Threads.cpp revision 3eaf66b860f9a0d8af0dd4d5ac6adb5b67d7b73a
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->mLatency = latency_l(); 2157 break; 2158 2159 case AUDIO_OUTPUT_CLOSED: 2160 default: 2161 break; 2162 } 2163 mAudioFlinger->ioConfigChanged(event, desc, pid); 2164} 2165 2166void AudioFlinger::PlaybackThread::writeCallback() 2167{ 2168 ALOG_ASSERT(mCallbackThread != 0); 2169 mCallbackThread->resetWriteBlocked(); 2170} 2171 2172void AudioFlinger::PlaybackThread::drainCallback() 2173{ 2174 ALOG_ASSERT(mCallbackThread != 0); 2175 mCallbackThread->resetDraining(); 2176} 2177 2178void AudioFlinger::PlaybackThread::resetWriteBlocked(uint32_t sequence) 2179{ 2180 Mutex::Autolock _l(mLock); 2181 // reject out of sequence requests 2182 if ((mWriteAckSequence & 1) && (sequence == mWriteAckSequence)) { 2183 mWriteAckSequence &= ~1; 2184 mWaitWorkCV.signal(); 2185 } 2186} 2187 2188void AudioFlinger::PlaybackThread::resetDraining(uint32_t sequence) 2189{ 2190 Mutex::Autolock _l(mLock); 2191 // reject out of sequence requests 2192 if ((mDrainSequence & 1) && (sequence == mDrainSequence)) { 2193 mDrainSequence &= ~1; 2194 mWaitWorkCV.signal(); 2195 } 2196} 2197 2198// static 2199int AudioFlinger::PlaybackThread::asyncCallback(stream_callback_event_t event, 2200 void *param __unused, 2201 void *cookie) 2202{ 2203 AudioFlinger::PlaybackThread *me = (AudioFlinger::PlaybackThread *)cookie; 2204 ALOGV("asyncCallback() event %d", event); 2205 switch (event) { 2206 case STREAM_CBK_EVENT_WRITE_READY: 2207 me->writeCallback(); 2208 break; 2209 case STREAM_CBK_EVENT_DRAIN_READY: 2210 me->drainCallback(); 2211 break; 2212 default: 2213 ALOGW("asyncCallback() unknown event %d", event); 2214 break; 2215 } 2216 return 0; 2217} 2218 2219void AudioFlinger::PlaybackThread::readOutputParameters_l() 2220{ 2221 // unfortunately we have no way of recovering from errors here, hence the LOG_ALWAYS_FATAL 2222 mSampleRate = mOutput->getSampleRate(); 2223 mChannelMask = mOutput->getChannelMask(); 2224 if (!audio_is_output_channel(mChannelMask)) { 2225 LOG_ALWAYS_FATAL("HAL channel mask %#x not valid for output", mChannelMask); 2226 } 2227 if ((mType == MIXER || mType == DUPLICATING) 2228 && !isValidPcmSinkChannelMask(mChannelMask)) { 2229 LOG_ALWAYS_FATAL("HAL channel mask %#x not supported for mixed output", 2230 mChannelMask); 2231 } 2232 mChannelCount = audio_channel_count_from_out_mask(mChannelMask); 2233 2234 // Get actual HAL format. 2235 mHALFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 2236 // Get format from the shim, which will be different than the HAL format 2237 // if playing compressed audio over HDMI passthrough. 2238 mFormat = mOutput->getFormat(); 2239 if (!audio_is_valid_format(mFormat)) { 2240 LOG_ALWAYS_FATAL("HAL format %#x not valid for output", mFormat); 2241 } 2242 if ((mType == MIXER || mType == DUPLICATING) 2243 && !isValidPcmSinkFormat(mFormat)) { 2244 LOG_FATAL("HAL format %#x not supported for mixed output", 2245 mFormat); 2246 } 2247 mFrameSize = mOutput->getFrameSize(); 2248 mBufferSize = mOutput->stream->common.get_buffer_size(&mOutput->stream->common); 2249 mFrameCount = mBufferSize / mFrameSize; 2250 if (mFrameCount & 15) { 2251 ALOGW("HAL output buffer size is %zu frames but AudioMixer requires multiples of 16 frames", 2252 mFrameCount); 2253 } 2254 2255 if ((mOutput->flags & AUDIO_OUTPUT_FLAG_NON_BLOCKING) && 2256 (mOutput->stream->set_callback != NULL)) { 2257 if (mOutput->stream->set_callback(mOutput->stream, 2258 AudioFlinger::PlaybackThread::asyncCallback, this) == 0) { 2259 mUseAsyncWrite = true; 2260 mCallbackThread = new AudioFlinger::AsyncCallbackThread(this); 2261 } 2262 } 2263 2264 mHwSupportsPause = false; 2265 if (mOutput->flags & AUDIO_OUTPUT_FLAG_DIRECT) { 2266 if (mOutput->stream->pause != NULL) { 2267 if (mOutput->stream->resume != NULL) { 2268 mHwSupportsPause = true; 2269 } else { 2270 ALOGW("direct output implements pause but not resume"); 2271 } 2272 } else if (mOutput->stream->resume != NULL) { 2273 ALOGW("direct output implements resume but not pause"); 2274 } 2275 } 2276 if (!mHwSupportsPause && mOutput->flags & AUDIO_OUTPUT_FLAG_HW_AV_SYNC) { 2277 LOG_ALWAYS_FATAL("HW_AV_SYNC requested but HAL does not implement pause and resume"); 2278 } 2279 2280 if (mType == DUPLICATING && mMixerBufferEnabled && mEffectBufferEnabled) { 2281 // For best precision, we use float instead of the associated output 2282 // device format (typically PCM 16 bit). 2283 2284 mFormat = AUDIO_FORMAT_PCM_FLOAT; 2285 mFrameSize = mChannelCount * audio_bytes_per_sample(mFormat); 2286 mBufferSize = mFrameSize * mFrameCount; 2287 2288 // TODO: We currently use the associated output device channel mask and sample rate. 2289 // (1) Perhaps use the ORed channel mask of all downstream MixerThreads 2290 // (if a valid mask) to avoid premature downmix. 2291 // (2) Perhaps use the maximum sample rate of all downstream MixerThreads 2292 // instead of the output device sample rate to avoid loss of high frequency information. 2293 // This may need to be updated as MixerThread/OutputTracks are added and not here. 2294 } 2295 2296 // Calculate size of normal sink buffer relative to the HAL output buffer size 2297 double multiplier = 1.0; 2298 if (mType == MIXER && (kUseFastMixer == FastMixer_Static || 2299 kUseFastMixer == FastMixer_Dynamic)) { 2300 size_t minNormalFrameCount = (kMinNormalSinkBufferSizeMs * mSampleRate) / 1000; 2301 size_t maxNormalFrameCount = (kMaxNormalSinkBufferSizeMs * mSampleRate) / 1000; 2302 // round up minimum and round down maximum to nearest 16 frames to satisfy AudioMixer 2303 minNormalFrameCount = (minNormalFrameCount + 15) & ~15; 2304 maxNormalFrameCount = maxNormalFrameCount & ~15; 2305 if (maxNormalFrameCount < minNormalFrameCount) { 2306 maxNormalFrameCount = minNormalFrameCount; 2307 } 2308 multiplier = (double) minNormalFrameCount / (double) mFrameCount; 2309 if (multiplier <= 1.0) { 2310 multiplier = 1.0; 2311 } else if (multiplier <= 2.0) { 2312 if (2 * mFrameCount <= maxNormalFrameCount) { 2313 multiplier = 2.0; 2314 } else { 2315 multiplier = (double) maxNormalFrameCount / (double) mFrameCount; 2316 } 2317 } else { 2318 // prefer an even multiplier, for compatibility with doubling of fast tracks due to HAL 2319 // SRC (it would be unusual for the normal sink buffer size to not be a multiple of fast 2320 // track, but we sometimes have to do this to satisfy the maximum frame count 2321 // constraint) 2322 // FIXME this rounding up should not be done if no HAL SRC 2323 uint32_t truncMult = (uint32_t) multiplier; 2324 if ((truncMult & 1)) { 2325 if ((truncMult + 1) * mFrameCount <= maxNormalFrameCount) { 2326 ++truncMult; 2327 } 2328 } 2329 multiplier = (double) truncMult; 2330 } 2331 } 2332 mNormalFrameCount = multiplier * mFrameCount; 2333 // round up to nearest 16 frames to satisfy AudioMixer 2334 if (mType == MIXER || mType == DUPLICATING) { 2335 mNormalFrameCount = (mNormalFrameCount + 15) & ~15; 2336 } 2337 ALOGI("HAL output buffer size %zu frames, normal sink buffer size %zu frames", mFrameCount, 2338 mNormalFrameCount); 2339 2340 // Check if we want to throttle the processing to no more than 2x normal rate 2341 mThreadThrottle = property_get_bool("af.thread.throttle", true /* default_value */); 2342 mThreadThrottleTimeMs = 0; 2343 mThreadThrottleEndMs = 0; 2344 mHalfBufferMs = mNormalFrameCount * 1000 / (2 * mSampleRate); 2345 2346 // mSinkBuffer is the sink buffer. Size is always multiple-of-16 frames. 2347 // Originally this was int16_t[] array, need to remove legacy implications. 2348 free(mSinkBuffer); 2349 mSinkBuffer = NULL; 2350 // For sink buffer size, we use the frame size from the downstream sink to avoid problems 2351 // with non PCM formats for compressed music, e.g. AAC, and Offload threads. 2352 const size_t sinkBufferSize = mNormalFrameCount * mFrameSize; 2353 (void)posix_memalign(&mSinkBuffer, 32, sinkBufferSize); 2354 2355 // We resize the mMixerBuffer according to the requirements of the sink buffer which 2356 // drives the output. 2357 free(mMixerBuffer); 2358 mMixerBuffer = NULL; 2359 if (mMixerBufferEnabled) { 2360 mMixerBufferFormat = AUDIO_FORMAT_PCM_FLOAT; // also valid: AUDIO_FORMAT_PCM_16_BIT. 2361 mMixerBufferSize = mNormalFrameCount * mChannelCount 2362 * audio_bytes_per_sample(mMixerBufferFormat); 2363 (void)posix_memalign(&mMixerBuffer, 32, mMixerBufferSize); 2364 } 2365 free(mEffectBuffer); 2366 mEffectBuffer = NULL; 2367 if (mEffectBufferEnabled) { 2368 mEffectBufferFormat = AUDIO_FORMAT_PCM_16_BIT; // Note: Effects support 16b only 2369 mEffectBufferSize = mNormalFrameCount * mChannelCount 2370 * audio_bytes_per_sample(mEffectBufferFormat); 2371 (void)posix_memalign(&mEffectBuffer, 32, mEffectBufferSize); 2372 } 2373 2374 // force reconfiguration of effect chains and engines to take new buffer size and audio 2375 // parameters into account 2376 // Note that mLock is not held when readOutputParameters_l() is called from the constructor 2377 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 2378 // matter. 2379 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 2380 Vector< sp<EffectChain> > effectChains = mEffectChains; 2381 for (size_t i = 0; i < effectChains.size(); i ++) { 2382 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 2383 } 2384} 2385 2386 2387status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames) 2388{ 2389 if (halFrames == NULL || dspFrames == NULL) { 2390 return BAD_VALUE; 2391 } 2392 Mutex::Autolock _l(mLock); 2393 if (initCheck() != NO_ERROR) { 2394 return INVALID_OPERATION; 2395 } 2396 int64_t framesWritten = mBytesWritten / mFrameSize; 2397 *halFrames = framesWritten; 2398 2399 if (isSuspended()) { 2400 // return an estimation of rendered frames when the output is suspended 2401 size_t latencyFrames = (latency_l() * mSampleRate) / 1000; 2402 *dspFrames = (uint32_t) 2403 (framesWritten >= (int64_t)latencyFrames ? framesWritten - latencyFrames : 0); 2404 return NO_ERROR; 2405 } else { 2406 status_t status; 2407 uint32_t frames; 2408 status = mOutput->getRenderPosition(&frames); 2409 *dspFrames = (size_t)frames; 2410 return status; 2411 } 2412} 2413 2414uint32_t AudioFlinger::PlaybackThread::hasAudioSession(audio_session_t sessionId) const 2415{ 2416 Mutex::Autolock _l(mLock); 2417 uint32_t result = 0; 2418 if (getEffectChain_l(sessionId) != 0) { 2419 result = EFFECT_SESSION; 2420 } 2421 2422 for (size_t i = 0; i < mTracks.size(); ++i) { 2423 sp<Track> track = mTracks[i]; 2424 if (sessionId == track->sessionId() && !track->isInvalid()) { 2425 result |= TRACK_SESSION; 2426 break; 2427 } 2428 } 2429 2430 return result; 2431} 2432 2433uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(audio_session_t sessionId) 2434{ 2435 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 2436 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 2437 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 2438 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 2439 } 2440 for (size_t i = 0; i < mTracks.size(); i++) { 2441 sp<Track> track = mTracks[i]; 2442 if (sessionId == track->sessionId() && !track->isInvalid()) { 2443 return AudioSystem::getStrategyForStream(track->streamType()); 2444 } 2445 } 2446 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 2447} 2448 2449 2450AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 2451{ 2452 Mutex::Autolock _l(mLock); 2453 return mOutput; 2454} 2455 2456AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 2457{ 2458 Mutex::Autolock _l(mLock); 2459 AudioStreamOut *output = mOutput; 2460 mOutput = NULL; 2461 // FIXME FastMixer might also have a raw ptr to mOutputSink; 2462 // must push a NULL and wait for ack 2463 mOutputSink.clear(); 2464 mPipeSink.clear(); 2465 mNormalSink.clear(); 2466 return output; 2467} 2468 2469// this method must always be called either with ThreadBase mLock held or inside the thread loop 2470audio_stream_t* AudioFlinger::PlaybackThread::stream() const 2471{ 2472 if (mOutput == NULL) { 2473 return NULL; 2474 } 2475 return &mOutput->stream->common; 2476} 2477 2478uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() const 2479{ 2480 return (uint32_t)((uint32_t)((mNormalFrameCount * 1000) / mSampleRate) * 1000); 2481} 2482 2483status_t AudioFlinger::PlaybackThread::setSyncEvent(const sp<SyncEvent>& event) 2484{ 2485 if (!isValidSyncEvent(event)) { 2486 return BAD_VALUE; 2487 } 2488 2489 Mutex::Autolock _l(mLock); 2490 2491 for (size_t i = 0; i < mTracks.size(); ++i) { 2492 sp<Track> track = mTracks[i]; 2493 if (event->triggerSession() == track->sessionId()) { 2494 (void) track->setSyncEvent(event); 2495 return NO_ERROR; 2496 } 2497 } 2498 2499 return NAME_NOT_FOUND; 2500} 2501 2502bool AudioFlinger::PlaybackThread::isValidSyncEvent(const sp<SyncEvent>& event) const 2503{ 2504 return event->type() == AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE; 2505} 2506 2507void AudioFlinger::PlaybackThread::threadLoop_removeTracks( 2508 const Vector< sp<Track> >& tracksToRemove) 2509{ 2510 size_t count = tracksToRemove.size(); 2511 if (count > 0) { 2512 for (size_t i = 0 ; i < count ; i++) { 2513 const sp<Track>& track = tracksToRemove.itemAt(i); 2514 if (track->isExternalTrack()) { 2515 AudioSystem::stopOutput(mId, track->streamType(), 2516 track->sessionId()); 2517#ifdef ADD_BATTERY_DATA 2518 // to track the speaker usage 2519 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 2520#endif 2521 if (track->isTerminated()) { 2522 AudioSystem::releaseOutput(mId, track->streamType(), 2523 track->sessionId()); 2524 } 2525 } 2526 } 2527 } 2528} 2529 2530void AudioFlinger::PlaybackThread::checkSilentMode_l() 2531{ 2532 if (!mMasterMute) { 2533 char value[PROPERTY_VALUE_MAX]; 2534 if (property_get("ro.audio.silent", value, "0") > 0) { 2535 char *endptr; 2536 unsigned long ul = strtoul(value, &endptr, 0); 2537 if (*endptr == '\0' && ul != 0) { 2538 ALOGD("Silence is golden"); 2539 // The setprop command will not allow a property to be changed after 2540 // the first time it is set, so we don't have to worry about un-muting. 2541 setMasterMute_l(true); 2542 } 2543 } 2544 } 2545} 2546 2547// shared by MIXER and DIRECT, overridden by DUPLICATING 2548ssize_t AudioFlinger::PlaybackThread::threadLoop_write() 2549{ 2550 // FIXME rewrite to reduce number of system calls 2551 mLastWriteTime = systemTime(); 2552 mInWrite = true; 2553 ssize_t bytesWritten; 2554 const size_t offset = mCurrentWriteLength - mBytesRemaining; 2555 2556 // If an NBAIO sink is present, use it to write the normal mixer's submix 2557 if (mNormalSink != 0) { 2558 2559 const size_t count = mBytesRemaining / mFrameSize; 2560 2561 ATRACE_BEGIN("write"); 2562 // update the setpoint when AudioFlinger::mScreenState changes 2563 uint32_t screenState = AudioFlinger::mScreenState; 2564 if (screenState != mScreenState) { 2565 mScreenState = screenState; 2566 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 2567 if (pipe != NULL) { 2568 pipe->setAvgFrames((mScreenState & 1) ? 2569 (pipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2570 } 2571 } 2572 ssize_t framesWritten = mNormalSink->write((char *)mSinkBuffer + offset, count); 2573 ATRACE_END(); 2574 if (framesWritten > 0) { 2575 bytesWritten = framesWritten * mFrameSize; 2576 } else { 2577 bytesWritten = framesWritten; 2578 } 2579 // otherwise use the HAL / AudioStreamOut directly 2580 } else { 2581 // Direct output and offload threads 2582 2583 if (mUseAsyncWrite) { 2584 ALOGW_IF(mWriteAckSequence & 1, "threadLoop_write(): out of sequence write request"); 2585 mWriteAckSequence += 2; 2586 mWriteAckSequence |= 1; 2587 ALOG_ASSERT(mCallbackThread != 0); 2588 mCallbackThread->setWriteBlocked(mWriteAckSequence); 2589 } 2590 // FIXME We should have an implementation of timestamps for direct output threads. 2591 // They are used e.g for multichannel PCM playback over HDMI. 2592 bytesWritten = mOutput->write((char *)mSinkBuffer + offset, mBytesRemaining); 2593 2594 if (mUseAsyncWrite && 2595 ((bytesWritten < 0) || (bytesWritten == (ssize_t)mBytesRemaining))) { 2596 // do not wait for async callback in case of error of full write 2597 mWriteAckSequence &= ~1; 2598 ALOG_ASSERT(mCallbackThread != 0); 2599 mCallbackThread->setWriteBlocked(mWriteAckSequence); 2600 } 2601 } 2602 2603 mNumWrites++; 2604 mInWrite = false; 2605 mStandby = false; 2606 return bytesWritten; 2607} 2608 2609void AudioFlinger::PlaybackThread::threadLoop_drain() 2610{ 2611 if (mOutput->stream->drain) { 2612 ALOGV("draining %s", (mMixerStatus == MIXER_DRAIN_TRACK) ? "early" : "full"); 2613 if (mUseAsyncWrite) { 2614 ALOGW_IF(mDrainSequence & 1, "threadLoop_drain(): out of sequence drain request"); 2615 mDrainSequence |= 1; 2616 ALOG_ASSERT(mCallbackThread != 0); 2617 mCallbackThread->setDraining(mDrainSequence); 2618 } 2619 mOutput->stream->drain(mOutput->stream, 2620 (mMixerStatus == MIXER_DRAIN_TRACK) ? AUDIO_DRAIN_EARLY_NOTIFY 2621 : AUDIO_DRAIN_ALL); 2622 } 2623} 2624 2625void AudioFlinger::PlaybackThread::threadLoop_exit() 2626{ 2627 { 2628 Mutex::Autolock _l(mLock); 2629 for (size_t i = 0; i < mTracks.size(); i++) { 2630 sp<Track> track = mTracks[i]; 2631 track->invalidate(); 2632 } 2633 } 2634} 2635 2636/* 2637The derived values that are cached: 2638 - mSinkBufferSize from frame count * frame size 2639 - mActiveSleepTimeUs from activeSleepTimeUs() 2640 - mIdleSleepTimeUs from idleSleepTimeUs() 2641 - mStandbyDelayNs from mActiveSleepTimeUs (DIRECT only) or forced to at least 2642 kDefaultStandbyTimeInNsecs when connected to an A2DP device. 2643 - maxPeriod from frame count and sample rate (MIXER only) 2644 2645The parameters that affect these derived values are: 2646 - frame count 2647 - frame size 2648 - sample rate 2649 - device type: A2DP or not 2650 - device latency 2651 - format: PCM or not 2652 - active sleep time 2653 - idle sleep time 2654*/ 2655 2656void AudioFlinger::PlaybackThread::cacheParameters_l() 2657{ 2658 mSinkBufferSize = mNormalFrameCount * mFrameSize; 2659 mActiveSleepTimeUs = activeSleepTimeUs(); 2660 mIdleSleepTimeUs = idleSleepTimeUs(); 2661 2662 // make sure standby delay is not too short when connected to an A2DP sink to avoid 2663 // truncating audio when going to standby. 