Threads.cpp revision 4a8308b11b92e608cdaf29f73f7919e75706f9a2
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#include "AutoPark.h" 75 76// ---------------------------------------------------------------------------- 77 78// Note: the following macro is used for extremely verbose logging message. In 79// order to run with ALOG_ASSERT turned on, we need to have LOG_NDEBUG set to 80// 0; but one side effect of this is to turn all LOGV's as well. Some messages 81// are so verbose that we want to suppress them even when we have ALOG_ASSERT 82// turned on. Do not uncomment the #def below unless you really know what you 83// are doing and want to see all of the extremely verbose messages. 84//#define VERY_VERY_VERBOSE_LOGGING 85#ifdef VERY_VERY_VERBOSE_LOGGING 86#define ALOGVV ALOGV 87#else 88#define ALOGVV(a...) do { } while(0) 89#endif 90 91// TODO: Move these macro/inlines to a header file. 92#define max(a, b) ((a) > (b) ? (a) : (b)) 93template <typename T> 94static inline T min(const T& a, const T& b) 95{ 96 return a < b ? a : b; 97} 98 99#ifndef ARRAY_SIZE 100#define ARRAY_SIZE(a) (sizeof(a) / sizeof(a[0])) 101#endif 102 103namespace android { 104 105// retry counts for buffer fill timeout 106// 50 * ~20msecs = 1 second 107static const int8_t kMaxTrackRetries = 50; 108static const int8_t kMaxTrackStartupRetries = 50; 109// allow less retry attempts on direct output thread. 110// direct outputs can be a scarce resource in audio hardware and should 111// be released as quickly as possible. 112static const int8_t kMaxTrackRetriesDirect = 2; 113// retry count before removing active track in case of underrun on offloaded thread: 114// we need to make sure that AudioTrack client has enough time to send large buffers 115//FIXME may be more appropriate if expressed in time units. Need to revise how underrun is handled 116// for offloaded tracks 117static const int8_t kMaxTrackRetriesOffload = 10; 118static const int8_t kMaxTrackStartupRetriesOffload = 100; 119 120 121// don't warn about blocked writes or record buffer overflows more often than this 122static const nsecs_t kWarningThrottleNs = seconds(5); 123 124// RecordThread loop sleep time upon application overrun or audio HAL read error 125static const int kRecordThreadSleepUs = 5000; 126 127// maximum time to wait in sendConfigEvent_l() for a status to be received 128static const nsecs_t kConfigEventTimeoutNs = seconds(2); 129 130// minimum sleep time for the mixer thread loop when tracks are active but in underrun 131static const uint32_t kMinThreadSleepTimeUs = 5000; 132// maximum divider applied to the active sleep time in the mixer thread loop 133static const uint32_t kMaxThreadSleepTimeShift = 2; 134 135// minimum normal sink buffer size, expressed in milliseconds rather than frames 136// FIXME This should be based on experimentally observed scheduling jitter 137static const uint32_t kMinNormalSinkBufferSizeMs = 20; 138// maximum normal sink buffer size 139static const uint32_t kMaxNormalSinkBufferSizeMs = 24; 140 141// minimum capture buffer size in milliseconds to _not_ need a fast capture thread 142// FIXME This should be based on experimentally observed scheduling jitter 143static const uint32_t kMinNormalCaptureBufferSizeMs = 12; 144 145// Offloaded output thread standby delay: allows track transition without going to standby 146static const nsecs_t kOffloadStandbyDelayNs = seconds(1); 147 148// Direct output thread minimum sleep time in idle or active(underrun) state 149static const nsecs_t kDirectMinSleepTimeUs = 10000; 150 151// Offloaded output bit rate in bits per second when unknown. 152// Used for sleep time calculation, so use a high default bitrate to be conservative on sleep time. 153static const uint32_t kOffloadDefaultBitRateBps = 1500000; 154 155 156// Whether to use fast mixer 157static const enum { 158 FastMixer_Never, // never initialize or use: for debugging only 159 FastMixer_Always, // always initialize and use, even if not needed: for debugging only 160 // normal mixer multiplier is 1 161 FastMixer_Static, // initialize if needed, then use all the time if initialized, 162 // multiplier is calculated based on min & max normal mixer buffer size 163 FastMixer_Dynamic, // initialize if needed, then use dynamically depending on track load, 164 // multiplier is calculated based on min & max normal mixer buffer size 165 // FIXME for FastMixer_Dynamic: 166 // Supporting this option will require fixing HALs that can't handle large writes. 167 // For example, one HAL implementation returns an error from a large write, 168 // and another HAL implementation corrupts memory, possibly in the sample rate converter. 169 // We could either fix the HAL implementations, or provide a wrapper that breaks 170 // up large writes into smaller ones, and the wrapper would need to deal with scheduler. 171} kUseFastMixer = FastMixer_Static; 172 173// Whether to use fast capture 174static const enum { 175 FastCapture_Never, // never initialize or use: for debugging only 176 FastCapture_Always, // always initialize and use, even if not needed: for debugging only 177 FastCapture_Static, // initialize if needed, then use all the time if initialized 178} kUseFastCapture = FastCapture_Static; 179 180// Priorities for requestPriority 181static const int kPriorityAudioApp = 2; 182static const int kPriorityFastMixer = 3; 183static const int kPriorityFastCapture = 3; 184 185// IAudioFlinger::createTrack() reports back to client the total size of shared memory area 186// for the track. The client then sub-divides this into smaller buffers for its use. 187// Currently the client uses N-buffering by default, but doesn't tell us about the value of N. 188// So for now we just assume that client is double-buffered for fast tracks. 189// FIXME It would be better for client to tell AudioFlinger the value of N, 190// so AudioFlinger could allocate the right amount of memory. 191// See the client's minBufCount and mNotificationFramesAct calculations for details. 192 193// This is the default value, if not specified by property. 194static const int kFastTrackMultiplier = 2; 195 196// The minimum and maximum allowed values 197static const int kFastTrackMultiplierMin = 1; 198static const int kFastTrackMultiplierMax = 2; 199 200// The actual value to use, which can be specified per-device via property af.fast_track_multiplier. 201static int sFastTrackMultiplier = kFastTrackMultiplier; 202 203// See Thread::readOnlyHeap(). 204// Initially this heap is used to allocate client buffers for "fast" AudioRecord. 205// Eventually it will be the single buffer that FastCapture writes into via HAL read(), 206// and that all "fast" AudioRecord clients read from. In either case, the size can be small. 207static const size_t kRecordThreadReadOnlyHeapSize = 0x2000; 208 209// ---------------------------------------------------------------------------- 210 211static pthread_once_t sFastTrackMultiplierOnce = PTHREAD_ONCE_INIT; 212 213static void sFastTrackMultiplierInit() 214{ 215 char value[PROPERTY_VALUE_MAX]; 216 if (property_get("af.fast_track_multiplier", value, NULL) > 0) { 217 char *endptr; 218 unsigned long ul = strtoul(value, &endptr, 0); 219 if (*endptr == '\0' && kFastTrackMultiplierMin <= ul && ul <= kFastTrackMultiplierMax) { 220 sFastTrackMultiplier = (int) ul; 221 } 222 } 223} 224 225// ---------------------------------------------------------------------------- 226 227#ifdef ADD_BATTERY_DATA 228// To collect the amplifier usage 229static void addBatteryData(uint32_t params) { 230 sp<IMediaPlayerService> service = IMediaDeathNotifier::getMediaPlayerService(); 231 if (service == NULL) { 232 // it already logged 233 return; 234 } 235 236 service->addBatteryData(params); 237} 238#endif 239 240// Track the CLOCK_BOOTTIME versus CLOCK_MONOTONIC timebase offset 241struct { 242 // call when you acquire a partial wakelock 243 void acquire(const sp<IBinder> &wakeLockToken) { 244 pthread_mutex_lock(&mLock); 245 if (wakeLockToken.get() == nullptr) { 246 adjustTimebaseOffset(&mBoottimeOffset, ExtendedTimestamp::TIMEBASE_BOOTTIME); 247 } else { 248 if (mCount == 0) { 249 adjustTimebaseOffset(&mBoottimeOffset, ExtendedTimestamp::TIMEBASE_BOOTTIME); 250 } 251 ++mCount; 252 } 253 pthread_mutex_unlock(&mLock); 254 } 255 256 // call when you release a partial wakelock. 257 void release(const sp<IBinder> &wakeLockToken) { 258 if (wakeLockToken.get() == nullptr) { 259 return; 260 } 261 pthread_mutex_lock(&mLock); 262 if (--mCount < 0) { 263 ALOGE("negative wakelock count"); 264 mCount = 0; 265 } 266 pthread_mutex_unlock(&mLock); 267 } 268 269 // retrieves the boottime timebase offset from monotonic. 270 int64_t getBoottimeOffset() { 271 pthread_mutex_lock(&mLock); 272 int64_t boottimeOffset = mBoottimeOffset; 273 pthread_mutex_unlock(&mLock); 274 return boottimeOffset; 275 } 276 277 // Adjusts the timebase offset between TIMEBASE_MONOTONIC 278 // and the selected timebase. 279 // Currently only TIMEBASE_BOOTTIME is allowed. 280 // 281 // This only needs to be called upon acquiring the first partial wakelock 282 // after all other partial wakelocks are released. 283 // 284 // We do an empirical measurement of the offset rather than parsing 285 // /proc/timer_list since the latter is not a formal kernel ABI. 286 static void adjustTimebaseOffset(int64_t *offset, ExtendedTimestamp::Timebase timebase) { 287 int clockbase; 288 switch (timebase) { 289 case ExtendedTimestamp::TIMEBASE_BOOTTIME: 290 clockbase = SYSTEM_TIME_BOOTTIME; 291 break; 292 default: 293 LOG_ALWAYS_FATAL("invalid timebase %d", timebase); 294 break; 295 } 296 // try three times to get the clock offset, choose the one 297 // with the minimum gap in measurements. 298 const int tries = 3; 299 nsecs_t bestGap, measured; 300 for (int i = 0; i < tries; ++i) { 301 const nsecs_t tmono = systemTime(SYSTEM_TIME_MONOTONIC); 302 const nsecs_t tbase = systemTime(clockbase); 303 const nsecs_t tmono2 = systemTime(SYSTEM_TIME_MONOTONIC); 304 const nsecs_t gap = tmono2 - tmono; 305 if (i == 0 || gap < bestGap) { 306 bestGap = gap; 307 measured = tbase - ((tmono + tmono2) >> 1); 308 } 309 } 310 311 // to avoid micro-adjusting, we don't change the timebase 312 // unless it is significantly different. 313 // 314 // Assumption: It probably takes more than toleranceNs to 315 // suspend and resume the device. 316 static int64_t toleranceNs = 10000; // 10 us 317 if (llabs(*offset - measured) > toleranceNs) { 318 ALOGV("Adjusting timebase offset old: %lld new: %lld", 319 (long long)*offset, (long long)measured); 320 *offset = measured; 321 } 322 } 323 324 pthread_mutex_t mLock; 325 int32_t mCount; 326 int64_t mBoottimeOffset; 327} gBoottime = { PTHREAD_MUTEX_INITIALIZER, 0, 0 }; // static, so use POD initialization 328 329// ---------------------------------------------------------------------------- 330// CPU Stats 331// ---------------------------------------------------------------------------- 332 333class CpuStats { 334public: 335 CpuStats(); 336 void sample(const String8 &title); 337#ifdef DEBUG_CPU_USAGE 338private: 339 ThreadCpuUsage mCpuUsage; // instantaneous thread CPU usage in wall clock ns 340 CentralTendencyStatistics mWcStats; // statistics on thread CPU usage in wall clock ns 341 342 CentralTendencyStatistics mHzStats; // statistics on thread CPU usage in cycles 343 344 int mCpuNum; // thread's current CPU number 345 int mCpukHz; // frequency of thread's current CPU in kHz 346#endif 347}; 348 349CpuStats::CpuStats() 350#ifdef DEBUG_CPU_USAGE 351 : mCpuNum(-1), mCpukHz(-1) 352#endif 353{ 354} 355 356void CpuStats::sample(const String8 &title 357#ifndef DEBUG_CPU_USAGE 358 __unused 359#endif 360 ) { 361#ifdef DEBUG_CPU_USAGE 362 // get current thread's delta CPU time in wall clock ns 363 double wcNs; 364 bool valid = mCpuUsage.sampleAndEnable(wcNs); 365 366 // record sample for wall clock statistics 367 if (valid) { 368 mWcStats.sample(wcNs); 369 } 370 371 // get the current CPU number 372 int cpuNum = sched_getcpu(); 373 374 // get the current CPU frequency in kHz 375 int cpukHz = mCpuUsage.getCpukHz(cpuNum); 376 377 // check if either CPU number or frequency changed 378 if (cpuNum != mCpuNum || cpukHz != mCpukHz) { 379 mCpuNum = cpuNum; 380 mCpukHz = cpukHz; 381 // ignore sample for purposes of cycles 382 valid = false; 383 } 384 385 // if no change in CPU number or frequency, then record sample for cycle statistics 386 if (valid && mCpukHz > 0) { 387 double cycles = wcNs * cpukHz * 0.000001; 388 mHzStats.sample(cycles); 389 } 390 391 unsigned n = mWcStats.n(); 392 // mCpuUsage.elapsed() is expensive, so don't call it every loop 393 if ((n & 127) == 1) { 394 long long elapsed = mCpuUsage.elapsed(); 395 if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) { 396 double perLoop = elapsed / (double) n; 397 double perLoop100 = perLoop * 0.01; 398 double perLoop1k = perLoop * 0.001; 399 double mean = mWcStats.mean(); 400 double stddev = mWcStats.stddev(); 401 double minimum = mWcStats.minimum(); 402 double maximum = mWcStats.maximum(); 403 double meanCycles = mHzStats.mean(); 404 double stddevCycles = mHzStats.stddev(); 405 double minCycles = mHzStats.minimum(); 406 double maxCycles = mHzStats.maximum(); 407 mCpuUsage.resetElapsed(); 408 mWcStats.reset(); 409 mHzStats.reset(); 410 ALOGD("CPU usage for %s over past %.1f secs\n" 411 " (%u mixer loops at %.1f mean ms per loop):\n" 412 " us per mix loop: mean=%.0f stddev=%.0f min=%.0f max=%.0f\n" 413 " %% of wall: mean=%.1f stddev=%.1f min=%.1f max=%.1f\n" 414 " MHz: mean=%.1f, stddev=%.1f, min=%.1f max=%.1f", 415 title.string(), 416 elapsed * .000000001, n, perLoop * .000001, 417 mean * .001, 418 stddev * .001, 419 minimum * .001, 420 maximum * .001, 421 mean / perLoop100, 422 stddev / perLoop100, 423 minimum / perLoop100, 424 maximum / perLoop100, 425 meanCycles / perLoop1k, 426 stddevCycles / perLoop1k, 427 minCycles / perLoop1k, 428 maxCycles / perLoop1k); 429 430 } 431 } 432#endif 433}; 434 435// ---------------------------------------------------------------------------- 436// ThreadBase 437// ---------------------------------------------------------------------------- 438 439// static 440const char *AudioFlinger::ThreadBase::threadTypeToString(AudioFlinger::ThreadBase::type_t type) 441{ 442 switch (type) { 443 case MIXER: 444 return "MIXER"; 445 case DIRECT: 446 return "DIRECT"; 447 case DUPLICATING: 448 return "DUPLICATING"; 449 case RECORD: 450 return "RECORD"; 451 case OFFLOAD: 452 return "OFFLOAD"; 453 default: 454 return "unknown"; 455 } 456} 457 458String8 devicesToString(audio_devices_t devices) 459{ 460 static const struct mapping { 461 audio_devices_t mDevices; 462 const char * mString; 463 } mappingsOut[] = { 464 {AUDIO_DEVICE_OUT_EARPIECE, "EARPIECE"}, 465 {AUDIO_DEVICE_OUT_SPEAKER, "SPEAKER"}, 466 {AUDIO_DEVICE_OUT_WIRED_HEADSET, "WIRED_HEADSET"}, 467 {AUDIO_DEVICE_OUT_WIRED_HEADPHONE, "WIRED_HEADPHONE"}, 468 {AUDIO_DEVICE_OUT_BLUETOOTH_SCO, "BLUETOOTH_SCO"}, 469 {AUDIO_DEVICE_OUT_BLUETOOTH_SCO_HEADSET, "BLUETOOTH_SCO_HEADSET"}, 470 {AUDIO_DEVICE_OUT_BLUETOOTH_SCO_CARKIT, "BLUETOOTH_SCO_CARKIT"}, 471 {AUDIO_DEVICE_OUT_BLUETOOTH_A2DP, "BLUETOOTH_A2DP"}, 472 {AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES,"BLUETOOTH_A2DP_HEADPHONES"}, 473 {AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER, "BLUETOOTH_A2DP_SPEAKER"}, 474 {AUDIO_DEVICE_OUT_AUX_DIGITAL, "AUX_DIGITAL"}, 475 {AUDIO_DEVICE_OUT_HDMI, "HDMI"}, 476 {AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET,"ANLG_DOCK_HEADSET"}, 477 {AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET,"DGTL_DOCK_HEADSET"}, 478 {AUDIO_DEVICE_OUT_USB_ACCESSORY, "USB_ACCESSORY"}, 479 {AUDIO_DEVICE_OUT_USB_DEVICE, "USB_DEVICE"}, 480 {AUDIO_DEVICE_OUT_TELEPHONY_TX, "TELEPHONY_TX"}, 481 {AUDIO_DEVICE_OUT_LINE, "LINE"}, 482 {AUDIO_DEVICE_OUT_HDMI_ARC, "HDMI_ARC"}, 483 {AUDIO_DEVICE_OUT_SPDIF, "SPDIF"}, 484 {AUDIO_DEVICE_OUT_FM, "FM"}, 485 {AUDIO_DEVICE_OUT_AUX_LINE, "AUX_LINE"}, 486 {AUDIO_DEVICE_OUT_SPEAKER_SAFE, "SPEAKER_SAFE"}, 487 {AUDIO_DEVICE_OUT_IP, "IP"}, 488 {AUDIO_DEVICE_OUT_BUS, "BUS"}, 489 {AUDIO_DEVICE_NONE, "NONE"}, // must be last 490 }, mappingsIn[] = { 491 {AUDIO_DEVICE_IN_COMMUNICATION, "COMMUNICATION"}, 492 {AUDIO_DEVICE_IN_AMBIENT, "AMBIENT"}, 493 {AUDIO_DEVICE_IN_BUILTIN_MIC, "BUILTIN_MIC"}, 494 {AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET, "BLUETOOTH_SCO_HEADSET"}, 495 {AUDIO_DEVICE_IN_WIRED_HEADSET, "WIRED_HEADSET"}, 496 {AUDIO_DEVICE_IN_AUX_DIGITAL, "AUX_DIGITAL"}, 497 {AUDIO_DEVICE_IN_VOICE_CALL, "VOICE_CALL"}, 498 {AUDIO_DEVICE_IN_TELEPHONY_RX, "TELEPHONY_RX"}, 499 {AUDIO_DEVICE_IN_BACK_MIC, "BACK_MIC"}, 500 {AUDIO_DEVICE_IN_REMOTE_SUBMIX, "REMOTE_SUBMIX"}, 501 {AUDIO_DEVICE_IN_ANLG_DOCK_HEADSET, "ANLG_DOCK_HEADSET"}, 502 {AUDIO_DEVICE_IN_DGTL_DOCK_HEADSET, "DGTL_DOCK_HEADSET"}, 503 {AUDIO_DEVICE_IN_USB_ACCESSORY, "USB_ACCESSORY"}, 504 {AUDIO_DEVICE_IN_USB_DEVICE, "USB_DEVICE"}, 505 {AUDIO_DEVICE_IN_FM_TUNER, "FM_TUNER"}, 506 {AUDIO_DEVICE_IN_TV_TUNER, "TV_TUNER"}, 507 {AUDIO_DEVICE_IN_LINE, "LINE"}, 508 {AUDIO_DEVICE_IN_SPDIF, "SPDIF"}, 509 {AUDIO_DEVICE_IN_BLUETOOTH_A2DP, "BLUETOOTH_A2DP"}, 510 {AUDIO_DEVICE_IN_LOOPBACK, "LOOPBACK"}, 511 {AUDIO_DEVICE_IN_IP, "IP"}, 512 {AUDIO_DEVICE_IN_BUS, "BUS"}, 513 {AUDIO_DEVICE_NONE, "NONE"}, // must be last 514 }; 515 String8 result; 516 audio_devices_t allDevices = AUDIO_DEVICE_NONE; 517 const mapping *entry; 518 if (devices & AUDIO_DEVICE_BIT_IN) { 519 devices &= ~AUDIO_DEVICE_BIT_IN; 520 entry = mappingsIn; 521 } else { 522 entry = mappingsOut; 523 } 524 for ( ; entry->mDevices != AUDIO_DEVICE_NONE; entry++) { 525 allDevices = (audio_devices_t) (allDevices | entry->mDevices); 526 if (devices & entry->mDevices) { 527 if (!result.isEmpty()) { 528 result.append("|"); 529 } 530 result.append(entry->mString); 531 } 532 } 533 if (devices & ~allDevices) { 534 if (!result.isEmpty()) { 535 result.append("|"); 536 } 537 result.appendFormat("0x%X", devices & ~allDevices); 538 } 539 if (result.isEmpty()) { 540 result.append(entry->mString); 541 } 542 return result; 543} 544 545String8 inputFlagsToString(audio_input_flags_t flags) 546{ 547 static const struct mapping { 548 audio_input_flags_t mFlag; 549 const char * mString; 550 } mappings[] = { 551 {AUDIO_INPUT_FLAG_FAST, "FAST"}, 552 {AUDIO_INPUT_FLAG_HW_HOTWORD, "HW_HOTWORD"}, 553 {AUDIO_INPUT_FLAG_RAW, "RAW"}, 554 {AUDIO_INPUT_FLAG_SYNC, "SYNC"}, 555 {AUDIO_INPUT_FLAG_NONE, "NONE"}, // must be last 556 }; 557 String8 result; 558 audio_input_flags_t allFlags = AUDIO_INPUT_FLAG_NONE; 559 const mapping *entry; 560 for (entry = mappings; entry->mFlag != AUDIO_INPUT_FLAG_NONE; entry++) { 561 allFlags = (audio_input_flags_t) (allFlags | entry->mFlag); 562 if (flags & entry->mFlag) { 563 if (!result.isEmpty()) { 564 result.append("|"); 565 } 566 result.append(entry->mString); 567 } 568 } 569 if (flags & ~allFlags) { 570 if (!result.isEmpty()) { 571 result.append("|"); 572 } 573 result.appendFormat("0x%X", flags & ~allFlags); 574 } 575 if (result.isEmpty()) { 576 result.append(entry->mString); 577 } 578 return result; 579} 580 581String8 outputFlagsToString(audio_output_flags_t flags) 582{ 583 static const struct mapping { 584 audio_output_flags_t mFlag; 585 const char * mString; 586 } mappings[] = { 587 {AUDIO_OUTPUT_FLAG_DIRECT, "DIRECT"}, 588 {AUDIO_OUTPUT_FLAG_PRIMARY, "PRIMARY"}, 589 {AUDIO_OUTPUT_FLAG_FAST, "FAST"}, 590 {AUDIO_OUTPUT_FLAG_DEEP_BUFFER, "DEEP_BUFFER"}, 591 {AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD,"COMPRESS_OFFLOAD"}, 592 {AUDIO_OUTPUT_FLAG_NON_BLOCKING, "NON_BLOCKING"}, 593 {AUDIO_OUTPUT_FLAG_HW_AV_SYNC, "HW_AV_SYNC"}, 594 {AUDIO_OUTPUT_FLAG_RAW, "RAW"}, 595 {AUDIO_OUTPUT_FLAG_SYNC, "SYNC"}, 596 {AUDIO_OUTPUT_FLAG_IEC958_NONAUDIO, "IEC958_NONAUDIO"}, 597 {AUDIO_OUTPUT_FLAG_NONE, "NONE"}, // must be last 598 }; 599 String8 result; 600 audio_output_flags_t allFlags = AUDIO_OUTPUT_FLAG_NONE; 601 const mapping *entry; 602 for (entry = mappings; entry->mFlag != AUDIO_OUTPUT_FLAG_NONE; entry++) { 603 allFlags = (audio_output_flags_t) (allFlags | entry->mFlag); 604 if (flags & entry->mFlag) { 605 if (!result.isEmpty()) { 606 result.append("|"); 607 } 608 result.append(entry->mString); 609 } 610 } 611 if (flags & ~allFlags) { 612 if (!result.isEmpty()) { 613 result.append("|"); 614 } 615 result.appendFormat("0x%X", flags & ~allFlags); 616 } 617 if (result.isEmpty()) { 618 result.append(entry->mString); 619 } 620 return result; 621} 622 623const char *sourceToString(audio_source_t source) 624{ 625 switch (source) { 626 case AUDIO_SOURCE_DEFAULT: return "default"; 627 case AUDIO_SOURCE_MIC: return "mic"; 628 case AUDIO_SOURCE_VOICE_UPLINK: return "voice uplink"; 629 case AUDIO_SOURCE_VOICE_DOWNLINK: return "voice downlink"; 630 case AUDIO_SOURCE_VOICE_CALL: return "voice call"; 631 case AUDIO_SOURCE_CAMCORDER: return "camcorder"; 632 case AUDIO_SOURCE_VOICE_RECOGNITION: return "voice recognition"; 633 case AUDIO_SOURCE_VOICE_COMMUNICATION: return "voice communication"; 634 case AUDIO_SOURCE_REMOTE_SUBMIX: return "remote submix"; 635 case AUDIO_SOURCE_UNPROCESSED: return "unprocessed"; 636 case AUDIO_SOURCE_FM_TUNER: return "FM tuner"; 637 case AUDIO_SOURCE_HOTWORD: return "hotword"; 638 default: return "unknown"; 639 } 640} 641 642AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, 643 audio_devices_t outDevice, audio_devices_t inDevice, type_t type, bool systemReady) 644 : Thread(false /*canCallJava*/), 645 mType(type), 646 mAudioFlinger(audioFlinger), 647 // mSampleRate, mFrameCount, mChannelMask, mChannelCount, mFrameSize, mFormat, mBufferSize 648 // are set by PlaybackThread::readOutputParameters_l() or 649 // RecordThread::readInputParameters_l() 650 //FIXME: mStandby should be true here. Is this some kind of hack? 651 mStandby(false), mOutDevice(outDevice), mInDevice(inDevice), 652 mPrevOutDevice(AUDIO_DEVICE_NONE), mPrevInDevice(AUDIO_DEVICE_NONE), 653 mAudioSource(AUDIO_SOURCE_DEFAULT), mId(id), 654 // mName will be set by concrete (non-virtual) subclass 655 mDeathRecipient(new PMDeathRecipient(this)), 656 mSystemReady(systemReady), 657 mNotifiedBatteryStart(false) 658{ 659 memset(&mPatch, 0, sizeof(struct audio_patch)); 660} 661 662AudioFlinger::ThreadBase::~ThreadBase() 663{ 664 // mConfigEvents should be empty, but just in case it isn't, free the memory it owns 665 mConfigEvents.clear(); 666 667 // do not lock the mutex in destructor 668 releaseWakeLock_l(); 669 if (mPowerManager != 0) { 670 sp<IBinder> binder = IInterface::asBinder(mPowerManager); 671 binder->unlinkToDeath(mDeathRecipient); 672 } 673} 674 675status_t AudioFlinger::ThreadBase::readyToRun() 676{ 677 status_t status = initCheck(); 678 if (status == NO_ERROR) { 679 ALOGI("AudioFlinger's thread %p ready to run", this); 680 } else { 681 ALOGE("No working audio driver found."); 682 } 683 return status; 684} 685 686void AudioFlinger::ThreadBase::exit() 687{ 688 ALOGV("ThreadBase::exit"); 689 // do any cleanup required for exit to succeed 690 preExit(); 691 { 692 // This lock prevents the following race in thread (uniprocessor for illustration): 693 // if (!exitPending()) { 694 // // context switch from here to exit() 695 // // exit() calls requestExit(), what exitPending() observes 696 // // exit() calls signal(), which is dropped since no waiters 697 // // context switch back from exit() to here 698 // mWaitWorkCV.wait(...); 699 // // now thread is hung 700 // } 701 AutoMutex lock(mLock); 702 requestExit(); 703 mWaitWorkCV.broadcast(); 704 } 705 // When Thread::requestExitAndWait is made virtual and this method is renamed to 706 // "virtual status_t requestExitAndWait()", replace by "return Thread::requestExitAndWait();" 707 requestExitAndWait(); 708} 709 710status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs) 711{ 712 ALOGV("ThreadBase::setParameters() %s", keyValuePairs.string()); 713 Mutex::Autolock _l(mLock); 714 715 return sendSetParameterConfigEvent_l(keyValuePairs); 716} 717 718// sendConfigEvent_l() must be called with ThreadBase::mLock held 719// Can temporarily release the lock if waiting for a reply from processConfigEvents_l(). 720status_t AudioFlinger::ThreadBase::sendConfigEvent_l(sp<ConfigEvent>& event) 721{ 722 status_t status = NO_ERROR; 723 724 if (event->mRequiresSystemReady && !mSystemReady) { 725 event->mWaitStatus = false; 726 mPendingConfigEvents.add(event); 727 return status; 728 } 729 mConfigEvents.add(event); 730 ALOGV("sendConfigEvent_l() num events %zu event %d", mConfigEvents.size(), event->mType); 731 mWaitWorkCV.signal(); 732 mLock.unlock(); 733 { 734 Mutex::Autolock _l(event->mLock); 735 while (event->mWaitStatus) { 736 if (event->mCond.waitRelative(event->mLock, kConfigEventTimeoutNs) != NO_ERROR) { 737 event->mStatus = TIMED_OUT; 738 event->mWaitStatus = false; 739 } 740 } 741 status = event->mStatus; 742 } 743 mLock.lock(); 744 return status; 745} 746 747void AudioFlinger::ThreadBase::sendIoConfigEvent(audio_io_config_event event, pid_t pid) 748{ 749 Mutex::Autolock _l(mLock); 750 sendIoConfigEvent_l(event, pid); 751} 752 753// sendIoConfigEvent_l() must be called with ThreadBase::mLock held 754void AudioFlinger::ThreadBase::sendIoConfigEvent_l(audio_io_config_event event, pid_t pid) 755{ 756 sp<ConfigEvent> configEvent = (ConfigEvent *)new IoConfigEvent(event, pid); 757 sendConfigEvent_l(configEvent); 758} 759 760void AudioFlinger::ThreadBase::sendPrioConfigEvent(pid_t pid, pid_t tid, int32_t prio) 761{ 762 Mutex::Autolock _l(mLock); 763 sendPrioConfigEvent_l(pid, tid, prio); 764} 765 766// sendPrioConfigEvent_l() must be called with ThreadBase::mLock held 767void AudioFlinger::ThreadBase::sendPrioConfigEvent_l(pid_t pid, pid_t tid, int32_t prio) 768{ 769 sp<ConfigEvent> configEvent = (ConfigEvent *)new PrioConfigEvent(pid, tid, prio); 770 sendConfigEvent_l(configEvent); 771} 772 773// sendSetParameterConfigEvent_l() must be called with ThreadBase::mLock held 774status_t AudioFlinger::ThreadBase::sendSetParameterConfigEvent_l(const String8& keyValuePair) 775{ 776 sp<ConfigEvent> configEvent; 777 AudioParameter param(keyValuePair); 778 int value; 779 if (param.getInt(String8(AUDIO_PARAMETER_MONO_OUTPUT), value) == NO_ERROR) { 780 setMasterMono_l(value != 0); 781 if (param.size() == 1) { 782 return NO_ERROR; // should be a solo parameter - we don't pass down 783 } 784 param.remove(String8(AUDIO_PARAMETER_MONO_OUTPUT)); 785 configEvent = new SetParameterConfigEvent(param.toString()); 786 } else { 787 configEvent = new SetParameterConfigEvent(keyValuePair); 788 } 789 return sendConfigEvent_l(configEvent); 790} 791 792status_t AudioFlinger::ThreadBase::sendCreateAudioPatchConfigEvent( 793 const struct audio_patch *patch, 794 audio_patch_handle_t *handle) 795{ 796 Mutex::Autolock _l(mLock); 797 sp<ConfigEvent> configEvent = (ConfigEvent *)new CreateAudioPatchConfigEvent(*patch, *handle); 798 status_t status = sendConfigEvent_l(configEvent); 799 if (status == NO_ERROR) { 800 CreateAudioPatchConfigEventData *data = 801 (CreateAudioPatchConfigEventData *)configEvent->mData.get(); 802 *handle = data->mHandle; 803 } 804 return status; 805} 806 807status_t AudioFlinger::ThreadBase::sendReleaseAudioPatchConfigEvent( 808 const audio_patch_handle_t handle) 809{ 810 Mutex::Autolock _l(mLock); 811 sp<ConfigEvent> configEvent = (ConfigEvent *)new ReleaseAudioPatchConfigEvent(handle); 812 return sendConfigEvent_l(configEvent); 813} 814 815 816// post condition: mConfigEvents.isEmpty() 817void AudioFlinger::ThreadBase::processConfigEvents_l() 818{ 819 bool configChanged = false; 820 821 while (!mConfigEvents.isEmpty()) { 822 ALOGV("processConfigEvents_l() remaining events %zu", mConfigEvents.size()); 823 sp<ConfigEvent> event = mConfigEvents[0]; 824 mConfigEvents.removeAt(0); 825 switch (event->mType) { 826 case CFG_EVENT_PRIO: { 827 PrioConfigEventData *data = (PrioConfigEventData *)event->mData.get(); 828 // FIXME Need to understand why this has to be done asynchronously 829 int err = requestPriority(data->mPid, data->mTid, data->mPrio, 830 true /*asynchronous*/); 831 if (err != 0) { 832 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 833 data->mPrio, data->mPid, data->mTid, err); 834 } 835 } break; 836 case CFG_EVENT_IO: { 837 IoConfigEventData *data = (IoConfigEventData *)event->mData.get(); 838 ioConfigChanged(data->mEvent, data->mPid); 839 } break; 840 case CFG_EVENT_SET_PARAMETER: { 841 SetParameterConfigEventData *data = (SetParameterConfigEventData *)event->mData.get(); 842 if (checkForNewParameter_l(data->mKeyValuePairs, event->mStatus)) { 843 configChanged = true; 844 } 845 } break; 846 case CFG_EVENT_CREATE_AUDIO_PATCH: { 847 CreateAudioPatchConfigEventData *data = 848 (CreateAudioPatchConfigEventData *)event->mData.get(); 849 event->mStatus = createAudioPatch_l(&data->mPatch, &data->mHandle); 850 } break; 851 case CFG_EVENT_RELEASE_AUDIO_PATCH: { 852 ReleaseAudioPatchConfigEventData *data = 853 (ReleaseAudioPatchConfigEventData *)event->mData.get(); 854 event->mStatus = releaseAudioPatch_l(data->mHandle); 855 } break; 856 default: 857 ALOG_ASSERT(false, "processConfigEvents_l() unknown event type %d", event->mType); 858 break; 859 } 860 { 861 Mutex::Autolock _l(event->mLock); 862 if (event->mWaitStatus) { 863 event->mWaitStatus = false; 864 event->mCond.signal(); 865 } 866 } 867 ALOGV_IF(mConfigEvents.isEmpty(), "processConfigEvents_l() DONE thread %p", this); 868 } 869 870 if (configChanged) { 871 cacheParameters_l(); 872 } 873} 874 875String8 channelMaskToString(audio_channel_mask_t mask, bool output) { 876 String8 s; 877 const audio_channel_representation_t representation = 878 audio_channel_mask_get_representation(mask); 879 880 switch (representation) { 881 case AUDIO_CHANNEL_REPRESENTATION_POSITION: { 882 if (output) { 883 if (mask & AUDIO_CHANNEL_OUT_FRONT_LEFT) s.append("front-left, "); 884 if (mask & AUDIO_CHANNEL_OUT_FRONT_RIGHT) s.append("front-right, "); 885 if (mask & AUDIO_CHANNEL_OUT_FRONT_CENTER) s.append("front-center, "); 886 if (mask & AUDIO_CHANNEL_OUT_LOW_FREQUENCY) s.append("low freq, "); 887 if (mask & AUDIO_CHANNEL_OUT_BACK_LEFT) s.append("back-left, "); 888 if (mask & AUDIO_CHANNEL_OUT_BACK_RIGHT) s.append("back-right, "); 889 if (mask & AUDIO_CHANNEL_OUT_FRONT_LEFT_OF_CENTER) s.append("front-left-of-center, "); 890 if (mask & AUDIO_CHANNEL_OUT_FRONT_RIGHT_OF_CENTER) s.append("front-right-of-center, "); 891 if (mask & AUDIO_CHANNEL_OUT_BACK_CENTER) s.append("back-center, "); 892 if (mask & AUDIO_CHANNEL_OUT_SIDE_LEFT) s.append("side-left, "); 893 if (mask & AUDIO_CHANNEL_OUT_SIDE_RIGHT) s.append("side-right, "); 894 if (mask & AUDIO_CHANNEL_OUT_TOP_CENTER) s.append("top-center ,"); 895 if (mask & AUDIO_CHANNEL_OUT_TOP_FRONT_LEFT) s.append("top-front-left, "); 896 if (mask & AUDIO_CHANNEL_OUT_TOP_FRONT_CENTER) s.append("top-front-center, "); 897 if (mask & AUDIO_CHANNEL_OUT_TOP_FRONT_RIGHT) s.append("top-front-right, "); 898 if (mask & AUDIO_CHANNEL_OUT_TOP_BACK_LEFT) s.append("top-back-left, "); 899 if (mask & AUDIO_CHANNEL_OUT_TOP_BACK_CENTER) s.append("top-back-center, " ); 900 if (mask & AUDIO_CHANNEL_OUT_TOP_BACK_RIGHT) s.append("top-back-right, " ); 901 if (mask & ~AUDIO_CHANNEL_OUT_ALL) s.append("unknown, "); 902 } else { 903 if (mask & AUDIO_CHANNEL_IN_LEFT) s.append("left, "); 904 if (mask & AUDIO_CHANNEL_IN_RIGHT) s.append("right, "); 905 if (mask & AUDIO_CHANNEL_IN_FRONT) s.append("front, "); 906 if (mask & AUDIO_CHANNEL_IN_BACK) s.append("back, "); 907 if (mask & AUDIO_CHANNEL_IN_LEFT_PROCESSED) s.append("left-processed, "); 908 if (mask & AUDIO_CHANNEL_IN_RIGHT_PROCESSED) s.append("right-processed, "); 909 if (mask & AUDIO_CHANNEL_IN_FRONT_PROCESSED) s.append("front-processed, "); 910 if (mask & AUDIO_CHANNEL_IN_BACK_PROCESSED) s.append("back-processed, "); 911 if (mask & AUDIO_CHANNEL_IN_PRESSURE) s.append("pressure, "); 912 if (mask & AUDIO_CHANNEL_IN_X_AXIS) s.append("X, "); 913 if (mask & AUDIO_CHANNEL_IN_Y_AXIS) s.append("Y, "); 914 if (mask & AUDIO_CHANNEL_IN_Z_AXIS) s.append("Z, "); 915 if (mask & AUDIO_CHANNEL_IN_VOICE_UPLINK) s.append("voice-uplink, "); 916 if (mask & AUDIO_CHANNEL_IN_VOICE_DNLINK) s.append("voice-dnlink, "); 917 if (mask & ~AUDIO_CHANNEL_IN_ALL) s.append("unknown, "); 918 } 919 const int len = s.length(); 920 if (len > 2) { 921 (void) s.lockBuffer(len); // needed? 