fast capture is not already running, or cblk change 5793 if (mFastCapture != 0) { 5794 FastCaptureStateQueue *sq = mFastCapture->sq(); 5795 FastCaptureState *state = sq->begin(); 5796 bool didModify = false; 5797 FastCaptureStateQueue::block_t block = FastCaptureStateQueue::BLOCK_UNTIL_PUSHED; 5798 if (state->mCommand != FastCaptureState::READ_WRITE /* FIXME && 5799 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)*/) { 5800 if (state->mCommand == FastCaptureState::COLD_IDLE) { 5801 int32_t old = android_atomic_inc(&mFastCaptureFutex); 5802 if (old == -1) { 5803 (void) syscall(__NR_futex, &mFastCaptureFutex, FUTEX_WAKE_PRIVATE, 1); 5804 } 5805 } 5806 state->mCommand = FastCaptureState::READ_WRITE; 5807#if 0 // FIXME 5808 mFastCaptureDumpState.increaseSamplingN(mAudioFlinger->isLowRamDevice() ? 5809 FastThreadDumpState::kSamplingNforLowRamDevice : 5810 FastThreadDumpState::kSamplingN); 5811#endif 5812 didModify = true; 5813 } 5814 audio_track_cblk_t *cblkOld = state->mCblk; 5815 audio_track_cblk_t *cblkNew = fastTrack != 0 ? fastTrack->cblk() : NULL; 5816 if (cblkNew != cblkOld) { 5817 state->mCblk = cblkNew; 5818 // block until acked if removing a fast track 5819 if (cblkOld != NULL) { 5820 block = FastCaptureStateQueue::BLOCK_UNTIL_ACKED; 5821 } 5822 didModify = true; 5823 } 5824 sq->end(didModify); 5825 if (didModify) { 5826 sq->push(block); 5827#if 0 5828 if (kUseFastCapture == FastCapture_Dynamic) { 5829 mNormalSource = mPipeSource; 5830 } 5831#endif 5832 } 5833 } 5834 5835 // now run the fast track destructor with thread mutex unlocked 5836 fastTrackToRemove.clear(); 5837 5838 // Read from HAL to keep up with fastest client if multiple active tracks, not slowest one. 5839 // Only the client(s) that are too slow will overrun. But if even the fastest client is too 5840 // slow, then this RecordThread will overrun by not calling HAL read often enough. 5841 // If destination is non-contiguous, first read past the nominal end of buffer, then 5842 // copy to the right place. Permitted because mRsmpInBuffer was over-allocated. 5843 5844 int32_t rear = mRsmpInRear & (mRsmpInFramesP2 - 1); 5845 ssize_t framesRead; 5846 5847 // If an NBAIO source is present, use it to read the normal capture's data 5848 if (mPipeSource != 0) { 5849 size_t framesToRead = mBufferSize / mFrameSize; 5850 framesRead = mPipeSource->read((uint8_t*)mRsmpInBuffer + rear * mFrameSize, 5851 framesToRead, AudioBufferProvider::kInvalidPTS); 5852 if (framesRead == 0) { 5853 // since pipe is non-blocking, simulate blocking input 5854 sleepUs = (framesToRead * 1000000LL) / mSampleRate; 5855 } 5856 // otherwise use the HAL / AudioStreamIn directly 5857 } else { 5858 ssize_t bytesRead = mInput->stream->read(mInput->stream, 5859 (uint8_t*)mRsmpInBuffer + rear * mFrameSize, mBufferSize); 5860 if (bytesRead < 0) { 5861 framesRead = bytesRead; 5862 } else { 5863 framesRead = bytesRead / mFrameSize; 5864 } 5865 } 5866 5867 if (framesRead < 0 || (framesRead == 0 && mPipeSource == 0)) { 5868 ALOGE("read failed: framesRead=%d", framesRead); 5869 // Force input into standby so that it tries to recover at next read attempt 5870 inputStandBy(); 5871 sleepUs = kRecordThreadSleepUs; 5872 } 5873 if (framesRead <= 0) { 5874 goto unlock; 5875 } 5876 ALOG_ASSERT(framesRead > 0); 5877 5878 if (mTeeSink != 0) { 5879 (void) mTeeSink->write((uint8_t*)mRsmpInBuffer + rear * mFrameSize, framesRead); 5880 } 5881 // If destination is non-contiguous, we now correct for reading past end of buffer. 5882 { 5883 size_t part1 = mRsmpInFramesP2 - rear; 5884 if ((size_t) framesRead > part1) { 5885 memcpy(mRsmpInBuffer, (uint8_t*)mRsmpInBuffer + mRsmpInFramesP2 * mFrameSize, 5886 (framesRead - part1) * mFrameSize); 5887 } 5888 } 5889 rear = mRsmpInRear += framesRead; 5890 5891 size = activeTracks.size(); 5892 // loop over each active track 5893 for (size_t i = 0; i < size; i++) { 5894 activeTrack = activeTracks[i]; 5895 5896 // skip fast tracks, as those are handled directly by FastCapture 5897 if (activeTrack->isFastTrack()) { 5898 continue; 5899 } 5900 5901 // TODO: This code probably should be moved to RecordTrack. 5902 // TODO: Update the activeTrack buffer converter in case of reconfigure. 5903 5904 enum { 5905 OVERRUN_UNKNOWN, 5906 OVERRUN_TRUE, 5907 OVERRUN_FALSE 5908 } overrun = OVERRUN_UNKNOWN; 5909 5910 // loop over getNextBuffer to handle circular sink 5911 for (;;) { 5912 5913 activeTrack->mSink.frameCount = ~0; 5914 status_t status = activeTrack->getNextBuffer(&activeTrack->mSink); 5915 size_t framesOut = activeTrack->mSink.frameCount; 5916 LOG_ALWAYS_FATAL_IF((status == OK) != (framesOut > 0)); 5917 5918 // check available frames and handle overrun conditions 5919 // if the record track isn't draining fast enough. 5920 bool hasOverrun; 5921 size_t framesIn; 5922 activeTrack->mResamplerBufferProvider->sync(&framesIn, &hasOverrun); 5923 if (hasOverrun) { 5924 overrun = OVERRUN_TRUE; 5925 } 5926 if (framesOut == 0 || framesIn == 0) { 5927 break; 5928 } 5929 5930 // Don't allow framesOut to be larger than what is possible with resampling 5931 // from framesIn. 5932 // This isn't strictly necessary but helps limit buffer resizing in 5933 // RecordBufferConverter. TODO: remove when no longer needed. 5934 framesOut = min(framesOut, 5935 destinationFramesPossible( 5936 framesIn, mSampleRate, activeTrack->mSampleRate)); 5937 // process frames from the RecordThread buffer provider to the RecordTrack buffer 5938 framesOut = activeTrack->mRecordBufferConverter->convert( 5939 activeTrack->mSink.raw, activeTrack->mResamplerBufferProvider, framesOut); 5940 5941 if (framesOut > 0 && (overrun == OVERRUN_UNKNOWN)) { 5942 overrun = OVERRUN_FALSE; 5943 } 5944 5945 if (activeTrack->mFramesToDrop == 0) { 5946 if (framesOut > 0) { 5947 activeTrack->mSink.frameCount = framesOut; 5948 activeTrack->releaseBuffer(&activeTrack->mSink); 5949 } 5950 } else { 5951 // FIXME could do a partial drop of framesOut 5952 if (activeTrack->mFramesToDrop > 0) { 5953 activeTrack->mFramesToDrop -= framesOut; 5954 if (activeTrack->mFramesToDrop <= 0) { 5955 activeTrack->clearSyncStartEvent(); 5956 } 5957 } else { 5958 activeTrack->mFramesToDrop += framesOut; 5959 if (activeTrack->mFramesToDrop >= 0 || activeTrack->mSyncStartEvent == 0 || 5960 activeTrack->mSyncStartEvent->isCancelled()) { 5961 ALOGW("Synced record %s, session %d, trigger session %d", 5962 (activeTrack->mFramesToDrop >= 0) ? "timed out" : "cancelled", 5963 activeTrack->sessionId(), 5964 (activeTrack->mSyncStartEvent != 0) ? 