2664 mStandbyDelayNs = AudioFlinger::mStandbyTimeInNsecs; 2665 if ((mOutDevice & AUDIO_DEVICE_OUT_ALL_A2DP) != 0) { 2666 if (mStandbyDelayNs < kDefaultStandbyTimeInNsecs) { 2667 mStandbyDelayNs = kDefaultStandbyTimeInNsecs; 2668 } 2669 } 2670} 2671 2672void AudioFlinger::PlaybackThread::invalidateTracks(audio_stream_type_t streamType) 2673{ 2674 ALOGV("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %zu", 2675 this, streamType, mTracks.size()); 2676 Mutex::Autolock _l(mLock); 2677 2678 size_t size = mTracks.size(); 2679 for (size_t i = 0; i < size; i++) { 2680 sp<Track> t = mTracks[i]; 2681 if (t->streamType() == streamType && t->isExternalTrack()) { 2682 t->invalidate(); 2683 } 2684 } 2685} 2686 2687status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 2688{ 2689 audio_session_t session = chain->sessionId(); 2690 int16_t* buffer = reinterpret_cast<int16_t*>(mEffectBufferEnabled 2691 ? mEffectBuffer : mSinkBuffer); 2692 bool ownsBuffer = false; 2693 2694 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 2695 if (session > AUDIO_SESSION_OUTPUT_MIX) { 2696 // Only one effect chain can be present in direct output thread and it uses 2697 // the sink buffer as input 2698 if (mType != DIRECT) { 2699 size_t numSamples = mNormalFrameCount * mChannelCount; 2700 buffer = new int16_t[numSamples]; 2701 memset(buffer, 0, numSamples * sizeof(int16_t)); 2702 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 2703 ownsBuffer = true; 2704 } 2705 2706 // Attach all tracks with same session ID to this chain. 2707 for (size_t i = 0; i < mTracks.size(); ++i) { 2708 sp<Track> track = mTracks[i]; 2709 if (session == track->sessionId()) { 2710 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), 2711 buffer); 2712 track->setMainBuffer(buffer); 2713 chain->incTrackCnt(); 2714 } 2715 } 2716 2717 // indicate all active tracks in the chain 2718 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 2719 sp<Track> track = mActiveTracks[i].promote(); 2720 if (track == 0) { 2721 continue; 2722 } 2723 if (session == track->sessionId()) { 2724 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 2725 chain->incActiveTrackCnt(); 2726 } 2727 } 2728 } 2729 chain->setThread(this); 2730 chain->setInBuffer(buffer, ownsBuffer); 2731 chain->setOutBuffer(reinterpret_cast<int16_t*>(mEffectBufferEnabled 2732 ? mEffectBuffer : mSinkBuffer)); 2733 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 2734 // chains list in order to be processed last as it contains output stage effects. 2735 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 2736 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 2737 // after track specific effects and before output stage. 2738 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 2739 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX. 2740 // Effect chain for other sessions are inserted at beginning of effect 2741 // chains list to be processed before output mix effects. Relative order between other 2742 // sessions is not important. 2743 static_assert(AUDIO_SESSION_OUTPUT_MIX == 0 && 2744 AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX, 2745 "audio_session_t constants misdefined"); 2746 size_t size = mEffectChains.size(); 2747 size_t i = 0; 2748 for (i = 0; i < size; i++) { 2749 if (mEffectChains[i]->sessionId() < session) { 2750 break; 2751 } 2752 } 2753 mEffectChains.insertAt(chain, i); 2754 checkSuspendOnAddEffectChain_l(chain); 2755 2756 return NO_ERROR; 2757} 2758 2759size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 2760{ 2761 audio_session_t session = chain->sessionId(); 2762 2763 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 2764 2765 for (size_t i = 0; i < mEffectChains.size(); i++) { 2766 if (chain == mEffectChains[i]) { 2767 mEffectChains.removeAt(i); 2768 // detach all active tracks from the chain 2769 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 2770 sp<Track> track = mActiveTracks[i].promote(); 2771 if (track == 0) { 2772 continue; 2773 } 2774 if (session == track->sessionId()) { 2775 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 2776 chain.get(), session); 2777 chain->decActiveTrackCnt(); 2778 } 2779 } 2780 2781 // detach all tracks with same session ID from this chain 2782 for (size_t i = 0; i < mTracks.size(); ++i) { 2783 sp<Track> track = mTracks[i]; 2784 if (session == track->sessionId()) { 2785 track->setMainBuffer(reinterpret_cast<int16_t*>(mSinkBuffer)); 2786 chain->decTrackCnt(); 2787 } 2788 } 2789 break; 2790 } 2791 } 2792 return mEffectChains.size(); 2793} 2794 2795status_t AudioFlinger::PlaybackThread::attachAuxEffect( 2796 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2797{ 2798 Mutex::Autolock _l(mLock); 2799 return attachAuxEffect_l(track, EffectId); 2800} 2801 2802status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 2803 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2804{ 2805 status_t status = NO_ERROR; 2806 2807 if (EffectId == 0) { 2808 track->setAuxBuffer(0, NULL); 2809 } else { 2810 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 2811 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 2812 if (effect != 0) { 2813 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 2814 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 2815 } else { 2816 status = INVALID_OPERATION; 2817 } 2818 } else { 2819 status = BAD_VALUE; 2820 } 2821 } 2822 return status; 2823} 2824 2825void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 2826{ 2827 for (size_t i = 0; i < mTracks.size(); ++i) { 2828 sp<Track> track = mTracks[i]; 2829 if (track->auxEffectId() == effectId) { 2830 attachAuxEffect_l(track, 0); 2831 } 2832 } 2833} 2834 2835bool AudioFlinger::PlaybackThread::threadLoop() 2836{ 2837 Vector< sp<Track> > tracksToRemove; 2838 2839 mStandbyTimeNs = systemTime(); 2840 2841 // MIXER 2842 nsecs_t lastWarning = 0; 2843 2844 // DUPLICATING 2845 // FIXME could this be made local to while loop? 2846 writeFrames = 0; 2847 2848 int lastGeneration = 0; 2849 2850 cacheParameters_l(); 2851 mSleepTimeUs = mIdleSleepTimeUs; 2852 2853 if (mType == MIXER) { 2854 sleepTimeShift = 0; 2855 } 2856 2857 CpuStats cpuStats; 2858 const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid())); 2859 2860 acquireWakeLock(); 2861 2862 // mNBLogWriter->log can only be called while thread mutex mLock is held. 2863 // So if you need to log when mutex is unlocked, set logString to a non-NULL string, 2864 // and then that string will be logged at the next convenient opportunity. 2865 const char *logString = NULL; 2866 2867 checkSilentMode_l(); 2868 2869 while (!exitPending()) 2870 { 2871 cpuStats.sample(myName); 2872 2873 Vector< sp<EffectChain> > effectChains; 2874 2875 { // scope for mLock 2876 2877 Mutex::Autolock _l(mLock); 2878 2879 processConfigEvents_l(); 2880 2881 if (logString != NULL) { 2882 mNBLogWriter->logTimestamp(); 2883 mNBLogWriter->log(logString); 2884 logString = NULL; 2885 } 2886 2887 // Gather the framesReleased counters for all active tracks, 2888 // and associate with the sink frames written out. We need 2889 // this to convert the sink timestamp to the track timestamp. 2890 if (mNormalSink != 0) { 2891 // Note: The DuplicatingThread may not have a mNormalSink. 2892 // We always fetch the timestamp here because often the downstream 2893 // sink will block whie writing. 2894 ExtendedTimestamp timestamp; // use private copy to fetch 2895 (void) mNormalSink->getTimestamp(timestamp); 2896 // copy over kernel info 2897 mTimestamp.mPosition[ExtendedTimestamp::LOCATION_KERNEL] = 2898 timestamp.mPosition[ExtendedTimestamp::LOCATION_KERNEL]; 2899 mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL] = 2900 timestamp.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL]; 2901 } 2902 // mFramesWritten for non-offloaded tracks are contiguous 2903 // even after standby() is called. This is useful for the track frame 2904 // to sink frame mapping. 2905 mTimestamp.mPosition[ExtendedTimestamp::LOCATION_SERVER] = mFramesWritten; 2906 mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_SERVER] = systemTime(); 2907 const size_t size = mActiveTracks.size(); 2908 for (size_t i = 0; i < size; ++i) { 2909 sp<Track> t = mActiveTracks[i].promote(); 2910 if (t != 0 && !t->isFastTrack()) { 2911 t->updateTrackFrameInfo( 2912 t->mAudioTrackServerProxy->framesReleased(), 2913 mFramesWritten, 2914 mTimestamp); 2915 } 2916 } 2917 2918 saveOutputTracks(); 2919 if (mSignalPending) { 2920 // A signal was raised while we were unlocked 2921 mSignalPending = false; 2922 } else if (waitingAsyncCallback_l()) { 2923 if (exitPending()) { 2924 break; 2925 } 2926 bool released = false; 2927 // The following works around a bug in the offload driver. Ideally we would release 2928 // the wake lock every time, but that causes the last offload buffer(s) to be 2929 // dropped while the device is on battery, so we need to hold a wake lock during 2930 // the drain phase. 2931 if (mBytesRemaining && !(mDrainSequence & 1)) { 2932 releaseWakeLock_l(); 2933 released = true; 2934 } 2935 mWakeLockUids.clear(); 2936 mActiveTracksGeneration++; 2937 ALOGV("wait async completion"); 2938 mWaitWorkCV.wait(mLock); 2939 ALOGV("async completion/wake"); 2940 if (released) { 2941 acquireWakeLock_l(); 2942 } 2943 mStandbyTimeNs = systemTime() + mStandbyDelayNs; 2944 mSleepTimeUs = 0; 2945 2946 continue; 2947 } 2948 if ((!mActiveTracks.size() && systemTime() > mStandbyTimeNs) || 2949 isSuspended()) { 2950 // put audio hardware into standby after short delay 2951 if (shouldStandby_l()) { 2952 2953 threadLoop_standby(); 2954 2955 mStandby = true; 2956 } 2957 2958 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2959 // we're about to wait, flush the binder command buffer 2960 IPCThreadState::self()->flushCommands(); 2961 2962 clearOutputTracks(); 2963 2964 if (exitPending()) { 2965 break; 2966 } 2967 2968 releaseWakeLock_l(); 2969 mWakeLockUids.clear(); 2970 mActiveTracksGeneration++; 2971 // wait until we have something to do... 2972 ALOGV("%s going to sleep", myName.string()); 2973 mWaitWorkCV.wait(mLock); 2974 ALOGV("%s waking up", myName.string()); 2975 acquireWakeLock_l(); 2976 2977 mMixerStatus = MIXER_IDLE; 2978 mMixerStatusIgnoringFastTracks = MIXER_IDLE; 2979 mBytesWritten = 0; 2980 mBytesRemaining = 0; 2981 checkSilentMode_l(); 2982 2983 mStandbyTimeNs = systemTime() + mStandbyDelayNs; 2984 mSleepTimeUs = mIdleSleepTimeUs; 2985 if (mType == MIXER) { 2986 sleepTimeShift = 0; 2987 } 2988 2989 continue; 2990 } 2991 } 2992 // mMixerStatusIgnoringFastTracks is also updated internally 2993 mMixerStatus = prepareTracks_l(&tracksToRemove); 2994 2995 // compare with previously applied list 2996 if (lastGeneration != mActiveTracksGeneration) { 2997 // update wakelock 2998 updateWakeLockUids_l(mWakeLockUids); 2999 lastGeneration = mActiveTracksGeneration; 3000 } 3001 3002 // prevent any changes in effect chain list and in each effect chain 3003 // during mixing and effect process as the audio buffers could be deleted 3004 // or modified if an effect is created or deleted 3005 lockEffectChains_l(effectChains); 3006 } // mLock scope ends 3007 3008 if (mBytesRemaining == 0) { 3009 mCurrentWriteLength = 0; 3010 if (mMixerStatus == MIXER_TRACKS_READY) { 3011 // threadLoop_mix() sets mCurrentWriteLength 3012 threadLoop_mix(); 3013 } else if ((mMixerStatus != MIXER_DRAIN_TRACK) 3014 && (mMixerStatus != MIXER_DRAIN_ALL)) { 3015 // threadLoop_sleepTime sets mSleepTimeUs to 0 if data 3016 // must be written to HAL 3017 threadLoop_sleepTime(); 3018 if (mSleepTimeUs == 0) { 3019 mCurrentWriteLength = mSinkBufferSize; 3020 } 3021 } 3022 // Either threadLoop_mix() or threadLoop_sleepTime() should have set 3023 // mMixerBuffer with data if mMixerBufferValid is true and mSleepTimeUs == 0. 3024 // Merge mMixerBuffer data into mEffectBuffer (if any effects are valid) 3025 // or mSinkBuffer (if there are no effects). 3026 // 3027 // This is done pre-effects computation; if effects change to 3028 // support higher precision, this needs to move. 3029 // 3030 // mMixerBufferValid is only set true by MixerThread::prepareTracks_l(). 3031 // TODO use mSleepTimeUs == 0 as an additional condition. 3032 if (mMixerBufferValid) { 3033 void *buffer = mEffectBufferValid ? mEffectBuffer : mSinkBuffer; 3034 audio_format_t format = mEffectBufferValid ? mEffectBufferFormat : mFormat; 3035 3036 // mono blend occurs for mixer threads only (not direct or offloaded) 3037 // and is handled here if we're going directly to the sink. 3038 if (requireMonoBlend() && !mEffectBufferValid) { 3039 mono_blend(mMixerBuffer, mMixerBufferFormat, mChannelCount, mNormalFrameCount, 3040 true /*limit*/); 3041 } 3042 3043 memcpy_by_audio_format(buffer, format, mMixerBuffer, mMixerBufferFormat, 3044 mNormalFrameCount * mChannelCount); 3045 } 3046 3047 mBytesRemaining = mCurrentWriteLength; 3048 if (isSuspended()) { 3049 mSleepTimeUs = suspendSleepTimeUs(); 3050 // simulate write to HAL when suspended 3051 mBytesWritten += mSinkBufferSize; 3052 mFramesWritten += mSinkBufferSize / mFrameSize; 3053 mBytesRemaining = 0; 3054 } 3055 3056 // only process effects if we're going to write 3057 if (mSleepTimeUs == 0 && mType != OFFLOAD) { 3058 for (size_t i = 0; i < effectChains.size(); i ++) { 3059 effectChains[i]->process_l(); 3060 } 3061 } 3062 } 3063 // Process effect chains for offloaded thread even if no audio 3064 // was read from audio track: process only updates effect state 3065 // and thus does have to be synchronized with audio writes but may have 3066 // to be called while waiting for async write callback 3067 if (mType == OFFLOAD) { 3068 for (size_t i = 0; i < effectChains.size(); i ++) { 3069 effectChains[i]->process_l(); 3070 } 3071 } 3072 3073 // Only if the Effects buffer is enabled and there is data in the 3074 // Effects buffer (buffer valid), we need to 3075 // copy into the sink buffer. 3076 // TODO use mSleepTimeUs == 0 as an additional condition. 3077 if (mEffectBufferValid) { 3078 //ALOGV("writing effect buffer to sink buffer format %#x", mFormat); 3079 3080 if (requireMonoBlend()) { 3081 mono_blend(mEffectBuffer, mEffectBufferFormat, mChannelCount, mNormalFrameCount, 3082 true /*limit*/); 3083 } 3084 3085 memcpy_by_audio_format(mSinkBuffer, mFormat, mEffectBuffer, mEffectBufferFormat, 3086 mNormalFrameCount * mChannelCount); 3087 } 3088 3089 // enable changes in effect chain 3090 unlockEffectChains(effectChains); 3091 3092 if (!waitingAsyncCallback()) { 3093 // mSleepTimeUs == 0 means we must write to audio hardware 3094 if (mSleepTimeUs == 0) { 3095 ssize_t ret = 0; 3096 if (mBytesRemaining) { 3097 ret = threadLoop_write(); 3098 if (ret < 0) { 3099 mBytesRemaining = 0; 3100 } else { 3101 mBytesWritten += ret; 3102 mBytesRemaining -= ret; 3103 mFramesWritten += ret / mFrameSize; 3104 } 3105 } else if ((mMixerStatus == MIXER_DRAIN_TRACK) || 3106 (mMixerStatus == MIXER_DRAIN_ALL)) { 3107 threadLoop_drain(); 3108 } 3109 if (mType == MIXER && !mStandby) { 3110 // write blocked detection 3111 nsecs_t now = systemTime(); 3112 nsecs_t delta = now - mLastWriteTime; 3113 if (delta > maxPeriod) { 3114 mNumDelayedWrites++; 3115 if ((now - lastWarning) > kWarningThrottleNs) { 3116 ATRACE_NAME("underrun"); 3117 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 3118 (unsigned long long) ns2ms(delta), mNumDelayedWrites, this); 3119 lastWarning = now; 3120 } 3121 } 3122 3123 if (mThreadThrottle 3124 && mMixerStatus == MIXER_TRACKS_READY // we are mixing (active tracks) 3125 && ret > 0) { // we wrote something 3126 // Limit MixerThread data processing to no more than twice the 3127 // expected processing rate. 3128 // 3129 // This helps prevent underruns with NuPlayer and other applications 3130 // which may set up buffers that are close to the minimum size, or use 3131 // deep buffers, and rely on a double-buffering sleep strategy to fill. 3132 // 3133 // The throttle smooths out sudden large data drains from the device, 3134 // e.g. when it comes out of standby, which often causes problems with 3135 // (1) mixer threads without a fast mixer (which has its own warm-up) 3136 // (2) minimum buffer sized tracks (even if the track is full, 3137 // the app won't fill fast enough to handle the sudden draw). 3138 3139 const int32_t deltaMs = delta / 1000000; 3140 const int32_t throttleMs = mHalfBufferMs - deltaMs; 3141 if ((signed)mHalfBufferMs >= throttleMs && throttleMs > 0) { 3142 usleep(throttleMs * 1000); 3143 // notify of throttle start on verbose log 3144 ALOGV_IF(mThreadThrottleEndMs == mThreadThrottleTimeMs, 3145 "mixer(%p) throttle begin:" 3146 " ret(%zd) deltaMs(%d) requires sleep %d ms", 3147 this, ret, deltaMs, throttleMs); 3148 mThreadThrottleTimeMs += throttleMs; 3149 } else { 3150 uint32_t diff = mThreadThrottleTimeMs - mThreadThrottleEndMs; 3151 if (diff > 0) { 3152 // notify of throttle end on debug log 3153 ALOGD("mixer(%p) throttle end: throttle time(%u)", this, diff); 3154 mThreadThrottleEndMs = mThreadThrottleTimeMs; 3155 } 3156 } 3157 } 3158 } 3159 3160 } else { 3161 ATRACE_BEGIN("sleep"); 3162 if ((mType == OFFLOAD) && !audio_has_proportional_frames(mFormat)) { 3163 Mutex::Autolock _l(mLock); 3164 if (!mSignalPending && !exitPending()) { 3165 // If more than one buffer has been written to the audio HAL since exiting 3166 // standby or last flush, do not sleep more than one buffer duration 3167 // since last write and not less than kDirectMinSleepTimeUs. 3168 // Wake up if a command is received 3169 uint32_t timeoutUs = mSleepTimeUs; 3170 if (mBytesWritten >= (int64_t) mBufferSize) { 3171 nsecs_t now = systemTime(); 3172 uint32_t deltaUs = (uint32_t)((now - mLastWriteTime) / 1000); 3173 if (timeoutUs + deltaUs > mBufferDurationUs) { 3174 if (mBufferDurationUs > deltaUs) { 3175 timeoutUs = mBufferDurationUs - deltaUs; 3176 if (timeoutUs < kDirectMinSleepTimeUs) { 3177 timeoutUs = kDirectMinSleepTimeUs; 3178 } 3179 } else { 3180 timeoutUs = kDirectMinSleepTimeUs; 3181 } 3182 } 3183 } 3184 mWaitWorkCV.waitRelative(mLock, microseconds((nsecs_t)timeoutUs)); 3185 } 3186 } else { 3187 usleep(mSleepTimeUs); 3188 } 3189 ATRACE_END(); 3190 } 3191 } 3192 3193 // Finally let go of removed track(s), without the lock held 3194 // since we can't guarantee the destructors won't acquire that 3195 // same lock. This will also mutate and push a new fast mixer state. 3196 threadLoop_removeTracks(tracksToRemove); 3197 tracksToRemove.clear(); 3198 3199 // FIXME I don't understand the need for this here; 3200 // it was in the original code but maybe the 3201 // assignment in saveOutputTracks() makes this unnecessary? 