Threads.cpp revision e0a269a5f75956efdf78a9cacaefc428b352730c
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/conversion.h> 40#include <audio_utils/primitives.h> 41#include <audio_utils/format.h> 42#include <audio_utils/minifloat.h> 43 44// NBAIO implementations 45#include <media/nbaio/AudioStreamInSource.h> 46#include <media/nbaio/AudioStreamOutSink.h> 47#include <media/nbaio/MonoPipe.h> 48#include <media/nbaio/MonoPipeReader.h> 49#include <media/nbaio/Pipe.h> 50#include <media/nbaio/PipeReader.h> 51#include <media/nbaio/SourceAudioBufferProvider.h> 52#include <mediautils/BatteryNotifier.h> 53 54#include <powermanager/PowerManager.h> 55 56#include "AudioFlinger.h" 57#include "AudioMixer.h" 58#include "BufferProviders.h" 59#include "FastMixer.h" 60#include "FastCapture.h" 61#include "ServiceUtilities.h" 62#include "mediautils/SchedulingPolicyService.h" 63 64#ifdef ADD_BATTERY_DATA 65#include <media/IMediaPlayerService.h> 66#include <media/IMediaDeathNotifier.h> 67#endif 68 69#ifdef DEBUG_CPU_USAGE 70#include <cpustats/CentralTendencyStatistics.h> 71#include <cpustats/ThreadCpuUsage.h> 72#endif 73 74// ---------------------------------------------------------------------------- 75 76// Note: the following macro is used for extremely verbose logging message. In 77// order to run with ALOG_ASSERT turned on, we need to have LOG_NDEBUG set to 78// 0; but one side effect of this is to turn all LOGV's as well. Some messages 79// are so verbose that we want to suppress them even when we have ALOG_ASSERT 80// turned on. Do not uncomment the #def below unless you really know what you 81// are doing and want to see all of the extremely verbose messages. 82//#define VERY_VERY_VERBOSE_LOGGING 83#ifdef VERY_VERY_VERBOSE_LOGGING 84#define ALOGVV ALOGV 85#else 86#define ALOGVV(a...) do { } while(0) 87#endif 88 89// TODO: Move these macro/inlines to a header file. 90#define max(a, b) ((a) > (b) ? (a) : (b)) 91template <typename T> 92static inline T min(const T& a, const T& b) 93{ 94 return a < b ? a : b; 95} 96 97#ifndef ARRAY_SIZE 98#define ARRAY_SIZE(a) (sizeof(a) / sizeof(a[0])) 99#endif 100 101namespace android { 102 103// retry counts for buffer fill timeout 104// 50 * ~20msecs = 1 second 105static const int8_t kMaxTrackRetries = 50; 106static const int8_t kMaxTrackStartupRetries = 50; 107// allow less retry attempts on direct output thread. 108// direct outputs can be a scarce resource in audio hardware and should 109// be released as quickly as possible. 110static const int8_t kMaxTrackRetriesDirect = 2; 111// retry count before removing active track in case of underrun on offloaded thread: 112// we need to make sure that AudioTrack client has enough time to send large buffers 113//FIXME may be more appropriate if expressed in time units. Need to revise how underrun is handled 114// for offloaded tracks 115static const int8_t kMaxTrackRetriesOffload = 10; 116static const int8_t kMaxTrackStartupRetriesOffload = 100; 117 118 119// don't warn about blocked writes or record buffer overflows more often than this 120static const nsecs_t kWarningThrottleNs = seconds(5); 121 122// RecordThread loop sleep time upon application overrun or audio HAL read error 123static const int kRecordThreadSleepUs = 5000; 124 125// maximum time to wait in sendConfigEvent_l() for a status to be received 126static const nsecs_t kConfigEventTimeoutNs = seconds(2); 127 128// minimum sleep time for the mixer thread loop when tracks are active but in underrun 129static const uint32_t kMinThreadSleepTimeUs = 5000; 130// maximum divider applied to the active sleep time in the mixer thread loop 131static const uint32_t kMaxThreadSleepTimeShift = 2; 132 133// minimum normal sink buffer size, expressed in milliseconds rather than frames 134// FIXME This should be based on experimentally observed scheduling jitter 135static const uint32_t kMinNormalSinkBufferSizeMs = 20; 136// maximum normal sink buffer size 137static const uint32_t kMaxNormalSinkBufferSizeMs = 24; 138 139// minimum capture buffer size in milliseconds to _not_ need a fast capture thread 140// FIXME This should be based on experimentally observed scheduling jitter 141static const uint32_t kMinNormalCaptureBufferSizeMs = 12; 142 143// Offloaded output thread standby delay: allows track transition without going to standby 144static const nsecs_t kOffloadStandbyDelayNs = seconds(1); 145 146// Direct output thread minimum sleep time in idle or active(underrun) state 147static const nsecs_t kDirectMinSleepTimeUs = 10000; 148 149// Offloaded output bit rate in bits per second when unknown. 150// Used for sleep time calculation, so use a high default bitrate to be conservative on sleep time. 151static const uint32_t kOffloadDefaultBitRateBps = 1500000; 152 153 154// Whether to use fast mixer 155static const enum { 156 FastMixer_Never, // never initialize or use: for debugging only 157 FastMixer_Always, // always initialize and use, even if not needed: for debugging only 158 // normal mixer multiplier is 1 159 FastMixer_Static, // initialize if needed, then use all the time if initialized, 160 // multiplier is calculated based on min & max normal mixer buffer size 161 FastMixer_Dynamic, // initialize if needed, then use dynamically depending on track load, 162 // multiplier is calculated based on min & max normal mixer buffer size 163 // FIXME for FastMixer_Dynamic: 164 // Supporting this option will require fixing HALs that can't handle large writes. 165 // For example, one HAL implementation returns an error from a large write, 166 // and another HAL implementation corrupts memory, possibly in the sample rate converter. 167 // We could either fix the HAL implementations, or provide a wrapper that breaks 168 // up large writes into smaller ones, and the wrapper would need to deal with scheduler. 169} kUseFastMixer = FastMixer_Static; 170 171// Whether to use fast capture 172static const enum { 173 FastCapture_Never, // never initialize or use: for debugging only 174 FastCapture_Always, // always initialize and use, even if not needed: for debugging only 175 FastCapture_Static, // initialize if needed, then use all the time if initialized 176} kUseFastCapture = FastCapture_Static; 177 178// Priorities for requestPriority 179static const int kPriorityAudioApp = 2; 180static const int kPriorityFastMixer = 3; 181static const int kPriorityFastCapture = 3; 182 183// IAudioFlinger::createTrack() reports back to client the total size of shared memory area 184// for the track. The client then sub-divides this into smaller buffers for its use. 185// Currently the client uses N-buffering by default, but doesn't tell us about the value of N. 186// So for now we just assume that client is double-buffered for fast tracks. 187// FIXME It would be better for client to tell AudioFlinger the value of N, 188// so AudioFlinger could allocate the right amount of memory. 189// See the client's minBufCount and mNotificationFramesAct calculations for details. 190 191// This is the default value, if not specified by property. 192static const int kFastTrackMultiplier = 2; 193 194// The minimum and maximum allowed values 195static const int kFastTrackMultiplierMin = 1; 196static const int kFastTrackMultiplierMax = 2; 197 198// The actual value to use, which can be specified per-device via property af.fast_track_multiplier. 199static int sFastTrackMultiplier = kFastTrackMultiplier; 200 201// See Thread::readOnlyHeap(). 202// Initially this heap is used to allocate client buffers for "fast" AudioRecord. 203// Eventually it will be the single buffer that FastCapture writes into via HAL read(), 204// and that all "fast" AudioRecord clients read from. In either case, the size can be small. 205static const size_t kRecordThreadReadOnlyHeapSize = 0x2000; 206 207// ---------------------------------------------------------------------------- 208 209static pthread_once_t sFastTrackMultiplierOnce = PTHREAD_ONCE_INIT; 210 211static void sFastTrackMultiplierInit() 212{ 213 char value[PROPERTY_VALUE_MAX]; 214 if (property_get("af.fast_track_multiplier", value, NULL) > 0) { 215 char *endptr; 216 unsigned long ul = strtoul(value, &endptr, 0); 217 if (*endptr == '\0' && kFastTrackMultiplierMin <= ul && ul <= kFastTrackMultiplierMax) { 218 sFastTrackMultiplier = (int) ul; 219 } 220 } 221} 222 223// ---------------------------------------------------------------------------- 224 225#ifdef ADD_BATTERY_DATA 226// To collect the amplifier usage 227static void addBatteryData(uint32_t params) { 228 sp<IMediaPlayerService> service = IMediaDeathNotifier::getMediaPlayerService(); 229 if (service == NULL) { 230 // it already logged 231 return; 232 } 233 234 service->addBatteryData(params); 235} 236#endif 237 238// Track the CLOCK_BOOTTIME versus CLOCK_MONOTONIC timebase offset 239struct { 240 // call when you acquire a partial wakelock 241 void acquire(const sp<IBinder> &wakeLockToken) { 242 pthread_mutex_lock(&mLock); 243 if (wakeLockToken.get() == nullptr) { 244 adjustTimebaseOffset(&mBoottimeOffset, ExtendedTimestamp::TIMEBASE_BOOTTIME); 245 } else { 246 if (mCount == 0) { 247 adjustTimebaseOffset(&mBoottimeOffset, ExtendedTimestamp::TIMEBASE_BOOTTIME); 248 } 249 ++mCount; 250 } 251 pthread_mutex_unlock(&mLock); 252 } 253 254 // call when you release a partial wakelock. 255 void release(const sp<IBinder> &wakeLockToken) { 256 if (wakeLockToken.get() == nullptr) { 257 return; 258 } 259 pthread_mutex_lock(&mLock); 260 if (--mCount < 0) { 261 ALOGE("negative wakelock count"); 262 mCount = 0; 263 } 264 pthread_mutex_unlock(&mLock); 265 } 266 267 // retrieves the boottime timebase offset from monotonic. 268 int64_t getBoottimeOffset() { 269 pthread_mutex_lock(&mLock); 270 int64_t boottimeOffset = mBoottimeOffset; 271 pthread_mutex_unlock(&mLock); 272 return boottimeOffset; 273 } 274 275 // Adjusts the timebase offset between TIMEBASE_MONOTONIC 276 // and the selected timebase. 277 // Currently only TIMEBASE_BOOTTIME is allowed. 278 // 279 // This only needs to be called upon acquiring the first partial wakelock 280 // after all other partial wakelocks are released. 281 // 282 // We do an empirical measurement of the offset rather than parsing 283 // /proc/timer_list since the latter is not a formal kernel ABI. 284 static void adjustTimebaseOffset(int64_t *offset, ExtendedTimestamp::Timebase timebase) { 285 int clockbase; 286 switch (timebase) { 287 case ExtendedTimestamp::TIMEBASE_BOOTTIME: 288 clockbase = SYSTEM_TIME_BOOTTIME; 289 break; 290 default: 291 LOG_ALWAYS_FATAL("invalid timebase %d", timebase); 292 break; 293 } 294 // try three times to get the clock offset, choose the one 295 // with the minimum gap in measurements. 296 const int tries = 3; 297 nsecs_t bestGap, measured; 298 for (int i = 0; i < tries; ++i) { 299 const nsecs_t tmono = systemTime(SYSTEM_TIME_MONOTONIC); 300 const nsecs_t tbase = systemTime(clockbase); 301 const nsecs_t tmono2 = systemTime(SYSTEM_TIME_MONOTONIC); 302 const nsecs_t gap = tmono2 - tmono; 303 if (i == 0 || gap < bestGap) { 304 bestGap = gap; 305 measured = tbase - ((tmono + tmono2) >> 1); 306 } 307 } 308 309 // to avoid micro-adjusting, we don't change the timebase 310 // unless it is significantly different. 311 // 312 // Assumption: It probably takes more than toleranceNs to 313 // suspend and resume the device. 314 static int64_t toleranceNs = 10000; // 10 us 315 if (llabs(*offset - measured) > toleranceNs) { 316 ALOGV("Adjusting timebase offset old: %lld new: %lld", 317 (long long)*offset, (long long)measured); 318 *offset = measured; 319 } 320 } 321 322 pthread_mutex_t mLock; 323 int32_t mCount; 324 int64_t mBoottimeOffset; 325} gBoottime = { PTHREAD_MUTEX_INITIALIZER, 0, 0 }; // static, so use POD initialization 326 327// ---------------------------------------------------------------------------- 328// CPU Stats 329// ---------------------------------------------------------------------------- 330 331class CpuStats { 332public: 333 CpuStats(); 334 void sample(const String8 &title); 335#ifdef DEBUG_CPU_USAGE 336private: 337 ThreadCpuUsage mCpuUsage; // instantaneous thread CPU usage in wall clock ns 338 CentralTendencyStatistics mWcStats; // statistics on thread CPU usage in wall clock ns 339 340 CentralTendencyStatistics mHzStats; // statistics on thread CPU usage in cycles 341 342 int mCpuNum; // thread's current CPU number 343 int mCpukHz; // frequency of thread's current CPU in kHz 344#endif 345}; 346 347CpuStats::CpuStats() 348#ifdef DEBUG_CPU_USAGE 349 : mCpuNum(-1), mCpukHz(-1) 350#endif 351{ 352} 353 354void CpuStats::sample(const String8 &title 355#ifndef DEBUG_CPU_USAGE 356 __unused 357#endif 358 ) { 359#ifdef DEBUG_CPU_USAGE 360 // get current thread's delta CPU time in wall clock ns 361 double wcNs; 362 bool valid = mCpuUsage.sampleAndEnable(wcNs); 363 364 // record sample for wall clock statistics 365 if (valid) { 366 mWcStats.sample(wcNs); 367 } 368 369 // get the current CPU number 370 int cpuNum = sched_getcpu(); 371 372 // get the current CPU frequency in kHz 373 int cpukHz = mCpuUsage.getCpukHz(cpuNum); 374 375 // check if either CPU number or frequency changed 376 if (cpuNum != mCpuNum || cpukHz != mCpukHz) { 377 mCpuNum = cpuNum; 378 mCpukHz = cpukHz; 379 // ignore sample for purposes of cycles 380 valid = false; 381 } 382 383 // if no change in CPU number or frequency, then record sample for cycle statistics 384 if (valid && mCpukHz > 0) { 385 double cycles = wcNs * cpukHz * 0.000001; 386 mHzStats.sample(cycles); 387 } 388 389 unsigned n = mWcStats.n(); 390 // mCpuUsage.elapsed() is expensive, so don't call it every loop 391 if ((n & 127) == 1) { 392 long long elapsed = mCpuUsage.elapsed(); 393 if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) { 394 double perLoop = elapsed / (double) n; 395 double perLoop100 = perLoop * 0.01; 396 double perLoop1k = perLoop * 0.001; 397 double mean = mWcStats.mean(); 398 double stddev = mWcStats.stddev(); 399 double minimum = mWcStats.minimum(); 400 double maximum = mWcStats.maximum(); 401 double meanCycles = mHzStats.mean(); 402 double stddevCycles = mHzStats.stddev(); 403 double minCycles = mHzStats.minimum(); 404 double maxCycles = mHzStats.maximum(); 405 mCpuUsage.resetElapsed(); 406 mWcStats.reset(); 407 mHzStats.reset(); 408 ALOGD("CPU usage for %s over past %.1f secs\n" 409 " (%u mixer loops at %.1f mean ms per loop):\n" 410 " us per mix loop: mean=%.0f stddev=%.0f min=%.0f max=%.0f\n" 411 " %% of wall: mean=%.1f stddev=%.1f min=%.1f max=%.1f\n" 412 " MHz: mean=%.1f, stddev=%.1f, min=%.1f max=%.1f", 413 title.string(), 414 elapsed * .000000001, n, perLoop * .000001, 415 mean * .001, 416 stddev * .001, 417 minimum * .001, 418 maximum * .001, 419 mean / perLoop100, 420 stddev / perLoop100, 421 minimum / perLoop100, 422 maximum / perLoop100, 423 meanCycles / perLoop1k, 424 stddevCycles / perLoop1k, 425 minCycles / perLoop1k, 426 maxCycles / perLoop1k); 427 428 } 429 } 430#endif 431}; 432 433// ---------------------------------------------------------------------------- 434// ThreadBase 435// ---------------------------------------------------------------------------- 436 437// static 438const char *AudioFlinger::ThreadBase::threadTypeToString(AudioFlinger::ThreadBase::type_t type) 439{ 440 switch (type) { 441 case MIXER: 442 return "MIXER"; 443 case DIRECT: 444 return "DIRECT"; 445 case DUPLICATING: 446 return "DUPLICATING"; 447 case RECORD: 448 return "RECORD"; 449 case OFFLOAD: 450 return "OFFLOAD"; 451 default: 452 return "unknown"; 453 } 454} 455 456String8 devicesToString(audio_devices_t devices) 457{ 458 static const struct mapping { 459 audio_devices_t mDevices; 460 const char * mString; 461 } mappingsOut[] = { 462 {AUDIO_DEVICE_OUT_EARPIECE, "EARPIECE"}, 463 {AUDIO_DEVICE_OUT_SPEAKER, "SPEAKER"}, 464 {AUDIO_DEVICE_OUT_WIRED_HEADSET, "WIRED_HEADSET"}, 465 {AUDIO_DEVICE_OUT_WIRED_HEADPHONE, "WIRED_HEADPHONE"}, 466 {AUDIO_DEVICE_OUT_BLUETOOTH_SCO, "BLUETOOTH_SCO"}, 467 {AUDIO_DEVICE_OUT_BLUETOOTH_SCO_HEADSET, "BLUETOOTH_SCO_HEADSET"}, 468 {AUDIO_DEVICE_OUT_BLUETOOTH_SCO_CARKIT, "BLUETOOTH_SCO_CARKIT"}, 469 {AUDIO_DEVICE_OUT_BLUETOOTH_A2DP, "BLUETOOTH_A2DP"}, 470 {AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES,"BLUETOOTH_A2DP_HEADPHONES"}, 471 {AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER, "BLUETOOTH_A2DP_SPEAKER"}, 472 {AUDIO_DEVICE_OUT_AUX_DIGITAL, "AUX_DIGITAL"}, 473 {AUDIO_DEVICE_OUT_HDMI, "HDMI"}, 474 {AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET,"ANLG_DOCK_HEADSET"}, 475 {AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET,"DGTL_DOCK_HEADSET"}, 476 {AUDIO_DEVICE_OUT_USB_ACCESSORY, "USB_ACCESSORY"}, 477 {AUDIO_DEVICE_OUT_USB_DEVICE, "USB_DEVICE"}, 478 {AUDIO_DEVICE_OUT_TELEPHONY_TX, "TELEPHONY_TX"}, 479 {AUDIO_DEVICE_OUT_LINE, "LINE"}, 480 {AUDIO_DEVICE_OUT_HDMI_ARC, "HDMI_ARC"}, 481 {AUDIO_DEVICE_OUT_SPDIF, "SPDIF"}, 482 {AUDIO_DEVICE_OUT_FM, "FM"}, 483 {AUDIO_DEVICE_OUT_AUX_LINE, "AUX_LINE"}, 484 {AUDIO_DEVICE_OUT_SPEAKER_SAFE, "SPEAKER_SAFE"}, 485 {AUDIO_DEVICE_OUT_IP, "IP"}, 486 {AUDIO_DEVICE_OUT_BUS, "BUS"}, 487 {AUDIO_DEVICE_NONE, "NONE"}, // must be last 488 }, mappingsIn[] = { 489 {AUDIO_DEVICE_IN_COMMUNICATION, "COMMUNICATION"}, 490 {AUDIO_DEVICE_IN_AMBIENT, "AMBIENT"}, 491 {AUDIO_DEVICE_IN_BUILTIN_MIC, "BUILTIN_MIC"}, 492 {AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET, "BLUETOOTH_SCO_HEADSET"}, 493 {AUDIO_DEVICE_IN_WIRED_HEADSET, "WIRED_HEADSET"}, 494 {AUDIO_DEVICE_IN_AUX_DIGITAL, "AUX_DIGITAL"}, 495 {AUDIO_DEVICE_IN_VOICE_CALL, "VOICE_CALL"}, 496 {AUDIO_DEVICE_IN_TELEPHONY_RX, "TELEPHONY_RX"}, 497 {AUDIO_DEVICE_IN_BACK_MIC, "BACK_MIC"}, 498 {AUDIO_DEVICE_IN_REMOTE_SUBMIX, "REMOTE_SUBMIX"}, 499 {AUDIO_DEVICE_IN_ANLG_DOCK_HEADSET, "ANLG_DOCK_HEADSET"}, 500 {AUDIO_DEVICE_IN_DGTL_DOCK_HEADSET, "DGTL_DOCK_HEADSET"}, 501 {AUDIO_DEVICE_IN_USB_ACCESSORY, "USB_ACCESSORY"}, 502 {AUDIO_DEVICE_IN_USB_DEVICE, "USB_DEVICE"}, 503 {AUDIO_DEVICE_IN_FM_TUNER, "FM_TUNER"}, 504 {AUDIO_DEVICE_IN_TV_TUNER, "TV_TUNER"}, 505 {AUDIO_DEVICE_IN_LINE, "LINE"}, 506 {AUDIO_DEVICE_IN_SPDIF, "SPDIF"}, 507 {AUDIO_DEVICE_IN_BLUETOOTH_A2DP, "BLUETOOTH_A2DP"}, 508 {AUDIO_DEVICE_IN_LOOPBACK, "LOOPBACK"}, 509 {AUDIO_DEVICE_IN_IP, "IP"}, 510 {AUDIO_DEVICE_IN_BUS, "BUS"}, 511 {AUDIO_DEVICE_NONE, "NONE"}, // must be last 512 }; 513 String8 result; 514 audio_devices_t allDevices = AUDIO_DEVICE_NONE; 515 const mapping *entry; 516 if (devices & AUDIO_DEVICE_BIT_IN) { 517 devices &= ~AUDIO_DEVICE_BIT_IN; 518 entry = mappingsIn; 519 } else { 520 entry = mappingsOut; 521 } 522 for ( ; entry->mDevices != AUDIO_DEVICE_NONE; entry++) { 523 allDevices = (audio_devices_t) (allDevices | entry->mDevices); 524 if (devices & entry->mDevices) { 525 if (!result.isEmpty()) { 526 result.append("|"); 527 } 528 result.append(entry->mString); 529 } 530 } 531 if (devices & ~allDevices) { 532 if (!result.isEmpty()) { 533 result.append("|"); 534 } 535 result.appendFormat("0x%X", devices & ~allDevices); 536 } 537 if (result.isEmpty()) { 538 result.append(entry->mString); 539 } 540 return result; 541} 542 543String8 inputFlagsToString(audio_input_flags_t flags) 544{ 545 static const struct mapping { 546 audio_input_flags_t mFlag; 547 const char * mString; 548 } mappings[] = { 549 {AUDIO_INPUT_FLAG_FAST, "FAST"}, 550 {AUDIO_INPUT_FLAG_HW_HOTWORD, "HW_HOTWORD"}, 551 {AUDIO_INPUT_FLAG_RAW, "RAW"}, 552 {AUDIO_INPUT_FLAG_SYNC, "SYNC"}, 553 {AUDIO_INPUT_FLAG_NONE, "NONE"}, // must be last 554 }; 555 String8 result; 556 audio_input_flags_t allFlags = AUDIO_INPUT_FLAG_NONE; 557 const mapping *entry; 558 for (entry = mappings; entry->mFlag != AUDIO_INPUT_FLAG_NONE; entry++) { 559 allFlags = (audio_input_flags_t) (allFlags | entry->mFlag); 560 if (flags & entry->mFlag) { 561 if (!result.isEmpty()) { 562 result.append("|"); 563 } 564 result.append(entry->mString); 565 } 566 } 567 if (flags & ~allFlags) { 568 if (!result.isEmpty()) { 569 result.append("|"); 570 } 571 result.appendFormat("0x%X", flags & ~allFlags); 572 } 573 if (result.isEmpty()) { 574 result.append(entry->mString); 575 } 576 return result; 577} 578 579String8 outputFlagsToString(audio_output_flags_t flags) 580{ 581 static const struct mapping { 582 audio_output_flags_t mFlag; 583 const char * mString; 584 } mappings[] = { 585 {AUDIO_OUTPUT_FLAG_DIRECT, "DIRECT"}, 586 {AUDIO_OUTPUT_FLAG_PRIMARY, "PRIMARY"}, 587 {AUDIO_OUTPUT_FLAG_FAST, "FAST"}, 588 {AUDIO_OUTPUT_FLAG_DEEP_BUFFER, "DEEP_BUFFER"}, 589 {AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD,"COMPRESS_OFFLOAD"}, 590 {AUDIO_OUTPUT_FLAG_NON_BLOCKING, "NON_BLOCKING"}, 591 {AUDIO_OUTPUT_FLAG_HW_AV_SYNC, "HW_AV_SYNC"}, 592 {AUDIO_OUTPUT_FLAG_RAW, "RAW"}, 593 {AUDIO_OUTPUT_FLAG_SYNC, "SYNC"}, 594 {AUDIO_OUTPUT_FLAG_IEC958_NONAUDIO, "IEC958_NONAUDIO"}, 595 {AUDIO_OUTPUT_FLAG_NONE, "NONE"}, // must be last 596 }; 597 String8 result; 598 audio_output_flags_t allFlags = AUDIO_OUTPUT_FLAG_NONE; 599 const mapping *entry; 600 for (entry = mappings; entry->mFlag != AUDIO_OUTPUT_FLAG_NONE; entry++) { 601 allFlags = (audio_output_flags_t) (allFlags | entry->mFlag); 602 if (flags & entry->mFlag) { 603 if (!result.isEmpty()) { 604 result.append("|"); 605 } 606 result.append(entry->mString); 607 } 608 } 609 if (flags & ~allFlags) { 610 if (!result.isEmpty()) { 611 result.append("|"); 612 } 613 result.appendFormat("0x%X", flags & ~allFlags); 614 } 615 if (result.isEmpty()) { 616 result.append(entry->mString); 617 } 618 return result; 619} 620 621const char *sourceToString(audio_source_t source) 622{ 623 switch (source) { 624 case AUDIO_SOURCE_DEFAULT: return "default"; 625 case AUDIO_SOURCE_MIC: return "mic"; 626 case AUDIO_SOURCE_VOICE_UPLINK: return "voice uplink"; 627 case AUDIO_SOURCE_VOICE_DOWNLINK: return "voice downlink"; 628 case AUDIO_SOURCE_VOICE_CALL: return "voice call"; 629 case AUDIO_SOURCE_CAMCORDER: return "camcorder"; 630 case AUDIO_SOURCE_VOICE_RECOGNITION: return "voice recognition"; 631 case AUDIO_SOURCE_VOICE_COMMUNICATION: return "voice communication"; 632 case AUDIO_SOURCE_REMOTE_SUBMIX: return "remote submix"; 633 case AUDIO_SOURCE_UNPROCESSED: return "unprocessed"; 634 case AUDIO_SOURCE_FM_TUNER: return "FM tuner"; 635 case AUDIO_SOURCE_HOTWORD: return "hotword"; 636 default: return "unknown"; 637 } 638} 639 640AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, 641 audio_devices_t outDevice, audio_devices_t inDevice, type_t type, bool systemReady) 642 : Thread(false /*canCallJava*/), 643 mType(type), 644 mAudioFlinger(audioFlinger), 645 // mSampleRate, mFrameCount, mChannelMask, mChannelCount, mFrameSize, mFormat, mBufferSize 646 // are set by PlaybackThread::readOutputParameters_l() or 647 // RecordThread::readInputParameters_l() 648 //FIXME: mStandby should be true here. Is this some kind of hack? 649 mStandby(false), mOutDevice(outDevice), mInDevice(inDevice), 650 mPrevOutDevice(AUDIO_DEVICE_NONE), mPrevInDevice(AUDIO_DEVICE_NONE), 651 mAudioSource(AUDIO_SOURCE_DEFAULT), mId(id), 652 // mName will be set by concrete (non-virtual) subclass 653 mDeathRecipient(new PMDeathRecipient(this)), 654 mSystemReady(systemReady), 655 mNotifiedBatteryStart(false) 656{ 657 memset(&mPatch, 0, sizeof(struct audio_patch)); 658} 659 660AudioFlinger::ThreadBase::~ThreadBase() 661{ 662 // mConfigEvents should be empty, but just in case it isn't, free the memory it owns 663 mConfigEvents.clear(); 664 665 // do not lock the mutex in destructor 666 releaseWakeLock_l(); 667 if (mPowerManager != 0) { 668 sp<IBinder> binder = IInterface::asBinder(mPowerManager); 669 binder->unlinkToDeath(mDeathRecipient); 670 } 671} 672 673status_t AudioFlinger::ThreadBase::readyToRun() 674{ 675 status_t status = initCheck(); 676 if (status == NO_ERROR) { 677 ALOGI("AudioFlinger's thread %p ready to run", this); 678 } else { 679 ALOGE("No working audio driver found."); 680 } 681 return status; 682} 683 684void AudioFlinger::ThreadBase::exit() 685{ 686 ALOGV("ThreadBase::exit"); 687 // do any cleanup required for exit to succeed 688 preExit(); 689 { 690 // This lock prevents the following race in thread (uniprocessor for illustration): 691 // if (!exitPending()) { 692 // // context switch from here to exit() 693 // // exit() calls requestExit(), what exitPending() observes 694 // // exit() calls signal(), which is dropped since no waiters 695 // // context switch back from exit() to here 696 // mWaitWorkCV.wait(...); 697 // // now thread is hung 698 // } 699 AutoMutex lock(mLock); 700 requestExit(); 701 mWaitWorkCV.broadcast(); 702 } 703 // When Thread::requestExitAndWait is made virtual and this method is renamed to 704 // "virtual status_t requestExitAndWait()", replace by "return Thread::requestExitAndWait();" 705 requestExitAndWait(); 706} 707 708status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs) 709{ 710 ALOGV("ThreadBase::setParameters() %s", keyValuePairs.string()); 711 Mutex::Autolock _l(mLock); 712 713 return sendSetParameterConfigEvent_l(keyValuePairs); 714} 715 716// sendConfigEvent_l() must be called with ThreadBase::mLock held 717// Can temporarily release the lock if waiting for a reply from processConfigEvents_l(). 718status_t AudioFlinger::ThreadBase::sendConfigEvent_l(sp<ConfigEvent>& event) 719{ 720 status_t status = NO_ERROR; 721 722 if (event->mRequiresSystemReady && !mSystemReady) { 723 event->mWaitStatus = false; 724 mPendingConfigEvents.add(event); 725 return status; 726 } 727 mConfigEvents.add(event); 728 ALOGV("sendConfigEvent_l() num events %zu event %d", mConfigEvents.size(), event->mType); 729 mWaitWorkCV.signal(); 730 mLock.unlock(); 731 { 732 Mutex::Autolock _l(event->mLock); 733 while (event->mWaitStatus) { 734 if (event->mCond.waitRelative(event->mLock, kConfigEventTimeoutNs) != NO_ERROR) { 735 event->mStatus = TIMED_OUT; 736 event->mWaitStatus = false; 737 } 738 } 739 status = event->mStatus; 740 } 741 mLock.lock(); 742 return status; 743} 744 745void AudioFlinger::ThreadBase::sendIoConfigEvent(audio_io_config_event event, pid_t pid) 746{ 747 Mutex::Autolock _l(mLock); 748 sendIoConfigEvent_l(event, pid); 749} 750 751// sendIoConfigEvent_l() must be called with ThreadBase::mLock held 752void AudioFlinger::ThreadBase::sendIoConfigEvent_l(audio_io_config_event event, pid_t pid) 753{ 754 sp<ConfigEvent> configEvent = (ConfigEvent *)new IoConfigEvent(event, pid); 755 sendConfigEvent_l(configEvent); 756} 757 758void AudioFlinger::ThreadBase::sendPrioConfigEvent(pid_t pid, pid_t tid, int32_t prio) 759{ 760 Mutex::Autolock _l(mLock); 761 sendPrioConfigEvent_l(pid, tid, prio); 762} 763 764// sendPrioConfigEvent_l() must be called with ThreadBase::mLock held 765void AudioFlinger::ThreadBase::sendPrioConfigEvent_l(pid_t pid, pid_t tid, int32_t prio) 766{ 767 sp<ConfigEvent> configEvent = (ConfigEvent *)new PrioConfigEvent(pid, tid, prio); 768 sendConfigEvent_l(configEvent); 769} 770 771// sendSetParameterConfigEvent_l() must be called with ThreadBase::mLock held 772status_t AudioFlinger::ThreadBase::sendSetParameterConfigEvent_l(const String8& keyValuePair) 773{ 774 sp<ConfigEvent> configEvent; 775 AudioParameter param(keyValuePair); 776 int value; 777 if (param.getInt(String8(AUDIO_PARAMETER_MONO_OUTPUT), value) == NO_ERROR) { 778 setMasterMono_l(value != 0); 779 if (param.size() == 1) { 780 return NO_ERROR; // should be a solo parameter - we don't pass down 781 } 782 param.remove(String8(AUDIO_PARAMETER_MONO_OUTPUT)); 783 configEvent = new SetParameterConfigEvent(param.toString()); 784 } else { 785 configEvent = new SetParameterConfigEvent(keyValuePair); 786 } 787 return sendConfigEvent_l(configEvent); 788} 789 790status_t AudioFlinger::ThreadBase::sendCreateAudioPatchConfigEvent( 791 const struct audio_patch *patch, 792 audio_patch_handle_t *handle) 793{ 794 Mutex::Autolock _l(mLock); 795 sp<ConfigEvent> configEvent = (ConfigEvent *)new CreateAudioPatchConfigEvent(*patch, *handle); 796 status_t status = sendConfigEvent_l(configEvent); 797 if (status == NO_ERROR) { 798 CreateAudioPatchConfigEventData *data = 799 (CreateAudioPatchConfigEventData *)configEvent->mData.get(); 800 *handle = data->mHandle; 801 } 802 return status; 803} 804 805status_t AudioFlinger::ThreadBase::sendReleaseAudioPatchConfigEvent( 806 const audio_patch_handle_t handle) 807{ 808 Mutex::Autolock _l(mLock); 809 sp<ConfigEvent> configEvent = (ConfigEvent *)new ReleaseAudioPatchConfigEvent(handle); 810 return sendConfigEvent_l(configEvent); 811} 812 813 814// post condition: mConfigEvents.isEmpty() 815void AudioFlinger::ThreadBase::processConfigEvents_l() 816{ 817 bool configChanged = false; 818 819 while (!mConfigEvents.isEmpty()) { 820 ALOGV("processConfigEvents_l() remaining events %zu", mConfigEvents.size()); 821 sp<ConfigEvent> event = mConfigEvents[0]; 822 mConfigEvents.removeAt(0); 823 switch (event->mType) { 824 case CFG_EVENT_PRIO: { 825 PrioConfigEventData *data = (PrioConfigEventData *)event->mData.get(); 826 // FIXME Need to understand why this has to be done asynchronously 827 int err = requestPriority(data->mPid, data->mTid, data->mPrio, 828 true /*asynchronous*/); 829 if (err != 0) { 830 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 831 data->mPrio, data->mPid, data->mTid, err); 832 } 833 } break; 834 case CFG_EVENT_IO: { 835 IoConfigEventData *data = (IoConfigEventData *)event->mData.get(); 836 ioConfigChanged(data->mEvent, data->mPid); 837 } break; 838 case CFG_EVENT_SET_PARAMETER: { 839 SetParameterConfigEventData *data = (SetParameterConfigEventData *)event->mData.get(); 840 if (checkForNewParameter_l(data->mKeyValuePairs, event->mStatus)) { 841 configChanged = true; 842 } 843 } break; 844 case CFG_EVENT_CREATE_AUDIO_PATCH: { 845 CreateAudioPatchConfigEventData *data = 846 (CreateAudioPatchConfigEventData *)event->mData.get(); 847 event->mStatus = createAudioPatch_l(&data->mPatch, &data->mHandle); 848 } break; 849 case CFG_EVENT_RELEASE_AUDIO_PATCH: { 850 ReleaseAudioPatchConfigEventData *data = 851 (ReleaseAudioPatchConfigEventData *)event->mData.get(); 852 event->mStatus = releaseAudioPatch_l(data->mHandle); 853 } break; 854 default: 855 ALOG_ASSERT(false, "processConfigEvents_l() unknown event type %d", event->mType); 856 break; 857 } 858 { 859 Mutex::Autolock _l(event->mLock); 860 if (event->mWaitStatus) { 861 event->mWaitStatus = false; 862 event->mCond.signal(); 863 } 864 } 865 ALOGV_IF(mConfigEvents.isEmpty(), "processConfigEvents_l() DONE thread %p", this); 866 } 867 868 if (configChanged) { 869 cacheParameters_l(); 870 } 871} 872 873String8 channelMaskToString(audio_channel_mask_t mask, bool output) { 874 String8 s; 875 const audio_channel_representation_t representation = 876 audio_channel_mask_get_representation(mask); 877 878 switch (representation) { 879 case AUDIO_CHANNEL_REPRESENTATION_POSITION: { 880 if (output) { 881 if (mask & AUDIO_CHANNEL_OUT_FRONT_LEFT) s.append("front-left, "); 882 if (mask & AUDIO_CHANNEL_OUT_FRONT_RIGHT) s.append("front-right, "); 883 if (mask & AUDIO_CHANNEL_OUT_FRONT_CENTER) s.append("front-center, "); 884 if (mask & AUDIO_CHANNEL_OUT_LOW_FREQUENCY) s.append("low freq, "); 885 if (mask & AUDIO_CHANNEL_OUT_BACK_LEFT) s.append("back-left, "); 886 if (mask & AUDIO_CHANNEL_OUT_BACK_RIGHT) s.append("back-right, "); 887 if (mask & AUDIO_CHANNEL_OUT_FRONT_LEFT_OF_CENTER) s.append("front-left-of-center, "); 888 if (mask & AUDIO_CHANNEL_OUT_FRONT_RIGHT_OF_CENTER) s.append("front-right-of-center, "); 889 if (mask & AUDIO_CHANNEL_OUT_BACK_CENTER) s.append("back-center, "); 890 if (mask & AUDIO_CHANNEL_OUT_SIDE_LEFT) s.append("side-left, "); 891 if (mask & AUDIO_CHANNEL_OUT_SIDE_RIGHT) s.append("side-right, "); 892 if (mask & AUDIO_CHANNEL_OUT_TOP_CENTER) s.append("top-center ,"); 893 if (mask & AUDIO_CHANNEL_OUT_TOP_FRONT_LEFT) s.append("top-front-left, "); 894 if (mask & AUDIO_CHANNEL_OUT_TOP_FRONT_CENTER) s.append("top-front-center, "); 895 if (mask & AUDIO_CHANNEL_OUT_TOP_FRONT_RIGHT) s.append("top-front-right, "); 896 if (mask & AUDIO_CHANNEL_OUT_TOP_BACK_LEFT) s.append("top-back-left, "); 897 if (mask & AUDIO_CHANNEL_OUT_TOP_BACK_CENTER) s.append("top-back-center, " ); 898 if (mask & AUDIO_CHANNEL_OUT_TOP_BACK_RIGHT) s.append("top-back-right, " ); 899 if (mask & ~AUDIO_CHANNEL_OUT_ALL) s.append("unknown, "); 900 } else { 901 if (mask & AUDIO_CHANNEL_IN_LEFT) s.append("left, "); 902 if (mask & AUDIO_CHANNEL_IN_RIGHT) s.append("right, "); 903 if (mask & AUDIO_CHANNEL_IN_FRONT) s.append("front, "); 904 if (mask & AUDIO_CHANNEL_IN_BACK) s.append("back, "); 905 if (mask & AUDIO_CHANNEL_IN_LEFT_PROCESSED) s.append("left-processed, "); 906 if (mask & AUDIO_CHANNEL_IN_RIGHT_PROCESSED) s.append("right-processed, "); 907 if (mask & AUDIO_CHANNEL_IN_FRONT_PROCESSED) s.append("front-processed, "); 908 if (mask & AUDIO_CHANNEL_IN_BACK_PROCESSED) s.append("back-processed, "); 909 if (mask & AUDIO_CHANNEL_IN_PRESSURE) s.append("pressure, "); 910 if (mask & AUDIO_CHANNEL_IN_X_AXIS) s.append("X, "); 911 if (mask & AUDIO_CHANNEL_IN_Y_AXIS) s.append("Y, "); 912 if (mask & AUDIO_CHANNEL_IN_Z_AXIS) s.append("Z, "); 913 if (mask & AUDIO_CHANNEL_IN_VOICE_UPLINK) s.append("voice-uplink, "); 914 if (mask & AUDIO_CHANNEL_IN_VOICE_DNLINK) s.append("voice-dnlink, "); 915 if (mask & ~AUDIO_CHANNEL_IN_ALL) s.append("unknown, "); 916 } 917 const int len = s.length(); 918 if (len > 2) { 919 (void) s.lockBuffer(len); // needed? 