922 s.unlockBuffer(len - 2); // remove trailing ", " 923 } 924 return s; 925 } 926 case AUDIO_CHANNEL_REPRESENTATION_INDEX: 927 s.appendFormat("index mask, bits:%#x", audio_channel_mask_get_bits(mask)); 928 return s; 929 default: 930 s.appendFormat("unknown mask, representation:%d bits:%#x", 931 representation, audio_channel_mask_get_bits(mask)); 932 return s; 933 } 934} 935 936void AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args __unused) 937{ 938 const size_t SIZE = 256; 939 char buffer[SIZE]; 940 String8 result; 941 942 bool locked = AudioFlinger::dumpTryLock(mLock); 943 if (!locked) { 944 dprintf(fd, "thread %p may be deadlocked\n", this); 945 } 946 947 dprintf(fd, " Thread name: %s\n", mThreadName); 948 dprintf(fd, " I/O handle: %d\n", mId); 949 dprintf(fd, " TID: %d\n", getTid()); 950 dprintf(fd, " Standby: %s\n", mStandby ? "yes" : "no"); 951 dprintf(fd, " Sample rate: %u Hz\n", mSampleRate); 952 dprintf(fd, " HAL frame count: %zu\n", mFrameCount); 953 dprintf(fd, " HAL format: 0x%x (%s)\n", mHALFormat, formatToString(mHALFormat)); 954 dprintf(fd, " HAL buffer size: %zu bytes\n", mBufferSize); 955 dprintf(fd, " Channel count: %u\n", mChannelCount); 956 dprintf(fd, " Channel mask: 0x%08x (%s)\n", mChannelMask, 957 channelMaskToString(mChannelMask, mType != RECORD).string()); 958 dprintf(fd, " Processing format: 0x%x (%s)\n", mFormat, formatToString(mFormat)); 959 dprintf(fd, " Processing frame size: %zu bytes\n", mFrameSize); 960 dprintf(fd, " Pending config events:"); 961 size_t numConfig = mConfigEvents.size(); 962 if (numConfig) { 963 for (size_t i = 0; i < numConfig; i++) { 964 mConfigEvents[i]->dump(buffer, SIZE); 965 dprintf(fd, "\n %s", buffer); 966 } 967 dprintf(fd, "\n"); 968 } else { 969 dprintf(fd, " none\n"); 970 } 971 dprintf(fd, " Output device: %#x (%s)\n", mOutDevice, devicesToString(mOutDevice).string()); 972 dprintf(fd, " Input device: %#x (%s)\n", mInDevice, devicesToString(mInDevice).string()); 973 dprintf(fd, " Audio source: %d (%s)\n", mAudioSource, sourceToString(mAudioSource)); 974 975 if (locked) { 976 mLock.unlock(); 977 } 978} 979 980void AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args) 981{ 982 const size_t SIZE = 256; 983 char buffer[SIZE]; 984 String8 result; 985 986 size_t numEffectChains = mEffectChains.size(); 987 snprintf(buffer, SIZE, " %zu Effect Chains\n", numEffectChains); 988 write(fd, buffer, strlen(buffer)); 989 990 for (size_t i = 0; i < numEffectChains; ++i) { 991 sp<EffectChain> chain = mEffectChains[i]; 992 if (chain != 0) { 993 chain->dump(fd, args); 994 } 995 } 996} 997 998void AudioFlinger::ThreadBase::acquireWakeLock(int uid) 999{ 1000 Mutex::Autolock _l(mLock); 1001 acquireWakeLock_l(uid); 1002} 1003 1004String16 AudioFlinger::ThreadBase::getWakeLockTag() 1005{ 1006 switch (mType) { 1007 case MIXER: 1008 return String16("AudioMix"); 1009 case DIRECT: 1010 return String16("AudioDirectOut"); 1011 case DUPLICATING: 1012 return String16("AudioDup"); 1013 case RECORD: 1014 return String16("AudioIn"); 1015 case OFFLOAD: 1016 return String16("AudioOffload"); 1017 default: 1018 ALOG_ASSERT(false); 1019 return String16("AudioUnknown"); 1020 } 1021} 1022 1023void AudioFlinger::ThreadBase::acquireWakeLock_l(int uid) 1024{ 1025 getPowerManager_l(); 1026 if (mPowerManager != 0) { 1027 sp<IBinder> binder = new BBinder(); 1028 status_t status; 1029 if (uid >= 0) { 1030 status = mPowerManager->acquireWakeLockWithUid(POWERMANAGER_PARTIAL_WAKE_LOCK, 1031 binder, 1032 getWakeLockTag(), 1033 String16("audioserver"), 1034 uid, 1035 true /* FIXME force oneway contrary to .aidl */); 1036 } else { 1037 status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK, 1038 binder, 1039 getWakeLockTag(), 1040 String16("audioserver"), 1041 true /* FIXME force oneway contrary to .aidl */); 1042 } 1043 if (status == NO_ERROR) { 1044 mWakeLockToken = binder; 1045 } 1046 ALOGV("acquireWakeLock_l() %s status %d", mThreadName, status); 1047 } 1048 1049 if (!mNotifiedBatteryStart) { 1050 BatteryNotifier::getInstance().noteStartAudio(); 1051 mNotifiedBatteryStart = true; 1052 } 1053 gBoottime.acquire(mWakeLockToken); 1054 mTimestamp.mTimebaseOffset[ExtendedTimestamp::TIMEBASE_BOOTTIME] = 1055 gBoottime.getBoottimeOffset(); 1056} 1057 1058void AudioFlinger::ThreadBase::releaseWakeLock() 1059{ 1060 Mutex::Autolock _l(mLock); 1061 releaseWakeLock_l(); 1062} 1063 1064void AudioFlinger::ThreadBase::releaseWakeLock_l() 1065{ 1066 gBoottime.release(mWakeLockToken); 1067 if (mWakeLockToken != 0) { 1068 ALOGV("releaseWakeLock_l() %s", mThreadName); 1069 if (mPowerManager != 0) { 1070 mPowerManager->releaseWakeLock(mWakeLockToken, 0, 1071 true /* FIXME force oneway contrary to .aidl */); 1072 } 1073 mWakeLockToken.clear(); 1074 } 1075 1076 if (mNotifiedBatteryStart) { 1077 BatteryNotifier::getInstance().noteStopAudio(); 1078 mNotifiedBatteryStart = false; 1079 } 1080} 1081 1082void AudioFlinger::ThreadBase::updateWakeLockUids(const SortedVector<int> &uids) { 1083 Mutex::Autolock _l(mLock); 1084 updateWakeLockUids_l(uids); 1085} 1086 1087void AudioFlinger::ThreadBase::getPowerManager_l() { 1088 if (mSystemReady && mPowerManager == 0) { 1089 // use checkService() to avoid blocking if power service is not up yet 1090 sp<IBinder> binder = 1091 defaultServiceManager()->checkService(String16("power")); 1092 if (binder == 0) { 1093 ALOGW("Thread %s cannot connect to the power manager service", mThreadName); 1094 } else { 1095 mPowerManager = interface_cast<IPowerManager>(binder); 1096 binder->linkToDeath(mDeathRecipient); 1097 } 1098 } 1099} 1100 1101void AudioFlinger::ThreadBase::updateWakeLockUids_l(const SortedVector<int> &uids) { 1102 getPowerManager_l(); 1103 if (mWakeLockToken == NULL) { // token may be NULL if AudioFlinger::systemReady() not called. 1104 if (mSystemReady) { 1105 ALOGE("no wake lock to update, but system ready!"); 1106 } else { 1107 ALOGW("no wake lock to update, system not ready yet"); 1108 } 1109 return; 1110 } 1111 if (mPowerManager != 0) { 1112 sp<IBinder> binder = new BBinder(); 1113 status_t status; 1114 status = mPowerManager->updateWakeLockUids(mWakeLockToken, uids.size(), uids.array(), 1115 true /* FIXME force oneway contrary to .aidl */); 1116 ALOGV("updateWakeLockUids_l() %s status %d", mThreadName, status); 1117 } 1118} 1119 1120void AudioFlinger::ThreadBase::clearPowerManager() 1121{ 1122 Mutex::Autolock _l(mLock); 1123 releaseWakeLock_l(); 1124 mPowerManager.clear(); 1125} 1126 1127void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who __unused) 1128{ 1129 sp<ThreadBase> thread = mThread.promote(); 1130 if (thread != 0) { 1131 thread->clearPowerManager(); 1132 } 1133 ALOGW("power manager service died !!!"); 1134} 1135 1136void AudioFlinger::ThreadBase::setEffectSuspended( 1137 const effect_uuid_t *type, bool suspend, audio_session_t sessionId) 1138{ 1139 Mutex::Autolock _l(mLock); 1140 setEffectSuspended_l(type, suspend, sessionId); 1141} 1142 1143void AudioFlinger::ThreadBase::setEffectSuspended_l( 1144 const effect_uuid_t *type, bool suspend, audio_session_t sessionId) 1145{ 1146 sp<EffectChain> chain = getEffectChain_l(sessionId); 1147 if (chain != 0) { 1148 if (type != NULL) { 1149 chain->setEffectSuspended_l(type, suspend); 1150 } else { 1151 chain->setEffectSuspendedAll_l(suspend); 1152 } 1153 } 1154 1155 updateSuspendedSessions_l(type, suspend, sessionId); 1156} 1157 1158void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain) 1159{ 1160 ssize_t index = mSuspendedSessions.indexOfKey(chain->sessionId()); 1161 if (index < 0) { 1162 return; 1163 } 1164 1165 const KeyedVector <int, sp<SuspendedSessionDesc> >& sessionEffects = 1166 mSuspendedSessions.valueAt(index); 1167 1168 for (size_t i = 0; i < sessionEffects.size(); i++) { 1169 sp<SuspendedSessionDesc> desc = sessionEffects.valueAt(i); 1170 for (int j = 0; j < desc->mRefCount; j++) { 1171 if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) { 1172 chain->setEffectSuspendedAll_l(true); 1173 } else { 1174 ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x", 1175 desc->mType.timeLow); 1176 chain->setEffectSuspended_l(&desc->mType, true); 1177 } 1178 } 1179 } 1180} 1181 1182void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type, 1183 bool suspend, 1184 audio_session_t sessionId) 1185{ 1186 ssize_t index = mSuspendedSessions.indexOfKey(sessionId); 1187 1188 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects; 1189 1190 if (suspend) { 1191 if (index >= 0) { 1192 sessionEffects = mSuspendedSessions.valueAt(index); 1193 } else { 1194 mSuspendedSessions.add(sessionId, sessionEffects); 1195 } 1196 } else { 1197 if (index < 0) { 1198 return; 1199 } 1200 sessionEffects = mSuspendedSessions.valueAt(index); 1201 } 1202 1203 1204 int key = EffectChain::kKeyForSuspendAll; 1205 if (type != NULL) { 1206 key = type->timeLow; 1207 } 1208 index = sessionEffects.indexOfKey(key); 1209 1210 sp<SuspendedSessionDesc> desc; 1211 if (suspend) { 1212 if (index >= 0) { 1213 desc = sessionEffects.valueAt(index); 1214 } else { 1215 desc = new SuspendedSessionDesc(); 1216 if (type != NULL) { 1217 desc->mType = *type; 1218 } 1219 sessionEffects.add(key, desc); 1220 ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key); 1221 } 1222 desc->mRefCount++; 1223 } else { 1224 if (index < 0) { 1225 return; 1226 } 1227 desc = sessionEffects.valueAt(index); 1228 if (--desc->mRefCount == 0) { 1229 ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key); 1230 sessionEffects.removeItemsAt(index); 1231 if (sessionEffects.isEmpty()) { 1232 ALOGV("updateSuspendedSessions_l() restore removing session %d", 1233 sessionId); 1234 mSuspendedSessions.removeItem(sessionId); 1235 } 1236 } 1237 } 1238 if (!sessionEffects.isEmpty()) { 1239 mSuspendedSessions.replaceValueFor(sessionId, sessionEffects); 1240 } 1241} 1242 1243void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 1244 bool enabled, 1245 audio_session_t sessionId) 1246{ 1247 Mutex::Autolock _l(mLock); 1248 checkSuspendOnEffectEnabled_l(effect, enabled, sessionId); 1249} 1250 1251void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 1252 bool enabled, 1253 audio_session_t sessionId) 1254{ 1255 if (mType != RECORD) { 1256 // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on 1257 // another session. This gives the priority to well behaved effect control panels 1258 // and applications not using global effects. 1259 // Enabling post processing in AUDIO_SESSION_OUTPUT_STAGE session does not affect 1260 // global effects 1261 if ((sessionId != AUDIO_SESSION_OUTPUT_MIX) && (sessionId != AUDIO_SESSION_OUTPUT_STAGE)) { 1262 setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX); 1263 } 1264 } 1265 1266 sp<EffectChain> chain = getEffectChain_l(sessionId); 1267 if (chain != 0) { 1268 chain->checkSuspendOnEffectEnabled(effect, enabled); 1269 } 1270} 1271 1272// ThreadBase::createEffect_l() must be called with AudioFlinger::mLock held 1273sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 1274 const sp<AudioFlinger::Client>& client, 1275 const sp<IEffectClient>& effectClient, 1276 int32_t priority, 1277 audio_session_t sessionId, 1278 effect_descriptor_t *desc, 1279 int *enabled, 1280 status_t *status) 1281{ 1282 sp<EffectModule> effect; 1283 sp<EffectHandle> handle; 1284 status_t lStatus; 1285 sp<EffectChain> chain; 1286 bool chainCreated = false; 1287 bool effectCreated = false; 1288 bool effectRegistered = false; 1289 1290 lStatus = initCheck(); 1291 if (lStatus != NO_ERROR) { 1292 ALOGW("createEffect_l() Audio driver not initialized."); 1293 goto Exit; 1294 } 1295 1296 // Reject any effect on Direct output threads for now, since the format of 1297 // mSinkBuffer is not guaranteed to be compatible with effect processing (PCM 16 stereo). 1298 if (mType == DIRECT) { 1299 ALOGW("createEffect_l() Cannot add effect %s on Direct output type thread %s", 1300 desc->name, mThreadName); 1301 lStatus = BAD_VALUE; 1302 goto Exit; 1303 } 1304 1305 // Reject any effect on mixer or duplicating multichannel sinks. 1306 // TODO: fix both format and multichannel issues with effects. 1307 if ((mType == MIXER || mType == DUPLICATING) && mChannelCount != FCC_2) { 1308 ALOGW("createEffect_l() Cannot add effect %s for multichannel(%d) %s threads", 1309 desc->name, mChannelCount, mType == MIXER ? "MIXER" : "DUPLICATING"); 1310 lStatus = BAD_VALUE; 1311 goto Exit; 1312 } 1313 1314 // Allow global effects only on offloaded and mixer threads 1315 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 1316 switch (mType) { 1317 case MIXER: 1318 case OFFLOAD: 1319 break; 1320 case DIRECT: 1321 case DUPLICATING: 1322 case RECORD: 1323 default: 1324 ALOGW("createEffect_l() Cannot add global effect %s on thread %s", 1325 desc->name, mThreadName); 1326 lStatus = BAD_VALUE; 1327 goto Exit; 1328 } 1329 } 1330 1331 // Only Pre processor effects are allowed on input threads and only on input threads 1332 if ((mType == RECORD) != ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { 1333 ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d", 1334 desc->name, desc->flags, mType); 1335 lStatus = BAD_VALUE; 1336 goto Exit; 1337 } 1338 1339 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 1340 1341 { // scope for mLock 1342 Mutex::Autolock _l(mLock); 1343 1344 // check for existing effect chain with the requested audio session 1345 chain = getEffectChain_l(sessionId); 1346 if (chain == 0) { 1347 // create a new chain for this session 1348 ALOGV("createEffect_l() new effect chain for session %d", sessionId); 1349 chain = new EffectChain(this, sessionId); 1350 addEffectChain_l(chain); 1351 chain->setStrategy(getStrategyForSession_l(sessionId)); 1352 chainCreated = true; 1353 } else { 1354 effect = chain->getEffectFromDesc_l(desc); 1355 } 1356 1357 ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get()); 1358 1359 if (effect == 0) { 1360 audio_unique_id_t id = mAudioFlinger->nextUniqueId(AUDIO_UNIQUE_ID_USE_EFFECT); 1361 // Check CPU and memory usage 1362 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 1363 if (lStatus != NO_ERROR) { 1364 goto Exit; 1365 } 1366 effectRegistered = true; 1367 // create a new effect module if none present in the chain 1368 effect = new EffectModule(this, chain, desc, id, sessionId); 1369 lStatus = effect->status(); 1370 if (lStatus != NO_ERROR) { 1371 goto Exit; 1372 } 1373 effect->setOffloaded(mType == OFFLOAD, mId); 1374 1375 lStatus = chain->addEffect_l(effect); 1376 if (lStatus != NO_ERROR) { 1377 goto Exit; 1378 } 1379 effectCreated = true; 1380 1381 effect->setDevice(mOutDevice); 1382 effect->setDevice(mInDevice); 1383 effect->setMode(mAudioFlinger->getMode()); 1384 effect->setAudioSource(mAudioSource); 1385 } 1386 // create effect handle and connect it to effect module 1387 handle = new EffectHandle(effect, client, effectClient, priority); 1388 lStatus = handle->initCheck(); 1389 if (lStatus == OK) { 1390 lStatus = effect->addHandle(handle.get()); 1391 } 1392 if (enabled != NULL) { 1393 *enabled = (int)effect->isEnabled(); 1394 } 1395 } 1396 1397Exit: 1398 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 1399 Mutex::Autolock _l(mLock); 1400 if (effectCreated) { 1401 chain->removeEffect_l(effect); 1402 } 1403 if (effectRegistered) { 1404 AudioSystem::unregisterEffect(effect->id()); 1405 } 1406 if (chainCreated) { 1407 removeEffectChain_l(chain); 1408 } 1409 handle.clear(); 1410 } 1411 1412 *status = lStatus; 1413 return handle; 1414} 1415 1416sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect(audio_session_t sessionId, 1417 int effectId) 1418{ 1419 Mutex::Autolock _l(mLock); 1420 return getEffect_l(sessionId, effectId); 1421} 1422 1423sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(audio_session_t sessionId, 1424 int effectId) 1425{ 1426 sp<EffectChain> chain = getEffectChain_l(sessionId); 1427 return chain != 0 ? chain->getEffectFromId_l(effectId) : 0; 1428} 1429 1430// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 1431// PlaybackThread::mLock held 1432status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 1433{ 1434 // check for existing effect chain with the requested audio session 1435 audio_session_t sessionId = effect->sessionId(); 1436 sp<EffectChain> chain = getEffectChain_l(sessionId); 1437 bool chainCreated = false; 1438 1439 ALOGD_IF((mType == OFFLOAD) && !effect->isOffloadable(), 1440 "addEffect_l() on offloaded thread %p: effect %s does not support offload flags %x", 1441 this, effect->desc().name, effect->desc().flags); 1442 1443 if (chain == 0) { 1444 // create a new chain for this session 1445 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 1446 chain = new EffectChain(this, sessionId); 1447 addEffectChain_l(chain); 1448 chain->setStrategy(getStrategyForSession_l(sessionId)); 1449 chainCreated = true; 1450 } 1451 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 1452 1453 if (chain->getEffectFromId_l(effect->id()) != 0) { 1454 ALOGW("addEffect_l() %p effect %s already present in chain %p", 1455 this, effect->desc().name, chain.get()); 1456 return BAD_VALUE; 1457 } 1458 1459 effect->setOffloaded(mType == OFFLOAD, mId); 1460 1461 status_t status = chain->addEffect_l(effect); 1462 if (status != NO_ERROR) { 1463 if (chainCreated) { 1464 removeEffectChain_l(chain); 1465 } 1466 return status; 1467 } 1468 1469 effect->setDevice(mOutDevice); 1470 effect->setDevice(mInDevice); 1471 effect->setMode(mAudioFlinger->getMode()); 1472 effect->setAudioSource(mAudioSource); 1473 return NO_ERROR; 1474} 1475 1476void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 1477 1478 ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 1479 effect_descriptor_t desc = effect->desc(); 1480 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 1481 detachAuxEffect_l(effect->id()); 1482 } 1483 1484 sp<EffectChain> chain = effect->chain().promote(); 1485 if (chain != 0) { 1486 // remove effect chain if removing last effect 1487 if (chain->removeEffect_l(effect) == 0) { 1488 removeEffectChain_l(chain); 1489 } 1490 } else { 1491 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 1492 } 1493} 1494 1495void AudioFlinger::ThreadBase::lockEffectChains_l( 1496 Vector< sp<AudioFlinger::EffectChain> >& effectChains) 1497{ 1498 effectChains = mEffectChains; 1499 for (size_t i = 0; i < mEffectChains.size(); i++) { 1500 mEffectChains[i]->lock(); 1501 } 1502} 1503 1504void AudioFlinger::ThreadBase::unlockEffectChains( 1505 const Vector< sp<AudioFlinger::EffectChain> >& effectChains) 1506{ 1507 for (size_t i = 0; i < effectChains.size(); i++) { 1508 effectChains[i]->unlock(); 1509 } 1510} 1511 1512sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(audio_session_t sessionId) 1513{ 1514 Mutex::Autolock _l(mLock); 1515 return getEffectChain_l(sessionId); 1516} 1517 1518sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(audio_session_t sessionId) 1519 const 1520{ 1521 size_t size = mEffectChains.size(); 1522 for (size_t i = 0; i < size; i++) { 1523 if (mEffectChains[i]->sessionId() == sessionId) { 1524 return mEffectChains[i]; 1525 } 1526 } 1527 return 0; 1528} 1529 1530void AudioFlinger::ThreadBase::setMode(audio_mode_t mode) 1531{ 1532 Mutex::Autolock _l(mLock); 1533 size_t size = mEffectChains.size(); 1534 for (size_t i = 0; i < size; i++) { 1535 mEffectChains[i]->setMode_l(mode); 1536 } 1537} 1538 1539void AudioFlinger::ThreadBase::getAudioPortConfig(struct audio_port_config *config) 1540{ 1541 config->type = AUDIO_PORT_TYPE_MIX; 1542 config->ext.mix.handle = mId; 1543 config->sample_rate = mSampleRate; 1544 config->format = mFormat; 1545 config->channel_mask = mChannelMask; 1546 config->config_mask = AUDIO_PORT_CONFIG_SAMPLE_RATE|AUDIO_PORT_CONFIG_CHANNEL_MASK| 1547 AUDIO_PORT_CONFIG_FORMAT; 1548} 1549 1550void AudioFlinger::ThreadBase::systemReady() 1551{ 1552 Mutex::Autolock _l(mLock); 1553 if (mSystemReady) { 1554 return; 1555 } 1556 mSystemReady = true; 1557 1558 for (size_t i = 0; i < mPendingConfigEvents.size(); i++) { 1559 sendConfigEvent_l(mPendingConfigEvents.editItemAt(i)); 1560 } 1561 mPendingConfigEvents.clear(); 1562} 1563 1564 1565// ---------------------------------------------------------------------------- 1566// Playback 1567// ---------------------------------------------------------------------------- 1568 1569AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, 1570 AudioStreamOut* output, 1571 audio_io_handle_t id, 1572 audio_devices_t device, 1573 type_t type, 1574 bool systemReady, 1575 uint32_t bitRate) 1576 : ThreadBase(audioFlinger, id, device, AUDIO_DEVICE_NONE, type, systemReady), 1577 mNormalFrameCount(0), mSinkBuffer(NULL), 1578 mMixerBufferEnabled(AudioFlinger::kEnableExtendedPrecision), 1579 mMixerBuffer(NULL), 1580 mMixerBufferSize(0), 1581 mMixerBufferFormat(AUDIO_FORMAT_INVALID), 1582 mMixerBufferValid(false), 1583 mEffectBufferEnabled(AudioFlinger::kEnableExtendedPrecision), 1584 mEffectBuffer(NULL), 1585 mEffectBufferSize(0), 1586 mEffectBufferFormat(AUDIO_FORMAT_INVALID), 1587 mEffectBufferValid(false), 1588 mSuspended(0), mBytesWritten(0), 1589 mFramesWritten(0), 1590 mActiveTracksGeneration(0), 1591 // mStreamTypes[] initialized in constructor body 1592 mOutput(output), 1593 mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false), 1594 mMixerStatus(MIXER_IDLE), 1595 mMixerStatusIgnoringFastTracks(MIXER_IDLE), 1596 mStandbyDelayNs(AudioFlinger::mStandbyTimeInNsecs), 1597 mBytesRemaining(0), 1598 mCurrentWriteLength(0), 1599 mUseAsyncWrite(false), 1600 mWriteAckSequence(0), 1601 mDrainSequence(0), 1602 mSignalPending(false), 1603 mScreenState(AudioFlinger::mScreenState), 1604 // index 0 is reserved for normal mixer's submix 1605 mFastTrackAvailMask(((1 << FastMixerState::kMaxFastTracks) - 1) & ~1), 1606 mHwSupportsPause(false), mHwPaused(false), mFlushPending(false) 1607{ 1608 snprintf(mThreadName, kThreadNameLength, "AudioOut_%X", id); 1609 mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mThreadName); 1610 1611 // Assumes constructor is called by AudioFlinger with it's mLock held, but 1612 // it would be safer to explicitly pass initial masterVolume/masterMute as 1613 // parameter. 1614 // 1615 // If the HAL we are using has support for master volume or master mute, 1616 // then do not attenuate or mute during mixing (just leave the volume at 1.0 1617 // and the mute set to false). 1618 mMasterVolume = audioFlinger->masterVolume_l(); 1619 mMasterMute = audioFlinger->masterMute_l(); 1620 if (mOutput && mOutput->audioHwDev) { 1621 if (mOutput->audioHwDev->canSetMasterVolume()) { 1622 mMasterVolume = 1.0; 1623 } 1624 1625 if (mOutput->audioHwDev->canSetMasterMute()) { 1626 mMasterMute = false; 1627 } 1628 } 1629 1630 readOutputParameters_l(); 1631 1632 // ++ operator does not compile 1633 for (audio_stream_type_t stream = AUDIO_STREAM_MIN; stream < AUDIO_STREAM_CNT; 1634 stream = (audio_stream_type_t) (stream + 1)) { 1635 mStreamTypes[stream].volume = mAudioFlinger->streamVolume_l(stream); 1636 mStreamTypes[stream].mute = mAudioFlinger->streamMute_l(stream); 1637 } 1638 1639 if (audio_has_proportional_frames(mFormat)) { 1640 mBufferDurationUs = (uint32_t)((mNormalFrameCount * 1000000LL) / mSampleRate); 1641 } else { 1642 bitRate = bitRate != 0 ? bitRate : kOffloadDefaultBitRateBps; 1643 mBufferDurationUs = (uint32_t)((mBufferSize * 8 * 1000000LL) / bitRate); 1644 } 1645} 1646 1647AudioFlinger::PlaybackThread::~PlaybackThread() 1648{ 1649 mAudioFlinger->unregisterWriter(mNBLogWriter); 1650 free(mSinkBuffer); 1651 free(mMixerBuffer); 1652 free(mEffectBuffer); 1653} 1654 1655void AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) 1656{ 1657 dumpInternals(fd, args); 1658 dumpTracks(fd, args); 1659 dumpEffectChains(fd, args); 1660} 1661 1662void AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args __unused) 1663{ 1664 const size_t SIZE = 256; 1665 char buffer[SIZE]; 1666 String8 result; 1667 1668 result.appendFormat(" Stream volumes in dB: "); 1669 for (int i = 0; i < AUDIO_STREAM_CNT; ++i) { 1670 const stream_type_t *st = &mStreamTypes[i]; 1671 if (i > 0) { 1672 result.appendFormat(", "); 1673 } 1674 result.appendFormat("%d:%.2g", i, 20.0 * log10(st->volume)); 1675 if (st->mute) { 1676 result.append("M"); 1677 } 1678 } 1679 result.append("\n"); 1680 write(fd, result.string(), result.length()); 1681 result.clear(); 1682 1683 // These values are "raw"; they will wrap around. See prepareTracks_l() for a better way. 1684 FastTrackUnderruns underruns = getFastTrackUnderruns(0); 1685 dprintf(fd, " Normal mixer raw underrun counters: partial=%u empty=%u\n", 1686 underruns.mBitFields.mPartial, underruns.mBitFields.mEmpty); 1687 1688 size_t numtracks = mTracks.size(); 1689 size_t numactive = mActiveTracks.size(); 1690 dprintf(fd, " %zu Tracks", numtracks); 1691 size_t numactiveseen = 0; 1692 if (numtracks) { 1693 dprintf(fd, " of which %zu are active\n", numactive); 1694 Track::appendDumpHeader(result); 1695 for (size_t i = 0; i < numtracks; ++i) { 1696 sp<Track> track = mTracks[i]; 1697 if (track != 0) { 1698 bool active = mActiveTracks.indexOf(track) >= 0; 1699 if (active) { 1700 numactiveseen++; 1701 } 1702 track->dump(buffer, SIZE, active); 1703 result.append(buffer); 1704 } 1705 } 1706 } else { 1707 result.append("\n"); 1708 } 1709 if (numactiveseen != numactive) { 1710 // some tracks in the active list were not in the tracks list 1711 snprintf(buffer, SIZE, " The following tracks are in the active list but" 1712 " not in the track list\n"); 1713 result.append(buffer); 1714 Track::appendDumpHeader(result); 1715 for (size_t i = 0; i < numactive; ++i) { 1716 sp<Track> track = mActiveTracks[i].promote(); 1717 if (track != 0 && mTracks.indexOf(track) < 0) { 1718 track->dump(buffer, SIZE, true); 1719 result.append(buffer); 1720 } 1721 } 1722 } 1723 1724 write(fd, result.string(), result.size()); 1725} 1726 1727void AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 1728{ 1729 dprintf(fd, "\nOutput thread %p type %d (%s):\n", this, type(), threadTypeToString(type())); 1730 1731 dumpBase(fd, args); 1732 1733 dprintf(fd, " Normal frame count: %zu\n", mNormalFrameCount); 1734 dprintf(fd, " Last write occurred (msecs): %llu\n", 1735 (unsigned long long) ns2ms(systemTime() - mLastWriteTime)); 1736 dprintf(fd, " Total writes: %d\n", mNumWrites); 1737 dprintf(fd, " Delayed writes: %d\n", mNumDelayedWrites); 1738 dprintf(fd, " Blocked in write: %s\n", mInWrite ? "yes" : "no"); 1739 dprintf(fd, " Suspend count: %d\n", mSuspended); 1740 dprintf(fd, " Sink buffer : %p\n", mSinkBuffer); 1741 dprintf(fd, " Mixer buffer: %p\n", mMixerBuffer); 1742 dprintf(fd, " Effect buffer: %p\n", mEffectBuffer); 1743 dprintf(fd, " Fast track availMask=%#x\n", mFastTrackAvailMask); 1744 dprintf(fd, " Standby delay ns=%lld\n", (long long)mStandbyDelayNs); 1745 AudioStreamOut *output = mOutput; 1746 audio_output_flags_t flags = output != NULL ? output->flags : AUDIO_OUTPUT_FLAG_NONE; 1747 String8 flagsAsString = outputFlagsToString(flags); 1748 dprintf(fd, " AudioStreamOut: %p flags %#x (%s)\n", output, flags, flagsAsString.string()); 1749} 1750 1751// Thread virtuals 1752 1753void AudioFlinger::PlaybackThread::onFirstRef() 1754{ 1755 run(mThreadName, ANDROID_PRIORITY_URGENT_AUDIO); 1756} 1757 1758// ThreadBase virtuals 1759void AudioFlinger::PlaybackThread::preExit() 1760{ 1761 ALOGV(" preExit()"); 1762 // FIXME this is using hard-coded strings but in the future, this functionality will be 1763 // converted to use audio HAL extensions required to support tunneling 1764 mOutput->stream->common.set_parameters(&mOutput->stream->common, "exiting=1"); 1765} 1766 1767// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 1768sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 1769 const sp<AudioFlinger::Client>& client, 1770 audio_stream_type_t streamType, 1771 uint32_t sampleRate, 1772 audio_format_t format, 1773 audio_channel_mask_t channelMask, 1774 size_t *pFrameCount, 1775 const sp<IMemory>& sharedBuffer, 1776 audio_session_t sessionId, 1777 IAudioFlinger::track_flags_t *flags, 1778 pid_t tid, 1779 int uid, 1780 status_t *status) 1781{ 1782 size_t frameCount = *pFrameCount; 1783 sp<Track> track; 1784 status_t lStatus; 1785 1786 // client expresses a preference for FAST, but we get the final say 1787 if (*flags & IAudioFlinger::TRACK_FAST) { 1788 if ( 1789 // PCM data 1790 audio_is_linear_pcm(format) && 1791 // TODO: extract as a data library function that checks that a computationally 1792 // expensive downmixer is not required: isFastOutputChannelConversion() 1793 (channelMask == mChannelMask || 1794 mChannelMask != AUDIO_CHANNEL_OUT_STEREO || 1795 (channelMask == AUDIO_CHANNEL_OUT_MONO 1796 /* && mChannelMask == AUDIO_CHANNEL_OUT_STEREO */)) && 1797 // hardware sample rate 1798 (sampleRate == mSampleRate) && 1799 // normal mixer has an associated fast mixer 1800 hasFastMixer() && 1801 // there are sufficient fast track slots available 1802 (mFastTrackAvailMask != 0) 1803 // FIXME test that MixerThread for this fast track has a capable output HAL 1804 // FIXME add a permission test also? 1805 ) { 1806 // static tracks can have any nonzero framecount, streaming tracks check against minimum. 1807 if (sharedBuffer == 0) { 1808 // read the fast track multiplier property the first time it is needed 1809 int ok = pthread_once(&sFastTrackMultiplierOnce, sFastTrackMultiplierInit); 1810 if (ok != 0) { 1811 ALOGE("%s pthread_once failed: %d", __func__, ok); 1812 } 1813 frameCount = max(frameCount, mFrameCount * sFastTrackMultiplier); // incl framecount 0 1814 } 1815 ALOGV("AUDIO_OUTPUT_FLAG_FAST accepted: frameCount=%zu mFrameCount=%zu", 1816 frameCount, mFrameCount); 1817 } else { 1818 ALOGV("AUDIO_OUTPUT_FLAG_FAST denied: sharedBuffer=%p frameCount=%zu " 1819 "mFrameCount=%zu format=%#x mFormat=%#x isLinear=%d channelMask=%#x " 1820 "sampleRate=%u mSampleRate=%u " 1821 "hasFastMixer=%d tid=%d fastTrackAvailMask=%#x", 1822 sharedBuffer.get(), frameCount, mFrameCount, format, mFormat, 1823 audio_is_linear_pcm(format), 1824 channelMask, sampleRate, mSampleRate, hasFastMixer(), tid, mFastTrackAvailMask); 1825 *flags &= ~IAudioFlinger::TRACK_FAST; 1826 } 1827 } 1828 // For normal PCM streaming tracks, update minimum frame count. 1829 // For compatibility with AudioTrack calculation, buffer depth is forced 1830 // to be at least 2 x the normal mixer frame count and cover audio hardware latency. 1831 // This is probably too conservative, but legacy application code may depend on it. 1832 // If you change this calculation, also review the start threshold which is related. 1833 if (!(*flags & IAudioFlinger::TRACK_FAST) 1834 && audio_has_proportional_frames(format) && sharedBuffer == 0) { 1835 // this must match AudioTrack.cpp calculateMinFrameCount(). 1836 // TODO: Move to a common library 1837 uint32_t latencyMs = mOutput->stream->get_latency(mOutput->stream); 1838 uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate); 1839 if (minBufCount < 2) { 1840 minBufCount = 2; 1841 } 1842 // For normal mixing tracks, if speed is > 1.0f (normal), AudioTrack 1843 // or the client should compute and pass in a larger buffer request. 1844 size_t minFrameCount = 1845 minBufCount * sourceFramesNeededWithTimestretch( 1846 sampleRate, mNormalFrameCount, 1847 mSampleRate, AUDIO_TIMESTRETCH_SPEED_NORMAL /*speed*/); 1848 if (frameCount < minFrameCount) { // including frameCount == 0 1849 frameCount = minFrameCount; 1850 } 1851 } 1852 *pFrameCount = frameCount; 1853 1854 switch (mType) { 1855 1856 case DIRECT: 1857 if (audio_is_linear_pcm(format)) { // TODO maybe use audio_has_proportional_frames()? 