3202 clearOutputTracks(); 3203 3204 // Effect chains will be actually deleted here if they were removed from 3205 // mEffectChains list during mixing or effects processing 3206 effectChains.clear(); 3207 3208 // FIXME Note that the above .clear() is no longer necessary since effectChains 3209 // is now local to this block, but will keep it for now (at least until merge done). 3210 } 3211 3212 threadLoop_exit(); 3213 3214 if (!mStandby) { 3215 threadLoop_standby(); 3216 mStandby = true; 3217 } 3218 3219 releaseWakeLock(); 3220 mWakeLockUids.clear(); 3221 mActiveTracksGeneration++; 3222 3223 ALOGV("Thread %p type %d exiting", this, mType); 3224 return false; 3225} 3226 3227// removeTracks_l() must be called with ThreadBase::mLock held 3228void AudioFlinger::PlaybackThread::removeTracks_l(const Vector< sp<Track> >& tracksToRemove) 3229{ 3230 size_t count = tracksToRemove.size(); 3231 if (count > 0) { 3232 for (size_t i=0 ; i<count ; i++) { 3233 const sp<Track>& track = tracksToRemove.itemAt(i); 3234 mActiveTracks.remove(track); 3235 mWakeLockUids.remove(track->uid()); 3236 mActiveTracksGeneration++; 3237 ALOGV("removeTracks_l removing track on session %d", track->sessionId()); 3238 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 3239 if (chain != 0) { 3240 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), 3241 track->sessionId()); 3242 chain->decActiveTrackCnt(); 3243 } 3244 if (track->isTerminated()) { 3245 removeTrack_l(track); 3246 } 3247 } 3248 } 3249 3250} 3251 3252status_t AudioFlinger::PlaybackThread::getTimestamp_l(AudioTimestamp& timestamp) 3253{ 3254 if (mNormalSink != 0) { 3255 ExtendedTimestamp ets; 3256 status_t status = mNormalSink->getTimestamp(ets); 3257 if (status == NO_ERROR) { 3258 status = ets.getBestTimestamp(×tamp); 3259 } 3260 return status; 3261 } 3262 if ((mType == OFFLOAD || mType == DIRECT) 3263 && mOutput != NULL && mOutput->stream->get_presentation_position) { 3264 uint64_t position64; 3265 int ret = mOutput->getPresentationPosition(&position64, ×tamp.mTime); 3266 if (ret == 0) { 3267 timestamp.mPosition = (uint32_t)position64; 3268 return NO_ERROR; 3269 } 3270 } 3271 return INVALID_OPERATION; 3272} 3273 3274status_t AudioFlinger::MixerThread::createAudioPatch_l(const struct audio_patch *patch, 3275 audio_patch_handle_t *handle) 3276{ 3277 AutoPark<FastMixer> park(mFastMixer); 3278 3279 status_t status = PlaybackThread::createAudioPatch_l(patch, handle); 3280 3281 return status; 3282} 3283 3284status_t AudioFlinger::PlaybackThread::createAudioPatch_l(const struct audio_patch *patch, 3285 audio_patch_handle_t *handle) 3286{ 3287 status_t status = NO_ERROR; 3288 3289 // store new device and send to effects 3290 audio_devices_t type = AUDIO_DEVICE_NONE; 3291 for (unsigned int i = 0; i < patch->num_sinks; i++) { 3292 type |= patch->sinks[i].ext.device.type; 3293 } 3294 3295#ifdef ADD_BATTERY_DATA 3296 // when changing the audio output device, call addBatteryData to notify 3297 // the change 3298 if (mOutDevice != type) { 3299 uint32_t params = 0; 3300 // check whether speaker is on 3301 if (type & AUDIO_DEVICE_OUT_SPEAKER) { 3302 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 3303 } 3304 3305 audio_devices_t deviceWithoutSpeaker 3306 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 3307 // check if any other device (except speaker) is on 3308 if (type & deviceWithoutSpeaker) { 3309 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 3310 } 3311 3312 if (params != 0) { 3313 addBatteryData(params); 3314 } 3315 } 3316#endif 3317 3318 for (size_t i = 0; i < mEffectChains.size(); i++) { 3319 mEffectChains[i]->setDevice_l(type); 3320 } 3321 3322 // mPrevOutDevice is the latest device set by createAudioPatch_l(). It is not set when 3323 // the thread is created so that the first patch creation triggers an ioConfigChanged callback 3324 bool configChanged = mPrevOutDevice != type; 3325 mOutDevice = type; 3326 mPatch = *patch; 3327 3328 if (mOutput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 3329 audio_hw_device_t *hwDevice = mOutput->audioHwDev->hwDevice(); 3330 status = hwDevice->create_audio_patch(hwDevice, 3331 patch->num_sources, 3332 patch->sources, 3333 patch->num_sinks, 3334 patch->sinks, 3335 handle); 3336 } else { 3337 char *address; 3338 if (strcmp(patch->sinks[0].ext.device.address, "") != 0) { 3339 //FIXME: we only support address on first sink with HAL version < 3.0 3340 address = audio_device_address_to_parameter( 3341 patch->sinks[0].ext.device.type, 3342 patch->sinks[0].ext.device.address); 3343 } else { 3344 address = (char *)calloc(1, 1); 3345 } 3346 AudioParameter param = AudioParameter(String8(address)); 3347 free(address); 3348 param.addInt(String8(AUDIO_PARAMETER_STREAM_ROUTING), (int)type); 3349 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3350 param.toString().string()); 3351 *handle = AUDIO_PATCH_HANDLE_NONE; 3352 } 3353 if (configChanged) { 3354 mPrevOutDevice = type; 3355 sendIoConfigEvent_l(AUDIO_OUTPUT_CONFIG_CHANGED); 3356 } 3357 return status; 3358} 3359 3360status_t AudioFlinger::MixerThread::releaseAudioPatch_l(const audio_patch_handle_t handle) 3361{ 3362 AutoPark<FastMixer> park(mFastMixer); 3363 3364 status_t status = PlaybackThread::releaseAudioPatch_l(handle); 3365 3366 return status; 3367} 3368 3369status_t AudioFlinger::PlaybackThread::releaseAudioPatch_l(const audio_patch_handle_t handle) 3370{ 3371 status_t status = NO_ERROR; 3372 3373 mOutDevice = AUDIO_DEVICE_NONE; 3374 3375 if (mOutput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 3376 audio_hw_device_t *hwDevice = mOutput->audioHwDev->hwDevice(); 3377 status = hwDevice->release_audio_patch(hwDevice, handle); 3378 } else { 3379 AudioParameter param; 3380 param.addInt(String8(AUDIO_PARAMETER_STREAM_ROUTING), 0); 3381 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3382 param.toString().string()); 3383 } 3384 return status; 3385} 3386 3387void AudioFlinger::PlaybackThread::addPatchTrack(const sp<PatchTrack>& track) 3388{ 3389 Mutex::Autolock _l(mLock); 3390 mTracks.add(track); 3391} 3392 3393void AudioFlinger::PlaybackThread::deletePatchTrack(const sp<PatchTrack>& track) 3394{ 3395 Mutex::Autolock _l(mLock); 3396 destroyTrack_l(track); 3397} 3398 3399void AudioFlinger::PlaybackThread::getAudioPortConfig(struct audio_port_config *config) 3400{ 3401 ThreadBase::getAudioPortConfig(config); 3402 config->role = AUDIO_PORT_ROLE_SOURCE; 3403 config->ext.mix.hw_module = mOutput->audioHwDev->handle(); 3404 config->ext.mix.usecase.stream = AUDIO_STREAM_DEFAULT; 3405} 3406 3407// ---------------------------------------------------------------------------- 3408 3409AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 3410 audio_io_handle_t id, audio_devices_t device, bool systemReady, type_t type) 3411 : PlaybackThread(audioFlinger, output, id, device, type, systemReady), 3412 // mAudioMixer below 3413 // mFastMixer below 3414 mFastMixerFutex(0), 3415 mMasterMono(false) 3416 // mOutputSink below 3417 // mPipeSink below 3418 // mNormalSink below 3419{ 3420 ALOGV("MixerThread() id=%d device=%#x type=%d", id, device, type); 3421 ALOGV("mSampleRate=%u, mChannelMask=%#x, mChannelCount=%u, mFormat=%d, mFrameSize=%zu, " 3422 "mFrameCount=%zu, mNormalFrameCount=%zu", 3423 mSampleRate, mChannelMask, mChannelCount, mFormat, mFrameSize, mFrameCount, 3424 mNormalFrameCount); 3425 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 3426 3427 if (type == DUPLICATING) { 3428 // The Duplicating thread uses the AudioMixer and delivers data to OutputTracks 3429 // (downstream MixerThreads) in DuplicatingThread::threadLoop_write(). 3430 // Do not create or use mFastMixer, mOutputSink, mPipeSink, or mNormalSink. 3431 return; 3432 } 3433 // create an NBAIO sink for the HAL output stream, and negotiate 3434 mOutputSink = new AudioStreamOutSink(output->stream); 3435 size_t numCounterOffers = 0; 3436 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount, mFormat)}; 3437#if !LOG_NDEBUG 3438 ssize_t index = 3439#else 3440 (void) 3441#endif 3442 mOutputSink->negotiate(offers, 1, NULL, numCounterOffers); 3443 ALOG_ASSERT(index == 0); 3444 3445 // initialize fast mixer depending on configuration 3446 bool initFastMixer; 3447 switch (kUseFastMixer) { 3448 case FastMixer_Never: 3449 initFastMixer = false; 3450 break; 3451 case FastMixer_Always: 3452 initFastMixer = true; 3453 break; 3454 case FastMixer_Static: 3455 case FastMixer_Dynamic: 3456 initFastMixer = mFrameCount < mNormalFrameCount; 3457 break; 3458 } 3459 if (initFastMixer) { 3460 audio_format_t fastMixerFormat; 3461 if (mMixerBufferEnabled && mEffectBufferEnabled) { 3462 fastMixerFormat = AUDIO_FORMAT_PCM_FLOAT; 3463 } else { 3464 fastMixerFormat = AUDIO_FORMAT_PCM_16_BIT; 3465 } 3466 if (mFormat != fastMixerFormat) { 3467 // change our Sink format to accept our intermediate precision 3468 mFormat = fastMixerFormat; 3469 free(mSinkBuffer); 3470 mFrameSize = mChannelCount * audio_bytes_per_sample(mFormat); 3471 const size_t sinkBufferSize = mNormalFrameCount * mFrameSize; 3472 (void)posix_memalign(&mSinkBuffer, 32, sinkBufferSize); 3473 } 3474 3475 // create a MonoPipe to connect our submix to FastMixer 3476 NBAIO_Format format = mOutputSink->format(); 3477#ifdef TEE_SINK 3478 NBAIO_Format origformat = format; 3479#endif 3480 // adjust format to match that of the Fast Mixer 3481 ALOGV("format changed from %d to %d", format.mFormat, fastMixerFormat); 3482 format.mFormat = fastMixerFormat; 3483 format.mFrameSize = audio_bytes_per_sample(format.mFormat) * format.mChannelCount; 3484 3485 // This pipe depth compensates for scheduling latency of the normal mixer thread. 3486 // When it wakes up after a maximum latency, it runs a few cycles quickly before 3487 // finally blocking. Note the pipe implementation rounds up the request to a power of 2. 3488 MonoPipe *monoPipe = new MonoPipe(mNormalFrameCount * 4, format, true /*writeCanBlock*/); 3489 const NBAIO_Format offers[1] = {format}; 3490 size_t numCounterOffers = 0; 3491#if !LOG_NDEBUG 3492 ssize_t index = 3493#else 3494 (void) 3495#endif 3496 monoPipe->negotiate(offers, 1, NULL, numCounterOffers); 3497 ALOG_ASSERT(index == 0); 3498 monoPipe->setAvgFrames((mScreenState & 1) ? 3499 (monoPipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 3500 mPipeSink = monoPipe; 3501 3502#ifdef TEE_SINK 3503 if (mTeeSinkOutputEnabled) { 3504 // create a Pipe to archive a copy of FastMixer's output for dumpsys 3505 Pipe *teeSink = new Pipe(mTeeSinkOutputFrames, origformat); 3506 const NBAIO_Format offers2[1] = {origformat}; 3507 numCounterOffers = 0; 3508 index = teeSink->negotiate(offers2, 1, NULL, numCounterOffers); 3509 ALOG_ASSERT(index == 0); 3510 mTeeSink = teeSink; 3511 PipeReader *teeSource = new PipeReader(*teeSink); 3512 numCounterOffers = 0; 3513 index = teeSource->negotiate(offers2, 1, NULL, numCounterOffers); 3514 ALOG_ASSERT(index == 0); 3515 mTeeSource = teeSource; 3516 } 3517#endif 3518 3519 // create fast mixer and configure it initially with just one fast track for our submix 3520 mFastMixer = new FastMixer(); 3521 FastMixerStateQueue *sq = mFastMixer->sq(); 3522#ifdef STATE_QUEUE_DUMP 3523 sq->setObserverDump(&mStateQueueObserverDump); 3524 sq->setMutatorDump(&mStateQueueMutatorDump); 3525#endif 3526 FastMixerState *state = sq->begin(); 3527 FastTrack *fastTrack = &state->mFastTracks[0]; 3528 // wrap the source side of the MonoPipe to make it an AudioBufferProvider 3529 fastTrack->mBufferProvider = new SourceAudioBufferProvider(new MonoPipeReader(monoPipe)); 3530 fastTrack->mVolumeProvider = NULL; 3531 fastTrack->mChannelMask = mChannelMask; // mPipeSink channel mask for audio to FastMixer 3532 fastTrack->mFormat = mFormat; // mPipeSink format for audio to FastMixer 3533 fastTrack->mGeneration++; 3534 state->mFastTracksGen++; 3535 state->mTrackMask = 1; 3536 // fast mixer will use the HAL output sink 3537 state->mOutputSink = mOutputSink.get(); 3538 state->mOutputSinkGen++; 3539 state->mFrameCount = mFrameCount; 3540 state->mCommand = FastMixerState::COLD_IDLE; 3541 // already done in constructor initialization list 3542 //mFastMixerFutex = 0; 3543 state->mColdFutexAddr = &mFastMixerFutex; 3544 state->mColdGen++; 3545 state->mDumpState = &mFastMixerDumpState; 3546#ifdef TEE_SINK 3547 state->mTeeSink = mTeeSink.get(); 3548#endif 3549 mFastMixerNBLogWriter = audioFlinger->newWriter_l(kFastMixerLogSize, "FastMixer"); 3550 state->mNBLogWriter = mFastMixerNBLogWriter.get(); 3551 sq->end(); 3552 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3553 3554 // start the fast mixer 3555 mFastMixer->run("FastMixer", PRIORITY_URGENT_AUDIO); 3556 pid_t tid = mFastMixer->getTid(); 3557 sendPrioConfigEvent(getpid_cached, tid, kPriorityFastMixer); 3558 3559#ifdef AUDIO_WATCHDOG 3560 // create and start the watchdog 3561 mAudioWatchdog = new AudioWatchdog(); 3562 mAudioWatchdog->setDump(&mAudioWatchdogDump); 3563 mAudioWatchdog->run("AudioWatchdog", PRIORITY_URGENT_AUDIO); 3564 tid = mAudioWatchdog->getTid(); 3565 sendPrioConfigEvent(getpid_cached, tid, kPriorityFastMixer); 3566#endif 3567 3568 } 3569 3570 switch (kUseFastMixer) { 3571 case FastMixer_Never: 3572 case FastMixer_Dynamic: 3573 mNormalSink = mOutputSink; 3574 break; 3575 case FastMixer_Always: 3576 mNormalSink = mPipeSink; 3577 break; 3578 case FastMixer_Static: 3579 mNormalSink = initFastMixer ? mPipeSink : mOutputSink; 3580 break; 3581 } 3582} 3583 3584AudioFlinger::MixerThread::~MixerThread() 3585{ 3586 if (mFastMixer != 0) { 3587 FastMixerStateQueue *sq = mFastMixer->sq(); 3588 FastMixerState *state = sq->begin(); 3589 if (state->mCommand == FastMixerState::COLD_IDLE) { 3590 int32_t old = android_atomic_inc(&mFastMixerFutex); 3591 if (old == -1) { 3592 (void) syscall(__NR_futex, &mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 3593 } 3594 } 3595 state->mCommand = FastMixerState::EXIT; 3596 sq->end(); 3597 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3598 mFastMixer->join(); 3599 // Though the fast mixer thread has exited, it's state queue is still valid. 3600 // We'll use that extract the final state which contains one remaining fast track 3601 // corresponding to our sub-mix. 3602 state = sq->begin(); 3603 ALOG_ASSERT(state->mTrackMask == 1); 3604 FastTrack *fastTrack = &state->mFastTracks[0]; 3605 ALOG_ASSERT(fastTrack->mBufferProvider != NULL); 3606 delete fastTrack->mBufferProvider; 3607 sq->end(false /*didModify*/); 3608 mFastMixer.clear(); 3609#ifdef AUDIO_WATCHDOG 3610 if (mAudioWatchdog != 0) { 3611 mAudioWatchdog->requestExit(); 3612 mAudioWatchdog->requestExitAndWait(); 3613 mAudioWatchdog.clear(); 3614 } 3615#endif 3616 } 3617 mAudioFlinger->unregisterWriter(mFastMixerNBLogWriter); 3618 delete mAudioMixer; 3619} 3620 3621 3622uint32_t AudioFlinger::MixerThread::correctLatency_l(uint32_t latency) const 3623{ 3624 if (mFastMixer != 0) { 3625 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 3626 latency += (pipe->getAvgFrames() * 1000) / mSampleRate; 3627 } 3628 return latency; 3629} 3630 3631 3632void AudioFlinger::MixerThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) 3633{ 3634 PlaybackThread::threadLoop_removeTracks(tracksToRemove); 3635} 3636 3637ssize_t AudioFlinger::MixerThread::threadLoop_write() 3638{ 3639 // FIXME we should only do one push per cycle; confirm this is true 3640 // Start the fast mixer if it's not already running 3641 if (mFastMixer != 0) { 3642 FastMixerStateQueue *sq = mFastMixer->sq(); 3643 FastMixerState *state = sq->begin(); 3644 if (state->mCommand != FastMixerState::MIX_WRITE && 3645 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)) { 3646 if (state->mCommand == FastMixerState::COLD_IDLE) { 3647 3648 // FIXME workaround for first HAL write being CPU bound on some devices 3649 ATRACE_BEGIN("write"); 3650 mOutput->write((char *)mSinkBuffer, 0); 3651 ATRACE_END(); 3652 3653 int32_t old = android_atomic_inc(&mFastMixerFutex); 3654 if (old == -1) { 3655 (void) syscall(__NR_futex, &mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 3656 } 3657#ifdef AUDIO_WATCHDOG 3658 if (mAudioWatchdog != 0) { 3659 mAudioWatchdog->resume(); 3660 } 3661#endif 3662 } 3663 state->mCommand = FastMixerState::MIX_WRITE; 3664#ifdef FAST_THREAD_STATISTICS 3665 mFastMixerDumpState.increaseSamplingN(mAudioFlinger->isLowRamDevice() ? 3666 FastThreadDumpState::kSamplingNforLowRamDevice : FastThreadDumpState::kSamplingN); 3667#endif 3668 sq->end(); 3669 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3670 if (kUseFastMixer == FastMixer_Dynamic) { 3671 mNormalSink = mPipeSink; 3672 } 3673 } else { 3674 sq->end(false /*didModify*/); 3675 } 3676 } 3677 return PlaybackThread::threadLoop_write(); 3678} 3679 3680void AudioFlinger::MixerThread::threadLoop_standby() 3681{ 3682 // Idle the fast mixer if it's currently running 3683 if (mFastMixer != 0) { 3684 FastMixerStateQueue *sq = mFastMixer->sq(); 3685 FastMixerState *state = sq->begin(); 3686 if (!(state->mCommand & FastMixerState::IDLE)) { 3687 state->mCommand = FastMixerState::COLD_IDLE; 3688 state->mColdFutexAddr = &mFastMixerFutex; 3689 state->mColdGen++; 3690 mFastMixerFutex = 0; 3691 sq->end(); 3692 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 3693 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3694 if (kUseFastMixer == FastMixer_Dynamic) { 3695 mNormalSink = mOutputSink; 3696 } 3697#ifdef AUDIO_WATCHDOG 3698 if (mAudioWatchdog != 0) { 3699 mAudioWatchdog->pause(); 3700 } 3701#endif 3702 } else { 3703 sq->end(false /*didModify*/); 3704 } 3705 } 3706 PlaybackThread::threadLoop_standby(); 3707} 3708 3709bool AudioFlinger::PlaybackThread::waitingAsyncCallback_l() 3710{ 3711 return false; 3712} 3713 3714bool AudioFlinger::PlaybackThread::shouldStandby_l() 3715{ 3716 return !mStandby; 3717} 3718 3719bool AudioFlinger::PlaybackThread::waitingAsyncCallback() 3720{ 3721 Mutex::Autolock _l(mLock); 3722 return waitingAsyncCallback_l(); 3723} 3724 3725// shared by MIXER and DIRECT, overridden by DUPLICATING 3726void AudioFlinger::PlaybackThread::threadLoop_standby() 3727{ 3728 ALOGV("Audio hardware entering standby, mixer %p, suspend count %d", this, mSuspended); 3729 mOutput->standby(); 3730 if (mUseAsyncWrite != 0) { 3731 // discard any pending drain or write ack by incrementing sequence 3732 mWriteAckSequence = (mWriteAckSequence + 2) & ~1; 3733 mDrainSequence = (mDrainSequence + 2) & ~1; 3734 ALOG_ASSERT(mCallbackThread != 0); 3735 mCallbackThread->setWriteBlocked(mWriteAckSequence); 3736 mCallbackThread->setDraining(mDrainSequence); 3737 } 3738 mHwPaused = false; 3739} 3740 3741void AudioFlinger::PlaybackThread::onAddNewTrack_l() 3742{ 3743 ALOGV("signal playback thread"); 3744 broadcast_l(); 3745} 3746 3747void AudioFlinger::MixerThread::threadLoop_mix() 3748{ 3749 // mix buffers... 