920 s.unlockBuffer(len - 2); // remove trailing ", " 921 } 922 return s; 923 } 924 case AUDIO_CHANNEL_REPRESENTATION_INDEX: 925 s.appendFormat("index mask, bits:%#x", audio_channel_mask_get_bits(mask)); 926 return s; 927 default: 928 s.appendFormat("unknown mask, representation:%d bits:%#x", 929 representation, audio_channel_mask_get_bits(mask)); 930 return s; 931 } 932} 933 934void AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args __unused) 935{ 936 const size_t SIZE = 256; 937 char buffer[SIZE]; 938 String8 result; 939 940 bool locked = AudioFlinger::dumpTryLock(mLock); 941 if (!locked) { 942 dprintf(fd, "thread %p may be deadlocked\n", this); 943 } 944 945 dprintf(fd, " Thread name: %s\n", mThreadName); 946 dprintf(fd, " I/O handle: %d\n", mId); 947 dprintf(fd, " TID: %d\n", getTid()); 948 dprintf(fd, " Standby: %s\n", mStandby ? "yes" : "no"); 949 dprintf(fd, " Sample rate: %u Hz\n", mSampleRate); 950 dprintf(fd, " HAL frame count: %zu\n", mFrameCount); 951 dprintf(fd, " HAL format: 0x%x (%s)\n", mHALFormat, formatToString(mHALFormat)); 952 dprintf(fd, " HAL buffer size: %zu bytes\n", mBufferSize); 953 dprintf(fd, " Channel count: %u\n", mChannelCount); 954 dprintf(fd, " Channel mask: 0x%08x (%s)\n", mChannelMask, 955 channelMaskToString(mChannelMask, mType != RECORD).string()); 956 dprintf(fd, " Processing format: 0x%x (%s)\n", mFormat, formatToString(mFormat)); 957 dprintf(fd, " Processing frame size: %zu bytes\n", mFrameSize); 958 dprintf(fd, " Pending config events:"); 959 size_t numConfig = mConfigEvents.size(); 960 if (numConfig) { 961 for (size_t i = 0; i < numConfig; i++) { 962 mConfigEvents[i]->dump(buffer, SIZE); 963 dprintf(fd, "\n %s", buffer); 964 } 965 dprintf(fd, "\n"); 966 } else { 967 dprintf(fd, " none\n"); 968 } 969 dprintf(fd, " Output device: %#x (%s)\n", mOutDevice, devicesToString(mOutDevice).string()); 970 dprintf(fd, " Input device: %#x (%s)\n", mInDevice, devicesToString(mInDevice).string()); 971 dprintf(fd, " Audio source: %d (%s)\n", mAudioSource, sourceToString(mAudioSource)); 972 973 if (locked) { 974 mLock.unlock(); 975 } 976} 977 978void AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args) 979{ 980 const size_t SIZE = 256; 981 char buffer[SIZE]; 982 String8 result; 983 984 size_t numEffectChains = mEffectChains.size(); 985 snprintf(buffer, SIZE, " %zu Effect Chains\n", numEffectChains); 986 write(fd, buffer, strlen(buffer)); 987 988 for (size_t i = 0; i < numEffectChains; ++i) { 989 sp<EffectChain> chain = mEffectChains[i]; 990 if (chain != 0) { 991 chain->dump(fd, args); 992 } 993 } 994} 995 996void AudioFlinger::ThreadBase::acquireWakeLock(int uid) 997{ 998 Mutex::Autolock _l(mLock); 999 acquireWakeLock_l(uid); 1000} 1001 1002String16 AudioFlinger::ThreadBase::getWakeLockTag() 1003{ 1004 switch (mType) { 1005 case MIXER: 1006 return String16("AudioMix"); 1007 case DIRECT: 1008 return String16("AudioDirectOut"); 1009 case DUPLICATING: 1010 return String16("AudioDup"); 1011 case RECORD: 1012 return String16("AudioIn"); 1013 case OFFLOAD: 1014 return String16("AudioOffload"); 1015 default: 1016 ALOG_ASSERT(false); 1017 return String16("AudioUnknown"); 1018 } 1019} 1020 1021void AudioFlinger::ThreadBase::acquireWakeLock_l(int uid) 1022{ 1023 getPowerManager_l(); 1024 if (mPowerManager != 0) { 1025 sp<IBinder> binder = new BBinder(); 1026 status_t status; 1027 if (uid >= 0) { 1028 status = mPowerManager->acquireWakeLockWithUid(POWERMANAGER_PARTIAL_WAKE_LOCK, 1029 binder, 1030 getWakeLockTag(), 1031 String16("audioserver"), 1032 uid, 1033 true /* FIXME force oneway contrary to .aidl */); 1034 } else { 1035 status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK, 1036 binder, 1037 getWakeLockTag(), 1038 String16("audioserver"), 1039 true /* FIXME force oneway contrary to .aidl */); 1040 } 1041 if (status == NO_ERROR) { 1042 mWakeLockToken = binder; 1043 } 1044 ALOGV("acquireWakeLock_l() %s status %d", mThreadName, status); 1045 } 1046 1047 if (!mNotifiedBatteryStart) { 1048 BatteryNotifier::getInstance().noteStartAudio(); 1049 mNotifiedBatteryStart = true; 1050 } 1051 gBoottime.acquire(mWakeLockToken); 1052 mTimestamp.mTimebaseOffset[ExtendedTimestamp::TIMEBASE_BOOTTIME] = 1053 gBoottime.getBoottimeOffset(); 1054} 1055 1056void AudioFlinger::ThreadBase::releaseWakeLock() 1057{ 1058 Mutex::Autolock _l(mLock); 1059 releaseWakeLock_l(); 1060} 1061 1062void AudioFlinger::ThreadBase::releaseWakeLock_l() 1063{ 1064 gBoottime.release(mWakeLockToken); 1065 if (mWakeLockToken != 0) { 1066 ALOGV("releaseWakeLock_l() %s", mThreadName); 1067 if (mPowerManager != 0) { 1068 mPowerManager->releaseWakeLock(mWakeLockToken, 0, 1069 true /* FIXME force oneway contrary to .aidl */); 1070 } 1071 mWakeLockToken.clear(); 1072 } 1073 1074 if (mNotifiedBatteryStart) { 1075 BatteryNotifier::getInstance().noteStopAudio(); 1076 mNotifiedBatteryStart = false; 1077 } 1078} 1079 1080void AudioFlinger::ThreadBase::updateWakeLockUids(const SortedVector<int> &uids) { 1081 Mutex::Autolock _l(mLock); 1082 updateWakeLockUids_l(uids); 1083} 1084 1085void AudioFlinger::ThreadBase::getPowerManager_l() { 1086 if (mSystemReady && mPowerManager == 0) { 1087 // use checkService() to avoid blocking if power service is not up yet 1088 sp<IBinder> binder = 1089 defaultServiceManager()->checkService(String16("power")); 1090 if (binder == 0) { 1091 ALOGW("Thread %s cannot connect to the power manager service", mThreadName); 1092 } else { 1093 mPowerManager = interface_cast<IPowerManager>(binder); 1094 binder->linkToDeath(mDeathRecipient); 1095 } 1096 } 1097} 1098 1099void AudioFlinger::ThreadBase::updateWakeLockUids_l(const SortedVector<int> &uids) { 1100 getPowerManager_l(); 1101 if (mWakeLockToken == NULL) { // token may be NULL if AudioFlinger::systemReady() not called. 1102 if (mSystemReady) { 1103 ALOGE("no wake lock to update, but system ready!"); 1104 } else { 1105 ALOGW("no wake lock to update, system not ready yet"); 1106 } 1107 return; 1108 } 1109 if (mPowerManager != 0) { 1110 sp<IBinder> binder = new BBinder(); 1111 status_t status; 1112 status = mPowerManager->updateWakeLockUids(mWakeLockToken, uids.size(), uids.array(), 1113 true /* FIXME force oneway contrary to .aidl */); 1114 ALOGV("updateWakeLockUids_l() %s status %d", mThreadName, status); 1115 } 1116} 1117 1118void AudioFlinger::ThreadBase::clearPowerManager() 1119{ 1120 Mutex::Autolock _l(mLock); 1121 releaseWakeLock_l(); 1122 mPowerManager.clear(); 1123} 1124 1125void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who __unused) 1126{ 1127 sp<ThreadBase> thread = mThread.promote(); 1128 if (thread != 0) { 1129 thread->clearPowerManager(); 1130 } 1131 ALOGW("power manager service died !!!"); 1132} 1133 1134void AudioFlinger::ThreadBase::setEffectSuspended( 1135 const effect_uuid_t *type, bool suspend, audio_session_t sessionId) 1136{ 1137 Mutex::Autolock _l(mLock); 1138 setEffectSuspended_l(type, suspend, sessionId); 1139} 1140 1141void AudioFlinger::ThreadBase::setEffectSuspended_l( 1142 const effect_uuid_t *type, bool suspend, audio_session_t sessionId) 1143{ 1144 sp<EffectChain> chain = getEffectChain_l(sessionId); 1145 if (chain != 0) { 1146 if (type != NULL) { 1147 chain->setEffectSuspended_l(type, suspend); 1148 } else { 1149 chain->setEffectSuspendedAll_l(suspend); 1150 } 1151 } 1152 1153 updateSuspendedSessions_l(type, suspend, sessionId); 1154} 1155 1156void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain) 1157{ 1158 ssize_t index = mSuspendedSessions.indexOfKey(chain->sessionId()); 1159 if (index < 0) { 1160 return; 1161 } 1162 1163 const KeyedVector <int, sp<SuspendedSessionDesc> >& sessionEffects = 1164 mSuspendedSessions.valueAt(index); 1165 1166 for (size_t i = 0; i < sessionEffects.size(); i++) { 1167 sp<SuspendedSessionDesc> desc = sessionEffects.valueAt(i); 1168 for (int j = 0; j < desc->mRefCount; j++) { 1169 if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) { 1170 chain->setEffectSuspendedAll_l(true); 1171 } else { 1172 ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x", 1173 desc->mType.timeLow); 1174 chain->setEffectSuspended_l(&desc->mType, true); 1175 } 1176 } 1177 } 1178} 1179 1180void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type, 1181 bool suspend, 1182 audio_session_t sessionId) 1183{ 1184 ssize_t index = mSuspendedSessions.indexOfKey(sessionId); 1185 1186 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects; 1187 1188 if (suspend) { 1189 if (index >= 0) { 1190 sessionEffects = mSuspendedSessions.valueAt(index); 1191 } else { 1192 mSuspendedSessions.add(sessionId, sessionEffects); 1193 } 1194 } else { 1195 if (index < 0) { 1196 return; 1197 } 1198 sessionEffects = mSuspendedSessions.valueAt(index); 1199 } 1200 1201 1202 int key = EffectChain::kKeyForSuspendAll; 1203 if (type != NULL) { 1204 key = type->timeLow; 1205 } 1206 index = sessionEffects.indexOfKey(key); 1207 1208 sp<SuspendedSessionDesc> desc; 1209 if (suspend) { 1210 if (index >= 0) { 1211 desc = sessionEffects.valueAt(index); 1212 } else { 1213 desc = new SuspendedSessionDesc(); 1214 if (type != NULL) { 1215 desc->mType = *type; 1216 } 1217 sessionEffects.add(key, desc); 1218 ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key); 1219 } 1220 desc->mRefCount++; 1221 } else { 1222 if (index < 0) { 1223 return; 1224 } 1225 desc = sessionEffects.valueAt(index); 1226 if (--desc->mRefCount == 0) { 1227 ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key); 1228 sessionEffects.removeItemsAt(index); 1229 if (sessionEffects.isEmpty()) { 1230 ALOGV("updateSuspendedSessions_l() restore removing session %d", 1231 sessionId); 1232 mSuspendedSessions.removeItem(sessionId); 1233 } 1234 } 1235 } 1236 if (!sessionEffects.isEmpty()) { 1237 mSuspendedSessions.replaceValueFor(sessionId, sessionEffects); 1238 } 1239} 1240 1241void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 1242 bool enabled, 1243 audio_session_t sessionId) 1244{ 1245 Mutex::Autolock _l(mLock); 1246 checkSuspendOnEffectEnabled_l(effect, enabled, sessionId); 1247} 1248 1249void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 1250 bool enabled, 1251 audio_session_t sessionId) 1252{ 1253 if (mType != RECORD) { 1254 // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on 1255 // another session. This gives the priority to well behaved effect control panels 1256 // and applications not using global effects. 1257 // Enabling post processing in AUDIO_SESSION_OUTPUT_STAGE session does not affect 1258 // global effects 1259 if ((sessionId != AUDIO_SESSION_OUTPUT_MIX) && (sessionId != AUDIO_SESSION_OUTPUT_STAGE)) { 1260 setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX); 1261 } 1262 } 1263 1264 sp<EffectChain> chain = getEffectChain_l(sessionId); 1265 if (chain != 0) { 1266 chain->checkSuspendOnEffectEnabled(effect, enabled); 1267 } 1268} 1269 1270// ThreadBase::createEffect_l() must be called with AudioFlinger::mLock held 1271sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 1272 const sp<AudioFlinger::Client>& client, 1273 const sp<IEffectClient>& effectClient, 1274 int32_t priority, 1275 audio_session_t sessionId, 1276 effect_descriptor_t *desc, 1277 int *enabled, 1278 status_t *status) 1279{ 1280 sp<EffectModule> effect; 1281 sp<EffectHandle> handle; 1282 status_t lStatus; 1283 sp<EffectChain> chain; 1284 bool chainCreated = false; 1285 bool effectCreated = false; 1286 bool effectRegistered = false; 1287 1288 lStatus = initCheck(); 1289 if (lStatus != NO_ERROR) { 1290 ALOGW("createEffect_l() Audio driver not initialized."); 1291 goto Exit; 1292 } 1293 1294 // Reject any effect on Direct output threads for now, since the format of 1295 // mSinkBuffer is not guaranteed to be compatible with effect processing (PCM 16 stereo). 1296 if (mType == DIRECT) { 1297 ALOGW("createEffect_l() Cannot add effect %s on Direct output type thread %s", 1298 desc->name, mThreadName); 1299 lStatus = BAD_VALUE; 1300 goto Exit; 1301 } 1302 1303 // Reject any effect on mixer or duplicating multichannel sinks. 1304 // TODO: fix both format and multichannel issues with effects. 1305 if ((mType == MIXER || mType == DUPLICATING) && mChannelCount != FCC_2) { 1306 ALOGW("createEffect_l() Cannot add effect %s for multichannel(%d) %s threads", 1307 desc->name, mChannelCount, mType == MIXER ? "MIXER" : "DUPLICATING"); 1308 lStatus = BAD_VALUE; 1309 goto Exit; 1310 } 1311 1312 // Allow global effects only on offloaded and mixer threads 1313 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 1314 switch (mType) { 1315 case MIXER: 1316 case OFFLOAD: 1317 break; 1318 case DIRECT: 1319 case DUPLICATING: 1320 case RECORD: 1321 default: 1322 ALOGW("createEffect_l() Cannot add global effect %s on thread %s", 1323 desc->name, mThreadName); 1324 lStatus = BAD_VALUE; 1325 goto Exit; 1326 } 1327 } 1328 1329 // Only Pre processor effects are allowed on input threads and only on input threads 1330 if ((mType == RECORD) != ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { 1331 ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d", 1332 desc->name, desc->flags, mType); 1333 lStatus = BAD_VALUE; 1334 goto Exit; 1335 } 1336 1337 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 1338 1339 { // scope for mLock 1340 Mutex::Autolock _l(mLock); 1341 1342 // check for existing effect chain with the requested audio session 1343 chain = getEffectChain_l(sessionId); 1344 if (chain == 0) { 1345 // create a new chain for this session 1346 ALOGV("createEffect_l() new effect chain for session %d", sessionId); 1347 chain = new EffectChain(this, sessionId); 1348 addEffectChain_l(chain); 1349 chain->setStrategy(getStrategyForSession_l(sessionId)); 1350 chainCreated = true; 1351 } else { 1352 effect = chain->getEffectFromDesc_l(desc); 1353 } 1354 1355 ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get()); 1356 1357 if (effect == 0) { 1358 audio_unique_id_t id = mAudioFlinger->nextUniqueId(AUDIO_UNIQUE_ID_USE_EFFECT); 1359 // Check CPU and memory usage 1360 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 1361 if (lStatus != NO_ERROR) { 1362 goto Exit; 1363 } 1364 effectRegistered = true; 1365 // create a new effect module if none present in the chain 1366 effect = new EffectModule(this, chain, desc, id, sessionId); 1367 lStatus = effect->status(); 1368 if (lStatus != NO_ERROR) { 1369 goto Exit; 1370 } 1371 effect->setOffloaded(mType == OFFLOAD, mId); 1372 1373 lStatus = chain->addEffect_l(effect); 1374 if (lStatus != NO_ERROR) { 1375 goto Exit; 1376 } 1377 effectCreated = true; 1378 1379 effect->setDevice(mOutDevice); 1380 effect->setDevice(mInDevice); 1381 effect->setMode(mAudioFlinger->getMode()); 1382 effect->setAudioSource(mAudioSource); 1383 } 1384 // create effect handle and connect it to effect module 1385 handle = new EffectHandle(effect, client, effectClient, priority); 1386 lStatus = handle->initCheck(); 1387 if (lStatus == OK) { 1388 lStatus = effect->addHandle(handle.get()); 1389 } 1390 if (enabled != NULL) { 1391 *enabled = (int)effect->isEnabled(); 1392 } 1393 } 1394 1395Exit: 1396 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 1397 Mutex::Autolock _l(mLock); 1398 if (effectCreated) { 1399 chain->removeEffect_l(effect); 1400 } 1401 if (effectRegistered) { 1402 AudioSystem::unregisterEffect(effect->id()); 1403 } 1404 if (chainCreated) { 1405 removeEffectChain_l(chain); 1406 } 1407 handle.clear(); 1408 } 1409 1410 *status = lStatus; 1411 return handle; 1412} 1413 1414sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect(audio_session_t sessionId, 1415 int effectId) 1416{ 1417 Mutex::Autolock _l(mLock); 1418 return getEffect_l(sessionId, effectId); 1419} 1420 1421sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(audio_session_t sessionId, 1422 int effectId) 1423{ 1424 sp<EffectChain> chain = getEffectChain_l(sessionId); 1425 return chain != 0 ? chain->getEffectFromId_l(effectId) : 0; 1426} 1427 1428// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 1429// PlaybackThread::mLock held 1430status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 1431{ 1432 // check for existing effect chain with the requested audio session 1433 audio_session_t sessionId = effect->sessionId(); 1434 sp<EffectChain> chain = getEffectChain_l(sessionId); 1435 bool chainCreated = false; 1436 1437 ALOGD_IF((mType == OFFLOAD) && !effect->isOffloadable(), 1438 "addEffect_l() on offloaded thread %p: effect %s does not support offload flags %x", 1439 this, effect->desc().name, effect->desc().flags); 1440 1441 if (chain == 0) { 1442 // create a new chain for this session 1443 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 1444 chain = new EffectChain(this, sessionId); 1445 addEffectChain_l(chain); 1446 chain->setStrategy(getStrategyForSession_l(sessionId)); 1447 chainCreated = true; 1448 } 1449 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 1450 1451 if (chain->getEffectFromId_l(effect->id()) != 0) { 1452 ALOGW("addEffect_l() %p effect %s already present in chain %p", 1453 this, effect->desc().name, chain.get()); 1454 return BAD_VALUE; 1455 } 1456 1457 effect->setOffloaded(mType == OFFLOAD, mId); 1458 1459 status_t status = chain->addEffect_l(effect); 1460 if (status != NO_ERROR) { 1461 if (chainCreated) { 1462 removeEffectChain_l(chain); 1463 } 1464 return status; 1465 } 1466 1467 effect->setDevice(mOutDevice); 1468 effect->setDevice(mInDevice); 1469 effect->setMode(mAudioFlinger->getMode()); 1470 effect->setAudioSource(mAudioSource); 1471 return NO_ERROR; 1472} 1473 1474void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 1475 1476 ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 1477 effect_descriptor_t desc = effect->desc(); 1478 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 1479 detachAuxEffect_l(effect->id()); 1480 } 1481 1482 sp<EffectChain> chain = effect->chain().promote(); 1483 if (chain != 0) { 1484 // remove effect chain if removing last effect 1485 if (chain->removeEffect_l(effect) == 0) { 1486 removeEffectChain_l(chain); 1487 } 1488 } else { 1489 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 1490 } 1491} 1492 1493void AudioFlinger::ThreadBase::lockEffectChains_l( 1494 Vector< sp<AudioFlinger::EffectChain> >& effectChains) 1495{ 1496 effectChains = mEffectChains; 1497 for (size_t i = 0; i < mEffectChains.size(); i++) { 1498 mEffectChains[i]->lock(); 1499 } 1500} 1501 1502void AudioFlinger::ThreadBase::unlockEffectChains( 1503 const Vector< sp<AudioFlinger::EffectChain> >& effectChains) 1504{ 1505 for (size_t i = 0; i < effectChains.size(); i++) { 1506 effectChains[i]->unlock(); 1507 } 1508} 1509 1510sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(audio_session_t sessionId) 1511{ 1512 Mutex::Autolock _l(mLock); 1513 return getEffectChain_l(sessionId); 1514} 1515 1516sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(audio_session_t sessionId) 1517 const 1518{ 1519 size_t size = mEffectChains.size(); 1520 for (size_t i = 0; i < size; i++) { 1521 if (mEffectChains[i]->sessionId() == sessionId) { 1522 return mEffectChains[i]; 1523 } 1524 } 1525 return 0; 1526} 1527 1528void AudioFlinger::ThreadBase::setMode(audio_mode_t mode) 1529{ 1530 Mutex::Autolock _l(mLock); 1531 size_t size = mEffectChains.size(); 1532 for (size_t i = 0; i < size; i++) { 1533 mEffectChains[i]->setMode_l(mode); 1534 } 1535} 1536 1537void AudioFlinger::ThreadBase::getAudioPortConfig(struct audio_port_config *config) 1538{ 1539 config->type = AUDIO_PORT_TYPE_MIX; 1540 config->ext.mix.handle = mId; 1541 config->sample_rate = mSampleRate; 1542 config->format = mFormat; 1543 config->channel_mask = mChannelMask; 1544 config->config_mask = AUDIO_PORT_CONFIG_SAMPLE_RATE|AUDIO_PORT_CONFIG_CHANNEL_MASK| 1545 AUDIO_PORT_CONFIG_FORMAT; 1546} 1547 1548void AudioFlinger::ThreadBase::systemReady() 1549{ 1550 Mutex::Autolock _l(mLock); 1551 if (mSystemReady) { 1552 return; 1553 } 1554 mSystemReady = true; 1555 1556 for (size_t i = 0; i < mPendingConfigEvents.size(); i++) { 1557 sendConfigEvent_l(mPendingConfigEvents.editItemAt(i)); 1558 } 1559 mPendingConfigEvents.clear(); 1560} 1561 1562 1563// ---------------------------------------------------------------------------- 1564// Playback 1565// ---------------------------------------------------------------------------- 1566 1567AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, 1568 AudioStreamOut* output, 1569 audio_io_handle_t id, 1570 audio_devices_t device, 1571 type_t type, 1572 bool systemReady, 1573 uint32_t bitRate) 1574 : ThreadBase(audioFlinger, id, device, AUDIO_DEVICE_NONE, type, systemReady), 1575 mNormalFrameCount(0), mSinkBuffer(NULL), 1576 mMixerBufferEnabled(AudioFlinger::kEnableExtendedPrecision), 1577 mMixerBuffer(NULL), 1578 mMixerBufferSize(0), 1579 mMixerBufferFormat(AUDIO_FORMAT_INVALID), 1580 mMixerBufferValid(false), 1581 mEffectBufferEnabled(AudioFlinger::kEnableExtendedPrecision), 1582 mEffectBuffer(NULL), 1583 mEffectBufferSize(0), 1584 mEffectBufferFormat(AUDIO_FORMAT_INVALID), 1585 mEffectBufferValid(false), 1586 mSuspended(0), mBytesWritten(0), 1587 mFramesWritten(0), 1588 mActiveTracksGeneration(0), 1589 // mStreamTypes[] initialized in constructor body 1590 mOutput(output), 1591 mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false), 1592 mMixerStatus(MIXER_IDLE), 1593 mMixerStatusIgnoringFastTracks(MIXER_IDLE), 1594 mStandbyDelayNs(AudioFlinger::mStandbyTimeInNsecs), 1595 mBytesRemaining(0), 1596 mCurrentWriteLength(0), 1597 mUseAsyncWrite(false), 1598 mWriteAckSequence(0), 1599 mDrainSequence(0), 1600 mSignalPending(false), 1601 mScreenState(AudioFlinger::mScreenState), 1602 // index 0 is reserved for normal mixer's submix 1603 mFastTrackAvailMask(((1 << FastMixerState::kMaxFastTracks) - 1) & ~1), 1604 mHwSupportsPause(false), mHwPaused(false), mFlushPending(false) 1605{ 1606 snprintf(mThreadName, kThreadNameLength, "AudioOut_%X", id); 1607 mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mThreadName); 1608 1609 // Assumes constructor is called by AudioFlinger with it's mLock held, but 1610 // it would be safer to explicitly pass initial masterVolume/masterMute as 1611 // parameter. 1612 // 1613 // If the HAL we are using has support for master volume or master mute, 1614 // then do not attenuate or mute during mixing (just leave the volume at 1.0 1615 // and the mute set to false). 1616 mMasterVolume = audioFlinger->masterVolume_l(); 1617 mMasterMute = audioFlinger->masterMute_l(); 1618 if (mOutput && mOutput->audioHwDev) { 1619 if (mOutput->audioHwDev->canSetMasterVolume()) { 1620 mMasterVolume = 1.0; 1621 } 1622 1623 if (mOutput->audioHwDev->canSetMasterMute()) { 1624 mMasterMute = false; 1625 } 1626 } 1627 1628 readOutputParameters_l(); 1629 1630 // ++ operator does not compile 1631 for (audio_stream_type_t stream = AUDIO_STREAM_MIN; stream < AUDIO_STREAM_CNT; 1632 stream = (audio_stream_type_t) (stream + 1)) { 1633 mStreamTypes[stream].volume = mAudioFlinger->streamVolume_l(stream); 1634 mStreamTypes[stream].mute = mAudioFlinger->streamMute_l(stream); 1635 } 1636 1637 if (audio_has_proportional_frames(mFormat)) { 1638 mBufferDurationUs = (uint32_t)((mNormalFrameCount * 1000000LL) / mSampleRate); 1639 } else { 1640 bitRate = bitRate != 0 ? bitRate : kOffloadDefaultBitRateBps; 1641 mBufferDurationUs = (uint32_t)((mBufferSize * 8 * 1000000LL) / bitRate); 1642 } 1643} 1644 1645AudioFlinger::PlaybackThread::~PlaybackThread() 1646{ 1647 mAudioFlinger->unregisterWriter(mNBLogWriter); 1648 free(mSinkBuffer); 1649 free(mMixerBuffer); 1650 free(mEffectBuffer); 1651} 1652 1653void AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) 1654{ 1655 dumpInternals(fd, args); 1656 dumpTracks(fd, args); 1657 dumpEffectChains(fd, args); 1658} 1659 1660void AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args __unused) 1661{ 1662 const size_t SIZE = 256; 1663 char buffer[SIZE]; 1664 String8 result; 1665 1666 result.appendFormat(" Stream volumes in dB: "); 1667 for (int i = 0; i < AUDIO_STREAM_CNT; ++i) { 1668 const stream_type_t *st = &mStreamTypes[i]; 1669 if (i > 0) { 1670 result.appendFormat(", "); 1671 } 1672 result.appendFormat("%d:%.2g", i, 20.0 * log10(st->volume)); 1673 if (st->mute) { 1674 result.append("M"); 1675 } 1676 } 1677 result.append("\n"); 1678 write(fd, result.string(), result.length()); 1679 result.clear(); 1680 1681 // These values are "raw"; they will wrap around. See prepareTracks_l() for a better way. 1682 FastTrackUnderruns underruns = getFastTrackUnderruns(0); 1683 dprintf(fd, " Normal mixer raw underrun counters: partial=%u empty=%u\n", 1684 underruns.mBitFields.mPartial, underruns.mBitFields.mEmpty); 1685 1686 size_t numtracks = mTracks.size(); 1687 size_t numactive = mActiveTracks.size(); 1688 dprintf(fd, " %zu Tracks", numtracks); 1689 size_t numactiveseen = 0; 1690 if (numtracks) { 1691 dprintf(fd, " of which %zu are active\n", numactive); 1692 Track::appendDumpHeader(result); 1693 for (size_t i = 0; i < numtracks; ++i) { 1694 sp<Track> track = mTracks[i]; 1695 if (track != 0) { 1696 bool active = mActiveTracks.indexOf(track) >= 0; 1697 if (active) { 1698 numactiveseen++; 1699 } 1700 track->dump(buffer, SIZE, active); 1701 result.append(buffer); 1702 } 1703 } 1704 } else { 1705 result.append("\n"); 1706 } 1707 if (numactiveseen != numactive) { 1708 // some tracks in the active list were not in the tracks list 1709 snprintf(buffer, SIZE, " The following tracks are in the active list but" 1710 " not in the track list\n"); 1711 result.append(buffer); 1712 Track::appendDumpHeader(result); 1713 for (size_t i = 0; i < numactive; ++i) { 1714 sp<Track> track = mActiveTracks[i].promote(); 1715 if (track != 0 && mTracks.indexOf(track) < 0) { 1716 track->dump(buffer, SIZE, true); 1717 result.append(buffer); 1718 } 1719 } 1720 } 1721 1722 write(fd, result.string(), result.size()); 1723} 1724 1725void AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 1726{ 1727 dprintf(fd, "\nOutput thread %p type %d (%s):\n", this, type(), threadTypeToString(type())); 1728 1729 dumpBase(fd, args); 1730 1731 dprintf(fd, " Normal frame count: %zu\n", mNormalFrameCount); 1732 dprintf(fd, " Last write occurred (msecs): %llu\n", 1733 (unsigned long long) ns2ms(systemTime() - mLastWriteTime)); 1734 dprintf(fd, " Total writes: %d\n", mNumWrites); 1735 dprintf(fd, " Delayed writes: %d\n", mNumDelayedWrites); 1736 dprintf(fd, " Blocked in write: %s\n", mInWrite ? "yes" : "no"); 1737 dprintf(fd, " Suspend count: %d\n", mSuspended); 1738 dprintf(fd, " Sink buffer : %p\n", mSinkBuffer); 1739 dprintf(fd, " Mixer buffer: %p\n", mMixerBuffer); 1740 dprintf(fd, " Effect buffer: %p\n", mEffectBuffer); 1741 dprintf(fd, " Fast track availMask=%#x\n", mFastTrackAvailMask); 1742 dprintf(fd, " Standby delay ns=%lld\n", (long long)mStandbyDelayNs); 1743 AudioStreamOut *output = mOutput; 1744 audio_output_flags_t flags = output != NULL ? output->flags : AUDIO_OUTPUT_FLAG_NONE; 1745 String8 flagsAsString = outputFlagsToString(flags); 1746 dprintf(fd, " AudioStreamOut: %p flags %#x (%s)\n", output, flags, flagsAsString.string()); 1747} 1748 1749// Thread virtuals 1750 1751void AudioFlinger::PlaybackThread::onFirstRef() 1752{ 1753 run(mThreadName, ANDROID_PRIORITY_URGENT_AUDIO); 1754} 1755 1756// ThreadBase virtuals 1757void AudioFlinger::PlaybackThread::preExit() 1758{ 1759 ALOGV(" preExit()"); 1760 // FIXME this is using hard-coded strings but in the future, this functionality will be 1761 // converted to use audio HAL extensions required to support tunneling 1762 mOutput->stream->common.set_parameters(&mOutput->stream->common, "exiting=1"); 1763} 1764 1765// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 1766sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 1767 const sp<AudioFlinger::Client>& client, 1768 audio_stream_type_t streamType, 1769 uint32_t sampleRate, 1770 audio_format_t format, 1771 audio_channel_mask_t channelMask, 1772 size_t *pFrameCount, 1773 const sp<IMemory>& sharedBuffer, 1774 audio_session_t sessionId, 1775 IAudioFlinger::track_flags_t *flags, 1776 pid_t tid, 1777 int uid, 1778 status_t *status) 1779{ 1780 size_t frameCount = *pFrameCount; 1781 sp<Track> track; 1782 status_t lStatus; 1783 1784 // client expresses a preference for FAST, but we get the final say 1785 if (*flags & IAudioFlinger::TRACK_FAST) { 1786 if ( 1787 // PCM data 1788 audio_is_linear_pcm(format) && 1789 // TODO: extract as a data library function that checks that a computationally 1790 // expensive downmixer is not required: isFastOutputChannelConversion() 1791 (channelMask == mChannelMask || 1792 mChannelMask != AUDIO_CHANNEL_OUT_STEREO || 1793 (channelMask == AUDIO_CHANNEL_OUT_MONO 1794 /* && mChannelMask == AUDIO_CHANNEL_OUT_STEREO */)) && 1795 // hardware sample rate 1796 (sampleRate == mSampleRate) && 1797 // normal mixer has an associated fast mixer 1798 hasFastMixer() && 1799 // there are sufficient fast track slots available 1800 (mFastTrackAvailMask != 0) 1801 // FIXME test that MixerThread for this fast track has a capable output HAL 1802 // FIXME add a permission test also? 1803 ) { 1804 // static tracks can have any nonzero framecount, streaming tracks check against minimum. 1805 if (sharedBuffer == 0) { 1806 // read the fast track multiplier property the first time it is needed 1807 int ok = pthread_once(&sFastTrackMultiplierOnce, sFastTrackMultiplierInit); 1808 if (ok != 0) { 1809 ALOGE("%s pthread_once failed: %d", __func__, ok); 1810 } 1811 frameCount = max(frameCount, mFrameCount * sFastTrackMultiplier); // incl framecount 0 1812 } 1813 ALOGV("AUDIO_OUTPUT_FLAG_FAST accepted: frameCount=%zu mFrameCount=%zu", 1814 frameCount, mFrameCount); 1815 } else { 1816 ALOGV("AUDIO_OUTPUT_FLAG_FAST denied: sharedBuffer=%p frameCount=%zu " 1817 "mFrameCount=%zu format=%#x mFormat=%#x isLinear=%d channelMask=%#x " 1818 "sampleRate=%u mSampleRate=%u " 1819 "hasFastMixer=%d tid=%d fastTrackAvailMask=%#x", 1820 sharedBuffer.get(), frameCount, mFrameCount, format, mFormat, 1821 audio_is_linear_pcm(format), 1822 channelMask, sampleRate, mSampleRate, hasFastMixer(), tid, mFastTrackAvailMask); 1823 *flags &= ~IAudioFlinger::TRACK_FAST; 1824 } 1825 } 1826 // For normal PCM streaming tracks, update minimum frame count. 1827 // For compatibility with AudioTrack calculation, buffer depth is forced 1828 // to be at least 2 x the normal mixer frame count and cover audio hardware latency. 1829 // This is probably too conservative, but legacy application code may depend on it. 1830 // If you change this calculation, also review the start threshold which is related. 1831 if (!(*flags & IAudioFlinger::TRACK_FAST) 1832 && audio_has_proportional_frames(format) && sharedBuffer == 0) { 1833 // this must match AudioTrack.cpp calculateMinFrameCount(). 1834 // TODO: Move to a common library 1835 uint32_t latencyMs = mOutput->stream->get_latency(mOutput->stream); 1836 uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate); 1837 if (minBufCount < 2) { 1838 minBufCount = 2; 1839 } 1840 // For normal mixing tracks, if speed is > 1.0f (normal), AudioTrack 1841 // or the client should compute and pass in a larger buffer request. 1842 size_t minFrameCount = 1843 minBufCount * sourceFramesNeededWithTimestretch( 1844 sampleRate, mNormalFrameCount, 1845 mSampleRate, AUDIO_TIMESTRETCH_SPEED_NORMAL /*speed*/); 1846 if (frameCount < minFrameCount) { // including frameCount == 0 1847 frameCount = minFrameCount; 1848 } 1849 } 1850 *pFrameCount = frameCount; 1851 1852 switch (mType) { 1853 1854 case DIRECT: 1855 if (audio_is_linear_pcm(format)) { // TODO maybe use audio_has_proportional_frames()? 