1858 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1859 ALOGE("createTrack_l() Bad parameter: sampleRate %u format %#x, channelMask 0x%08x " 1860 "for output %p with format %#x", 1861 sampleRate, format, channelMask, mOutput, mFormat); 1862 lStatus = BAD_VALUE; 1863 goto Exit; 1864 } 1865 } 1866 break; 1867 1868 case OFFLOAD: 1869 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1870 ALOGE("createTrack_l() Bad parameter: sampleRate %d format %#x, channelMask 0x%08x \"" 1871 "for output %p with format %#x", 1872 sampleRate, format, channelMask, mOutput, mFormat); 1873 lStatus = BAD_VALUE; 1874 goto Exit; 1875 } 1876 break; 1877 1878 default: 1879 if (!audio_is_linear_pcm(format)) { 1880 ALOGE("createTrack_l() Bad parameter: format %#x \"" 1881 "for output %p with format %#x", 1882 format, mOutput, mFormat); 1883 lStatus = BAD_VALUE; 1884 goto Exit; 1885 } 1886 if (sampleRate > mSampleRate * AUDIO_RESAMPLER_DOWN_RATIO_MAX) { 1887 ALOGE("Sample rate out of range: %u mSampleRate %u", sampleRate, mSampleRate); 1888 lStatus = BAD_VALUE; 1889 goto Exit; 1890 } 1891 break; 1892 1893 } 1894 1895 lStatus = initCheck(); 1896 if (lStatus != NO_ERROR) { 1897 ALOGE("createTrack_l() audio driver not initialized"); 1898 goto Exit; 1899 } 1900 1901 { // scope for mLock 1902 Mutex::Autolock _l(mLock); 1903 1904 // all tracks in same audio session must share the same routing strategy otherwise 1905 // conflicts will happen when tracks are moved from one output to another by audio policy 1906 // manager 1907 uint32_t strategy = AudioSystem::getStrategyForStream(streamType); 1908 for (size_t i = 0; i < mTracks.size(); ++i) { 1909 sp<Track> t = mTracks[i]; 1910 if (t != 0 && t->isExternalTrack()) { 1911 uint32_t actual = AudioSystem::getStrategyForStream(t->streamType()); 1912 if (sessionId == t->sessionId() && strategy != actual) { 1913 ALOGE("createTrack_l() mismatched strategy; expected %u but found %u", 1914 strategy, actual); 1915 lStatus = BAD_VALUE; 1916 goto Exit; 1917 } 1918 } 1919 } 1920 1921 track = new Track(this, client, streamType, sampleRate, format, 1922 channelMask, frameCount, NULL, sharedBuffer, 1923 sessionId, uid, *flags, TrackBase::TYPE_DEFAULT); 1924 1925 lStatus = track != 0 ? track->initCheck() : (status_t) NO_MEMORY; 1926 if (lStatus != NO_ERROR) { 1927 ALOGE("createTrack_l() initCheck failed %d; no control block?", lStatus); 1928 // track must be cleared from the caller as the caller has the AF lock 1929 goto Exit; 1930 } 1931 mTracks.add(track); 1932 1933 sp<EffectChain> chain = getEffectChain_l(sessionId); 1934 if (chain != 0) { 1935 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1936 track->setMainBuffer(chain->inBuffer()); 1937 chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType())); 1938 chain->incTrackCnt(); 1939 } 1940 1941 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 1942 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 1943 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 1944 // so ask activity manager to do this on our behalf 1945 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 1946 } 1947 } 1948 1949 lStatus = NO_ERROR; 1950 1951Exit: 1952 *status = lStatus; 1953 return track; 1954} 1955 1956uint32_t AudioFlinger::PlaybackThread::correctLatency_l(uint32_t latency) const 1957{ 1958 return latency; 1959} 1960 1961uint32_t AudioFlinger::PlaybackThread::latency() const 1962{ 1963 Mutex::Autolock _l(mLock); 1964 return latency_l(); 1965} 1966uint32_t AudioFlinger::PlaybackThread::latency_l() const 1967{ 1968 if (initCheck() == NO_ERROR) { 1969 return correctLatency_l(mOutput->stream->get_latency(mOutput->stream)); 1970 } else { 1971 return 0; 1972 } 1973} 1974 1975void AudioFlinger::PlaybackThread::setMasterVolume(float value) 1976{ 1977 Mutex::Autolock _l(mLock); 1978 // Don't apply master volume in SW if our HAL can do it for us. 1979 if (mOutput && mOutput->audioHwDev && 1980 mOutput->audioHwDev->canSetMasterVolume()) { 1981 mMasterVolume = 1.0; 1982 } else { 1983 mMasterVolume = value; 1984 } 1985} 1986 1987void AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1988{ 1989 Mutex::Autolock _l(mLock); 1990 // Don't apply master mute in SW if our HAL can do it for us. 1991 if (mOutput && mOutput->audioHwDev && 1992 mOutput->audioHwDev->canSetMasterMute()) { 1993 mMasterMute = false; 1994 } else { 1995 mMasterMute = muted; 1996 } 1997} 1998 1999void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) 2000{ 2001 Mutex::Autolock _l(mLock); 2002 mStreamTypes[stream].volume = value; 2003 broadcast_l(); 2004} 2005 2006void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) 2007{ 2008 Mutex::Autolock _l(mLock); 2009 mStreamTypes[stream].mute = muted; 2010 broadcast_l(); 2011} 2012 2013float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const 2014{ 2015 Mutex::Autolock _l(mLock); 2016 return mStreamTypes[stream].volume; 2017} 2018 2019// addTrack_l() must be called with ThreadBase::mLock held 2020status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 2021{ 2022 status_t status = ALREADY_EXISTS; 2023 2024 if (mActiveTracks.indexOf(track) < 0) { 2025 // the track is newly added, make sure it fills up all its 2026 // buffers before playing. This is to ensure the client will 2027 // effectively get the latency it requested. 2028 if (track->isExternalTrack()) { 2029 TrackBase::track_state state = track->mState; 2030 mLock.unlock(); 2031 status = AudioSystem::startOutput(mId, track->streamType(), 2032 track->sessionId()); 2033 mLock.lock(); 2034 // abort track was stopped/paused while we released the lock 2035 if (state != track->mState) { 2036 if (status == NO_ERROR) { 2037 mLock.unlock(); 2038 AudioSystem::stopOutput(mId, track->streamType(), 2039 track->sessionId()); 2040 mLock.lock(); 2041 } 2042 return INVALID_OPERATION; 2043 } 2044 // abort if start is rejected by audio policy manager 2045 if (status != NO_ERROR) { 2046 return PERMISSION_DENIED; 2047 } 2048#ifdef ADD_BATTERY_DATA 2049 // to track the speaker usage 2050 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 2051#endif 2052 } 2053 2054 // set retry count for buffer fill 2055 if (track->isOffloaded()) { 2056 track->mRetryCount = kMaxTrackStartupRetriesOffload; 2057 } else { 2058 track->mRetryCount = kMaxTrackStartupRetries; 2059 } 2060 2061 track->mFillingUpStatus = track->sharedBuffer() != 0 ? Track::FS_FILLED : Track::FS_FILLING; 2062 track->mResetDone = false; 2063 track->mPresentationCompleteFrames = 0; 2064 mActiveTracks.add(track); 2065 mWakeLockUids.add(track->uid()); 2066 mActiveTracksGeneration++; 2067 mLatestActiveTrack = track; 2068 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 2069 if (chain != 0) { 2070 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), 2071 track->sessionId()); 2072 chain->incActiveTrackCnt(); 2073 } 2074 2075 status = NO_ERROR; 2076 } 2077 2078 onAddNewTrack_l(); 2079 return status; 2080} 2081 2082bool AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 2083{ 2084 track->terminate(); 2085 // active tracks are removed by threadLoop() 2086 bool trackActive = (mActiveTracks.indexOf(track) >= 0); 2087 track->mState = TrackBase::STOPPED; 2088 if (!trackActive) { 2089 removeTrack_l(track); 2090 } else if (track->isFastTrack() || track->isOffloaded() || track->isDirect()) { 2091 track->mState = TrackBase::STOPPING_1; 2092 } 2093 2094 return trackActive; 2095} 2096 2097void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 2098{ 2099 track->triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 2100 mTracks.remove(track); 2101 deleteTrackName_l(track->name()); 2102 // redundant as track is about to be destroyed, for dumpsys only 2103 track->mName = -1; 2104 if (track->isFastTrack()) { 2105 int index = track->mFastIndex; 2106 ALOG_ASSERT(0 < index && index < (int)FastMixerState::kMaxFastTracks); 2107 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << index))); 2108 mFastTrackAvailMask |= 1 << index; 2109 // redundant as track is about to be destroyed, for dumpsys only 2110 track->mFastIndex = -1; 2111 } 2112 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 2113 if (chain != 0) { 2114 chain->decTrackCnt(); 2115 } 2116} 2117 2118void AudioFlinger::PlaybackThread::broadcast_l() 2119{ 2120 // Thread could be blocked waiting for async 2121 // so signal it to handle state changes immediately 2122 // If threadLoop is currently unlocked a signal of mWaitWorkCV will 2123 // be lost so we also flag to prevent it blocking on mWaitWorkCV 2124 mSignalPending = true; 2125 mWaitWorkCV.broadcast(); 2126} 2127 2128String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 2129{ 2130 Mutex::Autolock _l(mLock); 2131 if (initCheck() != NO_ERROR) { 2132 return String8(); 2133 } 2134 2135 char *s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 2136 const String8 out_s8(s); 2137 free(s); 2138 return out_s8; 2139} 2140 2141void AudioFlinger::PlaybackThread::ioConfigChanged(audio_io_config_event event, pid_t pid) { 2142 sp<AudioIoDescriptor> desc = new AudioIoDescriptor(); 2143 ALOGV("PlaybackThread::ioConfigChanged, thread %p, event %d", this, event); 2144 2145 desc->mIoHandle = mId; 2146 2147 switch (event) { 2148 case AUDIO_OUTPUT_OPENED: 2149 case AUDIO_OUTPUT_CONFIG_CHANGED: 2150 desc->mPatch = mPatch; 2151 desc->mChannelMask = mChannelMask; 2152 desc->mSamplingRate = mSampleRate; 2153 desc->mFormat = mFormat; 2154 desc->mFrameCount = mNormalFrameCount; // FIXME see 2155 // AudioFlinger::frameCount(audio_io_handle_t) 2156 desc->mFrameCountHAL = mFrameCount; 2157 desc->mLatency = latency_l(); 2158 break; 2159 2160 case AUDIO_OUTPUT_CLOSED: 2161 default: 2162 break; 2163 } 2164 mAudioFlinger->ioConfigChanged(event, desc, pid); 2165} 2166 2167void AudioFlinger::PlaybackThread::writeCallback() 2168{ 2169 ALOG_ASSERT(mCallbackThread != 0); 2170 mCallbackThread->resetWriteBlocked(); 2171} 2172 2173void AudioFlinger::PlaybackThread::drainCallback() 2174{ 2175 ALOG_ASSERT(mCallbackThread != 0); 2176 mCallbackThread->resetDraining(); 2177} 2178 2179void AudioFlinger::PlaybackThread::resetWriteBlocked(uint32_t sequence) 2180{ 2181 Mutex::Autolock _l(mLock); 2182 // reject out of sequence requests 2183 if ((mWriteAckSequence & 1) && (sequence == mWriteAckSequence)) { 2184 mWriteAckSequence &= ~1; 2185 mWaitWorkCV.signal(); 2186 } 2187} 2188 2189void AudioFlinger::PlaybackThread::resetDraining(uint32_t sequence) 2190{ 2191 Mutex::Autolock _l(mLock); 2192 // reject out of sequence requests 2193 if ((mDrainSequence & 1) && (sequence == mDrainSequence)) { 2194 mDrainSequence &= ~1; 2195 mWaitWorkCV.signal(); 2196 } 2197} 2198 2199// static 2200int AudioFlinger::PlaybackThread::asyncCallback(stream_callback_event_t event, 2201 void *param __unused, 2202 void *cookie) 2203{ 2204 AudioFlinger::PlaybackThread *me = (AudioFlinger::PlaybackThread *)cookie; 2205 ALOGV("asyncCallback() event %d", event); 2206 switch (event) { 2207 case STREAM_CBK_EVENT_WRITE_READY: 2208 me->writeCallback(); 2209 break; 2210 case STREAM_CBK_EVENT_DRAIN_READY: 2211 me->drainCallback(); 2212 break; 2213 default: 2214 ALOGW("asyncCallback() unknown event %d", event); 2215 break; 2216 } 2217 return 0; 2218} 2219 2220void AudioFlinger::PlaybackThread::readOutputParameters_l() 2221{ 2222 // unfortunately we have no way of recovering from errors here, hence the LOG_ALWAYS_FATAL 2223 mSampleRate = mOutput->getSampleRate(); 2224 mChannelMask = mOutput->getChannelMask(); 2225 if (!audio_is_output_channel(mChannelMask)) { 2226 LOG_ALWAYS_FATAL("HAL channel mask %#x not valid for output", mChannelMask); 2227 } 2228 if ((mType == MIXER || mType == DUPLICATING) 2229 && !isValidPcmSinkChannelMask(mChannelMask)) { 2230 LOG_ALWAYS_FATAL("HAL channel mask %#x not supported for mixed output", 2231 mChannelMask); 2232 } 2233 mChannelCount = audio_channel_count_from_out_mask(mChannelMask); 2234 2235 // Get actual HAL format. 2236 mHALFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 2237 // Get format from the shim, which will be different than the HAL format 2238 // if playing compressed audio over HDMI passthrough. 2239 mFormat = mOutput->getFormat(); 2240 if (!audio_is_valid_format(mFormat)) { 2241 LOG_ALWAYS_FATAL("HAL format %#x not valid for output", mFormat); 2242 } 2243 if ((mType == MIXER || mType == DUPLICATING) 2244 && !isValidPcmSinkFormat(mFormat)) { 2245 LOG_FATAL("HAL format %#x not supported for mixed output", 2246 mFormat); 2247 } 2248 mFrameSize = mOutput->getFrameSize(); 2249 mBufferSize = mOutput->stream->common.get_buffer_size(&mOutput->stream->common); 2250 mFrameCount = mBufferSize / mFrameSize; 2251 if (mFrameCount & 15) { 2252 ALOGW("HAL output buffer size is %zu frames but AudioMixer requires multiples of 16 frames", 2253 mFrameCount); 2254 } 2255 2256 if ((mOutput->flags & AUDIO_OUTPUT_FLAG_NON_BLOCKING) && 2257 (mOutput->stream->set_callback != NULL)) { 2258 if (mOutput->stream->set_callback(mOutput->stream, 2259 AudioFlinger::PlaybackThread::asyncCallback, this) == 0) { 2260 mUseAsyncWrite = true; 2261 mCallbackThread = new AudioFlinger::AsyncCallbackThread(this); 2262 } 2263 } 2264 2265 mHwSupportsPause = false; 2266 if (mOutput->flags & AUDIO_OUTPUT_FLAG_DIRECT) { 2267 if (mOutput->stream->pause != NULL) { 2268 if (mOutput->stream->resume != NULL) { 2269 mHwSupportsPause = true; 2270 } else { 2271 ALOGW("direct output implements pause but not resume"); 2272 } 2273 } else if (mOutput->stream->resume != NULL) { 2274 ALOGW("direct output implements resume but not pause"); 2275 } 2276 } 2277 if (!mHwSupportsPause && mOutput->flags & AUDIO_OUTPUT_FLAG_HW_AV_SYNC) { 2278 LOG_ALWAYS_FATAL("HW_AV_SYNC requested but HAL does not implement pause and resume"); 2279 } 2280 2281 if (mType == DUPLICATING && mMixerBufferEnabled && mEffectBufferEnabled) { 2282 // For best precision, we use float instead of the associated output 2283 // device format (typically PCM 16 bit). 2284 2285 mFormat = AUDIO_FORMAT_PCM_FLOAT; 2286 mFrameSize = mChannelCount * audio_bytes_per_sample(mFormat); 2287 mBufferSize = mFrameSize * mFrameCount; 2288 2289 // TODO: We currently use the associated output device channel mask and sample rate. 2290 // (1) Perhaps use the ORed channel mask of all downstream MixerThreads 2291 // (if a valid mask) to avoid premature downmix. 2292 // (2) Perhaps use the maximum sample rate of all downstream MixerThreads 2293 // instead of the output device sample rate to avoid loss of high frequency information. 2294 // This may need to be updated as MixerThread/OutputTracks are added and not here. 2295 } 2296 2297 // Calculate size of normal sink buffer relative to the HAL output buffer size 2298 double multiplier = 1.0; 2299 if (mType == MIXER && (kUseFastMixer == FastMixer_Static || 2300 kUseFastMixer == FastMixer_Dynamic)) { 2301 size_t minNormalFrameCount = (kMinNormalSinkBufferSizeMs * mSampleRate) / 1000; 2302 size_t maxNormalFrameCount = (kMaxNormalSinkBufferSizeMs * mSampleRate) / 1000; 2303 // round up minimum and round down maximum to nearest 16 frames to satisfy AudioMixer 2304 minNormalFrameCount = (minNormalFrameCount + 15) & ~15; 2305 maxNormalFrameCount = maxNormalFrameCount & ~15; 2306 if (maxNormalFrameCount < minNormalFrameCount) { 2307 maxNormalFrameCount = minNormalFrameCount; 2308 } 2309 multiplier = (double) minNormalFrameCount / (double) mFrameCount; 2310 if (multiplier <= 1.0) { 2311 multiplier = 1.0; 2312 } else if (multiplier <= 2.0) { 2313 if (2 * mFrameCount <= maxNormalFrameCount) { 2314 multiplier = 2.0; 2315 } else { 2316 multiplier = (double) maxNormalFrameCount / (double) mFrameCount; 2317 } 2318 } else { 2319 // prefer an even multiplier, for compatibility with doubling of fast tracks due to HAL 2320 // SRC (it would be unusual for the normal sink buffer size to not be a multiple of fast 2321 // track, but we sometimes have to do this to satisfy the maximum frame count 2322 // constraint) 2323 // FIXME this rounding up should not be done if no HAL SRC 2324 uint32_t truncMult = (uint32_t) multiplier; 2325 if ((truncMult & 1)) { 2326 if ((truncMult + 1) * mFrameCount <= maxNormalFrameCount) { 2327 ++truncMult; 2328 } 2329 } 2330 multiplier = (double) truncMult; 2331 } 2332 } 2333 mNormalFrameCount = multiplier * mFrameCount; 2334 // round up to nearest 16 frames to satisfy AudioMixer 2335 if (mType == MIXER || mType == DUPLICATING) { 2336 mNormalFrameCount = (mNormalFrameCount + 15) & ~15; 2337 } 2338 ALOGI("HAL output buffer size %zu frames, normal sink buffer size %zu frames", mFrameCount, 2339 mNormalFrameCount); 2340 2341 // Check if we want to throttle the processing to no more than 2x normal rate 2342 mThreadThrottle = property_get_bool("af.thread.throttle", true /* default_value */); 2343 mThreadThrottleTimeMs = 0; 2344 mThreadThrottleEndMs = 0; 2345 mHalfBufferMs = mNormalFrameCount * 1000 / (2 * mSampleRate); 2346 2347 // mSinkBuffer is the sink buffer. Size is always multiple-of-16 frames. 2348 // Originally this was int16_t[] array, need to remove legacy implications. 2349 free(mSinkBuffer); 2350 mSinkBuffer = NULL; 2351 // For sink buffer size, we use the frame size from the downstream sink to avoid problems 2352 // with non PCM formats for compressed music, e.g. AAC, and Offload threads. 2353 const size_t sinkBufferSize = mNormalFrameCount * mFrameSize; 2354 (void)posix_memalign(&mSinkBuffer, 32, sinkBufferSize); 2355 2356 // We resize the mMixerBuffer according to the requirements of the sink buffer which 2357 // drives the output. 2358 free(mMixerBuffer); 2359 mMixerBuffer = NULL; 2360 if (mMixerBufferEnabled) { 2361 mMixerBufferFormat = AUDIO_FORMAT_PCM_FLOAT; // also valid: AUDIO_FORMAT_PCM_16_BIT. 2362 mMixerBufferSize = mNormalFrameCount * mChannelCount 2363 * audio_bytes_per_sample(mMixerBufferFormat); 2364 (void)posix_memalign(&mMixerBuffer, 32, mMixerBufferSize); 2365 } 2366 free(mEffectBuffer); 2367 mEffectBuffer = NULL; 2368 if (mEffectBufferEnabled) { 2369 mEffectBufferFormat = AUDIO_FORMAT_PCM_16_BIT; // Note: Effects support 16b only 2370 mEffectBufferSize = mNormalFrameCount * mChannelCount 2371 * audio_bytes_per_sample(mEffectBufferFormat); 2372 (void)posix_memalign(&mEffectBuffer, 32, mEffectBufferSize); 2373 } 2374 2375 // force reconfiguration of effect chains and engines to take new buffer size and audio 2376 // parameters into account 2377 // Note that mLock is not held when readOutputParameters_l() is called from the constructor 2378 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 2379 // matter. 2380 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 2381 Vector< sp<EffectChain> > effectChains = mEffectChains; 2382 for (size_t i = 0; i < effectChains.size(); i ++) { 2383 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 2384 } 2385} 2386 2387 2388status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames) 2389{ 2390 if (halFrames == NULL || dspFrames == NULL) { 2391 return BAD_VALUE; 2392 } 2393 Mutex::Autolock _l(mLock); 2394 if (initCheck() != NO_ERROR) { 2395 return INVALID_OPERATION; 2396 } 2397 int64_t framesWritten = mBytesWritten / mFrameSize; 2398 *halFrames = framesWritten; 2399 2400 if (isSuspended()) { 2401 // return an estimation of rendered frames when the output is suspended 2402 size_t latencyFrames = (latency_l() * mSampleRate) / 1000; 2403 *dspFrames = (uint32_t) 2404 (framesWritten >= (int64_t)latencyFrames ? framesWritten - latencyFrames : 0); 2405 return NO_ERROR; 2406 } else { 2407 status_t status; 2408 uint32_t frames; 2409 status = mOutput->getRenderPosition(&frames); 2410 *dspFrames = (size_t)frames; 2411 return status; 2412 } 2413} 2414 2415uint32_t AudioFlinger::PlaybackThread::hasAudioSession(audio_session_t sessionId) const 2416{ 2417 Mutex::Autolock _l(mLock); 2418 uint32_t result = 0; 2419 if (getEffectChain_l(sessionId) != 0) { 2420 result = EFFECT_SESSION; 2421 } 2422 2423 for (size_t i = 0; i < mTracks.size(); ++i) { 2424 sp<Track> track = mTracks[i]; 2425 if (sessionId == track->sessionId() && !track->isInvalid()) { 2426 result |= TRACK_SESSION; 2427 break; 2428 } 2429 } 2430 2431 return result; 2432} 2433 2434uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(audio_session_t sessionId) 2435{ 2436 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 2437 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 2438 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 2439 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 2440 } 2441 for (size_t i = 0; i < mTracks.size(); i++) { 2442 sp<Track> track = mTracks[i]; 2443 if (sessionId == track->sessionId() && !track->isInvalid()) { 2444 return AudioSystem::getStrategyForStream(track->streamType()); 2445 } 2446 } 2447 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 2448} 2449 2450 2451AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 2452{ 2453 Mutex::Autolock _l(mLock); 2454 return mOutput; 2455} 2456 2457AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 2458{ 2459 Mutex::Autolock _l(mLock); 2460 AudioStreamOut *output = mOutput; 2461 mOutput = NULL; 2462 // FIXME FastMixer might also have a raw ptr to mOutputSink; 2463 // must push a NULL and wait for ack 2464 mOutputSink.clear(); 2465 mPipeSink.clear(); 2466 mNormalSink.clear(); 2467 return output; 2468} 2469 2470// this method must always be called either with ThreadBase mLock held or inside the thread loop 2471audio_stream_t* AudioFlinger::PlaybackThread::stream() const 2472{ 2473 if (mOutput == NULL) { 2474 return NULL; 2475 } 2476 return &mOutput->stream->common; 2477} 2478 2479uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() const 2480{ 2481 return (uint32_t)((uint32_t)((mNormalFrameCount * 1000) / mSampleRate) * 1000); 2482} 2483 2484status_t AudioFlinger::PlaybackThread::setSyncEvent(const sp<SyncEvent>& event) 2485{ 2486 if (!isValidSyncEvent(event)) { 2487 return BAD_VALUE; 2488 } 2489 2490 Mutex::Autolock _l(mLock); 2491 2492 for (size_t i = 0; i < mTracks.size(); ++i) { 2493 sp<Track> track = mTracks[i]; 2494 if (event->triggerSession() == track->sessionId()) { 2495 (void) track->setSyncEvent(event); 2496 return NO_ERROR; 2497 } 2498 } 2499 2500 return NAME_NOT_FOUND; 2501} 2502 2503bool AudioFlinger::PlaybackThread::isValidSyncEvent(const sp<SyncEvent>& event) const 2504{ 2505 return event->type() == AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE; 2506} 2507 2508void AudioFlinger::PlaybackThread::threadLoop_removeTracks( 2509 const Vector< sp<Track> >& tracksToRemove) 2510{ 2511 size_t count = tracksToRemove.size(); 2512 if (count > 0) { 2513 for (size_t i = 0 ; i < count ; i++) { 2514 const sp<Track>& track = tracksToRemove.itemAt(i); 2515 if (track->isExternalTrack()) { 2516 AudioSystem::stopOutput(mId, track->streamType(), 2517 track->sessionId()); 2518#ifdef ADD_BATTERY_DATA 2519 // to track the speaker usage 2520 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 2521#endif 2522 if (track->isTerminated()) { 2523 AudioSystem::releaseOutput(mId, track->streamType(), 2524 track->sessionId()); 2525 } 2526 } 2527 } 2528 } 2529} 2530 2531void AudioFlinger::PlaybackThread::checkSilentMode_l() 2532{ 2533 if (!mMasterMute) { 2534 char value[PROPERTY_VALUE_MAX]; 2535 if (mOutDevice == AUDIO_DEVICE_OUT_REMOTE_SUBMIX) { 2536 ALOGD("ro.audio.silent will be ignored for threads on AUDIO_DEVICE_OUT_REMOTE_SUBMIX"); 2537 return; 2538 } 2539 if (property_get("ro.audio.silent", value, "0") > 0) { 2540 char *endptr; 2541 unsigned long ul = strtoul(value, &endptr, 0); 2542 if (*endptr == '\0' && ul != 0) { 2543 ALOGD("Silence is golden"); 2544 // The setprop command will not allow a property to be changed after 2545 // the first time it is set, so we don't have to worry about un-muting. 2546 setMasterMute_l(true); 2547 } 2548 } 2549 } 2550} 2551 2552// shared by MIXER and DIRECT, overridden by DUPLICATING 2553ssize_t AudioFlinger::PlaybackThread::threadLoop_write() 2554{ 2555 // FIXME rewrite to reduce number of system calls 2556 mLastWriteTime = systemTime(); 2557 mInWrite = true; 2558 ssize_t bytesWritten; 2559 const size_t offset = mCurrentWriteLength - mBytesRemaining; 2560 2561 // If an NBAIO sink is present, use it to write the normal mixer's submix 2562 if (mNormalSink != 0) { 2563 2564 const size_t count = mBytesRemaining / mFrameSize; 2565 2566 ATRACE_BEGIN("write"); 2567 // update the setpoint when AudioFlinger::mScreenState changes 2568 uint32_t screenState = AudioFlinger::mScreenState; 2569 if (screenState != mScreenState) { 2570 mScreenState = screenState; 2571 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 2572 if (pipe != NULL) { 2573 pipe->setAvgFrames((mScreenState & 1) ? 2574 (pipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2575 } 2576 } 2577 ssize_t framesWritten = mNormalSink->write((char *)mSinkBuffer + offset, count); 2578 ATRACE_END(); 2579 if (framesWritten > 0) { 2580 bytesWritten = framesWritten * mFrameSize; 2581 } else { 2582 bytesWritten = framesWritten; 2583 } 2584 // otherwise use the HAL / AudioStreamOut directly 2585 } else { 2586 // Direct output and offload threads 2587 2588 if (mUseAsyncWrite) { 2589 ALOGW_IF(mWriteAckSequence & 1, "threadLoop_write(): out of sequence write request"); 2590 mWriteAckSequence += 2; 2591 mWriteAckSequence |= 1; 2592 ALOG_ASSERT(mCallbackThread != 0); 2593 mCallbackThread->setWriteBlocked(mWriteAckSequence); 2594 } 2595 // FIXME We should have an implementation of timestamps for direct output threads. 2596 // They are used e.g for multichannel PCM playback over HDMI. 2597 bytesWritten = mOutput->write((char *)mSinkBuffer + offset, mBytesRemaining); 2598 2599 if (mUseAsyncWrite && 2600 ((bytesWritten < 0) || (bytesWritten == (ssize_t)mBytesRemaining))) { 2601 // do not wait for async callback in case of error of full write 2602 mWriteAckSequence &= ~1; 2603 ALOG_ASSERT(mCallbackThread != 0); 2604 mCallbackThread->setWriteBlocked(mWriteAckSequence); 2605 } 2606 } 2607 2608 mNumWrites++; 2609 mInWrite = false; 2610 mStandby = false; 2611 return bytesWritten; 2612} 2613 2614void AudioFlinger::PlaybackThread::threadLoop_drain() 2615{ 2616 if (mOutput->stream->drain) { 2617 ALOGV("draining %s", (mMixerStatus == MIXER_DRAIN_TRACK) ? "early" : "full"); 2618 if (mUseAsyncWrite) { 2619 ALOGW_IF(mDrainSequence & 1, "threadLoop_drain(): out of sequence drain request"); 2620 mDrainSequence |= 1; 2621 ALOG_ASSERT(mCallbackThread != 0); 2622 mCallbackThread->setDraining(mDrainSequence); 2623 } 2624 mOutput->stream->drain(mOutput->stream, 2625 (mMixerStatus == MIXER_DRAIN_TRACK) ? AUDIO_DRAIN_EARLY_NOTIFY 2626 : AUDIO_DRAIN_ALL); 2627 } 2628} 2629 2630void AudioFlinger::PlaybackThread::threadLoop_exit() 2631{ 2632 { 2633 Mutex::Autolock _l(mLock); 2634 for (size_t i = 0; i < mTracks.size(); i++) { 2635 sp<Track> track = mTracks[i]; 2636 track->invalidate(); 2637 } 2638 } 2639} 2640 2641/* 2642The derived values that are cached: 2643 - mSinkBufferSize from frame count * frame size 2644 - mActiveSleepTimeUs from activeSleepTimeUs() 2645 - mIdleSleepTimeUs from idleSleepTimeUs() 2646 - mStandbyDelayNs from mActiveSleepTimeUs (DIRECT only) or forced to at least 2647 kDefaultStandbyTimeInNsecs when connected to an A2DP device. 2648 - maxPeriod from frame count and sample rate (MIXER only) 2649 2650The parameters that affect these derived values are: 2651 - frame count 2652 - frame size 2653 - sample rate 2654 - device type: A2DP or not 2655 - device latency 2656 - format: PCM or not 2657 - active sleep time 2658 - idle sleep time 2659*/ 2660 2661void AudioFlinger::PlaybackThread::cacheParameters_l() 2662{ 2663 mSinkBufferSize = mNormalFrameCount * mFrameSize; 2664 mActiveSleepTimeUs = activeSleepTimeUs(); 2665 mIdleSleepTimeUs = idleSleepTimeUs(); 2666 2667 // make sure standby delay is not too short when connected to an A2DP sink to avoid 2668 // truncating audio when going to standby. 2669 mStandbyDelayNs = AudioFlinger::mStandbyTimeInNsecs; 2670 if ((mOutDevice & AUDIO_DEVICE_OUT_ALL_A2DP) != 0) { 2671 if (mStandbyDelayNs < kDefaultStandbyTimeInNsecs) { 2672 mStandbyDelayNs = kDefaultStandbyTimeInNsecs; 2673 } 2674 } 2675} 2676 2677void AudioFlinger::PlaybackThread::invalidateTracks(audio_stream_type_t streamType) 2678{ 2679 ALOGV("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %zu", 2680 this, streamType, mTracks.size()); 2681 Mutex::Autolock _l(mLock); 2682 2683 size_t size = mTracks.size(); 2684 for (size_t i = 0; i < size; i++) { 2685 sp<Track> t = mTracks[i]; 2686 if (t->streamType() == streamType && t->isExternalTrack()) { 2687 t->invalidate(); 2688 } 2689 } 2690} 2691 2692status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 2693{ 2694 audio_session_t session = chain->sessionId(); 2695 int16_t* buffer = reinterpret_cast<int16_t*>(mEffectBufferEnabled 2696 ? mEffectBuffer : mSinkBuffer); 2697 bool ownsBuffer = false; 2698 2699 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 2700 if (session > AUDIO_SESSION_OUTPUT_MIX) { 2701 // Only one effect chain can be present in direct output thread and it uses 2702 // the sink buffer as input 2703 if (mType != DIRECT) { 2704 size_t numSamples = mNormalFrameCount * mChannelCount; 2705 buffer = new int16_t[numSamples]; 2706 memset(buffer, 0, numSamples * sizeof(int16_t)); 2707 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 2708 ownsBuffer = true; 2709 } 2710 2711 // Attach all tracks with same session ID to this chain. 2712 for (size_t i = 0; i < mTracks.size(); ++i) { 2713 sp<Track> track = mTracks[i]; 2714 if (session == track->sessionId()) { 2715 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), 2716 buffer); 2717 track->setMainBuffer(buffer); 2718 chain->incTrackCnt(); 2719 } 2720 } 2721 2722 // indicate all active tracks in the chain 2723 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 2724 sp<Track> track = mActiveTracks[i].promote(); 2725 if (track == 0) { 2726 continue; 2727 } 2728 if (session == track->sessionId()) { 2729 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 2730 chain->incActiveTrackCnt(); 2731 } 2732 } 2733 } 2734 chain->setThread(this); 2735 chain->setInBuffer(buffer, ownsBuffer); 2736 chain->setOutBuffer(reinterpret_cast<int16_t*>(mEffectBufferEnabled 2737 ? mEffectBuffer : mSinkBuffer)); 2738 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 2739 // chains list in order to be processed last as it contains output stage effects. 2740 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 2741 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 2742 // after track specific effects and before output stage. 2743 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 2744 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX. 2745 // Effect chain for other sessions are inserted at beginning of effect 2746 // chains list to be processed before output mix effects. Relative order between other 2747 // sessions is not important. 