3750 mAudioMixer->process(); 3751 mCurrentWriteLength = mSinkBufferSize; 3752 // increase sleep time progressively when application underrun condition clears. 3753 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 3754 // that a steady state of alternating ready/not ready conditions keeps the sleep time 3755 // such that we would underrun the audio HAL. 3756 if ((mSleepTimeUs == 0) && (sleepTimeShift > 0)) { 3757 sleepTimeShift--; 3758 } 3759 mSleepTimeUs = 0; 3760 mStandbyTimeNs = systemTime() + mStandbyDelayNs; 3761 //TODO: delay standby when effects have a tail 3762 3763} 3764 3765void AudioFlinger::MixerThread::threadLoop_sleepTime() 3766{ 3767 // If no tracks are ready, sleep once for the duration of an output 3768 // buffer size, then write 0s to the output 3769 if (mSleepTimeUs == 0) { 3770 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3771 mSleepTimeUs = mActiveSleepTimeUs >> sleepTimeShift; 3772 if (mSleepTimeUs < kMinThreadSleepTimeUs) { 3773 mSleepTimeUs = kMinThreadSleepTimeUs; 3774 } 3775 // reduce sleep time in case of consecutive application underruns to avoid 3776 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 3777 // duration we would end up writing less data than needed by the audio HAL if 3778 // the condition persists. 3779 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 3780 sleepTimeShift++; 3781 } 3782 } else { 3783 mSleepTimeUs = mIdleSleepTimeUs; 3784 } 3785 } else if (mBytesWritten != 0 || (mMixerStatus == MIXER_TRACKS_ENABLED)) { 3786 // clear out mMixerBuffer or mSinkBuffer, to ensure buffers are cleared 3787 // before effects processing or output. 3788 if (mMixerBufferValid) { 3789 memset(mMixerBuffer, 0, mMixerBufferSize); 3790 } else { 3791 memset(mSinkBuffer, 0, mSinkBufferSize); 3792 } 3793 mSleepTimeUs = 0; 3794 ALOGV_IF(mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED), 3795 "anticipated start"); 3796 } 3797 // TODO add standby time extension fct of effect tail 3798} 3799 3800// prepareTracks_l() must be called with ThreadBase::mLock held 3801AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 3802 Vector< sp<Track> > *tracksToRemove) 3803{ 3804 3805 mixer_state mixerStatus = MIXER_IDLE; 3806 // find out which tracks need to be processed 3807 size_t count = mActiveTracks.size(); 3808 size_t mixedTracks = 0; 3809 size_t tracksWithEffect = 0; 3810 // counts only _active_ fast tracks 3811 size_t fastTracks = 0; 3812 uint32_t resetMask = 0; // bit mask of fast tracks that need to be reset 3813 3814 float masterVolume = mMasterVolume; 3815 bool masterMute = mMasterMute; 3816 3817 if (masterMute) { 3818 masterVolume = 0; 3819 } 3820 // Delegate master volume control to effect in output mix effect chain if needed 3821 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 3822 if (chain != 0) { 3823 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 3824 chain->setVolume_l(&v, &v); 3825 masterVolume = (float)((v + (1 << 23)) >> 24); 3826 chain.clear(); 3827 } 3828 3829 // prepare a new state to push 3830 FastMixerStateQueue *sq = NULL; 3831 FastMixerState *state = NULL; 3832 bool didModify = false; 3833 FastMixerStateQueue::block_t block = FastMixerStateQueue::BLOCK_UNTIL_PUSHED; 3834 if (mFastMixer != 0) { 3835 sq = mFastMixer->sq(); 3836 state = sq->begin(); 3837 } 3838 3839 mMixerBufferValid = false; // mMixerBuffer has no valid data until appropriate tracks found. 3840 mEffectBufferValid = false; // mEffectBuffer has no valid data until tracks found. 3841 3842 for (size_t i=0 ; i<count ; i++) { 3843 const sp<Track> t = mActiveTracks[i].promote(); 3844 if (t == 0) { 3845 continue; 3846 } 3847 3848 // this const just means the local variable doesn't change 3849 Track* const track = t.get(); 3850 3851 // process fast tracks 3852 if (track->isFastTrack()) { 3853 3854 // It's theoretically possible (though unlikely) for a fast track to be created 3855 // and then removed within the same normal mix cycle. This is not a problem, as 3856 // the track never becomes active so it's fast mixer slot is never touched. 3857 // The converse, of removing an (active) track and then creating a new track 3858 // at the identical fast mixer slot within the same normal mix cycle, 3859 // is impossible because the slot isn't marked available until the end of each cycle. 3860 int j = track->mFastIndex; 3861 ALOG_ASSERT(0 < j && j < (int)FastMixerState::kMaxFastTracks); 3862 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << j))); 3863 FastTrack *fastTrack = &state->mFastTracks[j]; 3864 3865 // Determine whether the track is currently in underrun condition, 3866 // and whether it had a recent underrun. 3867 FastTrackDump *ftDump = &mFastMixerDumpState.mTracks[j]; 3868 FastTrackUnderruns underruns = ftDump->mUnderruns; 3869 uint32_t recentFull = (underruns.mBitFields.mFull - 3870 track->mObservedUnderruns.mBitFields.mFull) & UNDERRUN_MASK; 3871 uint32_t recentPartial = (underruns.mBitFields.mPartial - 3872 track->mObservedUnderruns.mBitFields.mPartial) & UNDERRUN_MASK; 3873 uint32_t recentEmpty = (underruns.mBitFields.mEmpty - 3874 track->mObservedUnderruns.mBitFields.mEmpty) & UNDERRUN_MASK; 3875 uint32_t recentUnderruns = recentPartial + recentEmpty; 3876 track->mObservedUnderruns = underruns; 3877 // don't count underruns that occur while stopping or pausing 3878 // or stopped which can occur when flush() is called while active 3879 if (!(track->isStopping() || track->isPausing() || track->isStopped()) && 3880 recentUnderruns > 0) { 3881 // FIXME fast mixer will pull & mix partial buffers, but we count as a full underrun 3882 track->mAudioTrackServerProxy->tallyUnderrunFrames(recentUnderruns * mFrameCount); 3883 } else { 3884 track->mAudioTrackServerProxy->tallyUnderrunFrames(0); 3885 } 3886 3887 // This is similar to the state machine for normal tracks, 3888 // with a few modifications for fast tracks. 3889 bool isActive = true; 3890 switch (track->mState) { 3891 case TrackBase::STOPPING_1: 3892 // track stays active in STOPPING_1 state until first underrun 3893 if (recentUnderruns > 0 || track->isTerminated()) { 3894 track->mState = TrackBase::STOPPING_2; 3895 } 3896 break; 3897 case TrackBase::PAUSING: 3898 // ramp down is not yet implemented 3899 track->setPaused(); 3900 break; 3901 case TrackBase::RESUMING: 3902 // ramp up is not yet implemented 3903 track->mState = TrackBase::ACTIVE; 3904 break; 3905 case TrackBase::ACTIVE: 3906 if (recentFull > 0 || recentPartial > 0) { 3907 // track has provided at least some frames recently: reset retry count 3908 track->mRetryCount = kMaxTrackRetries; 3909 } 3910 if (recentUnderruns == 0) { 3911 // no recent underruns: stay active 3912 break; 3913 } 3914 // there has recently been an underrun of some kind 3915 if (track->sharedBuffer() == 0) { 3916 // were any of the recent underruns "empty" (no frames available)? 3917 if (recentEmpty == 0) { 3918 // no, then ignore the partial underruns as they are allowed indefinitely 3919 break; 3920 } 3921 // there has recently been an "empty" underrun: decrement the retry counter 3922 if (--(track->mRetryCount) > 0) { 3923 break; 3924 } 3925 // indicate to client process that the track was disabled because of underrun; 3926 // it will then automatically call start() when data is available 3927 track->disable(); 3928 // remove from active list, but state remains ACTIVE [confusing but true] 3929 isActive = false; 3930 break; 3931 } 3932 // fall through 3933 case TrackBase::STOPPING_2: 3934 case TrackBase::PAUSED: 3935 case TrackBase::STOPPED: 3936 case TrackBase::FLUSHED: // flush() while active 3937 // Check for presentation complete if track is inactive 3938 // We have consumed all the buffers of this track. 3939 // This would be incomplete if we auto-paused on underrun 3940 { 3941 size_t audioHALFrames = 3942 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 3943 int64_t framesWritten = mBytesWritten / mFrameSize; 3944 if (!(mStandby || track->presentationComplete(framesWritten, audioHALFrames))) { 3945 // track stays in active list until presentation is complete 3946 break; 3947 } 3948 } 3949 if (track->isStopping_2()) { 3950 track->mState = TrackBase::STOPPED; 3951 } 3952 if (track->isStopped()) { 3953 // Can't reset directly, as fast mixer is still polling this track 3954 // track->reset(); 3955 // So instead mark this track as needing to be reset after push with ack 3956 resetMask |= 1 << i; 3957 } 3958 isActive = false; 3959 break; 3960 case TrackBase::IDLE: 3961 default: 3962 LOG_ALWAYS_FATAL("unexpected track state %d", track->mState); 3963 } 3964 3965 if (isActive) { 3966 // was it previously inactive? 3967 if (!(state->mTrackMask & (1 << j))) { 3968 ExtendedAudioBufferProvider *eabp = track; 3969 VolumeProvider *vp = track; 3970 fastTrack->mBufferProvider = eabp; 3971 fastTrack->mVolumeProvider = vp; 3972 fastTrack->mChannelMask = track->mChannelMask; 3973 fastTrack->mFormat = track->mFormat; 3974 fastTrack->mGeneration++; 3975 state->mTrackMask |= 1 << j; 3976 didModify = true; 3977 // no acknowledgement required for newly active tracks 3978 } 3979 // cache the combined master volume and stream type volume for fast mixer; this 3980 // lacks any synchronization or barrier so VolumeProvider may read a stale value 3981 track->mCachedVolume = masterVolume * mStreamTypes[track->streamType()].volume; 3982 ++fastTracks; 3983 } else { 3984 // was it previously active? 3985 if (state->mTrackMask & (1 << j)) { 3986 fastTrack->mBufferProvider = NULL; 3987 fastTrack->mGeneration++; 3988 state->mTrackMask &= ~(1 << j); 3989 didModify = true; 3990 // If any fast tracks were removed, we must wait for acknowledgement 3991 // because we're about to decrement the last sp<> on those tracks. 3992 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3993 } else { 3994 LOG_ALWAYS_FATAL("fast track %d should have been active; " 3995 "mState=%d, mTrackMask=%#x, recentUnderruns=%u, isShared=%d", 3996 j, track->mState, state->mTrackMask, recentUnderruns, 3997 track->sharedBuffer() != 0); 3998 } 3999 tracksToRemove->add(track); 4000 // Avoids a misleading display in dumpsys 4001 track->mObservedUnderruns.mBitFields.mMostRecent = UNDERRUN_FULL; 4002 } 4003 continue; 4004 } 4005 4006 { // local variable scope to avoid goto warning 4007 4008 audio_track_cblk_t* cblk = track->cblk(); 4009 4010 // The first time a track is added we wait 4011 // for all its buffers to be filled before processing it 4012 int name = track->name(); 4013 // make sure that we have enough frames to mix one full buffer. 4014 // enforce this condition only once to enable draining the buffer in case the client 4015 // app does not call stop() and relies on underrun to stop: 4016 // hence the test on (mMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 4017 // during last round 4018 size_t desiredFrames; 4019 const uint32_t sampleRate = track->mAudioTrackServerProxy->getSampleRate(); 4020 AudioPlaybackRate playbackRate = track->mAudioTrackServerProxy->getPlaybackRate(); 4021 4022 desiredFrames = sourceFramesNeededWithTimestretch( 4023 sampleRate, mNormalFrameCount, mSampleRate, playbackRate.mSpeed); 4024 // TODO: ONLY USED FOR LEGACY RESAMPLERS, remove when they are removed. 4025 // add frames already consumed but not yet released by the resampler 4026 // because mAudioTrackServerProxy->framesReady() will include these frames 4027 desiredFrames += mAudioMixer->getUnreleasedFrames(track->name()); 4028 4029 uint32_t minFrames = 1; 4030 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() && 4031 (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) { 4032 minFrames = desiredFrames; 4033 } 4034 4035 size_t framesReady = track->framesReady(); 4036 if (ATRACE_ENABLED()) { 4037 // I wish we had formatted trace names 4038 char traceName[16]; 4039 strcpy(traceName, "nRdy"); 4040 int name = track->name(); 4041 if (AudioMixer::TRACK0 <= name && 4042 name < (int) (AudioMixer::TRACK0 + AudioMixer::MAX_NUM_TRACKS)) { 4043 name -= AudioMixer::TRACK0; 4044 traceName[4] = (name / 10) + '0'; 4045 traceName[5] = (name % 10) + '0'; 4046 } else { 4047 traceName[4] = '?'; 4048 traceName[5] = '?'; 4049 } 4050 traceName[6] = '\0'; 4051 ATRACE_INT(traceName, framesReady); 4052 } 4053 if ((framesReady >= minFrames) && track->isReady() && 4054 !track->isPaused() && !track->isTerminated()) 4055 { 4056 ALOGVV("track %d s=%08x [OK] on thread %p", name, cblk->mServer, this); 4057 4058 mixedTracks++; 4059 4060 // track->mainBuffer() != mSinkBuffer or mMixerBuffer means 4061 // there is an effect chain connected to the track 4062 chain.clear(); 4063 if (track->mainBuffer() != mSinkBuffer && 4064 track->mainBuffer() != mMixerBuffer) { 4065 if (mEffectBufferEnabled) { 4066 mEffectBufferValid = true; // Later can set directly. 4067 } 4068 chain = getEffectChain_l(track->sessionId()); 4069 // Delegate volume control to effect in track effect chain if needed 4070 if (chain != 0) { 4071 tracksWithEffect++; 4072 } else { 4073 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on " 4074 "session %d", 4075 name, track->sessionId()); 4076 } 4077 } 4078 4079 4080 int param = AudioMixer::VOLUME; 4081 if (track->mFillingUpStatus == Track::FS_FILLED) { 4082 // no ramp for the first volume setting 4083 track->mFillingUpStatus = Track::FS_ACTIVE; 4084 if (track->mState == TrackBase::RESUMING) { 4085 track->mState = TrackBase::ACTIVE; 4086 param = AudioMixer::RAMP_VOLUME; 4087 } 4088 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 4089 // FIXME should not make a decision based on mServer 4090 } else if (cblk->mServer != 0) { 4091 // If the track is stopped before the first frame was mixed, 4092 // do not apply ramp 4093 param = AudioMixer::RAMP_VOLUME; 4094 } 4095 4096 // compute volume for this track 4097 uint32_t vl, vr; // in U8.24 integer format 4098 float vlf, vrf, vaf; // in [0.0, 1.0] float format 4099 if (track->isPausing() || mStreamTypes[track->streamType()].mute) { 4100 vl = vr = 0; 4101 vlf = vrf = vaf = 0.; 4102 if (track->isPausing()) { 4103 track->setPaused(); 4104 } 4105 } else { 4106 4107 // read original volumes with volume control 4108 float typeVolume = mStreamTypes[track->streamType()].volume; 4109 float v = masterVolume * typeVolume; 4110 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 4111 gain_minifloat_packed_t vlr = proxy->getVolumeLR(); 4112 vlf = float_from_gain(gain_minifloat_unpack_left(vlr)); 4113 vrf = float_from_gain(gain_minifloat_unpack_right(vlr)); 4114 // track volumes come from shared memory, so can't be trusted and must be clamped 4115 if (vlf > GAIN_FLOAT_UNITY) { 4116 ALOGV("Track left volume out of range: %.3g", vlf); 4117 vlf = GAIN_FLOAT_UNITY; 4118 } 4119 if (vrf > GAIN_FLOAT_UNITY) { 4120 ALOGV("Track right volume out of range: %.3g", vrf); 4121 vrf = GAIN_FLOAT_UNITY; 4122 } 4123 // now apply the master volume and stream type volume 4124 vlf *= v; 4125 vrf *= v; 4126 // assuming master volume and stream type volume each go up to 1.0, 4127 // then derive vl and vr as U8.24 versions for the effect chain 4128 const float scaleto8_24 = MAX_GAIN_INT * MAX_GAIN_INT; 4129 vl = (uint32_t) (scaleto8_24 * vlf); 4130 vr = (uint32_t) (scaleto8_24 * vrf); 4131 // vl and vr are now in U8.24 format 4132 uint16_t sendLevel = proxy->getSendLevel_U4_12(); 4133 // send level comes from shared memory and so may be corrupt 4134 if (sendLevel > MAX_GAIN_INT) { 4135 ALOGV("Track send level out of range: %04X", sendLevel); 4136 sendLevel = MAX_GAIN_INT; 4137 } 4138 // vaf is represented as [0.0, 1.0] float by rescaling sendLevel 4139 vaf = v * sendLevel * (1. / MAX_GAIN_INT); 4140 } 4141 4142 // Delegate volume control to effect in track effect chain if needed 4143 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 4144 // Do not ramp volume if volume is controlled by effect 4145 param = AudioMixer::VOLUME; 4146 // Update remaining floating point volume levels 4147 vlf = (float)vl / (1 << 24); 4148 vrf = (float)vr / (1 << 24); 4149 track->mHasVolumeController = true; 4150 } else { 4151 // force no volume ramp when volume controller was just disabled or removed 4152 // from effect chain to avoid volume spike 4153 if (track->mHasVolumeController) { 4154 param = AudioMixer::VOLUME; 4155 } 4156 track->mHasVolumeController = false; 4157 } 4158 4159 // XXX: these things DON'T need to be done each time 4160 mAudioMixer->setBufferProvider(name, track); 4161 mAudioMixer->enable(name); 4162 4163 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, &vlf); 4164 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, &vrf); 4165 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, &vaf); 4166 mAudioMixer->setParameter( 4167 name, 4168 AudioMixer::TRACK, 4169 AudioMixer::FORMAT, (void *)track->format()); 4170 mAudioMixer->setParameter( 4171 name, 4172 AudioMixer::TRACK, 4173 AudioMixer::CHANNEL_MASK, (void *)(uintptr_t)track->channelMask()); 4174 mAudioMixer->setParameter( 4175 name, 4176 AudioMixer::TRACK, 4177 AudioMixer::MIXER_CHANNEL_MASK, (void *)(uintptr_t)mChannelMask); 4178 // limit track sample rate to 2 x output sample rate, which changes at re-configuration 4179 uint32_t maxSampleRate = mSampleRate * AUDIO_RESAMPLER_DOWN_RATIO_MAX; 4180 uint32_t reqSampleRate = track->mAudioTrackServerProxy->getSampleRate(); 4181 if (reqSampleRate == 0) { 4182 reqSampleRate = mSampleRate; 4183 } else if (reqSampleRate > maxSampleRate) { 4184 reqSampleRate = maxSampleRate; 4185 } 4186 mAudioMixer->setParameter( 4187 name, 4188 AudioMixer::RESAMPLE, 4189 AudioMixer::SAMPLE_RATE, 4190 (void *)(uintptr_t)reqSampleRate); 4191 4192 AudioPlaybackRate playbackRate = track->mAudioTrackServerProxy->getPlaybackRate(); 4193 mAudioMixer->setParameter( 4194 name, 4195 AudioMixer::TIMESTRETCH, 4196 AudioMixer::PLAYBACK_RATE, 4197 &playbackRate); 4198 4199 /* 4200 * Select the appropriate output buffer for the track. 4201 * 4202 * Tracks with effects go into their own effects chain buffer 4203 * and from there into either mEffectBuffer or mSinkBuffer. 4204 * 4205 * Other tracks can use mMixerBuffer for higher precision 4206 * channel accumulation. If this buffer is enabled 4207 * (mMixerBufferEnabled true), then selected tracks will accumulate 4208 * into it. 4209 * 4210 */ 4211 if (mMixerBufferEnabled 4212 && (track->mainBuffer() == mSinkBuffer 4213 || track->mainBuffer() == mMixerBuffer)) { 4214 mAudioMixer->setParameter( 4215 name, 4216 AudioMixer::TRACK, 4217 AudioMixer::MIXER_FORMAT, (void *)mMixerBufferFormat); 4218 mAudioMixer->setParameter( 4219 name, 4220 AudioMixer::TRACK, 4221 AudioMixer::MAIN_BUFFER, (void *)mMixerBuffer); 4222 // TODO: override track->mainBuffer()? 