1856 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1857 ALOGE("createTrack_l() Bad parameter: sampleRate %u format %#x, channelMask 0x%08x " 1858 "for output %p with format %#x", 1859 sampleRate, format, channelMask, mOutput, mFormat); 1860 lStatus = BAD_VALUE; 1861 goto Exit; 1862 } 1863 } 1864 break; 1865 1866 case OFFLOAD: 1867 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1868 ALOGE("createTrack_l() Bad parameter: sampleRate %d format %#x, channelMask 0x%08x \"" 1869 "for output %p with format %#x", 1870 sampleRate, format, channelMask, mOutput, mFormat); 1871 lStatus = BAD_VALUE; 1872 goto Exit; 1873 } 1874 break; 1875 1876 default: 1877 if (!audio_is_linear_pcm(format)) { 1878 ALOGE("createTrack_l() Bad parameter: format %#x \"" 1879 "for output %p with format %#x", 1880 format, mOutput, mFormat); 1881 lStatus = BAD_VALUE; 1882 goto Exit; 1883 } 1884 if (sampleRate > mSampleRate * AUDIO_RESAMPLER_DOWN_RATIO_MAX) { 1885 ALOGE("Sample rate out of range: %u mSampleRate %u", sampleRate, mSampleRate); 1886 lStatus = BAD_VALUE; 1887 goto Exit; 1888 } 1889 break; 1890 1891 } 1892 1893 lStatus = initCheck(); 1894 if (lStatus != NO_ERROR) { 1895 ALOGE("createTrack_l() audio driver not initialized"); 1896 goto Exit; 1897 } 1898 1899 { // scope for mLock 1900 Mutex::Autolock _l(mLock); 1901 1902 // all tracks in same audio session must share the same routing strategy otherwise 1903 // conflicts will happen when tracks are moved from one output to another by audio policy 1904 // manager 1905 uint32_t strategy = AudioSystem::getStrategyForStream(streamType); 1906 for (size_t i = 0; i < mTracks.size(); ++i) { 1907 sp<Track> t = mTracks[i]; 1908 if (t != 0 && t->isExternalTrack()) { 1909 uint32_t actual = AudioSystem::getStrategyForStream(t->streamType()); 1910 if (sessionId == t->sessionId() && strategy != actual) { 1911 ALOGE("createTrack_l() mismatched strategy; expected %u but found %u", 1912 strategy, actual); 1913 lStatus = BAD_VALUE; 1914 goto Exit; 1915 } 1916 } 1917 } 1918 1919 track = new Track(this, client, streamType, sampleRate, format, 1920 channelMask, frameCount, NULL, sharedBuffer, 1921 sessionId, uid, *flags, TrackBase::TYPE_DEFAULT); 1922 1923 lStatus = track != 0 ? track->initCheck() : (status_t) NO_MEMORY; 1924 if (lStatus != NO_ERROR) { 1925 ALOGE("createTrack_l() initCheck failed %d; no control block?", lStatus); 1926 // track must be cleared from the caller as the caller has the AF lock 1927 goto Exit; 1928 } 1929 mTracks.add(track); 1930 1931 sp<EffectChain> chain = getEffectChain_l(sessionId); 1932 if (chain != 0) { 1933 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1934 track->setMainBuffer(chain->inBuffer()); 1935 chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType())); 1936 chain->incTrackCnt(); 1937 } 1938 1939 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 1940 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 1941 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 1942 // so ask activity manager to do this on our behalf 1943 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 1944 } 1945 } 1946 1947 lStatus = NO_ERROR; 1948 1949Exit: 1950 *status = lStatus; 1951 return track; 1952} 1953 1954uint32_t AudioFlinger::PlaybackThread::correctLatency_l(uint32_t latency) const 1955{ 1956 return latency; 1957} 1958 1959uint32_t AudioFlinger::PlaybackThread::latency() const 1960{ 1961 Mutex::Autolock _l(mLock); 1962 return latency_l(); 1963} 1964uint32_t AudioFlinger::PlaybackThread::latency_l() const 1965{ 1966 if (initCheck() == NO_ERROR) { 1967 return correctLatency_l(mOutput->stream->get_latency(mOutput->stream)); 1968 } else { 1969 return 0; 1970 } 1971} 1972 1973void AudioFlinger::PlaybackThread::setMasterVolume(float value) 1974{ 1975 Mutex::Autolock _l(mLock); 1976 // Don't apply master volume in SW if our HAL can do it for us. 1977 if (mOutput && mOutput->audioHwDev && 1978 mOutput->audioHwDev->canSetMasterVolume()) { 1979 mMasterVolume = 1.0; 1980 } else { 1981 mMasterVolume = value; 1982 } 1983} 1984 1985void AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1986{ 1987 Mutex::Autolock _l(mLock); 1988 // Don't apply master mute in SW if our HAL can do it for us. 1989 if (mOutput && mOutput->audioHwDev && 1990 mOutput->audioHwDev->canSetMasterMute()) { 1991 mMasterMute = false; 1992 } else { 1993 mMasterMute = muted; 1994 } 1995} 1996 1997void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) 1998{ 1999 Mutex::Autolock _l(mLock); 2000 mStreamTypes[stream].volume = value; 2001 broadcast_l(); 2002} 2003 2004void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) 2005{ 2006 Mutex::Autolock _l(mLock); 2007 mStreamTypes[stream].mute = muted; 2008 broadcast_l(); 2009} 2010 2011float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const 2012{ 2013 Mutex::Autolock _l(mLock); 2014 return mStreamTypes[stream].volume; 2015} 2016 2017// addTrack_l() must be called with ThreadBase::mLock held 2018status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 2019{ 2020 status_t status = ALREADY_EXISTS; 2021 2022 if (mActiveTracks.indexOf(track) < 0) { 2023 // the track is newly added, make sure it fills up all its 2024 // buffers before playing. This is to ensure the client will 2025 // effectively get the latency it requested. 2026 if (track->isExternalTrack()) { 2027 TrackBase::track_state state = track->mState; 2028 mLock.unlock(); 2029 status = AudioSystem::startOutput(mId, track->streamType(), 2030 track->sessionId()); 2031 mLock.lock(); 2032 // abort track was stopped/paused while we released the lock 2033 if (state != track->mState) { 2034 if (status == NO_ERROR) { 2035 mLock.unlock(); 2036 AudioSystem::stopOutput(mId, track->streamType(), 2037 track->sessionId()); 2038 mLock.lock(); 2039 } 2040 return INVALID_OPERATION; 2041 } 2042 // abort if start is rejected by audio policy manager 2043 if (status != NO_ERROR) { 2044 return PERMISSION_DENIED; 2045 } 2046#ifdef ADD_BATTERY_DATA 2047 // to track the speaker usage 2048 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 2049#endif 2050 } 2051 2052 // set retry count for buffer fill 2053 if (track->isOffloaded()) { 2054 track->mRetryCount = kMaxTrackStartupRetriesOffload; 2055 } else { 2056 track->mRetryCount = kMaxTrackStartupRetries; 2057 } 2058 2059 track->mFillingUpStatus = track->sharedBuffer() != 0 ? Track::FS_FILLED : Track::FS_FILLING; 2060 track->mResetDone = false; 2061 track->mPresentationCompleteFrames = 0; 2062 mActiveTracks.add(track); 2063 mWakeLockUids.add(track->uid()); 2064 mActiveTracksGeneration++; 2065 mLatestActiveTrack = track; 2066 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 2067 if (chain != 0) { 2068 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), 2069 track->sessionId()); 2070 chain->incActiveTrackCnt(); 2071 } 2072 2073 status = NO_ERROR; 2074 } 2075 2076 onAddNewTrack_l(); 2077 return status; 2078} 2079 2080bool AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 2081{ 2082 track->terminate(); 2083 // active tracks are removed by threadLoop() 2084 bool trackActive = (mActiveTracks.indexOf(track) >= 0); 2085 track->mState = TrackBase::STOPPED; 2086 if (!trackActive) { 2087 removeTrack_l(track); 2088 } else if (track->isFastTrack() || track->isOffloaded() || track->isDirect()) { 2089 track->mState = TrackBase::STOPPING_1; 2090 } 2091 2092 return trackActive; 2093} 2094 2095void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 2096{ 2097 track->triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 2098 mTracks.remove(track); 2099 deleteTrackName_l(track->name()); 2100 // redundant as track is about to be destroyed, for dumpsys only 2101 track->mName = -1; 2102 if (track->isFastTrack()) { 2103 int index = track->mFastIndex; 2104 ALOG_ASSERT(0 < index && index < (int)FastMixerState::kMaxFastTracks); 2105 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << index))); 2106 mFastTrackAvailMask |= 1 << index; 2107 // redundant as track is about to be destroyed, for dumpsys only 2108 track->mFastIndex = -1; 2109 } 2110 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 2111 if (chain != 0) { 2112 chain->decTrackCnt(); 2113 } 2114} 2115 2116void AudioFlinger::PlaybackThread::broadcast_l() 2117{ 2118 // Thread could be blocked waiting for async 2119 // so signal it to handle state changes immediately 2120 // If threadLoop is currently unlocked a signal of mWaitWorkCV will 2121 // be lost so we also flag to prevent it blocking on mWaitWorkCV 2122 mSignalPending = true; 2123 mWaitWorkCV.broadcast(); 2124} 2125 2126String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 2127{ 2128 Mutex::Autolock _l(mLock); 2129 if (initCheck() != NO_ERROR) { 2130 return String8(); 2131 } 2132 2133 char *s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 2134 const String8 out_s8(s); 2135 free(s); 2136 return out_s8; 2137} 2138 2139void AudioFlinger::PlaybackThread::ioConfigChanged(audio_io_config_event event, pid_t pid) { 2140 sp<AudioIoDescriptor> desc = new AudioIoDescriptor(); 2141 ALOGV("PlaybackThread::ioConfigChanged, thread %p, event %d", this, event); 2142 2143 desc->mIoHandle = mId; 2144 2145 switch (event) { 2146 case AUDIO_OUTPUT_OPENED: 2147 case AUDIO_OUTPUT_CONFIG_CHANGED: 2148 desc->mPatch = mPatch; 2149 desc->mChannelMask = mChannelMask; 2150 desc->mSamplingRate = mSampleRate; 2151 desc->mFormat = mFormat; 2152 desc->mFrameCount = mNormalFrameCount; // FIXME see 2153 // AudioFlinger::frameCount(audio_io_handle_t) 2154 desc->mLatency = latency_l(); 2155 break; 2156 2157 case AUDIO_OUTPUT_CLOSED: 2158 default: 2159 break; 2160 } 2161 mAudioFlinger->ioConfigChanged(event, desc, pid); 2162} 2163 2164void AudioFlinger::PlaybackThread::writeCallback() 2165{ 2166 ALOG_ASSERT(mCallbackThread != 0); 2167 mCallbackThread->resetWriteBlocked(); 2168} 2169 2170void AudioFlinger::PlaybackThread::drainCallback() 2171{ 2172 ALOG_ASSERT(mCallbackThread != 0); 2173 mCallbackThread->resetDraining(); 2174} 2175 2176void AudioFlinger::PlaybackThread::resetWriteBlocked(uint32_t sequence) 2177{ 2178 Mutex::Autolock _l(mLock); 2179 // reject out of sequence requests 2180 if ((mWriteAckSequence & 1) && (sequence == mWriteAckSequence)) { 2181 mWriteAckSequence &= ~1; 2182 mWaitWorkCV.signal(); 2183 } 2184} 2185 2186void AudioFlinger::PlaybackThread::resetDraining(uint32_t sequence) 2187{ 2188 Mutex::Autolock _l(mLock); 2189 // reject out of sequence requests 2190 if ((mDrainSequence & 1) && (sequence == mDrainSequence)) { 2191 mDrainSequence &= ~1; 2192 mWaitWorkCV.signal(); 2193 } 2194} 2195 2196// static 2197int AudioFlinger::PlaybackThread::asyncCallback(stream_callback_event_t event, 2198 void *param __unused, 2199 void *cookie) 2200{ 2201 AudioFlinger::PlaybackThread *me = (AudioFlinger::PlaybackThread *)cookie; 2202 ALOGV("asyncCallback() event %d", event); 2203 switch (event) { 2204 case STREAM_CBK_EVENT_WRITE_READY: 2205 me->writeCallback(); 2206 break; 2207 case STREAM_CBK_EVENT_DRAIN_READY: 2208 me->drainCallback(); 2209 break; 2210 default: 2211 ALOGW("asyncCallback() unknown event %d", event); 2212 break; 2213 } 2214 return 0; 2215} 2216 2217void AudioFlinger::PlaybackThread::readOutputParameters_l() 2218{ 2219 // unfortunately we have no way of recovering from errors here, hence the LOG_ALWAYS_FATAL 2220 mSampleRate = mOutput->getSampleRate(); 2221 mChannelMask = mOutput->getChannelMask(); 2222 if (!audio_is_output_channel(mChannelMask)) { 2223 LOG_ALWAYS_FATAL("HAL channel mask %#x not valid for output", mChannelMask); 2224 } 2225 if ((mType == MIXER || mType == DUPLICATING) 2226 && !isValidPcmSinkChannelMask(mChannelMask)) { 2227 LOG_ALWAYS_FATAL("HAL channel mask %#x not supported for mixed output", 2228 mChannelMask); 2229 } 2230 mChannelCount = audio_channel_count_from_out_mask(mChannelMask); 2231 2232 // Get actual HAL format. 2233 mHALFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 2234 // Get format from the shim, which will be different than the HAL format 2235 // if playing compressed audio over HDMI passthrough. 2236 mFormat = mOutput->getFormat(); 2237 if (!audio_is_valid_format(mFormat)) { 2238 LOG_ALWAYS_FATAL("HAL format %#x not valid for output", mFormat); 2239 } 2240 if ((mType == MIXER || mType == DUPLICATING) 2241 && !isValidPcmSinkFormat(mFormat)) { 2242 LOG_FATAL("HAL format %#x not supported for mixed output", 2243 mFormat); 2244 } 2245 mFrameSize = mOutput->getFrameSize(); 2246 mBufferSize = mOutput->stream->common.get_buffer_size(&mOutput->stream->common); 2247 mFrameCount = mBufferSize / mFrameSize; 2248 if (mFrameCount & 15) { 2249 ALOGW("HAL output buffer size is %zu frames but AudioMixer requires multiples of 16 frames", 2250 mFrameCount); 2251 } 2252 2253 if ((mOutput->flags & AUDIO_OUTPUT_FLAG_NON_BLOCKING) && 2254 (mOutput->stream->set_callback != NULL)) { 2255 if (mOutput->stream->set_callback(mOutput->stream, 2256 AudioFlinger::PlaybackThread::asyncCallback, this) == 0) { 2257 mUseAsyncWrite = true; 2258 mCallbackThread = new AudioFlinger::AsyncCallbackThread(this); 2259 } 2260 } 2261 2262 mHwSupportsPause = false; 2263 if (mOutput->flags & AUDIO_OUTPUT_FLAG_DIRECT) { 2264 if (mOutput->stream->pause != NULL) { 2265 if (mOutput->stream->resume != NULL) { 2266 mHwSupportsPause = true; 2267 } else { 2268 ALOGW("direct output implements pause but not resume"); 2269 } 2270 } else if (mOutput->stream->resume != NULL) { 2271 ALOGW("direct output implements resume but not pause"); 2272 } 2273 } 2274 if (!mHwSupportsPause && mOutput->flags & AUDIO_OUTPUT_FLAG_HW_AV_SYNC) { 2275 LOG_ALWAYS_FATAL("HW_AV_SYNC requested but HAL does not implement pause and resume"); 2276 } 2277 2278 if (mType == DUPLICATING && mMixerBufferEnabled && mEffectBufferEnabled) { 2279 // For best precision, we use float instead of the associated output 2280 // device format (typically PCM 16 bit). 2281 2282 mFormat = AUDIO_FORMAT_PCM_FLOAT; 2283 mFrameSize = mChannelCount * audio_bytes_per_sample(mFormat); 2284 mBufferSize = mFrameSize * mFrameCount; 2285 2286 // TODO: We currently use the associated output device channel mask and sample rate. 2287 // (1) Perhaps use the ORed channel mask of all downstream MixerThreads 2288 // (if a valid mask) to avoid premature downmix. 2289 // (2) Perhaps use the maximum sample rate of all downstream MixerThreads 2290 // instead of the output device sample rate to avoid loss of high frequency information. 2291 // This may need to be updated as MixerThread/OutputTracks are added and not here. 2292 } 2293 2294 // Calculate size of normal sink buffer relative to the HAL output buffer size 2295 double multiplier = 1.0; 2296 if (mType == MIXER && (kUseFastMixer == FastMixer_Static || 2297 kUseFastMixer == FastMixer_Dynamic)) { 2298 size_t minNormalFrameCount = (kMinNormalSinkBufferSizeMs * mSampleRate) / 1000; 2299 size_t maxNormalFrameCount = (kMaxNormalSinkBufferSizeMs * mSampleRate) / 1000; 2300 // round up minimum and round down maximum to nearest 16 frames to satisfy AudioMixer 2301 minNormalFrameCount = (minNormalFrameCount + 15) & ~15; 2302 maxNormalFrameCount = maxNormalFrameCount & ~15; 2303 if (maxNormalFrameCount < minNormalFrameCount) { 2304 maxNormalFrameCount = minNormalFrameCount; 2305 } 2306 multiplier = (double) minNormalFrameCount / (double) mFrameCount; 2307 if (multiplier <= 1.0) { 2308 multiplier = 1.0; 2309 } else if (multiplier <= 2.0) { 2310 if (2 * mFrameCount <= maxNormalFrameCount) { 2311 multiplier = 2.0; 2312 } else { 2313 multiplier = (double) maxNormalFrameCount / (double) mFrameCount; 2314 } 2315 } else { 2316 // prefer an even multiplier, for compatibility with doubling of fast tracks due to HAL 2317 // SRC (it would be unusual for the normal sink buffer size to not be a multiple of fast 2318 // track, but we sometimes have to do this to satisfy the maximum frame count 2319 // constraint) 2320 // FIXME this rounding up should not be done if no HAL SRC 2321 uint32_t truncMult = (uint32_t) multiplier; 2322 if ((truncMult & 1)) { 2323 if ((truncMult + 1) * mFrameCount <= maxNormalFrameCount) { 2324 ++truncMult; 2325 } 2326 } 2327 multiplier = (double) truncMult; 2328 } 2329 } 2330 mNormalFrameCount = multiplier * mFrameCount; 2331 // round up to nearest 16 frames to satisfy AudioMixer 2332 if (mType == MIXER || mType == DUPLICATING) { 2333 mNormalFrameCount = (mNormalFrameCount + 15) & ~15; 2334 } 2335 ALOGI("HAL output buffer size %zu frames, normal sink buffer size %zu frames", mFrameCount, 2336 mNormalFrameCount); 2337 2338 // Check if we want to throttle the processing to no more than 2x normal rate 2339 mThreadThrottle = property_get_bool("af.thread.throttle", true /* default_value */); 2340 mThreadThrottleTimeMs = 0; 2341 mThreadThrottleEndMs = 0; 2342 mHalfBufferMs = mNormalFrameCount * 1000 / (2 * mSampleRate); 2343 2344 // mSinkBuffer is the sink buffer. Size is always multiple-of-16 frames. 2345 // Originally this was int16_t[] array, need to remove legacy implications. 2346 free(mSinkBuffer); 2347 mSinkBuffer = NULL; 2348 // For sink buffer size, we use the frame size from the downstream sink to avoid problems 2349 // with non PCM formats for compressed music, e.g. AAC, and Offload threads. 2350 const size_t sinkBufferSize = mNormalFrameCount * mFrameSize; 2351 (void)posix_memalign(&mSinkBuffer, 32, sinkBufferSize); 2352 2353 // We resize the mMixerBuffer according to the requirements of the sink buffer which 2354 // drives the output. 2355 free(mMixerBuffer); 2356 mMixerBuffer = NULL; 2357 if (mMixerBufferEnabled) { 2358 mMixerBufferFormat = AUDIO_FORMAT_PCM_FLOAT; // also valid: AUDIO_FORMAT_PCM_16_BIT. 2359 mMixerBufferSize = mNormalFrameCount * mChannelCount 2360 * audio_bytes_per_sample(mMixerBufferFormat); 2361 (void)posix_memalign(&mMixerBuffer, 32, mMixerBufferSize); 2362 } 2363 free(mEffectBuffer); 2364 mEffectBuffer = NULL; 2365 if (mEffectBufferEnabled) { 2366 mEffectBufferFormat = AUDIO_FORMAT_PCM_16_BIT; // Note: Effects support 16b only 2367 mEffectBufferSize = mNormalFrameCount * mChannelCount 2368 * audio_bytes_per_sample(mEffectBufferFormat); 2369 (void)posix_memalign(&mEffectBuffer, 32, mEffectBufferSize); 2370 } 2371 2372 // force reconfiguration of effect chains and engines to take new buffer size and audio 2373 // parameters into account 2374 // Note that mLock is not held when readOutputParameters_l() is called from the constructor 2375 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 2376 // matter. 2377 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 2378 Vector< sp<EffectChain> > effectChains = mEffectChains; 2379 for (size_t i = 0; i < effectChains.size(); i ++) { 2380 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 2381 } 2382} 2383 2384 2385status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames) 2386{ 2387 if (halFrames == NULL || dspFrames == NULL) { 2388 return BAD_VALUE; 2389 } 2390 Mutex::Autolock _l(mLock); 2391 if (initCheck() != NO_ERROR) { 2392 return INVALID_OPERATION; 2393 } 2394 int64_t framesWritten = mBytesWritten / mFrameSize; 2395 *halFrames = framesWritten; 2396 2397 if (isSuspended()) { 2398 // return an estimation of rendered frames when the output is suspended 2399 size_t latencyFrames = (latency_l() * mSampleRate) / 1000; 2400 *dspFrames = (uint32_t) 2401 (framesWritten >= (int64_t)latencyFrames ? framesWritten - latencyFrames : 0); 2402 return NO_ERROR; 2403 } else { 2404 status_t status; 2405 uint32_t frames; 2406 status = mOutput->getRenderPosition(&frames); 2407 *dspFrames = (size_t)frames; 2408 return status; 2409 } 2410} 2411 2412uint32_t AudioFlinger::PlaybackThread::hasAudioSession(audio_session_t sessionId) const 2413{ 2414 Mutex::Autolock _l(mLock); 2415 uint32_t result = 0; 2416 if (getEffectChain_l(sessionId) != 0) { 2417 result = EFFECT_SESSION; 2418 } 2419 2420 for (size_t i = 0; i < mTracks.size(); ++i) { 2421 sp<Track> track = mTracks[i]; 2422 if (sessionId == track->sessionId() && !track->isInvalid()) { 2423 result |= TRACK_SESSION; 2424 break; 2425 } 2426 } 2427 2428 return result; 2429} 2430 2431uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(audio_session_t sessionId) 2432{ 2433 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 2434 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 2435 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 2436 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 2437 } 2438 for (size_t i = 0; i < mTracks.size(); i++) { 2439 sp<Track> track = mTracks[i]; 2440 if (sessionId == track->sessionId() && !track->isInvalid()) { 2441 return AudioSystem::getStrategyForStream(track->streamType()); 2442 } 2443 } 2444 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 2445} 2446 2447 2448AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 2449{ 2450 Mutex::Autolock _l(mLock); 2451 return mOutput; 2452} 2453 2454AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 2455{ 2456 Mutex::Autolock _l(mLock); 2457 AudioStreamOut *output = mOutput; 2458 mOutput = NULL; 2459 // FIXME FastMixer might also have a raw ptr to mOutputSink; 2460 // must push a NULL and wait for ack 2461 mOutputSink.clear(); 2462 mPipeSink.clear(); 2463 mNormalSink.clear(); 2464 return output; 2465} 2466 2467// this method must always be called either with ThreadBase mLock held or inside the thread loop 2468audio_stream_t* AudioFlinger::PlaybackThread::stream() const 2469{ 2470 if (mOutput == NULL) { 2471 return NULL; 2472 } 2473 return &mOutput->stream->common; 2474} 2475 2476uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() const 2477{ 2478 return (uint32_t)((uint32_t)((mNormalFrameCount * 1000) / mSampleRate) * 1000); 2479} 2480 2481status_t AudioFlinger::PlaybackThread::setSyncEvent(const sp<SyncEvent>& event) 2482{ 2483 if (!isValidSyncEvent(event)) { 2484 return BAD_VALUE; 2485 } 2486 2487 Mutex::Autolock _l(mLock); 2488 2489 for (size_t i = 0; i < mTracks.size(); ++i) { 2490 sp<Track> track = mTracks[i]; 2491 if (event->triggerSession() == track->sessionId()) { 2492 (void) track->setSyncEvent(event); 2493 return NO_ERROR; 2494 } 2495 } 2496 2497 return NAME_NOT_FOUND; 2498} 2499 2500bool AudioFlinger::PlaybackThread::isValidSyncEvent(const sp<SyncEvent>& event) const 2501{ 2502 return event->type() == AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE; 2503} 2504 2505void AudioFlinger::PlaybackThread::threadLoop_removeTracks( 2506 const Vector< sp<Track> >& tracksToRemove) 2507{ 2508 size_t count = tracksToRemove.size(); 2509 if (count > 0) { 2510 for (size_t i = 0 ; i < count ; i++) { 2511 const sp<Track>& track = tracksToRemove.itemAt(i); 2512 if (track->isExternalTrack()) { 2513 AudioSystem::stopOutput(mId, track->streamType(), 2514 track->sessionId()); 2515#ifdef ADD_BATTERY_DATA 2516 // to track the speaker usage 2517 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 2518#endif 2519 if (track->isTerminated()) { 2520 AudioSystem::releaseOutput(mId, track->streamType(), 2521 track->sessionId()); 2522 } 2523 } 2524 } 2525 } 2526} 2527 2528void AudioFlinger::PlaybackThread::checkSilentMode_l() 2529{ 2530 if (!mMasterMute) { 2531 char value[PROPERTY_VALUE_MAX]; 2532 if (property_get("ro.audio.silent", value, "0") > 0) { 2533 char *endptr; 2534 unsigned long ul = strtoul(value, &endptr, 0); 2535 if (*endptr == '\0' && ul != 0) { 2536 ALOGD("Silence is golden"); 2537 // The setprop command will not allow a property to be changed after 2538 // the first time it is set, so we don't have to worry about un-muting. 2539 setMasterMute_l(true); 2540 } 2541 } 2542 } 2543} 2544 2545// shared by MIXER and DIRECT, overridden by DUPLICATING 2546ssize_t AudioFlinger::PlaybackThread::threadLoop_write() 2547{ 2548 // FIXME rewrite to reduce number of system calls 2549 mLastWriteTime = systemTime(); 2550 mInWrite = true; 2551 ssize_t bytesWritten; 2552 const size_t offset = mCurrentWriteLength - mBytesRemaining; 2553 2554 // If an NBAIO sink is present, use it to write the normal mixer's submix 2555 if (mNormalSink != 0) { 2556 2557 const size_t count = mBytesRemaining / mFrameSize; 2558 2559 ATRACE_BEGIN("write"); 2560 // update the setpoint when AudioFlinger::mScreenState changes 2561 uint32_t screenState = AudioFlinger::mScreenState; 2562 if (screenState != mScreenState) { 2563 mScreenState = screenState; 2564 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 2565 if (pipe != NULL) { 2566 pipe->setAvgFrames((mScreenState & 1) ? 2567 (pipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2568 } 2569 } 2570 ssize_t framesWritten = mNormalSink->write((char *)mSinkBuffer + offset, count); 2571 ATRACE_END(); 2572 if (framesWritten > 0) { 2573 bytesWritten = framesWritten * mFrameSize; 2574 } else { 2575 bytesWritten = framesWritten; 2576 } 2577 // otherwise use the HAL / AudioStreamOut directly 2578 } else { 2579 // Direct output and offload threads 2580 2581 if (mUseAsyncWrite) { 2582 ALOGW_IF(mWriteAckSequence & 1, "threadLoop_write(): out of sequence write request"); 2583 mWriteAckSequence += 2; 2584 mWriteAckSequence |= 1; 2585 ALOG_ASSERT(mCallbackThread != 0); 2586 mCallbackThread->setWriteBlocked(mWriteAckSequence); 2587 } 2588 // FIXME We should have an implementation of timestamps for direct output threads. 2589 // They are used e.g for multichannel PCM playback over HDMI. 2590 bytesWritten = mOutput->write((char *)mSinkBuffer + offset, mBytesRemaining); 2591 2592 if (mUseAsyncWrite && 2593 ((bytesWritten < 0) || (bytesWritten == (ssize_t)mBytesRemaining))) { 2594 // do not wait for async callback in case of error of full write 2595 mWriteAckSequence &= ~1; 2596 ALOG_ASSERT(mCallbackThread != 0); 2597 mCallbackThread->setWriteBlocked(mWriteAckSequence); 2598 } 2599 } 2600 2601 mNumWrites++; 2602 mInWrite = false; 2603 mStandby = false; 2604 return bytesWritten; 2605} 2606 2607void AudioFlinger::PlaybackThread::threadLoop_drain() 2608{ 2609 if (mOutput->stream->drain) { 2610 ALOGV("draining %s", (mMixerStatus == MIXER_DRAIN_TRACK) ? "early" : "full"); 2611 if (mUseAsyncWrite) { 2612 ALOGW_IF(mDrainSequence & 1, "threadLoop_drain(): out of sequence drain request"); 2613 mDrainSequence |= 1; 2614 ALOG_ASSERT(mCallbackThread != 0); 2615 mCallbackThread->setDraining(mDrainSequence); 2616 } 2617 mOutput->stream->drain(mOutput->stream, 2618 (mMixerStatus == MIXER_DRAIN_TRACK) ? AUDIO_DRAIN_EARLY_NOTIFY 2619 : AUDIO_DRAIN_ALL); 2620 } 2621} 2622 2623void AudioFlinger::PlaybackThread::threadLoop_exit() 2624{ 2625 { 2626 Mutex::Autolock _l(mLock); 2627 for (size_t i = 0; i < mTracks.size(); i++) { 2628 sp<Track> track = mTracks[i]; 2629 track->invalidate(); 2630 } 2631 } 2632} 2633 2634/* 2635The derived values that are cached: 2636 - mSinkBufferSize from frame count * frame size 2637 - mActiveSleepTimeUs from activeSleepTimeUs() 2638 - mIdleSleepTimeUs from idleSleepTimeUs() 2639 - mStandbyDelayNs from mActiveSleepTimeUs (DIRECT only) or forced to at least 2640 kDefaultStandbyTimeInNsecs when connected to an A2DP device. 2641 - maxPeriod from frame count and sample rate (MIXER only) 2642 2643The parameters that affect these derived values are: 2644 - frame count 2645 - frame size 2646 - sample rate 2647 - device type: A2DP or not 2648 - device latency 2649 - format: PCM or not 2650 - active sleep time 2651 - idle sleep time 2652*/ 2653 2654void AudioFlinger::PlaybackThread::cacheParameters_l() 2655{ 2656 mSinkBufferSize = mNormalFrameCount * mFrameSize; 2657 mActiveSleepTimeUs = activeSleepTimeUs(); 2658 mIdleSleepTimeUs = idleSleepTimeUs(); 2659 2660 // make sure standby delay is not too short when connected to an A2DP sink to avoid 2661 // truncating audio when going to standby. 2662 mStandbyDelayNs = AudioFlinger::mStandbyTimeInNsecs; 2663 if ((mOutDevice & AUDIO_DEVICE_OUT_ALL_A2DP) != 0) { 2664 if (mStandbyDelayNs < kDefaultStandbyTimeInNsecs) { 2665 mStandbyDelayNs = kDefaultStandbyTimeInNsecs; 2666 } 2667 } 2668} 2669 2670void AudioFlinger::PlaybackThread::invalidateTracks(audio_stream_type_t streamType) 2671{ 2672 ALOGV("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %zu", 2673 this, streamType, mTracks.size()); 2674 Mutex::Autolock _l(mLock); 2675 2676 size_t size = mTracks.size(); 2677 for (size_t i = 0; i < size; i++) { 2678 sp<Track> t = mTracks[i]; 2679 if (t->streamType() == streamType && t->isExternalTrack()) { 2680 t->invalidate(); 2681 } 2682 } 2683} 2684 2685status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 2686{ 2687 audio_session_t session = chain->sessionId(); 2688 int16_t* buffer = reinterpret_cast<int16_t*>(mEffectBufferEnabled 2689 ? mEffectBuffer : mSinkBuffer); 2690 bool ownsBuffer = false; 2691 2692 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 2693 if (session > AUDIO_SESSION_OUTPUT_MIX) { 2694 // Only one effect chain can be present in direct output thread and it uses 2695 // the sink buffer as input 2696 if (mType != DIRECT) { 2697 size_t numSamples = mNormalFrameCount * mChannelCount; 2698 buffer = new int16_t[numSamples]; 2699 memset(buffer, 0, numSamples * sizeof(int16_t)); 2700 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 2701 ownsBuffer = true; 2702 } 2703 2704 // Attach all tracks with same session ID to this chain. 2705 for (size_t i = 0; i < mTracks.size(); ++i) { 2706 sp<Track> track = mTracks[i]; 2707 if (session == track->sessionId()) { 2708 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), 2709 buffer); 2710 track->setMainBuffer(buffer); 2711 chain->incTrackCnt(); 2712 } 2713 } 2714 2715 // indicate all active tracks in the chain 2716 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 2717 sp<Track> track = mActiveTracks[i].promote(); 2718 if (track == 0) { 2719 continue; 2720 } 2721 if (session == track->sessionId()) { 2722 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 2723 chain->incActiveTrackCnt(); 2724 } 2725 } 2726 } 2727 chain->setThread(this); 2728 chain->setInBuffer(buffer, ownsBuffer); 2729 chain->setOutBuffer(reinterpret_cast<int16_t*>(mEffectBufferEnabled 2730 ? mEffectBuffer : mSinkBuffer)); 2731 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 2732 // chains list in order to be processed last as it contains output stage effects. 2733 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 2734 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 2735 // after track specific effects and before output stage. 2736 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 2737 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX. 2738 // Effect chain for other sessions are inserted at beginning of effect 2739 // chains list to be processed before output mix effects. Relative order between other 2740 // sessions is not important. 