2748 static_assert(AUDIO_SESSION_OUTPUT_MIX == 0 && 2749 AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX, 2750 "audio_session_t constants misdefined"); 2751 size_t size = mEffectChains.size(); 2752 size_t i = 0; 2753 for (i = 0; i < size; i++) { 2754 if (mEffectChains[i]->sessionId() < session) { 2755 break; 2756 } 2757 } 2758 mEffectChains.insertAt(chain, i); 2759 checkSuspendOnAddEffectChain_l(chain); 2760 2761 return NO_ERROR; 2762} 2763 2764size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 2765{ 2766 audio_session_t session = chain->sessionId(); 2767 2768 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 2769 2770 for (size_t i = 0; i < mEffectChains.size(); i++) { 2771 if (chain == mEffectChains[i]) { 2772 mEffectChains.removeAt(i); 2773 // detach all active tracks from the chain 2774 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 2775 sp<Track> track = mActiveTracks[i].promote(); 2776 if (track == 0) { 2777 continue; 2778 } 2779 if (session == track->sessionId()) { 2780 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 2781 chain.get(), session); 2782 chain->decActiveTrackCnt(); 2783 } 2784 } 2785 2786 // detach all tracks with same session ID from this chain 2787 for (size_t i = 0; i < mTracks.size(); ++i) { 2788 sp<Track> track = mTracks[i]; 2789 if (session == track->sessionId()) { 2790 track->setMainBuffer(reinterpret_cast<int16_t*>(mSinkBuffer)); 2791 chain->decTrackCnt(); 2792 } 2793 } 2794 break; 2795 } 2796 } 2797 return mEffectChains.size(); 2798} 2799 2800status_t AudioFlinger::PlaybackThread::attachAuxEffect( 2801 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2802{ 2803 Mutex::Autolock _l(mLock); 2804 return attachAuxEffect_l(track, EffectId); 2805} 2806 2807status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 2808 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2809{ 2810 status_t status = NO_ERROR; 2811 2812 if (EffectId == 0) { 2813 track->setAuxBuffer(0, NULL); 2814 } else { 2815 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 2816 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 2817 if (effect != 0) { 2818 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 2819 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 2820 } else { 2821 status = INVALID_OPERATION; 2822 } 2823 } else { 2824 status = BAD_VALUE; 2825 } 2826 } 2827 return status; 2828} 2829 2830void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 2831{ 2832 for (size_t i = 0; i < mTracks.size(); ++i) { 2833 sp<Track> track = mTracks[i]; 2834 if (track->auxEffectId() == effectId) { 2835 attachAuxEffect_l(track, 0); 2836 } 2837 } 2838} 2839 2840bool AudioFlinger::PlaybackThread::threadLoop() 2841{ 2842 Vector< sp<Track> > tracksToRemove; 2843 2844 mStandbyTimeNs = systemTime(); 2845 2846 // MIXER 2847 nsecs_t lastWarning = 0; 2848 2849 // DUPLICATING 2850 // FIXME could this be made local to while loop? 2851 writeFrames = 0; 2852 2853 int lastGeneration = 0; 2854 2855 cacheParameters_l(); 2856 mSleepTimeUs = mIdleSleepTimeUs; 2857 2858 if (mType == MIXER) { 2859 sleepTimeShift = 0; 2860 } 2861 2862 CpuStats cpuStats; 2863 const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid())); 2864 2865 acquireWakeLock(); 2866 2867 // mNBLogWriter->log can only be called while thread mutex mLock is held. 2868 // So if you need to log when mutex is unlocked, set logString to a non-NULL string, 2869 // and then that string will be logged at the next convenient opportunity. 2870 const char *logString = NULL; 2871 2872 checkSilentMode_l(); 2873 2874 while (!exitPending()) 2875 { 2876 cpuStats.sample(myName); 2877 2878 Vector< sp<EffectChain> > effectChains; 2879 2880 { // scope for mLock 2881 2882 Mutex::Autolock _l(mLock); 2883 2884 processConfigEvents_l(); 2885 2886 if (logString != NULL) { 2887 mNBLogWriter->logTimestamp(); 2888 mNBLogWriter->log(logString); 2889 logString = NULL; 2890 } 2891 2892 // Gather the framesReleased counters for all active tracks, 2893 // and associate with the sink frames written out. We need 2894 // this to convert the sink timestamp to the track timestamp. 2895 if (mNormalSink != 0) { 2896 // Note: The DuplicatingThread may not have a mNormalSink. 2897 // We always fetch the timestamp here because often the downstream 2898 // sink will block whie writing. 2899 ExtendedTimestamp timestamp; // use private copy to fetch 2900 (void) mNormalSink->getTimestamp(timestamp); 2901 // copy over kernel info 2902 mTimestamp.mPosition[ExtendedTimestamp::LOCATION_KERNEL] = 2903 timestamp.mPosition[ExtendedTimestamp::LOCATION_KERNEL]; 2904 mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL] = 2905 timestamp.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL]; 2906 } 2907 // mFramesWritten for non-offloaded tracks are contiguous 2908 // even after standby() is called. This is useful for the track frame 2909 // to sink frame mapping. 2910 mTimestamp.mPosition[ExtendedTimestamp::LOCATION_SERVER] = mFramesWritten; 2911 mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_SERVER] = systemTime(); 2912 const size_t size = mActiveTracks.size(); 2913 for (size_t i = 0; i < size; ++i) { 2914 sp<Track> t = mActiveTracks[i].promote(); 2915 if (t != 0 && !t->isFastTrack()) { 2916 t->updateTrackFrameInfo( 2917 t->mAudioTrackServerProxy->framesReleased(), 2918 mFramesWritten, 2919 mTimestamp); 2920 } 2921 } 2922 2923 saveOutputTracks(); 2924 if (mSignalPending) { 2925 // A signal was raised while we were unlocked 2926 mSignalPending = false; 2927 } else if (waitingAsyncCallback_l()) { 2928 if (exitPending()) { 2929 break; 2930 } 2931 bool released = false; 2932 if (!keepWakeLock()) { 2933 releaseWakeLock_l(); 2934 released = true; 2935 } 2936 mWakeLockUids.clear(); 2937 mActiveTracksGeneration++; 2938 ALOGV("wait async completion"); 2939 mWaitWorkCV.wait(mLock); 2940 ALOGV("async completion/wake"); 2941 if (released) { 2942 acquireWakeLock_l(); 2943 } 2944 mStandbyTimeNs = systemTime() + mStandbyDelayNs; 2945 mSleepTimeUs = 0; 2946 2947 continue; 2948 } 2949 if ((!mActiveTracks.size() && systemTime() > mStandbyTimeNs) || 2950 isSuspended()) { 2951 // put audio hardware into standby after short delay 2952 if (shouldStandby_l()) { 2953 2954 threadLoop_standby(); 2955 2956 mStandby = true; 2957 } 2958 2959 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2960 // we're about to wait, flush the binder command buffer 2961 IPCThreadState::self()->flushCommands(); 2962 2963 clearOutputTracks(); 2964 2965 if (exitPending()) { 2966 break; 2967 } 2968 2969 releaseWakeLock_l(); 2970 mWakeLockUids.clear(); 2971 mActiveTracksGeneration++; 2972 // wait until we have something to do... 2973 ALOGV("%s going to sleep", myName.string()); 2974 mWaitWorkCV.wait(mLock); 2975 ALOGV("%s waking up", myName.string()); 2976 acquireWakeLock_l(); 2977 2978 mMixerStatus = MIXER_IDLE; 2979 mMixerStatusIgnoringFastTracks = MIXER_IDLE; 2980 mBytesWritten = 0; 2981 mBytesRemaining = 0; 2982 checkSilentMode_l(); 2983 2984 mStandbyTimeNs = systemTime() + mStandbyDelayNs; 2985 mSleepTimeUs = mIdleSleepTimeUs; 2986 if (mType == MIXER) { 2987 sleepTimeShift = 0; 2988 } 2989 2990 continue; 2991 } 2992 } 2993 // mMixerStatusIgnoringFastTracks is also updated internally 2994 mMixerStatus = prepareTracks_l(&tracksToRemove); 2995 2996 // compare with previously applied list 2997 if (lastGeneration != mActiveTracksGeneration) { 2998 // update wakelock 2999 updateWakeLockUids_l(mWakeLockUids); 3000 lastGeneration = mActiveTracksGeneration; 3001 } 3002 3003 // prevent any changes in effect chain list and in each effect chain 3004 // during mixing and effect process as the audio buffers could be deleted 3005 // or modified if an effect is created or deleted 3006 lockEffectChains_l(effectChains); 3007 } // mLock scope ends 3008 3009 if (mBytesRemaining == 0) { 3010 mCurrentWriteLength = 0; 3011 if (mMixerStatus == MIXER_TRACKS_READY) { 3012 // threadLoop_mix() sets mCurrentWriteLength 3013 threadLoop_mix(); 3014 } else if ((mMixerStatus != MIXER_DRAIN_TRACK) 3015 && (mMixerStatus != MIXER_DRAIN_ALL)) { 3016 // threadLoop_sleepTime sets mSleepTimeUs to 0 if data 3017 // must be written to HAL 3018 threadLoop_sleepTime(); 3019 if (mSleepTimeUs == 0) { 3020 mCurrentWriteLength = mSinkBufferSize; 3021 } 3022 } 3023 // Either threadLoop_mix() or threadLoop_sleepTime() should have set 3024 // mMixerBuffer with data if mMixerBufferValid is true and mSleepTimeUs == 0. 3025 // Merge mMixerBuffer data into mEffectBuffer (if any effects are valid) 3026 // or mSinkBuffer (if there are no effects). 3027 // 3028 // This is done pre-effects computation; if effects change to 3029 // support higher precision, this needs to move. 3030 // 3031 // mMixerBufferValid is only set true by MixerThread::prepareTracks_l(). 3032 // TODO use mSleepTimeUs == 0 as an additional condition. 3033 if (mMixerBufferValid) { 3034 void *buffer = mEffectBufferValid ? mEffectBuffer : mSinkBuffer; 3035 audio_format_t format = mEffectBufferValid ? mEffectBufferFormat : mFormat; 3036 3037 // mono blend occurs for mixer threads only (not direct or offloaded) 3038 // and is handled here if we're going directly to the sink. 3039 if (requireMonoBlend() && !mEffectBufferValid) { 3040 mono_blend(mMixerBuffer, mMixerBufferFormat, mChannelCount, mNormalFrameCount, 3041 true /*limit*/); 3042 } 3043 3044 memcpy_by_audio_format(buffer, format, mMixerBuffer, mMixerBufferFormat, 3045 mNormalFrameCount * mChannelCount); 3046 } 3047 3048 mBytesRemaining = mCurrentWriteLength; 3049 if (isSuspended()) { 3050 mSleepTimeUs = suspendSleepTimeUs(); 3051 // simulate write to HAL when suspended 3052 mBytesWritten += mSinkBufferSize; 3053 mFramesWritten += mSinkBufferSize / mFrameSize; 3054 mBytesRemaining = 0; 3055 } 3056 3057 // only process effects if we're going to write 3058 if (mSleepTimeUs == 0 && mType != OFFLOAD) { 3059 for (size_t i = 0; i < effectChains.size(); i ++) { 3060 effectChains[i]->process_l(); 3061 } 3062 } 3063 } 3064 // Process effect chains for offloaded thread even if no audio 3065 // was read from audio track: process only updates effect state 3066 // and thus does have to be synchronized with audio writes but may have 3067 // to be called while waiting for async write callback 3068 if (mType == OFFLOAD) { 3069 for (size_t i = 0; i < effectChains.size(); i ++) { 3070 effectChains[i]->process_l(); 3071 } 3072 } 3073 3074 // Only if the Effects buffer is enabled and there is data in the 3075 // Effects buffer (buffer valid), we need to 3076 // copy into the sink buffer. 3077 // TODO use mSleepTimeUs == 0 as an additional condition. 3078 if (mEffectBufferValid) { 3079 //ALOGV("writing effect buffer to sink buffer format %#x", mFormat); 3080 3081 if (requireMonoBlend()) { 3082 mono_blend(mEffectBuffer, mEffectBufferFormat, mChannelCount, mNormalFrameCount, 3083 true /*limit*/); 3084 } 3085 3086 memcpy_by_audio_format(mSinkBuffer, mFormat, mEffectBuffer, mEffectBufferFormat, 3087 mNormalFrameCount * mChannelCount); 3088 } 3089 3090 // enable changes in effect chain 3091 unlockEffectChains(effectChains); 3092 3093 if (!waitingAsyncCallback()) { 3094 // mSleepTimeUs == 0 means we must write to audio hardware 3095 if (mSleepTimeUs == 0) { 3096 ssize_t ret = 0; 3097 if (mBytesRemaining) { 3098 ret = threadLoop_write(); 3099 if (ret < 0) { 3100 mBytesRemaining = 0; 3101 } else { 3102 mBytesWritten += ret; 3103 mBytesRemaining -= ret; 3104 mFramesWritten += ret / mFrameSize; 3105 } 3106 } else if ((mMixerStatus == MIXER_DRAIN_TRACK) || 3107 (mMixerStatus == MIXER_DRAIN_ALL)) { 3108 threadLoop_drain(); 3109 } 3110 if (mType == MIXER && !mStandby) { 3111 // write blocked detection 3112 nsecs_t now = systemTime(); 3113 nsecs_t delta = now - mLastWriteTime; 3114 if (delta > maxPeriod) { 3115 mNumDelayedWrites++; 3116 if ((now - lastWarning) > kWarningThrottleNs) { 3117 ATRACE_NAME("underrun"); 3118 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 3119 (unsigned long long) ns2ms(delta), mNumDelayedWrites, this); 3120 lastWarning = now; 3121 } 3122 } 3123 3124 if (mThreadThrottle 3125 && mMixerStatus == MIXER_TRACKS_READY // we are mixing (active tracks) 3126 && ret > 0) { // we wrote something 3127 // Limit MixerThread data processing to no more than twice the 3128 // expected processing rate. 3129 // 3130 // This helps prevent underruns with NuPlayer and other applications 3131 // which may set up buffers that are close to the minimum size, or use 3132 // deep buffers, and rely on a double-buffering sleep strategy to fill. 3133 // 3134 // The throttle smooths out sudden large data drains from the device, 3135 // e.g. when it comes out of standby, which often causes problems with 3136 // (1) mixer threads without a fast mixer (which has its own warm-up) 3137 // (2) minimum buffer sized tracks (even if the track is full, 3138 // the app won't fill fast enough to handle the sudden draw). 3139 3140 const int32_t deltaMs = delta / 1000000; 3141 const int32_t throttleMs = mHalfBufferMs - deltaMs; 3142 if ((signed)mHalfBufferMs >= throttleMs && throttleMs > 0) { 3143 usleep(throttleMs * 1000); 3144 // notify of throttle start on verbose log 3145 ALOGV_IF(mThreadThrottleEndMs == mThreadThrottleTimeMs, 3146 "mixer(%p) throttle begin:" 3147 " ret(%zd) deltaMs(%d) requires sleep %d ms", 3148 this, ret, deltaMs, throttleMs); 3149 mThreadThrottleTimeMs += throttleMs; 3150 } else { 3151 uint32_t diff = mThreadThrottleTimeMs - mThreadThrottleEndMs; 3152 if (diff > 0) { 3153 // notify of throttle end on debug log 3154 ALOGD("mixer(%p) throttle end: throttle time(%u)", this, diff); 3155 mThreadThrottleEndMs = mThreadThrottleTimeMs; 3156 } 3157 } 3158 } 3159 } 3160 3161 } else { 3162 ATRACE_BEGIN("sleep"); 3163 if ((mType == OFFLOAD) && !audio_has_proportional_frames(mFormat)) { 3164 Mutex::Autolock _l(mLock); 3165 if (!mSignalPending && !exitPending()) { 3166 // If more than one buffer has been written to the audio HAL since exiting 3167 // standby or last flush, do not sleep more than one buffer duration 3168 // since last write and not less than kDirectMinSleepTimeUs. 3169 // Wake up if a command is received 3170 uint32_t timeoutUs = mSleepTimeUs; 3171 if (mBytesWritten >= (int64_t) mBufferSize) { 3172 nsecs_t now = systemTime(); 3173 uint32_t deltaUs = (uint32_t)((now - mLastWriteTime) / 1000); 3174 if (timeoutUs + deltaUs > mBufferDurationUs) { 3175 if (mBufferDurationUs > deltaUs) { 3176 timeoutUs = mBufferDurationUs - deltaUs; 3177 if (timeoutUs < kDirectMinSleepTimeUs) { 3178 timeoutUs = kDirectMinSleepTimeUs; 3179 } 3180 } else { 3181 timeoutUs = kDirectMinSleepTimeUs; 3182 } 3183 } 3184 } 3185 mWaitWorkCV.waitRelative(mLock, microseconds((nsecs_t)timeoutUs)); 3186 } 3187 } else { 3188 usleep(mSleepTimeUs); 3189 } 3190 ATRACE_END(); 3191 } 3192 } 3193 3194 // Finally let go of removed track(s), without the lock held 3195 // since we can't guarantee the destructors won't acquire that 3196 // same lock. This will also mutate and push a new fast mixer state. 3197 threadLoop_removeTracks(tracksToRemove); 3198 tracksToRemove.clear(); 3199 3200 // FIXME I don't understand the need for this here; 3201 // it was in the original code but maybe the 3202 // assignment in saveOutputTracks() makes this unnecessary? 3203 clearOutputTracks(); 3204 3205 // Effect chains will be actually deleted here if they were removed from 3206 // mEffectChains list during mixing or effects processing 3207 effectChains.clear(); 3208 3209 // FIXME Note that the above .clear() is no longer necessary since effectChains 3210 // is now local to this block, but will keep it for now (at least until merge done). 3211 } 3212 3213 threadLoop_exit(); 3214 3215 if (!mStandby) { 3216 threadLoop_standby(); 3217 mStandby = true; 3218 } 3219 3220 releaseWakeLock(); 3221 mWakeLockUids.clear(); 3222 mActiveTracksGeneration++; 3223 3224 ALOGV("Thread %p type %d exiting", this, mType); 3225 return false; 3226} 3227 3228// removeTracks_l() must be called with ThreadBase::mLock held 3229void AudioFlinger::PlaybackThread::removeTracks_l(const Vector< sp<Track> >& tracksToRemove) 3230{ 3231 size_t count = tracksToRemove.size(); 3232 if (count > 0) { 3233 for (size_t i=0 ; i<count ; i++) { 3234 const sp<Track>& track = tracksToRemove.itemAt(i); 3235 mActiveTracks.remove(track); 3236 mWakeLockUids.remove(track->uid()); 3237 mActiveTracksGeneration++; 3238 ALOGV("removeTracks_l removing track on session %d", track->sessionId()); 3239 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 3240 if (chain != 0) { 3241 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), 3242 track->sessionId()); 3243 chain->decActiveTrackCnt(); 3244 } 3245 if (track->isTerminated()) { 3246 removeTrack_l(track); 3247 } 3248 } 3249 } 3250 3251} 3252 3253status_t AudioFlinger::PlaybackThread::getTimestamp_l(AudioTimestamp& timestamp) 3254{ 3255 if (mNormalSink != 0) { 3256 ExtendedTimestamp ets; 3257 status_t status = mNormalSink->getTimestamp(ets); 3258 if (status == NO_ERROR) { 3259 status = ets.getBestTimestamp(×tamp); 3260 } 3261 return status; 3262 } 3263 if ((mType == OFFLOAD || mType == DIRECT) 3264 && mOutput != NULL && mOutput->stream->get_presentation_position) { 3265 uint64_t position64; 3266 int ret = mOutput->getPresentationPosition(&position64, ×tamp.mTime); 3267 if (ret == 0) { 3268 timestamp.mPosition = (uint32_t)position64; 3269 return NO_ERROR; 3270 } 3271 } 3272 return INVALID_OPERATION; 3273} 3274 3275status_t AudioFlinger::MixerThread::createAudioPatch_l(const struct audio_patch *patch, 3276 audio_patch_handle_t *handle) 3277{ 3278 AutoPark<FastMixer> park(mFastMixer); 3279 3280 status_t status = PlaybackThread::createAudioPatch_l(patch, handle); 3281 3282 return status; 3283} 3284 3285status_t AudioFlinger::PlaybackThread::createAudioPatch_l(const struct audio_patch *patch, 3286 audio_patch_handle_t *handle) 3287{ 3288 status_t status = NO_ERROR; 3289 3290 // store new device and send to effects 3291 audio_devices_t type = AUDIO_DEVICE_NONE; 3292 for (unsigned int i = 0; i < patch->num_sinks; i++) { 3293 type |= patch->sinks[i].ext.device.type; 3294 } 3295 3296#ifdef ADD_BATTERY_DATA 3297 // when changing the audio output device, call addBatteryData to notify 3298 // the change 3299 if (mOutDevice != type) { 3300 uint32_t params = 0; 3301 // check whether speaker is on 3302 if (type & AUDIO_DEVICE_OUT_SPEAKER) { 3303 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 3304 } 3305 3306 audio_devices_t deviceWithoutSpeaker 3307 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 3308 // check if any other device (except speaker) is on 3309 if (type & deviceWithoutSpeaker) { 3310 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 3311 } 3312 3313 if (params != 0) { 3314 addBatteryData(params); 3315 } 3316 } 3317#endif 3318 3319 for (size_t i = 0; i < mEffectChains.size(); i++) { 3320 mEffectChains[i]->setDevice_l(type); 3321 } 3322 3323 // mPrevOutDevice is the latest device set by createAudioPatch_l(). It is not set when 3324 // the thread is created so that the first patch creation triggers an ioConfigChanged callback 3325 bool configChanged = mPrevOutDevice != type; 3326 mOutDevice = type; 3327 mPatch = *patch; 3328 3329 if (mOutput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 3330 audio_hw_device_t *hwDevice = mOutput->audioHwDev->hwDevice(); 3331 status = hwDevice->create_audio_patch(hwDevice, 3332 patch->num_sources, 3333 patch->sources, 3334 patch->num_sinks, 3335 patch->sinks, 3336 handle); 3337 } else { 3338 char *address; 3339 if (strcmp(patch->sinks[0].ext.device.address, "") != 0) { 3340 //FIXME: we only support address on first sink with HAL version < 3.0 3341 address = audio_device_address_to_parameter( 3342 patch->sinks[0].ext.device.type, 3343 patch->sinks[0].ext.device.address); 3344 } else { 3345 address = (char *)calloc(1, 1); 3346 } 3347 AudioParameter param = AudioParameter(String8(address)); 3348 free(address); 3349 param.addInt(String8(AUDIO_PARAMETER_STREAM_ROUTING), (int)type); 3350 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3351 param.toString().string()); 3352 *handle = AUDIO_PATCH_HANDLE_NONE; 3353 } 3354 if (configChanged) { 3355 mPrevOutDevice = type; 3356 sendIoConfigEvent_l(AUDIO_OUTPUT_CONFIG_CHANGED); 3357 } 3358 return status; 3359} 3360 3361status_t AudioFlinger::MixerThread::releaseAudioPatch_l(const audio_patch_handle_t handle) 3362{ 3363 AutoPark<FastMixer> park(mFastMixer); 3364 3365 status_t status = PlaybackThread::releaseAudioPatch_l(handle); 3366 3367 return status; 3368} 3369 3370status_t AudioFlinger::PlaybackThread::releaseAudioPatch_l(const audio_patch_handle_t handle) 3371{ 3372 status_t status = NO_ERROR; 3373 3374 mOutDevice = AUDIO_DEVICE_NONE; 3375 3376 if (mOutput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 3377 audio_hw_device_t *hwDevice = mOutput->audioHwDev->hwDevice(); 3378 status = hwDevice->release_audio_patch(hwDevice, handle); 3379 } else { 3380 AudioParameter param; 3381 param.addInt(String8(AUDIO_PARAMETER_STREAM_ROUTING), 0); 3382 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3383 param.toString().string()); 3384 } 3385 return status; 3386} 3387 3388void AudioFlinger::PlaybackThread::addPatchTrack(const sp<PatchTrack>& track) 3389{ 3390 Mutex::Autolock _l(mLock); 3391 mTracks.add(track); 3392} 3393 3394void AudioFlinger::PlaybackThread::deletePatchTrack(const sp<PatchTrack>& track) 3395{ 3396 Mutex::Autolock _l(mLock); 3397 destroyTrack_l(track); 3398} 3399 3400void AudioFlinger::PlaybackThread::getAudioPortConfig(struct audio_port_config *config) 3401{ 3402 ThreadBase::getAudioPortConfig(config); 3403 config->role = AUDIO_PORT_ROLE_SOURCE; 3404 config->ext.mix.hw_module = mOutput->audioHwDev->handle(); 3405 config->ext.mix.usecase.stream = AUDIO_STREAM_DEFAULT; 3406} 3407 3408// ---------------------------------------------------------------------------- 3409 3410AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 3411 audio_io_handle_t id, audio_devices_t device, bool systemReady, type_t type) 3412 : PlaybackThread(audioFlinger, output, id, device, type, systemReady), 3413 // mAudioMixer below 3414 // mFastMixer below 3415 mFastMixerFutex(0), 3416 mMasterMono(false) 3417 // mOutputSink below 3418 // mPipeSink below 3419 // mNormalSink below 3420{ 3421 ALOGV("MixerThread() id=%d device=%#x type=%d", id, device, type); 3422 ALOGV("mSampleRate=%u, mChannelMask=%#x, mChannelCount=%u, mFormat=%d, mFrameSize=%zu, " 3423 "mFrameCount=%zu, mNormalFrameCount=%zu", 3424 mSampleRate, mChannelMask, mChannelCount, mFormat, mFrameSize, mFrameCount, 3425 mNormalFrameCount); 3426 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 3427 3428 if (type == DUPLICATING) { 3429 // The Duplicating thread uses the AudioMixer and delivers data to OutputTracks 3430 // (downstream MixerThreads) in DuplicatingThread::threadLoop_write(). 3431 // Do not create or use mFastMixer, mOutputSink, mPipeSink, or mNormalSink. 3432 return; 3433 } 3434 // create an NBAIO sink for the HAL output stream, and negotiate 3435 mOutputSink = new AudioStreamOutSink(output->stream); 3436 size_t numCounterOffers = 0; 3437 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount, mFormat)}; 3438#if !LOG_NDEBUG 3439 ssize_t index = 3440#else 3441 (void) 3442#endif 3443 mOutputSink->negotiate(offers, 1, NULL, numCounterOffers); 3444 ALOG_ASSERT(index == 0); 3445 3446 // initialize fast mixer depending on configuration 3447 bool initFastMixer; 3448 switch (kUseFastMixer) { 3449 case FastMixer_Never: 3450 initFastMixer = false; 3451 break; 3452 case FastMixer_Always: 3453 initFastMixer = true; 3454 break; 3455 case FastMixer_Static: 3456 case FastMixer_Dynamic: 3457 initFastMixer = mFrameCount < mNormalFrameCount; 3458 break; 3459 } 3460 if (initFastMixer) { 3461 audio_format_t fastMixerFormat; 3462 if (mMixerBufferEnabled && mEffectBufferEnabled) { 3463 fastMixerFormat = AUDIO_FORMAT_PCM_FLOAT; 3464 } else { 3465 fastMixerFormat = AUDIO_FORMAT_PCM_16_BIT; 3466 } 3467 if (mFormat != fastMixerFormat) { 3468 // change our Sink format to accept our intermediate precision 3469 mFormat = fastMixerFormat; 3470 free(mSinkBuffer); 3471 mFrameSize = mChannelCount * audio_bytes_per_sample(mFormat); 3472 const size_t sinkBufferSize = mNormalFrameCount * mFrameSize; 3473 (void)posix_memalign(&mSinkBuffer, 32, sinkBufferSize); 3474 } 3475 3476 // create a MonoPipe to connect our submix to FastMixer 3477 NBAIO_Format format = mOutputSink->format(); 3478#ifdef TEE_SINK 3479 NBAIO_Format origformat = format; 3480#endif 3481 // adjust format to match that of the Fast Mixer 3482 ALOGV("format changed from %d to %d", format.mFormat, fastMixerFormat); 3483 format.mFormat = fastMixerFormat; 3484 format.mFrameSize = audio_bytes_per_sample(format.mFormat) * format.mChannelCount; 3485 3486 // This pipe depth compensates for scheduling latency of the normal mixer thread. 3487 // When it wakes up after a maximum latency, it runs a few cycles quickly before 3488 // finally blocking. Note the pipe implementation rounds up the request to a power of 2. 3489 MonoPipe *monoPipe = new MonoPipe(mNormalFrameCount * 4, format, true /*writeCanBlock*/); 3490 const NBAIO_Format offers[1] = {format}; 3491 size_t numCounterOffers = 0; 3492#if !LOG_NDEBUG || defined(TEE_SINK) 3493 ssize_t index = 3494#else 3495 (void) 3496#endif 3497 monoPipe->negotiate(offers, 1, NULL, numCounterOffers); 3498 ALOG_ASSERT(index == 0); 3499 monoPipe->setAvgFrames((mScreenState & 1) ? 3500 (monoPipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 3501 mPipeSink = monoPipe; 3502 3503#ifdef TEE_SINK 3504 if (mTeeSinkOutputEnabled) { 3505 // create a Pipe to archive a copy of FastMixer's output for dumpsys 3506 Pipe *teeSink = new Pipe(mTeeSinkOutputFrames, origformat); 3507 const NBAIO_Format offers2[1] = {origformat}; 3508 numCounterOffers = 0; 3509 index = teeSink->negotiate(offers2, 1, NULL, numCounterOffers); 3510 ALOG_ASSERT(index == 0); 3511 mTeeSink = teeSink; 3512 PipeReader *teeSource = new PipeReader(*teeSink); 3513 numCounterOffers = 0; 3514 index = teeSource->negotiate(offers2, 1, NULL, numCounterOffers); 3515 ALOG_ASSERT(index == 0); 3516 mTeeSource = teeSource; 3517 } 3518#endif 3519 3520 // create fast mixer and configure it initially with just one fast track for our submix 3521 mFastMixer = new FastMixer(); 3522 FastMixerStateQueue *sq = mFastMixer->sq(); 3523#ifdef STATE_QUEUE_DUMP 3524 sq->setObserverDump(&mStateQueueObserverDump); 3525 sq->setMutatorDump(&mStateQueueMutatorDump); 3526#endif 3527 FastMixerState *state = sq->begin(); 3528 FastTrack *fastTrack = &state->mFastTracks[0]; 3529 // wrap the source side of the MonoPipe to make it an AudioBufferProvider 3530 fastTrack->mBufferProvider = new SourceAudioBufferProvider(new MonoPipeReader(monoPipe)); 3531 fastTrack->mVolumeProvider = NULL; 3532 fastTrack->mChannelMask = mChannelMask; // mPipeSink channel mask for audio to FastMixer 3533 fastTrack->mFormat = mFormat; // mPipeSink format for audio to FastMixer 3534 fastTrack->mGeneration++; 3535 state->mFastTracksGen++; 3536 state->mTrackMask = 1; 3537 // fast mixer will use the HAL output sink 3538 state->mOutputSink = mOutputSink.get(); 3539 state->mOutputSinkGen++; 3540 state->mFrameCount = mFrameCount; 3541 state->mCommand = FastMixerState::COLD_IDLE; 3542 // already done in constructor initialization list 3543 //mFastMixerFutex = 0; 3544 state->mColdFutexAddr = &mFastMixerFutex; 3545 state->mColdGen++; 3546 state->mDumpState = &mFastMixerDumpState; 3547#ifdef TEE_SINK 3548 state->mTeeSink = mTeeSink.get(); 3549#endif 3550 mFastMixerNBLogWriter = audioFlinger->newWriter_l(kFastMixerLogSize, "FastMixer"); 3551 state->mNBLogWriter = mFastMixerNBLogWriter.get(); 3552 sq->end(); 3553 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3554 3555 // start the fast mixer 3556 mFastMixer->run("FastMixer", PRIORITY_URGENT_AUDIO); 3557 pid_t tid = mFastMixer->getTid(); 3558 sendPrioConfigEvent(getpid_cached, tid, kPriorityFastMixer); 3559 3560#ifdef AUDIO_WATCHDOG 3561 // create and start the watchdog 3562 mAudioWatchdog = new AudioWatchdog(); 3563 mAudioWatchdog->setDump(&mAudioWatchdogDump); 3564 mAudioWatchdog->run("AudioWatchdog", PRIORITY_URGENT_AUDIO); 3565 tid = mAudioWatchdog->getTid(); 3566 sendPrioConfigEvent(getpid_cached, tid, kPriorityFastMixer); 3567#endif 3568 3569 } 3570 3571 switch (kUseFastMixer) { 3572 case FastMixer_Never: 3573 case FastMixer_Dynamic: 3574 mNormalSink = mOutputSink; 3575 break; 3576 case FastMixer_Always: 3577 mNormalSink = mPipeSink; 3578 break; 3579 case FastMixer_Static: 3580 mNormalSink = initFastMixer ? mPipeSink : mOutputSink; 3581 break; 3582 } 3583} 3584 3585AudioFlinger::MixerThread::~MixerThread() 3586{ 3587 if (mFastMixer != 0) { 3588 FastMixerStateQueue *sq = mFastMixer->sq(); 3589 FastMixerState *state = sq->begin(); 3590 if (state->mCommand == FastMixerState::COLD_IDLE) { 3591 int32_t old = android_atomic_inc(&mFastMixerFutex); 3592 if (old == -1) { 3593 (void) syscall(__NR_futex, &mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 3594 } 3595 } 3596 state->mCommand = FastMixerState::EXIT; 3597 sq->end(); 3598 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3599 mFastMixer->join(); 3600 // Though the fast mixer thread has exited, it's state queue is still valid. 3601 // We'll use that extract the final state which contains one remaining fast track 3602 // corresponding to our sub-mix. 3603 state = sq->begin(); 3604 ALOG_ASSERT(state->mTrackMask == 1); 3605 FastTrack *fastTrack = &state->mFastTracks[0]; 3606 ALOG_ASSERT(fastTrack->mBufferProvider != NULL); 3607 delete fastTrack->mBufferProvider; 3608 sq->end(false /*didModify*/); 3609 mFastMixer.clear(); 3610#ifdef AUDIO_WATCHDOG 3611 if (mAudioWatchdog != 0) { 3612 mAudioWatchdog->requestExit(); 3613 mAudioWatchdog->requestExitAndWait(); 3614 mAudioWatchdog.clear(); 3615 } 3616#endif 3617 } 3618 mAudioFlinger->unregisterWriter(mFastMixerNBLogWriter); 3619 delete mAudioMixer; 3620} 3621 3622 3623uint32_t AudioFlinger::MixerThread::correctLatency_l(uint32_t latency) const 3624{ 3625 if (mFastMixer != 0) { 3626 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 3627 latency += (pipe->getAvgFrames() * 1000) / mSampleRate; 3628 } 3629 return latency; 3630} 3631 3632 3633void AudioFlinger::MixerThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) 3634{ 3635 PlaybackThread::threadLoop_removeTracks(tracksToRemove); 3636} 3637 3638ssize_t AudioFlinger::MixerThread::threadLoop_write() 3639{ 3640 // FIXME we should only do one push per cycle; confirm this is true 3641 // Start the fast mixer if it's not already running 3642 if (mFastMixer != 0) { 3643 FastMixerStateQueue *sq = mFastMixer->sq(); 3644 FastMixerState *state = sq->begin(); 3645 if (state->mCommand != FastMixerState::MIX_WRITE && 3646 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)) { 3647 if (state->mCommand == FastMixerState::COLD_IDLE) { 3648 3649 // FIXME workaround for first HAL write being CPU bound on some devices 3650 ATRACE_BEGIN("write"); 3651 mOutput->write((char *)mSinkBuffer, 0); 3652 ATRACE_END(); 3653 3654 int32_t old = android_atomic_inc(&mFastMixerFutex); 3655 if (old == -1) { 3656 (void) syscall(__NR_futex, &mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 3657 } 3658#ifdef AUDIO_WATCHDOG 3659 if (mAudioWatchdog != 0) { 3660 mAudioWatchdog->resume(); 3661 } 3662#endif 3663 } 3664 state->mCommand = FastMixerState::MIX_WRITE; 3665#ifdef FAST_THREAD_STATISTICS 3666 mFastMixerDumpState.increaseSamplingN(mAudioFlinger->isLowRamDevice() ? 3667 FastThreadDumpState::kSamplingNforLowRamDevice : FastThreadDumpState::kSamplingN); 3668#endif 3669 sq->end(); 3670 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3671 if (kUseFastMixer == FastMixer_Dynamic) { 3672 mNormalSink = mPipeSink; 3673 } 3674 } else { 3675 sq->end(false /*didModify*/); 3676 } 3677 } 3678 return PlaybackThread::threadLoop_write(); 3679} 3680 3681void AudioFlinger::MixerThread::threadLoop_standby() 3682{ 3683 // Idle the fast mixer if it's currently running 3684 if (mFastMixer != 0) { 3685 FastMixerStateQueue *sq = mFastMixer->sq(); 3686 FastMixerState *state = sq->begin(); 3687 if (!(state->mCommand & FastMixerState::IDLE)) { 3688 state->mCommand = FastMixerState::COLD_IDLE; 3689 state->mColdFutexAddr = &mFastMixerFutex; 3690 state->mColdGen++; 3691 mFastMixerFutex = 0; 3692 sq->end(); 3693 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 3694 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3695 if (kUseFastMixer == FastMixer_Dynamic) { 3696 mNormalSink = mOutputSink; 3697 } 3698#ifdef AUDIO_WATCHDOG 3699 if (mAudioWatchdog != 0) { 3700 mAudioWatchdog->pause(); 3701 } 3702#endif 3703 } else { 3704 sq->end(false /*didModify*/); 3705 } 3706 } 3707 PlaybackThread::threadLoop_standby(); 3708} 3709 3710bool AudioFlinger::PlaybackThread::waitingAsyncCallback_l() 3711{ 3712 return false; 3713} 3714 3715bool AudioFlinger::PlaybackThread::shouldStandby_l() 3716{ 3717 return !mStandby; 3718} 3719 3720bool AudioFlinger::PlaybackThread::waitingAsyncCallback() 3721{ 3722 Mutex::Autolock _l(mLock); 3723 return waitingAsyncCallback_l(); 3724} 3725 3726// shared by MIXER and DIRECT, overridden by DUPLICATING 3727void AudioFlinger::PlaybackThread::threadLoop_standby() 3728{ 3729 ALOGV("Audio hardware entering standby, mixer %p, suspend count %d", this, mSuspended); 3730 mOutput->standby(); 3731 if (mUseAsyncWrite != 0) { 3732 // discard any pending drain or write ack by incrementing sequence 3733 mWriteAckSequence = (mWriteAckSequence + 2) & ~1; 3734 mDrainSequence = (mDrainSequence + 2) & ~1; 3735 ALOG_ASSERT(mCallbackThread != 0); 3736 mCallbackThread->setWriteBlocked(mWriteAckSequence); 3737 mCallbackThread->setDraining(mDrainSequence); 3738 } 3739 mHwPaused = false; 3740} 3741 3742void AudioFlinger::PlaybackThread::onAddNewTrack_l() 3743{ 3744 ALOGV("signal playback thread"); 3745 broadcast_l(); 3746} 3747 3748void AudioFlinger::MixerThread::threadLoop_mix() 3749{ 3750 // mix buffers... 