4223 mMixerBufferValid = true; 4224 } else { 4225 mAudioMixer->setParameter( 4226 name, 4227 AudioMixer::TRACK, 4228 AudioMixer::MIXER_FORMAT, (void *)AUDIO_FORMAT_PCM_16_BIT); 4229 mAudioMixer->setParameter( 4230 name, 4231 AudioMixer::TRACK, 4232 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 4233 } 4234 mAudioMixer->setParameter( 4235 name, 4236 AudioMixer::TRACK, 4237 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 4238 4239 // reset retry count 4240 track->mRetryCount = kMaxTrackRetries; 4241 4242 // If one track is ready, set the mixer ready if: 4243 // - the mixer was not ready during previous round OR 4244 // - no other track is not ready 4245 if (mMixerStatusIgnoringFastTracks != MIXER_TRACKS_READY || 4246 mixerStatus != MIXER_TRACKS_ENABLED) { 4247 mixerStatus = MIXER_TRACKS_READY; 4248 } 4249 } else { 4250 if (framesReady < desiredFrames && !track->isStopped() && !track->isPaused()) { 4251 ALOGV("track(%p) underrun, framesReady(%zu) < framesDesired(%zd)", 4252 track, framesReady, desiredFrames); 4253 track->mAudioTrackServerProxy->tallyUnderrunFrames(desiredFrames); 4254 } else { 4255 track->mAudioTrackServerProxy->tallyUnderrunFrames(0); 4256 } 4257 4258 // clear effect chain input buffer if an active track underruns to avoid sending 4259 // previous audio buffer again to effects 4260 chain = getEffectChain_l(track->sessionId()); 4261 if (chain != 0) { 4262 chain->clearInputBuffer(); 4263 } 4264 4265 ALOGVV("track %d s=%08x [NOT READY] on thread %p", name, cblk->mServer, this); 4266 if ((track->sharedBuffer() != 0) || track->isTerminated() || 4267 track->isStopped() || track->isPaused()) { 4268 // We have consumed all the buffers of this track. 4269 // Remove it from the list of active tracks. 4270 // TODO: use actual buffer filling status instead of latency when available from 4271 // audio HAL 4272 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 4273 int64_t framesWritten = mBytesWritten / mFrameSize; 4274 if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) { 4275 if (track->isStopped()) { 4276 track->reset(); 4277 } 4278 tracksToRemove->add(track); 4279 } 4280 } else { 4281 // No buffers for this track. Give it a few chances to 4282 // fill a buffer, then remove it from active list. 4283 if (--(track->mRetryCount) <= 0) { 4284 ALOGI("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 4285 tracksToRemove->add(track); 4286 // indicate to client process that the track was disabled because of underrun; 4287 // it will then automatically call start() when data is available 4288 track->disable(); 4289 // If one track is not ready, mark the mixer also not ready if: 4290 // - the mixer was ready during previous round OR 4291 // - no other track is ready 4292 } else if (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY || 4293 mixerStatus != MIXER_TRACKS_READY) { 4294 mixerStatus = MIXER_TRACKS_ENABLED; 4295 } 4296 } 4297 mAudioMixer->disable(name); 4298 } 4299 4300 } // local variable scope to avoid goto warning 4301 4302 } 4303 4304 // Push the new FastMixer state if necessary 4305 bool pauseAudioWatchdog = false; 4306 if (didModify) { 4307 state->mFastTracksGen++; 4308 // if the fast mixer was active, but now there are no fast tracks, then put it in cold idle 4309 if (kUseFastMixer == FastMixer_Dynamic && 4310 state->mCommand == FastMixerState::MIX_WRITE && state->mTrackMask <= 1) { 4311 state->mCommand = FastMixerState::COLD_IDLE; 4312 state->mColdFutexAddr = &mFastMixerFutex; 4313 state->mColdGen++; 4314 mFastMixerFutex = 0; 4315 if (kUseFastMixer == FastMixer_Dynamic) { 4316 mNormalSink = mOutputSink; 4317 } 4318 // If we go into cold idle, need to wait for acknowledgement 4319 // so that fast mixer stops doing I/O. 4320 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 4321 pauseAudioWatchdog = true; 4322 } 4323 } 4324 if (sq != NULL) { 4325 sq->end(didModify); 4326 sq->push(block); 4327 } 4328#ifdef AUDIO_WATCHDOG 4329 if (pauseAudioWatchdog && mAudioWatchdog != 0) { 4330 mAudioWatchdog->pause(); 4331 } 4332#endif 4333 4334 // Now perform the deferred reset on fast tracks that have stopped 4335 while (resetMask != 0) { 4336 size_t i = __builtin_ctz(resetMask); 4337 ALOG_ASSERT(i < count); 4338 resetMask &= ~(1 << i); 4339 sp<Track> t = mActiveTracks[i].promote(); 4340 if (t == 0) { 4341 continue; 4342 } 4343 Track* track = t.get(); 4344 ALOG_ASSERT(track->isFastTrack() && track->isStopped()); 4345 track->reset(); 4346 } 4347 4348 // remove all the tracks that need to be... 4349 removeTracks_l(*tracksToRemove); 4350 4351 if (getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX) != 0) { 4352 mEffectBufferValid = true; 4353 } 4354 4355 if (mEffectBufferValid) { 4356 // as long as there are effects we should clear the effects buffer, to avoid 4357 // passing a non-clean buffer to the effect chain 4358 memset(mEffectBuffer, 0, mEffectBufferSize); 4359 } 4360 // sink or mix buffer must be cleared if all tracks are connected to an 4361 // effect chain as in this case the mixer will not write to the sink or mix buffer 4362 // and track effects will accumulate into it 4363 if ((mBytesRemaining == 0) && ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 4364 (mixedTracks == 0 && fastTracks > 0))) { 4365 // FIXME as a performance optimization, should remember previous zero status 4366 if (mMixerBufferValid) { 4367 memset(mMixerBuffer, 0, mMixerBufferSize); 4368 // TODO: In testing, mSinkBuffer below need not be cleared because 4369 // the PlaybackThread::threadLoop() copies mMixerBuffer into mSinkBuffer 4370 // after mixing. 4371 // 4372 // To enforce this guarantee: 4373 // ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 4374 // (mixedTracks == 0 && fastTracks > 0)) 4375 // must imply MIXER_TRACKS_READY. 4376 // Later, we may clear buffers regardless, and skip much of this logic. 4377 } 4378 // FIXME as a performance optimization, should remember previous zero status 4379 memset(mSinkBuffer, 0, mNormalFrameCount * mFrameSize); 4380 } 4381 4382 // if any fast tracks, then status is ready 4383 mMixerStatusIgnoringFastTracks = mixerStatus; 4384 if (fastTracks > 0) { 4385 mixerStatus = MIXER_TRACKS_READY; 4386 } 4387 return mixerStatus; 4388} 4389 4390// getTrackName_l() must be called with ThreadBase::mLock held 4391int AudioFlinger::MixerThread::getTrackName_l(audio_channel_mask_t channelMask, 4392 audio_format_t format, audio_session_t sessionId) 4393{ 4394 return mAudioMixer->getTrackName(channelMask, format, sessionId); 4395} 4396 4397// deleteTrackName_l() must be called with ThreadBase::mLock held 4398void AudioFlinger::MixerThread::deleteTrackName_l(int name) 4399{ 4400 ALOGV("remove track (%d) and delete from mixer", name); 4401 mAudioMixer->deleteTrackName(name); 4402} 4403 4404// checkForNewParameter_l() must be called with ThreadBase::mLock held 4405bool AudioFlinger::MixerThread::checkForNewParameter_l(const String8& keyValuePair, 4406 status_t& status) 4407{ 4408 bool reconfig = false; 4409 bool a2dpDeviceChanged = false; 4410 4411 status = NO_ERROR; 4412 4413 AutoPark<FastMixer> park(mFastMixer); 4414 4415 AudioParameter param = AudioParameter(keyValuePair); 4416 int value; 4417 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 4418 reconfig = true; 4419 } 4420 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 4421 if (!isValidPcmSinkFormat((audio_format_t) value)) { 4422 status = BAD_VALUE; 4423 } else { 4424 // no need to save value, since it's constant 4425 reconfig = true; 4426 } 4427 } 4428 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 4429 if (!isValidPcmSinkChannelMask((audio_channel_mask_t) value)) { 4430 status = BAD_VALUE; 4431 } else { 4432 // no need to save value, since it's constant 4433 reconfig = true; 4434 } 4435 } 4436 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 4437 // do not accept frame count changes if tracks are open as the track buffer 4438 // size depends on frame count and correct behavior would not be guaranteed 4439 // if frame count is changed after track creation 4440 if (!mTracks.isEmpty()) { 4441 status = INVALID_OPERATION; 4442 } else { 4443 reconfig = true; 4444 } 4445 } 4446 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 4447#ifdef ADD_BATTERY_DATA 4448 // when changing the audio output device, call addBatteryData to notify 4449 // the change 4450 if (mOutDevice != value) { 4451 uint32_t params = 0; 4452 // check whether speaker is on 4453 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 4454 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 4455 } 4456 4457 audio_devices_t deviceWithoutSpeaker 4458 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 4459 // check if any other device (except speaker) is on 4460 if (value & deviceWithoutSpeaker) { 4461 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 4462 } 4463 4464 if (params != 0) { 4465 addBatteryData(params); 4466 } 4467 } 4468#endif 4469 4470 // forward device change to effects that have requested to be 4471 // aware of attached audio device. 4472 if (value != AUDIO_DEVICE_NONE) { 4473 a2dpDeviceChanged = 4474 (mOutDevice & AUDIO_DEVICE_OUT_ALL_A2DP) != (value & AUDIO_DEVICE_OUT_ALL_A2DP); 4475 mOutDevice = value; 4476 for (size_t i = 0; i < mEffectChains.size(); i++) { 4477 mEffectChains[i]->setDevice_l(mOutDevice); 4478 } 4479 } 4480 } 4481 4482 if (status == NO_ERROR) { 4483 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 4484 keyValuePair.string()); 4485 if (!mStandby && status == INVALID_OPERATION) { 4486 mOutput->standby(); 4487 mStandby = true; 4488 mBytesWritten = 0; 4489 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 4490 keyValuePair.string()); 4491 } 4492 if (status == NO_ERROR && reconfig) { 4493 readOutputParameters_l(); 4494 delete mAudioMixer; 4495 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 4496 for (size_t i = 0; i < mTracks.size() ; i++) { 4497 int name = getTrackName_l(mTracks[i]->mChannelMask, 4498 mTracks[i]->mFormat, mTracks[i]->mSessionId); 4499 if (name < 0) { 4500 break; 4501 } 4502 mTracks[i]->mName = name; 4503 } 4504 sendIoConfigEvent_l(AUDIO_OUTPUT_CONFIG_CHANGED); 4505 } 4506 } 4507 4508 return reconfig || a2dpDeviceChanged; 4509} 4510 4511 4512void AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 4513{ 4514 PlaybackThread::dumpInternals(fd, args); 4515 dprintf(fd, " Thread throttle time (msecs): %u\n", mThreadThrottleTimeMs); 4516 dprintf(fd, " AudioMixer tracks: 0x%08x\n", mAudioMixer->trackNames()); 4517 dprintf(fd, " Master mono: %s\n", mMasterMono ? "on" : "off"); 4518 4519 // Make a non-atomic copy of fast mixer dump state so it won't change underneath us 4520 // while we are dumping it. It may be inconsistent, but it won't mutate! 4521 // This is a large object so we place it on the heap. 4522 // FIXME 25972958: Need an intelligent copy constructor that does not touch unused pages. 4523 const FastMixerDumpState *copy = new FastMixerDumpState(mFastMixerDumpState); 4524 copy->dump(fd); 4525 delete copy; 4526 4527#ifdef STATE_QUEUE_DUMP 4528 // Similar for state queue 4529 StateQueueObserverDump observerCopy = mStateQueueObserverDump; 4530 observerCopy.dump(fd); 4531 StateQueueMutatorDump mutatorCopy = mStateQueueMutatorDump; 4532 mutatorCopy.dump(fd); 4533#endif 4534 4535#ifdef TEE_SINK 4536 // Write the tee output to a .wav file 4537 dumpTee(fd, mTeeSource, mId); 4538#endif 4539 4540#ifdef AUDIO_WATCHDOG 4541 if (mAudioWatchdog != 0) { 4542 // Make a non-atomic copy of audio watchdog dump so it won't change underneath us 4543 AudioWatchdogDump wdCopy = mAudioWatchdogDump; 4544 wdCopy.dump(fd); 4545 } 4546#endif 4547} 4548 4549uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const 4550{ 4551 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000) / 2; 4552} 4553 4554uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() const 4555{ 4556 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000); 4557} 4558 4559void AudioFlinger::MixerThread::cacheParameters_l() 4560{ 4561 PlaybackThread::cacheParameters_l(); 4562 4563 // FIXME: Relaxed timing because of a certain device that can't meet latency 4564 // Should be reduced to 2x after the vendor fixes the driver issue 4565 // increase threshold again due to low power audio mode. The way this warning 4566 // threshold is calculated and its usefulness should be reconsidered anyway. 4567 maxPeriod = seconds(mNormalFrameCount) / mSampleRate * 15; 4568} 4569 4570// ---------------------------------------------------------------------------- 4571 4572AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 4573 AudioStreamOut* output, audio_io_handle_t id, audio_devices_t device, bool systemReady, 4574 uint32_t bitRate) 4575 : PlaybackThread(audioFlinger, output, id, device, DIRECT, systemReady, bitRate) 4576 // mLeftVolFloat, mRightVolFloat 4577{ 4578} 4579 4580AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 4581 AudioStreamOut* output, audio_io_handle_t id, uint32_t device, 4582 ThreadBase::type_t type, bool systemReady, uint32_t bitRate) 4583 : PlaybackThread(audioFlinger, output, id, device, type, systemReady, bitRate) 4584 // mLeftVolFloat, mRightVolFloat 4585{ 4586} 4587 4588AudioFlinger::DirectOutputThread::~DirectOutputThread() 4589{ 4590} 4591 4592void AudioFlinger::DirectOutputThread::processVolume_l(Track *track, bool lastTrack) 4593{ 4594 float left, right; 4595 4596 if (mMasterMute || mStreamTypes[track->streamType()].mute) { 4597 left = right = 0; 4598 } else { 4599 float typeVolume = mStreamTypes[track->streamType()].volume; 4600 float v = mMasterVolume * typeVolume; 4601 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 4602 gain_minifloat_packed_t vlr = proxy->getVolumeLR(); 4603 left = float_from_gain(gain_minifloat_unpack_left(vlr)); 4604 if (left > GAIN_FLOAT_UNITY) { 4605 left = GAIN_FLOAT_UNITY; 4606 } 4607 left *= v; 4608 right = float_from_gain(gain_minifloat_unpack_right(vlr)); 4609 if (right > GAIN_FLOAT_UNITY) { 4610 right = GAIN_FLOAT_UNITY; 4611 } 4612 right *= v; 4613 } 4614 4615 if (lastTrack) { 4616 if (left != mLeftVolFloat || right != mRightVolFloat) { 4617 mLeftVolFloat = left; 4618 mRightVolFloat = right; 4619 4620 // Convert volumes from float to 8.24 4621 uint32_t vl = (uint32_t)(left * (1 << 24)); 4622 uint32_t vr = (uint32_t)(right * (1 << 24)); 4623 4624 // Delegate volume control to effect in track effect chain if needed 4625 // only one effect chain can be present on DirectOutputThread, so if 4626 // there is one, the track is connected to it 4627 if (!mEffectChains.isEmpty()) { 4628 mEffectChains[0]->setVolume_l(&vl, &vr); 4629 left = (float)vl / (1 << 24); 4630 right = (float)vr / (1 << 24); 4631 } 4632 if (mOutput->stream->set_volume) { 4633 mOutput->stream->set_volume(mOutput->stream, left, right); 4634 } 4635 } 4636 } 4637} 4638 4639void AudioFlinger::DirectOutputThread::onAddNewTrack_l() 4640{ 4641 sp<Track> previousTrack = mPreviousTrack.promote(); 4642 sp<Track> latestTrack = mLatestActiveTrack.promote(); 4643 4644 if (previousTrack != 0 && latestTrack != 0) { 4645 if (mType == DIRECT) { 4646 if (previousTrack.get() != latestTrack.get()) { 4647 mFlushPending = true; 4648 } 4649 } else /* mType == OFFLOAD */ { 4650 if (previousTrack->sessionId() != latestTrack->sessionId()) { 4651 mFlushPending = true; 4652 } 4653 } 4654 } 4655 PlaybackThread::onAddNewTrack_l(); 4656} 4657 4658AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l( 4659 Vector< sp<Track> > *tracksToRemove 4660) 4661{ 4662 size_t count = mActiveTracks.size(); 4663 mixer_state mixerStatus = MIXER_IDLE; 4664 bool doHwPause = false; 4665 bool doHwResume = false; 4666 4667 // find out which tracks need to be processed 4668 for (size_t i = 0; i < count; i++) { 4669 sp<Track> t = mActiveTracks[i].promote(); 4670 // The track died recently 4671 if (t == 0) { 4672 continue; 4673 } 4674 4675 if (t->isInvalid()) { 4676 ALOGW("An invalidated track shouldn't be in active list"); 4677 tracksToRemove->add(t); 4678 continue; 4679 } 4680 4681 Track* const track = t.get(); 4682#ifdef VERY_VERY_VERBOSE_LOGGING 4683 audio_track_cblk_t* cblk = track->cblk(); 4684#endif 4685 // Only consider last track started for volume and mixer state control. 4686 // In theory an older track could underrun and restart after the new one starts 4687 // but as we only care about the transition phase between two tracks on a 4688 // direct output, it is not a problem to ignore the underrun case. 4689 sp<Track> l = mLatestActiveTrack.promote(); 4690 bool last = l.get() == track; 4691 4692 if (track->isPausing()) { 4693 track->setPaused(); 4694 if (mHwSupportsPause && last && !mHwPaused) { 4695 doHwPause = true; 4696 mHwPaused = true; 4697 } 4698 tracksToRemove->add(track); 4699 } else if (track->isFlushPending()) { 4700 track->flushAck(); 4701 if (last) { 4702 mFlushPending = true; 4703 } 4704 } else if (track->isResumePending()) { 4705 track->resumeAck(); 4706 if (last && mHwPaused) { 4707 doHwResume = true; 4708 mHwPaused = false; 4709 } 4710 } 4711 4712 // The first time a track is added we wait 4713 // for all its buffers to be filled before processing it. 4714 // Allow draining the buffer in case the client 4715 // app does not call stop() and relies on underrun to stop: 4716 // hence the test on (track->mRetryCount > 1). 4717 // If retryCount<=1 then track is about to underrun and be removed. 4718 // Do not use a high threshold for compressed audio. 4719 uint32_t minFrames; 4720 if ((track->sharedBuffer() == 0) && !track->isStopping_1() && !track->isPausing() 4721 && (track->mRetryCount > 1) && audio_has_proportional_frames(mFormat)) { 4722 minFrames = mNormalFrameCount; 4723 } else { 4724 minFrames = 1; 4725 } 4726 4727 if ((track->framesReady() >= minFrames) && track->isReady() && !track->isPaused() && 4728 !track->isStopping_2() && !track->isStopped()) 4729 { 4730 ALOGVV("track %d s=%08x [OK]", track->name(), cblk->mServer); 4731 4732 if (track->mFillingUpStatus == Track::FS_FILLED) { 4733 track->mFillingUpStatus = Track::FS_ACTIVE; 4734 // make sure processVolume_l() will apply new volume even if 0 4735 mLeftVolFloat = mRightVolFloat = -1.0; 4736 if (!mHwSupportsPause) { 4737 track->resumeAck(); 4738 } 4739 } 4740 4741 // compute volume for this track 4742 processVolume_l(track, last); 4743 if (last) { 4744 sp<Track> previousTrack = mPreviousTrack.promote(); 4745 if (previousTrack != 0) { 4746 if (track != previousTrack.get()) { 4747 // Flush any data still being written from last track 4748 mBytesRemaining = 0; 4749 // Invalidate previous track to force a seek when resuming. 