2741 static_assert(AUDIO_SESSION_OUTPUT_MIX == 0 && 2742 AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX, 2743 "audio_session_t constants misdefined"); 2744 size_t size = mEffectChains.size(); 2745 size_t i = 0; 2746 for (i = 0; i < size; i++) { 2747 if (mEffectChains[i]->sessionId() < session) { 2748 break; 2749 } 2750 } 2751 mEffectChains.insertAt(chain, i); 2752 checkSuspendOnAddEffectChain_l(chain); 2753 2754 return NO_ERROR; 2755} 2756 2757size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 2758{ 2759 audio_session_t session = chain->sessionId(); 2760 2761 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 2762 2763 for (size_t i = 0; i < mEffectChains.size(); i++) { 2764 if (chain == mEffectChains[i]) { 2765 mEffectChains.removeAt(i); 2766 // detach all active tracks from the chain 2767 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 2768 sp<Track> track = mActiveTracks[i].promote(); 2769 if (track == 0) { 2770 continue; 2771 } 2772 if (session == track->sessionId()) { 2773 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 2774 chain.get(), session); 2775 chain->decActiveTrackCnt(); 2776 } 2777 } 2778 2779 // detach all tracks with same session ID from this chain 2780 for (size_t i = 0; i < mTracks.size(); ++i) { 2781 sp<Track> track = mTracks[i]; 2782 if (session == track->sessionId()) { 2783 track->setMainBuffer(reinterpret_cast<int16_t*>(mSinkBuffer)); 2784 chain->decTrackCnt(); 2785 } 2786 } 2787 break; 2788 } 2789 } 2790 return mEffectChains.size(); 2791} 2792 2793status_t AudioFlinger::PlaybackThread::attachAuxEffect( 2794 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2795{ 2796 Mutex::Autolock _l(mLock); 2797 return attachAuxEffect_l(track, EffectId); 2798} 2799 2800status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 2801 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2802{ 2803 status_t status = NO_ERROR; 2804 2805 if (EffectId == 0) { 2806 track->setAuxBuffer(0, NULL); 2807 } else { 2808 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 2809 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 2810 if (effect != 0) { 2811 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 2812 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 2813 } else { 2814 status = INVALID_OPERATION; 2815 } 2816 } else { 2817 status = BAD_VALUE; 2818 } 2819 } 2820 return status; 2821} 2822 2823void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 2824{ 2825 for (size_t i = 0; i < mTracks.size(); ++i) { 2826 sp<Track> track = mTracks[i]; 2827 if (track->auxEffectId() == effectId) { 2828 attachAuxEffect_l(track, 0); 2829 } 2830 } 2831} 2832 2833bool AudioFlinger::PlaybackThread::threadLoop() 2834{ 2835 Vector< sp<Track> > tracksToRemove; 2836 2837 mStandbyTimeNs = systemTime(); 2838 2839 // MIXER 2840 nsecs_t lastWarning = 0; 2841 2842 // DUPLICATING 2843 // FIXME could this be made local to while loop? 2844 writeFrames = 0; 2845 2846 int lastGeneration = 0; 2847 2848 cacheParameters_l(); 2849 mSleepTimeUs = mIdleSleepTimeUs; 2850 2851 if (mType == MIXER) { 2852 sleepTimeShift = 0; 2853 } 2854 2855 CpuStats cpuStats; 2856 const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid())); 2857 2858 acquireWakeLock(); 2859 2860 // mNBLogWriter->log can only be called while thread mutex mLock is held. 2861 // So if you need to log when mutex is unlocked, set logString to a non-NULL string, 2862 // and then that string will be logged at the next convenient opportunity. 2863 const char *logString = NULL; 2864 2865 checkSilentMode_l(); 2866 2867 while (!exitPending()) 2868 { 2869 cpuStats.sample(myName); 2870 2871 Vector< sp<EffectChain> > effectChains; 2872 2873 { // scope for mLock 2874 2875 Mutex::Autolock _l(mLock); 2876 2877 processConfigEvents_l(); 2878 2879 if (logString != NULL) { 2880 mNBLogWriter->logTimestamp(); 2881 mNBLogWriter->log(logString); 2882 logString = NULL; 2883 } 2884 2885 // Gather the framesReleased counters for all active tracks, 2886 // and associate with the sink frames written out. We need 2887 // this to convert the sink timestamp to the track timestamp. 2888 if (mNormalSink != 0) { 2889 // Note: The DuplicatingThread may not have a mNormalSink. 2890 // We always fetch the timestamp here because often the downstream 2891 // sink will block whie writing. 2892 ExtendedTimestamp timestamp; // use private copy to fetch 2893 (void) mNormalSink->getTimestamp(timestamp); 2894 // copy over kernel info 2895 mTimestamp.mPosition[ExtendedTimestamp::LOCATION_KERNEL] = 2896 timestamp.mPosition[ExtendedTimestamp::LOCATION_KERNEL]; 2897 mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL] = 2898 timestamp.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL]; 2899 } 2900 // mFramesWritten for non-offloaded tracks are contiguous 2901 // even after standby() is called. This is useful for the track frame 2902 // to sink frame mapping. 2903 mTimestamp.mPosition[ExtendedTimestamp::LOCATION_SERVER] = mFramesWritten; 2904 mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_SERVER] = systemTime(); 2905 const size_t size = mActiveTracks.size(); 2906 for (size_t i = 0; i < size; ++i) { 2907 sp<Track> t = mActiveTracks[i].promote(); 2908 if (t != 0 && !t->isFastTrack()) { 2909 t->updateTrackFrameInfo( 2910 t->mAudioTrackServerProxy->framesReleased(), 2911 mFramesWritten, 2912 mTimestamp); 2913 } 2914 } 2915 2916 saveOutputTracks(); 2917 if (mSignalPending) { 2918 // A signal was raised while we were unlocked 2919 mSignalPending = false; 2920 } else if (waitingAsyncCallback_l()) { 2921 if (exitPending()) { 2922 break; 2923 } 2924 bool released = false; 2925 // The following works around a bug in the offload driver. Ideally we would release 2926 // the wake lock every time, but that causes the last offload buffer(s) to be 2927 // dropped while the device is on battery, so we need to hold a wake lock during 2928 // the drain phase. 2929 if (mBytesRemaining && !(mDrainSequence & 1)) { 2930 releaseWakeLock_l(); 2931 released = true; 2932 } 2933 mWakeLockUids.clear(); 2934 mActiveTracksGeneration++; 2935 ALOGV("wait async completion"); 2936 mWaitWorkCV.wait(mLock); 2937 ALOGV("async completion/wake"); 2938 if (released) { 2939 acquireWakeLock_l(); 2940 } 2941 mStandbyTimeNs = systemTime() + mStandbyDelayNs; 2942 mSleepTimeUs = 0; 2943 2944 continue; 2945 } 2946 if ((!mActiveTracks.size() && systemTime() > mStandbyTimeNs) || 2947 isSuspended()) { 2948 // put audio hardware into standby after short delay 2949 if (shouldStandby_l()) { 2950 2951 threadLoop_standby(); 2952 2953 mStandby = true; 2954 } 2955 2956 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2957 // we're about to wait, flush the binder command buffer 2958 IPCThreadState::self()->flushCommands(); 2959 2960 clearOutputTracks(); 2961 2962 if (exitPending()) { 2963 break; 2964 } 2965 2966 releaseWakeLock_l(); 2967 mWakeLockUids.clear(); 2968 mActiveTracksGeneration++; 2969 // wait until we have something to do... 2970 ALOGV("%s going to sleep", myName.string()); 2971 mWaitWorkCV.wait(mLock); 2972 ALOGV("%s waking up", myName.string()); 2973 acquireWakeLock_l(); 2974 2975 mMixerStatus = MIXER_IDLE; 2976 mMixerStatusIgnoringFastTracks = MIXER_IDLE; 2977 mBytesWritten = 0; 2978 mBytesRemaining = 0; 2979 checkSilentMode_l(); 2980 2981 mStandbyTimeNs = systemTime() + mStandbyDelayNs; 2982 mSleepTimeUs = mIdleSleepTimeUs; 2983 if (mType == MIXER) { 2984 sleepTimeShift = 0; 2985 } 2986 2987 continue; 2988 } 2989 } 2990 // mMixerStatusIgnoringFastTracks is also updated internally 2991 mMixerStatus = prepareTracks_l(&tracksToRemove); 2992 2993 // compare with previously applied list 2994 if (lastGeneration != mActiveTracksGeneration) { 2995 // update wakelock 2996 updateWakeLockUids_l(mWakeLockUids); 2997 lastGeneration = mActiveTracksGeneration; 2998 } 2999 3000 // prevent any changes in effect chain list and in each effect chain 3001 // during mixing and effect process as the audio buffers could be deleted 3002 // or modified if an effect is created or deleted 3003 lockEffectChains_l(effectChains); 3004 } // mLock scope ends 3005 3006 if (mBytesRemaining == 0) { 3007 mCurrentWriteLength = 0; 3008 if (mMixerStatus == MIXER_TRACKS_READY) { 3009 // threadLoop_mix() sets mCurrentWriteLength 3010 threadLoop_mix(); 3011 } else if ((mMixerStatus != MIXER_DRAIN_TRACK) 3012 && (mMixerStatus != MIXER_DRAIN_ALL)) { 3013 // threadLoop_sleepTime sets mSleepTimeUs to 0 if data 3014 // must be written to HAL 3015 threadLoop_sleepTime(); 3016 if (mSleepTimeUs == 0) { 3017 mCurrentWriteLength = mSinkBufferSize; 3018 } 3019 } 3020 // Either threadLoop_mix() or threadLoop_sleepTime() should have set 3021 // mMixerBuffer with data if mMixerBufferValid is true and mSleepTimeUs == 0. 3022 // Merge mMixerBuffer data into mEffectBuffer (if any effects are valid) 3023 // or mSinkBuffer (if there are no effects). 3024 // 3025 // This is done pre-effects computation; if effects change to 3026 // support higher precision, this needs to move. 3027 // 3028 // mMixerBufferValid is only set true by MixerThread::prepareTracks_l(). 3029 // TODO use mSleepTimeUs == 0 as an additional condition. 3030 if (mMixerBufferValid) { 3031 void *buffer = mEffectBufferValid ? mEffectBuffer : mSinkBuffer; 3032 audio_format_t format = mEffectBufferValid ? mEffectBufferFormat : mFormat; 3033 3034 // mono blend occurs for mixer threads only (not direct or offloaded) 3035 // and is handled here if we're going directly to the sink. 3036 if (requireMonoBlend() && !mEffectBufferValid) { 3037 mono_blend(mMixerBuffer, mMixerBufferFormat, mChannelCount, mNormalFrameCount, 3038 true /*limit*/); 3039 } 3040 3041 memcpy_by_audio_format(buffer, format, mMixerBuffer, mMixerBufferFormat, 3042 mNormalFrameCount * mChannelCount); 3043 } 3044 3045 mBytesRemaining = mCurrentWriteLength; 3046 if (isSuspended()) { 3047 mSleepTimeUs = suspendSleepTimeUs(); 3048 // simulate write to HAL when suspended 3049 mBytesWritten += mSinkBufferSize; 3050 mFramesWritten += mSinkBufferSize / mFrameSize; 3051 mBytesRemaining = 0; 3052 } 3053 3054 // only process effects if we're going to write 3055 if (mSleepTimeUs == 0 && mType != OFFLOAD) { 3056 for (size_t i = 0; i < effectChains.size(); i ++) { 3057 effectChains[i]->process_l(); 3058 } 3059 } 3060 } 3061 // Process effect chains for offloaded thread even if no audio 3062 // was read from audio track: process only updates effect state 3063 // and thus does have to be synchronized with audio writes but may have 3064 // to be called while waiting for async write callback 3065 if (mType == OFFLOAD) { 3066 for (size_t i = 0; i < effectChains.size(); i ++) { 3067 effectChains[i]->process_l(); 3068 } 3069 } 3070 3071 // Only if the Effects buffer is enabled and there is data in the 3072 // Effects buffer (buffer valid), we need to 3073 // copy into the sink buffer. 3074 // TODO use mSleepTimeUs == 0 as an additional condition. 3075 if (mEffectBufferValid) { 3076 //ALOGV("writing effect buffer to sink buffer format %#x", mFormat); 3077 3078 if (requireMonoBlend()) { 3079 mono_blend(mEffectBuffer, mEffectBufferFormat, mChannelCount, mNormalFrameCount, 3080 true /*limit*/); 3081 } 3082 3083 memcpy_by_audio_format(mSinkBuffer, mFormat, mEffectBuffer, mEffectBufferFormat, 3084 mNormalFrameCount * mChannelCount); 3085 } 3086 3087 // enable changes in effect chain 3088 unlockEffectChains(effectChains); 3089 3090 if (!waitingAsyncCallback()) { 3091 // mSleepTimeUs == 0 means we must write to audio hardware 3092 if (mSleepTimeUs == 0) { 3093 ssize_t ret = 0; 3094 if (mBytesRemaining) { 3095 ret = threadLoop_write(); 3096 if (ret < 0) { 3097 mBytesRemaining = 0; 3098 } else { 3099 mBytesWritten += ret; 3100 mBytesRemaining -= ret; 3101 mFramesWritten += ret / mFrameSize; 3102 } 3103 } else if ((mMixerStatus == MIXER_DRAIN_TRACK) || 3104 (mMixerStatus == MIXER_DRAIN_ALL)) { 3105 threadLoop_drain(); 3106 } 3107 if (mType == MIXER && !mStandby) { 3108 // write blocked detection 3109 nsecs_t now = systemTime(); 3110 nsecs_t delta = now - mLastWriteTime; 3111 if (delta > maxPeriod) { 3112 mNumDelayedWrites++; 3113 if ((now - lastWarning) > kWarningThrottleNs) { 3114 ATRACE_NAME("underrun"); 3115 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 3116 (unsigned long long) ns2ms(delta), mNumDelayedWrites, this); 3117 lastWarning = now; 3118 } 3119 } 3120 3121 if (mThreadThrottle 3122 && mMixerStatus == MIXER_TRACKS_READY // we are mixing (active tracks) 3123 && ret > 0) { // we wrote something 3124 // Limit MixerThread data processing to no more than twice the 3125 // expected processing rate. 3126 // 3127 // This helps prevent underruns with NuPlayer and other applications 3128 // which may set up buffers that are close to the minimum size, or use 3129 // deep buffers, and rely on a double-buffering sleep strategy to fill. 3130 // 3131 // The throttle smooths out sudden large data drains from the device, 3132 // e.g. when it comes out of standby, which often causes problems with 3133 // (1) mixer threads without a fast mixer (which has its own warm-up) 3134 // (2) minimum buffer sized tracks (even if the track is full, 3135 // the app won't fill fast enough to handle the sudden draw). 3136 3137 const int32_t deltaMs = delta / 1000000; 3138 const int32_t throttleMs = mHalfBufferMs - deltaMs; 3139 if ((signed)mHalfBufferMs >= throttleMs && throttleMs > 0) { 3140 usleep(throttleMs * 1000); 3141 // notify of throttle start on verbose log 3142 ALOGV_IF(mThreadThrottleEndMs == mThreadThrottleTimeMs, 3143 "mixer(%p) throttle begin:" 3144 " ret(%zd) deltaMs(%d) requires sleep %d ms", 3145 this, ret, deltaMs, throttleMs); 3146 mThreadThrottleTimeMs += throttleMs; 3147 } else { 3148 uint32_t diff = mThreadThrottleTimeMs - mThreadThrottleEndMs; 3149 if (diff > 0) { 3150 // notify of throttle end on debug log 3151 ALOGD("mixer(%p) throttle end: throttle time(%u)", this, diff); 3152 mThreadThrottleEndMs = mThreadThrottleTimeMs; 3153 } 3154 } 3155 } 3156 } 3157 3158 } else { 3159 ATRACE_BEGIN("sleep"); 3160 if ((mType == OFFLOAD) && !audio_has_proportional_frames(mFormat)) { 3161 Mutex::Autolock _l(mLock); 3162 if (!mSignalPending && !exitPending()) { 3163 // Do not sleep more than one buffer duration since last write and not 3164 // less than kDirectMinSleepTimeUs 3165 // Wake up if a command is received 3166 nsecs_t now = systemTime(); 3167 uint32_t deltaUs = (uint32_t)((now - mLastWriteTime) / 1000); 3168 uint32_t timeoutUs = mSleepTimeUs; 3169 if (timeoutUs + deltaUs > mBufferDurationUs) { 3170 if (mBufferDurationUs > deltaUs) { 3171 timeoutUs = mBufferDurationUs - deltaUs; 3172 if (timeoutUs < kDirectMinSleepTimeUs) { 3173 timeoutUs = kDirectMinSleepTimeUs; 3174 } 3175 } else { 3176 timeoutUs = kDirectMinSleepTimeUs; 3177 } 3178 } 3179 mWaitWorkCV.waitRelative(mLock, microseconds((nsecs_t)timeoutUs)); 3180 } 3181 } else { 3182 usleep(mSleepTimeUs); 3183 } 3184 ATRACE_END(); 3185 } 3186 } 3187 3188 // Finally let go of removed track(s), without the lock held 3189 // since we can't guarantee the destructors won't acquire that 3190 // same lock. This will also mutate and push a new fast mixer state. 3191 threadLoop_removeTracks(tracksToRemove); 3192 tracksToRemove.clear(); 3193 3194 // FIXME I don't understand the need for this here; 3195 // it was in the original code but maybe the 3196 // assignment in saveOutputTracks() makes this unnecessary? 3197 clearOutputTracks(); 3198 3199 // Effect chains will be actually deleted here if they were removed from 3200 // mEffectChains list during mixing or effects processing 3201 effectChains.clear(); 3202 3203 // FIXME Note that the above .clear() is no longer necessary since effectChains 3204 // is now local to this block, but will keep it for now (at least until merge done). 3205 } 3206 3207 threadLoop_exit(); 3208 3209 if (!mStandby) { 3210 threadLoop_standby(); 3211 mStandby = true; 3212 } 3213 3214 releaseWakeLock(); 3215 mWakeLockUids.clear(); 3216 mActiveTracksGeneration++; 3217 3218 ALOGV("Thread %p type %d exiting", this, mType); 3219 return false; 3220} 3221 3222// removeTracks_l() must be called with ThreadBase::mLock held 3223void AudioFlinger::PlaybackThread::removeTracks_l(const Vector< sp<Track> >& tracksToRemove) 3224{ 3225 size_t count = tracksToRemove.size(); 3226 if (count > 0) { 3227 for (size_t i=0 ; i<count ; i++) { 3228 const sp<Track>& track = tracksToRemove.itemAt(i); 3229 mActiveTracks.remove(track); 3230 mWakeLockUids.remove(track->uid()); 3231 mActiveTracksGeneration++; 3232 ALOGV("removeTracks_l removing track on session %d", track->sessionId()); 3233 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 3234 if (chain != 0) { 3235 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), 3236 track->sessionId()); 3237 chain->decActiveTrackCnt(); 3238 } 3239 if (track->isTerminated()) { 3240 removeTrack_l(track); 3241 } 3242 } 3243 } 3244 3245} 3246 3247status_t AudioFlinger::PlaybackThread::getTimestamp_l(AudioTimestamp& timestamp) 3248{ 3249 if (mNormalSink != 0) { 3250 ExtendedTimestamp ets; 3251 status_t status = mNormalSink->getTimestamp(ets); 3252 if (status == NO_ERROR) { 3253 status = ets.getBestTimestamp(×tamp); 3254 } 3255 return status; 3256 } 3257 if ((mType == OFFLOAD || mType == DIRECT) 3258 && mOutput != NULL && mOutput->stream->get_presentation_position) { 3259 uint64_t position64; 3260 int ret = mOutput->getPresentationPosition(&position64, ×tamp.mTime); 3261 if (ret == 0) { 3262 timestamp.mPosition = (uint32_t)position64; 3263 return NO_ERROR; 3264 } 3265 } 3266 return INVALID_OPERATION; 3267} 3268 3269status_t AudioFlinger::MixerThread::createAudioPatch_l(const struct audio_patch *patch, 3270 audio_patch_handle_t *handle) 3271{ 3272 // if !&IDLE, holds the FastMixer state to restore after new parameters processed 3273 FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE; 3274 if (mFastMixer != 0) { 3275 FastMixerStateQueue *sq = mFastMixer->sq(); 3276 FastMixerState *state = sq->begin(); 3277 if (!(state->mCommand & FastMixerState::IDLE)) { 3278 previousCommand = state->mCommand; 3279 state->mCommand = FastMixerState::HOT_IDLE; 3280 sq->end(); 3281 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3282 } else { 3283 sq->end(false /*didModify*/); 3284 } 3285 } 3286 status_t status = PlaybackThread::createAudioPatch_l(patch, handle); 3287 3288 if (!(previousCommand & FastMixerState::IDLE)) { 3289 ALOG_ASSERT(mFastMixer != 0); 3290 FastMixerStateQueue *sq = mFastMixer->sq(); 3291 FastMixerState *state = sq->begin(); 3292 ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE); 3293 state->mCommand = previousCommand; 3294 sq->end(); 3295 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3296 } 3297 3298 return status; 3299} 3300 3301status_t AudioFlinger::PlaybackThread::createAudioPatch_l(const struct audio_patch *patch, 3302 audio_patch_handle_t *handle) 3303{ 3304 status_t status = NO_ERROR; 3305 3306 // store new device and send to effects 3307 audio_devices_t type = AUDIO_DEVICE_NONE; 3308 for (unsigned int i = 0; i < patch->num_sinks; i++) { 3309 type |= patch->sinks[i].ext.device.type; 3310 } 3311 3312#ifdef ADD_BATTERY_DATA 3313 // when changing the audio output device, call addBatteryData to notify 3314 // the change 3315 if (mOutDevice != type) { 3316 uint32_t params = 0; 3317 // check whether speaker is on 3318 if (type & AUDIO_DEVICE_OUT_SPEAKER) { 3319 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 3320 } 3321 3322 audio_devices_t deviceWithoutSpeaker 3323 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 3324 // check if any other device (except speaker) is on 3325 if (type & deviceWithoutSpeaker) { 3326 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 3327 } 3328 3329 if (params != 0) { 3330 addBatteryData(params); 3331 } 3332 } 3333#endif 3334 3335 for (size_t i = 0; i < mEffectChains.size(); i++) { 3336 mEffectChains[i]->setDevice_l(type); 3337 } 3338 3339 // mPrevOutDevice is the latest device set by createAudioPatch_l(). It is not set when 3340 // the thread is created so that the first patch creation triggers an ioConfigChanged callback 3341 bool configChanged = mPrevOutDevice != type; 3342 mOutDevice = type; 3343 mPatch = *patch; 3344 3345 if (mOutput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 3346 audio_hw_device_t *hwDevice = mOutput->audioHwDev->hwDevice(); 3347 status = hwDevice->create_audio_patch(hwDevice, 3348 patch->num_sources, 3349 patch->sources, 3350 patch->num_sinks, 3351 patch->sinks, 3352 handle); 3353 } else { 3354 char *address; 3355 if (strcmp(patch->sinks[0].ext.device.address, "") != 0) { 3356 //FIXME: we only support address on first sink with HAL version < 3.0 3357 address = audio_device_address_to_parameter( 3358 patch->sinks[0].ext.device.type, 3359 patch->sinks[0].ext.device.address); 3360 } else { 3361 address = (char *)calloc(1, 1); 3362 } 3363 AudioParameter param = AudioParameter(String8(address)); 3364 free(address); 3365 param.addInt(String8(AUDIO_PARAMETER_STREAM_ROUTING), (int)type); 3366 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3367 param.toString().string()); 3368 *handle = AUDIO_PATCH_HANDLE_NONE; 3369 } 3370 if (configChanged) { 3371 mPrevOutDevice = type; 3372 sendIoConfigEvent_l(AUDIO_OUTPUT_CONFIG_CHANGED); 3373 } 3374 return status; 3375} 3376 3377status_t AudioFlinger::MixerThread::releaseAudioPatch_l(const audio_patch_handle_t handle) 3378{ 3379 // if !&IDLE, holds the FastMixer state to restore after new parameters processed 3380 FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE; 3381 if (mFastMixer != 0) { 3382 FastMixerStateQueue *sq = mFastMixer->sq(); 3383 FastMixerState *state = sq->begin(); 3384 if (!(state->mCommand & FastMixerState::IDLE)) { 3385 previousCommand = state->mCommand; 3386 state->mCommand = FastMixerState::HOT_IDLE; 3387 sq->end(); 3388 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3389 } else { 3390 sq->end(false /*didModify*/); 3391 } 3392 } 3393 3394 status_t status = PlaybackThread::releaseAudioPatch_l(handle); 3395 3396 if (!(previousCommand & FastMixerState::IDLE)) { 3397 ALOG_ASSERT(mFastMixer != 0); 3398 FastMixerStateQueue *sq = mFastMixer->sq(); 3399 FastMixerState *state = sq->begin(); 3400 ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE); 3401 state->mCommand = previousCommand; 3402 sq->end(); 3403 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3404 } 3405 3406 return status; 3407} 3408 3409status_t AudioFlinger::PlaybackThread::releaseAudioPatch_l(const audio_patch_handle_t handle) 3410{ 3411 status_t status = NO_ERROR; 3412 3413 mOutDevice = AUDIO_DEVICE_NONE; 3414 3415 if (mOutput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 3416 audio_hw_device_t *hwDevice = mOutput->audioHwDev->hwDevice(); 3417 status = hwDevice->release_audio_patch(hwDevice, handle); 3418 } else { 3419 AudioParameter param; 3420 param.addInt(String8(AUDIO_PARAMETER_STREAM_ROUTING), 0); 3421 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3422 param.toString().string()); 3423 } 3424 return status; 3425} 3426 3427void AudioFlinger::PlaybackThread::addPatchTrack(const sp<PatchTrack>& track) 3428{ 3429 Mutex::Autolock _l(mLock); 3430 mTracks.add(track); 3431} 3432 3433void AudioFlinger::PlaybackThread::deletePatchTrack(const sp<PatchTrack>& track) 3434{ 3435 Mutex::Autolock _l(mLock); 3436 destroyTrack_l(track); 3437} 3438 3439void AudioFlinger::PlaybackThread::getAudioPortConfig(struct audio_port_config *config) 3440{ 3441 ThreadBase::getAudioPortConfig(config); 3442 config->role = AUDIO_PORT_ROLE_SOURCE; 3443 config->ext.mix.hw_module = mOutput->audioHwDev->handle(); 3444 config->ext.mix.usecase.stream = AUDIO_STREAM_DEFAULT; 3445} 3446 3447// ---------------------------------------------------------------------------- 3448 3449AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 3450 audio_io_handle_t id, audio_devices_t device, bool systemReady, type_t type) 3451 : PlaybackThread(audioFlinger, output, id, device, type, systemReady), 3452 // mAudioMixer below 3453 // mFastMixer below 3454 mFastMixerFutex(0), 3455 mMasterMono(false) 3456 // mOutputSink below 3457 // mPipeSink below 3458 // mNormalSink below 3459{ 3460 ALOGV("MixerThread() id=%d device=%#x type=%d", id, device, type); 3461 ALOGV("mSampleRate=%u, mChannelMask=%#x, mChannelCount=%u, mFormat=%d, mFrameSize=%zu, " 3462 "mFrameCount=%zu, mNormalFrameCount=%zu", 3463 mSampleRate, mChannelMask, mChannelCount, mFormat, mFrameSize, mFrameCount, 3464 mNormalFrameCount); 3465 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 3466 3467 if (type == DUPLICATING) { 3468 // The Duplicating thread uses the AudioMixer and delivers data to OutputTracks 3469 // (downstream MixerThreads) in DuplicatingThread::threadLoop_write(). 3470 // Do not create or use mFastMixer, mOutputSink, mPipeSink, or mNormalSink. 3471 return; 3472 } 3473 // create an NBAIO sink for the HAL output stream, and negotiate 3474 mOutputSink = new AudioStreamOutSink(output->stream); 3475 size_t numCounterOffers = 0; 3476 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount, mFormat)}; 3477#if !LOG_NDEBUG 3478 ssize_t index = 3479#else 3480 (void) 3481#endif 3482 mOutputSink->negotiate(offers, 1, NULL, numCounterOffers); 3483 ALOG_ASSERT(index == 0); 3484 3485 // initialize fast mixer depending on configuration 3486 bool initFastMixer; 3487 switch (kUseFastMixer) { 3488 case FastMixer_Never: 3489 initFastMixer = false; 3490 break; 3491 case FastMixer_Always: 3492 initFastMixer = true; 3493 break; 3494 case FastMixer_Static: 3495 case FastMixer_Dynamic: 3496 initFastMixer = mFrameCount < mNormalFrameCount; 3497 break; 3498 } 3499 if (initFastMixer) { 3500 audio_format_t fastMixerFormat; 3501 if (mMixerBufferEnabled && mEffectBufferEnabled) { 3502 fastMixerFormat = AUDIO_FORMAT_PCM_FLOAT; 3503 } else { 3504 fastMixerFormat = AUDIO_FORMAT_PCM_16_BIT; 3505 } 3506 if (mFormat != fastMixerFormat) { 3507 // change our Sink format to accept our intermediate precision 3508 mFormat = fastMixerFormat; 3509 free(mSinkBuffer); 3510 mFrameSize = mChannelCount * audio_bytes_per_sample(mFormat); 3511 const size_t sinkBufferSize = mNormalFrameCount * mFrameSize; 3512 (void)posix_memalign(&mSinkBuffer, 32, sinkBufferSize); 3513 } 3514 3515 // create a MonoPipe to connect our submix to FastMixer 3516 NBAIO_Format format = mOutputSink->format(); 3517#ifdef TEE_SINK 3518 NBAIO_Format origformat = format; 3519#endif 3520 // adjust format to match that of the Fast Mixer 3521 ALOGV("format changed from %d to %d", format.mFormat, fastMixerFormat); 3522 format.mFormat = fastMixerFormat; 3523 format.mFrameSize = audio_bytes_per_sample(format.mFormat) * format.mChannelCount; 3524 3525 // This pipe depth compensates for scheduling latency of the normal mixer thread. 3526 // When it wakes up after a maximum latency, it runs a few cycles quickly before 3527 // finally blocking. Note the pipe implementation rounds up the request to a power of 2. 3528 MonoPipe *monoPipe = new MonoPipe(mNormalFrameCount * 4, format, true /*writeCanBlock*/); 3529 const NBAIO_Format offers[1] = {format}; 3530 size_t numCounterOffers = 0; 3531#if !LOG_NDEBUG 3532 ssize_t index = 3533#else 3534 (void) 3535#endif 3536 monoPipe->negotiate(offers, 1, NULL, numCounterOffers); 3537 ALOG_ASSERT(index == 0); 3538 monoPipe->setAvgFrames((mScreenState & 1) ? 3539 (monoPipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 3540 mPipeSink = monoPipe; 3541 3542#ifdef TEE_SINK 3543 if (mTeeSinkOutputEnabled) { 3544 // create a Pipe to archive a copy of FastMixer's output for dumpsys 3545 Pipe *teeSink = new Pipe(mTeeSinkOutputFrames, origformat); 3546 const NBAIO_Format offers2[1] = {origformat}; 3547 numCounterOffers = 0; 3548 index = teeSink->negotiate(offers2, 1, NULL, numCounterOffers); 3549 ALOG_ASSERT(index == 0); 3550 mTeeSink = teeSink; 3551 PipeReader *teeSource = new PipeReader(*teeSink); 3552 numCounterOffers = 0; 3553 index = teeSource->negotiate(offers2, 1, NULL, numCounterOffers); 3554 ALOG_ASSERT(index == 0); 3555 mTeeSource = teeSource; 3556 } 3557#endif 3558 3559 // create fast mixer and configure it initially with just one fast track for our submix 3560 mFastMixer = new FastMixer(); 3561 FastMixerStateQueue *sq = mFastMixer->sq(); 3562#ifdef STATE_QUEUE_DUMP 3563 sq->setObserverDump(&mStateQueueObserverDump); 3564 sq->setMutatorDump(&mStateQueueMutatorDump); 3565#endif 3566 FastMixerState *state = sq->begin(); 3567 FastTrack *fastTrack = &state->mFastTracks[0]; 3568 // wrap the source side of the MonoPipe to make it an AudioBufferProvider 3569 fastTrack->mBufferProvider = new SourceAudioBufferProvider(new MonoPipeReader(monoPipe)); 3570 fastTrack->mVolumeProvider = NULL; 3571 fastTrack->mChannelMask = mChannelMask; // mPipeSink channel mask for audio to FastMixer 3572 fastTrack->mFormat = mFormat; // mPipeSink format for audio to FastMixer 3573 fastTrack->mGeneration++; 3574 state->mFastTracksGen++; 3575 state->mTrackMask = 1; 3576 // fast mixer will use the HAL output sink 3577 state->mOutputSink = mOutputSink.get(); 3578 state->mOutputSinkGen++; 3579 state->mFrameCount = mFrameCount; 3580 state->mCommand = FastMixerState::COLD_IDLE; 3581 // already done in constructor initialization list 3582 //mFastMixerFutex = 0; 3583 state->mColdFutexAddr = &mFastMixerFutex; 3584 state->mColdGen++; 3585 state->mDumpState = &mFastMixerDumpState; 3586#ifdef TEE_SINK 3587 state->mTeeSink = mTeeSink.get(); 3588#endif 3589 mFastMixerNBLogWriter = audioFlinger->newWriter_l(kFastMixerLogSize, "FastMixer"); 3590 state->mNBLogWriter = mFastMixerNBLogWriter.get(); 3591 sq->end(); 3592 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3593 3594 // start the fast mixer 3595 mFastMixer->run("FastMixer", PRIORITY_URGENT_AUDIO); 3596 pid_t tid = mFastMixer->getTid(); 3597 sendPrioConfigEvent(getpid_cached, tid, kPriorityFastMixer); 3598 3599#ifdef AUDIO_WATCHDOG 3600 // create and start the watchdog 3601 mAudioWatchdog = new AudioWatchdog(); 3602 mAudioWatchdog->setDump(&mAudioWatchdogDump); 3603 mAudioWatchdog->run("AudioWatchdog", PRIORITY_URGENT_AUDIO); 3604 tid = mAudioWatchdog->getTid(); 3605 sendPrioConfigEvent(getpid_cached, tid, kPriorityFastMixer); 3606#endif 3607 3608 } 3609 3610 switch (kUseFastMixer) { 3611 case FastMixer_Never: 3612 case FastMixer_Dynamic: 3613 mNormalSink = mOutputSink; 3614 break; 3615 case FastMixer_Always: 3616 mNormalSink = mPipeSink; 3617 break; 3618 case FastMixer_Static: 3619 mNormalSink = initFastMixer ? mPipeSink : mOutputSink; 3620 break; 3621 } 3622} 3623 3624AudioFlinger::MixerThread::~MixerThread() 3625{ 3626 if (mFastMixer != 0) { 3627 FastMixerStateQueue *sq = mFastMixer->sq(); 3628 FastMixerState *state = sq->begin(); 3629 if (state->mCommand == FastMixerState::COLD_IDLE) { 3630 int32_t old = android_atomic_inc(&mFastMixerFutex); 3631 if (old == -1) { 3632 (void) syscall(__NR_futex, &mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 3633 } 3634 } 3635 state->mCommand = FastMixerState::EXIT; 3636 sq->end(); 3637 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3638 mFastMixer->join(); 3639 // Though the fast mixer thread has exited, it's state queue is still valid. 3640 // We'll use that extract the final state which contains one remaining fast track 3641 // corresponding to our sub-mix. 3642 state = sq->begin(); 3643 ALOG_ASSERT(state->mTrackMask == 1); 3644 FastTrack *fastTrack = &state->mFastTracks[0]; 3645 ALOG_ASSERT(fastTrack->mBufferProvider != NULL); 3646 delete fastTrack->mBufferProvider; 3647 sq->end(false /*didModify*/); 3648 mFastMixer.clear(); 3649#ifdef AUDIO_WATCHDOG 3650 if (mAudioWatchdog != 0) { 3651 mAudioWatchdog->requestExit(); 3652 mAudioWatchdog->requestExitAndWait(); 3653 mAudioWatchdog.clear(); 3654 } 3655#endif 3656 } 3657 mAudioFlinger->unregisterWriter(mFastMixerNBLogWriter); 3658 delete mAudioMixer; 3659} 3660 3661 3662uint32_t AudioFlinger::MixerThread::correctLatency_l(uint32_t latency) const 3663{ 3664 if (mFastMixer != 0) { 3665 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 3666 latency += (pipe->getAvgFrames() * 1000) / mSampleRate; 3667 } 3668 return latency; 3669} 3670 3671 3672void AudioFlinger::MixerThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) 3673{ 3674 PlaybackThread::threadLoop_removeTracks(tracksToRemove); 3675} 3676 3677ssize_t AudioFlinger::MixerThread::threadLoop_write() 3678{ 3679 // FIXME we should only do one push per cycle; confirm this is true 3680 // Start the fast mixer if it's not already running 3681 if (mFastMixer != 0) { 3682 FastMixerStateQueue *sq = mFastMixer->sq(); 3683 FastMixerState *state = sq->begin(); 3684 if (state->mCommand != FastMixerState::MIX_WRITE && 3685 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)) { 3686 if (state->mCommand == FastMixerState::COLD_IDLE) { 3687 3688 // FIXME workaround for first HAL write being CPU bound on some devices 3689 ATRACE_BEGIN("write"); 3690 mOutput->write((char *)mSinkBuffer, 0); 3691 ATRACE_END(); 3692 3693 int32_t old = android_atomic_inc(&mFastMixerFutex); 3694 if (old == -1) { 3695 (void) syscall(__NR_futex, &mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 3696 } 3697#ifdef AUDIO_WATCHDOG 3698 if (mAudioWatchdog != 0) { 3699 mAudioWatchdog->resume(); 3700 } 3701#endif 3702 } 3703 state->mCommand = FastMixerState::MIX_WRITE; 3704#ifdef FAST_THREAD_STATISTICS 3705 mFastMixerDumpState.increaseSamplingN(mAudioFlinger->isLowRamDevice() ? 3706 FastThreadDumpState::kSamplingNforLowRamDevice : FastThreadDumpState::kSamplingN); 3707#endif 3708 sq->end(); 3709 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3710 if (kUseFastMixer == FastMixer_Dynamic) { 3711 mNormalSink = mPipeSink; 3712 } 3713 } else { 3714 sq->end(false /*didModify*/); 3715 } 3716 } 3717 return PlaybackThread::threadLoop_write(); 3718} 3719 3720void AudioFlinger::MixerThread::threadLoop_standby() 3721{ 3722 // Idle the fast mixer if it's currently running 3723 if (mFastMixer != 0) { 3724 FastMixerStateQueue *sq = mFastMixer->sq(); 3725 FastMixerState *state = sq->begin(); 3726 if (!