5965 activeTrack->mSyncStartEvent->triggerSession() : 0); 5966 activeTrack->clearSyncStartEvent(); 5967 } 5968 } 5969 } 5970 5971 if (framesOut == 0) { 5972 break; 5973 } 5974 } 5975 5976 switch (overrun) { 5977 case OVERRUN_TRUE: 5978 // client isn't retrieving buffers fast enough 5979 if (!activeTrack->setOverflow()) { 5980 nsecs_t now = systemTime(); 5981 // FIXME should lastWarning per track? 5982 if ((now - lastWarning) > kWarningThrottleNs) { 5983 ALOGW("RecordThread: buffer overflow"); 5984 lastWarning = now; 5985 } 5986 } 5987 break; 5988 case OVERRUN_FALSE: 5989 activeTrack->clearOverflow(); 5990 break; 5991 case OVERRUN_UNKNOWN: 5992 break; 5993 } 5994 5995 } 5996 5997unlock: 5998 // enable changes in effect chain 5999 unlockEffectChains(effectChains); 6000 // effectChains doesn't need to be cleared, since it is cleared by destructor at scope end 6001 } 6002 6003 standbyIfNotAlreadyInStandby(); 6004 6005 { 6006 Mutex::Autolock _l(mLock); 6007 for (size_t i = 0; i < mTracks.size(); i++) { 6008 sp<RecordTrack> track = mTracks[i]; 6009 track->invalidate(); 6010 } 6011 mActiveTracks.clear(); 6012 mActiveTracksGen++; 6013 mStartStopCond.broadcast(); 6014 } 6015 6016 releaseWakeLock(); 6017 6018 ALOGV("RecordThread %p exiting", this); 6019 return false; 6020} 6021 6022void AudioFlinger::RecordThread::standbyIfNotAlreadyInStandby() 6023{ 6024 if (!mStandby) { 6025 inputStandBy(); 6026 mStandby = true; 6027 } 6028} 6029 6030void AudioFlinger::RecordThread::inputStandBy() 6031{ 6032 // Idle the fast capture if it's currently running 6033 if (mFastCapture != 0) { 6034 FastCaptureStateQueue *sq = mFastCapture->sq(); 6035 FastCaptureState *state = sq->begin(); 6036 if (!(state->mCommand & FastCaptureState::IDLE)) { 6037 state->mCommand = FastCaptureState::COLD_IDLE; 6038 state->mColdFutexAddr = &mFastCaptureFutex; 6039 state->mColdGen++; 6040 mFastCaptureFutex = 0; 6041 sq->end(); 6042 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 6043 sq->push(FastCaptureStateQueue::BLOCK_UNTIL_ACKED); 6044#if 0 6045 if (kUseFastCapture == FastCapture_Dynamic) { 6046 // FIXME 6047 } 6048#endif 6049#ifdef AUDIO_WATCHDOG 6050 // FIXME 6051#endif 6052 } else { 6053 sq->end(false /*didModify*/); 6054 } 6055 } 6056 mInput->stream->common.standby(&mInput->stream->common); 6057} 6058 6059// RecordThread::createRecordTrack_l() must be called with AudioFlinger::mLock held 6060sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 6061 const sp<AudioFlinger::Client>& client, 6062 uint32_t sampleRate, 6063 audio_format_t format, 6064 audio_channel_mask_t channelMask, 6065 size_t *pFrameCount, 6066 int sessionId, 6067 size_t *notificationFrames, 6068 int uid, 6069 IAudioFlinger::track_flags_t *flags, 6070 pid_t tid, 6071 status_t *status) 6072{ 6073 size_t frameCount = *pFrameCount; 6074 sp<RecordTrack> track; 6075 status_t lStatus; 6076 6077 // client expresses a preference for FAST, but we get the final say 6078 if (*flags & IAudioFlinger::TRACK_FAST) { 6079 if ( 6080 // we formerly checked for a callback handler (non-0 tid), 6081 // but that is no longer required for TRANSFER_OBTAIN mode 6082 // 6083 // frame count is not specified, or is exactly the pipe depth 6084 ((frameCount == 0) || (frameCount == mPipeFramesP2)) && 6085 // PCM data 6086 audio_is_linear_pcm(format) && 6087 // native format 6088 (format == mFormat) && 6089 // native channel mask 6090 (channelMask == mChannelMask) && 6091 // native hardware sample rate 6092 (sampleRate == mSampleRate) && 6093 // record thread has an associated fast capture 6094 hasFastCapture() && 6095 // there are sufficient fast track slots available 6096 mFastTrackAvail 6097 ) { 6098 ALOGV("AUDIO_INPUT_FLAG_FAST accepted: frameCount=%u mFrameCount=%u", 6099 frameCount, mFrameCount); 6100 } else { 6101 ALOGV("AUDIO_INPUT_FLAG_FAST denied: frameCount=%u mFrameCount=%u mPipeFramesP2=%u " 6102 "format=%#x isLinear=%d channelMask=%#x sampleRate=%u mSampleRate=%u " 6103 "hasFastCapture=%d tid=%d mFastTrackAvail=%d", 6104 frameCount, mFrameCount, mPipeFramesP2, 6105 format, audio_is_linear_pcm(format), channelMask, sampleRate, mSampleRate, 6106 hasFastCapture(), tid, mFastTrackAvail); 6107 *flags &= ~IAudioFlinger::TRACK_FAST; 6108 } 6109 } 6110 6111 // compute track buffer size in frames, and suggest the notification frame count 6112 if (*flags & IAudioFlinger::TRACK_FAST) { 6113 // fast track: frame count is exactly the pipe depth 6114 frameCount = mPipeFramesP2; 6115 // ignore requested notificationFrames, and always notify exactly once every HAL buffer 6116 *notificationFrames = mFrameCount; 6117 } else { 6118 // not fast track: max notification period is resampled equivalent of one HAL buffer time 6119 // or 20 ms if there is a fast capture 6120 // TODO This could be a roundupRatio inline, and const 6121 size_t maxNotificationFrames = ((int64_t) (hasFastCapture() ? mSampleRate/50 : mFrameCount) 6122 * sampleRate + mSampleRate - 1) / mSampleRate; 6123 // minimum number of notification periods is at least kMinNotifications, 6124 // and at least kMinMs rounded up to a whole notification period (minNotificationsByMs) 6125 static const size_t kMinNotifications = 3; 6126 static const uint32_t kMinMs = 30; 6127 // TODO This could be a roundupRatio inline 6128 const size_t minFramesByMs = (sampleRate * kMinMs + 1000 - 1) / 1000; 6129 // TODO This could be a roundupRatio inline 6130 const size_t minNotificationsByMs = (minFramesByMs + maxNotificationFrames - 1) / 6131 maxNotificationFrames; 6132 const size_t minFrameCount = maxNotificationFrames * 6133 max(kMinNotifications, minNotificationsByMs); 6134 frameCount = max(frameCount, minFrameCount); 6135 if (*notificationFrames == 0 || *notificationFrames > maxNotificationFrames) { 6136 *notificationFrames = maxNotificationFrames; 