4750 previousTrack->invalidate(); 4751 } 4752 } 4753 mPreviousTrack = track; 4754 4755 // reset retry count 4756 track->mRetryCount = kMaxTrackRetriesDirect; 4757 mActiveTrack = t; 4758 mixerStatus = MIXER_TRACKS_READY; 4759 if (mHwPaused) { 4760 doHwResume = true; 4761 mHwPaused = false; 4762 } 4763 } 4764 } else { 4765 // clear effect chain input buffer if the last active track started underruns 4766 // to avoid sending previous audio buffer again to effects 4767 if (!mEffectChains.isEmpty() && last) { 4768 mEffectChains[0]->clearInputBuffer(); 4769 } 4770 if (track->isStopping_1()) { 4771 track->mState = TrackBase::STOPPING_2; 4772 if (last && mHwPaused) { 4773 doHwResume = true; 4774 mHwPaused = false; 4775 } 4776 } 4777 if ((track->sharedBuffer() != 0) || track->isStopped() || 4778 track->isStopping_2() || track->isPaused()) { 4779 // We have consumed all the buffers of this track. 4780 // Remove it from the list of active tracks. 4781 size_t audioHALFrames; 4782 if (audio_has_proportional_frames(mFormat)) { 4783 audioHALFrames = (latency_l() * mSampleRate) / 1000; 4784 } else { 4785 audioHALFrames = 0; 4786 } 4787 4788 int64_t framesWritten = mBytesWritten / mFrameSize; 4789 if (mStandby || !last || 4790 track->presentationComplete(framesWritten, audioHALFrames)) { 4791 if (track->isStopping_2()) { 4792 track->mState = TrackBase::STOPPED; 4793 } 4794 if (track->isStopped()) { 4795 track->reset(); 4796 } 4797 tracksToRemove->add(track); 4798 } 4799 } else { 4800 // No buffers for this track. Give it a few chances to 4801 // fill a buffer, then remove it from active list. 4802 // Only consider last track started for mixer state control 4803 if (--(track->mRetryCount) <= 0) { 4804 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 4805 tracksToRemove->add(track); 4806 // indicate to client process that the track was disabled because of underrun; 4807 // it will then automatically call start() when data is available 4808 track->disable(); 4809 } else if (last) { 4810 ALOGW("pause because of UNDERRUN, framesReady = %zu," 4811 "minFrames = %u, mFormat = %#x", 4812 track->framesReady(), minFrames, mFormat); 4813 mixerStatus = MIXER_TRACKS_ENABLED; 4814 if (mHwSupportsPause && !mHwPaused && !mStandby) { 4815 doHwPause = true; 4816 mHwPaused = true; 4817 } 4818 } 4819 } 4820 } 4821 } 4822 4823 // if an active track did not command a flush, check for pending flush on stopped tracks 4824 if (!mFlushPending) { 4825 for (size_t i = 0; i < mTracks.size(); i++) { 4826 if (mTracks[i]->isFlushPending()) { 4827 mTracks[i]->flushAck(); 4828 mFlushPending = true; 4829 } 4830 } 4831 } 4832 4833 // make sure the pause/flush/resume sequence is executed in the right order. 4834 // If a flush is pending and a track is active but the HW is not paused, force a HW pause 4835 // before flush and then resume HW. This can happen in case of pause/flush/resume 4836 // if resume is received before pause is executed. 4837 if (mHwSupportsPause && !mStandby && 4838 (doHwPause || (mFlushPending && !mHwPaused && (count != 0)))) { 4839 mOutput->stream->pause(mOutput->stream); 4840 } 4841 if (mFlushPending) { 4842 flushHw_l(); 4843 } 4844 if (mHwSupportsPause && !mStandby && doHwResume) { 4845 mOutput->stream->resume(mOutput->stream); 4846 } 4847 // remove all the tracks that need to be... 4848 removeTracks_l(*tracksToRemove); 4849 4850 return mixerStatus; 4851} 4852 4853void AudioFlinger::DirectOutputThread::threadLoop_mix() 4854{ 4855 size_t frameCount = mFrameCount; 4856 int8_t *curBuf = (int8_t *)mSinkBuffer; 4857 // output audio to hardware 4858 while (frameCount) { 4859 AudioBufferProvider::Buffer buffer; 4860 buffer.frameCount = frameCount; 4861 status_t status = mActiveTrack->getNextBuffer(&buffer); 4862 if (status != NO_ERROR || buffer.raw == NULL) { 4863 // no need to pad with 0 for compressed audio 4864 if (audio_has_proportional_frames(mFormat)) { 4865 memset(curBuf, 0, frameCount * mFrameSize); 4866 } 4867 break; 4868 } 4869 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 4870 frameCount -= buffer.frameCount; 4871 curBuf += buffer.frameCount * mFrameSize; 4872 mActiveTrack->releaseBuffer(&buffer); 4873 } 4874 mCurrentWriteLength = curBuf - (int8_t *)mSinkBuffer; 4875 mSleepTimeUs = 0; 4876 mStandbyTimeNs = systemTime() + mStandbyDelayNs; 4877 mActiveTrack.clear(); 4878} 4879 4880void AudioFlinger::DirectOutputThread::threadLoop_sleepTime() 4881{ 4882 // do not write to HAL when paused 4883 if (mHwPaused || (usesHwAvSync() && mStandby)) { 4884 mSleepTimeUs = mIdleSleepTimeUs; 4885 return; 4886 } 4887 if (mSleepTimeUs == 0) { 4888 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4889 // For compressed offload, use faster sleep time when underruning until more than an 4890 // entire buffer was written to the audio HAL 4891 if (!audio_has_proportional_frames(mFormat) && 4892 (mType == OFFLOAD) && (mBytesWritten < (int64_t) mBufferSize)) { 4893 mSleepTimeUs = kDirectMinSleepTimeUs; 4894 } else { 4895 mSleepTimeUs = mActiveSleepTimeUs; 4896 } 4897 } else { 4898 mSleepTimeUs = mIdleSleepTimeUs; 4899 } 4900 } else if (mBytesWritten != 0 && audio_has_proportional_frames(mFormat)) { 4901 memset(mSinkBuffer, 0, mFrameCount * mFrameSize); 4902 mSleepTimeUs = 0; 4903 } 4904} 4905 4906void AudioFlinger::DirectOutputThread::threadLoop_exit() 4907{ 4908 { 4909 Mutex::Autolock _l(mLock); 4910 for (size_t i = 0; i < mTracks.size(); i++) { 4911 if (mTracks[i]->isFlushPending()) { 4912 mTracks[i]->flushAck(); 4913 mFlushPending = true; 4914 } 4915 } 4916 if (mFlushPending) { 4917 flushHw_l(); 4918 } 4919 } 4920 PlaybackThread::threadLoop_exit(); 4921} 4922 4923// must be called with thread mutex locked 4924bool AudioFlinger::DirectOutputThread::shouldStandby_l() 4925{ 4926 bool trackPaused = false; 4927 bool trackStopped = false; 4928 4929 if ((mType == DIRECT) && audio_is_linear_pcm(mFormat) && !usesHwAvSync()) { 4930 return !mStandby; 4931 } 4932 4933 // do not put the HAL in standby when paused. AwesomePlayer clear the offloaded AudioTrack 4934 // after a timeout and we will enter standby then. 4935 if (mTracks.size() > 0) { 4936 trackPaused = mTracks[mTracks.size() - 1]->isPaused(); 4937 trackStopped = mTracks[mTracks.size() - 1]->isStopped() || 4938 mTracks[mTracks.size() - 1]->mState == TrackBase::IDLE; 4939 } 4940 4941 return !mStandby && !(trackPaused || (mHwPaused && !trackStopped)); 4942} 4943 4944// getTrackName_l() must be called with ThreadBase::mLock held 4945int AudioFlinger::DirectOutputThread::getTrackName_l(audio_channel_mask_t channelMask __unused, 4946 audio_format_t format __unused, audio_session_t sessionId __unused) 4947{ 4948 return 0; 4949} 4950 4951// deleteTrackName_l() must be called with ThreadBase::mLock held 4952void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name __unused) 4953{ 4954} 4955 4956// checkForNewParameter_l() must be called with ThreadBase::mLock held 4957bool AudioFlinger::DirectOutputThread::checkForNewParameter_l(const String8& keyValuePair, 4958 status_t& status) 4959{ 4960 bool reconfig = false; 4961 bool a2dpDeviceChanged = false; 4962 4963 status = NO_ERROR; 4964 4965 AudioParameter param = AudioParameter(keyValuePair); 4966 int value; 4967 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 4968 // forward device change to effects that have requested to be 4969 // aware of attached audio device. 4970 if (value != AUDIO_DEVICE_NONE) { 4971 a2dpDeviceChanged = 4972 (mOutDevice & AUDIO_DEVICE_OUT_ALL_A2DP) != (value & AUDIO_DEVICE_OUT_ALL_A2DP); 4973 mOutDevice = value; 4974 for (size_t i = 0; i < mEffectChains.size(); i++) { 4975 mEffectChains[i]->setDevice_l(mOutDevice); 4976 } 4977 } 4978 } 4979 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 4980 // do not accept frame count changes if tracks are open as the track buffer 4981 // size depends on frame count and correct behavior would not be garantied 4982 // if frame count is changed after track creation 4983 if (!mTracks.isEmpty()) { 4984 status = INVALID_OPERATION; 4985 } else { 4986 reconfig = true; 4987 } 4988 } 4989 if (status == NO_ERROR) { 4990 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 4991 keyValuePair.string()); 4992 if (!mStandby && status == INVALID_OPERATION) { 4993 mOutput->standby(); 4994 mStandby = true; 4995 mBytesWritten = 0; 4996 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 4997 keyValuePair.string()); 4998 } 4999 if (status == NO_ERROR && reconfig) { 5000 readOutputParameters_l(); 5001 sendIoConfigEvent_l(AUDIO_OUTPUT_CONFIG_CHANGED); 5002 } 5003 } 5004 5005 return reconfig || a2dpDeviceChanged; 5006} 5007 5008uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() const 5009{ 5010 uint32_t time; 5011 if (audio_has_proportional_frames(mFormat)) { 5012 time = PlaybackThread::activeSleepTimeUs(); 5013 } else { 5014 time = kDirectMinSleepTimeUs; 5015 } 5016 return time; 5017} 5018 5019uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() const 5020{ 5021 uint32_t time; 5022 if (audio_has_proportional_frames(mFormat)) { 5023 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 5024 } else { 5025 time = kDirectMinSleepTimeUs; 5026 } 5027 return time; 5028} 5029 5030uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() const 5031{ 5032 uint32_t time; 5033 if (audio_has_proportional_frames(mFormat)) { 5034 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 5035 } else { 5036 time = kDirectMinSleepTimeUs; 5037 } 5038 return time; 5039} 5040 5041void AudioFlinger::DirectOutputThread::cacheParameters_l() 5042{ 5043 PlaybackThread::cacheParameters_l(); 5044 5045 // use shorter standby delay as on normal output to release 5046 // hardware resources as soon as possible 5047 // no delay on outputs with HW A/V sync 5048 if (usesHwAvSync()) { 5049 mStandbyDelayNs = 0; 5050 } else if ((mType == OFFLOAD) && !audio_has_proportional_frames(mFormat)) { 5051 mStandbyDelayNs = kOffloadStandbyDelayNs; 5052 } else { 5053 mStandbyDelayNs = microseconds(mActiveSleepTimeUs*2); 5054 } 5055} 5056 5057void AudioFlinger::DirectOutputThread::flushHw_l() 5058{ 5059 mOutput->flush(); 5060 mHwPaused = false; 5061 mFlushPending = false; 5062} 5063 5064// ---------------------------------------------------------------------------- 5065 5066AudioFlinger::AsyncCallbackThread::AsyncCallbackThread( 5067 const wp<AudioFlinger::PlaybackThread>& playbackThread) 5068 : Thread(false /*canCallJava*/), 5069 mPlaybackThread(playbackThread), 5070 mWriteAckSequence(0), 5071 mDrainSequence(0) 5072{ 5073} 5074 5075AudioFlinger::AsyncCallbackThread::~AsyncCallbackThread() 5076{ 5077} 5078 5079void AudioFlinger::AsyncCallbackThread::onFirstRef() 5080{ 5081 run("Offload Cbk", ANDROID_PRIORITY_URGENT_AUDIO); 5082} 5083 5084bool AudioFlinger::AsyncCallbackThread::threadLoop() 5085{ 5086 while (!exitPending()) { 5087 uint32_t writeAckSequence; 5088 uint32_t drainSequence; 5089 5090 { 5091 Mutex::Autolock _l(mLock); 5092 while (!((mWriteAckSequence & 1) || 5093 (mDrainSequence & 1) || 5094 exitPending())) { 5095 mWaitWorkCV.wait(mLock); 5096 } 5097 5098 if (exitPending()) { 5099 break; 5100 } 5101 ALOGV("AsyncCallbackThread mWriteAckSequence %d mDrainSequence %d", 5102 mWriteAckSequence, mDrainSequence); 5103 writeAckSequence = mWriteAckSequence; 5104 mWriteAckSequence &= ~1; 5105 drainSequence = mDrainSequence; 5106 mDrainSequence &= ~1; 5107 } 5108 { 5109 sp<AudioFlinger::PlaybackThread> playbackThread = mPlaybackThread.promote(); 5110 if (playbackThread != 0) { 5111 if (writeAckSequence & 1) { 5112 playbackThread->resetWriteBlocked(writeAckSequence >> 1); 5113 } 5114 if (drainSequence & 1) { 5115 playbackThread->resetDraining(drainSequence >> 1); 5116 } 5117 } 5118 } 5119 } 5120 return false; 5121} 5122 5123void AudioFlinger::AsyncCallbackThread::exit() 5124{ 5125 ALOGV("AsyncCallbackThread::exit"); 5126 Mutex::Autolock _l(mLock); 5127 requestExit(); 5128 mWaitWorkCV.broadcast(); 5129} 5130 5131void AudioFlinger::AsyncCallbackThread::setWriteBlocked(uint32_t sequence) 5132{ 5133 Mutex::Autolock _l(mLock); 5134 // bit 0 is cleared 5135 mWriteAckSequence = sequence << 1; 5136} 5137 5138void AudioFlinger::AsyncCallbackThread::resetWriteBlocked() 5139{ 5140 Mutex::Autolock _l(mLock); 5141 // ignore unexpected callbacks 5142 if (mWriteAckSequence & 2) { 5143 mWriteAckSequence |= 1; 5144 mWaitWorkCV.signal(); 5145 } 5146} 5147 5148void AudioFlinger::AsyncCallbackThread::setDraining(uint32_t sequence) 5149{ 5150 Mutex::Autolock _l(mLock); 5151 // bit 0 is cleared 5152 mDrainSequence = sequence << 1; 5153} 5154 5155void AudioFlinger::AsyncCallbackThread::resetDraining() 5156{ 5157 Mutex::Autolock _l(mLock); 5158 // ignore unexpected callbacks 5159 if (mDrainSequence & 2) { 5160 mDrainSequence |= 1; 5161 mWaitWorkCV.signal(); 5162 } 5163} 5164 5165 5166// ---------------------------------------------------------------------------- 5167AudioFlinger::OffloadThread::OffloadThread(const sp<AudioFlinger>& audioFlinger, 5168 AudioStreamOut* output, audio_io_handle_t id, uint32_t device, bool systemReady, 5169 uint32_t bitRate) 5170 : DirectOutputThread(audioFlinger, output, id, device, OFFLOAD, systemReady, bitRate), 5171 mPausedBytesRemaining(0) 5172{ 5173 //FIXME: mStandby should be set to true by ThreadBase constructor 5174 mStandby = true; 5175} 5176 5177void AudioFlinger::OffloadThread::threadLoop_exit() 5178{ 5179 if (mFlushPending || mHwPaused) { 5180 // If a flush is pending or track was paused, just discard buffered data 5181 flushHw_l(); 5182 } else { 5183 mMixerStatus = MIXER_DRAIN_ALL; 5184 threadLoop_drain(); 5185 } 5186 if (mUseAsyncWrite) { 5187 ALOG_ASSERT(mCallbackThread != 0); 5188 mCallbackThread->exit(); 5189 } 5190 PlaybackThread::threadLoop_exit(); 5191} 5192 5193AudioFlinger::PlaybackThread::mixer_state AudioFlinger::OffloadThread::prepareTracks_l( 5194 Vector< sp<Track> > *tracksToRemove 5195) 5196{ 5197 size_t count = mActiveTracks.size(); 5198 5199 mixer_state mixerStatus = MIXER_IDLE; 5200 bool doHwPause = false; 5201 bool doHwResume = false; 5202 5203 ALOGV("OffloadThread::prepareTracks_l active tracks %zu", count); 5204 5205 // find out which tracks need to be processed 5206 for (size_t i = 0; i < count; i++) { 5207 sp<Track> t = mActiveTracks[i].promote(); 5208 // The track died recently 5209 if (t == 0) { 5210 continue; 5211 } 5212 Track* const track = t.get(); 5213#ifdef VERY_VERY_VERBOSE_LOGGING 5214 audio_track_cblk_t* cblk = track->cblk(); 5215#endif 5216 // Only consider last track started for volume and mixer state control. 5217 // In theory an older track could underrun and restart after the new one starts 5218 // but as we only care about the transition phase between two tracks on a 5219 // direct output, it is not a problem to ignore the underrun case. 5220 sp<Track> l = mLatestActiveTrack.promote(); 5221 bool last = l.get() == track; 5222 5223 if (track->isInvalid()) { 5224 ALOGW("An invalidated track shouldn't be in active list"); 5225 tracksToRemove->add(track); 5226 continue; 5227 } 5228 5229 if (track->mState == TrackBase::IDLE) { 5230 ALOGW("An idle track shouldn't be in active list"); 5231 continue; 5232 } 5233 5234 if (track->isPausing()) { 5235 track->setPaused(); 5236 if (last) { 5237 if (mHwSupportsPause && !mHwPaused) { 5238 doHwPause = true; 5239 mHwPaused = true; 5240 } 5241 // If we were part way through writing the mixbuffer to 5242 // the HAL we must save this until we resume 5243 // BUG - this will be wrong if a different track is made active, 5244 // in that case we want to discard the pending data in the 5245 // mixbuffer and tell the client to present it again when the 5246 // track is resumed 5247 mPausedWriteLength = mCurrentWriteLength; 5248 mPausedBytesRemaining = mBytesRemaining; 5249 mBytesRemaining = 0; // stop writing 5250 } 5251 tracksToRemove->add(track); 5252 } else if (track->isFlushPending()) { 5253 track->mRetryCount = kMaxTrackRetriesOffload; 5254 track->flushAck(); 5255 if (last) { 5256 mFlushPending = true; 5257 } 5258 } else if (track->isResumePending()){ 5259 track->resumeAck(); 5260 if (last) { 5261 if (mPausedBytesRemaining) { 5262 // Need to continue write that was interrupted 5263 mCurrentWriteLength = mPausedWriteLength; 5264 mBytesRemaining = mPausedBytesRemaining; 5265 mPausedBytesRemaining = 0; 5266 } 5267 if (mHwPaused) { 5268 doHwResume = true; 5269 mHwPaused = false; 5270 // threadLoop_mix() will handle the case that we need to 5271 // resume an interrupted write 5272 } 5273 // enable write to audio HAL 5274 mSleepTimeUs = 0; 5275 5276 // Do not handle new data in this iteration even if track->framesReady() 5277 mixerStatus = MIXER_TRACKS_ENABLED; 5278 } 5279 } else if (track->framesReady() && track->isReady() && 5280 !track->isPaused() && !track->isTerminated() && !track->isStopping_2()) { 5281 ALOGVV("OffloadThread: track %d s=%08x [OK]", track->name(), cblk->mServer); 5282 if (track->mFillingUpStatus == Track::FS_FILLED) { 5283 track->mFillingUpStatus = Track::FS_ACTIVE; 5284 // make sure processVolume_l() will apply new volume even if 0 5285 mLeftVolFloat = mRightVolFloat = -1.0; 5286 } 5287 5288 if (last) { 5289 sp<Track> previousTrack = mPreviousTrack.promote(); 5290 if (previousTrack != 0) { 5291 if (track != previousTrack.get()) { 5292 // Flush any data still being written from last track 5293 mBytesRemaining = 0; 5294 if (mPausedBytesRemaining) { 5295 // Last track was paused so we also need to flush saved 5296 // mixbuffer state and invalidate track so that it will 5297 // re-submit that unwritten data when it is next resumed 5298 mPausedBytesRemaining = 0; 5299 // Invalidate is a bit drastic - would be more efficient 5300 // to have a flag to tell client that some of the 5301 // previously written data was lost 5302 previousTrack->invalidate(); 5303 } 5304 // flush data already sent to the DSP if changing audio session as audio 5305 // comes from a different source. Also invalidate previous track to force a 5306 // seek when resuming. 