3751 mAudioMixer->process(); 3752 mCurrentWriteLength = mSinkBufferSize; 3753 // increase sleep time progressively when application underrun condition clears. 3754 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 3755 // that a steady state of alternating ready/not ready conditions keeps the sleep time 3756 // such that we would underrun the audio HAL. 3757 if ((mSleepTimeUs == 0) && (sleepTimeShift > 0)) { 3758 sleepTimeShift--; 3759 } 3760 mSleepTimeUs = 0; 3761 mStandbyTimeNs = systemTime() + mStandbyDelayNs; 3762 //TODO: delay standby when effects have a tail 3763 3764} 3765 3766void AudioFlinger::MixerThread::threadLoop_sleepTime() 3767{ 3768 // If no tracks are ready, sleep once for the duration of an output 3769 // buffer size, then write 0s to the output 3770 if (mSleepTimeUs == 0) { 3771 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3772 mSleepTimeUs = mActiveSleepTimeUs >> sleepTimeShift; 3773 if (mSleepTimeUs < kMinThreadSleepTimeUs) { 3774 mSleepTimeUs = kMinThreadSleepTimeUs; 3775 } 3776 // reduce sleep time in case of consecutive application underruns to avoid 3777 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 3778 // duration we would end up writing less data than needed by the audio HAL if 3779 // the condition persists. 3780 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 3781 sleepTimeShift++; 3782 } 3783 } else { 3784 mSleepTimeUs = mIdleSleepTimeUs; 3785 } 3786 } else if (mBytesWritten != 0 || (mMixerStatus == MIXER_TRACKS_ENABLED)) { 3787 // clear out mMixerBuffer or mSinkBuffer, to ensure buffers are cleared 3788 // before effects processing or output. 3789 if (mMixerBufferValid) { 3790 memset(mMixerBuffer, 0, mMixerBufferSize); 3791 } else { 3792 memset(mSinkBuffer, 0, mSinkBufferSize); 3793 } 3794 mSleepTimeUs = 0; 3795 ALOGV_IF(mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED), 3796 "anticipated start"); 3797 } 3798 // TODO add standby time extension fct of effect tail 3799} 3800 3801// prepareTracks_l() must be called with ThreadBase::mLock held 3802AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 3803 Vector< sp<Track> > *tracksToRemove) 3804{ 3805 3806 mixer_state mixerStatus = MIXER_IDLE; 3807 // find out which tracks need to be processed 3808 size_t count = mActiveTracks.size(); 3809 size_t mixedTracks = 0; 3810 size_t tracksWithEffect = 0; 3811 // counts only _active_ fast tracks 3812 size_t fastTracks = 0; 3813 uint32_t resetMask = 0; // bit mask of fast tracks that need to be reset 3814 3815 float masterVolume = mMasterVolume; 3816 bool masterMute = mMasterMute; 3817 3818 if (masterMute) { 3819 masterVolume = 0; 3820 } 3821 // Delegate master volume control to effect in output mix effect chain if needed 3822 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 3823 if (chain != 0) { 3824 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 3825 chain->setVolume_l(&v, &v); 3826 masterVolume = (float)((v + (1 << 23)) >> 24); 3827 chain.clear(); 3828 } 3829 3830 // prepare a new state to push 3831 FastMixerStateQueue *sq = NULL; 3832 FastMixerState *state = NULL; 3833 bool didModify = false; 3834 FastMixerStateQueue::block_t block = FastMixerStateQueue::BLOCK_UNTIL_PUSHED; 3835 if (mFastMixer != 0) { 3836 sq = mFastMixer->sq(); 3837 state = sq->begin(); 3838 } 3839 3840 mMixerBufferValid = false; // mMixerBuffer has no valid data until appropriate tracks found. 3841 mEffectBufferValid = false; // mEffectBuffer has no valid data until tracks found. 3842 3843 for (size_t i=0 ; i<count ; i++) { 3844 const sp<Track> t = mActiveTracks[i].promote(); 3845 if (t == 0) { 3846 continue; 3847 } 3848 3849 // this const just means the local variable doesn't change 3850 Track* const track = t.get(); 3851 3852 // process fast tracks 3853 if (track->isFastTrack()) { 3854 3855 // It's theoretically possible (though unlikely) for a fast track to be created 3856 // and then removed within the same normal mix cycle. This is not a problem, as 3857 // the track never becomes active so it's fast mixer slot is never touched. 3858 // The converse, of removing an (active) track and then creating a new track 3859 // at the identical fast mixer slot within the same normal mix cycle, 3860 // is impossible because the slot isn't marked available until the end of each cycle. 3861 int j = track->mFastIndex; 3862 ALOG_ASSERT(0 < j && j < (int)FastMixerState::kMaxFastTracks); 3863 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << j))); 3864 FastTrack *fastTrack = &state->mFastTracks[j]; 3865 3866 // Determine whether the track is currently in underrun condition, 3867 // and whether it had a recent underrun. 3868 FastTrackDump *ftDump = &mFastMixerDumpState.mTracks[j]; 3869 FastTrackUnderruns underruns = ftDump->mUnderruns; 3870 uint32_t recentFull = (underruns.mBitFields.mFull - 3871 track->mObservedUnderruns.mBitFields.mFull) & UNDERRUN_MASK; 3872 uint32_t recentPartial = (underruns.mBitFields.mPartial - 3873 track->mObservedUnderruns.mBitFields.mPartial) & UNDERRUN_MASK; 3874 uint32_t recentEmpty = (underruns.mBitFields.mEmpty - 3875 track->mObservedUnderruns.mBitFields.mEmpty) & UNDERRUN_MASK; 3876 uint32_t recentUnderruns = recentPartial + recentEmpty; 3877 track->mObservedUnderruns = underruns; 3878 // don't count underruns that occur while stopping or pausing 3879 // or stopped which can occur when flush() is called while active 3880 if (!(track->isStopping() || track->isPausing() || track->isStopped()) && 3881 recentUnderruns > 0) { 3882 // FIXME fast mixer will pull & mix partial buffers, but we count as a full underrun 3883 track->mAudioTrackServerProxy->tallyUnderrunFrames(recentUnderruns * mFrameCount); 3884 } else { 3885 track->mAudioTrackServerProxy->tallyUnderrunFrames(0); 3886 } 3887 3888 // This is similar to the state machine for normal tracks, 3889 // with a few modifications for fast tracks. 3890 bool isActive = true; 3891 switch (track->mState) { 3892 case TrackBase::STOPPING_1: 3893 // track stays active in STOPPING_1 state until first underrun 3894 if (recentUnderruns > 0 || track->isTerminated()) { 3895 track->mState = TrackBase::STOPPING_2; 3896 } 3897 break; 3898 case TrackBase::PAUSING: 3899 // ramp down is not yet implemented 3900 track->setPaused(); 3901 break; 3902 case TrackBase::RESUMING: 3903 // ramp up is not yet implemented 3904 track->mState = TrackBase::ACTIVE; 3905 break; 3906 case TrackBase::ACTIVE: 3907 if (recentFull > 0 || recentPartial > 0) { 3908 // track has provided at least some frames recently: reset retry count 3909 track->mRetryCount = kMaxTrackRetries; 3910 } 3911 if (recentUnderruns == 0) { 3912 // no recent underruns: stay active 3913 break; 3914 } 3915 // there has recently been an underrun of some kind 3916 if (track->sharedBuffer() == 0) { 3917 // were any of the recent underruns "empty" (no frames available)? 3918 if (recentEmpty == 0) { 3919 // no, then ignore the partial underruns as they are allowed indefinitely 3920 break; 3921 } 3922 // there has recently been an "empty" underrun: decrement the retry counter 3923 if (--(track->mRetryCount) > 0) { 3924 break; 3925 } 3926 // indicate to client process that the track was disabled because of underrun; 3927 // it will then automatically call start() when data is available 3928 track->disable(); 3929 // remove from active list, but state remains ACTIVE [confusing but true] 3930 isActive = false; 3931 break; 3932 } 3933 // fall through 3934 case TrackBase::STOPPING_2: 3935 case TrackBase::PAUSED: 3936 case TrackBase::STOPPED: 3937 case TrackBase::FLUSHED: // flush() while active 3938 // Check for presentation complete if track is inactive 3939 // We have consumed all the buffers of this track. 3940 // This would be incomplete if we auto-paused on underrun 3941 { 3942 size_t audioHALFrames = 3943 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 3944 int64_t framesWritten = mBytesWritten / mFrameSize; 3945 if (!(mStandby || track->presentationComplete(framesWritten, audioHALFrames))) { 3946 // track stays in active list until presentation is complete 3947 break; 3948 } 3949 } 3950 if (track->isStopping_2()) { 3951 track->mState = TrackBase::STOPPED; 3952 } 3953 if (track->isStopped()) { 3954 // Can't reset directly, as fast mixer is still polling this track 3955 // track->reset(); 3956 // So instead mark this track as needing to be reset after push with ack 3957 resetMask |= 1 << i; 3958 } 3959 isActive = false; 3960 break; 3961 case TrackBase::IDLE: 3962 default: 3963 LOG_ALWAYS_FATAL("unexpected track state %d", track->mState); 3964 } 3965 3966 if (isActive) { 3967 // was it previously inactive? 3968 if (!(state->mTrackMask & (1 << j))) { 3969 ExtendedAudioBufferProvider *eabp = track; 3970 VolumeProvider *vp = track; 3971 fastTrack->mBufferProvider = eabp; 3972 fastTrack->mVolumeProvider = vp; 3973 fastTrack->mChannelMask = track->mChannelMask; 3974 fastTrack->mFormat = track->mFormat; 3975 fastTrack->mGeneration++; 3976 state->mTrackMask |= 1 << j; 3977 didModify = true; 3978 // no acknowledgement required for newly active tracks 3979 } 3980 // cache the combined master volume and stream type volume for fast mixer; this 3981 // lacks any synchronization or barrier so VolumeProvider may read a stale value 3982 track->mCachedVolume = masterVolume * mStreamTypes[track->streamType()].volume; 3983 ++fastTracks; 3984 } else { 3985 // was it previously active? 3986 if (state->mTrackMask & (1 << j)) { 3987 fastTrack->mBufferProvider = NULL; 3988 fastTrack->mGeneration++; 3989 state->mTrackMask &= ~(1 << j); 3990 didModify = true; 3991 // If any fast tracks were removed, we must wait for acknowledgement 3992 // because we're about to decrement the last sp<> on those tracks. 3993 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3994 } else { 3995 LOG_ALWAYS_FATAL("fast track %d should have been active; " 3996 "mState=%d, mTrackMask=%#x, recentUnderruns=%u, isShared=%d", 3997 j, track->mState, state->mTrackMask, recentUnderruns, 3998 track->sharedBuffer() != 0); 3999 } 4000 tracksToRemove->add(track); 4001 // Avoids a misleading display in dumpsys 4002 track->mObservedUnderruns.mBitFields.mMostRecent = UNDERRUN_FULL; 4003 } 4004 continue; 4005 } 4006 4007 { // local variable scope to avoid goto warning 4008 4009 audio_track_cblk_t* cblk = track->cblk(); 4010 4011 // The first time a track is added we wait 4012 // for all its buffers to be filled before processing it 4013 int name = track->name(); 4014 // make sure that we have enough frames to mix one full buffer. 4015 // enforce this condition only once to enable draining the buffer in case the client 4016 // app does not call stop() and relies on underrun to stop: 4017 // hence the test on (mMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 4018 // during last round 4019 size_t desiredFrames; 4020 const uint32_t sampleRate = track->mAudioTrackServerProxy->getSampleRate(); 4021 AudioPlaybackRate playbackRate = track->mAudioTrackServerProxy->getPlaybackRate(); 4022 4023 desiredFrames = sourceFramesNeededWithTimestretch( 4024 sampleRate, mNormalFrameCount, mSampleRate, playbackRate.mSpeed); 4025 // TODO: ONLY USED FOR LEGACY RESAMPLERS, remove when they are removed. 4026 // add frames already consumed but not yet released by the resampler 4027 // because mAudioTrackServerProxy->framesReady() will include these frames 4028 desiredFrames += mAudioMixer->getUnreleasedFrames(track->name()); 4029 4030 uint32_t minFrames = 1; 4031 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() && 4032 (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) { 4033 minFrames = desiredFrames; 4034 } 4035 4036 size_t framesReady = track->framesReady(); 4037 if (ATRACE_ENABLED()) { 4038 // I wish we had formatted trace names 4039 char traceName[16]; 4040 strcpy(traceName, "nRdy"); 4041 int name = track->name(); 4042 if (AudioMixer::TRACK0 <= name && 4043 name < (int) (AudioMixer::TRACK0 + AudioMixer::MAX_NUM_TRACKS)) { 4044 name -= AudioMixer::TRACK0; 4045 traceName[4] = (name / 10) + '0'; 4046 traceName[5] = (name % 10) + '0'; 4047 } else { 4048 traceName[4] = '?'; 4049 traceName[5] = '?'; 4050 } 4051 traceName[6] = '\0'; 4052 ATRACE_INT(traceName, framesReady); 4053 } 4054 if ((framesReady >= minFrames) && track->isReady() && 4055 !track->isPaused() && !track->isTerminated()) 4056 { 4057 ALOGVV("track %d s=%08x [OK] on thread %p", name, cblk->mServer, this); 4058 4059 mixedTracks++; 4060 4061 // track->mainBuffer() != mSinkBuffer or mMixerBuffer means 4062 // there is an effect chain connected to the track 4063 chain.clear(); 4064 if (track->mainBuffer() != mSinkBuffer && 4065 track->mainBuffer() != mMixerBuffer) { 4066 if (mEffectBufferEnabled) { 4067 mEffectBufferValid = true; // Later can set directly. 4068 } 4069 chain = getEffectChain_l(track->sessionId()); 4070 // Delegate volume control to effect in track effect chain if needed 4071 if (chain != 0) { 4072 tracksWithEffect++; 4073 } else { 4074 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on " 4075 "session %d", 4076 name, track->sessionId()); 4077 } 4078 } 4079 4080 4081 int param = AudioMixer::VOLUME; 4082 if (track->mFillingUpStatus == Track::FS_FILLED) { 4083 // no ramp for the first volume setting 4084 track->mFillingUpStatus = Track::FS_ACTIVE; 4085 if (track->mState == TrackBase::RESUMING) { 4086 track->mState = TrackBase::ACTIVE; 4087 param = AudioMixer::RAMP_VOLUME; 4088 } 4089 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 4090 // FIXME should not make a decision based on mServer 4091 } else if (cblk->mServer != 0) { 4092 // If the track is stopped before the first frame was mixed, 4093 // do not apply ramp 4094 param = AudioMixer::RAMP_VOLUME; 4095 } 4096 4097 // compute volume for this track 4098 uint32_t vl, vr; // in U8.24 integer format 4099 float vlf, vrf, vaf; // in [0.0, 1.0] float format 4100 if (track->isPausing() || mStreamTypes[track->streamType()].mute) { 4101 vl = vr = 0; 4102 vlf = vrf = vaf = 0.; 4103 if (track->isPausing()) { 4104 track->setPaused(); 4105 } 4106 } else { 4107 4108 // read original volumes with volume control 4109 float typeVolume = mStreamTypes[track->streamType()].volume; 4110 float v = masterVolume * typeVolume; 4111 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 4112 gain_minifloat_packed_t vlr = proxy->getVolumeLR(); 4113 vlf = float_from_gain(gain_minifloat_unpack_left(vlr)); 4114 vrf = float_from_gain(gain_minifloat_unpack_right(vlr)); 4115 // track volumes come from shared memory, so can't be trusted and must be clamped 4116 if (vlf > GAIN_FLOAT_UNITY) { 4117 ALOGV("Track left volume out of range: %.3g", vlf); 4118 vlf = GAIN_FLOAT_UNITY; 4119 } 4120 if (vrf > GAIN_FLOAT_UNITY) { 4121 ALOGV("Track right volume out of range: %.3g", vrf); 4122 vrf = GAIN_FLOAT_UNITY; 4123 } 4124 // now apply the master volume and stream type volume 4125 vlf *= v; 4126 vrf *= v; 4127 // assuming master volume and stream type volume each go up to 1.0, 4128 // then derive vl and vr as U8.24 versions for the effect chain 4129 const float scaleto8_24 = MAX_GAIN_INT * MAX_GAIN_INT; 4130 vl = (uint32_t) (scaleto8_24 * vlf); 4131 vr = (uint32_t) (scaleto8_24 * vrf); 4132 // vl and vr are now in U8.24 format 4133 uint16_t sendLevel = proxy->getSendLevel_U4_12(); 4134 // send level comes from shared memory and so may be corrupt 4135 if (sendLevel > MAX_GAIN_INT) { 4136 ALOGV("Track send level out of range: %04X", sendLevel); 4137 sendLevel = MAX_GAIN_INT; 4138 } 4139 // vaf is represented as [0.0, 1.0] float by rescaling sendLevel 4140 vaf = v * sendLevel * (1. / MAX_GAIN_INT); 4141 } 4142 4143 // Delegate volume control to effect in track effect chain if needed 4144 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 4145 // Do not ramp volume if volume is controlled by effect 4146 param = AudioMixer::VOLUME; 4147 // Update remaining floating point volume levels 4148 vlf = (float)vl / (1 << 24); 4149 vrf = (float)vr / (1 << 24); 4150 track->mHasVolumeController = true; 4151 } else { 4152 // force no volume ramp when volume controller was just disabled or removed 4153 // from effect chain to avoid volume spike 4154 if (track->mHasVolumeController) { 4155 param = AudioMixer::VOLUME; 4156 } 4157 track->mHasVolumeController = false; 4158 } 4159 4160 // XXX: these things DON'T need to be done each time 4161 mAudioMixer->setBufferProvider(name, track); 4162 mAudioMixer->enable(name); 4163 4164 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, &vlf); 4165 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, &vrf); 4166 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, &vaf); 4167 mAudioMixer->setParameter( 4168 name, 4169 AudioMixer::TRACK, 4170 AudioMixer::FORMAT, (void *)track->format()); 4171 mAudioMixer->setParameter( 4172 name, 4173 AudioMixer::TRACK, 4174 AudioMixer::CHANNEL_MASK, (void *)(uintptr_t)track->channelMask()); 4175 mAudioMixer->setParameter( 4176 name, 4177 AudioMixer::TRACK, 4178 AudioMixer::MIXER_CHANNEL_MASK, (void *)(uintptr_t)mChannelMask); 4179 // limit track sample rate to 2 x output sample rate, which changes at re-configuration 4180 uint32_t maxSampleRate = mSampleRate * AUDIO_RESAMPLER_DOWN_RATIO_MAX; 4181 uint32_t reqSampleRate = track->mAudioTrackServerProxy->getSampleRate(); 4182 if (reqSampleRate == 0) { 4183 reqSampleRate = mSampleRate; 4184 } else if (reqSampleRate > maxSampleRate) { 4185 reqSampleRate = maxSampleRate; 4186 } 4187 mAudioMixer->setParameter( 4188 name, 4189 AudioMixer::RESAMPLE, 4190 AudioMixer::SAMPLE_RATE, 4191 (void *)(uintptr_t)reqSampleRate); 4192 4193 AudioPlaybackRate playbackRate = track->mAudioTrackServerProxy->getPlaybackRate(); 4194 mAudioMixer->setParameter( 4195 name, 4196 AudioMixer::TIMESTRETCH, 4197 AudioMixer::PLAYBACK_RATE, 4198 &playbackRate); 4199 4200 /* 4201 * Select the appropriate output buffer for the track. 4202 * 4203 * Tracks with effects go into their own effects chain buffer 4204 * and from there into either mEffectBuffer or mSinkBuffer. 4205 * 4206 * Other tracks can use mMixerBuffer for higher precision 4207 * channel accumulation. If this buffer is enabled 4208 * (mMixerBufferEnabled true), then selected tracks will accumulate 4209 * into it. 4210 * 4211 */ 4212 if (mMixerBufferEnabled 4213 && (track->mainBuffer() == mSinkBuffer 4214 || track->mainBuffer() == mMixerBuffer)) { 4215 mAudioMixer->setParameter( 4216 name, 4217 AudioMixer::TRACK, 4218 AudioMixer::MIXER_FORMAT, (void *)mMixerBufferFormat); 4219 mAudioMixer->setParameter( 4220 name, 4221 AudioMixer::TRACK, 4222 AudioMixer::MAIN_BUFFER, (void *)mMixerBuffer); 4223 // TODO: override track->mainBuffer()? 4224 mMixerBufferValid = true; 4225 } else { 4226 mAudioMixer->setParameter( 4227 name, 4228 AudioMixer::TRACK, 4229 AudioMixer::MIXER_FORMAT, (void *)AUDIO_FORMAT_PCM_16_BIT); 4230 mAudioMixer->setParameter( 4231 name, 4232 AudioMixer::TRACK, 4233 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 4234 } 4235 mAudioMixer->setParameter( 4236 name, 4237 AudioMixer::TRACK, 4238 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 4239 4240 // reset retry count 4241 track->mRetryCount = kMaxTrackRetries; 4242 4243 // If one track is ready, set the mixer ready if: 4244 // - the mixer was not ready during previous round OR 4245 // - no other track is not ready 4246 if (mMixerStatusIgnoringFastTracks != MIXER_TRACKS_READY || 4247 mixerStatus != MIXER_TRACKS_ENABLED) { 4248 mixerStatus = MIXER_TRACKS_READY; 4249 } 4250 } else { 4251 if (framesReady < desiredFrames && !track->isStopped() && !track->isPaused()) { 4252 ALOGV("track(%p) underrun, framesReady(%zu) < framesDesired(%zd)", 4253 track, framesReady, desiredFrames); 4254 track->mAudioTrackServerProxy->tallyUnderrunFrames(desiredFrames); 4255 } else { 4256 track->mAudioTrackServerProxy->tallyUnderrunFrames(0); 4257 } 4258 4259 // clear effect chain input buffer if an active track underruns to avoid sending 4260 // previous audio buffer again to effects 4261 chain = getEffectChain_l(track->sessionId()); 4262 if (chain != 0) { 4263 chain->clearInputBuffer(); 4264 } 4265 4266 ALOGVV("track %d s=%08x [NOT READY] on thread %p", name, cblk->mServer, this); 4267 if ((track->sharedBuffer() != 0) || track->isTerminated() || 4268 track->isStopped() || track->isPaused()) { 4269 // We have consumed all the buffers of this track. 4270 // Remove it from the list of active tracks. 4271 // TODO: use actual buffer filling status instead of latency when available from 4272 // audio HAL 4273 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 4274 int64_t framesWritten = mBytesWritten / mFrameSize; 4275 if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) { 4276 if (track->isStopped()) { 4277 track->reset(); 4278 } 4279 tracksToRemove->add(track); 4280 } 4281 } else { 4282 // No buffers for this track. Give it a few chances to 4283 // fill a buffer, then remove it from active list. 4284 if (--(track->mRetryCount) <= 0) { 4285 ALOGI("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 4286 tracksToRemove->add(track); 4287 // indicate to client process that the track was disabled because of underrun; 4288 // it will then automatically call start() when data is available 4289 track->disable(); 4290 // If one track is not ready, mark the mixer also not ready if: 4291 // - the mixer was ready during previous round OR 4292 // - no other track is ready 4293 } else if (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY || 4294 mixerStatus != MIXER_TRACKS_READY) { 4295 mixerStatus = MIXER_TRACKS_ENABLED; 4296 } 4297 } 4298 mAudioMixer->disable(name); 4299 } 4300 4301 } // local variable scope to avoid goto warning 4302 4303 } 4304 4305 // Push the new FastMixer state if necessary 4306 bool pauseAudioWatchdog = false; 4307 if (didModify) { 4308 state->mFastTracksGen++; 4309 // if the fast mixer was active, but now there are no fast tracks, then put it in cold idle 4310 if (kUseFastMixer == FastMixer_Dynamic && 4311 state->mCommand == FastMixerState::MIX_WRITE && state->mTrackMask <= 1) { 4312 state->mCommand = FastMixerState::COLD_IDLE; 4313 state->mColdFutexAddr = &mFastMixerFutex; 4314 state->mColdGen++; 4315 mFastMixerFutex = 0; 4316 if (kUseFastMixer == FastMixer_Dynamic) { 4317 mNormalSink = mOutputSink; 4318 } 4319 // If we go into cold idle, need to wait for acknowledgement 4320 // so that fast mixer stops doing I/O. 4321 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 4322 pauseAudioWatchdog = true; 4323 } 4324 } 4325 if (sq != NULL) { 4326 sq->end(didModify); 4327 sq->push(block); 4328 } 4329#ifdef AUDIO_WATCHDOG 4330 if (pauseAudioWatchdog && mAudioWatchdog != 0) { 4331 mAudioWatchdog->pause(); 4332 } 4333#endif 4334 4335 // Now perform the deferred reset on fast tracks that have stopped 4336 while (resetMask != 0) { 4337 size_t i = __builtin_ctz(resetMask); 4338 ALOG_ASSERT(i < count); 4339 resetMask &= ~(1 << i); 4340 sp<Track> t = mActiveTracks[i].promote(); 4341 if (t == 0) { 4342 continue; 4343 } 4344 Track* track = t.get(); 4345 ALOG_ASSERT(track->isFastTrack() && track->isStopped()); 4346 track->reset(); 4347 } 4348 4349 // remove all the tracks that need to be... 4350 removeTracks_l(*tracksToRemove); 4351 4352 if (getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX) != 0) { 4353 mEffectBufferValid = true; 4354 } 4355 4356 if (mEffectBufferValid) { 4357 // as long as there are effects we should clear the effects buffer, to avoid 4358 // passing a non-clean buffer to the effect chain 4359 memset(mEffectBuffer, 0, mEffectBufferSize); 4360 } 4361 // sink or mix buffer must be cleared if all tracks are connected to an 4362 // effect chain as in this case the mixer will not write to the sink or mix buffer 4363 // and track effects will accumulate into it 4364 if ((mBytesRemaining == 0) && ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 4365 (mixedTracks == 0 && fastTracks > 0))) { 4366 // FIXME as a performance optimization, should remember previous zero status 4367 if (mMixerBufferValid) { 4368 memset(mMixerBuffer, 0, mMixerBufferSize); 4369 // TODO: In testing, mSinkBuffer below need not be cleared because 4370 // the PlaybackThread::threadLoop() copies mMixerBuffer into mSinkBuffer 4371 // after mixing. 4372 // 4373 // To enforce this guarantee: 4374 // ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 4375 // (mixedTracks == 0 && fastTracks > 0)) 4376 // must imply MIXER_TRACKS_READY. 4377 // Later, we may clear buffers regardless, and skip much of this logic. 4378 } 4379 // FIXME as a performance optimization, should remember previous zero status 4380 memset(mSinkBuffer, 0, mNormalFrameCount * mFrameSize); 4381 } 4382 4383 // if any fast tracks, then status is ready 4384 mMixerStatusIgnoringFastTracks = mixerStatus; 4385 if (fastTracks > 0) { 4386 mixerStatus = MIXER_TRACKS_READY; 4387 } 4388 return mixerStatus; 4389} 4390 4391// getTrackName_l() must be called with ThreadBase::mLock held 4392int AudioFlinger::MixerThread::getTrackName_l(audio_channel_mask_t channelMask, 4393 audio_format_t format, audio_session_t sessionId) 4394{ 4395 return mAudioMixer->getTrackName(channelMask, format, sessionId); 4396} 4397 4398// deleteTrackName_l() must be called with ThreadBase::mLock held 4399void AudioFlinger::MixerThread::deleteTrackName_l(int name) 4400{ 4401 ALOGV("remove track (%d) and delete from mixer", name); 4402 mAudioMixer->deleteTrackName(name); 4403} 4404 4405// checkForNewParameter_l() must be called with ThreadBase::mLock held 4406bool AudioFlinger::MixerThread::checkForNewParameter_l(const String8& keyValuePair, 4407 status_t& status) 4408{ 4409 bool reconfig = false; 4410 bool a2dpDeviceChanged = false; 4411 4412 status = NO_ERROR; 4413 4414 AutoPark<FastMixer> park(mFastMixer); 4415 4416 AudioParameter param = AudioParameter(keyValuePair); 4417 int value; 4418 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 4419 reconfig = true; 4420 } 4421 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 4422 if (!isValidPcmSinkFormat((audio_format_t) value)) { 4423 status = BAD_VALUE; 4424 } else { 4425 // no need to save value, since it's constant 4426 reconfig = true; 4427 } 4428 } 4429 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 4430 if (!isValidPcmSinkChannelMask((audio_channel_mask_t) value)) { 4431 status = BAD_VALUE; 4432 } else { 4433 // no need to save value, since it's constant 4434 reconfig = true; 4435 } 4436 } 4437 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 4438 // do not accept frame count changes if tracks are open as the track buffer 4439 // size depends on frame count and correct behavior would not be guaranteed 4440 // if frame count is changed after track creation 4441 if (!mTracks.isEmpty()) { 4442 status = INVALID_OPERATION; 4443 } else { 4444 reconfig = true; 4445 } 4446 } 4447 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 4448#ifdef ADD_BATTERY_DATA 4449 // when changing the audio output device, call addBatteryData to notify 4450 // the change 4451 if (mOutDevice != value) { 4452 uint32_t params = 0; 4453 // check whether speaker is on 4454 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 4455 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 4456 } 4457 4458 audio_devices_t deviceWithoutSpeaker 4459 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 4460 // check if any other device (except speaker) is on 4461 if (value & deviceWithoutSpeaker) { 4462 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 4463 } 4464 4465 if (params != 0) { 4466 addBatteryData(params); 4467 } 4468 } 4469#endif 4470 4471 // forward device change to effects that have requested to be 4472 // aware of attached audio device. 4473 if (value != AUDIO_DEVICE_NONE) { 4474 a2dpDeviceChanged = 4475 (mOutDevice & AUDIO_DEVICE_OUT_ALL_A2DP) != (value & AUDIO_DEVICE_OUT_ALL_A2DP); 4476 mOutDevice = value; 4477 for (size_t i = 0; i < mEffectChains.size(); i++) { 4478 mEffectChains[i]->setDevice_l(mOutDevice); 4479 } 4480 } 4481 } 4482 4483 if (status == NO_ERROR) { 4484 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 4485 keyValuePair.string()); 4486 if (!mStandby && status == INVALID_OPERATION) { 4487 mOutput->standby(); 4488 mStandby = true; 4489 mBytesWritten = 0; 4490 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 4491 keyValuePair.string()); 4492 } 4493 if (status == NO_ERROR && reconfig) { 4494 readOutputParameters_l(); 4495 delete mAudioMixer; 4496 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 4497 for (size_t i = 0; i < mTracks.size() ; i++) { 4498 int name = getTrackName_l(mTracks[i]->mChannelMask, 4499 mTracks[i]->mFormat, mTracks[i]->mSessionId); 4500 if (name < 0) { 4501 break; 4502 } 4503 mTracks[i]->mName = name; 4504 } 4505 sendIoConfigEvent_l(AUDIO_OUTPUT_CONFIG_CHANGED); 4506 } 4507 } 4508 4509 return reconfig || a2dpDeviceChanged; 4510} 4511 4512 4513void AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 4514{ 4515 PlaybackThread::dumpInternals(fd, args); 4516 dprintf(fd, " Thread throttle time (msecs): %u\n", mThreadThrottleTimeMs); 4517 dprintf(fd, " AudioMixer tracks: 0x%08x\n", mAudioMixer->trackNames()); 4518 dprintf(fd, " Master mono: %s\n", mMasterMono ? "on" : "off"); 4519 4520 // Make a non-atomic copy of fast mixer dump state so it won't change underneath us 4521 // while we are dumping it. It may be inconsistent, but it won't mutate! 4522 // This is a large object so we place it on the heap. 4523 // FIXME 25972958: Need an intelligent copy constructor that does not touch unused pages. 4524 const FastMixerDumpState *copy = new FastMixerDumpState(mFastMixerDumpState); 4525 copy->dump(fd); 4526 delete copy; 4527 4528#ifdef STATE_QUEUE_DUMP 4529 // Similar for state queue 4530 StateQueueObserverDump observerCopy = mStateQueueObserverDump; 4531 observerCopy.dump(fd); 4532 StateQueueMutatorDump mutatorCopy = mStateQueueMutatorDump; 4533 mutatorCopy.dump(fd); 4534#endif 4535 4536#ifdef TEE_SINK 4537 // Write the tee output to a .wav file 4538 dumpTee(fd, mTeeSource, mId); 4539#endif 4540 4541#ifdef AUDIO_WATCHDOG 4542 if (mAudioWatchdog != 0) { 4543 // Make a non-atomic copy of audio watchdog dump so it won't change underneath us 4544 AudioWatchdogDump wdCopy = mAudioWatchdogDump; 4545 wdCopy.dump(fd); 4546 } 4547#endif 4548} 4549 4550uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const 4551{ 4552 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000) / 2; 4553} 4554 4555uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() const 4556{ 4557 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000); 4558} 4559 4560void AudioFlinger::MixerThread::cacheParameters_l() 4561{ 4562 PlaybackThread::cacheParameters_l(); 4563 4564 // FIXME: Relaxed timing because of a certain device that can't meet latency 4565 // Should be reduced to 2x after the vendor fixes the driver issue 4566 // increase threshold again due to low power audio mode. The way this warning 4567 // threshold is calculated and its usefulness should be reconsidered anyway. 