6137 } 6138 } 6139 *pFrameCount = frameCount; 6140 6141 lStatus = initCheck(); 6142 if (lStatus != NO_ERROR) { 6143 ALOGE("createRecordTrack_l() audio driver not initialized"); 6144 goto Exit; 6145 } 6146 6147 { // scope for mLock 6148 Mutex::Autolock _l(mLock); 6149 6150 track = new RecordTrack(this, client, sampleRate, 6151 format, channelMask, frameCount, NULL, sessionId, uid, 6152 *flags, TrackBase::TYPE_DEFAULT); 6153 6154 lStatus = track->initCheck(); 6155 if (lStatus != NO_ERROR) { 6156 ALOGE("createRecordTrack_l() initCheck failed %d; no control block?", lStatus); 6157 // track must be cleared from the caller as the caller has the AF lock 6158 goto Exit; 6159 } 6160 mTracks.add(track); 6161 6162 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 6163 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 6164 mAudioFlinger->btNrecIsOff(); 6165 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 6166 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 6167 6168 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 6169 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 6170 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 6171 // so ask activity manager to do this on our behalf 6172 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 6173 } 6174 } 6175 6176 lStatus = NO_ERROR; 6177 6178Exit: 6179 *status = lStatus; 6180 return track; 6181} 6182 6183status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, 6184 AudioSystem::sync_event_t event, 6185 int triggerSession) 6186{ 6187 ALOGV("RecordThread::start event %d, triggerSession %d", event, triggerSession); 6188 sp<ThreadBase> strongMe = this; 6189 status_t status = NO_ERROR; 6190 6191 if (event == AudioSystem::SYNC_EVENT_NONE) { 6192 recordTrack->clearSyncStartEvent(); 6193 } else if (event != AudioSystem::SYNC_EVENT_SAME) { 6194 recordTrack->mSyncStartEvent = mAudioFlinger->createSyncEvent(event, 6195 triggerSession, 6196 recordTrack->sessionId(), 6197 syncStartEventCallback, 6198 recordTrack); 6199 // Sync event can be cancelled by the trigger session if the track is not in a 6200 // compatible state in which case we start record immediately 6201 if (recordTrack->mSyncStartEvent->isCancelled()) { 6202 recordTrack->clearSyncStartEvent(); 6203 } else { 6204 // do not wait for the event for more than AudioSystem::kSyncRecordStartTimeOutMs 6205 recordTrack->mFramesToDrop = - 6206 ((AudioSystem::kSyncRecordStartTimeOutMs * recordTrack->mSampleRate) / 1000); 6207 } 6208 } 6209 6210 { 6211 // This section is a rendezvous between binder thread executing start() and RecordThread 6212 AutoMutex lock(mLock); 6213 if (mActiveTracks.indexOf(recordTrack) >= 0) { 6214 if (recordTrack->mState == TrackBase::PAUSING) { 6215 ALOGV("active record track PAUSING -> ACTIVE"); 6216 recordTrack->mState = TrackBase::ACTIVE; 6217 } else { 6218 ALOGV("active record track state %d", recordTrack->mState); 6219 } 6220 return status; 6221 } 6222 6223 // TODO consider other ways of handling this, such as changing the state to :STARTING and 6224 // adding the track to mActiveTracks after returning from AudioSystem::startInput(), 6225 // or using a separate command thread 6226 recordTrack->mState = TrackBase::STARTING_1; 6227 mActiveTracks.add(recordTrack); 6228 mActiveTracksGen++; 6229 status_t status = NO_ERROR; 6230 if (recordTrack->isExternalTrack()) { 6231 mLock.unlock(); 6232 status = AudioSystem::startInput(mId, (audio_session_t)recordTrack->sessionId()); 6233 mLock.lock(); 6234 // FIXME should verify that recordTrack is still in mActiveTracks 6235 if (status != NO_ERROR) { 6236 mActiveTracks.remove(recordTrack); 6237 mActiveTracksGen++; 6238 recordTrack->clearSyncStartEvent(); 6239 ALOGV("RecordThread::start error %d", status); 6240 return status; 6241 } 6242 } 6243 // Catch up with current buffer indices if thread is already running. 6244 // This is what makes a new client discard all buffered data. If the track's mRsmpInFront 6245 // was initialized to some value closer to the thread's mRsmpInFront, then the track could 6246 // see previously buffered data before it called start(), but with greater risk of overrun. 6247 6248 recordTrack->mResamplerBufferProvider->reset(); 6249 // clear any converter state as new data will be discontinuous 6250 recordTrack->mRecordBufferConverter->reset(); 6251 recordTrack->mState = TrackBase::STARTING_2; 6252 // signal thread to start 6253 mWaitWorkCV.broadcast(); 6254 if (mActiveTracks.indexOf(recordTrack) < 0) { 6255 ALOGV("Record failed to start"); 6256 status = BAD_VALUE; 6257 goto startError; 6258 } 6259 return status; 6260 } 6261 6262startError: 6263 if (recordTrack->isExternalTrack()) { 6264 AudioSystem::stopInput(mId, (audio_session_t)recordTrack->sessionId()); 6265 } 6266 recordTrack->clearSyncStartEvent(); 6267 // FIXME I wonder why we do not reset the state here? 6268 return status; 6269} 6270 6271void AudioFlinger::RecordThread::syncStartEventCallback(const wp<SyncEvent>& event) 6272{ 6273 sp<SyncEvent> strongEvent = event.promote(); 6274 6275 if (strongEvent != 0) { 6276 sp<RefBase> ptr = strongEvent->cookie().promote(); 6277 if (ptr != 0) { 6278 RecordTrack *recordTrack = (RecordTrack *)ptr.get(); 6279 recordTrack->handleSyncStartEvent(strongEvent); 6280 } 6281 } 6282} 6283 6284bool AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 6285 ALOGV("RecordThread::stop"); 6286 AutoMutex _l(mLock); 6287 if (mActiveTracks.indexOf(recordTrack) != 0 || recordTrack->mState == TrackBase::PAUSING) { 6288 return false; 6289 } 6290 // note that threadLoop may still be processing the track at this point [without lock] 6291 recordTrack->mState = TrackBase::PAUSING; 6292 // do not wait for mStartStopCond if exiting 6293 if (exitPending()) { 6294 return true; 6295 } 6296 // FIXME incorrect usage of wait: no explicit predicate or loop 6297 mStartStopCond.wait(mLock); 6298 // if we have been restarted, recordTrack is in mActiveTracks here 6299 if (exitPending() || mActiveTracks.indexOf(recordTrack) != 0) { 6300 ALOGV("Record stopped OK"); 6301 return true; 6302 } 6303 return false; 6304} 6305 6306bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event __unused) const 6307{ 6308 return false; 6309} 6310 6311status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event __unused) 6312{ 6313#if 0 // This branch is currently dead code, but is preserved in case it will be needed in future 6314 if (!isValidSyncEvent(event)) { 6315 return BAD_VALUE; 6316 } 6317 6318 int eventSession = event->triggerSession(); 6319 status_t ret = NAME_NOT_FOUND; 6320 6321 Mutex::Autolock _l(mLock); 6322 6323 for (size_t i = 0; i < mTracks.size(); i++) { 6324 sp<RecordTrack> track = mTracks[i]; 6325 if (eventSession == track->sessionId()) { 6326 (void) track->setSyncEvent(event); 6327 ret = NO_ERROR; 6328 } 6329 } 6330 return ret; 6331#else 6332 return BAD_VALUE; 6333#endif 6334} 6335 6336// destroyTrack_l() must be called with ThreadBase::mLock held 6337void AudioFlinger::RecordThread::destroyTrack_l(const sp<RecordTrack>& track) 6338{ 6339 track->terminate(); 6340 track->mState = TrackBase::STOPPED; 6341 // active tracks are removed by threadLoop() 6342 if (mActiveTracks.indexOf(track) < 0) { 6343 removeTrack_l(track); 6344 } 6345} 6346 6347void AudioFlinger::RecordThread::removeTrack_l(const sp<RecordTrack>& track) 6348{ 6349 mTracks.remove(track); 6350 // need anything related to effects here? 6351 if (track->isFastTrack()) { 6352 ALOG_ASSERT(!mFastTrackAvail); 6353 mFastTrackAvail = true; 6354 } 6355} 6356 6357void AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 6358{ 6359 dumpInternals(fd, args); 6360 dumpTracks(fd, args); 6361 dumpEffectChains(fd, args); 6362} 6363 6364void AudioFlinger::RecordThread::dumpInternals(int fd, const Vector<String16>& args) 6365{ 6366 dprintf(fd, "\nInput thread %p:\n", this); 6367 6368 dumpBase(fd, args); 6369 6370 if (mActiveTracks.size() == 0) { 6371 dprintf(fd, " No active record clients\n"); 6372 } 6373 dprintf(fd, " Fast capture thread: %s\n", hasFastCapture() ? "yes" : "no"); 6374 dprintf(fd, " Fast track available: %s\n", mFastTrackAvail ? "yes" : "no"); 6375 6376 // Make a non-atomic copy of fast capture dump state so it won't change underneath us 6377 const FastCaptureDumpState copy(mFastCaptureDumpState); 6378 copy.dump(fd); 6379} 6380 6381void AudioFlinger::RecordThread::dumpTracks(int fd, const Vector<String16>& args __unused) 6382{ 6383 const size_t SIZE = 256; 6384 char buffer[SIZE]; 6385 String8 result; 6386 6387 size_t numtracks = mTracks.size(); 6388 size_t numactive = mActiveTracks.size(); 6389 size_t numactiveseen = 0; 6390 dprintf(fd, " %d Tracks", numtracks); 6391 if (numtracks) { 6392 dprintf(fd, " of which %d are active\n", numactive); 6393 RecordTrack::appendDumpHeader(result); 6394 for (size_t i = 0; i < numtracks ; ++i) { 6395 sp<RecordTrack> track = mTracks[i]; 6396 if (track != 0) { 6397 bool active = mActiveTracks.indexOf(track) >= 0; 6398 if (active) { 6399 numactiveseen++; 6400 } 6401 track->dump(buffer, SIZE, active); 6402 result.append(buffer); 6403 } 6404 } 6405 } else { 6406 dprintf(fd, "\n"); 6407 } 6408 6409 if (numactiveseen != numactive) { 6410 snprintf(buffer, SIZE, " The following tracks are in the active list but" 6411 " not in the track list\n"); 6412 result.append(buffer); 6413 RecordTrack::appendDumpHeader(result); 6414 for (size_t i = 0; i < numactive; ++i) { 6415 sp<RecordTrack> track = mActiveTracks[i]; 6416 if (mTracks.indexOf(track) < 0) { 6417 track->dump(buffer, SIZE, true); 6418 result.append(buffer); 6419 } 6420 } 6421 6422 } 6423 write(fd, result.string(), result.size()); 6424} 6425 6426 6427void AudioFlinger::RecordThread::ResamplerBufferProvider::reset() 6428{ 6429 sp<ThreadBase> threadBase = mRecordTrack->mThread.promote(); 6430 RecordThread *recordThread = (RecordThread *) threadBase.get(); 6431 mRsmpInFront = recordThread->mRsmpInRear; 6432 mRsmpInUnrel = 0; 6433} 6434 6435void AudioFlinger::RecordThread::ResamplerBufferProvider::sync( 6436 size_t *framesAvailable, bool *hasOverrun) 6437{ 6438 sp<ThreadBase> threadBase = mRecordTrack->mThread.promote(); 6439 RecordThread *recordThread = (RecordThread *) threadBase.get(); 6440 const int32_t rear = recordThread->mRsmpInRear; 6441 const int32_t front = mRsmpInFront; 6442 const ssize_t filled = rear - front; 6443 6444 size_t framesIn; 6445 bool overrun = false; 6446 if (filled < 0) { 6447 // should not happen, but treat like a massive overrun and re-sync 6448 framesIn = 0; 6449 mRsmpInFront = rear; 6450 overrun = true; 6451 } else if ((size_t) filled <= recordThread->mRsmpInFrames) { 6452 framesIn = (size_t) filled; 6453 } else { 6454 // client is not keeping up with server, but give it latest data 6455 framesIn = recordThread->mRsmpInFrames; 6456 mRsmpInFront = /* front = */ rear - framesIn; 6457 overrun = true; 6458 } 6459 if (framesAvailable != NULL) { 6460 *framesAvailable = framesIn; 6461 } 6462 if (hasOverrun != NULL) { 6463 *hasOverrun = overrun; 6464 } 6465} 6466 6467// AudioBufferProvider interface 6468status_t AudioFlinger::RecordThread::ResamplerBufferProvider::getNextBuffer( 6469 AudioBufferProvider::Buffer* buffer, int64_t pts __unused) 6470{ 6471 sp<ThreadBase> threadBase = mRecordTrack->mThread.promote(); 6472 if (threadBase == 0) { 6473 buffer->frameCount = 0; 6474 buffer->raw = NULL; 6475 return NOT_ENOUGH_DATA; 6476 } 6477 RecordThread *recordThread = (RecordThread *) threadBase.get(); 6478 int32_t rear = recordThread->mRsmpInRear; 6479 int32_t front = mRsmpInFront; 6480 ssize_t filled = rear - front; 6481 // FIXME should not be P2 (don't want to increase latency) 6482 // FIXME if client not keeping up, discard 6483 LOG_ALWAYS_FATAL_IF(!(0 <= filled && (size_t) filled <= recordThread->mRsmpInFrames)); 6484 // 'filled' may be non-contiguous, so return only the first contiguous chunk 6485 front &= recordThread->mRsmpInFramesP2 - 1; 6486 size_t part1 = recordThread->mRsmpInFramesP2 - front; 6487 if (part1 > (size_t) filled) { 6488 part1 = filled; 6489 } 6490 size_t ask = buffer->frameCount; 6491 ALOG_ASSERT(ask > 0); 6492 if (part1 > ask) { 6493 part1 = ask; 6494 } 6495 if (part1 == 0) { 6496 // out of data is fine since the resampler will return a short-count. 