5307 if (previousTrack->sessionId() != track->sessionId()) { 5308 previousTrack->invalidate(); 5309 } 5310 } 5311 } 5312 mPreviousTrack = track; 5313 // reset retry count 5314 track->mRetryCount = kMaxTrackRetriesOffload; 5315 mActiveTrack = t; 5316 mixerStatus = MIXER_TRACKS_READY; 5317 } 5318 } else { 5319 ALOGVV("OffloadThread: track %d s=%08x [NOT READY]", track->name(), cblk->mServer); 5320 if (track->isStopping_1()) { 5321 // Hardware buffer can hold a large amount of audio so we must 5322 // wait for all current track's data to drain before we say 5323 // that the track is stopped. 5324 if (mBytesRemaining == 0) { 5325 // Only start draining when all data in mixbuffer 5326 // has been written 5327 ALOGV("OffloadThread: underrun and STOPPING_1 -> draining, STOPPING_2"); 5328 track->mState = TrackBase::STOPPING_2; // so presentation completes after drain 5329 // do not drain if no data was ever sent to HAL (mStandby == true) 5330 if (last && !mStandby) { 5331 // do not modify drain sequence if we are already draining. This happens 5332 // when resuming from pause after drain. 5333 if ((mDrainSequence & 1) == 0) { 5334 mSleepTimeUs = 0; 5335 mStandbyTimeNs = systemTime() + mStandbyDelayNs; 5336 mixerStatus = MIXER_DRAIN_TRACK; 5337 mDrainSequence += 2; 5338 } 5339 if (mHwPaused) { 5340 // It is possible to move from PAUSED to STOPPING_1 without 5341 // a resume so we must ensure hardware is running 5342 doHwResume = true; 5343 mHwPaused = false; 5344 } 5345 } 5346 } 5347 } else if (track->isStopping_2()) { 5348 // Drain has completed or we are in standby, signal presentation complete 5349 if (!(mDrainSequence & 1) || !last || mStandby) { 5350 track->mState = TrackBase::STOPPED; 5351 size_t audioHALFrames = 5352 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 5353 int64_t framesWritten = 5354 mBytesWritten / mOutput->getFrameSize(); 5355 track->presentationComplete(framesWritten, audioHALFrames); 5356 track->reset(); 5357 tracksToRemove->add(track); 5358 } 5359 } else { 5360 // No buffers for this track. Give it a few chances to 5361 // fill a buffer, then remove it from active list. 5362 if (--(track->mRetryCount) <= 0) { 5363 ALOGV("OffloadThread: BUFFER TIMEOUT: remove(%d) from active list", 5364 track->name()); 5365 tracksToRemove->add(track); 5366 // indicate to client process that the track was disabled because of underrun; 5367 // it will then automatically call start() when data is available 5368 track->disable(); 5369 } else if (last){ 5370 mixerStatus = MIXER_TRACKS_ENABLED; 5371 } 5372 } 5373 } 5374 // compute volume for this track 5375 processVolume_l(track, last); 5376 } 5377 5378 // make sure the pause/flush/resume sequence is executed in the right order. 5379 // If a flush is pending and a track is active but the HW is not paused, force a HW pause 5380 // before flush and then resume HW. This can happen in case of pause/flush/resume 5381 // if resume is received before pause is executed. 5382 if (!mStandby && (doHwPause || (mFlushPending && !mHwPaused && (count != 0)))) { 5383 mOutput->stream->pause(mOutput->stream); 5384 } 5385 if (mFlushPending) { 5386 flushHw_l(); 5387 } 5388 if (!mStandby && doHwResume) { 5389 mOutput->stream->resume(mOutput->stream); 5390 } 5391 5392 // remove all the tracks that need to be... 5393 removeTracks_l(*tracksToRemove); 5394 5395 return mixerStatus; 5396} 5397 5398// must be called with thread mutex locked 5399bool AudioFlinger::OffloadThread::waitingAsyncCallback_l() 5400{ 5401 ALOGVV("waitingAsyncCallback_l mWriteAckSequence %d mDrainSequence %d", 5402 mWriteAckSequence, mDrainSequence); 5403 if (mUseAsyncWrite && ((mWriteAckSequence & 1) || (mDrainSequence & 1))) { 5404 return true; 5405 } 5406 return false; 5407} 5408 5409bool AudioFlinger::OffloadThread::waitingAsyncCallback() 5410{ 5411 Mutex::Autolock _l(mLock); 5412 return waitingAsyncCallback_l(); 5413} 5414 5415void AudioFlinger::OffloadThread::flushHw_l() 5416{ 5417 DirectOutputThread::flushHw_l(); 5418 // Flush anything still waiting in the mixbuffer 5419 mCurrentWriteLength = 0; 5420 mBytesRemaining = 0; 5421 mPausedWriteLength = 0; 5422 mPausedBytesRemaining = 0; 5423 // reset bytes written count to reflect that DSP buffers are empty after flush. 5424 mBytesWritten = 0; 5425 5426 if (mUseAsyncWrite) { 5427 // discard any pending drain or write ack by incrementing sequence 5428 mWriteAckSequence = (mWriteAckSequence + 2) & ~1; 5429 mDrainSequence = (mDrainSequence + 2) & ~1; 5430 ALOG_ASSERT(mCallbackThread != 0); 5431 mCallbackThread->setWriteBlocked(mWriteAckSequence); 5432 mCallbackThread->setDraining(mDrainSequence); 5433 } 5434} 5435 5436uint32_t AudioFlinger::OffloadThread::activeSleepTimeUs() const 5437{ 5438 uint32_t time; 5439 if (audio_has_proportional_frames(mFormat)) { 5440 time = PlaybackThread::activeSleepTimeUs(); 5441 } else { 5442 // sleep time is half the duration of an audio HAL buffer. 5443 // Note: This can be problematic in case of underrun with variable bit rate and 5444 // current rate is much less than initial rate. 5445 time = (uint32_t)max(kDirectMinSleepTimeUs, mBufferDurationUs / 2); 5446 } 5447 return time; 5448} 5449 5450// ---------------------------------------------------------------------------- 5451 5452AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 5453 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id, bool systemReady) 5454 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->outDevice(), 5455 systemReady, DUPLICATING), 5456 mWaitTimeMs(UINT_MAX) 5457{ 5458 addOutputTrack(mainThread); 5459} 5460 5461AudioFlinger::DuplicatingThread::~DuplicatingThread() 5462{ 5463 for (size_t i = 0; i < mOutputTracks.size(); i++) { 5464 mOutputTracks[i]->destroy(); 5465 } 5466} 5467 5468void AudioFlinger::DuplicatingThread::threadLoop_mix() 5469{ 5470 // mix buffers... 5471 if (outputsReady(outputTracks)) { 5472 mAudioMixer->process(); 5473 } else { 5474 if (mMixerBufferValid) { 5475 memset(mMixerBuffer, 0, mMixerBufferSize); 5476 } else { 5477 memset(mSinkBuffer, 0, mSinkBufferSize); 5478 } 5479 } 5480 mSleepTimeUs = 0; 5481 writeFrames = mNormalFrameCount; 5482 mCurrentWriteLength = mSinkBufferSize; 5483 mStandbyTimeNs = systemTime() + mStandbyDelayNs; 5484} 5485 5486void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() 5487{ 5488 if (mSleepTimeUs == 0) { 5489 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 5490 mSleepTimeUs = mActiveSleepTimeUs; 5491 } else { 5492 mSleepTimeUs = mIdleSleepTimeUs; 5493 } 5494 } else if (mBytesWritten != 0) { 5495 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 5496 writeFrames = mNormalFrameCount; 5497 memset(mSinkBuffer, 0, mSinkBufferSize); 5498 } else { 5499 // flush remaining overflow buffers in output tracks 5500 writeFrames = 0; 5501 } 5502 mSleepTimeUs = 0; 5503 } 5504} 5505 5506ssize_t AudioFlinger::DuplicatingThread::threadLoop_write() 5507{ 5508 for (size_t i = 0; i < outputTracks.size(); i++) { 5509 outputTracks[i]->write(mSinkBuffer, writeFrames); 5510 } 5511 mStandby = false; 5512 return (ssize_t)mSinkBufferSize; 5513} 5514 5515void AudioFlinger::DuplicatingThread::threadLoop_standby() 5516{ 5517 // DuplicatingThread implements standby by stopping all tracks 5518 for (size_t i = 0; i < outputTracks.size(); i++) { 5519 outputTracks[i]->stop(); 5520 } 5521} 5522 5523void AudioFlinger::DuplicatingThread::saveOutputTracks() 5524{ 5525 outputTracks = mOutputTracks; 5526} 5527 5528void AudioFlinger::DuplicatingThread::clearOutputTracks() 5529{ 5530 outputTracks.clear(); 5531} 5532 5533void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 5534{ 5535 Mutex::Autolock _l(mLock); 5536 // The downstream MixerThread consumes thread->frameCount() amount of frames per mix pass. 5537 // Adjust for thread->sampleRate() to determine minimum buffer frame count. 5538 // Then triple buffer because Threads do not run synchronously and may not be clock locked. 5539 const size_t frameCount = 5540 3 * sourceFramesNeeded(mSampleRate, thread->frameCount(), thread->sampleRate()); 5541 // TODO: Consider asynchronous sample rate conversion to handle clock disparity 5542 // from different OutputTracks and their associated MixerThreads (e.g. one may 5543 // nearly empty and the other may be dropping data). 5544 5545 sp<OutputTrack> outputTrack = new OutputTrack(thread, 5546 this, 5547 mSampleRate, 5548 mFormat, 5549 mChannelMask, 5550 frameCount, 5551 IPCThreadState::self()->getCallingUid()); 5552 if (outputTrack->cblk() != NULL) { 5553 thread->setStreamVolume(AUDIO_STREAM_PATCH, 1.0f); 5554 mOutputTracks.add(outputTrack); 5555 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack.get(), thread); 5556 updateWaitTime_l(); 5557 } 5558} 5559 5560void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 5561{ 5562 Mutex::Autolock _l(mLock); 5563 for (size_t i = 0; i < mOutputTracks.size(); i++) { 5564 if (mOutputTracks[i]->thread() == thread) { 5565 mOutputTracks[i]->destroy(); 5566 mOutputTracks.removeAt(i); 5567 updateWaitTime_l(); 5568 if (thread->getOutput() == mOutput) { 5569 mOutput = NULL; 5570 } 5571 return; 5572 } 5573 } 5574 ALOGV("removeOutputTrack(): unknown thread: %p", thread); 5575} 5576 5577// caller must hold mLock 5578void AudioFlinger::DuplicatingThread::updateWaitTime_l() 5579{ 5580 mWaitTimeMs = UINT_MAX; 5581 for (size_t i = 0; i < mOutputTracks.size(); i++) { 5582 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 5583 if (strong != 0) { 5584 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 5585 if (waitTimeMs < mWaitTimeMs) { 5586 mWaitTimeMs = waitTimeMs; 5587 } 5588 } 5589 } 5590} 5591 5592 5593bool AudioFlinger::DuplicatingThread::outputsReady( 5594 const SortedVector< sp<OutputTrack> > &outputTracks) 5595{ 5596 for (size_t i = 0; i < outputTracks.size(); i++) { 5597 sp<ThreadBase> thread = outputTracks[i]->thread().promote(); 5598 if (thread == 0) { 5599 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", 5600 outputTracks[i].get()); 5601 return false; 5602 } 5603 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 5604 // see note at standby() declaration 5605 if (playbackThread->standby() && !playbackThread->isSuspended()) { 5606 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), 5607 thread.get()); 5608 return false; 5609 } 5610 } 5611 return true; 5612} 5613 5614uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() const 5615{ 5616 return (mWaitTimeMs * 1000) / 2; 5617} 5618 5619void AudioFlinger::DuplicatingThread::cacheParameters_l() 5620{ 5621 // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first 5622 updateWaitTime_l(); 5623 5624 MixerThread::cacheParameters_l(); 5625} 5626 5627// ---------------------------------------------------------------------------- 5628// Record 5629// ---------------------------------------------------------------------------- 5630 5631AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 5632 AudioStreamIn *input, 5633 audio_io_handle_t id, 5634 audio_devices_t outDevice, 5635 audio_devices_t inDevice, 5636 bool systemReady 5637#ifdef TEE_SINK 5638 , const sp<NBAIO_Sink>& teeSink 5639#endif 5640 ) : 5641 ThreadBase(audioFlinger, id, outDevice, inDevice, RECORD, systemReady), 5642 mInput(input), mActiveTracksGen(0), mRsmpInBuffer(NULL), 5643 // mRsmpInFrames and mRsmpInFramesP2 are set by readInputParameters_l() 5644 mRsmpInRear(0) 5645#ifdef TEE_SINK 5646 , mTeeSink(teeSink) 5647#endif 5648 , mReadOnlyHeap(new MemoryDealer(kRecordThreadReadOnlyHeapSize, 5649 "RecordThreadRO", MemoryHeapBase::READ_ONLY)) 5650 // mFastCapture below 5651 , mFastCaptureFutex(0) 5652 // mInputSource 5653 // mPipeSink 5654 // mPipeSource 5655 , mPipeFramesP2(0) 5656 // mPipeMemory 5657 // mFastCaptureNBLogWriter 5658 , mFastTrackAvail(false) 5659{ 5660 snprintf(mThreadName, kThreadNameLength, "AudioIn_%X", id); 5661 mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mThreadName); 5662 5663 readInputParameters_l(); 5664 5665 // create an NBAIO source for the HAL input stream, and negotiate 5666 mInputSource = new AudioStreamInSource(input->stream); 5667 size_t numCounterOffers = 0; 5668 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount, mFormat)}; 5669#if !LOG_NDEBUG 5670 ssize_t index = 5671#else 5672 (void) 5673#endif 5674 mInputSource->negotiate(offers, 1, NULL, numCounterOffers); 5675 ALOG_ASSERT(index == 0); 5676 5677 // initialize fast capture depending on configuration 5678 bool initFastCapture; 5679 switch (kUseFastCapture) { 5680 case FastCapture_Never: 5681 initFastCapture = false; 5682 break; 5683 case FastCapture_Always: 5684 initFastCapture = true; 5685 break; 5686 case FastCapture_Static: 5687 initFastCapture = (mFrameCount * 1000) / mSampleRate < kMinNormalCaptureBufferSizeMs; 5688 break; 5689 // case FastCapture_Dynamic: 5690 } 5691 5692 if (initFastCapture) { 5693 // create a Pipe for FastCapture to write to, and for us and fast tracks to read from 5694 NBAIO_Format format = mInputSource->format(); 5695 size_t pipeFramesP2 = roundup(mSampleRate / 25); // double-buffering of 20 ms each 5696 size_t pipeSize = pipeFramesP2 * Format_frameSize(format); 5697 void *pipeBuffer; 5698 const sp<MemoryDealer> roHeap(readOnlyHeap()); 5699 sp<IMemory> pipeMemory; 5700 if ((roHeap == 0) || 5701 (pipeMemory = roHeap->allocate(pipeSize)) == 0 || 5702 (pipeBuffer = pipeMemory->pointer()) == NULL) { 5703 ALOGE("not enough memory for pipe buffer size=%zu", pipeSize); 5704 goto failed; 5705 } 5706 // pipe will be shared directly with fast clients, so clear to avoid leaking old information 5707 memset(pipeBuffer, 0, pipeSize); 5708 Pipe *pipe = new Pipe(pipeFramesP2, format, pipeBuffer); 5709 const NBAIO_Format offers[1] = {format}; 5710 size_t numCounterOffers = 0; 5711 ssize_t index = pipe->negotiate(offers, 1, NULL, numCounterOffers); 5712 ALOG_ASSERT(index == 0); 5713 mPipeSink = pipe; 5714 PipeReader *pipeReader = new PipeReader(*pipe); 5715 numCounterOffers = 0; 5716 index = pipeReader->negotiate(offers, 1, NULL, numCounterOffers); 5717 ALOG_ASSERT(index == 0); 5718 mPipeSource = pipeReader; 5719 mPipeFramesP2 = pipeFramesP2; 5720 mPipeMemory = pipeMemory; 5721 5722 // create fast capture 5723 mFastCapture = new FastCapture(); 5724 FastCaptureStateQueue *sq = mFastCapture->sq(); 5725#ifdef STATE_QUEUE_DUMP 5726 // FIXME 5727#endif 5728 FastCaptureState *state = sq->begin(); 5729 state->mCblk = NULL; 5730 state->mInputSource = mInputSource.get(); 5731 state->mInputSourceGen++; 5732 state->mPipeSink = pipe; 5733 state->mPipeSinkGen++; 5734 state->mFrameCount = mFrameCount; 5735 state->mCommand = FastCaptureState::COLD_IDLE; 5736 // already done in constructor initialization list 5737 //mFastCaptureFutex = 0; 5738 state->mColdFutexAddr = &mFastCaptureFutex; 5739 state->mColdGen++; 5740 state->mDumpState = &mFastCaptureDumpState; 5741#ifdef TEE_SINK 5742 // FIXME 5743#endif 5744 mFastCaptureNBLogWriter = audioFlinger->newWriter_l(kFastCaptureLogSize, "FastCapture"); 5745 state->mNBLogWriter = mFastCaptureNBLogWriter.get(); 5746 sq->end(); 5747 sq->push(FastCaptureStateQueue::BLOCK_UNTIL_PUSHED); 5748 5749 // start the fast capture 5750 mFastCapture->run("FastCapture", ANDROID_PRIORITY_URGENT_AUDIO); 5751 pid_t tid = mFastCapture->getTid(); 5752 sendPrioConfigEvent(getpid_cached, tid, kPriorityFastCapture); 5753#ifdef AUDIO_WATCHDOG 5754 // FIXME 5755#endif 5756 5757 mFastTrackAvail = true; 5758 } 5759failed: ; 5760 5761 // FIXME mNormalSource 5762} 5763 5764AudioFlinger::RecordThread::~RecordThread() 5765{ 5766 if (mFastCapture != 0) { 5767 FastCaptureStateQueue *sq = mFastCapture->sq(); 5768 FastCaptureState *state = sq->begin(); 5769 if (state->mCommand == FastCaptureState::COLD_IDLE) { 5770 int32_t old = android_atomic_inc(&mFastCaptureFutex); 5771 if (old == -1) { 5772 (void) syscall(__NR_futex, &mFastCaptureFutex, FUTEX_WAKE_PRIVATE, 1); 5773 } 5774 } 5775 state->mCommand = FastCaptureState::EXIT; 5776 sq->end(); 5777 sq->push(FastCaptureStateQueue::BLOCK_UNTIL_PUSHED); 5778 mFastCapture->join(); 5779 mFastCapture.clear(); 5780 } 5781 mAudioFlinger->unregisterWriter(mFastCaptureNBLogWriter); 5782 mAudioFlinger->unregisterWriter(mNBLogWriter); 5783 free(mRsmpInBuffer); 5784} 5785 5786void AudioFlinger::RecordThread::onFirstRef() 5787{ 5788 run(mThreadName, PRIORITY_URGENT_AUDIO); 5789} 5790 5791bool AudioFlinger::RecordThread::threadLoop() 5792{ 5793 nsecs_t lastWarning = 0; 5794 5795 inputStandBy(); 5796 5797reacquire_wakelock: 5798 sp<RecordTrack> activeTrack; 5799 int activeTracksGen; 5800 { 5801 Mutex::Autolock _l(mLock); 5802 size_t size = mActiveTracks.size(); 5803 activeTracksGen = mActiveTracksGen; 5804 if (size > 0) { 5805 // FIXME an arbitrary choice 5806 activeTrack = mActiveTracks[0]; 5807 acquireWakeLock_l(activeTrack->uid()); 5808 if (size > 1) { 5809 SortedVector<int> tmp; 5810 for (size_t i = 0; i < size; i++) { 5811 tmp.add(mActiveTracks[i]->uid()); 5812 } 5813 updateWakeLockUids_l(tmp); 5814 } 5815 } else { 5816 acquireWakeLock_l(-1); 5817 } 5818 } 5819 5820 // used to request a deferred sleep, to be executed later while mutex is unlocked 5821 uint32_t sleepUs = 0; 5822 5823 // loop while there is work to do 5824 for (;;) { 5825 Vector< sp<EffectChain> > effectChains; 5826 5827 // sleep with mutex unlocked 5828 if (sleepUs > 0) { 5829 ATRACE_BEGIN("sleep"); 5830 usleep(sleepUs); 5831 ATRACE_END(); 5832 sleepUs = 0; 5833 } 5834 5835 // activeTracks accumulates a copy of a subset of mActiveTracks 5836 Vector< sp<RecordTrack> > activeTracks; 5837 5838 // reference to the (first and only) active fast track 5839 sp<RecordTrack> fastTrack; 5840 5841 // reference to a fast track which is about to be removed 5842 sp<RecordTrack> fastTrackToRemove; 5843 5844 { // scope for mLock 5845 Mutex::Autolock _l(mLock); 5846 5847 processConfigEvents_l(); 5848 5849 // check exitPending here because checkForNewParameters_l() and 5850 // checkForNewParameters_l() can temporarily release mLock 5851 if (exitPending()) { 5852 break; 5853 } 5854 5855 // if no active track(s), then standby and release wakelock 5856 size_t size = mActiveTracks.size(); 5857 if (size == 0) { 5858 standbyIfNotAlreadyInStandby(); 5859 // exitPending() can't become true here 5860 releaseWakeLock_l(); 5861 ALOGV("RecordThread: loop stopping"); 5862 // go to sleep 5863 mWaitWorkCV.wait(mLock); 5864 ALOGV("RecordThread: loop starting"); 5865 goto reacquire_wakelock; 5866 } 5867 5868 if (mActiveTracksGen != activeTracksGen) { 5869 activeTracksGen = mActiveTracksGen; 5870 SortedVector<int> tmp; 5871 for (size_t i = 0; i < size; i++) { 5872 tmp.add(mActiveTracks[i]->uid()); 