(state->mCommand & FastMixerState::IDLE)) { 3727 state->mCommand = FastMixerState::COLD_IDLE; 3728 state->mColdFutexAddr = &mFastMixerFutex; 3729 state->mColdGen++; 3730 mFastMixerFutex = 0; 3731 sq->end(); 3732 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 3733 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3734 if (kUseFastMixer == FastMixer_Dynamic) { 3735 mNormalSink = mOutputSink; 3736 } 3737#ifdef AUDIO_WATCHDOG 3738 if (mAudioWatchdog != 0) { 3739 mAudioWatchdog->pause(); 3740 } 3741#endif 3742 } else { 3743 sq->end(false /*didModify*/); 3744 } 3745 } 3746 PlaybackThread::threadLoop_standby(); 3747} 3748 3749bool AudioFlinger::PlaybackThread::waitingAsyncCallback_l() 3750{ 3751 return false; 3752} 3753 3754bool AudioFlinger::PlaybackThread::shouldStandby_l() 3755{ 3756 return !mStandby; 3757} 3758 3759bool AudioFlinger::PlaybackThread::waitingAsyncCallback() 3760{ 3761 Mutex::Autolock _l(mLock); 3762 return waitingAsyncCallback_l(); 3763} 3764 3765// shared by MIXER and DIRECT, overridden by DUPLICATING 3766void AudioFlinger::PlaybackThread::threadLoop_standby() 3767{ 3768 ALOGV("Audio hardware entering standby, mixer %p, suspend count %d", this, mSuspended); 3769 mOutput->standby(); 3770 if (mUseAsyncWrite != 0) { 3771 // discard any pending drain or write ack by incrementing sequence 3772 mWriteAckSequence = (mWriteAckSequence + 2) & ~1; 3773 mDrainSequence = (mDrainSequence + 2) & ~1; 3774 ALOG_ASSERT(mCallbackThread != 0); 3775 mCallbackThread->setWriteBlocked(mWriteAckSequence); 3776 mCallbackThread->setDraining(mDrainSequence); 3777 } 3778 mHwPaused = false; 3779} 3780 3781void AudioFlinger::PlaybackThread::onAddNewTrack_l() 3782{ 3783 ALOGV("signal playback thread"); 3784 broadcast_l(); 3785} 3786 3787void AudioFlinger::MixerThread::threadLoop_mix() 3788{ 3789 // mix buffers... 3790 mAudioMixer->process(); 3791 mCurrentWriteLength = mSinkBufferSize; 3792 // increase sleep time progressively when application underrun condition clears. 3793 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 3794 // that a steady state of alternating ready/not ready conditions keeps the sleep time 3795 // such that we would underrun the audio HAL. 3796 if ((mSleepTimeUs == 0) && (sleepTimeShift > 0)) { 3797 sleepTimeShift--; 3798 } 3799 mSleepTimeUs = 0; 3800 mStandbyTimeNs = systemTime() + mStandbyDelayNs; 3801 //TODO: delay standby when effects have a tail 3802 3803} 3804 3805void AudioFlinger::MixerThread::threadLoop_sleepTime() 3806{ 3807 // If no tracks are ready, sleep once for the duration of an output 3808 // buffer size, then write 0s to the output 3809 if (mSleepTimeUs == 0) { 3810 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3811 mSleepTimeUs = mActiveSleepTimeUs >> sleepTimeShift; 3812 if (mSleepTimeUs < kMinThreadSleepTimeUs) { 3813 mSleepTimeUs = kMinThreadSleepTimeUs; 3814 } 3815 // reduce sleep time in case of consecutive application underruns to avoid 3816 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 3817 // duration we would end up writing less data than needed by the audio HAL if 3818 // the condition persists. 3819 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 3820 sleepTimeShift++; 3821 } 3822 } else { 3823 mSleepTimeUs = mIdleSleepTimeUs; 3824 } 3825 } else if (mBytesWritten != 0 || (mMixerStatus == MIXER_TRACKS_ENABLED)) { 3826 // clear out mMixerBuffer or mSinkBuffer, to ensure buffers are cleared 3827 // before effects processing or output. 3828 if (mMixerBufferValid) { 3829 memset(mMixerBuffer, 0, mMixerBufferSize); 3830 } else { 3831 memset(mSinkBuffer, 0, mSinkBufferSize); 3832 } 3833 mSleepTimeUs = 0; 3834 ALOGV_IF(mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED), 3835 "anticipated start"); 3836 } 3837 // TODO add standby time extension fct of effect tail 3838} 3839 3840// prepareTracks_l() must be called with ThreadBase::mLock held 3841AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 3842 Vector< sp<Track> > *tracksToRemove) 3843{ 3844 3845 mixer_state mixerStatus = MIXER_IDLE; 3846 // find out which tracks need to be processed 3847 size_t count = mActiveTracks.size(); 3848 size_t mixedTracks = 0; 3849 size_t tracksWithEffect = 0; 3850 // counts only _active_ fast tracks 3851 size_t fastTracks = 0; 3852 uint32_t resetMask = 0; // bit mask of fast tracks that need to be reset 3853 3854 float masterVolume = mMasterVolume; 3855 bool masterMute = mMasterMute; 3856 3857 if (masterMute) { 3858 masterVolume = 0; 3859 } 3860 // Delegate master volume control to effect in output mix effect chain if needed 3861 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 3862 if (chain != 0) { 3863 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 3864 chain->setVolume_l(&v, &v); 3865 masterVolume = (float)((v + (1 << 23)) >> 24); 3866 chain.clear(); 3867 } 3868 3869 // prepare a new state to push 3870 FastMixerStateQueue *sq = NULL; 3871 FastMixerState *state = NULL; 3872 bool didModify = false; 3873 FastMixerStateQueue::block_t block = FastMixerStateQueue::BLOCK_UNTIL_PUSHED; 3874 if (mFastMixer != 0) { 3875 sq = mFastMixer->sq(); 3876 state = sq->begin(); 3877 } 3878 3879 mMixerBufferValid = false; // mMixerBuffer has no valid data until appropriate tracks found. 3880 mEffectBufferValid = false; // mEffectBuffer has no valid data until tracks found. 3881 3882 for (size_t i=0 ; i<count ; i++) { 3883 const sp<Track> t = mActiveTracks[i].promote(); 3884 if (t == 0) { 3885 continue; 3886 } 3887 3888 // this const just means the local variable doesn't change 3889 Track* const track = t.get(); 3890 3891 // process fast tracks 3892 if (track->isFastTrack()) { 3893 3894 // It's theoretically possible (though unlikely) for a fast track to be created 3895 // and then removed within the same normal mix cycle. This is not a problem, as 3896 // the track never becomes active so it's fast mixer slot is never touched. 3897 // The converse, of removing an (active) track and then creating a new track 3898 // at the identical fast mixer slot within the same normal mix cycle, 3899 // is impossible because the slot isn't marked available until the end of each cycle. 3900 int j = track->mFastIndex; 3901 ALOG_ASSERT(0 < j && j < (int)FastMixerState::kMaxFastTracks); 3902 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << j))); 3903 FastTrack *fastTrack = &state->mFastTracks[j]; 3904 3905 // Determine whether the track is currently in underrun condition, 3906 // and whether it had a recent underrun. 3907 FastTrackDump *ftDump = &mFastMixerDumpState.mTracks[j]; 3908 FastTrackUnderruns underruns = ftDump->mUnderruns; 3909 uint32_t recentFull = (underruns.mBitFields.mFull - 3910 track->mObservedUnderruns.mBitFields.mFull) & UNDERRUN_MASK; 3911 uint32_t recentPartial = (underruns.mBitFields.mPartial - 3912 track->mObservedUnderruns.mBitFields.mPartial) & UNDERRUN_MASK; 3913 uint32_t recentEmpty = (underruns.mBitFields.mEmpty - 3914 track->mObservedUnderruns.mBitFields.mEmpty) & UNDERRUN_MASK; 3915 uint32_t recentUnderruns = recentPartial + recentEmpty; 3916 track->mObservedUnderruns = underruns; 3917 // don't count underruns that occur while stopping or pausing 3918 // or stopped which can occur when flush() is called while active 3919 if (!(track->isStopping() || track->isPausing() || track->isStopped()) && 3920 recentUnderruns > 0) { 3921 // FIXME fast mixer will pull & mix partial buffers, but we count as a full underrun 3922 track->mAudioTrackServerProxy->tallyUnderrunFrames(recentUnderruns * mFrameCount); 3923 } else { 3924 track->mAudioTrackServerProxy->tallyUnderrunFrames(0); 3925 } 3926 3927 // This is similar to the state machine for normal tracks, 3928 // with a few modifications for fast tracks. 3929 bool isActive = true; 3930 switch (track->mState) { 3931 case TrackBase::STOPPING_1: 3932 // track stays active in STOPPING_1 state until first underrun 3933 if (recentUnderruns > 0 || track->isTerminated()) { 3934 track->mState = TrackBase::STOPPING_2; 3935 } 3936 break; 3937 case TrackBase::PAUSING: 3938 // ramp down is not yet implemented 3939 track->setPaused(); 3940 break; 3941 case TrackBase::RESUMING: 3942 // ramp up is not yet implemented 3943 track->mState = TrackBase::ACTIVE; 3944 break; 3945 case TrackBase::ACTIVE: 3946 if (recentFull > 0 || recentPartial > 0) { 3947 // track has provided at least some frames recently: reset retry count 3948 track->mRetryCount = kMaxTrackRetries; 3949 } 3950 if (recentUnderruns == 0) { 3951 // no recent underruns: stay active 3952 break; 3953 } 3954 // there has recently been an underrun of some kind 3955 if (track->sharedBuffer() == 0) { 3956 // were any of the recent underruns "empty" (no frames available)? 3957 if (recentEmpty == 0) { 3958 // no, then ignore the partial underruns as they are allowed indefinitely 3959 break; 3960 } 3961 // there has recently been an "empty" underrun: decrement the retry counter 3962 if (--(track->mRetryCount) > 0) { 3963 break; 3964 } 3965 // indicate to client process that the track was disabled because of underrun; 3966 // it will then automatically call start() when data is available 3967 track->disable(); 3968 // remove from active list, but state remains ACTIVE [confusing but true] 3969 isActive = false; 3970 break; 3971 } 3972 // fall through 3973 case TrackBase::STOPPING_2: 3974 case TrackBase::PAUSED: 3975 case TrackBase::STOPPED: 3976 case TrackBase::FLUSHED: // flush() while active 3977 // Check for presentation complete if track is inactive 3978 // We have consumed all the buffers of this track. 3979 // This would be incomplete if we auto-paused on underrun 3980 { 3981 size_t audioHALFrames = 3982 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 3983 int64_t framesWritten = mBytesWritten / mFrameSize; 3984 if (!(mStandby || track->presentationComplete(framesWritten, audioHALFrames))) { 3985 // track stays in active list until presentation is complete 3986 break; 3987 } 3988 } 3989 if (track->isStopping_2()) { 3990 track->mState = TrackBase::STOPPED; 3991 } 3992 if (track->isStopped()) { 3993 // Can't reset directly, as fast mixer is still polling this track 3994 // track->reset(); 3995 // So instead mark this track as needing to be reset after push with ack 3996 resetMask |= 1 << i; 3997 } 3998 isActive = false; 3999 break; 4000 case TrackBase::IDLE: 4001 default: 4002 LOG_ALWAYS_FATAL("unexpected track state %d", track->mState); 4003 } 4004 4005 if (isActive) { 4006 // was it previously inactive? 4007 if (!(state->mTrackMask & (1 << j))) { 4008 ExtendedAudioBufferProvider *eabp = track; 4009 VolumeProvider *vp = track; 4010 fastTrack->mBufferProvider = eabp; 4011 fastTrack->mVolumeProvider = vp; 4012 fastTrack->mChannelMask = track->mChannelMask; 4013 fastTrack->mFormat = track->mFormat; 4014 fastTrack->mGeneration++; 4015 state->mTrackMask |= 1 << j; 4016 didModify = true; 4017 // no acknowledgement required for newly active tracks 4018 } 4019 // cache the combined master volume and stream type volume for fast mixer; this 4020 // lacks any synchronization or barrier so VolumeProvider may read a stale value 4021 track->mCachedVolume = masterVolume * mStreamTypes[track->streamType()].volume; 4022 ++fastTracks; 4023 } else { 4024 // was it previously active? 4025 if (state->mTrackMask & (1 << j)) { 4026 fastTrack->mBufferProvider = NULL; 4027 fastTrack->mGeneration++; 4028 state->mTrackMask &= ~(1 << j); 4029 didModify = true; 4030 // If any fast tracks were removed, we must wait for acknowledgement 4031 // because we're about to decrement the last sp<> on those tracks. 4032 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 4033 } else { 4034 LOG_ALWAYS_FATAL("fast track %d should have been active; " 4035 "mState=%d, mTrackMask=%#x, recentUnderruns=%u, isShared=%d", 4036 j, track->mState, state->mTrackMask, recentUnderruns, 4037 track->sharedBuffer() != 0); 4038 } 4039 tracksToRemove->add(track); 4040 // Avoids a misleading display in dumpsys 4041 track->mObservedUnderruns.mBitFields.mMostRecent = UNDERRUN_FULL; 4042 } 4043 continue; 4044 } 4045 4046 { // local variable scope to avoid goto warning 4047 4048 audio_track_cblk_t* cblk = track->cblk(); 4049 4050 // The first time a track is added we wait 4051 // for all its buffers to be filled before processing it 4052 int name = track->name(); 4053 // make sure that we have enough frames to mix one full buffer. 4054 // enforce this condition only once to enable draining the buffer in case the client 4055 // app does not call stop() and relies on underrun to stop: 4056 // hence the test on (mMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 4057 // during last round 4058 size_t desiredFrames; 4059 const uint32_t sampleRate = track->mAudioTrackServerProxy->getSampleRate(); 4060 AudioPlaybackRate playbackRate = track->mAudioTrackServerProxy->getPlaybackRate(); 4061 4062 desiredFrames = sourceFramesNeededWithTimestretch( 4063 sampleRate, mNormalFrameCount, mSampleRate, playbackRate.mSpeed); 4064 // TODO: ONLY USED FOR LEGACY RESAMPLERS, remove when they are removed. 4065 // add frames already consumed but not yet released by the resampler 4066 // because mAudioTrackServerProxy->framesReady() will include these frames 4067 desiredFrames += mAudioMixer->getUnreleasedFrames(track->name()); 4068 4069 uint32_t minFrames = 1; 4070 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() && 4071 (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) { 4072 minFrames = desiredFrames; 4073 } 4074 4075 size_t framesReady = track->framesReady(); 4076 if (ATRACE_ENABLED()) { 4077 // I wish we had formatted trace names 4078 char traceName[16]; 4079 strcpy(traceName, "nRdy"); 4080 int name = track->name(); 4081 if (AudioMixer::TRACK0 <= name && 4082 name < (int) (AudioMixer::TRACK0 + AudioMixer::MAX_NUM_TRACKS)) { 4083 name -= AudioMixer::TRACK0; 4084 traceName[4] = (name / 10) + '0'; 4085 traceName[5] = (name % 10) + '0'; 4086 } else { 4087 traceName[4] = '?'; 4088 traceName[5] = '?'; 4089 } 4090 traceName[6] = '\0'; 4091 ATRACE_INT(traceName, framesReady); 4092 } 4093 if ((framesReady >= minFrames) && track->isReady() && 4094 !track->isPaused() && !track->isTerminated()) 4095 { 4096 ALOGVV("track %d s=%08x [OK] on thread %p", name, cblk->mServer, this); 4097 4098 mixedTracks++; 4099 4100 // track->mainBuffer() != mSinkBuffer or mMixerBuffer means 4101 // there is an effect chain connected to the track 4102 chain.clear(); 4103 if (track->mainBuffer() != mSinkBuffer && 4104 track->mainBuffer() != mMixerBuffer) { 4105 if (mEffectBufferEnabled) { 4106 mEffectBufferValid = true; // Later can set directly. 4107 } 4108 chain = getEffectChain_l(track->sessionId()); 4109 // Delegate volume control to effect in track effect chain if needed 4110 if (chain != 0) { 4111 tracksWithEffect++; 4112 } else { 4113 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on " 4114 "session %d", 4115 name, track->sessionId()); 4116 } 4117 } 4118 4119 4120 int param = AudioMixer::VOLUME; 4121 if (track->mFillingUpStatus == Track::FS_FILLED) { 4122 // no ramp for the first volume setting 4123 track->mFillingUpStatus = Track::FS_ACTIVE; 4124 if (track->mState == TrackBase::RESUMING) { 4125 track->mState = TrackBase::ACTIVE; 4126 param = AudioMixer::RAMP_VOLUME; 4127 } 4128 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 4129 // FIXME should not make a decision based on mServer 4130 } else if (cblk->mServer != 0) { 4131 // If the track is stopped before the first frame was mixed, 4132 // do not apply ramp 4133 param = AudioMixer::RAMP_VOLUME; 4134 } 4135 4136 // compute volume for this track 4137 uint32_t vl, vr; // in U8.24 integer format 4138 float vlf, vrf, vaf; // in [0.0, 1.0] float format 4139 if (track->isPausing() || mStreamTypes[track->streamType()].mute) { 4140 vl = vr = 0; 4141 vlf = vrf = vaf = 0.; 4142 if (track->isPausing()) { 4143 track->setPaused(); 4144 } 4145 } else { 4146 4147 // read original volumes with volume control 4148 float typeVolume = mStreamTypes[track->streamType()].volume; 4149 float v = masterVolume * typeVolume; 4150 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 4151 gain_minifloat_packed_t vlr = proxy->getVolumeLR(); 4152 vlf = float_from_gain(gain_minifloat_unpack_left(vlr)); 4153 vrf = float_from_gain(gain_minifloat_unpack_right(vlr)); 4154 // track volumes come from shared memory, so can't be trusted and must be clamped 4155 if (vlf > GAIN_FLOAT_UNITY) { 4156 ALOGV("Track left volume out of range: %.3g", vlf); 4157 vlf = GAIN_FLOAT_UNITY; 4158 } 4159 if (vrf > GAIN_FLOAT_UNITY) { 4160 ALOGV("Track right volume out of range: %.3g", vrf); 4161 vrf = GAIN_FLOAT_UNITY; 4162 } 4163 // now apply the master volume and stream type volume 4164 vlf *= v; 4165 vrf *= v; 4166 // assuming master volume and stream type volume each go up to 1.0, 4167 // then derive vl and vr as U8.24 versions for the effect chain 4168 const float scaleto8_24 = MAX_GAIN_INT * MAX_GAIN_INT; 4169 vl = (uint32_t) (scaleto8_24 * vlf); 4170 vr = (uint32_t) (scaleto8_24 * vrf); 4171 // vl and vr are now in U8.24 format 4172 uint16_t sendLevel = proxy->getSendLevel_U4_12(); 4173 // send level comes from shared memory and so may be corrupt 4174 if (sendLevel > MAX_GAIN_INT) { 4175 ALOGV("Track send level out of range: %04X", sendLevel); 4176 sendLevel = MAX_GAIN_INT; 4177 } 4178 // vaf is represented as [0.0, 1.0] float by rescaling sendLevel 4179 vaf = v * sendLevel * (1. / MAX_GAIN_INT); 4180 } 4181 4182 // Delegate volume control to effect in track effect chain if needed 4183 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 4184 // Do not ramp volume if volume is controlled by effect 4185 param = AudioMixer::VOLUME; 4186 // Update remaining floating point volume levels 4187 vlf = (float)vl / (1 << 24); 4188 vrf = (float)vr / (1 << 24); 4189 track->mHasVolumeController = true; 4190 } else { 4191 // force no volume ramp when volume controller was just disabled or removed 4192 // from effect chain to avoid volume spike 4193 if (track->mHasVolumeController) { 4194 param = AudioMixer::VOLUME; 4195 } 4196 track->mHasVolumeController = false; 4197 } 4198 4199 // XXX: these things DON'T need to be done each time 4200 mAudioMixer->setBufferProvider(name, track); 4201 mAudioMixer->enable(name); 4202 4203 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, &vlf); 4204 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, &vrf); 4205 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, &vaf); 4206 mAudioMixer->setParameter( 4207 name, 4208 AudioMixer::TRACK, 4209 AudioMixer::FORMAT, (void *)track->format()); 4210 mAudioMixer->setParameter( 4211 name, 4212 AudioMixer::TRACK, 4213 AudioMixer::CHANNEL_MASK, (void *)(uintptr_t)track->channelMask()); 4214 mAudioMixer->setParameter( 4215 name, 4216 AudioMixer::TRACK, 4217 AudioMixer::MIXER_CHANNEL_MASK, (void *)(uintptr_t)mChannelMask); 4218 // limit track sample rate to 2 x output sample rate, which changes at re-configuration 4219 uint32_t maxSampleRate = mSampleRate * AUDIO_RESAMPLER_DOWN_RATIO_MAX; 4220 uint32_t reqSampleRate = track->mAudioTrackServerProxy->getSampleRate(); 4221 if (reqSampleRate == 0) { 4222 reqSampleRate = mSampleRate; 4223 } else if (reqSampleRate > maxSampleRate) { 4224 reqSampleRate = maxSampleRate; 4225 } 4226 mAudioMixer->setParameter( 4227 name, 4228 AudioMixer::RESAMPLE, 4229 AudioMixer::SAMPLE_RATE, 4230 (void *)(uintptr_t)reqSampleRate); 4231 4232 AudioPlaybackRate playbackRate = track->mAudioTrackServerProxy->getPlaybackRate(); 4233 mAudioMixer->setParameter( 4234 name, 4235 AudioMixer::TIMESTRETCH, 4236 AudioMixer::PLAYBACK_RATE, 4237 &playbackRate); 4238 4239 /* 4240 * Select the appropriate output buffer for the track. 4241 * 4242 * Tracks with effects go into their own effects chain buffer 4243 * and from there into either mEffectBuffer or mSinkBuffer. 4244 * 4245 * Other tracks can use mMixerBuffer for higher precision 4246 * channel accumulation. If this buffer is enabled 4247 * (mMixerBufferEnabled true), then selected tracks will accumulate 4248 * into it. 4249 * 4250 */ 4251 if (mMixerBufferEnabled 4252 && (track->mainBuffer() == mSinkBuffer 4253 || track->mainBuffer() == mMixerBuffer)) { 4254 mAudioMixer->setParameter( 4255 name, 4256 AudioMixer::TRACK, 4257 AudioMixer::MIXER_FORMAT, (void *)mMixerBufferFormat); 4258 mAudioMixer->setParameter( 4259 name, 4260 AudioMixer::TRACK, 4261 AudioMixer::MAIN_BUFFER, (void *)mMixerBuffer); 4262 // TODO: override track->mainBuffer()? 4263 mMixerBufferValid = true; 4264 } else { 4265 mAudioMixer->setParameter( 4266 name, 4267 AudioMixer::TRACK, 4268 AudioMixer::MIXER_FORMAT, (void *)AUDIO_FORMAT_PCM_16_BIT); 4269 mAudioMixer->setParameter( 4270 name, 4271 AudioMixer::TRACK, 4272 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 4273 } 4274 mAudioMixer->setParameter( 4275 name, 4276 AudioMixer::TRACK, 4277 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 4278 4279 // reset retry count 4280 track->mRetryCount = kMaxTrackRetries; 4281 4282 // If one track is ready, set the mixer ready if: 4283 // - the mixer was not ready during previous round OR 4284 // - no other track is not ready 4285 if (mMixerStatusIgnoringFastTracks != MIXER_TRACKS_READY || 4286 mixerStatus != MIXER_TRACKS_ENABLED) { 4287 mixerStatus = MIXER_TRACKS_READY; 4288 } 4289 } else { 4290 if (framesReady < desiredFrames && !track->isStopped() && !track->isPaused()) { 4291 ALOGV("track(%p) underrun, framesReady(%zu) < framesDesired(%zd)", 4292 track, framesReady, desiredFrames); 4293 track->mAudioTrackServerProxy->tallyUnderrunFrames(desiredFrames); 4294 } else { 4295 track->mAudioTrackServerProxy->tallyUnderrunFrames(0); 4296 } 4297 4298 // clear effect chain input buffer if an active track underruns to avoid sending 4299 // previous audio buffer again to effects 4300 chain = getEffectChain_l(track->sessionId()); 4301 if (chain != 0) { 4302 chain->clearInputBuffer(); 4303 } 4304 4305 ALOGVV("track %d s=%08x [NOT READY] on thread %p", name, cblk->mServer, this); 4306 if ((track->sharedBuffer() != 0) || track->isTerminated() || 4307 track->isStopped() || track->isPaused()) { 4308 // We have consumed all the buffers of this track. 4309 // Remove it from the list of active tracks. 4310 // TODO: use actual buffer filling status instead of latency when available from 4311 // audio HAL 4312 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 4313 int64_t framesWritten = mBytesWritten / mFrameSize; 4314 if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) { 4315 if (track->isStopped()) { 4316 track->reset(); 4317 } 4318 tracksToRemove->add(track); 4319 } 4320 } else { 4321 // No buffers for this track. Give it a few chances to 4322 // fill a buffer, then remove it from active list. 4323 if (--(track->mRetryCount) <= 0) { 4324 ALOGI("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 4325 tracksToRemove->add(track); 4326 // indicate to client process that the track was disabled because of underrun; 4327 // it will then automatically call start() when data is available 4328 track->disable(); 4329 // If one track is not ready, mark the mixer also not ready if: 4330 // - the mixer was ready during previous round OR 4331 // - no other track is ready 4332 } else if (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY || 4333 mixerStatus != MIXER_TRACKS_READY) { 4334 mixerStatus = MIXER_TRACKS_ENABLED; 4335 } 4336 } 4337 mAudioMixer->disable(name); 4338 } 4339 4340 } // local variable scope to avoid goto warning 4341 4342 } 4343 4344 // Push the new FastMixer state if necessary 4345 bool pauseAudioWatchdog = false; 4346 if (didModify) { 4347 state->mFastTracksGen++; 4348 // if the fast mixer was active, but now there are no fast tracks, then put it in cold idle 4349 if (kUseFastMixer == FastMixer_Dynamic && 4350 state->mCommand == FastMixerState::MIX_WRITE && state->mTrackMask <= 1) { 4351 state->mCommand = FastMixerState::COLD_IDLE; 4352 state->mColdFutexAddr = &mFastMixerFutex; 4353 state->mColdGen++; 4354 mFastMixerFutex = 0; 4355 if (kUseFastMixer == FastMixer_Dynamic) { 4356 mNormalSink = mOutputSink; 4357 } 4358 // If we go into cold idle, need to wait for acknowledgement 4359 // so that fast mixer stops doing I/O. 4360 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 4361 pauseAudioWatchdog = true; 4362 } 4363 } 4364 if (sq != NULL) { 4365 sq->end(didModify); 4366 sq->push(block); 4367 } 4368#ifdef AUDIO_WATCHDOG 4369 if (pauseAudioWatchdog && mAudioWatchdog != 0) { 4370 mAudioWatchdog->pause(); 4371 } 4372#endif 4373 4374 // Now perform the deferred reset on fast tracks that have stopped 4375 while (resetMask != 0) { 4376 size_t i = __builtin_ctz(resetMask); 4377 ALOG_ASSERT(i < count); 4378 resetMask &= ~(1 << i); 4379 sp<Track> t = mActiveTracks[i].promote(); 4380 if (t == 0) { 4381 continue; 4382 } 4383 Track* track = t.get(); 4384 ALOG_ASSERT(track->isFastTrack() && track->isStopped()); 4385 track->reset(); 4386 } 4387 4388 // remove all the tracks that need to be... 4389 removeTracks_l(*tracksToRemove); 4390 4391 if (getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX) != 0) { 4392 mEffectBufferValid = true; 4393 } 4394 4395 if (mEffectBufferValid) { 4396 // as long as there are effects we should clear the effects buffer, to avoid 4397 // passing a non-clean buffer to the effect chain 4398 memset(mEffectBuffer, 0, mEffectBufferSize); 4399 } 4400 // sink or mix buffer must be cleared if all tracks are connected to an 4401 // effect chain as in this case the mixer will not write to the sink or mix buffer 4402 // and track effects will accumulate into it 4403 if ((mBytesRemaining == 0) && ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 4404 (mixedTracks == 0 && fastTracks > 0))) { 4405 // FIXME as a performance optimization, should remember previous zero status 4406 if (mMixerBufferValid) { 4407 memset(mMixerBuffer, 0, mMixerBufferSize); 4408 // TODO: In testing, mSinkBuffer below need not be cleared because 4409 // the PlaybackThread::threadLoop() copies mMixerBuffer into mSinkBuffer 4410 // after mixing. 4411 // 4412 // To enforce this guarantee: 4413 // ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 4414 // (mixedTracks == 0 && fastTracks > 0)) 4415 // must imply MIXER_TRACKS_READY. 4416 // Later, we may clear buffers regardless, and skip much of this logic. 4417 } 4418 // FIXME as a performance optimization, should remember previous zero status 4419 memset(mSinkBuffer, 0, mNormalFrameCount * mFrameSize); 4420 } 4421 4422 // if any fast tracks, then status is ready 4423 mMixerStatusIgnoringFastTracks = mixerStatus; 4424 if (fastTracks > 0) { 4425 mixerStatus = MIXER_TRACKS_READY; 4426 } 4427 return mixerStatus; 4428} 4429 4430// getTrackName_l() must be called with ThreadBase::mLock held 4431int AudioFlinger::MixerThread::getTrackName_l(audio_channel_mask_t channelMask, 4432 audio_format_t format, audio_session_t sessionId) 4433{ 4434 return mAudioMixer->getTrackName(channelMask, format, sessionId); 4435} 4436 4437// deleteTrackName_l() must be called with ThreadBase::mLock held 4438void AudioFlinger::MixerThread::deleteTrackName_l(int name) 4439{ 4440 ALOGV("remove track (%d) and delete from mixer", name); 4441 mAudioMixer->deleteTrackName(name); 4442} 4443 4444// checkForNewParameter_l() must be called with ThreadBase::mLock held 4445bool AudioFlinger::MixerThread::checkForNewParameter_l(const String8& keyValuePair, 4446 status_t& status) 4447{ 4448 bool reconfig = false; 4449 bool a2dpDeviceChanged = false; 4450 4451 status = NO_ERROR; 4452 4453 // if !&IDLE, holds the FastMixer state to restore after new parameters processed 4454 FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE; 4455 if (mFastMixer != 0) { 4456 FastMixerStateQueue *sq = mFastMixer->sq(); 4457 FastMixerState *state = sq->begin(); 4458 if (!(state->mCommand & FastMixerState::IDLE)) { 4459 previousCommand = state->mCommand; 4460 state->mCommand = FastMixerState::HOT_IDLE; 4461 sq->end(); 4462 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 4463 } else { 4464 sq->end(false /*didModify*/); 4465 } 4466 } 4467 4468 AudioParameter param = AudioParameter(keyValuePair); 4469 int value; 4470 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 4471 reconfig = true; 4472 } 4473 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 4474 if (!isValidPcmSinkFormat((audio_format_t) value)) { 4475 status = BAD_VALUE; 4476 } else { 4477 // no need to save value, since it's constant 4478 reconfig = true; 4479 } 4480 } 4481 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 4482 if (!isValidPcmSinkChannelMask((audio_channel_mask_t) value)) { 4483 status = BAD_VALUE; 4484 } else { 4485 // no need to save value, since it's constant 4486 reconfig = true; 4487 } 4488 } 4489 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 4490 // do not accept frame count changes if tracks are open as the track buffer 4491 // size depends on frame count and correct behavior would not be guaranteed 4492 // if frame count is changed after track creation 4493 if (!mTracks.isEmpty()) { 4494 status = INVALID_OPERATION; 4495 } else { 4496 reconfig = true; 4497 } 4498 } 4499 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 4500#ifdef ADD_BATTERY_DATA 4501 // when changing the audio output device, call addBatteryData to notify 4502 // the change 4503 if (mOutDevice != value) { 4504 uint32_t params = 0; 4505 // check whether speaker is on 4506 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 4507 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 4508 } 4509 4510 audio_devices_t deviceWithoutSpeaker 4511 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 4512 // check if any other device (except speaker) is on 4513 if (value & deviceWithoutSpeaker) { 4514 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 4515 } 4516 4517 if (params != 0) { 4518 addBatteryData(params); 4519 } 4520 } 4521#endif 4522 4523 // forward device change to effects that have requested to be 4524 // aware of attached audio device. 4525 if (value != AUDIO_DEVICE_NONE) { 4526 a2dpDeviceChanged = 4527 (mOutDevice & AUDIO_DEVICE_OUT_ALL_A2DP) != (value & AUDIO_DEVICE_OUT_ALL_A2DP); 4528 mOutDevice = value; 4529 for (size_t i = 0; i < mEffectChains.size(); i++) { 4530 mEffectChains[i]->setDevice_l(mOutDevice); 4531 } 4532 } 4533 } 4534 4535 if (status == NO_ERROR) { 4536 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 4537 keyValuePair.string()); 4538 if (!mStandby && status == INVALID_OPERATION) { 4539 mOutput->standby(); 4540 mStandby = true; 4541 mBytesWritten = 0; 4542 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 4543 keyValuePair.string()); 4544 } 4545 if (status == NO_ERROR && reconfig) { 4546 readOutputParameters_l(); 4547 delete mAudioMixer; 4548 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 4549 for (size_t i = 0; i < mTracks.size() ; i++) { 4550 int name = getTrackName_l(mTracks[i]->mChannelMask, 4551 mTracks[i]->mFormat, mTracks[i]->mSessionId); 4552 if (name < 0) { 4553 break; 4554 } 4555 mTracks[i]->mName = name; 4556 } 4557 sendIoConfigEvent_l(AUDIO_OUTPUT_CONFIG_CHANGED); 4558 } 4559 } 4560 4561 if (!(previousCommand & FastMixerState::IDLE)) { 4562 ALOG_ASSERT(mFastMixer != 0); 4563 FastMixerStateQueue *sq = mFastMixer->sq(); 4564 FastMixerState *state = sq->begin(); 4565 ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE); 4566 state->mCommand = previousCommand; 4567 sq->end(); 4568 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 4569 } 4570 4571 return reconfig || a2dpDeviceChanged; 4572} 4573 4574 4575void AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 4576{ 4577 PlaybackThread::dumpInternals(fd, args); 4578 dprintf(fd, " Thread throttle time (msecs): %u\n", mThreadThrottleTimeMs); 4579 dprintf(fd, " AudioMixer tracks: 0x%08x\n", mAudioMixer->trackNames()); 4580 dprintf(fd, " Master mono: %s\n", mMasterMono ? "on" : "off"); 4581 4582 // Make a non-atomic copy of fast mixer dump state so it won't change underneath us 4583 // while we are dumping it. It may be inconsistent, but it won't mutate! 4584 // This is a large object so we place it on the heap. 4585 // FIXME 25972958: Need an intelligent copy constructor that does not touch unused pages. 