6497 buffer->raw = NULL; 6498 buffer->frameCount = 0; 6499 mRsmpInUnrel = 0; 6500 return NOT_ENOUGH_DATA; 6501 } 6502 6503 buffer->raw = (uint8_t*)recordThread->mRsmpInBuffer + front * recordThread->mFrameSize; 6504 buffer->frameCount = part1; 6505 mRsmpInUnrel = part1; 6506 return NO_ERROR; 6507} 6508 6509// AudioBufferProvider interface 6510void AudioFlinger::RecordThread::ResamplerBufferProvider::releaseBuffer( 6511 AudioBufferProvider::Buffer* buffer) 6512{ 6513 size_t stepCount = buffer->frameCount; 6514 if (stepCount == 0) { 6515 return; 6516 } 6517 ALOG_ASSERT(stepCount <= mRsmpInUnrel); 6518 mRsmpInUnrel -= stepCount; 6519 mRsmpInFront += stepCount; 6520 buffer->raw = NULL; 6521 buffer->frameCount = 0; 6522} 6523 6524AudioFlinger::RecordThread::RecordBufferConverter::RecordBufferConverter( 6525 audio_channel_mask_t srcChannelMask, audio_format_t srcFormat, 6526 uint32_t srcSampleRate, 6527 audio_channel_mask_t dstChannelMask, audio_format_t dstFormat, 6528 uint32_t dstSampleRate) : 6529 mSrcChannelMask(AUDIO_CHANNEL_INVALID), // updateParameters will set following vars 6530 // mSrcFormat 6531 // mSrcSampleRate 6532 // mDstChannelMask 6533 // mDstFormat 6534 // mDstSampleRate 6535 // mSrcChannelCount 6536 // mDstChannelCount 6537 // mDstFrameSize 6538 mBuf(NULL), mBufFrames(0), mBufFrameSize(0), 6539 mResampler(NULL), 6540 mIsLegacyDownmix(false), 6541 mIsLegacyUpmix(false), 6542 mRequiresFloat(false), 6543 mInputConverterProvider(NULL) 6544{ 6545 (void)updateParameters(srcChannelMask, srcFormat, srcSampleRate, 6546 dstChannelMask, dstFormat, dstSampleRate); 6547} 6548 6549AudioFlinger::RecordThread::RecordBufferConverter::~RecordBufferConverter() { 6550 free(mBuf); 6551 delete mResampler; 6552 delete mInputConverterProvider; 6553} 6554 6555size_t AudioFlinger::RecordThread::RecordBufferConverter::convert(void *dst, 6556 AudioBufferProvider *provider, size_t frames) 6557{ 6558 if (mInputConverterProvider != NULL) { 6559 mInputConverterProvider->setBufferProvider(provider); 6560 provider = mInputConverterProvider; 6561 } 6562 6563 if (mResampler == NULL) { 6564 ALOGVV("NO RESAMPLING sampleRate:%u mSrcFormat:%#x mDstFormat:%#x", 6565 mSrcSampleRate, mSrcFormat, mDstFormat); 6566 6567 AudioBufferProvider::Buffer buffer; 6568 for (size_t i = frames; i > 0; ) { 6569 buffer.frameCount = i; 6570 status_t status = provider->getNextBuffer(&buffer, 0); 6571 if (status != OK || buffer.frameCount == 0) { 6572 frames -= i; // cannot fill request. 6573 break; 6574 } 6575 // format convert to destination buffer 6576 convertNoResampler(dst, buffer.raw, buffer.frameCount); 6577 6578 dst = (int8_t*)dst + buffer.frameCount * mDstFrameSize; 6579 i -= buffer.frameCount; 6580 provider->releaseBuffer(&buffer); 6581 } 6582 } else { 6583 ALOGVV("RESAMPLING mSrcSampleRate:%u mDstSampleRate:%u mSrcFormat:%#x mDstFormat:%#x", 6584 mSrcSampleRate, mDstSampleRate, mSrcFormat, mDstFormat); 6585 6586 // reallocate buffer if needed 6587 if (mBufFrameSize != 0 && mBufFrames < frames) { 6588 free(mBuf); 6589 mBufFrames = frames; 6590 (void)posix_memalign(&mBuf, 32, mBufFrames * mBufFrameSize); 6591 } 6592 // resampler accumulates, but we only have one source track 6593 memset(mBuf, 0, frames * mBufFrameSize); 6594 frames = mResampler->resample((int32_t*)mBuf, frames, provider); 6595 // format convert to destination buffer 6596 convertResampler(dst, mBuf, frames); 6597 } 6598 return frames; 6599} 6600 6601status_t AudioFlinger::RecordThread::RecordBufferConverter::updateParameters( 6602 audio_channel_mask_t srcChannelMask, audio_format_t srcFormat, 6603 uint32_t srcSampleRate, 6604 audio_channel_mask_t dstChannelMask, audio_format_t dstFormat, 6605 uint32_t dstSampleRate) 6606{ 6607 // quick evaluation if there is any change. 6608 if (mSrcFormat == srcFormat 6609 && mSrcChannelMask == srcChannelMask 6610 && mSrcSampleRate == srcSampleRate 6611 && mDstFormat == dstFormat 6612 && mDstChannelMask == dstChannelMask 6613 && mDstSampleRate == dstSampleRate) { 6614 return NO_ERROR; 6615 } 6616 6617 ALOGV("RecordBufferConverter updateParameters srcMask:%#x dstMask:%#x" 6618 " srcFormat:%#x dstFormat:%#x srcRate:%u dstRate:%u", 6619 srcChannelMask, dstChannelMask, srcFormat, dstFormat, srcSampleRate, dstSampleRate); 6620 const bool valid = 6621 audio_is_input_channel(srcChannelMask) 6622 && audio_is_input_channel(dstChannelMask) 6623 && audio_is_valid_format(srcFormat) && audio_is_linear_pcm(srcFormat) 6624 && audio_is_valid_format(dstFormat) && audio_is_linear_pcm(dstFormat) 6625 && (srcSampleRate <= dstSampleRate * AUDIO_RESAMPLER_DOWN_RATIO_MAX) 6626 ; // no upsampling checks for now 6627 if (!valid) { 6628 return BAD_VALUE; 6629 } 6630 6631 mSrcFormat = srcFormat; 6632 mSrcChannelMask = srcChannelMask; 6633 mSrcSampleRate = srcSampleRate; 6634 mDstFormat = dstFormat; 6635 mDstChannelMask = dstChannelMask; 6636 mDstSampleRate = dstSampleRate; 6637 6638 // compute derived parameters 6639 mSrcChannelCount = audio_channel_count_from_in_mask(srcChannelMask); 6640 mDstChannelCount = audio_channel_count_from_in_mask(dstChannelMask); 6641 mDstFrameSize = mDstChannelCount * audio_bytes_per_sample(mDstFormat); 6642 6643 // do we need to resample? 6644 delete mResampler; 6645 mResampler = NULL; 6646 if (mSrcSampleRate != mDstSampleRate) { 6647 mResampler = AudioResampler::create(AUDIO_FORMAT_PCM_FLOAT, 6648 mSrcChannelCount, mDstSampleRate); 6649 mResampler->setSampleRate(mSrcSampleRate); 6650 mResampler->setVolume(AudioMixer::UNITY_GAIN_FLOAT, AudioMixer::UNITY_GAIN_FLOAT); 6651 } 6652 6653 // are we running legacy channel conversion modes? 6654 mIsLegacyDownmix = (mSrcChannelMask == AUDIO_CHANNEL_IN_STEREO 6655 || mSrcChannelMask == AUDIO_CHANNEL_IN_FRONT_BACK) 6656 && mDstChannelMask == AUDIO_CHANNEL_IN_MONO; 6657 mIsLegacyUpmix = mSrcChannelMask == AUDIO_CHANNEL_IN_MONO 6658 && (mDstChannelMask == AUDIO_CHANNEL_IN_STEREO 6659 || mDstChannelMask == AUDIO_CHANNEL_IN_FRONT_BACK); 6660 6661 // do we need to process in float? 6662 mRequiresFloat = mResampler != NULL || mIsLegacyDownmix || mIsLegacyUpmix; 6663 6664 // do we need a staging buffer to convert for destination (we can still optimize this)? 