4568 maxPeriod = seconds(mNormalFrameCount) / mSampleRate * 15; 4569} 4570 4571// ---------------------------------------------------------------------------- 4572 4573AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 4574 AudioStreamOut* output, audio_io_handle_t id, audio_devices_t device, bool systemReady, 4575 uint32_t bitRate) 4576 : PlaybackThread(audioFlinger, output, id, device, DIRECT, systemReady, bitRate) 4577 // mLeftVolFloat, mRightVolFloat 4578{ 4579} 4580 4581AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 4582 AudioStreamOut* output, audio_io_handle_t id, uint32_t device, 4583 ThreadBase::type_t type, bool systemReady, uint32_t bitRate) 4584 : PlaybackThread(audioFlinger, output, id, device, type, systemReady, bitRate) 4585 // mLeftVolFloat, mRightVolFloat 4586{ 4587} 4588 4589AudioFlinger::DirectOutputThread::~DirectOutputThread() 4590{ 4591} 4592 4593void AudioFlinger::DirectOutputThread::processVolume_l(Track *track, bool lastTrack) 4594{ 4595 float left, right; 4596 4597 if (mMasterMute || mStreamTypes[track->streamType()].mute) { 4598 left = right = 0; 4599 } else { 4600 float typeVolume = mStreamTypes[track->streamType()].volume; 4601 float v = mMasterVolume * typeVolume; 4602 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 4603 gain_minifloat_packed_t vlr = proxy->getVolumeLR(); 4604 left = float_from_gain(gain_minifloat_unpack_left(vlr)); 4605 if (left > GAIN_FLOAT_UNITY) { 4606 left = GAIN_FLOAT_UNITY; 4607 } 4608 left *= v; 4609 right = float_from_gain(gain_minifloat_unpack_right(vlr)); 4610 if (right > GAIN_FLOAT_UNITY) { 4611 right = GAIN_FLOAT_UNITY; 4612 } 4613 right *= v; 4614 } 4615 4616 if (lastTrack) { 4617 if (left != mLeftVolFloat || right != mRightVolFloat) { 4618 mLeftVolFloat = left; 4619 mRightVolFloat = right; 4620 4621 // Convert volumes from float to 8.24 4622 uint32_t vl = (uint32_t)(left * (1 << 24)); 4623 uint32_t vr = (uint32_t)(right * (1 << 24)); 4624 4625 // Delegate volume control to effect in track effect chain if needed 4626 // only one effect chain can be present on DirectOutputThread, so if 4627 // there is one, the track is connected to it 4628 if (!mEffectChains.isEmpty()) { 4629 mEffectChains[0]->setVolume_l(&vl, &vr); 4630 left = (float)vl / (1 << 24); 4631 right = (float)vr / (1 << 24); 4632 } 4633 if (mOutput->stream->set_volume) { 4634 mOutput->stream->set_volume(mOutput->stream, left, right); 4635 } 4636 } 4637 } 4638} 4639 4640void AudioFlinger::DirectOutputThread::onAddNewTrack_l() 4641{ 4642 sp<Track> previousTrack = mPreviousTrack.promote(); 4643 sp<Track> latestTrack = mLatestActiveTrack.promote(); 4644 4645 if (previousTrack != 0 && latestTrack != 0) { 4646 if (mType == DIRECT) { 4647 if (previousTrack.get() != latestTrack.get()) { 4648 mFlushPending = true; 4649 } 4650 } else /* mType == OFFLOAD */ { 4651 if (previousTrack->sessionId() != latestTrack->sessionId()) { 4652 mFlushPending = true; 4653 } 4654 } 4655 } 4656 PlaybackThread::onAddNewTrack_l(); 4657} 4658 4659AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l( 4660 Vector< sp<Track> > *tracksToRemove 4661) 4662{ 4663 size_t count = mActiveTracks.size(); 4664 mixer_state mixerStatus = MIXER_IDLE; 4665 bool doHwPause = false; 4666 bool doHwResume = false; 4667 4668 // find out which tracks need to be processed 4669 for (size_t i = 0; i < count; i++) { 4670 sp<Track> t = mActiveTracks[i].promote(); 4671 // The track died recently 4672 if (t == 0) { 4673 continue; 4674 } 4675 4676 if (t->isInvalid()) { 4677 ALOGW("An invalidated track shouldn't be in active list"); 4678 tracksToRemove->add(t); 4679 continue; 4680 } 4681 4682 Track* const track = t.get(); 4683#ifdef VERY_VERY_VERBOSE_LOGGING 4684 audio_track_cblk_t* cblk = track->cblk(); 4685#endif 4686 // Only consider last track started for volume and mixer state control. 4687 // In theory an older track could underrun and restart after the new one starts 4688 // but as we only care about the transition phase between two tracks on a 4689 // direct output, it is not a problem to ignore the underrun case. 4690 sp<Track> l = mLatestActiveTrack.promote(); 4691 bool last = l.get() == track; 4692 4693 if (track->isPausing()) { 4694 track->setPaused(); 4695 if (mHwSupportsPause && last && !mHwPaused) { 4696 doHwPause = true; 4697 mHwPaused = true; 4698 } 4699 tracksToRemove->add(track); 4700 } else if (track->isFlushPending()) { 4701 track->flushAck(); 4702 if (last) { 4703 mFlushPending = true; 4704 } 4705 } else if (track->isResumePending()) { 4706 track->resumeAck(); 4707 if (last && mHwPaused) { 4708 doHwResume = true; 4709 mHwPaused = false; 4710 } 4711 } 4712 4713 // The first time a track is added we wait 4714 // for all its buffers to be filled before processing it. 4715 // Allow draining the buffer in case the client 4716 // app does not call stop() and relies on underrun to stop: 4717 // hence the test on (track->mRetryCount > 1). 4718 // If retryCount<=1 then track is about to underrun and be removed. 4719 // Do not use a high threshold for compressed audio. 4720 uint32_t minFrames; 4721 if ((track->sharedBuffer() == 0) && !track->isStopping_1() && !track->isPausing() 4722 && (track->mRetryCount > 1) && audio_has_proportional_frames(mFormat)) { 4723 minFrames = mNormalFrameCount; 4724 } else { 4725 minFrames = 1; 4726 } 4727 4728 if ((track->framesReady() >= minFrames) && track->isReady() && !track->isPaused() && 4729 !track->isStopping_2() && !track->isStopped()) 4730 { 4731 ALOGVV("track %d s=%08x [OK]", track->name(), cblk->mServer); 4732 4733 if (track->mFillingUpStatus == Track::FS_FILLED) { 4734 track->mFillingUpStatus = Track::FS_ACTIVE; 4735 // make sure processVolume_l() will apply new volume even if 0 4736 mLeftVolFloat = mRightVolFloat = -1.0; 4737 if (!mHwSupportsPause) { 4738 track->resumeAck(); 4739 } 4740 } 4741 4742 // compute volume for this track 4743 processVolume_l(track, last); 4744 if (last) { 4745 sp<Track> previousTrack = mPreviousTrack.promote(); 4746 if (previousTrack != 0) { 4747 if (track != previousTrack.get()) { 4748 // Flush any data still being written from last track 4749 mBytesRemaining = 0; 4750 // Invalidate previous track to force a seek when resuming. 4751 previousTrack->invalidate(); 4752 } 4753 } 4754 mPreviousTrack = track; 4755 4756 // reset retry count 4757 track->mRetryCount = kMaxTrackRetriesDirect; 4758 mActiveTrack = t; 4759 mixerStatus = MIXER_TRACKS_READY; 4760 if (mHwPaused) { 4761 doHwResume = true; 4762 mHwPaused = false; 4763 } 4764 } 4765 } else { 4766 // clear effect chain input buffer if the last active track started underruns 4767 // to avoid sending previous audio buffer again to effects 4768 if (!mEffectChains.isEmpty() && last) { 4769 mEffectChains[0]->clearInputBuffer(); 4770 } 4771 if (track->isStopping_1()) { 4772 track->mState = TrackBase::STOPPING_2; 4773 if (last && mHwPaused) { 4774 doHwResume = true; 4775 mHwPaused = false; 4776 } 4777 } 4778 if ((track->sharedBuffer() != 0) || track->isStopped() || 4779 track->isStopping_2() || track->isPaused()) { 4780 // We have consumed all the buffers of this track. 4781 // Remove it from the list of active tracks. 4782 size_t audioHALFrames; 4783 if (audio_has_proportional_frames(mFormat)) { 4784 audioHALFrames = (latency_l() * mSampleRate) / 1000; 4785 } else { 4786 audioHALFrames = 0; 4787 } 4788 4789 int64_t framesWritten = mBytesWritten / mFrameSize; 4790 if (mStandby || !last || 4791 track->presentationComplete(framesWritten, audioHALFrames)) { 4792 if (track->isStopping_2()) { 4793 track->mState = TrackBase::STOPPED; 4794 } 4795 if (track->isStopped()) { 4796 track->reset(); 4797 } 4798 tracksToRemove->add(track); 4799 } 4800 } else { 4801 // No buffers for this track. Give it a few chances to 4802 // fill a buffer, then remove it from active list. 4803 // Only consider last track started for mixer state control 4804 if (--(track->mRetryCount) <= 0) { 4805 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 4806 tracksToRemove->add(track); 4807 // indicate to client process that the track was disabled because of underrun; 4808 // it will then automatically call start() when data is available 4809 track->disable(); 4810 } else if (last) { 4811 ALOGW("pause because of UNDERRUN, framesReady = %zu," 4812 "minFrames = %u, mFormat = %#x", 4813 track->framesReady(), minFrames, mFormat); 4814 mixerStatus = MIXER_TRACKS_ENABLED; 4815 if (mHwSupportsPause && !mHwPaused && !mStandby) { 4816 doHwPause = true; 4817 mHwPaused = true; 4818 } 4819 } 4820 } 4821 } 4822 } 4823 4824 // if an active track did not command a flush, check for pending flush on stopped tracks 4825 if (!mFlushPending) { 4826 for (size_t i = 0; i < mTracks.size(); i++) { 4827 if (mTracks[i]->isFlushPending()) { 4828 mTracks[i]->flushAck(); 4829 mFlushPending = true; 4830 } 4831 } 4832 } 4833 4834 // make sure the pause/flush/resume sequence is executed in the right order. 4835 // If a flush is pending and a track is active but the HW is not paused, force a HW pause 4836 // before flush and then resume HW. This can happen in case of pause/flush/resume 4837 // if resume is received before pause is executed. 4838 if (mHwSupportsPause && !mStandby && 4839 (doHwPause || (mFlushPending && !mHwPaused && (count != 0)))) { 4840 mOutput->stream->pause(mOutput->stream); 4841 } 4842 if (mFlushPending) { 4843 flushHw_l(); 4844 } 4845 if (mHwSupportsPause && !mStandby && doHwResume) { 4846 mOutput->stream->resume(mOutput->stream); 4847 } 4848 // remove all the tracks that need to be... 4849 removeTracks_l(*tracksToRemove); 4850 4851 return mixerStatus; 4852} 4853 4854void AudioFlinger::DirectOutputThread::threadLoop_mix() 4855{ 4856 size_t frameCount = mFrameCount; 4857 int8_t *curBuf = (int8_t *)mSinkBuffer; 4858 // output audio to hardware 4859 while (frameCount) { 4860 AudioBufferProvider::Buffer buffer; 4861 buffer.frameCount = frameCount; 4862 status_t status = mActiveTrack->getNextBuffer(&buffer); 4863 if (status != NO_ERROR || buffer.raw == NULL) { 4864 // no need to pad with 0 for compressed audio 4865 if (audio_has_proportional_frames(mFormat)) { 4866 memset(curBuf, 0, frameCount * mFrameSize); 4867 } 4868 break; 4869 } 4870 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 4871 frameCount -= buffer.frameCount; 4872 curBuf += buffer.frameCount * mFrameSize; 4873 mActiveTrack->releaseBuffer(&buffer); 4874 } 4875 mCurrentWriteLength = curBuf - (int8_t *)mSinkBuffer; 4876 mSleepTimeUs = 0; 4877 mStandbyTimeNs = systemTime() + mStandbyDelayNs; 4878 mActiveTrack.clear(); 4879} 4880 4881void AudioFlinger::DirectOutputThread::threadLoop_sleepTime() 4882{ 4883 // do not write to HAL when paused 4884 if (mHwPaused || (usesHwAvSync() && mStandby)) { 4885 mSleepTimeUs = mIdleSleepTimeUs; 4886 return; 4887 } 4888 if (mSleepTimeUs == 0) { 4889 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4890 // For compressed offload, use faster sleep time when underruning until more than an 4891 // entire buffer was written to the audio HAL 4892 if (!audio_has_proportional_frames(mFormat) && 4893 (mType == OFFLOAD) && (mBytesWritten < (int64_t) mBufferSize)) { 4894 mSleepTimeUs = kDirectMinSleepTimeUs; 4895 } else { 4896 mSleepTimeUs = mActiveSleepTimeUs; 4897 } 4898 } else { 4899 mSleepTimeUs = mIdleSleepTimeUs; 4900 } 4901 } else if (mBytesWritten != 0 && audio_has_proportional_frames(mFormat)) { 4902 memset(mSinkBuffer, 0, mFrameCount * mFrameSize); 4903 mSleepTimeUs = 0; 4904 } 4905} 4906 4907void AudioFlinger::DirectOutputThread::threadLoop_exit() 4908{ 4909 { 4910 Mutex::Autolock _l(mLock); 4911 for (size_t i = 0; i < mTracks.size(); i++) { 4912 if (mTracks[i]->isFlushPending()) { 4913 mTracks[i]->flushAck(); 4914 mFlushPending = true; 4915 } 4916 } 4917 if (mFlushPending) { 4918 flushHw_l(); 4919 } 4920 } 4921 PlaybackThread::threadLoop_exit(); 4922} 4923 4924// must be called with thread mutex locked 4925bool AudioFlinger::DirectOutputThread::shouldStandby_l() 4926{ 4927 bool trackPaused = false; 4928 bool trackStopped = false; 4929 4930 if ((mType == DIRECT) && audio_is_linear_pcm(mFormat) && !usesHwAvSync()) { 4931 return !mStandby; 4932 } 4933 4934 // do not put the HAL in standby when paused. AwesomePlayer clear the offloaded AudioTrack 4935 // after a timeout and we will enter standby then. 4936 if (mTracks.size() > 0) { 4937 trackPaused = mTracks[mTracks.size() - 1]->isPaused(); 4938 trackStopped = mTracks[mTracks.size() - 1]->isStopped() || 4939 mTracks[mTracks.size() - 1]->mState == TrackBase::IDLE; 4940 } 4941 4942 return !mStandby && !(trackPaused || (mHwPaused && !trackStopped)); 4943} 4944 4945// getTrackName_l() must be called with ThreadBase::mLock held 4946int AudioFlinger::DirectOutputThread::getTrackName_l(audio_channel_mask_t channelMask __unused, 4947 audio_format_t format __unused, audio_session_t sessionId __unused) 4948{ 4949 return 0; 4950} 4951 4952// deleteTrackName_l() must be called with ThreadBase::mLock held 4953void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name __unused) 4954{ 4955} 4956 4957// checkForNewParameter_l() must be called with ThreadBase::mLock held 4958bool AudioFlinger::DirectOutputThread::checkForNewParameter_l(const String8& keyValuePair, 4959 status_t& status) 4960{ 4961 bool reconfig = false; 4962 bool a2dpDeviceChanged = false; 4963 4964 status = NO_ERROR; 4965 4966 AudioParameter param = AudioParameter(keyValuePair); 4967 int value; 4968 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 4969 // forward device change to effects that have requested to be 4970 // aware of attached audio device. 4971 if (value != AUDIO_DEVICE_NONE) { 4972 a2dpDeviceChanged = 4973 (mOutDevice & AUDIO_DEVICE_OUT_ALL_A2DP) != (value & AUDIO_DEVICE_OUT_ALL_A2DP); 4974 mOutDevice = value; 4975 for (size_t i = 0; i < mEffectChains.size(); i++) { 4976 mEffectChains[i]->setDevice_l(mOutDevice); 4977 } 4978 } 4979 } 4980 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 4981 // do not accept frame count changes if tracks are open as the track buffer 4982 // size depends on frame count and correct behavior would not be garantied 4983 // if frame count is changed after track creation 4984 if (!mTracks.isEmpty()) { 4985 status = INVALID_OPERATION; 4986 } else { 4987 reconfig = true; 4988 } 4989 } 4990 if (status == NO_ERROR) { 4991 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 4992 keyValuePair.string()); 4993 if (!mStandby && status == INVALID_OPERATION) { 4994 mOutput->standby(); 4995 mStandby = true; 4996 mBytesWritten = 0; 4997 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 4998 keyValuePair.string()); 4999 } 5000 if (status == NO_ERROR && reconfig) { 5001 readOutputParameters_l(); 5002 sendIoConfigEvent_l(AUDIO_OUTPUT_CONFIG_CHANGED); 5003 } 5004 } 5005 5006 return reconfig || a2dpDeviceChanged; 5007} 5008 5009uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() const 5010{ 5011 uint32_t time; 5012 if (audio_has_proportional_frames(mFormat)) { 5013 time = PlaybackThread::activeSleepTimeUs(); 5014 } else { 5015 time = kDirectMinSleepTimeUs; 5016 } 5017 return time; 5018} 5019 5020uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() const 5021{ 5022 uint32_t time; 5023 if (audio_has_proportional_frames(mFormat)) { 5024 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 5025 } else { 5026 time = kDirectMinSleepTimeUs; 5027 } 5028 return time; 5029} 5030 5031uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() const 5032{ 5033 uint32_t time; 5034 if (audio_has_proportional_frames(mFormat)) { 5035 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 5036 } else { 5037 time = kDirectMinSleepTimeUs; 5038 } 5039 return time; 5040} 5041 5042void AudioFlinger::DirectOutputThread::cacheParameters_l() 5043{ 5044 PlaybackThread::cacheParameters_l(); 5045 5046 // use shorter standby delay as on normal output to release 5047 // hardware resources as soon as possible 5048 // no delay on outputs with HW A/V sync 5049 if (usesHwAvSync()) { 5050 mStandbyDelayNs = 0; 5051 } else if ((mType == OFFLOAD) && !audio_has_proportional_frames(mFormat)) { 5052 mStandbyDelayNs = kOffloadStandbyDelayNs; 5053 } else { 5054 mStandbyDelayNs = microseconds(mActiveSleepTimeUs*2); 5055 } 5056} 5057 5058void AudioFlinger::DirectOutputThread::flushHw_l() 5059{ 5060 mOutput->flush(); 5061 mHwPaused = false; 5062 mFlushPending = false; 5063} 5064 5065// ---------------------------------------------------------------------------- 5066 5067AudioFlinger::AsyncCallbackThread::AsyncCallbackThread( 5068 const wp<AudioFlinger::PlaybackThread>& playbackThread) 5069 : Thread(false /*canCallJava*/), 5070 mPlaybackThread(playbackThread), 5071 mWriteAckSequence(0), 5072 mDrainSequence(0) 5073{ 5074} 5075 5076AudioFlinger::AsyncCallbackThread::~AsyncCallbackThread() 5077{ 5078} 5079 5080void AudioFlinger::AsyncCallbackThread::onFirstRef() 5081{ 5082 run("Offload Cbk", ANDROID_PRIORITY_URGENT_AUDIO); 5083} 5084 5085bool AudioFlinger::AsyncCallbackThread::threadLoop() 5086{ 5087 while (!exitPending()) { 5088 uint32_t writeAckSequence; 5089 uint32_t drainSequence; 5090 5091 { 5092 Mutex::Autolock _l(mLock); 5093 while (!((mWriteAckSequence & 1) || 5094 (mDrainSequence & 1) || 5095 exitPending())) { 5096 mWaitWorkCV.wait(mLock); 5097 } 5098 5099 if (exitPending()) { 5100 break; 5101 } 5102 ALOGV("AsyncCallbackThread mWriteAckSequence %d mDrainSequence %d", 5103 mWriteAckSequence, mDrainSequence); 5104 writeAckSequence = mWriteAckSequence; 5105 mWriteAckSequence &= ~1; 5106 drainSequence = mDrainSequence; 5107 mDrainSequence &= ~1; 5108 } 5109 { 5110 sp<AudioFlinger::PlaybackThread> playbackThread = mPlaybackThread.promote(); 5111 if (playbackThread != 0) { 5112 if (writeAckSequence & 1) { 5113 playbackThread->resetWriteBlocked(writeAckSequence >> 1); 5114 } 5115 if (drainSequence & 1) { 5116 playbackThread->resetDraining(drainSequence >> 1); 5117 } 5118 } 5119 } 5120 } 5121 return false; 5122} 5123 5124void AudioFlinger::AsyncCallbackThread::exit() 5125{ 5126 ALOGV("AsyncCallbackThread::exit"); 5127 Mutex::Autolock _l(mLock); 5128 requestExit(); 5129 mWaitWorkCV.broadcast(); 5130} 5131 5132void AudioFlinger::AsyncCallbackThread::setWriteBlocked(uint32_t sequence) 5133{ 5134 Mutex::Autolock _l(mLock); 5135 // bit 0 is cleared 5136 mWriteAckSequence = sequence << 1; 5137} 5138 5139void AudioFlinger::AsyncCallbackThread::resetWriteBlocked() 5140{ 5141 Mutex::Autolock _l(mLock); 5142 // ignore unexpected callbacks 5143 if (mWriteAckSequence & 2) { 5144 mWriteAckSequence |= 1; 5145 mWaitWorkCV.signal(); 5146 } 5147} 5148 5149void AudioFlinger::AsyncCallbackThread::setDraining(uint32_t sequence) 5150{ 5151 Mutex::Autolock _l(mLock); 5152 // bit 0 is cleared 5153 mDrainSequence = sequence << 1; 5154} 5155 5156void AudioFlinger::AsyncCallbackThread::resetDraining() 5157{ 5158 Mutex::Autolock _l(mLock); 5159 // ignore unexpected callbacks 5160 if (mDrainSequence & 2) { 5161 mDrainSequence |= 1; 5162 mWaitWorkCV.signal(); 5163 } 5164} 5165 5166 5167// ---------------------------------------------------------------------------- 5168AudioFlinger::OffloadThread::OffloadThread(const sp<AudioFlinger>& audioFlinger, 5169 AudioStreamOut* output, audio_io_handle_t id, uint32_t device, bool systemReady, 5170 uint32_t bitRate) 5171 : DirectOutputThread(audioFlinger, output, id, device, OFFLOAD, systemReady, bitRate), 5172 mPausedWriteLength(0), mPausedBytesRemaining(0), mKeepWakeLock(true) 5173{ 5174 //FIXME: mStandby should be set to true by ThreadBase constructor 5175 mStandby = true; 5176 mKeepWakeLock = property_get_bool("ro.audio.offload_wakelock", true /* default_value */); 5177} 5178 5179void AudioFlinger::OffloadThread::threadLoop_exit() 5180{ 5181 if (mFlushPending || mHwPaused) { 5182 // If a flush is pending or track was paused, just discard buffered data 5183 flushHw_l(); 5184 } else { 5185 mMixerStatus = MIXER_DRAIN_ALL; 5186 threadLoop_drain(); 5187 } 5188 if (mUseAsyncWrite) { 5189 ALOG_ASSERT(mCallbackThread != 0); 5190 mCallbackThread->exit(); 5191 } 5192 PlaybackThread::threadLoop_exit(); 5193} 5194 5195AudioFlinger::PlaybackThread::mixer_state AudioFlinger::OffloadThread::prepareTracks_l( 5196 Vector< sp<Track> > *tracksToRemove 5197) 5198{ 5199 size_t count = mActiveTracks.size(); 5200 5201 mixer_state mixerStatus = MIXER_IDLE; 5202 bool doHwPause = false; 5203 bool doHwResume = false; 5204 5205 ALOGV("OffloadThread::prepareTracks_l active tracks %zu", count); 5206 5207 // find out which tracks need to be processed 5208 for (size_t i = 0; i < count; i++) { 5209 sp<Track> t = mActiveTracks[i].promote(); 5210 // The track died recently 5211 if (t == 0) { 5212 continue; 5213 } 5214 Track* const track = t.get(); 5215#ifdef VERY_VERY_VERBOSE_LOGGING 5216 audio_track_cblk_t* cblk = track->cblk(); 5217#endif 5218 // Only consider last track started for volume and mixer state control. 5219 // In theory an older track could underrun and restart after the new one starts 5220 // but as we only care about the transition phase between two tracks on a 5221 // direct output, it is not a problem to ignore the underrun case. 5222 sp<Track> l = mLatestActiveTrack.promote(); 5223 bool last = l.get() == track; 5224 5225 if (track->isInvalid()) { 5226 ALOGW("An invalidated track shouldn't be in active list"); 5227 tracksToRemove->add(track); 5228 continue; 5229 } 5230 5231 if (track->mState == TrackBase::IDLE) { 5232 ALOGW("An idle track shouldn't be in active list"); 5233 continue; 5234 } 5235 5236 if (track->isPausing()) { 5237 track->setPaused(); 5238 if (last) { 5239 if (mHwSupportsPause && !mHwPaused) { 5240 doHwPause = true; 5241 mHwPaused = true; 5242 } 5243 // If we were part way through writing the mixbuffer to 5244 // the HAL we must save this until we resume 5245 // BUG - this will be wrong if a different track is made active, 5246 // in that case we want to discard the pending data in the 5247 // mixbuffer and tell the client to present it again when the 5248 // track is resumed 5249 mPausedWriteLength = mCurrentWriteLength; 5250 mPausedBytesRemaining = mBytesRemaining; 5251 mBytesRemaining = 0; // stop writing 5252 } 5253 tracksToRemove->add(track); 5254 } else if (track->isFlushPending()) { 5255 track->mRetryCount = kMaxTrackRetriesOffload; 5256 track->flushAck(); 5257 if (last) { 5258 mFlushPending = true; 5259 } 5260 } else if (track->isResumePending()){ 5261 track->resumeAck(); 5262 if (last) { 5263 if (mPausedBytesRemaining) { 5264 // Need to continue write that was interrupted 5265 mCurrentWriteLength = mPausedWriteLength; 5266 mBytesRemaining = mPausedBytesRemaining; 5267 mPausedBytesRemaining = 0; 5268 } 5269 if (mHwPaused) { 5270 doHwResume = true; 5271 mHwPaused = false; 5272 // threadLoop_mix() will handle the case that we need to 5273 // resume an interrupted write 5274 } 5275 // enable write to audio HAL 5276 mSleepTimeUs = 0; 5277 5278 // Do not handle new data in this iteration even if track->framesReady() 5279 mixerStatus = MIXER_TRACKS_ENABLED; 5280 } 5281 } else if (track->framesReady() && track->isReady() && 5282 !track->isPaused() && !track->isTerminated() && !track->isStopping_2()) { 5283 ALOGVV("OffloadThread: track %d s=%08x [OK]", track->name(), cblk->mServer); 5284 if (track->mFillingUpStatus == Track::FS_FILLED) { 5285 track->mFillingUpStatus = Track::FS_ACTIVE; 5286 // make sure processVolume_l() will apply new volume even if 0 5287 mLeftVolFloat = mRightVolFloat = -1.0; 5288 } 5289 5290 if (last) { 5291 sp<Track> previousTrack = mPreviousTrack.promote(); 5292 if (previousTrack != 0) { 5293 if (track != previousTrack.get()) { 5294 // Flush any data still being written from last track 5295 mBytesRemaining = 0; 5296 if (mPausedBytesRemaining) { 5297 // Last track was paused so we also need to flush saved 5298 // mixbuffer state and invalidate track so that it will 5299 // re-submit that unwritten data when it is next resumed 5300 mPausedBytesRemaining = 0; 5301 // Invalidate is a bit drastic - would be more efficient 5302 // to have a flag to tell client that some of the 5303 // previously written data was lost 5304 previousTrack->invalidate(); 5305 } 5306 // flush data already sent to the DSP if changing audio session as audio 5307 // comes from a different source. Also invalidate previous track to force a 5308 // seek when resuming. 5309 if (previousTrack->sessionId() != track->sessionId()) { 5310 previousTrack->invalidate(); 5311 } 5312 } 5313 } 5314 mPreviousTrack = track; 5315 // reset retry count 5316 track->mRetryCount = kMaxTrackRetriesOffload; 5317 mActiveTrack = t; 5318 mixerStatus = MIXER_TRACKS_READY; 5319 } 5320 } else { 5321 ALOGVV("OffloadThread: track %d s=%08x [NOT READY]", track->name(), cblk->mServer); 5322 if (track->isStopping_1()) { 5323 // Hardware buffer can hold a large amount of audio so we must 5324 // wait for all current track's data to drain before we say 5325 // that the track is stopped. 5326 if (mBytesRemaining == 0) { 5327 // Only start draining when all data in mixbuffer 5328 // has been written 5329 ALOGV("OffloadThread: underrun and STOPPING_1 -> draining, STOPPING_2"); 5330 track->mState = TrackBase::STOPPING_2; // so presentation completes after drain 5331 // do not drain if no data was ever sent to HAL (mStandby == true) 5332 if (last && !mStandby) { 5333 // do not modify drain sequence if we are already draining. This happens 5334 // when resuming from pause after drain. 5335 if ((mDrainSequence & 1) == 0) { 5336 mSleepTimeUs = 0; 5337 mStandbyTimeNs = systemTime() + mStandbyDelayNs; 5338 mixerStatus = MIXER_DRAIN_TRACK; 5339 mDrainSequence += 2; 5340 } 5341 if (mHwPaused) { 5342 // It is possible to move from PAUSED to STOPPING_1 without 5343 // a resume so we must ensure hardware is running 5344 doHwResume = true; 5345 mHwPaused = false; 5346 } 5347 } 5348 } 5349 } else if (track->isStopping_2()) { 5350 // Drain has completed or we are in standby, signal presentation complete 5351 if (!(mDrainSequence & 1) || !last || mStandby) { 5352 track->mState = TrackBase::STOPPED; 5353 size_t audioHALFrames = 5354 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 5355 int64_t framesWritten = 5356 mBytesWritten / mOutput->getFrameSize(); 5357 track->presentationComplete(framesWritten, audioHALFrames); 5358 track->reset(); 5359 tracksToRemove->add(track); 5360 } 5361 } else { 5362 // No buffers for this track. Give it a few chances to 5363 // fill a buffer, then remove it from active list. 5364 if (--(track->mRetryCount) <= 0) { 5365 ALOGV("OffloadThread: BUFFER TIMEOUT: remove(%d) from active list", 5366 track->name()); 5367 tracksToRemove->add(track); 5368 // indicate to client process that the track was disabled because of underrun; 5369 // it will then automatically call start() when data is available 5370 track->disable(); 5371 } else if (last){ 5372 mixerStatus = MIXER_TRACKS_ENABLED; 5373 } 5374 } 5375 } 5376 // compute volume for this track 5377 processVolume_l(track, last); 5378 } 5379 5380 // make sure the pause/flush/resume sequence is executed in the right order. 5381 // If a flush is pending and a track is active but the HW is not paused, force a HW pause 5382 // before flush and then resume HW. This can happen in case of pause/flush/resume 5383 // if resume is received before pause is executed. 5384 if (!mStandby && (doHwPause || (mFlushPending && !mHwPaused && (count != 0)))) { 5385 mOutput->stream->pause(mOutput->stream); 5386 } 5387 if (mFlushPending) { 5388 flushHw_l(); 5389 } 5390 if (!mStandby && doHwResume) { 5391 mOutput->stream->resume(mOutput->stream); 5392 } 5393 5394 // remove all the tracks that need to be... 5395 removeTracks_l(*tracksToRemove); 5396 5397 return mixerStatus; 5398} 5399 5400// must be called with thread mutex locked 5401bool AudioFlinger::OffloadThread::waitingAsyncCallback_l() 5402{ 5403 ALOGVV("waitingAsyncCallback_l mWriteAckSequence %d mDrainSequence %d", 5404 mWriteAckSequence, mDrainSequence); 5405 if (mUseAsyncWrite && ((mWriteAckSequence & 1) || (mDrainSequence & 1))) { 5406 return true; 5407 } 5408 return false; 5409} 5410 5411bool AudioFlinger::OffloadThread::waitingAsyncCallback() 5412{ 5413 Mutex::Autolock _l(mLock); 5414 return waitingAsyncCallback_l(); 5415} 5416 5417void AudioFlinger::OffloadThread::flushHw_l() 5418{ 5419 DirectOutputThread::flushHw_l(); 5420 // Flush anything still waiting in the mixbuffer 5421 mCurrentWriteLength = 0; 5422 mBytesRemaining = 0; 5423 mPausedWriteLength = 0; 5424 mPausedBytesRemaining = 0; 5425 // reset bytes written count to reflect that DSP buffers are empty after flush. 5426 mBytesWritten = 0; 5427 5428 if (mUseAsyncWrite) { 5429 // discard any pending drain or write ack by incrementing sequence 5430 mWriteAckSequence = (mWriteAckSequence + 2) & ~1; 5431 mDrainSequence = (mDrainSequence + 2) & ~1; 5432 ALOG_ASSERT(mCallbackThread != 0); 5433 mCallbackThread->setWriteBlocked(mWriteAckSequence); 5434 mCallbackThread->setDraining(mDrainSequence); 5435 } 5436} 5437 5438uint32_t AudioFlinger::OffloadThread::activeSleepTimeUs() const 5439{ 5440 uint32_t time; 5441 if (audio_has_proportional_frames(mFormat)) { 5442 time = PlaybackThread::activeSleepTimeUs(); 5443 } else { 5444 // sleep time is half the duration of an audio HAL buffer. 5445 // Note: This can be problematic in case of underrun with variable bit rate and 5446 // current rate is much less than initial rate. 5447 time = (uint32_t)max(kDirectMinSleepTimeUs, mBufferDurationUs / 2); 5448 } 5449 return time; 5450} 5451 5452// ---------------------------------------------------------------------------- 5453 5454AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 5455 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id, bool systemReady) 5456 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->outDevice(), 5457 systemReady, DUPLICATING), 5458 mWaitTimeMs(UINT_MAX) 5459{ 5460 addOutputTrack(mainThread); 5461} 5462 5463AudioFlinger::DuplicatingThread::~DuplicatingThread() 5464{ 5465 for (size_t i = 0; i < mOutputTracks.size(); i++) { 5466 mOutputTracks[i]->destroy(); 5467 } 5468} 5469 5470void AudioFlinger::DuplicatingThread::threadLoop_mix() 5471{ 5472 // mix buffers... 5473 if (outputsReady(outputTracks)) { 5474 mAudioMixer->process(); 5475 } else { 5476 if (mMixerBufferValid) { 5477 memset(mMixerBuffer, 0, mMixerBufferSize); 5478 } else { 5479 memset(mSinkBuffer, 0, mSinkBufferSize); 5480 } 5481 } 5482 mSleepTimeUs = 0; 5483 writeFrames = mNormalFrameCount; 5484 mCurrentWriteLength = mSinkBufferSize; 5485 mStandbyTimeNs = systemTime() + mStandbyDelayNs; 5486} 5487 5488void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() 5489{ 5490 if (mSleepTimeUs == 0) { 5491 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 5492 mSleepTimeUs = mActiveSleepTimeUs; 5493 } else { 5494 mSleepTimeUs = mIdleSleepTimeUs; 5495 } 5496 } else if (mBytesWritten != 0) { 5497 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 5498 writeFrames = mNormalFrameCount; 5499 memset(mSinkBuffer, 0, mSinkBufferSize); 5500 } else { 5501 // flush remaining overflow buffers in output tracks 5502 writeFrames = 0; 5503 } 5504 mSleepTimeUs = 0; 5505 } 5506} 5507 5508ssize_t AudioFlinger::DuplicatingThread::threadLoop_write() 5509{ 5510 for (size_t i = 0; i < outputTracks.size(); i++) { 5511 outputTracks[i]->write(mSinkBuffer, writeFrames); 5512 } 5513 mStandby = false; 5514 return (ssize_t)mSinkBufferSize; 5515} 5516 5517void AudioFlinger::DuplicatingThread::threadLoop_standby() 5518{ 5519 // DuplicatingThread implements standby by stopping all tracks 5520 for (size_t i = 0; i < outputTracks.size(); i++) { 5521 outputTracks[i]->stop(); 5522 } 5523} 5524 5525void AudioFlinger::DuplicatingThread::saveOutputTracks() 5526{ 5527 outputTracks = mOutputTracks; 5528} 5529 5530void AudioFlinger::DuplicatingThread::clearOutputTracks() 5531{ 5532 outputTracks.clear(); 5533} 5534 5535void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 5536{ 5537 Mutex::Autolock _l(mLock); 5538 // The downstream MixerThread consumes thread->frameCount() amount of frames per mix pass. 5539 // Adjust for thread->sampleRate() to determine minimum buffer frame count. 