4586 const FastMixerDumpState *copy = new FastMixerDumpState(mFastMixerDumpState); 4587 copy->dump(fd); 4588 delete copy; 4589 4590#ifdef STATE_QUEUE_DUMP 4591 // Similar for state queue 4592 StateQueueObserverDump observerCopy = mStateQueueObserverDump; 4593 observerCopy.dump(fd); 4594 StateQueueMutatorDump mutatorCopy = mStateQueueMutatorDump; 4595 mutatorCopy.dump(fd); 4596#endif 4597 4598#ifdef TEE_SINK 4599 // Write the tee output to a .wav file 4600 dumpTee(fd, mTeeSource, mId); 4601#endif 4602 4603#ifdef AUDIO_WATCHDOG 4604 if (mAudioWatchdog != 0) { 4605 // Make a non-atomic copy of audio watchdog dump so it won't change underneath us 4606 AudioWatchdogDump wdCopy = mAudioWatchdogDump; 4607 wdCopy.dump(fd); 4608 } 4609#endif 4610} 4611 4612uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const 4613{ 4614 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000) / 2; 4615} 4616 4617uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() const 4618{ 4619 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000); 4620} 4621 4622void AudioFlinger::MixerThread::cacheParameters_l() 4623{ 4624 PlaybackThread::cacheParameters_l(); 4625 4626 // FIXME: Relaxed timing because of a certain device that can't meet latency 4627 // Should be reduced to 2x after the vendor fixes the driver issue 4628 // increase threshold again due to low power audio mode. The way this warning 4629 // threshold is calculated and its usefulness should be reconsidered anyway. 4630 maxPeriod = seconds(mNormalFrameCount) / mSampleRate * 15; 4631} 4632 4633// ---------------------------------------------------------------------------- 4634 4635AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 4636 AudioStreamOut* output, audio_io_handle_t id, audio_devices_t device, bool systemReady, 4637 uint32_t bitRate) 4638 : PlaybackThread(audioFlinger, output, id, device, DIRECT, systemReady, bitRate) 4639 // mLeftVolFloat, mRightVolFloat 4640{ 4641} 4642 4643AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 4644 AudioStreamOut* output, audio_io_handle_t id, uint32_t device, 4645 ThreadBase::type_t type, bool systemReady, uint32_t bitRate) 4646 : PlaybackThread(audioFlinger, output, id, device, type, systemReady, bitRate) 4647 // mLeftVolFloat, mRightVolFloat 4648{ 4649} 4650 4651AudioFlinger::DirectOutputThread::~DirectOutputThread() 4652{ 4653} 4654 4655void AudioFlinger::DirectOutputThread::processVolume_l(Track *track, bool lastTrack) 4656{ 4657 float left, right; 4658 4659 if (mMasterMute || mStreamTypes[track->streamType()].mute) { 4660 left = right = 0; 4661 } else { 4662 float typeVolume = mStreamTypes[track->streamType()].volume; 4663 float v = mMasterVolume * typeVolume; 4664 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 4665 gain_minifloat_packed_t vlr = proxy->getVolumeLR(); 4666 left = float_from_gain(gain_minifloat_unpack_left(vlr)); 4667 if (left > GAIN_FLOAT_UNITY) { 4668 left = GAIN_FLOAT_UNITY; 4669 } 4670 left *= v; 4671 right = float_from_gain(gain_minifloat_unpack_right(vlr)); 4672 if (right > GAIN_FLOAT_UNITY) { 4673 right = GAIN_FLOAT_UNITY; 4674 } 4675 right *= v; 4676 } 4677 4678 if (lastTrack) { 4679 if (left != mLeftVolFloat || right != mRightVolFloat) { 4680 mLeftVolFloat = left; 4681 mRightVolFloat = right; 4682 4683 // Convert volumes from float to 8.24 4684 uint32_t vl = (uint32_t)(left * (1 << 24)); 4685 uint32_t vr = (uint32_t)(right * (1 << 24)); 4686 4687 // Delegate volume control to effect in track effect chain if needed 4688 // only one effect chain can be present on DirectOutputThread, so if 4689 // there is one, the track is connected to it 4690 if (!mEffectChains.isEmpty()) { 4691 mEffectChains[0]->setVolume_l(&vl, &vr); 4692 left = (float)vl / (1 << 24); 4693 right = (float)vr / (1 << 24); 4694 } 4695 if (mOutput->stream->set_volume) { 4696 mOutput->stream->set_volume(mOutput->stream, left, right); 4697 } 4698 } 4699 } 4700} 4701 4702void AudioFlinger::DirectOutputThread::onAddNewTrack_l() 4703{ 4704 sp<Track> previousTrack = mPreviousTrack.promote(); 4705 sp<Track> latestTrack = mLatestActiveTrack.promote(); 4706 4707 if (previousTrack != 0 && latestTrack != 0) { 4708 if (mType == DIRECT) { 4709 if (previousTrack.get() != latestTrack.get()) { 4710 mFlushPending = true; 4711 } 4712 } else /* mType == OFFLOAD */ { 4713 if (previousTrack->sessionId() != latestTrack->sessionId()) { 4714 mFlushPending = true; 4715 } 4716 } 4717 } 4718 PlaybackThread::onAddNewTrack_l(); 4719} 4720 4721AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l( 4722 Vector< sp<Track> > *tracksToRemove 4723) 4724{ 4725 size_t count = mActiveTracks.size(); 4726 mixer_state mixerStatus = MIXER_IDLE; 4727 bool doHwPause = false; 4728 bool doHwResume = false; 4729 4730 // find out which tracks need to be processed 4731 for (size_t i = 0; i < count; i++) { 4732 sp<Track> t = mActiveTracks[i].promote(); 4733 // The track died recently 4734 if (t == 0) { 4735 continue; 4736 } 4737 4738 if (t->isInvalid()) { 4739 ALOGW("An invalidated track shouldn't be in active list"); 4740 tracksToRemove->add(t); 4741 continue; 4742 } 4743 4744 Track* const track = t.get(); 4745#ifdef VERY_VERY_VERBOSE_LOGGING 4746 audio_track_cblk_t* cblk = track->cblk(); 4747#endif 4748 // Only consider last track started for volume and mixer state control. 4749 // In theory an older track could underrun and restart after the new one starts 4750 // but as we only care about the transition phase between two tracks on a 4751 // direct output, it is not a problem to ignore the underrun case. 4752 sp<Track> l = mLatestActiveTrack.promote(); 4753 bool last = l.get() == track; 4754 4755 if (track->isPausing()) { 4756 track->setPaused(); 4757 if (mHwSupportsPause && last && !mHwPaused) { 4758 doHwPause = true; 4759 mHwPaused = true; 4760 } 4761 tracksToRemove->add(track); 4762 } else if (track->isFlushPending()) { 4763 track->flushAck(); 4764 if (last) { 4765 mFlushPending = true; 4766 } 4767 } else if (track->isResumePending()) { 4768 track->resumeAck(); 4769 if (last && mHwPaused) { 4770 doHwResume = true; 4771 mHwPaused = false; 4772 } 4773 } 4774 4775 // The first time a track is added we wait 4776 // for all its buffers to be filled before processing it. 4777 // Allow draining the buffer in case the client 4778 // app does not call stop() and relies on underrun to stop: 4779 // hence the test on (track->mRetryCount > 1). 4780 // If retryCount<=1 then track is about to underrun and be removed. 4781 // Do not use a high threshold for compressed audio. 4782 uint32_t minFrames; 4783 if ((track->sharedBuffer() == 0) && !track->isStopping_1() && !track->isPausing() 4784 && (track->mRetryCount > 1) && audio_has_proportional_frames(mFormat)) { 4785 minFrames = mNormalFrameCount; 4786 } else { 4787 minFrames = 1; 4788 } 4789 4790 if ((track->framesReady() >= minFrames) && track->isReady() && !track->isPaused() && 4791 !track->isStopping_2() && !track->isStopped()) 4792 { 4793 ALOGVV("track %d s=%08x [OK]", track->name(), cblk->mServer); 4794 4795 if (track->mFillingUpStatus == Track::FS_FILLED) { 4796 track->mFillingUpStatus = Track::FS_ACTIVE; 4797 // make sure processVolume_l() will apply new volume even if 0 4798 mLeftVolFloat = mRightVolFloat = -1.0; 4799 if (!mHwSupportsPause) { 4800 track->resumeAck(); 4801 } 4802 } 4803 4804 // compute volume for this track 4805 processVolume_l(track, last); 4806 if (last) { 4807 sp<Track> previousTrack = mPreviousTrack.promote(); 4808 if (previousTrack != 0) { 4809 if (track != previousTrack.get()) { 4810 // Flush any data still being written from last track 4811 mBytesRemaining = 0; 4812 // Invalidate previous track to force a seek when resuming. 4813 previousTrack->invalidate(); 4814 } 4815 } 4816 mPreviousTrack = track; 4817 4818 // reset retry count 4819 track->mRetryCount = kMaxTrackRetriesDirect; 4820 mActiveTrack = t; 4821 mixerStatus = MIXER_TRACKS_READY; 4822 if (mHwPaused) { 4823 doHwResume = true; 4824 mHwPaused = false; 4825 } 4826 } 4827 } else { 4828 // clear effect chain input buffer if the last active track started underruns 4829 // to avoid sending previous audio buffer again to effects 4830 if (!mEffectChains.isEmpty() && last) { 4831 mEffectChains[0]->clearInputBuffer(); 4832 } 4833 if (track->isStopping_1()) { 4834 track->mState = TrackBase::STOPPING_2; 4835 if (last && mHwPaused) { 4836 doHwResume = true; 4837 mHwPaused = false; 4838 } 4839 } 4840 if ((track->sharedBuffer() != 0) || track->isStopped() || 4841 track->isStopping_2() || track->isPaused()) { 4842 // We have consumed all the buffers of this track. 4843 // Remove it from the list of active tracks. 4844 size_t audioHALFrames; 4845 if (audio_has_proportional_frames(mFormat)) { 4846 audioHALFrames = (latency_l() * mSampleRate) / 1000; 4847 } else { 4848 audioHALFrames = 0; 4849 } 4850 4851 int64_t framesWritten = mBytesWritten / mFrameSize; 4852 if (mStandby || !last || 4853 track->presentationComplete(framesWritten, audioHALFrames)) { 4854 if (track->isStopping_2()) { 4855 track->mState = TrackBase::STOPPED; 4856 } 4857 if (track->isStopped()) { 4858 track->reset(); 4859 } 4860 tracksToRemove->add(track); 4861 } 4862 } else { 4863 // No buffers for this track. Give it a few chances to 4864 // fill a buffer, then remove it from active list. 4865 // Only consider last track started for mixer state control 4866 if (--(track->mRetryCount) <= 0) { 4867 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 4868 tracksToRemove->add(track); 4869 // indicate to client process that the track was disabled because of underrun; 4870 // it will then automatically call start() when data is available 4871 track->disable(); 4872 } else if (last) { 4873 ALOGW("pause because of UNDERRUN, framesReady = %zu," 4874 "minFrames = %u, mFormat = %#x", 4875 track->framesReady(), minFrames, mFormat); 4876 mixerStatus = MIXER_TRACKS_ENABLED; 4877 if (mHwSupportsPause && !mHwPaused && !mStandby) { 4878 doHwPause = true; 4879 mHwPaused = true; 4880 } 4881 } 4882 } 4883 } 4884 } 4885 4886 // if an active track did not command a flush, check for pending flush on stopped tracks 4887 if (!mFlushPending) { 4888 for (size_t i = 0; i < mTracks.size(); i++) { 4889 if (mTracks[i]->isFlushPending()) { 4890 mTracks[i]->flushAck(); 4891 mFlushPending = true; 4892 } 4893 } 4894 } 4895 4896 // make sure the pause/flush/resume sequence is executed in the right order. 4897 // If a flush is pending and a track is active but the HW is not paused, force a HW pause 4898 // before flush and then resume HW. This can happen in case of pause/flush/resume 4899 // if resume is received before pause is executed. 4900 if (mHwSupportsPause && !mStandby && 4901 (doHwPause || (mFlushPending && !mHwPaused && (count != 0)))) { 4902 mOutput->stream->pause(mOutput->stream); 4903 } 4904 if (mFlushPending) { 4905 flushHw_l(); 4906 } 4907 if (mHwSupportsPause && !mStandby && doHwResume) { 4908 mOutput->stream->resume(mOutput->stream); 4909 } 4910 // remove all the tracks that need to be... 4911 removeTracks_l(*tracksToRemove); 4912 4913 return mixerStatus; 4914} 4915 4916void AudioFlinger::DirectOutputThread::threadLoop_mix() 4917{ 4918 size_t frameCount = mFrameCount; 4919 int8_t *curBuf = (int8_t *)mSinkBuffer; 4920 // output audio to hardware 4921 while (frameCount) { 4922 AudioBufferProvider::Buffer buffer; 4923 buffer.frameCount = frameCount; 4924 status_t status = mActiveTrack->getNextBuffer(&buffer); 4925 if (status != NO_ERROR || buffer.raw == NULL) { 4926 // no need to pad with 0 for compressed audio 4927 if (audio_has_proportional_frames(mFormat)) { 4928 memset(curBuf, 0, frameCount * mFrameSize); 4929 } 4930 break; 4931 } 4932 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 4933 frameCount -= buffer.frameCount; 4934 curBuf += buffer.frameCount * mFrameSize; 4935 mActiveTrack->releaseBuffer(&buffer); 4936 } 4937 mCurrentWriteLength = curBuf - (int8_t *)mSinkBuffer; 4938 mSleepTimeUs = 0; 4939 mStandbyTimeNs = systemTime() + mStandbyDelayNs; 4940 mActiveTrack.clear(); 4941} 4942 4943void AudioFlinger::DirectOutputThread::threadLoop_sleepTime() 4944{ 4945 // do not write to HAL when paused 4946 if (mHwPaused || (usesHwAvSync() && mStandby)) { 4947 mSleepTimeUs = mIdleSleepTimeUs; 4948 return; 4949 } 4950 if (mSleepTimeUs == 0) { 4951 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4952 // For compressed offload, use faster sleep time when underruning until more than an 4953 // entire buffer was written to the audio HAL 4954 if (!audio_has_proportional_frames(mFormat) && 4955 (mType == OFFLOAD) && (mBytesWritten < (int64_t) mBufferSize)) { 4956 mSleepTimeUs = kDirectMinSleepTimeUs; 4957 } else { 4958 mSleepTimeUs = mActiveSleepTimeUs; 4959 } 4960 } else { 4961 mSleepTimeUs = mIdleSleepTimeUs; 4962 } 4963 } else if (mBytesWritten != 0 && audio_has_proportional_frames(mFormat)) { 4964 memset(mSinkBuffer, 0, mFrameCount * mFrameSize); 4965 mSleepTimeUs = 0; 4966 } 4967} 4968 4969void AudioFlinger::DirectOutputThread::threadLoop_exit() 4970{ 4971 { 4972 Mutex::Autolock _l(mLock); 4973 for (size_t i = 0; i < mTracks.size(); i++) { 4974 if (mTracks[i]->isFlushPending()) { 4975 mTracks[i]->flushAck(); 4976 mFlushPending = true; 4977 } 4978 } 4979 if (mFlushPending) { 4980 flushHw_l(); 4981 } 4982 } 4983 PlaybackThread::threadLoop_exit(); 4984} 4985 4986// must be called with thread mutex locked 4987bool AudioFlinger::DirectOutputThread::shouldStandby_l() 4988{ 4989 bool trackPaused = false; 4990 bool trackStopped = false; 4991 4992 if ((mType == DIRECT) && audio_is_linear_pcm(mFormat) && !usesHwAvSync()) { 4993 return !mStandby; 4994 } 4995 4996 // do not put the HAL in standby when paused. AwesomePlayer clear the offloaded AudioTrack 4997 // after a timeout and we will enter standby then. 4998 if (mTracks.size() > 0) { 4999 trackPaused = mTracks[mTracks.size() - 1]->isPaused(); 5000 trackStopped = mTracks[mTracks.size() - 1]->isStopped() || 5001 mTracks[mTracks.size() - 1]->mState == TrackBase::IDLE; 5002 } 5003 5004 return !mStandby && !(trackPaused || (mHwPaused && !trackStopped)); 5005} 5006 5007// getTrackName_l() must be called with ThreadBase::mLock held 5008int AudioFlinger::DirectOutputThread::getTrackName_l(audio_channel_mask_t channelMask __unused, 5009 audio_format_t format __unused, audio_session_t sessionId __unused) 5010{ 5011 return 0; 5012} 5013 5014// deleteTrackName_l() must be called with ThreadBase::mLock held 5015void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name __unused) 5016{ 5017} 5018 5019// checkForNewParameter_l() must be called with ThreadBase::mLock held 5020bool AudioFlinger::DirectOutputThread::checkForNewParameter_l(const String8& keyValuePair, 5021 status_t& status) 5022{ 5023 bool reconfig = false; 5024 bool a2dpDeviceChanged = false; 5025 5026 status = NO_ERROR; 5027 5028 AudioParameter param = AudioParameter(keyValuePair); 5029 int value; 5030 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 5031 // forward device change to effects that have requested to be 5032 // aware of attached audio device. 5033 if (value != AUDIO_DEVICE_NONE) { 5034 a2dpDeviceChanged = 5035 (mOutDevice & AUDIO_DEVICE_OUT_ALL_A2DP) != (value & AUDIO_DEVICE_OUT_ALL_A2DP); 5036 mOutDevice = value; 5037 for (size_t i = 0; i < mEffectChains.size(); i++) { 5038 mEffectChains[i]->setDevice_l(mOutDevice); 5039 } 5040 } 5041 } 5042 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 5043 // do not accept frame count changes if tracks are open as the track buffer 5044 // size depends on frame count and correct behavior would not be garantied 5045 // if frame count is changed after track creation 5046 if (!mTracks.isEmpty()) { 5047 status = INVALID_OPERATION; 5048 } else { 5049 reconfig = true; 5050 } 5051 } 5052 if (status == NO_ERROR) { 5053 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 5054 keyValuePair.string()); 5055 if (!mStandby && status == INVALID_OPERATION) { 5056 mOutput->standby(); 5057 mStandby = true; 5058 mBytesWritten = 0; 5059 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 5060 keyValuePair.string()); 5061 } 5062 if (status == NO_ERROR && reconfig) { 5063 readOutputParameters_l(); 5064 sendIoConfigEvent_l(AUDIO_OUTPUT_CONFIG_CHANGED); 5065 } 5066 } 5067 5068 return reconfig || a2dpDeviceChanged; 5069} 5070 5071uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() const 5072{ 5073 uint32_t time; 5074 if (audio_has_proportional_frames(mFormat)) { 5075 time = PlaybackThread::activeSleepTimeUs(); 5076 } else { 5077 time = kDirectMinSleepTimeUs; 5078 } 5079 return time; 5080} 5081 5082uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() const 5083{ 5084 uint32_t time; 5085 if (audio_has_proportional_frames(mFormat)) { 5086 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 5087 } else { 5088 time = kDirectMinSleepTimeUs; 5089 } 5090 return time; 5091} 5092 5093uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() const 5094{ 5095 uint32_t time; 5096 if (audio_has_proportional_frames(mFormat)) { 5097 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 5098 } else { 5099 time = kDirectMinSleepTimeUs; 5100 } 5101 return time; 5102} 5103 5104void AudioFlinger::DirectOutputThread::cacheParameters_l() 5105{ 5106 PlaybackThread::cacheParameters_l(); 5107 5108 // use shorter standby delay as on normal output to release 5109 // hardware resources as soon as possible 5110 // no delay on outputs with HW A/V sync 5111 if (usesHwAvSync()) { 5112 mStandbyDelayNs = 0; 5113 } else if ((mType == OFFLOAD) && !audio_has_proportional_frames(mFormat)) { 5114 mStandbyDelayNs = kOffloadStandbyDelayNs; 5115 } else { 5116 mStandbyDelayNs = microseconds(mActiveSleepTimeUs*2); 5117 } 5118} 5119 5120void AudioFlinger::DirectOutputThread::flushHw_l() 5121{ 5122 mOutput->flush(); 5123 mHwPaused = false; 5124 mFlushPending = false; 5125} 5126 5127// ---------------------------------------------------------------------------- 5128 5129AudioFlinger::AsyncCallbackThread::AsyncCallbackThread( 5130 const wp<AudioFlinger::PlaybackThread>& playbackThread) 5131 : Thread(false /*canCallJava*/), 5132 mPlaybackThread(playbackThread), 5133 mWriteAckSequence(0), 5134 mDrainSequence(0) 5135{ 5136} 5137 5138AudioFlinger::AsyncCallbackThread::~AsyncCallbackThread() 5139{ 5140} 5141 5142void AudioFlinger::AsyncCallbackThread::onFirstRef() 5143{ 5144 run("Offload Cbk", ANDROID_PRIORITY_URGENT_AUDIO); 5145} 5146 5147bool AudioFlinger::AsyncCallbackThread::threadLoop() 5148{ 5149 while (!exitPending()) { 5150 uint32_t writeAckSequence; 5151 uint32_t drainSequence; 5152 5153 { 5154 Mutex::Autolock _l(mLock); 5155 while (!((mWriteAckSequence & 1) || 5156 (mDrainSequence & 1) || 5157 exitPending())) { 5158 mWaitWorkCV.wait(mLock); 5159 } 5160 5161 if (exitPending()) { 5162 break; 5163 } 5164 ALOGV("AsyncCallbackThread mWriteAckSequence %d mDrainSequence %d", 5165 mWriteAckSequence, mDrainSequence); 5166 writeAckSequence = mWriteAckSequence; 5167 mWriteAckSequence &= ~1; 5168 drainSequence = mDrainSequence; 5169 mDrainSequence &= ~1; 5170 } 5171 { 5172 sp<AudioFlinger::PlaybackThread> playbackThread = mPlaybackThread.promote(); 5173 if (playbackThread != 0) { 5174 if (writeAckSequence & 1) { 5175 playbackThread->resetWriteBlocked(writeAckSequence >> 1); 5176 } 5177 if (drainSequence & 1) { 5178 playbackThread->resetDraining(drainSequence >> 1); 5179 } 5180 } 5181 } 5182 } 5183 return false; 5184} 5185 5186void AudioFlinger::AsyncCallbackThread::exit() 5187{ 5188 ALOGV("AsyncCallbackThread::exit"); 5189 Mutex::Autolock _l(mLock); 5190 requestExit(); 5191 mWaitWorkCV.broadcast(); 5192} 5193 5194void AudioFlinger::AsyncCallbackThread::setWriteBlocked(uint32_t sequence) 5195{ 5196 Mutex::Autolock _l(mLock); 5197 // bit 0 is cleared 5198 mWriteAckSequence = sequence << 1; 5199} 5200 5201void AudioFlinger::AsyncCallbackThread::resetWriteBlocked() 5202{ 5203 Mutex::Autolock _l(mLock); 5204 // ignore unexpected callbacks 5205 if (mWriteAckSequence & 2) { 5206 mWriteAckSequence |= 1; 5207 mWaitWorkCV.signal(); 5208 } 5209} 5210 5211void AudioFlinger::AsyncCallbackThread::setDraining(uint32_t sequence) 5212{ 5213 Mutex::Autolock _l(mLock); 5214 // bit 0 is cleared 5215 mDrainSequence = sequence << 1; 5216} 5217 5218void AudioFlinger::AsyncCallbackThread::resetDraining() 5219{ 5220 Mutex::Autolock _l(mLock); 5221 // ignore unexpected callbacks 5222 if (mDrainSequence & 2) { 5223 mDrainSequence |= 1; 5224 mWaitWorkCV.signal(); 5225 } 5226} 5227 5228 5229// ---------------------------------------------------------------------------- 5230AudioFlinger::OffloadThread::OffloadThread(const sp<AudioFlinger>& audioFlinger, 5231 AudioStreamOut* output, audio_io_handle_t id, uint32_t device, bool systemReady, 5232 uint32_t bitRate) 5233 : DirectOutputThread(audioFlinger, output, id, device, OFFLOAD, systemReady, bitRate), 5234 mPausedBytesRemaining(0) 5235{ 5236 //FIXME: mStandby should be set to true by ThreadBase constructor 5237 mStandby = true; 5238} 5239 5240void AudioFlinger::OffloadThread::threadLoop_exit() 5241{ 5242 if (mFlushPending || mHwPaused) { 5243 // If a flush is pending or track was paused, just discard buffered data 5244 flushHw_l(); 5245 } else { 5246 mMixerStatus = MIXER_DRAIN_ALL; 5247 threadLoop_drain(); 5248 } 5249 if (mUseAsyncWrite) { 5250 ALOG_ASSERT(mCallbackThread != 0); 5251 mCallbackThread->exit(); 5252 } 5253 PlaybackThread::threadLoop_exit(); 5254} 5255 5256AudioFlinger::PlaybackThread::mixer_state AudioFlinger::OffloadThread::prepareTracks_l( 5257 Vector< sp<Track> > *tracksToRemove 5258) 5259{ 5260 size_t count = mActiveTracks.size(); 5261 5262 mixer_state mixerStatus = MIXER_IDLE; 5263 bool doHwPause = false; 5264 bool doHwResume = false; 5265 5266 ALOGV("OffloadThread::prepareTracks_l active tracks %zu", count); 5267 5268 // find out which tracks need to be processed 5269 for (size_t i = 0; i < count; i++) { 5270 sp<Track> t = mActiveTracks[i].promote(); 5271 // The track died recently 5272 if (t == 0) { 5273 continue; 5274 } 5275 Track* const track = t.get(); 5276#ifdef VERY_VERY_VERBOSE_LOGGING 5277 audio_track_cblk_t* cblk = track->cblk(); 5278#endif 5279 // Only consider last track started for volume and mixer state control. 5280 // In theory an older track could underrun and restart after the new one starts 5281 // but as we only care about the transition phase between two tracks on a 5282 // direct output, it is not a problem to ignore the underrun case. 5283 sp<Track> l = mLatestActiveTrack.promote(); 5284 bool last = l.get() == track; 5285 5286 if (track->isInvalid()) { 5287 ALOGW("An invalidated track shouldn't be in active list"); 5288 tracksToRemove->add(track); 5289 continue; 5290 } 5291 5292 if (track->mState == TrackBase::IDLE) { 5293 ALOGW("An idle track shouldn't be in active list"); 5294 continue; 5295 } 5296 5297 if (track->isPausing()) { 5298 track->setPaused(); 5299 if (last) { 5300 if (mHwSupportsPause && !mHwPaused) { 5301 doHwPause = true; 5302 mHwPaused = true; 5303 } 5304 // If we were part way through writing the mixbuffer to 5305 // the HAL we must save this until we resume 5306 // BUG - this will be wrong if a different track is made active, 5307 // in that case we want to discard the pending data in the 5308 // mixbuffer and tell the client to present it again when the 5309 // track is resumed 5310 mPausedWriteLength = mCurrentWriteLength; 5311 mPausedBytesRemaining = mBytesRemaining; 5312 mBytesRemaining = 0; // stop writing 5313 } 5314 tracksToRemove->add(track); 5315 } else if (track->isFlushPending()) { 5316 track->mRetryCount = kMaxTrackRetriesOffload; 5317 track->flushAck(); 5318 if (last) { 5319 mFlushPending = true; 5320 } 5321 } else if (track->isResumePending()){ 5322 track->resumeAck(); 5323 if (last) { 5324 if (mPausedBytesRemaining) { 5325 // Need to continue write that was interrupted 5326 mCurrentWriteLength = mPausedWriteLength; 5327 mBytesRemaining = mPausedBytesRemaining; 5328 mPausedBytesRemaining = 0; 5329 } 5330 if (mHwPaused) { 5331 doHwResume = true; 5332 mHwPaused = false; 5333 // threadLoop_mix() will handle the case that we need to 5334 // resume an interrupted write 5335 } 5336 // enable write to audio HAL 5337 mSleepTimeUs = 0; 5338 5339 // Do not handle new data in this iteration even if track->framesReady() 5340 mixerStatus = MIXER_TRACKS_ENABLED; 5341 } 5342 } else if (track->framesReady() && track->isReady() && 5343 !track->isPaused() && !track->isTerminated() && !track->isStopping_2()) { 5344 ALOGVV("OffloadThread: track %d s=%08x [OK]", track->name(), cblk->mServer); 5345 if (track->mFillingUpStatus == Track::FS_FILLED) { 5346 track->mFillingUpStatus = Track::FS_ACTIVE; 5347 // make sure processVolume_l() will apply new volume even if 0 5348 mLeftVolFloat = mRightVolFloat = -1.0; 5349 } 5350 5351 if (last) { 5352 sp<Track> previousTrack = mPreviousTrack.promote(); 5353 if (previousTrack != 0) { 5354 if (track != previousTrack.get()) { 5355 // Flush any data still being written from last track 5356 mBytesRemaining = 0; 5357 if (mPausedBytesRemaining) { 5358 // Last track was paused so we also need to flush saved 5359 // mixbuffer state and invalidate track so that it will 5360 // re-submit that unwritten data when it is next resumed 5361 mPausedBytesRemaining = 0; 5362 // Invalidate is a bit drastic - would be more efficient 5363 // to have a flag to tell client that some of the 5364 // previously written data was lost 5365 previousTrack->invalidate(); 5366 } 5367 // flush data already sent to the DSP if changing audio session as audio 5368 // comes from a different source. Also invalidate previous track to force a 5369 // seek when resuming. 5370 if (previousTrack->sessionId() != track->sessionId()) { 5371 previousTrack->invalidate(); 5372 } 5373 } 5374 } 5375 mPreviousTrack = track; 5376 // reset retry count 5377 track->mRetryCount = kMaxTrackRetriesOffload; 5378 mActiveTrack = t; 5379 mixerStatus = MIXER_TRACKS_READY; 5380 } 5381 } else { 5382 ALOGVV("OffloadThread: track %d s=%08x [NOT READY]", track->name(), cblk->mServer); 5383 if (track->isStopping_1()) { 5384 // Hardware buffer can hold a large amount of audio so we must 5385 // wait for all current track's data to drain before we say 5386 // that the track is stopped. 5387 if (mBytesRemaining == 0) { 5388 // Only start draining when all data in mixbuffer 5389 // has been written 5390 ALOGV("OffloadThread: underrun and STOPPING_1 -> draining, STOPPING_2"); 5391 track->mState = TrackBase::STOPPING_2; // so presentation completes after drain 5392 // do not drain if no data was ever sent to HAL (mStandby == true) 5393 if (last && !mStandby) { 5394 // do not modify drain sequence if we are already draining. This happens 5395 // when resuming from pause after drain. 5396 if ((mDrainSequence & 1) == 0) { 5397 mSleepTimeUs = 0; 5398 mStandbyTimeNs = systemTime() + mStandbyDelayNs; 5399 mixerStatus = MIXER_DRAIN_TRACK; 5400 mDrainSequence += 2; 5401 } 5402 if (mHwPaused) { 5403 // It is possible to move from PAUSED to STOPPING_1 without 5404 // a resume so we must ensure hardware is running 5405 doHwResume = true; 5406 mHwPaused = false; 5407 } 5408 } 5409 } 5410 } else if (track->isStopping_2()) { 5411 // Drain has completed or we are in standby, signal presentation complete 5412 if (!(mDrainSequence & 1) || !last || mStandby) { 5413 track->mState = TrackBase::STOPPED; 5414 size_t audioHALFrames = 5415 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 5416 int64_t framesWritten = 5417 mBytesWritten / mOutput->getFrameSize(); 5418 track->presentationComplete(framesWritten, audioHALFrames); 5419 track->reset(); 5420 tracksToRemove->add(track); 5421 } 5422 } else { 5423 // No buffers for this track. Give it a few chances to 5424 // fill a buffer, then remove it from active list. 5425 if (--(track->mRetryCount) <= 0) { 5426 ALOGV("OffloadThread: BUFFER TIMEOUT: remove(%d) from active list", 5427 track->name()); 5428 tracksToRemove->add(track); 5429 // indicate to client process that the track was disabled because of underrun; 5430 // it will then automatically call start() when data is available 5431 track->disable(); 5432 } else if (last){ 5433 mixerStatus = MIXER_TRACKS_ENABLED; 5434 } 5435 } 5436 } 5437 // compute volume for this track 5438 processVolume_l(track, last); 5439 } 5440 5441 // make sure the pause/flush/resume sequence is executed in the right order. 5442 // If a flush is pending and a track is active but the HW is not paused, force a HW pause 5443 // before flush and then resume HW. This can happen in case of pause/flush/resume 5444 // if resume is received before pause is executed. 5445 if (!mStandby && (doHwPause || (mFlushPending && !mHwPaused && (count != 0)))) { 5446 mOutput->stream->pause(mOutput->stream); 5447 } 5448 if (mFlushPending) { 5449 flushHw_l(); 5450 } 5451 if (!mStandby && doHwResume) { 5452 mOutput->stream->resume(mOutput->stream); 5453 } 5454 5455 // remove all the tracks that need to be... 5456 removeTracks_l(*tracksToRemove); 5457 5458 return mixerStatus; 5459} 5460 5461// must be called with thread mutex locked 5462bool AudioFlinger::OffloadThread::waitingAsyncCallback_l() 5463{ 5464 ALOGVV("waitingAsyncCallback_l mWriteAckSequence %d mDrainSequence %d", 5465 mWriteAckSequence, mDrainSequence); 5466 if (mUseAsyncWrite && ((mWriteAckSequence & 1) || (mDrainSequence & 1))) { 5467 return true; 5468 } 5469 return false; 5470} 5471 5472bool AudioFlinger::OffloadThread::waitingAsyncCallback() 5473{ 5474 Mutex::Autolock _l(mLock); 5475 return waitingAsyncCallback_l(); 5476} 5477 5478void AudioFlinger::OffloadThread::flushHw_l() 5479{ 5480 DirectOutputThread::flushHw_l(); 5481 // Flush anything still waiting in the mixbuffer 5482 mCurrentWriteLength = 0; 5483 mBytesRemaining = 0; 5484 mPausedWriteLength = 0; 5485 mPausedBytesRemaining = 0; 5486 5487 if (mUseAsyncWrite) { 5488 // discard any pending drain or write ack by incrementing sequence 5489 mWriteAckSequence = (mWriteAckSequence + 2) & ~1; 5490 mDrainSequence = (mDrainSequence + 2) & ~1; 5491 ALOG_ASSERT(mCallbackThread != 0); 5492 mCallbackThread->setWriteBlocked(mWriteAckSequence); 5493 mCallbackThread->setDraining(mDrainSequence); 5494 } 5495} 5496 5497uint32_t AudioFlinger::OffloadThread::activeSleepTimeUs() const 5498{ 5499 uint32_t time; 5500 if (audio_has_proportional_frames(mFormat)) { 5501 time = PlaybackThread::activeSleepTimeUs(); 5502 } else { 5503 // sleep time is half the duration of an audio HAL buffer. 5504 // Note: This can be problematic in case of underrun with variable bit rate and 5505 // current rate is much less than initial rate. 5506 time = (uint32_t)max(kDirectMinSleepTimeUs, mBufferDurationUs / 2); 5507 } 5508 return time; 5509} 5510 5511// ---------------------------------------------------------------------------- 5512 5513AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 5514 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id, bool systemReady) 5515 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->outDevice(), 5516 systemReady, DUPLICATING), 5517 mWaitTimeMs(UINT_MAX) 5518{ 5519 addOutputTrack(mainThread); 5520} 5521 5522AudioFlinger::DuplicatingThread::~DuplicatingThread() 5523{ 5524 for (size_t i = 0; i < mOutputTracks.size(); i++) { 5525 mOutputTracks[i]->destroy(); 5526 } 5527} 5528 5529void AudioFlinger::DuplicatingThread::threadLoop_mix() 5530{ 5531 // mix buffers... 5532 if (outputsReady(outputTracks)) { 5533 mAudioMixer->process(); 5534 } else { 5535 if (mMixerBufferValid) { 5536 memset(mMixerBuffer, 0, mMixerBufferSize); 5537 } else { 5538 memset(mSinkBuffer, 0, mSinkBufferSize); 5539 } 5540 } 5541 mSleepTimeUs = 0; 5542 writeFrames = mNormalFrameCount; 5543 mCurrentWriteLength = mSinkBufferSize; 5544 mStandbyTimeNs = systemTime() + mStandbyDelayNs; 5545} 5546 5547void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() 5548{ 5549 if (mSleepTimeUs == 0) { 5550 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 5551 mSleepTimeUs = mActiveSleepTimeUs; 5552 } else { 5553 mSleepTimeUs = mIdleSleepTimeUs; 5554 } 5555 } else if (mBytesWritten != 0) { 5556 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 5557 writeFrames = mNormalFrameCount; 5558 memset(mSinkBuffer, 0, mSinkBufferSize); 5559 } else { 5560 // flush remaining overflow buffers in output tracks 5561 writeFrames = 0; 5562 } 5563 mSleepTimeUs = 0; 5564 } 5565} 5566 5567ssize_t AudioFlinger::DuplicatingThread::threadLoop_write() 5568{ 5569 for (size_t i = 0; i < outputTracks.size(); i++) { 5570 outputTracks[i]->write(mSinkBuffer, writeFrames); 5571 } 5572 mStandby = false; 5573 return (ssize_t)mSinkBufferSize; 5574} 5575 5576void AudioFlinger::DuplicatingThread::threadLoop_standby() 5577{ 5578 // DuplicatingThread implements standby by stopping all tracks 5579 for (size_t i = 0; i < outputTracks.size(); i++) { 5580 outputTracks[i]->stop(); 5581 } 5582} 5583 5584void AudioFlinger::DuplicatingThread::saveOutputTracks() 5585{ 5586 outputTracks = mOutputTracks; 5587} 5588 5589void AudioFlinger::DuplicatingThread::clearOutputTracks() 5590{ 5591 outputTracks.clear(); 5592} 5593 5594void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 5595{ 5596 Mutex::Autolock _l(mLock); 5597 // The downstream MixerThread consumes thread->frameCount() amount of frames per mix pass. 5598 // Adjust for thread->sampleRate() to determine minimum buffer frame count. 