6665 // we use mBufFrameSize > 0 to indicate both frame size as well as buffer necessity 6666 if (mResampler != NULL) { 6667 mBufFrameSize = max(mSrcChannelCount, FCC_2) 6668 * audio_bytes_per_sample(AUDIO_FORMAT_PCM_FLOAT); 6669 } else if (mIsLegacyUpmix || mIsLegacyDownmix) { // legacy modes always float 6670 mBufFrameSize = mDstChannelCount * audio_bytes_per_sample(AUDIO_FORMAT_PCM_FLOAT); 6671 } else if (mSrcChannelMask != mDstChannelMask && mDstFormat != mSrcFormat) { 6672 mBufFrameSize = mDstChannelCount * audio_bytes_per_sample(mSrcFormat); 6673 } else { 6674 mBufFrameSize = 0; 6675 } 6676 mBufFrames = 0; // force the buffer to be resized. 6677 6678 // do we need an input converter buffer provider to give us float? 6679 delete mInputConverterProvider; 6680 mInputConverterProvider = NULL; 6681 if (mRequiresFloat && mSrcFormat != AUDIO_FORMAT_PCM_FLOAT) { 6682 mInputConverterProvider = new ReformatBufferProvider( 6683 audio_channel_count_from_in_mask(mSrcChannelMask), 6684 mSrcFormat, 6685 AUDIO_FORMAT_PCM_FLOAT, 6686 256 /* provider buffer frame count */); 6687 } 6688 6689 // do we need a remixer to do channel mask conversion 6690 if (!mIsLegacyDownmix && !mIsLegacyUpmix && mSrcChannelMask != mDstChannelMask) { 6691 (void) memcpy_by_index_array_initialization_from_channel_mask( 6692 mIdxAry, ARRAY_SIZE(mIdxAry), mDstChannelMask, mSrcChannelMask); 6693 } 6694 return NO_ERROR; 6695} 6696 6697void AudioFlinger::RecordThread::RecordBufferConverter::convertNoResampler( 6698 void *dst, const void *src, size_t frames) 6699{ 6700 // src is native type unless there is legacy upmix or downmix, whereupon it is float. 6701 if (mBufFrameSize != 0 && mBufFrames < frames) { 6702 free(mBuf); 6703 mBufFrames = frames; 6704 (void)posix_memalign(&mBuf, 32, mBufFrames * mBufFrameSize); 6705 } 6706 // do we need to do legacy upmix and downmix? 6707 if (mIsLegacyUpmix || mIsLegacyDownmix) { 6708 void *dstBuf = mBuf != NULL ? mBuf : dst; 6709 if (mIsLegacyUpmix) { 6710 upmix_to_stereo_float_from_mono_float((float *)dstBuf, 6711 (const float *)src, frames); 6712 } else /*mIsLegacyDownmix */ { 6713 downmix_to_mono_float_from_stereo_float((float *)dstBuf, 6714 (const float *)src, frames); 6715 } 6716 if (mBuf != NULL) { 6717 memcpy_by_audio_format(dst, mDstFormat, mBuf, AUDIO_FORMAT_PCM_FLOAT, 6718 frames * mDstChannelCount); 6719 } 6720 return; 6721 } 6722 // do we need to do channel mask conversion? 6723 if (mSrcChannelMask != mDstChannelMask) { 6724 void *dstBuf = mBuf != NULL ? mBuf : dst; 6725 memcpy_by_index_array(dstBuf, mDstChannelCount, 6726 src, mSrcChannelCount, mIdxAry, audio_bytes_per_sample(mSrcFormat), frames); 6727 if (dstBuf == dst) { 6728 return; // format is the same 6729 } 6730 } 6731 // convert to destination buffer 6732 const void *convertBuf = mBuf != NULL ? mBuf : src; 6733 memcpy_by_audio_format(dst, mDstFormat, convertBuf, mSrcFormat, 6734 frames * mDstChannelCount); 6735} 6736 6737void AudioFlinger::RecordThread::RecordBufferConverter::convertResampler( 6738 void *dst, /*not-a-const*/ void *src, size_t frames) 6739{ 6740 // src buffer format is ALWAYS float when entering this routine 6741 if (mIsLegacyUpmix) { 6742 ; // mono to stereo already handled by resampler 6743 } else if (mIsLegacyDownmix 6744 || (mSrcChannelMask == mDstChannelMask && mSrcChannelCount == 1)) { 6745 // the resampler outputs stereo for mono input channel (a feature?) 6746 // must convert to mono 6747 downmix_to_mono_float_from_stereo_float((float *)src, 6748 (const float *)src, frames); 6749 } else if (mSrcChannelMask != mDstChannelMask) { 6750 // convert to mono channel again for channel mask conversion (could be skipped 6751 // with further optimization). 6752 if (mSrcChannelCount == 1) { 6753 downmix_to_mono_float_from_stereo_float((float *)src, 6754 (const float *)src, frames); 6755 } 6756 // convert to destination format (in place, OK as float is larger than other types) 6757 if (mDstFormat != AUDIO_FORMAT_PCM_FLOAT) { 6758 memcpy_by_audio_format(src, mDstFormat, src, AUDIO_FORMAT_PCM_FLOAT, 6759 frames * mSrcChannelCount); 6760 } 6761 // channel convert and save to dst 6762 memcpy_by_index_array(dst, mDstChannelCount, 6763 src, mSrcChannelCount, mIdxAry, audio_bytes_per_sample(mDstFormat), frames); 6764 return; 6765 } 6766 // convert to destination format and save to dst 6767 memcpy_by_audio_format(dst, mDstFormat, src, AUDIO_FORMAT_PCM_FLOAT, 6768 frames * mDstChannelCount); 6769} 6770 6771bool AudioFlinger::RecordThread::checkForNewParameter_l(const String8& keyValuePair, 6772 status_t& status) 6773{ 6774 bool reconfig = false; 6775 6776 status = NO_ERROR; 6777 6778 audio_format_t reqFormat = mFormat; 6779 uint32_t samplingRate = mSampleRate; 6780 // TODO this may change if we want to support capture from HDMI PCM multi channel (e.g on TVs). 6781 audio_channel_mask_t channelMask = audio_channel_in_mask_from_count(mChannelCount); 6782 6783 AudioParameter param = AudioParameter(keyValuePair); 6784 int value; 6785 // TODO Investigate when this code runs. Check with audio policy when a sample rate and 6786 // channel count change can be requested. Do we mandate the first client defines the 6787 // HAL sampling rate and channel count or do we allow changes on the fly? 6788 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 6789 samplingRate = value; 6790 reconfig = true; 6791 } 6792 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 6793 if (!audio_is_linear_pcm((audio_format_t) value)) { 6794 status = BAD_VALUE; 6795 } else { 6796 reqFormat = (audio_format_t) value; 6797 reconfig = true; 6798 } 6799 } 6800 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 6801 audio_channel_mask_t mask = (audio_channel_mask_t) value; 6802 if (!audio_is_input_channel(mask) || 6803 audio_channel_count_from_in_mask(mask) > FCC_8) { 6804 status = BAD_VALUE; 6805 } else { 6806 channelMask = mask; 6807 reconfig = true; 6808 } 6809 } 6810 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 6811 // do not accept frame count changes if tracks are open as the track buffer 6812 // size depends on frame count and correct behavior would not be guaranteed 6813 // if frame count is changed after track creation 6814 if (mActiveTracks.size() > 0) { 6815 status = INVALID_OPERATION; 6816 } else { 6817 reconfig = true; 6818 } 6819 } 6820 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 6821 // forward device change to effects that have requested to be 6822 // aware of attached audio device. 