5540 // Then triple buffer because Threads do not run synchronously and may not be clock locked. 5541 const size_t frameCount = 5542 3 * sourceFramesNeeded(mSampleRate, thread->frameCount(), thread->sampleRate()); 5543 // TODO: Consider asynchronous sample rate conversion to handle clock disparity 5544 // from different OutputTracks and their associated MixerThreads (e.g. one may 5545 // nearly empty and the other may be dropping data). 5546 5547 sp<OutputTrack> outputTrack = new OutputTrack(thread, 5548 this, 5549 mSampleRate, 5550 mFormat, 5551 mChannelMask, 5552 frameCount, 5553 IPCThreadState::self()->getCallingUid()); 5554 if (outputTrack->cblk() != NULL) { 5555 thread->setStreamVolume(AUDIO_STREAM_PATCH, 1.0f); 5556 mOutputTracks.add(outputTrack); 5557 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack.get(), thread); 5558 updateWaitTime_l(); 5559 } 5560} 5561 5562void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 5563{ 5564 Mutex::Autolock _l(mLock); 5565 for (size_t i = 0; i < mOutputTracks.size(); i++) { 5566 if (mOutputTracks[i]->thread() == thread) { 5567 mOutputTracks[i]->destroy(); 5568 mOutputTracks.removeAt(i); 5569 updateWaitTime_l(); 5570 if (thread->getOutput() == mOutput) { 5571 mOutput = NULL; 5572 } 5573 return; 5574 } 5575 } 5576 ALOGV("removeOutputTrack(): unknown thread: %p", thread); 5577} 5578 5579// caller must hold mLock 5580void AudioFlinger::DuplicatingThread::updateWaitTime_l() 5581{ 5582 mWaitTimeMs = UINT_MAX; 5583 for (size_t i = 0; i < mOutputTracks.size(); i++) { 5584 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 5585 if (strong != 0) { 5586 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 5587 if (waitTimeMs < mWaitTimeMs) { 5588 mWaitTimeMs = waitTimeMs; 5589 } 5590 } 5591 } 5592} 5593 5594 5595bool AudioFlinger::DuplicatingThread::outputsReady( 5596 const SortedVector< sp<OutputTrack> > &outputTracks) 5597{ 5598 for (size_t i = 0; i < outputTracks.size(); i++) { 5599 sp<ThreadBase> thread = outputTracks[i]->thread().promote(); 5600 if (thread == 0) { 5601 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", 5602 outputTracks[i].get()); 5603 return false; 5604 } 5605 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 5606 // see note at standby() declaration 5607 if (playbackThread->standby() && !playbackThread->isSuspended()) { 5608 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), 5609 thread.get()); 5610 return false; 5611 } 5612 } 5613 return true; 5614} 5615 5616uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() const 5617{ 5618 return (mWaitTimeMs * 1000) / 2; 5619} 5620 5621void AudioFlinger::DuplicatingThread::cacheParameters_l() 5622{ 5623 // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first 5624 updateWaitTime_l(); 5625 5626 MixerThread::cacheParameters_l(); 5627} 5628 5629// ---------------------------------------------------------------------------- 5630// Record 5631// ---------------------------------------------------------------------------- 5632 5633AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 5634 AudioStreamIn *input, 5635 audio_io_handle_t id, 5636 audio_devices_t outDevice, 5637 audio_devices_t inDevice, 5638 bool systemReady 5639#ifdef TEE_SINK 5640 , const sp<NBAIO_Sink>& teeSink 5641#endif 5642 ) : 5643 ThreadBase(audioFlinger, id, outDevice, inDevice, RECORD, systemReady), 5644 mInput(input), mActiveTracksGen(0), mRsmpInBuffer(NULL), 5645 // mRsmpInFrames and mRsmpInFramesP2 are set by readInputParameters_l() 5646 mRsmpInRear(0) 5647#ifdef TEE_SINK 5648 , mTeeSink(teeSink) 5649#endif 5650 , mReadOnlyHeap(new MemoryDealer(kRecordThreadReadOnlyHeapSize, 5651 "RecordThreadRO", MemoryHeapBase::READ_ONLY)) 5652 // mFastCapture below 5653 , mFastCaptureFutex(0) 5654 // mInputSource 5655 // mPipeSink 5656 // mPipeSource 5657 , mPipeFramesP2(0) 5658 // mPipeMemory 5659 // mFastCaptureNBLogWriter 5660 , mFastTrackAvail(false) 5661{ 5662 snprintf(mThreadName, kThreadNameLength, "AudioIn_%X", id); 5663 mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mThreadName); 5664 5665 readInputParameters_l(); 5666 5667 // create an NBAIO source for the HAL input stream, and negotiate 5668 mInputSource = new AudioStreamInSource(input->stream); 5669 size_t numCounterOffers = 0; 5670 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount, mFormat)}; 5671#if !LOG_NDEBUG 5672 ssize_t index = 5673#else 5674 (void) 5675#endif 5676 mInputSource->negotiate(offers, 1, NULL, numCounterOffers); 5677 ALOG_ASSERT(index == 0); 5678 5679 // initialize fast capture depending on configuration 5680 bool initFastCapture; 5681 switch (kUseFastCapture) { 5682 case FastCapture_Never: 5683 initFastCapture = false; 5684 break; 5685 case FastCapture_Always: 5686 initFastCapture = true; 5687 break; 5688 case FastCapture_Static: 5689 initFastCapture = (mFrameCount * 1000) / mSampleRate < kMinNormalCaptureBufferSizeMs; 5690 break; 5691 // case FastCapture_Dynamic: 5692 } 5693 5694 if (initFastCapture) { 5695 // create a Pipe for FastCapture to write to, and for us and fast tracks to read from 5696 NBAIO_Format format = mInputSource->format(); 5697 size_t pipeFramesP2 = roundup(mSampleRate / 25); // double-buffering of 20 ms each 5698 size_t pipeSize = pipeFramesP2 * Format_frameSize(format); 5699 void *pipeBuffer; 5700 const sp<MemoryDealer> roHeap(readOnlyHeap()); 5701 sp<IMemory> pipeMemory; 5702 if ((roHeap == 0) || 5703 (pipeMemory = roHeap->allocate(pipeSize)) == 0 || 5704 (pipeBuffer = pipeMemory->pointer()) == NULL) { 5705 ALOGE("not enough memory for pipe buffer size=%zu", pipeSize); 5706 goto failed; 5707 } 5708 // pipe will be shared directly with fast clients, so clear to avoid leaking old information 5709 memset(pipeBuffer, 0, pipeSize); 5710 Pipe *pipe = new Pipe(pipeFramesP2, format, pipeBuffer); 5711 const NBAIO_Format offers[1] = {format}; 5712 size_t numCounterOffers = 0; 5713 ssize_t index = pipe->negotiate(offers, 1, NULL, numCounterOffers); 5714 ALOG_ASSERT(index == 0); 5715 mPipeSink = pipe; 5716 PipeReader *pipeReader = new PipeReader(*pipe); 5717 numCounterOffers = 0; 5718 index = pipeReader->negotiate(offers, 1, NULL, numCounterOffers); 5719 ALOG_ASSERT(index == 0); 5720 mPipeSource = pipeReader; 5721 mPipeFramesP2 = pipeFramesP2; 5722 mPipeMemory = pipeMemory; 5723 5724 // create fast capture 5725 mFastCapture = new FastCapture(); 5726 FastCaptureStateQueue *sq = mFastCapture->sq(); 5727#ifdef STATE_QUEUE_DUMP 5728 // FIXME 5729#endif 5730 FastCaptureState *state = sq->begin(); 5731 state->mCblk = NULL; 5732 state->mInputSource = mInputSource.get(); 5733 state->mInputSourceGen++; 5734 state->mPipeSink = pipe; 5735 state->mPipeSinkGen++; 5736 state->mFrameCount = mFrameCount; 5737 state->mCommand = FastCaptureState::COLD_IDLE; 5738 // already done in constructor initialization list 5739 //mFastCaptureFutex = 0; 5740 state->mColdFutexAddr = &mFastCaptureFutex; 5741 state->mColdGen++; 5742 state->mDumpState = &mFastCaptureDumpState; 5743#ifdef TEE_SINK 5744 // FIXME 5745#endif 5746 mFastCaptureNBLogWriter = audioFlinger->newWriter_l(kFastCaptureLogSize, "FastCapture"); 5747 state->mNBLogWriter = mFastCaptureNBLogWriter.get(); 5748 sq->end(); 5749 sq->push(FastCaptureStateQueue::BLOCK_UNTIL_PUSHED); 5750 5751 // start the fast capture 5752 mFastCapture->run("FastCapture", ANDROID_PRIORITY_URGENT_AUDIO); 5753 pid_t tid = mFastCapture->getTid(); 5754 sendPrioConfigEvent(getpid_cached, tid, kPriorityFastCapture); 5755#ifdef AUDIO_WATCHDOG 5756 // FIXME 5757#endif 5758 5759 mFastTrackAvail = true; 5760 } 5761failed: ; 5762 5763 // FIXME mNormalSource 5764} 5765 5766AudioFlinger::RecordThread::~RecordThread() 5767{ 5768 if (mFastCapture != 0) { 5769 FastCaptureStateQueue *sq = mFastCapture->sq(); 5770 FastCaptureState *state = sq->begin(); 5771 if (state->mCommand == FastCaptureState::COLD_IDLE) { 5772 int32_t old = android_atomic_inc(&mFastCaptureFutex); 5773 if (old == -1) { 5774 (void) syscall(__NR_futex, &mFastCaptureFutex, FUTEX_WAKE_PRIVATE, 1); 5775 } 5776 } 5777 state->mCommand = FastCaptureState::EXIT; 5778 sq->end(); 5779 sq->push(FastCaptureStateQueue::BLOCK_UNTIL_PUSHED); 5780 mFastCapture->join(); 5781 mFastCapture.clear(); 5782 } 5783 mAudioFlinger->unregisterWriter(mFastCaptureNBLogWriter); 5784 mAudioFlinger->unregisterWriter(mNBLogWriter); 5785 free(mRsmpInBuffer); 5786} 5787 5788void AudioFlinger::RecordThread::onFirstRef() 5789{ 5790 run(mThreadName, PRIORITY_URGENT_AUDIO); 5791} 5792 5793bool AudioFlinger::RecordThread::threadLoop() 5794{ 5795 nsecs_t lastWarning = 0; 5796 5797 inputStandBy(); 5798 5799reacquire_wakelock: 5800 sp<RecordTrack> activeTrack; 5801 int activeTracksGen; 5802 { 5803 Mutex::Autolock _l(mLock); 5804 size_t size = mActiveTracks.size(); 5805 activeTracksGen = mActiveTracksGen; 5806 if (size > 0) { 5807 // FIXME an arbitrary choice 5808 activeTrack = mActiveTracks[0]; 5809 acquireWakeLock_l(activeTrack->uid()); 5810 if (size > 1) { 5811 SortedVector<int> tmp; 5812 for (size_t i = 0; i < size; i++) { 5813 tmp.add(mActiveTracks[i]->uid()); 5814 } 5815 updateWakeLockUids_l(tmp); 5816 } 5817 } else { 5818 acquireWakeLock_l(-1); 5819 } 5820 } 5821 5822 // used to request a deferred sleep, to be executed later while mutex is unlocked 5823 uint32_t sleepUs = 0; 5824 5825 // loop while there is work to do 5826 for (;;) { 5827 Vector< sp<EffectChain> > effectChains; 5828 5829 // sleep with mutex unlocked 5830 if (sleepUs > 0) { 5831 ATRACE_BEGIN("sleep"); 5832 usleep(sleepUs); 5833 ATRACE_END(); 5834 sleepUs = 0; 5835 } 5836 5837 // activeTracks accumulates a copy of a subset of mActiveTracks 5838 Vector< sp<RecordTrack> > activeTracks; 5839 5840 // reference to the (first and only) active fast track 5841 sp<RecordTrack> fastTrack; 5842 5843 // reference to a fast track which is about to be removed 5844 sp<RecordTrack> fastTrackToRemove; 5845 5846 { // scope for mLock 5847 Mutex::Autolock _l(mLock); 5848 5849 processConfigEvents_l(); 5850 5851 // check exitPending here because checkForNewParameters_l() and 5852 // checkForNewParameters_l() can temporarily release mLock 5853 if (exitPending()) { 5854 break; 5855 } 5856 5857 // if no active track(s), then standby and release wakelock 5858 size_t size = mActiveTracks.size(); 5859 if (size == 0) { 5860 standbyIfNotAlreadyInStandby(); 5861 // exitPending() can't become true here 5862 releaseWakeLock_l(); 5863 ALOGV("RecordThread: loop stopping"); 5864 // go to sleep 5865 mWaitWorkCV.wait(mLock); 5866 ALOGV("RecordThread: loop starting"); 5867 goto reacquire_wakelock; 5868 } 5869 5870 if (mActiveTracksGen != activeTracksGen) { 5871 activeTracksGen = mActiveTracksGen; 5872 SortedVector<int> tmp; 5873 for (size_t i = 0; i < size; i++) { 5874 tmp.add(mActiveTracks[i]->uid()); 5875 } 5876 updateWakeLockUids_l(tmp); 5877 } 5878 5879 bool doBroadcast = false; 5880 for (size_t i = 0; i < size; ) { 5881 5882 activeTrack = mActiveTracks[i]; 5883 if (activeTrack->isTerminated()) { 5884 if (activeTrack->isFastTrack()) { 5885 ALOG_ASSERT(fastTrackToRemove == 0); 5886 fastTrackToRemove = activeTrack; 5887 } 5888 removeTrack_l(activeTrack); 5889 mActiveTracks.remove(activeTrack); 5890 mActiveTracksGen++; 5891 size--; 5892 continue; 5893 } 5894 5895 TrackBase::track_state activeTrackState = activeTrack->mState; 5896 switch (activeTrackState) { 5897 5898 case TrackBase::PAUSING: 5899 mActiveTracks.remove(activeTrack); 5900 mActiveTracksGen++; 5901 doBroadcast = true; 5902 size--; 5903 continue; 5904 5905 case TrackBase::STARTING_1: 5906 sleepUs = 10000; 5907 i++; 5908 continue; 5909 5910 case TrackBase::STARTING_2: 5911 doBroadcast = true; 5912 mStandby = false; 5913 activeTrack->mState = TrackBase::ACTIVE; 5914 break; 5915 5916 case TrackBase::ACTIVE: 5917 break; 5918 5919 case TrackBase::IDLE: 5920 i++; 5921 continue; 5922 5923 default: 5924 LOG_ALWAYS_FATAL("Unexpected activeTrackState %d", activeTrackState); 5925 } 5926 5927 activeTracks.add(activeTrack); 5928 i++; 5929 5930 if (activeTrack->isFastTrack()) { 5931 ALOG_ASSERT(!mFastTrackAvail); 5932 ALOG_ASSERT(fastTrack == 0); 5933 fastTrack = activeTrack; 5934 } 5935 } 5936 if (doBroadcast) { 5937 mStartStopCond.broadcast(); 5938 } 5939 5940 // sleep if there are no active tracks to process 5941 if (activeTracks.size() == 0) { 5942 if (sleepUs == 0) { 5943 sleepUs = kRecordThreadSleepUs; 5944 } 5945 continue; 5946 } 5947 sleepUs = 0; 5948 5949 lockEffectChains_l(effectChains); 5950 } 5951 5952 // thread mutex is now unlocked, mActiveTracks unknown, activeTracks.size() > 0 5953 5954 size_t size = effectChains.size(); 5955 for (size_t i = 0; i < size; i++) { 5956 // thread mutex is not locked, but effect chain is locked 5957 effectChains[i]->process_l(); 5958 } 5959 5960 // Push a new fast capture state if fast capture is not already running, or cblk change 5961 if (mFastCapture != 0) { 5962 FastCaptureStateQueue *sq = mFastCapture->sq(); 5963 FastCaptureState *state = sq->begin(); 5964 bool didModify = false; 5965 FastCaptureStateQueue::block_t block = FastCaptureStateQueue::BLOCK_UNTIL_PUSHED; 5966 if (state->mCommand != FastCaptureState::READ_WRITE /* FIXME && 5967 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)*/) { 5968 if (state->mCommand == FastCaptureState::COLD_IDLE) { 5969 int32_t old = android_atomic_inc(&mFastCaptureFutex); 5970 if (old == -1) { 5971 (void) syscall(__NR_futex, &mFastCaptureFutex, FUTEX_WAKE_PRIVATE, 1); 5972 } 5973 } 5974 state->mCommand = FastCaptureState::READ_WRITE; 5975#if 0 // FIXME 5976 mFastCaptureDumpState.increaseSamplingN(mAudioFlinger->isLowRamDevice() ? 5977 FastThreadDumpState::kSamplingNforLowRamDevice : 5978 FastThreadDumpState::kSamplingN); 5979#endif 5980 didModify = true; 5981 } 5982 audio_track_cblk_t *cblkOld = state->mCblk; 5983 audio_track_cblk_t *cblkNew = fastTrack != 0 ? fastTrack->cblk() : NULL; 5984 if (cblkNew != cblkOld) { 5985 state->mCblk = cblkNew; 5986 // block until acked if removing a fast track 5987 if (cblkOld != NULL) { 5988 block = FastCaptureStateQueue::BLOCK_UNTIL_ACKED; 5989 } 5990 didModify = true; 5991 } 5992 sq->end(didModify); 5993 if (didModify) { 5994 sq->push(block); 5995#if 0 5996 if (kUseFastCapture == FastCapture_Dynamic) { 5997 mNormalSource = mPipeSource; 5998 } 5999#endif 6000 } 6001 } 6002 6003 // now run the fast track destructor with thread mutex unlocked 6004 fastTrackToRemove.clear(); 6005 6006 // Read from HAL to keep up with fastest client if multiple active tracks, not slowest one. 6007 // Only the client(s) that are too slow will overrun. But if even the fastest client is too 6008 // slow, then this RecordThread will overrun by not calling HAL read often enough. 6009 // If destination is non-contiguous, first read past the nominal end of buffer, then 6010 // copy to the right place. Permitted because mRsmpInBuffer was over-allocated. 6011 6012 int32_t rear = mRsmpInRear & (mRsmpInFramesP2 - 1); 6013 ssize_t framesRead; 6014 6015 // If an NBAIO source is present, use it to read the normal capture's data 6016 if (mPipeSource != 0) { 6017 size_t framesToRead = mBufferSize / mFrameSize; 6018 framesRead = mPipeSource->read((uint8_t*)mRsmpInBuffer + rear * mFrameSize, 6019 framesToRead); 6020 if (framesRead == 0) { 6021 // since pipe is non-blocking, simulate blocking input 6022 sleepUs = (framesToRead * 1000000LL) / mSampleRate; 6023 } 6024 // otherwise use the HAL / AudioStreamIn directly 6025 } else { 6026 ATRACE_BEGIN("read"); 6027 ssize_t bytesRead = mInput->stream->read(mInput->stream, 6028 (uint8_t*)mRsmpInBuffer + rear * mFrameSize, mBufferSize); 6029 ATRACE_END(); 6030 if (bytesRead < 0) { 6031 framesRead = bytesRead; 6032 } else { 6033 framesRead = bytesRead / mFrameSize; 6034 } 6035 } 6036 6037 // Update server timestamp with server stats 6038 // systemTime() is optional if the hardware supports timestamps. 6039 mTimestamp.mPosition[ExtendedTimestamp::LOCATION_SERVER] += framesRead; 6040 mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_SERVER] = systemTime(); 6041 6042 // Update server timestamp with kernel stats 6043 if (mInput->stream->get_capture_position != nullptr) { 6044 int64_t position, time; 6045 int ret = mInput->stream->get_capture_position(mInput->stream, &position, &time); 6046 if (ret == NO_ERROR) { 6047 mTimestamp.mPosition[ExtendedTimestamp::LOCATION_KERNEL] = position; 6048 mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL] = time; 6049 // Note: In general record buffers should tend to be empty in 6050 // a properly running pipeline. 6051 // 6052 // Also, it is not advantageous to call get_presentation_position during the read 6053 // as the read obtains a lock, preventing the timestamp call from executing. 6054 } 6055 } 6056 // Use this to track timestamp information 6057 // ALOGD("%s", mTimestamp.toString().c_str()); 6058 6059 if (framesRead < 0 || (framesRead == 0 && mPipeSource == 0)) { 6060 ALOGE("read failed: framesRead=%zd", framesRead); 6061 // Force input into standby so that it tries to recover at next read attempt 6062 inputStandBy(); 6063 sleepUs = kRecordThreadSleepUs; 6064 } 6065 if (framesRead <= 0) { 6066 goto unlock; 6067 } 6068 ALOG_ASSERT(framesRead > 0); 6069 6070 if (mTeeSink != 0) { 6071 (void) mTeeSink->write((uint8_t*)mRsmpInBuffer + rear * mFrameSize, framesRead); 6072 } 6073 // If destination is non-contiguous, we now correct for reading past end of buffer. 6074 { 6075 size_t part1 = mRsmpInFramesP2 - rear; 6076 if ((size_t) framesRead > part1) { 6077 memcpy(mRsmpInBuffer, (uint8_t*)mRsmpInBuffer + mRsmpInFramesP2 * mFrameSize, 6078 (framesRead - part1) * mFrameSize); 6079 } 6080 } 6081 rear = mRsmpInRear += framesRead; 6082 6083 size = activeTracks.size(); 6084 // loop over each active track 6085 for (size_t i = 0; i < size; i++) { 6086 activeTrack = activeTracks[i]; 6087 6088 // skip fast tracks, as those are handled directly by FastCapture 6089 if (activeTrack->isFastTrack()) { 6090 continue; 6091 } 6092 6093 // TODO: This code probably should be moved to RecordTrack. 6094 // TODO: Update the activeTrack buffer converter in case of reconfigure. 6095 6096 enum { 6097 OVERRUN_UNKNOWN, 6098 OVERRUN_TRUE, 6099 OVERRUN_FALSE 6100 } overrun = OVERRUN_UNKNOWN; 6101 6102 // loop over getNextBuffer to handle circular sink 6103 for (;;) { 6104 6105 activeTrack->mSink.frameCount = ~0; 6106 status_t status = activeTrack->getNextBuffer(&activeTrack->mSink); 6107 size_t framesOut = activeTrack->mSink.frameCount; 6108 LOG_ALWAYS_FATAL_IF((status == OK) != (framesOut > 0)); 6109 6110 // check available frames and handle overrun conditions 6111 // if the record track isn't draining fast enough. 6112 bool hasOverrun; 6113 size_t framesIn; 6114 activeTrack->mResamplerBufferProvider->sync(&framesIn, &hasOverrun); 6115 if (hasOverrun) { 6116 overrun = OVERRUN_TRUE; 6117 } 6118 if (framesOut == 0 || framesIn == 0) { 6119 break; 6120 } 6121 6122 // Don't allow framesOut to be larger than what is possible with resampling 6123 // from framesIn. 6124 // This isn't strictly necessary but helps limit buffer resizing in 6125 // RecordBufferConverter. TODO: remove when no longer needed. 6126 framesOut = min(framesOut, 6127 destinationFramesPossible( 6128 framesIn, mSampleRate, activeTrack->mSampleRate)); 6129 // process frames from the RecordThread buffer provider to the RecordTrack buffer 6130 framesOut = activeTrack->mRecordBufferConverter->convert( 6131 activeTrack->mSink.raw, activeTrack->mResamplerBufferProvider, framesOut); 6132 6133 if (framesOut > 0 && (overrun == OVERRUN_UNKNOWN)) { 6134 overrun = OVERRUN_FALSE; 6135 } 6136 6137 if (activeTrack->mFramesToDrop == 0) { 6138 if (framesOut > 0) { 6139 activeTrack->mSink.frameCount = framesOut; 6140 activeTrack->releaseBuffer(&activeTrack->mSink); 6141 } 6142 } else { 6143 // FIXME could do a partial drop of framesOut 6144 if (activeTrack->mFramesToDrop > 0) { 6145 activeTrack->mFramesToDrop -= framesOut; 6146 if (activeTrack->mFramesToDrop <= 0) { 6147 activeTrack->clearSyncStartEvent(); 6148 } 6149 } else { 6150 activeTrack->mFramesToDrop += framesOut; 6151 if (activeTrack->mFramesToDrop >= 0 || activeTrack->mSyncStartEvent == 0 || 6152 activeTrack->mSyncStartEvent->isCancelled()) { 6153 ALOGW("Synced record %s, session %d, trigger session %d", 6154 (activeTrack->mFramesToDrop >= 0) ? "timed out" : "cancelled", 6155 activeTrack->sessionId(), 6156 (activeTrack->mSyncStartEvent != 0) ? 6157 activeTrack->mSyncStartEvent->triggerSession() : 6158 AUDIO_SESSION_NONE); 6159 activeTrack->clearSyncStartEvent(); 6160 } 6161 } 6162 } 6163 6164 if (framesOut == 0) { 6165 break; 6166 } 6167 } 6168 6169 switch (overrun) { 6170 case OVERRUN_TRUE: 6171 // client isn't retrieving buffers fast enough 6172 if (!activeTrack->setOverflow()) { 6173 nsecs_t now = systemTime(); 6174 // FIXME should lastWarning per track? 6175 if ((now - lastWarning) > kWarningThrottleNs) { 6176 ALOGW("RecordThread: buffer overflow"); 6177 lastWarning = now; 6178 } 6179 } 6180 break; 6181 case OVERRUN_FALSE: 6182 activeTrack->clearOverflow(); 6183 break; 6184 case OVERRUN_UNKNOWN: 6185 break; 6186 } 6187 6188 // update frame information and push timestamp out 6189 activeTrack->updateTrackFrameInfo( 6190 activeTrack->mServerProxy->framesReleased(), 6191 mTimestamp.mPosition[ExtendedTimestamp::LOCATION_SERVER], 6192 mSampleRate, mTimestamp); 6193 } 6194 6195unlock: 6196 // enable changes in effect chain 6197 unlockEffectChains(effectChains); 6198 // effectChains doesn't need to be cleared, since it is cleared by destructor at scope end 6199 } 6200 6201 standbyIfNotAlreadyInStandby(); 6202 6203 { 6204 Mutex::Autolock _l(mLock); 6205 for (size_t i = 0; i < mTracks.size(); i++) { 6206 sp<RecordTrack> track = mTracks[i]; 6207 track->invalidate(); 6208 } 6209 mActiveTracks.clear(); 6210 mActiveTracksGen++; 6211 mStartStopCond.broadcast(); 6212 } 6213 6214 releaseWakeLock(); 6215 6216 ALOGV("RecordThread %p exiting", this); 6217 return false; 6218} 6219 6220void AudioFlinger::RecordThread::standbyIfNotAlreadyInStandby() 6221{ 6222 if (!mStandby) { 6223 inputStandBy(); 6224 mStandby = true; 6225 } 6226} 6227 6228void AudioFlinger::RecordThread::inputStandBy() 6229{ 6230 // Idle the fast capture if it's currently running 6231 if (mFastCapture != 0) { 6232 FastCaptureStateQueue *sq = mFastCapture->sq(); 6233 FastCaptureState *state = sq->begin(); 6234 if (!(state->mCommand & FastCaptureState::IDLE)) { 6235 state->mCommand = FastCaptureState::COLD_IDLE; 6236 state->mColdFutexAddr = &mFastCaptureFutex; 6237 state->mColdGen++; 6238 mFastCaptureFutex = 0; 6239 sq->end(); 6240 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 6241 sq->push(FastCaptureStateQueue::BLOCK_UNTIL_ACKED); 6242#if 0 6243 if (kUseFastCapture == FastCapture_Dynamic) { 6244 // FIXME 6245 } 6246#endif 6247#ifdef AUDIO_WATCHDOG 6248 // FIXME 6249#endif 6250 } else { 6251 sq->end(false /*didModify*/); 6252 } 6253 } 6254 mInput->stream->common.standby(&mInput->stream->common); 6255} 6256 6257// RecordThread::createRecordTrack_l() must be called with AudioFlinger::mLock held 6258sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 6259 const sp<AudioFlinger::Client>& client, 6260 uint32_t sampleRate, 6261 audio_format_t format, 6262 audio_channel_mask_t channelMask, 6263 size_t *pFrameCount, 6264 audio_session_t sessionId, 6265 size_t *notificationFrames, 6266 int uid, 6267 IAudioFlinger::track_flags_t *flags, 6268 pid_t tid, 6269 status_t *status) 6270{ 6271 size_t frameCount = *pFrameCount; 6272 sp<RecordTrack> track; 6273 status_t lStatus; 6274 6275 // client expresses a preference for FAST, but we get the final say 6276 if (*flags & IAudioFlinger::TRACK_FAST) { 6277 if ( 6278 // we formerly checked for a callback handler (non-0 tid), 6279 // but that is no longer required for TRANSFER_OBTAIN mode 6280 // 6281 // frame count is not specified, or is exactly the pipe depth 6282 ((frameCount == 0) || (frameCount == mPipeFramesP2)) && 6283 // PCM data 6284 audio_is_linear_pcm(format) && 6285 // hardware format 6286 (format == mFormat) && 6287 // hardware channel mask 6288 (channelMask == mChannelMask) && 6289 // hardware sample rate 6290 (sampleRate == mSampleRate) && 6291 // record thread has an associated fast capture 6292 hasFastCapture() && 6293 // there are sufficient fast track slots available 6294 mFastTrackAvail 6295 ) { 6296 ALOGV("AUDIO_INPUT_FLAG_FAST accepted: frameCount=%zu mFrameCount=%zu", 6297 frameCount, mFrameCount); 6298 } else { 6299 ALOGV("AUDIO_INPUT_FLAG_FAST denied: frameCount=%zu mFrameCount=%zu mPipeFramesP2=%zu " 6300 "format=%#x isLinear=%d channelMask=%#x sampleRate=%u mSampleRate=%u " 6301 "hasFastCapture=%d tid=%d mFastTrackAvail=%d", 6302 frameCount, mFrameCount, mPipeFramesP2, 6303 format, audio_is_linear_pcm(format), channelMask, sampleRate, mSampleRate, 6304 hasFastCapture(), tid, mFastTrackAvail); 6305 *flags &= ~IAudioFlinger::TRACK_FAST; 6306 } 6307 } 6308 6309 // compute track buffer size in frames, and suggest the notification frame count 6310 if (*flags & IAudioFlinger::TRACK_FAST) { 6311 // fast track: frame count is exactly the pipe depth 6312 frameCount = mPipeFramesP2; 6313 // ignore requested notificationFrames, and always notify exactly once every HAL buffer 6314 *notificationFrames = mFrameCount; 6315 } else { 6316 // not fast track: max notification period is resampled equivalent of one HAL buffer time 6317 // or 20 ms if there is a fast capture 6318 // TODO This could be a roundupRatio inline, and const 6319 size_t maxNotificationFrames = ((int64_t) (hasFastCapture() ? mSampleRate/50 : mFrameCount) 6320 * sampleRate + mSampleRate - 1) / mSampleRate; 6321 // minimum number of notification periods is at least kMinNotifications, 6322 // and at least kMinMs rounded up to a whole notification period (minNotificationsByMs) 6323 static const size_t kMinNotifications = 3; 6324 static const uint32_t kMinMs = 30; 6325 // TODO This could be a roundupRatio inline 6326 const size_t minFramesByMs = (sampleRate * kMinMs + 1000 - 1) / 1000; 6327 // TODO This could be a roundupRatio inline 6328 const size_t minNotificationsByMs = (minFramesByMs + maxNotificationFrames - 1) / 6329 maxNotificationFrames; 6330 const size_t minFrameCount = maxNotificationFrames * 6331 max(kMinNotifications, minNotificationsByMs); 6332 frameCount = max(frameCount, minFrameCount); 6333 if (*notificationFrames == 0 || *notificationFrames > maxNotificationFrames) { 6334 *notificationFrames = maxNotificationFrames; 6335 } 6336 } 6337 *pFrameCount = frameCount; 6338 6339 lStatus = initCheck(); 6340 if (lStatus != NO_ERROR) { 6341 ALOGE("createRecordTrack_l() audio driver not initialized"); 6342 goto Exit; 6343 } 6344 6345 { // scope for mLock 6346 Mutex::Autolock _l(mLock); 6347 6348 track = new RecordTrack(this, client, sampleRate, 6349 format, channelMask, frameCount, NULL, sessionId, uid, 6350 *flags, TrackBase::TYPE_DEFAULT); 6351 6352 lStatus = track->initCheck(); 6353 if (lStatus != NO_ERROR) { 6354 ALOGE("createRecordTrack_l() initCheck failed %d; no control block?", lStatus); 6355 // track must be cleared from the caller as the caller has the AF lock 6356 goto Exit; 6357 } 6358 mTracks.add(track); 6359 6360 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 6361 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 6362 mAudioFlinger->btNrecIsOff(); 6363 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 6364 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 6365 6366 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 6367 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 6368 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 6369 // so ask activity manager to do this on our behalf 6370 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 6371 } 6372 } 6373 6374 lStatus = NO_ERROR; 6375 6376Exit: 6377 *status = lStatus; 6378 return track; 6379} 6380 6381status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, 6382 AudioSystem::sync_event_t event, 6383 audio_session_t triggerSession) 6384{ 6385 ALOGV("RecordThread::start event %d, triggerSession %d", event, triggerSession); 6386 sp<ThreadBase> strongMe = this; 6387 status_t status = NO_ERROR; 6388 6389 if (event == AudioSystem::SYNC_EVENT_NONE) { 6390 recordTrack->clearSyncStartEvent(); 6391 } else if (event != AudioSystem::SYNC_EVENT_SAME) { 6392 recordTrack->mSyncStartEvent = mAudioFlinger->createSyncEvent(event, 6393 triggerSession, 6394 recordTrack->sessionId(), 6395 syncStartEventCallback, 6396 recordTrack); 6397 // Sync event can be cancelled by the trigger session if the track is not in a 6398 // compatible state in which case we start record immediately 6399 if (recordTrack->mSyncStartEvent->isCancelled()) { 6400 recordTrack->clearSyncStartEvent(); 6401 } else { 6402 // do not wait for the event for more than AudioSystem::kSyncRecordStartTimeOutMs 6403 recordTrack->mFramesToDrop = - 6404 ((AudioSystem::kSyncRecordStartTimeOutMs * recordTrack->mSampleRate) / 1000); 6405 } 6406 } 6407 6408 { 6409 // This section is a rendezvous between binder thread executing start() and RecordThread 6410 AutoMutex lock(mLock); 6411 if (mActiveTracks.indexOf(recordTrack) >= 0) { 6412 if (recordTrack->mState == TrackBase::PAUSING) { 6413 ALOGV("active record track PAUSING -> ACTIVE"); 6414 recordTrack->mState = TrackBase::ACTIVE; 6415 } else { 6416 ALOGV("active record track state %d", recordTrack->mState); 6417 } 6418 return status; 6419 } 6420 6421 // TODO consider other ways of handling this, such as changing the state to :STARTING and 6422 // adding the track to mActiveTracks after returning from AudioSystem::startInput(), 6423 // or using a separate command thread 6424 recordTrack->mState = TrackBase::STARTING_1; 6425 mActiveTracks.add(recordTrack); 6426 mActiveTracksGen++; 6427 status_t status = NO_ERROR; 6428 if (recordTrack->isExternalTrack()) { 6429 mLock.unlock(); 6430 status = AudioSystem::startInput(mId, recordTrack->sessionId()); 6431 mLock.lock(); 6432 // FIXME should verify that recordTrack is still in mActiveTracks 6433 if (status != NO_ERROR) { 6434 mActiveTracks.remove(recordTrack); 6435 mActiveTracksGen++; 6436 recordTrack->clearSyncStartEvent(); 6437 ALOGV("RecordThread::start error %d", status); 6438 return status; 6439 } 6440 } 6441 // Catch up with current buffer indices if thread is already running. 6442 // This is what makes a new client discard all buffered data. If the track's mRsmpInFront 6443 // was initialized to some value closer to the thread's mRsmpInFront, then the track could 6444 // see previously buffered data before it called start(), but with greater risk of overrun. 