5599 // Then triple buffer because Threads do not run synchronously and may not be clock locked. 5600 const size_t frameCount = 5601 3 * sourceFramesNeeded(mSampleRate, thread->frameCount(), thread->sampleRate()); 5602 // TODO: Consider asynchronous sample rate conversion to handle clock disparity 5603 // from different OutputTracks and their associated MixerThreads (e.g. one may 5604 // nearly empty and the other may be dropping data). 5605 5606 sp<OutputTrack> outputTrack = new OutputTrack(thread, 5607 this, 5608 mSampleRate, 5609 mFormat, 5610 mChannelMask, 5611 frameCount, 5612 IPCThreadState::self()->getCallingUid()); 5613 if (outputTrack->cblk() != NULL) { 5614 thread->setStreamVolume(AUDIO_STREAM_PATCH, 1.0f); 5615 mOutputTracks.add(outputTrack); 5616 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack.get(), thread); 5617 updateWaitTime_l(); 5618 } 5619} 5620 5621void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 5622{ 5623 Mutex::Autolock _l(mLock); 5624 for (size_t i = 0; i < mOutputTracks.size(); i++) { 5625 if (mOutputTracks[i]->thread() == thread) { 5626 mOutputTracks[i]->destroy(); 5627 mOutputTracks.removeAt(i); 5628 updateWaitTime_l(); 5629 if (thread->getOutput() == mOutput) { 5630 mOutput = NULL; 5631 } 5632 return; 5633 } 5634 } 5635 ALOGV("removeOutputTrack(): unknown thread: %p", thread); 5636} 5637 5638// caller must hold mLock 5639void AudioFlinger::DuplicatingThread::updateWaitTime_l() 5640{ 5641 mWaitTimeMs = UINT_MAX; 5642 for (size_t i = 0; i < mOutputTracks.size(); i++) { 5643 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 5644 if (strong != 0) { 5645 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 5646 if (waitTimeMs < mWaitTimeMs) { 5647 mWaitTimeMs = waitTimeMs; 5648 } 5649 } 5650 } 5651} 5652 5653 5654bool AudioFlinger::DuplicatingThread::outputsReady( 5655 const SortedVector< sp<OutputTrack> > &outputTracks) 5656{ 5657 for (size_t i = 0; i < outputTracks.size(); i++) { 5658 sp<ThreadBase> thread = outputTracks[i]->thread().promote(); 5659 if (thread == 0) { 5660 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", 5661 outputTracks[i].get()); 5662 return false; 5663 } 5664 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 5665 // see note at standby() declaration 5666 if (playbackThread->standby() && !playbackThread->isSuspended()) { 5667 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), 5668 thread.get()); 5669 return false; 5670 } 5671 } 5672 return true; 5673} 5674 5675uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() const 5676{ 5677 return (mWaitTimeMs * 1000) / 2; 5678} 5679 5680void AudioFlinger::DuplicatingThread::cacheParameters_l() 5681{ 5682 // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first 5683 updateWaitTime_l(); 5684 5685 MixerThread::cacheParameters_l(); 5686} 5687 5688// ---------------------------------------------------------------------------- 5689// Record 5690// ---------------------------------------------------------------------------- 5691 5692AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 5693 AudioStreamIn *input, 5694 audio_io_handle_t id, 5695 audio_devices_t outDevice, 5696 audio_devices_t inDevice, 5697 bool systemReady 5698#ifdef TEE_SINK 5699 , const sp<NBAIO_Sink>& teeSink 5700#endif 5701 ) : 5702 ThreadBase(audioFlinger, id, outDevice, inDevice, RECORD, systemReady), 5703 mInput(input), mActiveTracksGen(0), mRsmpInBuffer(NULL), 5704 // mRsmpInFrames and mRsmpInFramesP2 are set by readInputParameters_l() 5705 mRsmpInRear(0) 5706#ifdef TEE_SINK 5707 , mTeeSink(teeSink) 5708#endif 5709 , mReadOnlyHeap(new MemoryDealer(kRecordThreadReadOnlyHeapSize, 5710 "RecordThreadRO", MemoryHeapBase::READ_ONLY)) 5711 // mFastCapture below 5712 , mFastCaptureFutex(0) 5713 // mInputSource 5714 // mPipeSink 5715 // mPipeSource 5716 , mPipeFramesP2(0) 5717 // mPipeMemory 5718 // mFastCaptureNBLogWriter 5719 , mFastTrackAvail(false) 5720{ 5721 snprintf(mThreadName, kThreadNameLength, "AudioIn_%X", id); 5722 mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mThreadName); 5723 5724 readInputParameters_l(); 5725 5726 // create an NBAIO source for the HAL input stream, and negotiate 5727 mInputSource = new AudioStreamInSource(input->stream); 5728 size_t numCounterOffers = 0; 5729 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount, mFormat)}; 5730#if !LOG_NDEBUG 5731 ssize_t index = 5732#else 5733 (void) 5734#endif 5735 mInputSource->negotiate(offers, 1, NULL, numCounterOffers); 5736 ALOG_ASSERT(index == 0); 5737 5738 // initialize fast capture depending on configuration 5739 bool initFastCapture; 5740 switch (kUseFastCapture) { 5741 case FastCapture_Never: 5742 initFastCapture = false; 5743 break; 5744 case FastCapture_Always: 5745 initFastCapture = true; 5746 break; 5747 case FastCapture_Static: 5748 initFastCapture = (mFrameCount * 1000) / mSampleRate < kMinNormalCaptureBufferSizeMs; 5749 break; 5750 // case FastCapture_Dynamic: 5751 } 5752 5753 if (initFastCapture) { 5754 // create a Pipe for FastCapture to write to, and for us and fast tracks to read from 5755 NBAIO_Format format = mInputSource->format(); 5756 size_t pipeFramesP2 = roundup(mSampleRate / 25); // double-buffering of 20 ms each 5757 size_t pipeSize = pipeFramesP2 * Format_frameSize(format); 5758 void *pipeBuffer; 5759 const sp<MemoryDealer> roHeap(readOnlyHeap()); 5760 sp<IMemory> pipeMemory; 5761 if ((roHeap == 0) || 5762 (pipeMemory = roHeap->allocate(pipeSize)) == 0 || 5763 (pipeBuffer = pipeMemory->pointer()) == NULL) { 5764 ALOGE("not enough memory for pipe buffer size=%zu", pipeSize); 5765 goto failed; 5766 } 5767 // pipe will be shared directly with fast clients, so clear to avoid leaking old information 5768 memset(pipeBuffer, 0, pipeSize); 5769 Pipe *pipe = new Pipe(pipeFramesP2, format, pipeBuffer); 5770 const NBAIO_Format offers[1] = {format}; 5771 size_t numCounterOffers = 0; 5772 ssize_t index = pipe->negotiate(offers, 1, NULL, numCounterOffers); 5773 ALOG_ASSERT(index == 0); 5774 mPipeSink = pipe; 5775 PipeReader *pipeReader = new PipeReader(*pipe); 5776 numCounterOffers = 0; 5777 index = pipeReader->negotiate(offers, 1, NULL, numCounterOffers); 5778 ALOG_ASSERT(index == 0); 5779 mPipeSource = pipeReader; 5780 mPipeFramesP2 = pipeFramesP2; 5781 mPipeMemory = pipeMemory; 5782 5783 // create fast capture 5784 mFastCapture = new FastCapture(); 5785 FastCaptureStateQueue *sq = mFastCapture->sq(); 5786#ifdef STATE_QUEUE_DUMP 5787 // FIXME 5788#endif 5789 FastCaptureState *state = sq->begin(); 5790 state->mCblk = NULL; 5791 state->mInputSource = mInputSource.get(); 5792 state->mInputSourceGen++; 5793 state->mPipeSink = pipe; 5794 state->mPipeSinkGen++; 5795 state->mFrameCount = mFrameCount; 5796 state->mCommand = FastCaptureState::COLD_IDLE; 5797 // already done in constructor initialization list 5798 //mFastCaptureFutex = 0; 5799 state->mColdFutexAddr = &mFastCaptureFutex; 5800 state->mColdGen++; 5801 state->mDumpState = &mFastCaptureDumpState; 5802#ifdef TEE_SINK 5803 // FIXME 5804#endif 5805 mFastCaptureNBLogWriter = audioFlinger->newWriter_l(kFastCaptureLogSize, "FastCapture"); 5806 state->mNBLogWriter = mFastCaptureNBLogWriter.get(); 5807 sq->end(); 5808 sq->push(FastCaptureStateQueue::BLOCK_UNTIL_PUSHED); 5809 5810 // start the fast capture 5811 mFastCapture->run("FastCapture", ANDROID_PRIORITY_URGENT_AUDIO); 5812 pid_t tid = mFastCapture->getTid(); 5813 sendPrioConfigEvent(getpid_cached, tid, kPriorityFastCapture); 5814#ifdef AUDIO_WATCHDOG 5815 // FIXME 5816#endif 5817 5818 mFastTrackAvail = true; 5819 } 5820failed: ; 5821 5822 // FIXME mNormalSource 5823} 5824 5825AudioFlinger::RecordThread::~RecordThread() 5826{ 5827 if (mFastCapture != 0) { 5828 FastCaptureStateQueue *sq = mFastCapture->sq(); 5829 FastCaptureState *state = sq->begin(); 5830 if (state->mCommand == FastCaptureState::COLD_IDLE) { 5831 int32_t old = android_atomic_inc(&mFastCaptureFutex); 5832 if (old == -1) { 5833 (void) syscall(__NR_futex, &mFastCaptureFutex, FUTEX_WAKE_PRIVATE, 1); 5834 } 5835 } 5836 state->mCommand = FastCaptureState::EXIT; 5837 sq->end(); 5838 sq->push(FastCaptureStateQueue::BLOCK_UNTIL_PUSHED); 5839 mFastCapture->join(); 5840 mFastCapture.clear(); 5841 } 5842 mAudioFlinger->unregisterWriter(mFastCaptureNBLogWriter); 5843 mAudioFlinger->unregisterWriter(mNBLogWriter); 5844 free(mRsmpInBuffer); 5845} 5846 5847void AudioFlinger::RecordThread::onFirstRef() 5848{ 5849 run(mThreadName, PRIORITY_URGENT_AUDIO); 5850} 5851 5852bool AudioFlinger::RecordThread::threadLoop() 5853{ 5854 nsecs_t lastWarning = 0; 5855 5856 inputStandBy(); 5857 5858reacquire_wakelock: 5859 sp<RecordTrack> activeTrack; 5860 int activeTracksGen; 5861 { 5862 Mutex::Autolock _l(mLock); 5863 size_t size = mActiveTracks.size(); 5864 activeTracksGen = mActiveTracksGen; 5865 if (size > 0) { 5866 // FIXME an arbitrary choice 5867 activeTrack = mActiveTracks[0]; 5868 acquireWakeLock_l(activeTrack->uid()); 5869 if (size > 1) { 5870 SortedVector<int> tmp; 5871 for (size_t i = 0; i < size; i++) { 5872 tmp.add(mActiveTracks[i]->uid()); 5873 } 5874 updateWakeLockUids_l(tmp); 5875 } 5876 } else { 5877 acquireWakeLock_l(-1); 5878 } 5879 } 5880 5881 // used to request a deferred sleep, to be executed later while mutex is unlocked 5882 uint32_t sleepUs = 0; 5883 5884 // loop while there is work to do 5885 for (;;) { 5886 Vector< sp<EffectChain> > effectChains; 5887 5888 // sleep with mutex unlocked 5889 if (sleepUs > 0) { 5890 ATRACE_BEGIN("sleep"); 5891 usleep(sleepUs); 5892 ATRACE_END(); 5893 sleepUs = 0; 5894 } 5895 5896 // activeTracks accumulates a copy of a subset of mActiveTracks 5897 Vector< sp<RecordTrack> > activeTracks; 5898 5899 // reference to the (first and only) active fast track 5900 sp<RecordTrack> fastTrack; 5901 5902 // reference to a fast track which is about to be removed 5903 sp<RecordTrack> fastTrackToRemove; 5904 5905 { // scope for mLock 5906 Mutex::Autolock _l(mLock); 5907 5908 processConfigEvents_l(); 5909 5910 // check exitPending here because checkForNewParameters_l() and 5911 // checkForNewParameters_l() can temporarily release mLock 5912 if (exitPending()) { 5913 break; 5914 } 5915 5916 // if no active track(s), then standby and release wakelock 5917 size_t size = mActiveTracks.size(); 5918 if (size == 0) { 5919 standbyIfNotAlreadyInStandby(); 5920 // exitPending() can't become true here 5921 releaseWakeLock_l(); 5922 ALOGV("RecordThread: loop stopping"); 5923 // go to sleep 5924 mWaitWorkCV.wait(mLock); 5925 ALOGV("RecordThread: loop starting"); 5926 goto reacquire_wakelock; 5927 } 5928 5929 if (mActiveTracksGen != activeTracksGen) { 5930 activeTracksGen = mActiveTracksGen; 5931 SortedVector<int> tmp; 5932 for (size_t i = 0; i < size; i++) { 5933 tmp.add(mActiveTracks[i]->uid()); 5934 } 5935 updateWakeLockUids_l(tmp); 5936 } 5937 5938 bool doBroadcast = false; 5939 for (size_t i = 0; i < size; ) { 5940 5941 activeTrack = mActiveTracks[i]; 5942 if (activeTrack->isTerminated()) { 5943 if (activeTrack->isFastTrack()) { 5944 ALOG_ASSERT(fastTrackToRemove == 0); 5945 fastTrackToRemove = activeTrack; 5946 } 5947 removeTrack_l(activeTrack); 5948 mActiveTracks.remove(activeTrack); 5949 mActiveTracksGen++; 5950 size--; 5951 continue; 5952 } 5953 5954 TrackBase::track_state activeTrackState = activeTrack->mState; 5955 switch (activeTrackState) { 5956 5957 case TrackBase::PAUSING: 5958 mActiveTracks.remove(activeTrack); 5959 mActiveTracksGen++; 5960 doBroadcast = true; 5961 size--; 5962 continue; 5963 5964 case TrackBase::STARTING_1: 5965 sleepUs = 10000; 5966 i++; 5967 continue; 5968 5969 case TrackBase::STARTING_2: 5970 doBroadcast = true; 5971 mStandby = false; 5972 activeTrack->mState = TrackBase::ACTIVE; 5973 break; 5974 5975 case TrackBase::ACTIVE: 5976 break; 5977 5978 case TrackBase::IDLE: 5979 i++; 5980 continue; 5981 5982 default: 5983 LOG_ALWAYS_FATAL("Unexpected activeTrackState %d", activeTrackState); 5984 } 5985 5986 activeTracks.add(activeTrack); 5987 i++; 5988 5989 if (activeTrack->isFastTrack()) { 5990 ALOG_ASSERT(!mFastTrackAvail); 5991 ALOG_ASSERT(fastTrack == 0); 5992 fastTrack = activeTrack; 5993 } 5994 } 5995 if (doBroadcast) { 5996 mStartStopCond.broadcast(); 5997 } 5998 5999 // sleep if there are no active tracks to process 6000 if (activeTracks.size() == 0) { 6001 if (sleepUs == 0) { 6002 sleepUs = kRecordThreadSleepUs; 6003 } 6004 continue; 6005 } 6006 sleepUs = 0; 6007 6008 lockEffectChains_l(effectChains); 6009 } 6010 6011 // thread mutex is now unlocked, mActiveTracks unknown, activeTracks.size() > 0 6012 6013 size_t size = effectChains.size(); 6014 for (size_t i = 0; i < size; i++) { 6015 // thread mutex is not locked, but effect chain is locked 6016 effectChains[i]->process_l(); 6017 } 6018 6019 // Push a new fast capture state if fast capture is not already running, or cblk change 6020 if (mFastCapture != 0) { 6021 FastCaptureStateQueue *sq = mFastCapture->sq(); 6022 FastCaptureState *state = sq->begin(); 6023 bool didModify = false; 6024 FastCaptureStateQueue::block_t block = FastCaptureStateQueue::BLOCK_UNTIL_PUSHED; 6025 if (state->mCommand != FastCaptureState::READ_WRITE /* FIXME && 6026 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)*/) { 6027 if (state->mCommand == FastCaptureState::COLD_IDLE) { 6028 int32_t old = android_atomic_inc(&mFastCaptureFutex); 6029 if (old == -1) { 6030 (void) syscall(__NR_futex, &mFastCaptureFutex, FUTEX_WAKE_PRIVATE, 1); 6031 } 6032 } 6033 state->mCommand = FastCaptureState::READ_WRITE; 6034#if 0 // FIXME 6035 mFastCaptureDumpState.increaseSamplingN(mAudioFlinger->isLowRamDevice() ? 6036 FastThreadDumpState::kSamplingNforLowRamDevice : 6037 FastThreadDumpState::kSamplingN); 6038#endif 6039 didModify = true; 6040 } 6041 audio_track_cblk_t *cblkOld = state->mCblk; 6042 audio_track_cblk_t *cblkNew = fastTrack != 0 ? fastTrack->cblk() : NULL; 6043 if (cblkNew != cblkOld) { 6044 state->mCblk = cblkNew; 6045 // block until acked if removing a fast track 6046 if (cblkOld != NULL) { 6047 block = FastCaptureStateQueue::BLOCK_UNTIL_ACKED; 6048 } 6049 didModify = true; 6050 } 6051 sq->end(didModify); 6052 if (didModify) { 6053 sq->push(block); 6054#if 0 6055 if (kUseFastCapture == FastCapture_Dynamic) { 6056 mNormalSource = mPipeSource; 6057 } 6058#endif 6059 } 6060 } 6061 6062 // now run the fast track destructor with thread mutex unlocked 6063 fastTrackToRemove.clear(); 6064 6065 // Read from HAL to keep up with fastest client if multiple active tracks, not slowest one. 6066 // Only the client(s) that are too slow will overrun. But if even the fastest client is too 6067 // slow, then this RecordThread will overrun by not calling HAL read often enough. 6068 // If destination is non-contiguous, first read past the nominal end of buffer, then 6069 // copy to the right place. Permitted because mRsmpInBuffer was over-allocated. 6070 6071 int32_t rear = mRsmpInRear & (mRsmpInFramesP2 - 1); 6072 ssize_t framesRead; 6073 6074 // If an NBAIO source is present, use it to read the normal capture's data 6075 if (mPipeSource != 0) { 6076 size_t framesToRead = mBufferSize / mFrameSize; 6077 framesRead = mPipeSource->read((uint8_t*)mRsmpInBuffer + rear * mFrameSize, 6078 framesToRead); 6079 if (framesRead == 0) { 6080 // since pipe is non-blocking, simulate blocking input 6081 sleepUs = (framesToRead * 1000000LL) / mSampleRate; 6082 } 6083 // otherwise use the HAL / AudioStreamIn directly 6084 } else { 6085 ATRACE_BEGIN("read"); 6086 ssize_t bytesRead = mInput->stream->read(mInput->stream, 6087 (uint8_t*)mRsmpInBuffer + rear * mFrameSize, mBufferSize); 6088 ATRACE_END(); 6089 if (bytesRead < 0) { 6090 framesRead = bytesRead; 6091 } else { 6092 framesRead = bytesRead / mFrameSize; 6093 } 6094 } 6095 6096 // Update server timestamp with server stats 6097 // systemTime() is optional if the hardware supports timestamps. 6098 mTimestamp.mPosition[ExtendedTimestamp::LOCATION_SERVER] += framesRead; 6099 mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_SERVER] = systemTime(); 6100 6101 // Update server timestamp with kernel stats 6102 if (mInput->stream->get_capture_position != nullptr) { 6103 int64_t position, time; 6104 int ret = mInput->stream->get_capture_position(mInput->stream, &position, &time); 6105 if (ret == NO_ERROR) { 6106 mTimestamp.mPosition[ExtendedTimestamp::LOCATION_KERNEL] = position; 6107 mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL] = time; 6108 // Note: In general record buffers should tend to be empty in 6109 // a properly running pipeline. 6110 // 6111 // Also, it is not advantageous to call get_presentation_position during the read 6112 // as the read obtains a lock, preventing the timestamp call from executing. 6113 } 6114 } 6115 // Use this to track timestamp information 6116 // ALOGD("%s", mTimestamp.toString().c_str()); 6117 6118 if (framesRead < 0 || (framesRead == 0 && mPipeSource == 0)) { 6119 ALOGE("read failed: framesRead=%zd", framesRead); 6120 // Force input into standby so that it tries to recover at next read attempt 6121 inputStandBy(); 6122 sleepUs = kRecordThreadSleepUs; 6123 } 6124 if (framesRead <= 0) { 6125 goto unlock; 6126 } 6127 ALOG_ASSERT(framesRead > 0); 6128 6129 if (mTeeSink != 0) { 6130 (void) mTeeSink->write((uint8_t*)mRsmpInBuffer + rear * mFrameSize, framesRead); 6131 } 6132 // If destination is non-contiguous, we now correct for reading past end of buffer. 6133 { 6134 size_t part1 = mRsmpInFramesP2 - rear; 6135 if ((size_t) framesRead > part1) { 6136 memcpy(mRsmpInBuffer, (uint8_t*)mRsmpInBuffer + mRsmpInFramesP2 * mFrameSize, 6137 (framesRead - part1) * mFrameSize); 6138 } 6139 } 6140 rear = mRsmpInRear += framesRead; 6141 6142 size = activeTracks.size(); 6143 // loop over each active track 6144 for (size_t i = 0; i < size; i++) { 6145 activeTrack = activeTracks[i]; 6146 6147 // skip fast tracks, as those are handled directly by FastCapture 6148 if (activeTrack->isFastTrack()) { 6149 continue; 6150 } 6151 6152 // TODO: This code probably should be moved to RecordTrack. 6153 // TODO: Update the activeTrack buffer converter in case of reconfigure. 6154 6155 enum { 6156 OVERRUN_UNKNOWN, 6157 OVERRUN_TRUE, 6158 OVERRUN_FALSE 6159 } overrun = OVERRUN_UNKNOWN; 6160 6161 // loop over getNextBuffer to handle circular sink 6162 for (;;) { 6163 6164 activeTrack->mSink.frameCount = ~0; 6165 status_t status = activeTrack->getNextBuffer(&activeTrack->mSink); 6166 size_t framesOut = activeTrack->mSink.frameCount; 6167 LOG_ALWAYS_FATAL_IF((status == OK) != (framesOut > 0)); 6168 6169 // check available frames and handle overrun conditions 6170 // if the record track isn't draining fast enough. 6171 bool hasOverrun; 6172 size_t framesIn; 6173 activeTrack->mResamplerBufferProvider->sync(&framesIn, &hasOverrun); 6174 if (hasOverrun) { 6175 overrun = OVERRUN_TRUE; 6176 } 6177 if (framesOut == 0 || framesIn == 0) { 6178 break; 6179 } 6180 6181 // Don't allow framesOut to be larger than what is possible with resampling 6182 // from framesIn. 6183 // This isn't strictly necessary but helps limit buffer resizing in 6184 // RecordBufferConverter. TODO: remove when no longer needed. 6185 framesOut = min(framesOut, 6186 destinationFramesPossible( 6187 framesIn, mSampleRate, activeTrack->mSampleRate)); 6188 // process frames from the RecordThread buffer provider to the RecordTrack buffer 6189 framesOut = activeTrack->mRecordBufferConverter->convert( 6190 activeTrack->mSink.raw, activeTrack->mResamplerBufferProvider, framesOut); 6191 6192 if (framesOut > 0 && (overrun == OVERRUN_UNKNOWN)) { 6193 overrun = OVERRUN_FALSE; 6194 } 6195 6196 if (activeTrack->mFramesToDrop == 0) { 6197 if (framesOut > 0) { 6198 activeTrack->mSink.frameCount = framesOut; 6199 activeTrack->releaseBuffer(&activeTrack->mSink); 6200 } 6201 } else { 6202 // FIXME could do a partial drop of framesOut 6203 if (activeTrack->mFramesToDrop > 0) { 6204 activeTrack->mFramesToDrop -= framesOut; 6205 if (activeTrack->mFramesToDrop <= 0) { 6206 activeTrack->clearSyncStartEvent(); 6207 } 6208 } else { 6209 activeTrack->mFramesToDrop += framesOut; 6210 if (activeTrack->mFramesToDrop >= 0 || activeTrack->mSyncStartEvent == 0 || 6211 activeTrack->mSyncStartEvent->isCancelled()) { 6212 ALOGW("Synced record %s, session %d, trigger session %d", 6213 (activeTrack->mFramesToDrop >= 0) ? "timed out" : "cancelled", 6214 activeTrack->sessionId(), 6215 (activeTrack->mSyncStartEvent != 0) ? 6216 activeTrack->mSyncStartEvent->triggerSession() : 6217 AUDIO_SESSION_NONE); 6218 activeTrack->clearSyncStartEvent(); 6219 } 6220 } 6221 } 6222 6223 if (framesOut == 0) { 6224 break; 6225 } 6226 } 6227 6228 switch (overrun) { 6229 case OVERRUN_TRUE: 6230 // client isn't retrieving buffers fast enough 6231 if (!activeTrack->setOverflow()) { 6232 nsecs_t now = systemTime(); 6233 // FIXME should lastWarning per track? 6234 if ((now - lastWarning) > kWarningThrottleNs) { 6235 ALOGW("RecordThread: buffer overflow"); 6236 lastWarning = now; 6237 } 6238 } 6239 break; 6240 case OVERRUN_FALSE: 6241 activeTrack->clearOverflow(); 6242 break; 6243 case OVERRUN_UNKNOWN: 6244 break; 6245 } 6246 6247 // update frame information and push timestamp out 6248 activeTrack->updateTrackFrameInfo( 6249 activeTrack->mServerProxy->framesReleased(), 6250 mTimestamp.mPosition[ExtendedTimestamp::LOCATION_SERVER], 6251 mSampleRate, mTimestamp); 6252 } 6253 6254unlock: 6255 // enable changes in effect chain 6256 unlockEffectChains(effectChains); 6257 // effectChains doesn't need to be cleared, since it is cleared by destructor at scope end 6258 } 6259 6260 standbyIfNotAlreadyInStandby(); 6261 6262 { 6263 Mutex::Autolock _l(mLock); 6264 for (size_t i = 0; i < mTracks.size(); i++) { 6265 sp<RecordTrack> track = mTracks[i]; 6266 track->invalidate(); 6267 } 6268 mActiveTracks.clear(); 6269 mActiveTracksGen++; 6270 mStartStopCond.broadcast(); 6271 } 6272 6273 releaseWakeLock(); 6274 6275 ALOGV("RecordThread %p exiting", this); 6276 return false; 6277} 6278 6279void AudioFlinger::RecordThread::standbyIfNotAlreadyInStandby() 6280{ 6281 if (!mStandby) { 6282 inputStandBy(); 6283 mStandby = true; 6284 } 6285} 6286 6287void AudioFlinger::RecordThread::inputStandBy() 6288{ 6289 // Idle the fast capture if it's currently running 6290 if (mFastCapture != 0) { 6291 FastCaptureStateQueue *sq = mFastCapture->sq(); 6292 FastCaptureState *state = sq->begin(); 6293 if (!(state->mCommand & FastCaptureState::IDLE)) { 6294 state->mCommand = FastCaptureState::COLD_IDLE; 6295 state->mColdFutexAddr = &mFastCaptureFutex; 6296 state->mColdGen++; 6297 mFastCaptureFutex = 0; 6298 sq->end(); 6299 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 6300 sq->push(FastCaptureStateQueue::BLOCK_UNTIL_ACKED); 6301#if 0 6302 if (kUseFastCapture == FastCapture_Dynamic) { 6303 // FIXME 6304 } 6305#endif 6306#ifdef AUDIO_WATCHDOG 6307 // FIXME 6308#endif 6309 } else { 6310 sq->end(false /*didModify*/); 6311 } 6312 } 6313 mInput->stream->common.standby(&mInput->stream->common); 6314} 6315 6316// RecordThread::createRecordTrack_l() must be called with AudioFlinger::mLock held 6317sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 6318 const sp<AudioFlinger::Client>& client, 6319 uint32_t sampleRate, 6320 audio_format_t format, 6321 audio_channel_mask_t channelMask, 6322 size_t *pFrameCount, 6323 audio_session_t sessionId, 6324 size_t *notificationFrames, 6325 int uid, 6326 IAudioFlinger::track_flags_t *flags, 6327 pid_t tid, 6328 status_t *status) 6329{ 6330 size_t frameCount = *pFrameCount; 6331 sp<RecordTrack> track; 6332 status_t lStatus; 6333 6334 // client expresses a preference for FAST, but we get the final say 6335 if (*flags & IAudioFlinger::TRACK_FAST) { 6336 if ( 6337 // we formerly checked for a callback handler (non-0 tid), 6338 // but that is no longer required for TRANSFER_OBTAIN mode 6339 // 6340 // frame count is not specified, or is exactly the pipe depth 6341 ((frameCount == 0) || (frameCount == mPipeFramesP2)) && 6342 // PCM data 6343 audio_is_linear_pcm(format) && 6344 // hardware format 6345 (format == mFormat) && 6346 // hardware channel mask 6347 (channelMask == mChannelMask) && 6348 // hardware sample rate 6349 (sampleRate == mSampleRate) && 6350 // record thread has an associated fast capture 6351 hasFastCapture() && 6352 // there are sufficient fast track slots available 6353 mFastTrackAvail 6354 ) { 6355 ALOGV("AUDIO_INPUT_FLAG_FAST accepted: frameCount=%zu mFrameCount=%zu", 6356 frameCount, mFrameCount); 6357 } else { 6358 ALOGV("AUDIO_INPUT_FLAG_FAST denied: frameCount=%zu mFrameCount=%zu mPipeFramesP2=%zu " 6359 "format=%#x isLinear=%d channelMask=%#x sampleRate=%u mSampleRate=%u " 6360 "hasFastCapture=%d tid=%d mFastTrackAvail=%d", 6361 frameCount, mFrameCount, mPipeFramesP2, 6362 format, audio_is_linear_pcm(format), channelMask, sampleRate, mSampleRate, 6363 hasFastCapture(), tid, mFastTrackAvail); 6364 *flags &= ~IAudioFlinger::TRACK_FAST; 6365 } 6366 } 6367 6368 // compute track buffer size in frames, and suggest the notification frame count 6369 if (*flags & IAudioFlinger::TRACK_FAST) { 6370 // fast track: frame count is exactly the pipe depth 6371 frameCount = mPipeFramesP2; 6372 // ignore requested notificationFrames, and always notify exactly once every HAL buffer 6373 *notificationFrames = mFrameCount; 6374 } else { 6375 // not fast track: max notification period is resampled equivalent of one HAL buffer time 6376 // or 20 ms if there is a fast capture 6377 // TODO This could be a roundupRatio inline, and const 6378 size_t maxNotificationFrames = ((int64_t) (hasFastCapture() ? mSampleRate/50 : mFrameCount) 6379 * sampleRate + mSampleRate - 1) / mSampleRate; 6380 // minimum number of notification periods is at least kMinNotifications, 6381 // and at least kMinMs rounded up to a whole notification period (minNotificationsByMs) 6382 static const size_t kMinNotifications = 3; 6383 static const uint32_t kMinMs = 30; 6384 // TODO This could be a roundupRatio inline 6385 const size_t minFramesByMs = (sampleRate * kMinMs + 1000 - 1) / 1000; 6386 // TODO This could be a roundupRatio inline 6387 const size_t minNotificationsByMs = (minFramesByMs + maxNotificationFrames - 1) / 6388 maxNotificationFrames; 6389 const size_t minFrameCount = maxNotificationFrames * 6390 max(kMinNotifications, minNotificationsByMs); 6391 frameCount = max(frameCount, minFrameCount); 6392 if (*notificationFrames == 0 || *notificationFrames > maxNotificationFrames) { 6393 *notificationFrames = maxNotificationFrames; 6394 } 6395 } 6396 *pFrameCount = frameCount; 6397 6398 lStatus = initCheck(); 6399 if (lStatus != NO_ERROR) { 6400 ALOGE("createRecordTrack_l() audio driver not initialized"); 6401 goto Exit; 6402 } 6403 6404 { // scope for mLock 6405 Mutex::Autolock _l(mLock); 6406 6407 track = new RecordTrack(this, client, sampleRate, 6408 format, channelMask, frameCount, NULL, sessionId, uid, 6409 *flags, TrackBase::TYPE_DEFAULT); 6410 6411 lStatus = track->initCheck(); 6412 if (lStatus != NO_ERROR) { 6413 ALOGE("createRecordTrack_l() initCheck failed %d; no control block?", lStatus); 6414 // track must be cleared from the caller as the caller has the AF lock 6415 goto Exit; 6416 } 6417 mTracks.add(track); 6418 6419 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 6420 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 6421 mAudioFlinger->btNrecIsOff(); 6422 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 6423 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 6424 6425 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 6426 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 6427 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 6428 // so ask activity manager to do this on our behalf 6429 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 6430 } 6431 } 6432 6433 lStatus = NO_ERROR; 6434 6435Exit: 6436 *status = lStatus; 6437 return track; 6438} 6439 6440status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, 6441 AudioSystem::sync_event_t event, 6442 audio_session_t triggerSession) 6443{ 6444 ALOGV("RecordThread::start event %d, triggerSession %d", event, triggerSession); 6445 sp<ThreadBase> strongMe = this; 6446 status_t status = NO_ERROR; 6447 6448 if (event == AudioSystem::SYNC_EVENT_NONE) { 6449 recordTrack->clearSyncStartEvent(); 6450 } else if (event != AudioSystem::SYNC_EVENT_SAME) { 6451 recordTrack->mSyncStartEvent = mAudioFlinger->createSyncEvent(event, 6452 triggerSession, 6453 recordTrack->sessionId(), 6454 syncStartEventCallback, 6455 recordTrack); 6456 // Sync event can be cancelled by the trigger session if the track is not in a 6457 // compatible state in which case we start record immediately 6458 if (recordTrack->mSyncStartEvent->isCancelled()) { 6459 recordTrack->clearSyncStartEvent(); 6460 } else { 6461 // do not wait for the event for more than AudioSystem::kSyncRecordStartTimeOutMs 6462 recordTrack->mFramesToDrop = - 6463 ((AudioSystem::kSyncRecordStartTimeOutMs * recordTrack->mSampleRate) / 1000); 6464 } 6465 } 6466 6467 { 6468 // This section is a rendezvous between binder thread executing start() and RecordThread 6469 AutoMutex lock(mLock); 6470 if (mActiveTracks.indexOf(recordTrack) >= 0) { 6471 if (recordTrack->mState == TrackBase::PAUSING) { 6472 ALOGV("active record track PAUSING -> ACTIVE"); 6473 recordTrack->mState = TrackBase::ACTIVE; 6474 } else { 6475 ALOGV("active record track state %d", recordTrack->mState); 6476 } 6477 return status; 6478 } 6479 6480 // TODO consider other ways of handling this, such as changing the state to :STARTING and 6481 // adding the track to mActiveTracks after returning from AudioSystem::startInput(), 6482 // or using a separate command thread 6483 recordTrack->mState = TrackBase::STARTING_1; 6484 mActiveTracks.add(recordTrack); 6485 mActiveTracksGen++; 6486 status_t status = NO_ERROR; 6487 if (recordTrack->isExternalTrack()) { 6488 mLock.unlock(); 6489 status = AudioSystem::startInput(mId, recordTrack->sessionId()); 6490 mLock.lock(); 6491 // FIXME should verify that recordTrack is still in mActiveTracks 6492 if (status != NO_ERROR) { 6493 mActiveTracks.remove(recordTrack); 6494 mActiveTracksGen++; 6495 recordTrack->clearSyncStartEvent(); 6496 ALOGV("RecordThread::start error %d", status); 6497 return status; 6498 } 6499 } 6500 // Catch up with current buffer indices if thread is already running. 6501 // This is what makes a new client discard all buffered data. If the track's mRsmpInFront 6502 // was initialized to some value closer to the thread's mRsmpInFront, then the track could 6503 // see previously buffered data before it called start(), but with greater risk of overrun. 