6823 for (size_t i = 0; i < mEffectChains.size(); i++) { 6824 mEffectChains[i]->setDevice_l(value); 6825 } 6826 6827 // store input device and output device but do not forward output device to audio HAL. 6828 // Note that status is ignored by the caller for output device 6829 // (see AudioFlinger::setParameters() 6830 if (audio_is_output_devices(value)) { 6831 mOutDevice = value; 6832 status = BAD_VALUE; 6833 } else { 6834 mInDevice = value; 6835 if (value != AUDIO_DEVICE_NONE) { 6836 mPrevInDevice = value; 6837 } 6838 // disable AEC and NS if the device is a BT SCO headset supporting those 6839 // pre processings 6840 if (mTracks.size() > 0) { 6841 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 6842 mAudioFlinger->btNrecIsOff(); 6843 for (size_t i = 0; i < mTracks.size(); i++) { 6844 sp<RecordTrack> track = mTracks[i]; 6845 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId()); 6846 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId()); 6847 } 6848 } 6849 } 6850 } 6851 if (param.getInt(String8(AudioParameter::keyInputSource), value) == NO_ERROR && 6852 mAudioSource != (audio_source_t)value) { 6853 // forward device change to effects that have requested to be 6854 // aware of attached audio device. 6855 for (size_t i = 0; i < mEffectChains.size(); i++) { 6856 mEffectChains[i]->setAudioSource_l((audio_source_t)value); 6857 } 6858 mAudioSource = (audio_source_t)value; 6859 } 6860 6861 if (status == NO_ERROR) { 6862 status = mInput->stream->common.set_parameters(&mInput->stream->common, 6863 keyValuePair.string()); 6864 if (status == INVALID_OPERATION) { 6865 inputStandBy(); 6866 status = mInput->stream->common.set_parameters(&mInput->stream->common, 6867 keyValuePair.string()); 6868 } 6869 if (reconfig) { 6870 if (status == BAD_VALUE && 6871 audio_is_linear_pcm(mInput->stream->common.get_format(&mInput->stream->common)) && 6872 audio_is_linear_pcm(reqFormat) && 6873 (mInput->stream->common.get_sample_rate(&mInput->stream->common) 6874 <= (AUDIO_RESAMPLER_DOWN_RATIO_MAX * samplingRate)) && 6875 audio_channel_count_from_in_mask( 6876 mInput->stream->common.get_channels(&mInput->stream->common)) <= FCC_8) { 6877 status = NO_ERROR; 6878 } 6879 if (status == NO_ERROR) { 6880 readInputParameters_l(); 6881 sendIoConfigEvent_l(AUDIO_INPUT_CONFIG_CHANGED); 6882 } 6883 } 6884 } 6885 6886 return reconfig; 6887} 6888 6889String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 6890{ 6891 Mutex::Autolock _l(mLock); 6892 if (initCheck() != NO_ERROR) { 6893 return String8(); 6894 } 6895 6896 char *s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 6897 const String8 out_s8(s); 6898 free(s); 6899 return out_s8; 6900} 6901 6902void AudioFlinger::RecordThread::ioConfigChanged(audio_io_config_event event, pid_t pid) { 6903 sp<AudioIoDescriptor> desc = new AudioIoDescriptor(); 6904 6905 desc->mIoHandle = mId; 6906 6907 switch (event) { 6908 case AUDIO_INPUT_OPENED: 6909 case AUDIO_INPUT_CONFIG_CHANGED: 6910 desc->mPatch = mPatch; 6911 desc->mChannelMask = mChannelMask; 6912 desc->mSamplingRate = mSampleRate; 6913 desc->mFormat = mFormat; 6914 desc->mFrameCount = mFrameCount; 6915 desc->mLatency = 0; 6916 break; 6917 6918 case AUDIO_INPUT_CLOSED: 6919 default: 6920 break; 6921 } 6922 mAudioFlinger->ioConfigChanged(event, desc, pid); 6923} 6924 6925void AudioFlinger::RecordThread::readInputParameters_l() 6926{ 6927 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 6928 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 6929 mChannelCount = audio_channel_count_from_in_mask(mChannelMask); 6930 if (mChannelCount > FCC_8) { 6931 ALOGE("HAL channel count %d > %d", mChannelCount, FCC_8); 6932 } 6933 mHALFormat = mInput->stream->common.get_format(&mInput->stream->common); 6934 mFormat = mHALFormat; 6935 if (!audio_is_linear_pcm(mFormat)) { 6936 ALOGE("HAL format %#x is not linear pcm", mFormat); 6937 } 6938 mFrameSize = audio_stream_in_frame_size(mInput->stream); 6939 mBufferSize = mInput->stream->common.get_buffer_size(&mInput->stream->common); 6940 mFrameCount = mBufferSize / mFrameSize; 6941 // This is the formula for calculating the temporary buffer size. 6942 // With 7 HAL buffers, we can guarantee ability to down-sample the input by ratio of 6:1 to 6943 // 1 full output buffer, regardless of the alignment of the available input. 6944 // The value is somewhat arbitrary, and could probably be even larger. 6945 // A larger value should allow more old data to be read after a track calls start(), 6946 // without increasing latency. 6947 // 6948 // Note this is independent of the maximum downsampling ratio permitted for capture. 6949 mRsmpInFrames = mFrameCount * 7; 6950 mRsmpInFramesP2 = roundup(mRsmpInFrames); 6951 free(mRsmpInBuffer); 6952 mRsmpInBuffer = NULL; 6953 6954 // TODO optimize audio capture buffer sizes ... 6955 // Here we calculate the size of the sliding buffer used as a source 6956 // for resampling. mRsmpInFramesP2 is currently roundup(mFrameCount * 7). 6957 // For current HAL frame counts, this is usually 2048 = 40 ms. It would 6958 // be better to have it derived from the pipe depth in the long term. 6959 // The current value is higher than necessary. However it should not add to latency. 6960 6961 // Over-allocate beyond mRsmpInFramesP2 to permit a HAL read past end of buffer 6962 size_t bufferSize = (mRsmpInFramesP2 + mFrameCount - 1) * mFrameSize; 6963 (void)posix_memalign(&mRsmpInBuffer, 32, bufferSize); 6964 memset(mRsmpInBuffer, 0, bufferSize); // if posix_memalign fails, will segv here. 6965 6966 // AudioRecord mSampleRate and mChannelCount are constant due to AudioRecord API constraints. 6967 // But if thread's mSampleRate or mChannelCount changes, how will that affect active tracks? 