6445 6446 recordTrack->mResamplerBufferProvider->reset(); 6447 // clear any converter state as new data will be discontinuous 6448 recordTrack->mRecordBufferConverter->reset(); 6449 recordTrack->mState = TrackBase::STARTING_2; 6450 // signal thread to start 6451 mWaitWorkCV.broadcast(); 6452 if (mActiveTracks.indexOf(recordTrack) < 0) { 6453 ALOGV("Record failed to start"); 6454 status = BAD_VALUE; 6455 goto startError; 6456 } 6457 return status; 6458 } 6459 6460startError: 6461 if (recordTrack->isExternalTrack()) { 6462 AudioSystem::stopInput(mId, recordTrack->sessionId()); 6463 } 6464 recordTrack->clearSyncStartEvent(); 6465 // FIXME I wonder why we do not reset the state here? 6466 return status; 6467} 6468 6469void AudioFlinger::RecordThread::syncStartEventCallback(const wp<SyncEvent>& event) 6470{ 6471 sp<SyncEvent> strongEvent = event.promote(); 6472 6473 if (strongEvent != 0) { 6474 sp<RefBase> ptr = strongEvent->cookie().promote(); 6475 if (ptr != 0) { 6476 RecordTrack *recordTrack = (RecordTrack *)ptr.get(); 6477 recordTrack->handleSyncStartEvent(strongEvent); 6478 } 6479 } 6480} 6481 6482bool AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 6483 ALOGV("RecordThread::stop"); 6484 AutoMutex _l(mLock); 6485 if (mActiveTracks.indexOf(recordTrack) != 0 || recordTrack->mState == TrackBase::PAUSING) { 6486 return false; 6487 } 6488 // note that threadLoop may still be processing the track at this point [without lock] 6489 recordTrack->mState = TrackBase::PAUSING; 6490 // do not wait for mStartStopCond if exiting 6491 if (exitPending()) { 6492 return true; 6493 } 6494 // FIXME incorrect usage of wait: no explicit predicate or loop 6495 mStartStopCond.wait(mLock); 6496 // if we have been restarted, recordTrack is in mActiveTracks here 6497 if (exitPending() || mActiveTracks.indexOf(recordTrack) != 0) { 6498 ALOGV("Record stopped OK"); 6499 return true; 6500 } 6501 return false; 6502} 6503 6504bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event __unused) const 6505{ 6506 return false; 6507} 6508 6509status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event __unused) 6510{ 6511#if 0 // This branch is currently dead code, but is preserved in case it will be needed in future 6512 if (!isValidSyncEvent(event)) { 6513 return BAD_VALUE; 6514 } 6515 6516 audio_session_t eventSession = event->triggerSession(); 6517 status_t ret = NAME_NOT_FOUND; 6518 6519 Mutex::Autolock _l(mLock); 6520 6521 for (size_t i = 0; i < mTracks.size(); i++) { 6522 sp<RecordTrack> track = mTracks[i]; 6523 if (eventSession == track->sessionId()) { 6524 (void) track->setSyncEvent(event); 6525 ret = NO_ERROR; 6526 } 6527 } 6528 return ret; 6529#else 6530 return BAD_VALUE; 6531#endif 6532} 6533 6534// destroyTrack_l() must be called with ThreadBase::mLock held 6535void AudioFlinger::RecordThread::destroyTrack_l(const sp<RecordTrack>& track) 6536{ 6537 track->terminate(); 6538 track->mState = TrackBase::STOPPED; 6539 // active tracks are removed by threadLoop() 6540 if (mActiveTracks.indexOf(track) < 0) { 6541 removeTrack_l(track); 6542 } 6543} 6544 6545void AudioFlinger::RecordThread::removeTrack_l(const sp<RecordTrack>& track) 6546{ 6547 mTracks.remove(track); 6548 // need anything related to effects here? 6549 if (track->isFastTrack()) { 6550 ALOG_ASSERT(!mFastTrackAvail); 6551 mFastTrackAvail = true; 6552 } 6553} 6554 6555void AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 6556{ 6557 dumpInternals(fd, args); 6558 dumpTracks(fd, args); 6559 dumpEffectChains(fd, args); 6560} 6561 6562void AudioFlinger::RecordThread::dumpInternals(int fd, const Vector<String16>& args) 6563{ 6564 dprintf(fd, "\nInput thread %p:\n", this); 6565 6566 dumpBase(fd, args); 6567 6568 if (mActiveTracks.size() == 0) { 6569 dprintf(fd, " No active record clients\n"); 6570 } 6571 dprintf(fd, " Fast capture thread: %s\n", hasFastCapture() ? "yes" : "no"); 6572 dprintf(fd, " Fast track available: %s\n", mFastTrackAvail ? "yes" : "no"); 6573 6574 // Make a non-atomic copy of fast capture dump state so it won't change underneath us 6575 // while we are dumping it. It may be inconsistent, but it won't mutate! 6576 // This is a large object so we place it on the heap. 6577 // FIXME 25972958: Need an intelligent copy constructor that does not touch unused pages. 6578 const FastCaptureDumpState *copy = new FastCaptureDumpState(mFastCaptureDumpState); 6579 copy->dump(fd); 6580 delete copy; 6581} 6582 6583void AudioFlinger::RecordThread::dumpTracks(int fd, const Vector<String16>& args __unused) 6584{ 6585 const size_t SIZE = 256; 6586 char buffer[SIZE]; 6587 String8 result; 6588 6589 size_t numtracks = mTracks.size(); 6590 size_t numactive = mActiveTracks.size(); 6591 size_t numactiveseen = 0; 6592 dprintf(fd, " %zu Tracks", numtracks); 6593 if (numtracks) { 6594 dprintf(fd, " of which %zu are active\n", numactive); 6595 RecordTrack::appendDumpHeader(result); 6596 for (size_t i = 0; i < numtracks ; ++i) { 6597 sp<RecordTrack> track = mTracks[i]; 6598 if (track != 0) { 6599 bool active = mActiveTracks.indexOf(track) >= 0; 6600 if (active) { 6601 numactiveseen++; 6602 } 6603 track->dump(buffer, SIZE, active); 6604 result.append(buffer); 6605 } 6606 } 6607 } else { 6608 dprintf(fd, "\n"); 6609 } 6610 6611 if (numactiveseen != numactive) { 6612 snprintf(buffer, SIZE, " The following tracks are in the active list but" 6613 " not in the track list\n"); 6614 result.append(buffer); 6615 RecordTrack::appendDumpHeader(result); 6616 for (size_t i = 0; i < numactive; ++i) { 6617 sp<RecordTrack> track = mActiveTracks[i]; 6618 if (mTracks.indexOf(track) < 0) { 6619 track->dump(buffer, SIZE, true); 6620 result.append(buffer); 6621 } 6622 } 6623 6624 } 6625 write(fd, result.string(), result.size()); 6626} 6627 6628 6629void AudioFlinger::RecordThread::ResamplerBufferProvider::reset() 6630{ 6631 sp<ThreadBase> threadBase = mRecordTrack->mThread.promote(); 6632 RecordThread *recordThread = (RecordThread *) threadBase.get(); 6633 mRsmpInFront = recordThread->mRsmpInRear; 6634 mRsmpInUnrel = 0; 6635} 6636 6637void AudioFlinger::RecordThread::ResamplerBufferProvider::sync( 6638 size_t *framesAvailable, bool *hasOverrun) 6639{ 6640 sp<ThreadBase> threadBase = mRecordTrack->mThread.promote(); 6641 RecordThread *recordThread = (RecordThread *) threadBase.get(); 6642 const int32_t rear = recordThread->mRsmpInRear; 6643 const int32_t front = mRsmpInFront; 6644 const ssize_t filled = rear - front; 6645 6646 size_t framesIn; 6647 bool overrun = false; 6648 if (filled < 0) { 6649 // should not happen, but treat like a massive overrun and re-sync 6650 framesIn = 0; 6651 mRsmpInFront = rear; 6652 overrun = true; 6653 } else if ((size_t) filled <= recordThread->mRsmpInFrames) { 6654 framesIn = (size_t) filled; 6655 } else { 6656 // client is not keeping up with server, but give it latest data 6657 framesIn = recordThread->mRsmpInFrames; 6658 mRsmpInFront = /* front = */ rear - framesIn; 6659 overrun = true; 6660 } 6661 if (framesAvailable != NULL) { 6662 *framesAvailable = framesIn; 6663 } 6664 if (hasOverrun != NULL) { 6665 *hasOverrun = overrun; 6666 } 6667} 6668 6669// AudioBufferProvider interface 6670status_t AudioFlinger::RecordThread::ResamplerBufferProvider::getNextBuffer( 6671 AudioBufferProvider::Buffer* buffer) 6672{ 6673 sp<ThreadBase> threadBase = mRecordTrack->mThread.promote(); 6674 if (threadBase == 0) { 6675 buffer->frameCount = 0; 6676 buffer->raw = NULL; 6677 return NOT_ENOUGH_DATA; 6678 } 6679 RecordThread *recordThread = (RecordThread *) threadBase.get(); 6680 int32_t rear = recordThread->mRsmpInRear; 6681 int32_t front = mRsmpInFront; 6682 ssize_t filled = rear - front; 6683 // FIXME should not be P2 (don't want to increase latency) 6684 // FIXME if client not keeping up, discard 6685 LOG_ALWAYS_FATAL_IF(!(0 <= filled && (size_t) filled <= recordThread->mRsmpInFrames)); 6686 // 'filled' may be non-contiguous, so return only the first contiguous chunk 6687 front &= recordThread->mRsmpInFramesP2 - 1; 6688 size_t part1 = recordThread->mRsmpInFramesP2 - front; 6689 if (part1 > (size_t) filled) { 6690 part1 = filled; 6691 } 6692 size_t ask = buffer->frameCount; 6693 ALOG_ASSERT(ask > 0); 6694 if (part1 > ask) { 6695 part1 = ask; 6696 } 6697 if (part1 == 0) { 6698 // out of data is fine since the resampler will return a short-count. 6699 buffer->raw = NULL; 6700 buffer->frameCount = 0; 6701 mRsmpInUnrel = 0; 6702 return NOT_ENOUGH_DATA; 6703 } 6704 6705 buffer->raw = (uint8_t*)recordThread->mRsmpInBuffer + front * recordThread->mFrameSize; 6706 buffer->frameCount = part1; 6707 mRsmpInUnrel = part1; 6708 return NO_ERROR; 6709} 6710 6711// AudioBufferProvider interface 6712void AudioFlinger::RecordThread::ResamplerBufferProvider::releaseBuffer( 6713 AudioBufferProvider::Buffer* buffer) 6714{ 6715 size_t stepCount = buffer->frameCount; 6716 if (stepCount == 0) { 6717 return; 6718 } 6719 ALOG_ASSERT(stepCount <= mRsmpInUnrel); 6720 mRsmpInUnrel -= stepCount; 6721 mRsmpInFront += stepCount; 6722 buffer->raw = NULL; 6723 buffer->frameCount = 0; 6724} 6725 6726AudioFlinger::RecordThread::RecordBufferConverter::RecordBufferConverter( 6727 audio_channel_mask_t srcChannelMask, audio_format_t srcFormat, 6728 uint32_t srcSampleRate, 6729 audio_channel_mask_t dstChannelMask, audio_format_t dstFormat, 6730 uint32_t dstSampleRate) : 6731 mSrcChannelMask(AUDIO_CHANNEL_INVALID), // updateParameters will set following vars 6732 // mSrcFormat 6733 // mSrcSampleRate 6734 // mDstChannelMask 6735 // mDstFormat 6736 // mDstSampleRate 6737 // mSrcChannelCount 6738 // mDstChannelCount 6739 // mDstFrameSize 6740 mBuf(NULL), mBufFrames(0), mBufFrameSize(0), 6741 mResampler(NULL), 6742 mIsLegacyDownmix(false), 6743 mIsLegacyUpmix(false), 6744 mRequiresFloat(false), 6745 mInputConverterProvider(NULL) 6746{ 6747 (void)updateParameters(srcChannelMask, srcFormat, srcSampleRate, 6748 dstChannelMask, dstFormat, dstSampleRate); 6749} 6750 6751AudioFlinger::RecordThread::RecordBufferConverter::~RecordBufferConverter() { 6752 free(mBuf); 6753 delete mResampler; 6754 delete mInputConverterProvider; 6755} 6756 6757size_t AudioFlinger::RecordThread::RecordBufferConverter::convert(void *dst, 6758 AudioBufferProvider *provider, size_t frames) 6759{ 6760 if (mInputConverterProvider != NULL) { 6761 mInputConverterProvider->setBufferProvider(provider); 6762 provider = mInputConverterProvider; 6763 } 6764 6765 if (mResampler == NULL) { 6766 ALOGVV("NO RESAMPLING sampleRate:%u mSrcFormat:%#x mDstFormat:%#x", 6767 mSrcSampleRate, mSrcFormat, mDstFormat); 6768 6769 AudioBufferProvider::Buffer buffer; 6770 for (size_t i = frames; i > 0; ) { 6771 buffer.frameCount = i; 6772 status_t status = provider->getNextBuffer(&buffer); 6773 if (status != OK || buffer.frameCount == 0) { 6774 frames -= i; // cannot fill request. 6775 break; 6776 } 6777 // format convert to destination buffer 6778 convertNoResampler(dst, buffer.raw, buffer.frameCount); 6779 6780 dst = (int8_t*)dst + buffer.frameCount * mDstFrameSize; 6781 i -= buffer.frameCount; 6782 provider->releaseBuffer(&buffer); 6783 } 6784 } else { 6785 ALOGVV("RESAMPLING mSrcSampleRate:%u mDstSampleRate:%u mSrcFormat:%#x mDstFormat:%#x", 6786 mSrcSampleRate, mDstSampleRate, mSrcFormat, mDstFormat); 6787 6788 // reallocate buffer if needed 6789 if (mBufFrameSize != 0 && mBufFrames < frames) { 6790 free(mBuf); 6791 mBufFrames = frames; 6792 (void)posix_memalign(&mBuf, 32, mBufFrames * mBufFrameSize); 6793 } 6794 // resampler accumulates, but we only have one source track 6795 memset(mBuf, 0, frames * mBufFrameSize); 6796 frames = mResampler->resample((int32_t*)mBuf, frames, provider); 6797 // format convert to destination buffer 6798 convertResampler(dst, mBuf, frames); 6799 } 6800 return frames; 6801} 6802 6803status_t AudioFlinger::RecordThread::RecordBufferConverter::updateParameters( 6804 audio_channel_mask_t srcChannelMask, audio_format_t srcFormat, 6805 uint32_t srcSampleRate, 6806 audio_channel_mask_t dstChannelMask, audio_format_t dstFormat, 6807 uint32_t dstSampleRate) 6808{ 6809 // quick evaluation if there is any change. 6810 if (mSrcFormat == srcFormat 6811 && mSrcChannelMask == srcChannelMask 6812 && mSrcSampleRate == srcSampleRate 6813 && mDstFormat == dstFormat 6814 && mDstChannelMask == dstChannelMask 6815 && mDstSampleRate == dstSampleRate) { 6816 return NO_ERROR; 6817 } 6818 6819 ALOGV("RecordBufferConverter updateParameters srcMask:%#x dstMask:%#x" 6820 " srcFormat:%#x dstFormat:%#x srcRate:%u dstRate:%u", 6821 srcChannelMask, dstChannelMask, srcFormat, dstFormat, srcSampleRate, dstSampleRate); 6822 const bool valid = 6823 audio_is_input_channel(srcChannelMask) 6824 && audio_is_input_channel(dstChannelMask) 6825 && audio_is_valid_format(srcFormat) && audio_is_linear_pcm(srcFormat) 6826 && audio_is_valid_format(dstFormat) && audio_is_linear_pcm(dstFormat) 6827 && (srcSampleRate <= dstSampleRate * AUDIO_RESAMPLER_DOWN_RATIO_MAX) 6828 ; // no upsampling checks for now 6829 if (!valid) { 6830 return BAD_VALUE; 6831 } 6832 6833 mSrcFormat = srcFormat; 6834 mSrcChannelMask = srcChannelMask; 6835 mSrcSampleRate = srcSampleRate; 6836 mDstFormat = dstFormat; 6837 mDstChannelMask = dstChannelMask; 6838 mDstSampleRate = dstSampleRate; 6839 6840 // compute derived parameters 6841 mSrcChannelCount = audio_channel_count_from_in_mask(srcChannelMask); 6842 mDstChannelCount = audio_channel_count_from_in_mask(dstChannelMask); 6843 mDstFrameSize = mDstChannelCount * audio_bytes_per_sample(mDstFormat); 6844 6845 // do we need to resample? 6846 delete mResampler; 6847 mResampler = NULL; 6848 if (mSrcSampleRate != mDstSampleRate) { 6849 mResampler = AudioResampler::create(AUDIO_FORMAT_PCM_FLOAT, 6850 mSrcChannelCount, mDstSampleRate); 6851 mResampler->setSampleRate(mSrcSampleRate); 6852 mResampler->setVolume(AudioMixer::UNITY_GAIN_FLOAT, AudioMixer::UNITY_GAIN_FLOAT); 6853 } 6854 6855 // are we running legacy channel conversion modes? 6856 mIsLegacyDownmix = (mSrcChannelMask == AUDIO_CHANNEL_IN_STEREO 6857 || mSrcChannelMask == AUDIO_CHANNEL_IN_FRONT_BACK) 6858 && mDstChannelMask == AUDIO_CHANNEL_IN_MONO; 6859 mIsLegacyUpmix = mSrcChannelMask == AUDIO_CHANNEL_IN_MONO 6860 && (mDstChannelMask == AUDIO_CHANNEL_IN_STEREO 6861 || mDstChannelMask == AUDIO_CHANNEL_IN_FRONT_BACK); 6862 6863 // do we need to process in float? 6864 mRequiresFloat = mResampler != NULL || mIsLegacyDownmix || mIsLegacyUpmix; 6865 6866 // do we need a staging buffer to convert for destination (we can still optimize this)? 6867 // we use mBufFrameSize > 0 to indicate both frame size as well as buffer necessity 6868 if (mResampler != NULL) { 6869 mBufFrameSize = max(mSrcChannelCount, FCC_2) 6870 * audio_bytes_per_sample(AUDIO_FORMAT_PCM_FLOAT); 6871 } else if (mIsLegacyUpmix || mIsLegacyDownmix) { // legacy modes always float 6872 mBufFrameSize = mDstChannelCount * audio_bytes_per_sample(AUDIO_FORMAT_PCM_FLOAT); 6873 } else if (mSrcChannelMask != mDstChannelMask && mDstFormat != mSrcFormat) { 6874 mBufFrameSize = mDstChannelCount * audio_bytes_per_sample(mSrcFormat); 6875 } else { 6876 mBufFrameSize = 0; 6877 } 6878 mBufFrames = 0; // force the buffer to be resized. 6879 6880 // do we need an input converter buffer provider to give us float? 6881 delete mInputConverterProvider; 6882 mInputConverterProvider = NULL; 6883 if (mRequiresFloat && mSrcFormat != AUDIO_FORMAT_PCM_FLOAT) { 6884 mInputConverterProvider = new ReformatBufferProvider( 6885 audio_channel_count_from_in_mask(mSrcChannelMask), 6886 mSrcFormat, 6887 AUDIO_FORMAT_PCM_FLOAT, 6888 256 /* provider buffer frame count */); 6889 } 6890 6891 // do we need a remixer to do channel mask conversion 6892 if (!mIsLegacyDownmix && !mIsLegacyUpmix && mSrcChannelMask != mDstChannelMask) { 6893 (void) memcpy_by_index_array_initialization_from_channel_mask( 6894 mIdxAry, ARRAY_SIZE(mIdxAry), mDstChannelMask, mSrcChannelMask); 6895 } 6896 return NO_ERROR; 6897} 6898 6899void AudioFlinger::RecordThread::RecordBufferConverter::convertNoResampler( 6900 void *dst, const void *src, size_t frames) 6901{ 6902 // src is native type unless there is legacy upmix or downmix, whereupon it is float. 6903 if (mBufFrameSize != 0 && mBufFrames < frames) { 6904 free(mBuf); 6905 mBufFrames = frames; 6906 (void)posix_memalign(&mBuf, 32, mBufFrames * mBufFrameSize); 6907 } 6908 // do we need to do legacy upmix and downmix? 6909 if (mIsLegacyUpmix || mIsLegacyDownmix) { 6910 void *dstBuf = mBuf != NULL ? mBuf : dst; 6911 if (mIsLegacyUpmix) { 6912 upmix_to_stereo_float_from_mono_float((float *)dstBuf, 6913 (const float *)src, frames); 6914 } else /*mIsLegacyDownmix */ { 6915 downmix_to_mono_float_from_stereo_float((float *)dstBuf, 6916 (const float *)src, frames); 6917 } 6918 if (mBuf != NULL) { 6919 memcpy_by_audio_format(dst, mDstFormat, mBuf, AUDIO_FORMAT_PCM_FLOAT, 6920 frames * mDstChannelCount); 6921 } 6922 return; 6923 } 6924 // do we need to do channel mask conversion? 6925 if (mSrcChannelMask != mDstChannelMask) { 6926 void *dstBuf = mBuf != NULL ? mBuf : dst; 6927 memcpy_by_index_array(dstBuf, mDstChannelCount, 6928 src, mSrcChannelCount, mIdxAry, audio_bytes_per_sample(mSrcFormat), frames); 6929 if (dstBuf == dst) { 6930 return; // format is the same 6931 } 6932 } 6933 // convert to destination buffer 6934 const void *convertBuf = mBuf != NULL ? mBuf : src; 6935 memcpy_by_audio_format(dst, mDstFormat, convertBuf, mSrcFormat, 6936 frames * mDstChannelCount); 6937} 6938 6939void AudioFlinger::RecordThread::RecordBufferConverter::convertResampler( 6940 void *dst, /*not-a-const*/ void *src, size_t frames) 6941{ 6942 // src buffer format is ALWAYS float when entering this routine 6943 if (mIsLegacyUpmix) { 6944 ; // mono to stereo already handled by resampler 6945 } else if (mIsLegacyDownmix 6946 || (mSrcChannelMask == mDstChannelMask && mSrcChannelCount == 1)) { 6947 // the resampler outputs stereo for mono input channel (a feature?) 6948 // must convert to mono 6949 downmix_to_mono_float_from_stereo_float((float *)src, 6950 (const float *)src, frames); 6951 } else if (mSrcChannelMask != mDstChannelMask) { 6952 // convert to mono channel again for channel mask conversion (could be skipped 6953 // with further optimization). 6954 if (mSrcChannelCount == 1) { 6955 downmix_to_mono_float_from_stereo_float((float *)src, 6956 (const float *)src, frames); 6957 } 6958 // convert to destination format (in place, OK as float is larger than other types) 6959 if (mDstFormat != AUDIO_FORMAT_PCM_FLOAT) { 6960 memcpy_by_audio_format(src, mDstFormat, src, AUDIO_FORMAT_PCM_FLOAT, 6961 frames * mSrcChannelCount); 6962 } 6963 // channel convert and save to dst 6964 memcpy_by_index_array(dst, mDstChannelCount, 6965 src, mSrcChannelCount, mIdxAry, audio_bytes_per_sample(mDstFormat), frames); 6966 return; 6967 } 6968 // convert to destination format and save to dst 6969 memcpy_by_audio_format(dst, mDstFormat, src, AUDIO_FORMAT_PCM_FLOAT, 6970 frames * mDstChannelCount); 6971} 6972 6973bool AudioFlinger::RecordThread::checkForNewParameter_l(const String8& keyValuePair, 6974 status_t& status) 6975{ 6976 bool reconfig = false; 6977 6978 status = NO_ERROR; 6979 6980 audio_format_t reqFormat = mFormat; 6981 uint32_t samplingRate = mSampleRate; 6982 // TODO this may change if we want to support capture from HDMI PCM multi channel (e.g on TVs). 6983 audio_channel_mask_t channelMask = audio_channel_in_mask_from_count(mChannelCount); 6984 6985 AudioParameter param = AudioParameter(keyValuePair); 6986 int value; 6987 6988 // scope for AutoPark extends to end of method 6989 AutoPark<FastCapture> park(mFastCapture); 6990 6991 // TODO Investigate when this code runs. Check with audio policy when a sample rate and 6992 // channel count change can be requested. Do we mandate the first client defines the 6993 // HAL sampling rate and channel count or do we allow changes on the fly? 6994 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 6995 samplingRate = value; 6996 reconfig = true; 6997 } 6998 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 6999 if (!audio_is_linear_pcm((audio_format_t) value)) { 7000 status = BAD_VALUE; 7001 } else { 7002 reqFormat = (audio_format_t) value; 7003 reconfig = true; 7004 } 7005 } 7006 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 7007 audio_channel_mask_t mask = (audio_channel_mask_t) value; 7008 if (!audio_is_input_channel(mask) || 7009 audio_channel_count_from_in_mask(mask) > FCC_8) { 7010 status = BAD_VALUE; 7011 } else { 7012 channelMask = mask; 7013 reconfig = true; 7014 } 7015 } 7016 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 7017 // do not accept frame count changes if tracks are open as the track buffer 7018 // size depends on frame count and correct behavior would not be guaranteed 7019 // if frame count is changed after track creation 7020 if (mActiveTracks.size() > 0) { 7021 status = INVALID_OPERATION; 7022 } else { 7023 reconfig = true; 7024 } 7025 } 7026 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 7027 // forward device change to effects that have requested to be 7028 // aware of attached audio device. 7029 for (size_t i = 0; i < mEffectChains.size(); i++) { 7030 mEffectChains[i]->setDevice_l(value); 7031 } 7032 7033 // store input device and output device but do not forward output device to audio HAL. 7034 // Note that status is ignored by the caller for output device 7035 // (see AudioFlinger::setParameters() 7036 if (audio_is_output_devices(value)) { 7037 mOutDevice = value; 7038 status = BAD_VALUE; 7039 } else { 7040 mInDevice = value; 7041 if (value != AUDIO_DEVICE_NONE) { 7042 mPrevInDevice = value; 7043 } 7044 // disable AEC and NS if the device is a BT SCO headset supporting those 7045 // pre processings 7046 if (mTracks.size() > 0) { 7047 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 7048 mAudioFlinger->btNrecIsOff(); 7049 for (size_t i = 0; i < mTracks.size(); i++) { 7050 sp<RecordTrack> track = mTracks[i]; 7051 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId()); 7052 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId()); 7053 } 7054 } 7055 } 7056 } 7057 if (param.getInt(String8(AudioParameter::keyInputSource), value) == NO_ERROR && 7058 mAudioSource != (audio_source_t)value) { 7059 // forward device change to effects that have requested to be 7060 // aware of attached audio device. 7061 for (size_t i = 0; i < mEffectChains.size(); i++) { 7062 mEffectChains[i]->setAudioSource_l((audio_source_t)value); 7063 } 7064 mAudioSource = (audio_source_t)value; 7065 } 7066 7067 if (status == NO_ERROR) { 7068 status = mInput->stream->common.set_parameters(&mInput->stream->common, 7069 keyValuePair.string()); 7070 if (status == INVALID_OPERATION) { 7071 inputStandBy(); 7072 status = mInput->stream->common.set_parameters(&mInput->stream->common, 7073 keyValuePair.string()); 7074 } 7075 if (reconfig) { 7076 if (status == BAD_VALUE && 7077 audio_is_linear_pcm(mInput->stream->common.get_format(&mInput->stream->common)) && 7078 audio_is_linear_pcm(reqFormat) && 7079 (mInput->stream->common.get_sample_rate(&mInput->stream->common) 7080 <= (AUDIO_RESAMPLER_DOWN_RATIO_MAX * samplingRate)) && 7081 audio_channel_count_from_in_mask( 7082 mInput->stream->common.get_channels(&mInput->stream->common)) <= FCC_8) { 7083 status = NO_ERROR; 7084 } 7085 if (status == NO_ERROR) { 7086 readInputParameters_l(); 7087 sendIoConfigEvent_l(AUDIO_INPUT_CONFIG_CHANGED); 7088 } 7089 } 7090 } 7091 7092 return reconfig; 7093} 7094 7095String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 7096{ 7097 Mutex::Autolock _l(mLock); 7098 if (initCheck() != NO_ERROR) { 7099 return String8(); 7100 } 7101 7102 char *s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 7103 const String8 out_s8(s); 7104 free(s); 7105 return out_s8; 7106} 7107 7108void AudioFlinger::RecordThread::ioConfigChanged(audio_io_config_event event, pid_t pid) { 7109 sp<AudioIoDescriptor> desc = new AudioIoDescriptor(); 7110 7111 desc->mIoHandle = mId; 7112 7113 switch (event) { 7114 case AUDIO_INPUT_OPENED: 7115 case AUDIO_INPUT_CONFIG_CHANGED: 7116 desc->mPatch = mPatch; 7117 desc->mChannelMask = mChannelMask; 7118 desc->mSamplingRate = mSampleRate; 7119 desc->mFormat = mFormat; 7120 desc->mFrameCount = mFrameCount; 7121 desc->mFrameCountHAL = mFrameCount; 7122 desc->mLatency = 0; 7123 break; 7124 7125 case AUDIO_INPUT_CLOSED: 7126 default: 7127 break; 7128 } 7129 mAudioFlinger->ioConfigChanged(event, desc, pid); 7130} 7131 7132void AudioFlinger::RecordThread::readInputParameters_l() 7133{ 7134 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 7135 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 7136 mChannelCount = audio_channel_count_from_in_mask(mChannelMask); 7137 if (mChannelCount > FCC_8) { 7138 ALOGE("HAL channel count %d > %d", mChannelCount, FCC_8); 7139 } 7140 mHALFormat = mInput->stream->common.get_format(&mInput->stream->common); 7141 mFormat = mHALFormat; 7142 if (!audio_is_linear_pcm(mFormat)) { 7143 ALOGE("HAL format %#x is not linear pcm", mFormat); 7144 } 7145 mFrameSize = audio_stream_in_frame_size(mInput->stream); 7146 mBufferSize = mInput->stream->common.get_buffer_size(&mInput->stream->common); 7147 mFrameCount = mBufferSize / mFrameSize; 7148 // This is the formula for calculating the temporary buffer size. 7149 // With 7 HAL buffers, we can guarantee ability to down-sample the input by ratio of 6:1 to 7150 // 1 full output buffer, regardless of the alignment of the available input. 7151 // The value is somewhat arbitrary, and could probably be even larger. 7152 // A larger value should allow more old data to be read after a track calls start(), 7153 // without increasing latency. 7154 // 7155 // Note this is independent of the maximum downsampling ratio permitted for capture. 7156 mRsmpInFrames = mFrameCount * 7; 7157 mRsmpInFramesP2 = roundup(mRsmpInFrames); 7158 free(mRsmpInBuffer); 7159 mRsmpInBuffer = NULL; 7160 7161 // TODO optimize audio capture buffer sizes ... 7162 // Here we calculate the size of the sliding buffer used as a source 7163 // for resampling. mRsmpInFramesP2 is currently roundup(mFrameCount * 7). 7164 // For current HAL frame counts, this is usually 2048 = 40 ms. It would 7165 // be better to have it derived from the pipe depth in the long term. 7166 // The current value is higher than necessary. However it should not add to latency. 7167 7168 // Over-allocate beyond mRsmpInFramesP2 to permit a HAL read past end of buffer 7169 size_t bufferSize = (mRsmpInFramesP2 + mFrameCount - 1) * mFrameSize; 7170 (void)posix_memalign(&mRsmpInBuffer, 32, bufferSize); 7171 memset(mRsmpInBuffer, 0, bufferSize); // if posix_memalign fails, will segv here. 7172 7173 // AudioRecord mSampleRate and mChannelCount are constant due to AudioRecord API constraints. 7174 // But if thread's mSampleRate or mChannelCount changes, how will that affect active tracks? 7175} 7176 7177uint32_t AudioFlinger::RecordThread::getInputFramesLost() 7178{ 7179 Mutex::Autolock _l(mLock); 7180 if (initCheck() != NO_ERROR) { 7181 return 0; 7182 } 7183 7184 return mInput->stream->get_input_frames_lost(mInput->stream); 7185} 7186 7187uint32_t AudioFlinger::RecordThread::hasAudioSession(audio_session_t sessionId) const 7188{ 7189 Mutex::Autolock _l(mLock); 7190 uint32_t result = 0; 7191 if (getEffectChain_l(sessionId) != 0) { 7192 result = EFFECT_SESSION; 7193 } 7194 7195 for (size_t i = 0; i < mTracks.size(); ++i) { 7196 if (sessionId == mTracks[i]->sessionId()) { 7197 result |= TRACK_SESSION; 7198 break; 7199 } 7200 } 7201 7202 return result; 7203} 7204 7205KeyedVector<audio_session_t, bool> AudioFlinger::RecordThread::sessionIds() const 7206{ 7207 KeyedVector<audio_session_t, bool> ids; 7208 Mutex::Autolock _l(mLock); 7209 for (size_t j = 0; j < mTracks.size(); ++j) { 7210 sp<RecordThread::RecordTrack> track = mTracks[j]; 7211 audio_session_t sessionId = track->sessionId(); 7212 if (ids.indexOfKey(sessionId) < 0) { 7213 ids.add(sessionId, true); 7214 } 7215 } 7216 return ids; 7217} 7218 7219AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 7220{ 7221 Mutex::Autolock _l(mLock); 7222 AudioStreamIn *input = mInput; 7223 mInput = NULL; 7224 return input; 7225} 7226 7227// this method must always be called either with ThreadBase mLock held or inside the thread loop 7228audio_stream_t* AudioFlinger::RecordThread::stream() const 7229{ 7230 if (mInput == NULL) { 7231 return NULL; 7232 } 7233 return &mInput->stream->common; 7234} 7235 7236status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 7237{ 7238 // only one chain per input thread 7239 if (mEffectChains.size() != 0) { 7240 ALOGW("addEffectChain_l() already one chain %p on thread %p", chain.get(), this); 7241 return INVALID_OPERATION; 7242 } 7243 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 7244 chain->setThread(this); 7245 chain->setInBuffer(NULL); 7246 chain->setOutBuffer(NULL); 7247 7248 checkSuspendOnAddEffectChain_l(chain); 7249 7250 // make sure enabled pre processing effects state is communicated to the HAL as we 7251 // just moved them to a new input stream. 7252 chain->syncHalEffectsState(); 7253 7254 mEffectChains.add(chain); 7255 7256 return NO_ERROR; 7257} 7258 7259size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 7260{ 7261 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 7262 ALOGW_IF(mEffectChains.size() != 1, 7263 "removeEffectChain_l() %p invalid chain size %zu on thread %p", 7264 chain.get(), mEffectChains.size(), this); 7265 if (mEffectChains.size() == 1) { 7266 mEffectChains.removeAt(0); 7267 } 7268 return 0; 7269} 7270 7271status_t AudioFlinger::RecordThread::createAudioPatch_l(const struct audio_patch *patch, 7272 audio_patch_handle_t *handle) 7273{ 7274 status_t status = NO_ERROR; 7275 7276 // store new device and send to effects 7277 mInDevice = patch->sources[0].ext.device.type; 7278 mPatch = *patch; 7279 for (size_t i = 0; i < mEffectChains.size(); i++) { 7280 mEffectChains[i]->setDevice_l(mInDevice); 7281 } 7282 7283 // disable AEC and NS if the device is a BT SCO headset supporting those 7284 // pre processings 7285 if (mTracks.size() > 0) { 7286 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 7287 mAudioFlinger->btNrecIsOff(); 7288 for (size_t i = 0; i < mTracks.size(); i++) { 7289 sp<RecordTrack> track = mTracks[i]; 7290 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId()); 7291 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId()); 7292 } 7293 } 7294 7295 // store new source and send to effects 7296 if (mAudioSource != patch->sinks[0].ext.mix.usecase.source) { 7297 mAudioSource = patch->sinks[0].ext.mix.usecase.source; 7298 for (size_t i = 0; i < mEffectChains.size(); i++) { 7299 mEffectChains[i]->setAudioSource_l(mAudioSource); 7300 } 7301 } 7302 7303 if (mInput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 7304 audio_hw_device_t *hwDevice = mInput->audioHwDev->hwDevice(); 7305 status = hwDevice->create_audio_patch(hwDevice, 7306 patch->num_sources, 7307 patch->sources, 7308 patch->num_sinks, 7309 patch->sinks, 7310 handle); 7311 } else { 7312 char *address; 7313 if (strcmp(patch->sources[0].ext.device.address, "") != 0) { 7314 address = audio_device_address_to_parameter( 7315 patch->sources[0].ext.device.type, 7316 patch->sources[0].ext.device.address); 7317 } else { 7318 address = (char *)calloc(1, 1); 7319 } 7320 AudioParameter param = AudioParameter(String8(address)); 7321 free(address); 7322 param.addInt(String8(AUDIO_PARAMETER_STREAM_ROUTING), 7323 (int)patch->sources[0].ext.device.type); 7324 param.addInt(String8(AUDIO_PARAMETER_STREAM_INPUT_SOURCE), 7325 (int)patch->sinks[0].ext.mix.usecase.source); 7326 status = mInput->stream->common.set_parameters(&mInput->stream->common, 7327 param.toString().string()); 7328 *handle = AUDIO_PATCH_HANDLE_NONE; 7329 } 7330 7331 if (mInDevice != mPrevInDevice) { 7332 sendIoConfigEvent_l(AUDIO_INPUT_CONFIG_CHANGED); 7333 mPrevInDevice = mInDevice; 7334 } 7335 7336 return status; 7337} 7338 7339status_t AudioFlinger::RecordThread::releaseAudioPatch_l(const audio_patch_handle_t handle) 7340{ 7341 status_t status = NO_ERROR; 7342 7343 mInDevice = AUDIO_DEVICE_NONE; 7344 7345 if (mInput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 7346 audio_hw_device_t *hwDevice = mInput->audioHwDev->hwDevice(); 7347 status = hwDevice->release_audio_patch(hwDevice, handle); 7348 } else { 7349 AudioParameter param; 7350 param.addInt(String8(AUDIO_PARAMETER_STREAM_ROUTING), 0); 7351 status = mInput->stream->common.set_parameters(&mInput->stream->common, 7352 param.toString().string()); 7353 } 7354 return status; 7355} 7356 7357void AudioFlinger::RecordThread::addPatchRecord(const sp<PatchRecord>& record) 7358{ 7359 Mutex::Autolock _l(mLock); 7360 mTracks.add(record); 7361} 7362 7363void AudioFlinger::RecordThread::deletePatchRecord(const sp<PatchRecord>& record) 7364{ 7365 Mutex::Autolock _l(mLock); 7366 destroyTrack_l(record); 7367} 7368 7369void AudioFlinger::RecordThread::getAudioPortConfig(struct audio_port_config *config) 7370{ 7371 ThreadBase::getAudioPortConfig(config); 7372 config->role = AUDIO_PORT_ROLE_SINK; 7373 config->ext.mix.hw_module = mInput->audioHwDev->handle(); 7374 config->ext.mix.usecase.source = mAudioSource; 7375} 7376 7377} // namespace android 7378