6504 6505 recordTrack->mResamplerBufferProvider->reset(); 6506 // clear any converter state as new data will be discontinuous 6507 recordTrack->mRecordBufferConverter->reset(); 6508 recordTrack->mState = TrackBase::STARTING_2; 6509 // signal thread to start 6510 mWaitWorkCV.broadcast(); 6511 if (mActiveTracks.indexOf(recordTrack) < 0) { 6512 ALOGV("Record failed to start"); 6513 status = BAD_VALUE; 6514 goto startError; 6515 } 6516 return status; 6517 } 6518 6519startError: 6520 if (recordTrack->isExternalTrack()) { 6521 AudioSystem::stopInput(mId, recordTrack->sessionId()); 6522 } 6523 recordTrack->clearSyncStartEvent(); 6524 // FIXME I wonder why we do not reset the state here? 6525 return status; 6526} 6527 6528void AudioFlinger::RecordThread::syncStartEventCallback(const wp<SyncEvent>& event) 6529{ 6530 sp<SyncEvent> strongEvent = event.promote(); 6531 6532 if (strongEvent != 0) { 6533 sp<RefBase> ptr = strongEvent->cookie().promote(); 6534 if (ptr != 0) { 6535 RecordTrack *recordTrack = (RecordTrack *)ptr.get(); 6536 recordTrack->handleSyncStartEvent(strongEvent); 6537 } 6538 } 6539} 6540 6541bool AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 6542 ALOGV("RecordThread::stop"); 6543 AutoMutex _l(mLock); 6544 if (mActiveTracks.indexOf(recordTrack) != 0 || recordTrack->mState == TrackBase::PAUSING) { 6545 return false; 6546 } 6547 // note that threadLoop may still be processing the track at this point [without lock] 6548 recordTrack->mState = TrackBase::PAUSING; 6549 // do not wait for mStartStopCond if exiting 6550 if (exitPending()) { 6551 return true; 6552 } 6553 // FIXME incorrect usage of wait: no explicit predicate or loop 6554 mStartStopCond.wait(mLock); 6555 // if we have been restarted, recordTrack is in mActiveTracks here 6556 if (exitPending() || mActiveTracks.indexOf(recordTrack) != 0) { 6557 ALOGV("Record stopped OK"); 6558 return true; 6559 } 6560 return false; 6561} 6562 6563bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event __unused) const 6564{ 6565 return false; 6566} 6567 6568status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event __unused) 6569{ 6570#if 0 // This branch is currently dead code, but is preserved in case it will be needed in future 6571 if (!isValidSyncEvent(event)) { 6572 return BAD_VALUE; 6573 } 6574 6575 audio_session_t eventSession = event->triggerSession(); 6576 status_t ret = NAME_NOT_FOUND; 6577 6578 Mutex::Autolock _l(mLock); 6579 6580 for (size_t i = 0; i < mTracks.size(); i++) { 6581 sp<RecordTrack> track = mTracks[i]; 6582 if (eventSession == track->sessionId()) { 6583 (void) track->setSyncEvent(event); 6584 ret = NO_ERROR; 6585 } 6586 } 6587 return ret; 6588#else 6589 return BAD_VALUE; 6590#endif 6591} 6592 6593// destroyTrack_l() must be called with ThreadBase::mLock held 6594void AudioFlinger::RecordThread::destroyTrack_l(const sp<RecordTrack>& track) 6595{ 6596 track->terminate(); 6597 track->mState = TrackBase::STOPPED; 6598 // active tracks are removed by threadLoop() 6599 if (mActiveTracks.indexOf(track) < 0) { 6600 removeTrack_l(track); 6601 } 6602} 6603 6604void AudioFlinger::RecordThread::removeTrack_l(const sp<RecordTrack>& track) 6605{ 6606 mTracks.remove(track); 6607 // need anything related to effects here? 6608 if (track->isFastTrack()) { 6609 ALOG_ASSERT(!mFastTrackAvail); 6610 mFastTrackAvail = true; 6611 } 6612} 6613 6614void AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 6615{ 6616 dumpInternals(fd, args); 6617 dumpTracks(fd, args); 6618 dumpEffectChains(fd, args); 6619} 6620 6621void AudioFlinger::RecordThread::dumpInternals(int fd, const Vector<String16>& args) 6622{ 6623 dprintf(fd, "\nInput thread %p:\n", this); 6624 6625 dumpBase(fd, args); 6626 6627 if (mActiveTracks.size() == 0) { 6628 dprintf(fd, " No active record clients\n"); 6629 } 6630 dprintf(fd, " Fast capture thread: %s\n", hasFastCapture() ? "yes" : "no"); 6631 dprintf(fd, " Fast track available: %s\n", mFastTrackAvail ? "yes" : "no"); 6632 6633 // Make a non-atomic copy of fast capture dump state so it won't change underneath us 6634 // while we are dumping it. It may be inconsistent, but it won't mutate! 6635 // This is a large object so we place it on the heap. 6636 // FIXME 25972958: Need an intelligent copy constructor that does not touch unused pages. 6637 const FastCaptureDumpState *copy = new FastCaptureDumpState(mFastCaptureDumpState); 6638 copy->dump(fd); 6639 delete copy; 6640} 6641 6642void AudioFlinger::RecordThread::dumpTracks(int fd, const Vector<String16>& args __unused) 6643{ 6644 const size_t SIZE = 256; 6645 char buffer[SIZE]; 6646 String8 result; 6647 6648 size_t numtracks = mTracks.size(); 6649 size_t numactive = mActiveTracks.size(); 6650 size_t numactiveseen = 0; 6651 dprintf(fd, " %zu Tracks", numtracks); 6652 if (numtracks) { 6653 dprintf(fd, " of which %zu are active\n", numactive); 6654 RecordTrack::appendDumpHeader(result); 6655 for (size_t i = 0; i < numtracks ; ++i) { 6656 sp<RecordTrack> track = mTracks[i]; 6657 if (track != 0) { 6658 bool active = mActiveTracks.indexOf(track) >= 0; 6659 if (active) { 6660 numactiveseen++; 6661 } 6662 track->dump(buffer, SIZE, active); 6663 result.append(buffer); 6664 } 6665 } 6666 } else { 6667 dprintf(fd, "\n"); 6668 } 6669 6670 if (numactiveseen != numactive) { 6671 snprintf(buffer, SIZE, " The following tracks are in the active list but" 6672 " not in the track list\n"); 6673 result.append(buffer); 6674 RecordTrack::appendDumpHeader(result); 6675 for (size_t i = 0; i < numactive; ++i) { 6676 sp<RecordTrack> track = mActiveTracks[i]; 6677 if (mTracks.indexOf(track) < 0) { 6678 track->dump(buffer, SIZE, true); 6679 result.append(buffer); 6680 } 6681 } 6682 6683 } 6684 write(fd, result.string(), result.size()); 6685} 6686 6687 6688void AudioFlinger::RecordThread::ResamplerBufferProvider::reset() 6689{ 6690 sp<ThreadBase> threadBase = mRecordTrack->mThread.promote(); 6691 RecordThread *recordThread = (RecordThread *) threadBase.get(); 6692 mRsmpInFront = recordThread->mRsmpInRear; 6693 mRsmpInUnrel = 0; 6694} 6695 6696void AudioFlinger::RecordThread::ResamplerBufferProvider::sync( 6697 size_t *framesAvailable, bool *hasOverrun) 6698{ 6699 sp<ThreadBase> threadBase = mRecordTrack->mThread.promote(); 6700 RecordThread *recordThread = (RecordThread *) threadBase.get(); 6701 const int32_t rear = recordThread->mRsmpInRear; 6702 const int32_t front = mRsmpInFront; 6703 const ssize_t filled = rear - front; 6704 6705 size_t framesIn; 6706 bool overrun = false; 6707 if (filled < 0) { 6708 // should not happen, but treat like a massive overrun and re-sync 6709 framesIn = 0; 6710 mRsmpInFront = rear; 6711 overrun = true; 6712 } else if ((size_t) filled <= recordThread->mRsmpInFrames) { 6713 framesIn = (size_t) filled; 6714 } else { 6715 // client is not keeping up with server, but give it latest data 6716 framesIn = recordThread->mRsmpInFrames; 6717 mRsmpInFront = /* front = */ rear - framesIn; 6718 overrun = true; 6719 } 6720 if (framesAvailable != NULL) { 6721 *framesAvailable = framesIn; 6722 } 6723 if (hasOverrun != NULL) { 6724 *hasOverrun = overrun; 6725 } 6726} 6727 6728// AudioBufferProvider interface 6729status_t AudioFlinger::RecordThread::ResamplerBufferProvider::getNextBuffer( 6730 AudioBufferProvider::Buffer* buffer) 6731{ 6732 sp<ThreadBase> threadBase = mRecordTrack->mThread.promote(); 6733 if (threadBase == 0) { 6734 buffer->frameCount = 0; 6735 buffer->raw = NULL; 6736 return NOT_ENOUGH_DATA; 6737 } 6738 RecordThread *recordThread = (RecordThread *) threadBase.get(); 6739 int32_t rear = recordThread->mRsmpInRear; 6740 int32_t front = mRsmpInFront; 6741 ssize_t filled = rear - front; 6742 // FIXME should not be P2 (don't want to increase latency) 6743 // FIXME if client not keeping up, discard 6744 LOG_ALWAYS_FATAL_IF(!(0 <= filled && (size_t) filled <= recordThread->mRsmpInFrames)); 6745 // 'filled' may be non-contiguous, so return only the first contiguous chunk 6746 front &= recordThread->mRsmpInFramesP2 - 1; 6747 size_t part1 = recordThread->mRsmpInFramesP2 - front; 6748 if (part1 > (size_t) filled) { 6749 part1 = filled; 6750 } 6751 size_t ask = buffer->frameCount; 6752 ALOG_ASSERT(ask > 0); 6753 if (part1 > ask) { 6754 part1 = ask; 6755 } 6756 if (part1 == 0) { 6757 // out of data is fine since the resampler will return a short-count. 6758 buffer->raw = NULL; 6759 buffer->frameCount = 0; 6760 mRsmpInUnrel = 0; 6761 return NOT_ENOUGH_DATA; 6762 } 6763 6764 buffer->raw = (uint8_t*)recordThread->mRsmpInBuffer + front * recordThread->mFrameSize; 6765 buffer->frameCount = part1; 6766 mRsmpInUnrel = part1; 6767 return NO_ERROR; 6768} 6769 6770// AudioBufferProvider interface 6771void AudioFlinger::RecordThread::ResamplerBufferProvider::releaseBuffer( 6772 AudioBufferProvider::Buffer* buffer) 6773{ 6774 size_t stepCount = buffer->frameCount; 6775 if (stepCount == 0) { 6776 return; 6777 } 6778 ALOG_ASSERT(stepCount <= mRsmpInUnrel); 6779 mRsmpInUnrel -= stepCount; 6780 mRsmpInFront += stepCount; 6781 buffer->raw = NULL; 6782 buffer->frameCount = 0; 6783} 6784 6785AudioFlinger::RecordThread::RecordBufferConverter::RecordBufferConverter( 6786 audio_channel_mask_t srcChannelMask, audio_format_t srcFormat, 6787 uint32_t srcSampleRate, 6788 audio_channel_mask_t dstChannelMask, audio_format_t dstFormat, 6789 uint32_t dstSampleRate) : 6790 mSrcChannelMask(AUDIO_CHANNEL_INVALID), // updateParameters will set following vars 6791 // mSrcFormat 6792 // mSrcSampleRate 6793 // mDstChannelMask 6794 // mDstFormat 6795 // mDstSampleRate 6796 // mSrcChannelCount 6797 // mDstChannelCount 6798 // mDstFrameSize 6799 mBuf(NULL), mBufFrames(0), mBufFrameSize(0), 6800 mResampler(NULL), 6801 mIsLegacyDownmix(false), 6802 mIsLegacyUpmix(false), 6803 mRequiresFloat(false), 6804 mInputConverterProvider(NULL) 6805{ 6806 (void)updateParameters(srcChannelMask, srcFormat, srcSampleRate, 6807 dstChannelMask, dstFormat, dstSampleRate); 6808} 6809 6810AudioFlinger::RecordThread::RecordBufferConverter::~RecordBufferConverter() { 6811 free(mBuf); 6812 delete mResampler; 6813 delete mInputConverterProvider; 6814} 6815 6816size_t AudioFlinger::RecordThread::RecordBufferConverter::convert(void *dst, 6817 AudioBufferProvider *provider, size_t frames) 6818{ 6819 if (mInputConverterProvider != NULL) { 6820 mInputConverterProvider->setBufferProvider(provider); 6821 provider = mInputConverterProvider; 6822 } 6823 6824 if (mResampler == NULL) { 6825 ALOGVV("NO RESAMPLING sampleRate:%u mSrcFormat:%#x mDstFormat:%#x", 6826 mSrcSampleRate, mSrcFormat, mDstFormat); 6827 6828 AudioBufferProvider::Buffer buffer; 6829 for (size_t i = frames; i > 0; ) { 6830 buffer.frameCount = i; 6831 status_t status = provider->getNextBuffer(&buffer); 6832 if (status != OK || buffer.frameCount == 0) { 6833 frames -= i; // cannot fill request. 6834 break; 6835 } 6836 // format convert to destination buffer 6837 convertNoResampler(dst, buffer.raw, buffer.frameCount); 6838 6839 dst = (int8_t*)dst + buffer.frameCount * mDstFrameSize; 6840 i -= buffer.frameCount; 6841 provider->releaseBuffer(&buffer); 6842 } 6843 } else { 6844 ALOGVV("RESAMPLING mSrcSampleRate:%u mDstSampleRate:%u mSrcFormat:%#x mDstFormat:%#x", 6845 mSrcSampleRate, mDstSampleRate, mSrcFormat, mDstFormat); 6846 6847 // reallocate buffer if needed 6848 if (mBufFrameSize != 0 && mBufFrames < frames) { 6849 free(mBuf); 6850 mBufFrames = frames; 6851 (void)posix_memalign(&mBuf, 32, mBufFrames * mBufFrameSize); 6852 } 6853 // resampler accumulates, but we only have one source track 6854 memset(mBuf, 0, frames * mBufFrameSize); 6855 frames = mResampler->resample((int32_t*)mBuf, frames, provider); 6856 // format convert to destination buffer 6857 convertResampler(dst, mBuf, frames); 6858 } 6859 return frames; 6860} 6861 6862status_t AudioFlinger::RecordThread::RecordBufferConverter::updateParameters( 6863 audio_channel_mask_t srcChannelMask, audio_format_t srcFormat, 6864 uint32_t srcSampleRate, 6865 audio_channel_mask_t dstChannelMask, audio_format_t dstFormat, 6866 uint32_t dstSampleRate) 6867{ 6868 // quick evaluation if there is any change. 6869 if (mSrcFormat == srcFormat 6870 && mSrcChannelMask == srcChannelMask 6871 && mSrcSampleRate == srcSampleRate 6872 && mDstFormat == dstFormat 6873 && mDstChannelMask == dstChannelMask 6874 && mDstSampleRate == dstSampleRate) { 6875 return NO_ERROR; 6876 } 6877 6878 ALOGV("RecordBufferConverter updateParameters srcMask:%#x dstMask:%#x" 6879 " srcFormat:%#x dstFormat:%#x srcRate:%u dstRate:%u", 6880 srcChannelMask, dstChannelMask, srcFormat, dstFormat, srcSampleRate, dstSampleRate); 6881 const bool valid = 6882 audio_is_input_channel(srcChannelMask) 6883 && audio_is_input_channel(dstChannelMask) 6884 && audio_is_valid_format(srcFormat) && audio_is_linear_pcm(srcFormat) 6885 && audio_is_valid_format(dstFormat) && audio_is_linear_pcm(dstFormat) 6886 && (srcSampleRate <= dstSampleRate * AUDIO_RESAMPLER_DOWN_RATIO_MAX) 6887 ; // no upsampling checks for now 6888 if (!valid) { 6889 return BAD_VALUE; 6890 } 6891 6892 mSrcFormat = srcFormat; 6893 mSrcChannelMask = srcChannelMask; 6894 mSrcSampleRate = srcSampleRate; 6895 mDstFormat = dstFormat; 6896 mDstChannelMask = dstChannelMask; 6897 mDstSampleRate = dstSampleRate; 6898 6899 // compute derived parameters 6900 mSrcChannelCount = audio_channel_count_from_in_mask(srcChannelMask); 6901 mDstChannelCount = audio_channel_count_from_in_mask(dstChannelMask); 6902 mDstFrameSize = mDstChannelCount * audio_bytes_per_sample(mDstFormat); 6903 6904 // do we need to resample? 6905 delete mResampler; 6906 mResampler = NULL; 6907 if (mSrcSampleRate != mDstSampleRate) { 6908 mResampler = AudioResampler::create(AUDIO_FORMAT_PCM_FLOAT, 6909 mSrcChannelCount, mDstSampleRate); 6910 mResampler->setSampleRate(mSrcSampleRate); 6911 mResampler->setVolume(AudioMixer::UNITY_GAIN_FLOAT, AudioMixer::UNITY_GAIN_FLOAT); 6912 } 6913 6914 // are we running legacy channel conversion modes? 6915 mIsLegacyDownmix = (mSrcChannelMask == AUDIO_CHANNEL_IN_STEREO 6916 || mSrcChannelMask == AUDIO_CHANNEL_IN_FRONT_BACK) 6917 && mDstChannelMask == AUDIO_CHANNEL_IN_MONO; 6918 mIsLegacyUpmix = mSrcChannelMask == AUDIO_CHANNEL_IN_MONO 6919 && (mDstChannelMask == AUDIO_CHANNEL_IN_STEREO 6920 || mDstChannelMask == AUDIO_CHANNEL_IN_FRONT_BACK); 6921 6922 // do we need to process in float? 6923 mRequiresFloat = mResampler != NULL || mIsLegacyDownmix || mIsLegacyUpmix; 6924 6925 // do we need a staging buffer to convert for destination (we can still optimize this)? 6926 // we use mBufFrameSize > 0 to indicate both frame size as well as buffer necessity 6927 if (mResampler != NULL) { 6928 mBufFrameSize = max(mSrcChannelCount, FCC_2) 6929 * audio_bytes_per_sample(AUDIO_FORMAT_PCM_FLOAT); 6930 } else if (mIsLegacyUpmix || mIsLegacyDownmix) { // legacy modes always float 6931 mBufFrameSize = mDstChannelCount * audio_bytes_per_sample(AUDIO_FORMAT_PCM_FLOAT); 6932 } else if (mSrcChannelMask != mDstChannelMask && mDstFormat != mSrcFormat) { 6933 mBufFrameSize = mDstChannelCount * audio_bytes_per_sample(mSrcFormat); 6934 } else { 6935 mBufFrameSize = 0; 6936 } 6937 mBufFrames = 0; // force the buffer to be resized. 6938 6939 // do we need an input converter buffer provider to give us float? 6940 delete mInputConverterProvider; 6941 mInputConverterProvider = NULL; 6942 if (mRequiresFloat && mSrcFormat != AUDIO_FORMAT_PCM_FLOAT) { 6943 mInputConverterProvider = new ReformatBufferProvider( 6944 audio_channel_count_from_in_mask(mSrcChannelMask), 6945 mSrcFormat, 6946 AUDIO_FORMAT_PCM_FLOAT, 6947 256 /* provider buffer frame count */); 6948 } 6949 6950 // do we need a remixer to do channel mask conversion 6951 if (!mIsLegacyDownmix && !mIsLegacyUpmix && mSrcChannelMask != mDstChannelMask) { 6952 (void) memcpy_by_index_array_initialization_from_channel_mask( 6953 mIdxAry, ARRAY_SIZE(mIdxAry), mDstChannelMask, mSrcChannelMask); 6954 } 6955 return NO_ERROR; 6956} 6957 6958void AudioFlinger::RecordThread::RecordBufferConverter::convertNoResampler( 6959 void *dst, const void *src, size_t frames) 6960{ 6961 // src is native type unless there is legacy upmix or downmix, whereupon it is float. 6962 if (mBufFrameSize != 0 && mBufFrames < frames) { 6963 free(mBuf); 6964 mBufFrames = frames; 6965 (void)posix_memalign(&mBuf, 32, mBufFrames * mBufFrameSize); 6966 } 6967 // do we need to do legacy upmix and downmix? 6968 if (mIsLegacyUpmix || mIsLegacyDownmix) { 6969 void *dstBuf = mBuf != NULL ? mBuf : dst; 6970 if (mIsLegacyUpmix) { 6971 upmix_to_stereo_float_from_mono_float((float *)dstBuf, 6972 (const float *)src, frames); 6973 } else /*mIsLegacyDownmix */ { 6974 downmix_to_mono_float_from_stereo_float((float *)dstBuf, 6975 (const float *)src, frames); 6976 } 6977 if (mBuf != NULL) { 6978 memcpy_by_audio_format(dst, mDstFormat, mBuf, AUDIO_FORMAT_PCM_FLOAT, 6979 frames * mDstChannelCount); 6980 } 6981 return; 6982 } 6983 // do we need to do channel mask conversion? 6984 if (mSrcChannelMask != mDstChannelMask) { 6985 void *dstBuf = mBuf != NULL ? mBuf : dst; 6986 memcpy_by_index_array(dstBuf, mDstChannelCount, 6987 src, mSrcChannelCount, mIdxAry, audio_bytes_per_sample(mSrcFormat), frames); 6988 if (dstBuf == dst) { 6989 return; // format is the same 6990 } 6991 } 6992 // convert to destination buffer 6993 const void *convertBuf = mBuf != NULL ? mBuf : src; 6994 memcpy_by_audio_format(dst, mDstFormat, convertBuf, mSrcFormat, 6995 frames * mDstChannelCount); 6996} 6997 6998void AudioFlinger::RecordThread::RecordBufferConverter::convertResampler( 6999 void *dst, /*not-a-const*/ void *src, size_t frames) 7000{ 7001 // src buffer format is ALWAYS float when entering this routine 7002 if (mIsLegacyUpmix) { 7003 ; // mono to stereo already handled by resampler 7004 } else if (mIsLegacyDownmix 7005 || (mSrcChannelMask == mDstChannelMask && mSrcChannelCount == 1)) { 7006 // the resampler outputs stereo for mono input channel (a feature?) 7007 // must convert to mono 7008 downmix_to_mono_float_from_stereo_float((float *)src, 7009 (const float *)src, frames); 7010 } else if (mSrcChannelMask != mDstChannelMask) { 7011 // convert to mono channel again for channel mask conversion (could be skipped 7012 // with further optimization). 7013 if (mSrcChannelCount == 1) { 7014 downmix_to_mono_float_from_stereo_float((float *)src, 7015 (const float *)src, frames); 7016 } 7017 // convert to destination format (in place, OK as float is larger than other types) 7018 if (mDstFormat != AUDIO_FORMAT_PCM_FLOAT) { 7019 memcpy_by_audio_format(src, mDstFormat, src, AUDIO_FORMAT_PCM_FLOAT, 7020 frames * mSrcChannelCount); 7021 } 7022 // channel convert and save to dst 7023 memcpy_by_index_array(dst, mDstChannelCount, 7024 src, mSrcChannelCount, mIdxAry, audio_bytes_per_sample(mDstFormat), frames); 7025 return; 7026 } 7027 // convert to destination format and save to dst 7028 memcpy_by_audio_format(dst, mDstFormat, src, AUDIO_FORMAT_PCM_FLOAT, 7029 frames * mDstChannelCount); 7030} 7031 7032bool AudioFlinger::RecordThread::checkForNewParameter_l(const String8& keyValuePair, 7033 status_t& status) 7034{ 7035 bool reconfig = false; 7036 7037 status = NO_ERROR; 7038 7039 audio_format_t reqFormat = mFormat; 7040 uint32_t samplingRate = mSampleRate; 7041 // TODO this may change if we want to support capture from HDMI PCM multi channel (e.g on TVs). 7042 audio_channel_mask_t channelMask = audio_channel_in_mask_from_count(mChannelCount); 7043 7044 AudioParameter param = AudioParameter(keyValuePair); 7045 int value; 7046 // TODO Investigate when this code runs. Check with audio policy when a sample rate and 7047 // channel count change can be requested. Do we mandate the first client defines the 7048 // HAL sampling rate and channel count or do we allow changes on the fly? 7049 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 7050 samplingRate = value; 7051 reconfig = true; 7052 } 7053 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 7054 if (!audio_is_linear_pcm((audio_format_t) value)) { 7055 status = BAD_VALUE; 7056 } else { 7057 reqFormat = (audio_format_t) value; 7058 reconfig = true; 7059 } 7060 } 7061 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 7062 audio_channel_mask_t mask = (audio_channel_mask_t) value; 7063 if (!audio_is_input_channel(mask) || 7064 audio_channel_count_from_in_mask(mask) > FCC_8) { 7065 status = BAD_VALUE; 7066 } else { 7067 channelMask = mask; 7068 reconfig = true; 7069 } 7070 } 7071 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 7072 // do not accept frame count changes if tracks are open as the track buffer 7073 // size depends on frame count and correct behavior would not be guaranteed 7074 // if frame count is changed after track creation 7075 if (mActiveTracks.size() > 0) { 7076 status = INVALID_OPERATION; 7077 } else { 7078 reconfig = true; 7079 } 7080 } 7081 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 7082 // forward device change to effects that have requested to be 7083 // aware of attached audio device. 7084 for (size_t i = 0; i < mEffectChains.size(); i++) { 7085 mEffectChains[i]->setDevice_l(value); 7086 } 7087 7088 // store input device and output device but do not forward output device to audio HAL. 7089 // Note that status is ignored by the caller for output device 7090 // (see AudioFlinger::setParameters() 7091 if (audio_is_output_devices(value)) { 7092 mOutDevice = value; 7093 status = BAD_VALUE; 7094 } else { 7095 mInDevice = value; 7096 if (value != AUDIO_DEVICE_NONE) { 7097 mPrevInDevice = value; 7098 } 7099 // disable AEC and NS if the device is a BT SCO headset supporting those 7100 // pre processings 7101 if (mTracks.size() > 0) { 7102 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 7103 mAudioFlinger->btNrecIsOff(); 7104 for (size_t i = 0; i < mTracks.size(); i++) { 7105 sp<RecordTrack> track = mTracks[i]; 7106 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId()); 7107 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId()); 7108 } 7109 } 7110 } 7111 } 7112 if (param.getInt(String8(AudioParameter::keyInputSource), value) == NO_ERROR && 7113 mAudioSource != (audio_source_t)value) { 7114 // forward device change to effects that have requested to be 7115 // aware of attached audio device. 7116 for (size_t i = 0; i < mEffectChains.size(); i++) { 7117 mEffectChains[i]->setAudioSource_l((audio_source_t)value); 7118 } 7119 mAudioSource = (audio_source_t)value; 7120 } 7121 7122 if (status == NO_ERROR) { 7123 status = mInput->stream->common.set_parameters(&mInput->stream->common, 7124 keyValuePair.string()); 7125 if (status == INVALID_OPERATION) { 7126 inputStandBy(); 7127 status = mInput->stream->common.set_parameters(&mInput->stream->common, 7128 keyValuePair.string()); 7129 } 7130 if (reconfig) { 7131 if (status == BAD_VALUE && 7132 audio_is_linear_pcm(mInput->stream->common.get_format(&mInput->stream->common)) && 7133 audio_is_linear_pcm(reqFormat) && 7134 (mInput->stream->common.get_sample_rate(&mInput->stream->common) 7135 <= (AUDIO_RESAMPLER_DOWN_RATIO_MAX * samplingRate)) && 7136 audio_channel_count_from_in_mask( 7137 mInput->stream->common.get_channels(&mInput->stream->common)) <= FCC_8) { 7138 status = NO_ERROR; 7139 } 7140 if (status == NO_ERROR) { 7141 readInputParameters_l(); 7142 sendIoConfigEvent_l(AUDIO_INPUT_CONFIG_CHANGED); 7143 } 7144 } 7145 } 7146 7147 return reconfig; 7148} 7149 7150String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 7151{ 7152 Mutex::Autolock _l(mLock); 7153 if (initCheck() != NO_ERROR) { 7154 return String8(); 7155 } 7156 7157 char *s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 7158 const String8 out_s8(s); 7159 free(s); 7160 return out_s8; 7161} 7162 7163void AudioFlinger::RecordThread::ioConfigChanged(audio_io_config_event event, pid_t pid) { 7164 sp<AudioIoDescriptor> desc = new AudioIoDescriptor(); 7165 7166 desc->mIoHandle = mId; 7167 7168 switch (event) { 7169 case AUDIO_INPUT_OPENED: 7170 case AUDIO_INPUT_CONFIG_CHANGED: 7171 desc->mPatch = mPatch; 7172 desc->mChannelMask = mChannelMask; 7173 desc->mSamplingRate = mSampleRate; 7174 desc->mFormat = mFormat; 7175 desc->mFrameCount = mFrameCount; 7176 desc->mLatency = 0; 7177 break; 7178 7179 case AUDIO_INPUT_CLOSED: 7180 default: 7181 break; 7182 } 7183 mAudioFlinger->ioConfigChanged(event, desc, pid); 7184} 7185 7186void AudioFlinger::RecordThread::readInputParameters_l() 7187{ 7188 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 7189 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 7190 mChannelCount = audio_channel_count_from_in_mask(mChannelMask); 7191 if (mChannelCount > FCC_8) { 7192 ALOGE("HAL channel count %d > %d", mChannelCount, FCC_8); 7193 } 7194 mHALFormat = mInput->stream->common.get_format(&mInput->stream->common); 7195 mFormat = mHALFormat; 7196 if (!audio_is_linear_pcm(mFormat)) { 7197 ALOGE("HAL format %#x is not linear pcm", mFormat); 7198 } 7199 mFrameSize = audio_stream_in_frame_size(mInput->stream); 7200 mBufferSize = mInput->stream->common.get_buffer_size(&mInput->stream->common); 7201 mFrameCount = mBufferSize / mFrameSize; 7202 // This is the formula for calculating the temporary buffer size. 7203 // With 7 HAL buffers, we can guarantee ability to down-sample the input by ratio of 6:1 to 7204 // 1 full output buffer, regardless of the alignment of the available input. 7205 // The value is somewhat arbitrary, and could probably be even larger. 7206 // A larger value should allow more old data to be read after a track calls start(), 7207 // without increasing latency. 7208 // 7209 // Note this is independent of the maximum downsampling ratio permitted for capture. 7210 mRsmpInFrames = mFrameCount * 7; 7211 mRsmpInFramesP2 = roundup(mRsmpInFrames); 7212 free(mRsmpInBuffer); 7213 mRsmpInBuffer = NULL; 7214 7215 // TODO optimize audio capture buffer sizes ... 7216 // Here we calculate the size of the sliding buffer used as a source 7217 // for resampling. mRsmpInFramesP2 is currently roundup(mFrameCount * 7). 7218 // For current HAL frame counts, this is usually 2048 = 40 ms. It would 7219 // be better to have it derived from the pipe depth in the long term. 7220 // The current value is higher than necessary. However it should not add to latency. 7221 7222 // Over-allocate beyond mRsmpInFramesP2 to permit a HAL read past end of buffer 7223 size_t bufferSize = (mRsmpInFramesP2 + mFrameCount - 1) * mFrameSize; 7224 (void)posix_memalign(&mRsmpInBuffer, 32, bufferSize); 7225 memset(mRsmpInBuffer, 0, bufferSize); // if posix_memalign fails, will segv here. 7226 7227 // AudioRecord mSampleRate and mChannelCount are constant due to AudioRecord API constraints. 7228 // But if thread's mSampleRate or mChannelCount changes, how will that affect active tracks? 7229} 7230 7231uint32_t AudioFlinger::RecordThread::getInputFramesLost() 7232{ 7233 Mutex::Autolock _l(mLock); 7234 if (initCheck() != NO_ERROR) { 7235 return 0; 7236 } 7237 7238 return mInput->stream->get_input_frames_lost(mInput->stream); 7239} 7240 7241uint32_t AudioFlinger::RecordThread::hasAudioSession(audio_session_t sessionId) const 7242{ 7243 Mutex::Autolock _l(mLock); 7244 uint32_t result = 0; 7245 if (getEffectChain_l(sessionId) != 0) { 7246 result = EFFECT_SESSION; 7247 } 7248 7249 for (size_t i = 0; i < mTracks.size(); ++i) { 7250 if (sessionId == mTracks[i]->sessionId()) { 7251 result |= TRACK_SESSION; 7252 break; 7253 } 7254 } 7255 7256 return result; 7257} 7258 7259KeyedVector<audio_session_t, bool> AudioFlinger::RecordThread::sessionIds() const 7260{ 7261 KeyedVector<audio_session_t, bool> ids; 7262 Mutex::Autolock _l(mLock); 7263 for (size_t j = 0; j < mTracks.size(); ++j) { 7264 sp<RecordThread::RecordTrack> track = mTracks[j]; 7265 audio_session_t sessionId = track->sessionId(); 7266 if (ids.indexOfKey(sessionId) < 0) { 7267 ids.add(sessionId, true); 7268 } 7269 } 7270 return ids; 7271} 7272 7273AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 7274{ 7275 Mutex::Autolock _l(mLock); 7276 AudioStreamIn *input = mInput; 7277 mInput = NULL; 7278 return input; 7279} 7280 7281// this method must always be called either with ThreadBase mLock held or inside the thread loop 7282audio_stream_t* AudioFlinger::RecordThread::stream() const 7283{ 7284 if (mInput == NULL) { 7285 return NULL; 7286 } 7287 return &mInput->stream->common; 7288} 7289 7290status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 7291{ 7292 // only one chain per input thread 7293 if (mEffectChains.size() != 0) { 7294 ALOGW("addEffectChain_l() already one chain %p on thread %p", chain.get(), this); 7295 return INVALID_OPERATION; 7296 } 7297 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 7298 chain->setThread(this); 7299 chain->setInBuffer(NULL); 7300 chain->setOutBuffer(NULL); 7301 7302 checkSuspendOnAddEffectChain_l(chain); 7303 7304 // make sure enabled pre processing effects state is communicated to the HAL as we 7305 // just moved them to a new input stream. 7306 chain->syncHalEffectsState(); 7307 7308 mEffectChains.add(chain); 7309 7310 return NO_ERROR; 7311} 7312 7313size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 7314{ 7315 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 7316 ALOGW_IF(mEffectChains.size() != 1, 7317 "removeEffectChain_l() %p invalid chain size %zu on thread %p", 7318 chain.get(), mEffectChains.size(), this); 7319 if (mEffectChains.size() == 1) { 7320 mEffectChains.removeAt(0); 7321 } 7322 return 0; 7323} 7324 7325status_t AudioFlinger::RecordThread::createAudioPatch_l(const struct audio_patch *patch, 7326 audio_patch_handle_t *handle) 7327{ 7328 status_t status = NO_ERROR; 7329 7330 // store new device and send to effects 7331 mInDevice = patch->sources[0].ext.device.type; 7332 mPatch = *patch; 7333 for (size_t i = 0; i < mEffectChains.size(); i++) { 7334 mEffectChains[i]->setDevice_l(mInDevice); 7335 } 7336 7337 // disable AEC and NS if the device is a BT SCO headset supporting those 7338 // pre processings 7339 if (mTracks.size() > 0) { 7340 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 7341 mAudioFlinger->btNrecIsOff(); 7342 for (size_t i = 0; i < mTracks.size(); i++) { 7343 sp<RecordTrack> track = mTracks[i]; 7344 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId()); 7345 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId()); 7346 } 7347 } 7348 7349 // store new source and send to effects 7350 if (mAudioSource != patch->sinks[0].ext.mix.usecase.source) { 7351 mAudioSource = patch->sinks[0].ext.mix.usecase.source; 7352 for (size_t i = 0; i < mEffectChains.size(); i++) { 7353 mEffectChains[i]->setAudioSource_l(mAudioSource); 7354 } 7355 } 7356 7357 if (mInput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 7358 audio_hw_device_t *hwDevice = mInput->audioHwDev->hwDevice(); 7359 status = hwDevice->create_audio_patch(hwDevice, 7360 patch->num_sources, 7361 patch->sources, 7362 patch->num_sinks, 7363 patch->sinks, 7364 handle); 7365 } else { 7366 char *address; 7367 if (strcmp(patch->sources[0].ext.device.address, "") != 0) { 7368 address = audio_device_address_to_parameter( 7369 patch->sources[0].ext.device.type, 7370 patch->sources[0].ext.device.address); 7371 } else { 7372 address = (char *)calloc(1, 1); 7373 } 7374 AudioParameter param = AudioParameter(String8(address)); 7375 free(address); 7376 param.addInt(String8(AUDIO_PARAMETER_STREAM_ROUTING), 7377 (int)patch->sources[0].ext.device.type); 7378 param.addInt(String8(AUDIO_PARAMETER_STREAM_INPUT_SOURCE), 7379 (int)patch->sinks[0].ext.mix.usecase.source); 7380 status = mInput->stream->common.set_parameters(&mInput->stream->common, 7381 param.toString().string()); 7382 *handle = AUDIO_PATCH_HANDLE_NONE; 7383 } 7384 7385 if (mInDevice != mPrevInDevice) { 7386 sendIoConfigEvent_l(AUDIO_INPUT_CONFIG_CHANGED); 7387 mPrevInDevice = mInDevice; 7388 } 7389 7390 return status; 7391} 7392 7393status_t AudioFlinger::RecordThread::releaseAudioPatch_l(const audio_patch_handle_t handle) 7394{ 7395 status_t status = NO_ERROR; 7396 7397 mInDevice = AUDIO_DEVICE_NONE; 7398 7399 if (mInput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 7400 audio_hw_device_t *hwDevice = mInput->audioHwDev->hwDevice(); 7401 status = hwDevice->release_audio_patch(hwDevice, handle); 7402 } else { 7403 AudioParameter param; 7404 param.addInt(String8(AUDIO_PARAMETER_STREAM_ROUTING), 0); 7405 status = mInput->stream->common.set_parameters(&mInput->stream->common, 7406 param.toString().string()); 7407 } 7408 return status; 7409} 7410 7411void AudioFlinger::RecordThread::addPatchRecord(const sp<PatchRecord>& record) 7412{ 7413 Mutex::Autolock _l(mLock); 7414 mTracks.add(record); 7415} 7416 7417void AudioFlinger::RecordThread::deletePatchRecord(const sp<PatchRecord>& record) 7418{ 7419 Mutex::Autolock _l(mLock); 7420 destroyTrack_l(record); 7421} 7422 7423void AudioFlinger::RecordThread::getAudioPortConfig(struct audio_port_config *config) 7424{ 7425 ThreadBase::getAudioPortConfig(config); 7426 config->role = AUDIO_PORT_ROLE_SINK; 7427 config->ext.mix.hw_module = mInput->audioHwDev->handle(); 7428 config->ext.mix.usecase.source = mAudioSource; 7429} 7430 7431} // namespace android 7432