6968} 6969 6970uint32_t AudioFlinger::RecordThread::getInputFramesLost() 6971{ 6972 Mutex::Autolock _l(mLock); 6973 if (initCheck() != NO_ERROR) { 6974 return 0; 6975 } 6976 6977 return mInput->stream->get_input_frames_lost(mInput->stream); 6978} 6979 6980uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) const 6981{ 6982 Mutex::Autolock _l(mLock); 6983 uint32_t result = 0; 6984 if (getEffectChain_l(sessionId) != 0) { 6985 result = EFFECT_SESSION; 6986 } 6987 6988 for (size_t i = 0; i < mTracks.size(); ++i) { 6989 if (sessionId == mTracks[i]->sessionId()) { 6990 result |= TRACK_SESSION; 6991 break; 6992 } 6993 } 6994 6995 return result; 6996} 6997 6998KeyedVector<int, bool> AudioFlinger::RecordThread::sessionIds() const 6999{ 7000 KeyedVector<int, bool> ids; 7001 Mutex::Autolock _l(mLock); 7002 for (size_t j = 0; j < mTracks.size(); ++j) { 7003 sp<RecordThread::RecordTrack> track = mTracks[j]; 7004 int sessionId = track->sessionId(); 7005 if (ids.indexOfKey(sessionId) < 0) { 7006 ids.add(sessionId, true); 7007 } 7008 } 7009 return ids; 7010} 7011 7012AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 7013{ 7014 Mutex::Autolock _l(mLock); 7015 AudioStreamIn *input = mInput; 7016 mInput = NULL; 7017 return input; 7018} 7019 7020// this method must always be called either with ThreadBase mLock held or inside the thread loop 7021audio_stream_t* AudioFlinger::RecordThread::stream() const 7022{ 7023 if (mInput == NULL) { 7024 return NULL; 7025 } 7026 return &mInput->stream->common; 7027} 7028 7029status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 7030{ 7031 // only one chain per input thread 7032 if (mEffectChains.size() != 0) { 7033 ALOGW("addEffectChain_l() already one chain %p on thread %p", chain.get(), this); 7034 return INVALID_OPERATION; 7035 } 7036 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 7037 chain->setThread(this); 7038 chain->setInBuffer(NULL); 7039 chain->setOutBuffer(NULL); 7040 7041 checkSuspendOnAddEffectChain_l(chain); 7042 7043 // make sure enabled pre processing effects state is communicated to the HAL as we 7044 // just moved them to a new input stream. 7045 chain->syncHalEffectsState(); 7046 7047 mEffectChains.add(chain); 7048 7049 return NO_ERROR; 7050} 7051 7052size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 7053{ 7054 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 7055 ALOGW_IF(mEffectChains.size() != 1, 7056 "removeEffectChain_l() %p invalid chain size %d on thread %p", 7057 chain.get(), mEffectChains.size(), this); 7058 if (mEffectChains.size() == 1) { 7059 mEffectChains.removeAt(0); 7060 } 7061 return 0; 7062} 7063 7064status_t AudioFlinger::RecordThread::createAudioPatch_l(const struct audio_patch *patch, 7065 audio_patch_handle_t *handle) 7066{ 7067 status_t status = NO_ERROR; 7068 7069 // store new device and send to effects 7070 mInDevice = patch->sources[0].ext.device.type; 7071 mPatch = *patch; 7072 for (size_t i = 0; i < mEffectChains.size(); i++) { 7073 mEffectChains[i]->setDevice_l(mInDevice); 7074 } 7075 7076 // disable AEC and NS if the device is a BT SCO headset supporting those 7077 // pre processings 7078 if (mTracks.size() > 0) { 7079 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 7080 mAudioFlinger->btNrecIsOff(); 7081 for (size_t i = 0; i < mTracks.size(); i++) { 7082 sp<RecordTrack> track = mTracks[i]; 7083 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId()); 7084 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId()); 7085 } 7086 } 7087 7088 // store new source and send to effects 7089 if (mAudioSource != patch->sinks[0].ext.mix.usecase.source) { 7090 mAudioSource = patch->sinks[0].ext.mix.usecase.source; 7091 for (size_t i = 0; i < mEffectChains.size(); i++) { 7092 mEffectChains[i]->setAudioSource_l(mAudioSource); 7093 } 7094 } 7095 7096 if (mInput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 7097 audio_hw_device_t *hwDevice = mInput->audioHwDev->hwDevice(); 7098 status = hwDevice->create_audio_patch(hwDevice, 7099 patch->num_sources, 7100 patch->sources, 7101 patch->num_sinks, 7102 patch->sinks, 7103 handle); 7104 } else { 7105 char *address; 7106 if (strcmp(patch->sources[0].ext.device.address, "") != 0) { 7107 address = audio_device_address_to_parameter( 7108 patch->sources[0].ext.device.type, 7109 patch->sources[0].ext.device.address); 7110 } else { 7111 address = (char *)calloc(1, 1); 7112 } 7113 AudioParameter param = AudioParameter(String8(address)); 7114 free(address); 7115 param.addInt(String8(AUDIO_PARAMETER_STREAM_ROUTING), 7116 (int)patch->sources[0].ext.device.type); 7117 param.addInt(String8(AUDIO_PARAMETER_STREAM_INPUT_SOURCE), 7118 (int)patch->sinks[0].ext.mix.usecase.source); 7119 status = mInput->stream->common.set_parameters(&mInput->stream->common, 7120 param.toString().string()); 7121 *handle = AUDIO_PATCH_HANDLE_NONE; 7122 } 7123 7124 if (mInDevice != mPrevInDevice) { 7125 sendIoConfigEvent_l(AUDIO_INPUT_CONFIG_CHANGED); 7126 mPrevInDevice = mInDevice; 7127 } 7128 7129 return status; 7130} 7131 7132status_t AudioFlinger::RecordThread::releaseAudioPatch_l(const audio_patch_handle_t handle) 7133{ 7134 status_t status = NO_ERROR; 7135 7136 mInDevice = AUDIO_DEVICE_NONE; 7137 7138 if (mInput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 7139 audio_hw_device_t *hwDevice = mInput->audioHwDev->hwDevice(); 7140 status = hwDevice->release_audio_patch(hwDevice, handle); 7141 } else { 7142 AudioParameter param; 7143 param.addInt(String8(AUDIO_PARAMETER_STREAM_ROUTING), 0); 7144 status = mInput->stream->common.set_parameters(&mInput->stream->common, 7145 param.toString().string()); 7146 } 7147 return status; 7148} 7149 7150void AudioFlinger::RecordThread::addPatchRecord(const sp<PatchRecord>& record) 7151{ 7152 Mutex::Autolock _l(mLock); 7153 mTracks.add(record); 7154} 7155 7156void AudioFlinger::RecordThread::deletePatchRecord(const sp<PatchRecord>& record) 7157{ 7158 Mutex::Autolock _l(mLock); 7159 destroyTrack_l(record); 7160} 7161 7162void AudioFlinger::RecordThread::getAudioPortConfig(struct audio_port_config *config) 7163{ 7164 ThreadBase::getAudioPortConfig(config); 7165 config->role = AUDIO_PORT_ROLE_SINK; 7166 config->ext.mix.hw_module = mInput->audioHwDev->handle(); 7167 config->ext.mix.usecase.source = mAudioSource; 7168} 7169 7170} // namespace android 7171