5873 } 5874 updateWakeLockUids_l(tmp); 5875 } 5876 5877 bool doBroadcast = false; 5878 for (size_t i = 0; i < size; ) { 5879 5880 activeTrack = mActiveTracks[i]; 5881 if (activeTrack->isTerminated()) { 5882 if (activeTrack->isFastTrack()) { 5883 ALOG_ASSERT(fastTrackToRemove == 0); 5884 fastTrackToRemove = activeTrack; 5885 } 5886 removeTrack_l(activeTrack); 5887 mActiveTracks.remove(activeTrack); 5888 mActiveTracksGen++; 5889 size--; 5890 continue; 5891 } 5892 5893 TrackBase::track_state activeTrackState = activeTrack->mState; 5894 switch (activeTrackState) { 5895 5896 case TrackBase::PAUSING: 5897 mActiveTracks.remove(activeTrack); 5898 mActiveTracksGen++; 5899 doBroadcast = true; 5900 size--; 5901 continue; 5902 5903 case TrackBase::STARTING_1: 5904 sleepUs = 10000; 5905 i++; 5906 continue; 5907 5908 case TrackBase::STARTING_2: 5909 doBroadcast = true; 5910 mStandby = false; 5911 activeTrack->mState = TrackBase::ACTIVE; 5912 break; 5913 5914 case TrackBase::ACTIVE: 5915 break; 5916 5917 case TrackBase::IDLE: 5918 i++; 5919 continue; 5920 5921 default: 5922 LOG_ALWAYS_FATAL("Unexpected activeTrackState %d", activeTrackState); 5923 } 5924 5925 activeTracks.add(activeTrack); 5926 i++; 5927 5928 if (activeTrack->isFastTrack()) { 5929 ALOG_ASSERT(!mFastTrackAvail); 5930 ALOG_ASSERT(fastTrack == 0); 5931 fastTrack = activeTrack; 5932 } 5933 } 5934 if (doBroadcast) { 5935 mStartStopCond.broadcast(); 5936 } 5937 5938 // sleep if there are no active tracks to process 5939 if (activeTracks.size() == 0) { 5940 if (sleepUs == 0) { 5941 sleepUs = kRecordThreadSleepUs; 5942 } 5943 continue; 5944 } 5945 sleepUs = 0; 5946 5947 lockEffectChains_l(effectChains); 5948 } 5949 5950 // thread mutex is now unlocked, mActiveTracks unknown, activeTracks.size() > 0 5951 5952 size_t size = effectChains.size(); 5953 for (size_t i = 0; i < size; i++) { 5954 // thread mutex is not locked, but effect chain is locked 5955 effectChains[i]->process_l(); 5956 } 5957 5958 // Push a new fast capture state if fast capture is not already running, or cblk change 5959 if (mFastCapture != 0) { 5960 FastCaptureStateQueue *sq = mFastCapture->sq(); 5961 FastCaptureState *state = sq->begin(); 5962 bool didModify = false; 5963 FastCaptureStateQueue::block_t block = FastCaptureStateQueue::BLOCK_UNTIL_PUSHED; 5964 if (state->mCommand != FastCaptureState::READ_WRITE /* FIXME && 5965 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)*/) { 5966 if (state->mCommand == FastCaptureState::COLD_IDLE) { 5967 int32_t old = android_atomic_inc(&mFastCaptureFutex); 5968 if (old == -1) { 5969 (void) syscall(__NR_futex, &mFastCaptureFutex, FUTEX_WAKE_PRIVATE, 1); 5970 } 5971 } 5972 state->mCommand = FastCaptureState::READ_WRITE; 5973#if 0 // FIXME 5974 mFastCaptureDumpState.increaseSamplingN(mAudioFlinger->isLowRamDevice() ? 5975 FastThreadDumpState::kSamplingNforLowRamDevice : 5976 FastThreadDumpState::kSamplingN); 5977#endif 5978 didModify = true; 5979 } 5980 audio_track_cblk_t *cblkOld = state->mCblk; 5981 audio_track_cblk_t *cblkNew = fastTrack != 0 ? fastTrack->cblk() : NULL; 5982 if (cblkNew != cblkOld) { 5983 state->mCblk = cblkNew; 5984 // block until acked if removing a fast track 5985 if (cblkOld != NULL) { 5986 block = FastCaptureStateQueue::BLOCK_UNTIL_ACKED; 5987 } 5988 didModify = true; 5989 } 5990 sq->end(didModify); 5991 if (didModify) { 5992 sq->push(block); 5993#if 0 5994 if (kUseFastCapture == FastCapture_Dynamic) { 5995 mNormalSource = mPipeSource; 5996 } 5997#endif 5998 } 5999 } 6000 6001 // now run the fast track destructor with thread mutex unlocked 6002 fastTrackToRemove.clear(); 6003 6004 // Read from HAL to keep up with fastest client if multiple active tracks, not slowest one. 6005 // Only the client(s) that are too slow will overrun. But if even the fastest client is too 6006 // slow, then this RecordThread will overrun by not calling HAL read often enough. 6007 // If destination is non-contiguous, first read past the nominal end of buffer, then 6008 // copy to the right place. Permitted because mRsmpInBuffer was over-allocated. 6009 6010 int32_t rear = mRsmpInRear & (mRsmpInFramesP2 - 1); 6011 ssize_t framesRead; 6012 6013 // If an NBAIO source is present, use it to read the normal capture's data 6014 if (mPipeSource != 0) { 6015 size_t framesToRead = mBufferSize / mFrameSize; 6016 framesRead = mPipeSource->read((uint8_t*)mRsmpInBuffer + rear * mFrameSize, 6017 framesToRead); 6018 if (framesRead == 0) { 6019 // since pipe is non-blocking, simulate blocking input 6020 sleepUs = (framesToRead * 1000000LL) / mSampleRate; 6021 } 6022 // otherwise use the HAL / AudioStreamIn directly 6023 } else { 6024 ATRACE_BEGIN("read"); 6025 ssize_t bytesRead = mInput->stream->read(mInput->stream, 6026 (uint8_t*)mRsmpInBuffer + rear * mFrameSize, mBufferSize); 6027 ATRACE_END(); 6028 if (bytesRead < 0) { 6029 framesRead = bytesRead; 6030 } else { 6031 framesRead = bytesRead / mFrameSize; 6032 } 6033 } 6034 6035 // Update server timestamp with server stats 6036 // systemTime() is optional if the hardware supports timestamps. 6037 mTimestamp.mPosition[ExtendedTimestamp::LOCATION_SERVER] += framesRead; 6038 mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_SERVER] = systemTime(); 6039 6040 // Update server timestamp with kernel stats 6041 if (mInput->stream->get_capture_position != nullptr) { 6042 int64_t position, time; 6043 int ret = mInput->stream->get_capture_position(mInput->stream, &position, &time); 6044 if (ret == NO_ERROR) { 6045 mTimestamp.mPosition[ExtendedTimestamp::LOCATION_KERNEL] = position; 6046 mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL] = time; 6047 // Note: In general record buffers should tend to be empty in 6048 // a properly running pipeline. 6049 // 6050 // Also, it is not advantageous to call get_presentation_position during the read 6051 // as the read obtains a lock, preventing the timestamp call from executing. 6052 } 6053 } 6054 // Use this to track timestamp information 6055 // ALOGD("%s", mTimestamp.toString().c_str()); 6056 6057 if (framesRead < 0 || (framesRead == 0 && mPipeSource == 0)) { 6058 ALOGE("read failed: framesRead=%zd", framesRead); 6059 // Force input into standby so that it tries to recover at next read attempt 6060 inputStandBy(); 6061 sleepUs = kRecordThreadSleepUs; 6062 } 6063 if (framesRead <= 0) { 6064 goto unlock; 6065 } 6066 ALOG_ASSERT(framesRead > 0); 6067 6068 if (mTeeSink != 0) { 6069 (void) mTeeSink->write((uint8_t*)mRsmpInBuffer + rear * mFrameSize, framesRead); 6070 } 6071 // If destination is non-contiguous, we now correct for reading past end of buffer. 6072 { 6073 size_t part1 = mRsmpInFramesP2 - rear; 6074 if ((size_t) framesRead > part1) { 6075 memcpy(mRsmpInBuffer, (uint8_t*)mRsmpInBuffer + mRsmpInFramesP2 * mFrameSize, 6076 (framesRead - part1) * mFrameSize); 6077 } 6078 } 6079 rear = mRsmpInRear += framesRead; 6080 6081 size = activeTracks.size(); 6082 // loop over each active track 6083 for (size_t i = 0; i < size; i++) { 6084 activeTrack = activeTracks[i]; 6085 6086 // skip fast tracks, as those are handled directly by FastCapture 6087 if (activeTrack->isFastTrack()) { 6088 continue; 6089 } 6090 6091 // TODO: This code probably should be moved to RecordTrack. 6092 // TODO: Update the activeTrack buffer converter in case of reconfigure. 6093 6094 enum { 6095 OVERRUN_UNKNOWN, 6096 OVERRUN_TRUE, 6097 OVERRUN_FALSE 6098 } overrun = OVERRUN_UNKNOWN; 6099 6100 // loop over getNextBuffer to handle circular sink 6101 for (;;) { 6102 6103 activeTrack->mSink.frameCount = ~0; 6104 status_t status = activeTrack->getNextBuffer(&activeTrack->mSink); 6105 size_t framesOut = activeTrack->mSink.frameCount; 6106 LOG_ALWAYS_FATAL_IF((status == OK) != (framesOut > 0)); 6107 6108 // check available frames and handle overrun conditions 6109 // if the record track isn't draining fast enough. 6110 bool hasOverrun; 6111 size_t framesIn; 6112 activeTrack->mResamplerBufferProvider->sync(&framesIn, &hasOverrun); 6113 if (hasOverrun) { 6114 overrun = OVERRUN_TRUE; 6115 } 6116 if (framesOut == 0 || framesIn == 0) { 6117 break; 6118 } 6119 6120 // Don't allow framesOut to be larger than what is possible with resampling 6121 // from framesIn. 6122 // This isn't strictly necessary but helps limit buffer resizing in 6123 // RecordBufferConverter. TODO: remove when no longer needed. 6124 framesOut = min(framesOut, 6125 destinationFramesPossible( 6126 framesIn, mSampleRate, activeTrack->mSampleRate)); 6127 // process frames from the RecordThread buffer provider to the RecordTrack buffer 6128 framesOut = activeTrack->mRecordBufferConverter->convert( 6129 activeTrack->mSink.raw, activeTrack->mResamplerBufferProvider, framesOut); 6130 6131 if (framesOut > 0 && (overrun == OVERRUN_UNKNOWN)) { 6132 overrun = OVERRUN_FALSE; 6133 } 6134 6135 if (activeTrack->mFramesToDrop == 0) { 6136 if (framesOut > 0) { 6137 activeTrack->mSink.frameCount = framesOut; 6138 activeTrack->releaseBuffer(&activeTrack->mSink); 6139 } 6140 } else { 6141 // FIXME could do a partial drop of framesOut 6142 if (activeTrack->mFramesToDrop > 0) { 6143 activeTrack->mFramesToDrop -= framesOut; 6144 if (activeTrack->mFramesToDrop <= 0) { 6145 activeTrack->clearSyncStartEvent(); 6146 } 6147 } else { 6148 activeTrack->mFramesToDrop += framesOut; 6149 if (activeTrack->mFramesToDrop >= 0 || activeTrack->mSyncStartEvent == 0 || 6150 activeTrack->mSyncStartEvent->isCancelled()) { 6151 ALOGW("Synced record %s, session %d, trigger session %d", 6152 (activeTrack->mFramesToDrop >= 0) ? "timed out" : "cancelled", 6153 activeTrack->sessionId(), 6154 (activeTrack->mSyncStartEvent != 0) ? 6155 activeTrack->mSyncStartEvent->triggerSession() : 6156 AUDIO_SESSION_NONE); 6157 activeTrack->clearSyncStartEvent(); 6158 } 6159 } 6160 } 6161 6162 if (framesOut == 0) { 6163 break; 6164 } 6165 } 6166 6167 switch (overrun) { 6168 case OVERRUN_TRUE: 6169 // client isn't retrieving buffers fast enough 6170 if (!activeTrack->setOverflow()) { 6171 nsecs_t now = systemTime(); 6172 // FIXME should lastWarning per track? 6173 if ((now - lastWarning) > kWarningThrottleNs) { 6174 ALOGW("RecordThread: buffer overflow"); 6175 lastWarning = now; 6176 } 6177 } 6178 break; 6179 case OVERRUN_FALSE: 6180 activeTrack->clearOverflow(); 6181 break; 6182 case OVERRUN_UNKNOWN: 6183 break; 6184 } 6185 6186 // update frame information and push timestamp out 6187 activeTrack->updateTrackFrameInfo( 6188 activeTrack->mServerProxy->framesReleased(), 6189 mTimestamp.mPosition[ExtendedTimestamp::LOCATION_SERVER], 6190 mSampleRate, mTimestamp); 6191 } 6192 6193unlock: 6194 // enable changes in effect chain 6195 unlockEffectChains(effectChains); 6196 // effectChains doesn't need to be cleared, since it is cleared by destructor at scope end 6197 } 6198 6199 standbyIfNotAlreadyInStandby(); 6200 6201 { 6202 Mutex::Autolock _l(mLock); 6203 for (size_t i = 0; i < mTracks.size(); i++) { 6204 sp<RecordTrack> track = mTracks[i]; 6205 track->invalidate(); 6206 } 6207 mActiveTracks.clear(); 6208 mActiveTracksGen++; 6209 mStartStopCond.broadcast(); 6210 } 6211 6212 releaseWakeLock(); 6213 6214 ALOGV("RecordThread %p exiting", this); 6215 return false; 6216} 6217 6218void AudioFlinger::RecordThread::standbyIfNotAlreadyInStandby() 6219{ 6220 if (!mStandby) { 6221 inputStandBy(); 6222 mStandby = true; 6223 } 6224} 6225 6226void AudioFlinger::RecordThread::inputStandBy() 6227{ 6228 // Idle the fast capture if it's currently running 6229 if (mFastCapture != 0) { 6230 FastCaptureStateQueue *sq = mFastCapture->sq(); 6231 FastCaptureState *state = sq->begin(); 6232 if (!(state->mCommand & FastCaptureState::IDLE)) { 6233 state->mCommand = FastCaptureState::COLD_IDLE; 6234 state->mColdFutexAddr = &mFastCaptureFutex; 6235 state->mColdGen++; 6236 mFastCaptureFutex = 0; 6237 sq->end(); 6238 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 6239 sq->push(FastCaptureStateQueue::BLOCK_UNTIL_ACKED); 6240#if 0 6241 if (kUseFastCapture == FastCapture_Dynamic) { 6242 // FIXME 6243 } 6244#endif 6245#ifdef AUDIO_WATCHDOG 6246 // FIXME 6247#endif 6248 } else { 6249 sq->end(false /*didModify*/); 6250 } 6251 } 6252 mInput->stream->common.standby(&mInput->stream->common); 6253} 6254 6255// RecordThread::createRecordTrack_l() must be called with AudioFlinger::mLock held 6256sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 6257 const sp<AudioFlinger::Client>& client, 6258 uint32_t sampleRate, 6259 audio_format_t format, 6260 audio_channel_mask_t channelMask, 6261 size_t *pFrameCount, 6262 audio_session_t sessionId, 6263 size_t *notificationFrames, 6264 int uid, 6265 IAudioFlinger::track_flags_t *flags, 6266 pid_t tid, 6267 status_t *status) 6268{ 6269 size_t frameCount = *pFrameCount; 6270 sp<RecordTrack> track; 6271 status_t lStatus; 6272 6273 // client expresses a preference for FAST, but we get the final say 6274 if (*flags & IAudioFlinger::TRACK_FAST) { 6275 if ( 6276 // we formerly checked for a callback handler (non-0 tid), 6277 // but that is no longer required for TRANSFER_OBTAIN mode 6278 // 6279 // frame count is not specified, or is exactly the pipe depth 6280 ((frameCount == 0) || (frameCount == mPipeFramesP2)) && 6281 // PCM data 6282 audio_is_linear_pcm(format) && 6283 // hardware format 6284 (format == mFormat) && 6285 // hardware channel mask 6286 (channelMask == mChannelMask) && 6287 // hardware sample rate 6288 (sampleRate == mSampleRate) && 6289 // record thread has an associated fast capture 6290 hasFastCapture() && 6291 // there are sufficient fast track slots available 6292 mFastTrackAvail 6293 ) { 6294 ALOGV("AUDIO_INPUT_FLAG_FAST accepted: frameCount=%zu mFrameCount=%zu", 6295 frameCount, mFrameCount); 6296 } else { 6297 ALOGV("AUDIO_INPUT_FLAG_FAST denied: frameCount=%zu mFrameCount=%zu mPipeFramesP2=%zu " 6298 "format=%#x isLinear=%d channelMask=%#x sampleRate=%u mSampleRate=%u " 6299 "hasFastCapture=%d tid=%d mFastTrackAvail=%d", 6300 frameCount, mFrameCount, mPipeFramesP2, 6301 format, audio_is_linear_pcm(format), channelMask, sampleRate, mSampleRate, 6302 hasFastCapture(), tid, mFastTrackAvail); 6303 *flags &= ~IAudioFlinger::TRACK_FAST; 6304 } 6305 } 6306 6307 // compute track buffer size in frames, and suggest the notification frame count 6308 if (*flags & IAudioFlinger::TRACK_FAST) { 6309 // fast track: frame count is exactly the pipe depth 6310 frameCount = mPipeFramesP2; 6311 // ignore requested notificationFrames, and always notify exactly once every HAL buffer 6312 *notificationFrames = mFrameCount; 6313 } else { 6314 // not fast track: max notification period is resampled equivalent of one HAL buffer time 6315 // or 20 ms if there is a fast capture 6316 // TODO This could be a roundupRatio inline, and const 6317 size_t maxNotificationFrames = ((int64_t) (hasFastCapture() ? mSampleRate/50 : mFrameCount) 6318 * sampleRate + mSampleRate - 1) / mSampleRate; 6319 // minimum number of notification periods is at least kMinNotifications, 6320 // and at least kMinMs rounded up to a whole notification period (minNotificationsByMs) 6321 static const size_t kMinNotifications = 3; 6322 static const uint32_t kMinMs = 30; 6323 // TODO This could be a roundupRatio inline 6324 const size_t minFramesByMs = (sampleRate * kMinMs + 1000 - 1) / 1000; 6325 // TODO This could be a roundupRatio inline 6326 const size_t minNotificationsByMs = (minFramesByMs + maxNotificationFrames - 1) / 6327 maxNotificationFrames; 6328 const size_t minFrameCount = maxNotificationFrames * 6329 max(kMinNotifications, minNotificationsByMs); 6330 frameCount = max(frameCount, minFrameCount); 6331 if (*notificationFrames == 0 || *notificationFrames > maxNotificationFrames) { 6332 *notificationFrames = maxNotificationFrames; 6333 } 6334 } 6335 *pFrameCount = frameCount; 6336 6337 lStatus = initCheck(); 6338 if (lStatus != NO_ERROR) { 6339 ALOGE("createRecordTrack_l() audio driver not initialized"); 6340 goto Exit; 6341 } 6342 6343 { // scope for mLock 6344 Mutex::Autolock _l(mLock); 6345 6346 track = new RecordTrack(this, client, sampleRate, 6347 format, channelMask, frameCount, NULL, sessionId, uid, 6348 *flags, TrackBase::TYPE_DEFAULT); 6349 6350 lStatus = track->initCheck(); 6351 if (lStatus != NO_ERROR) { 6352 ALOGE("createRecordTrack_l() initCheck failed %d; no control block?", lStatus); 6353 // track must be cleared from the caller as the caller has the AF lock 6354 goto Exit; 6355 } 6356 mTracks.add(track); 6357 6358 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 6359 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 6360 mAudioFlinger->btNrecIsOff(); 6361 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 6362 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 6363 6364 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 6365 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 6366 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 6367 // so ask activity manager to do this on our behalf 6368 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 6369 } 6370 } 6371 6372 lStatus = NO_ERROR; 6373 6374Exit: 6375 *status = lStatus; 6376 return track; 6377} 6378 6379status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, 6380 AudioSystem::sync_event_t event, 6381 audio_session_t triggerSession) 6382{ 6383 ALOGV("RecordThread::start event %d, triggerSession %d", event, triggerSession); 6384 sp<ThreadBase> strongMe = this; 6385 status_t status = NO_ERROR; 6386 6387 if (event == AudioSystem::SYNC_EVENT_NONE) { 6388 recordTrack->clearSyncStartEvent(); 6389 } else if (event != AudioSystem::SYNC_EVENT_SAME) { 6390 recordTrack->mSyncStartEvent = mAudioFlinger->createSyncEvent(event, 6391 triggerSession, 6392 recordTrack->sessionId(), 6393 syncStartEventCallback, 6394 recordTrack); 6395 // Sync event can be cancelled by the trigger session if the track is not in a 6396 // compatible state in which case we start record immediately 6397 if (recordTrack->mSyncStartEvent->isCancelled()) { 6398 recordTrack->clearSyncStartEvent(); 6399 } else { 6400 // do not wait for the event for more than AudioSystem::kSyncRecordStartTimeOutMs 6401 recordTrack->mFramesToDrop = - 6402 ((AudioSystem::kSyncRecordStartTimeOutMs * recordTrack->mSampleRate) / 1000); 6403 } 6404 } 6405 6406 { 6407 // This section is a rendezvous between binder thread executing start() and RecordThread 6408 AutoMutex lock(mLock); 6409 if (mActiveTracks.indexOf(recordTrack) >= 0) { 6410 if (recordTrack->mState == TrackBase::PAUSING) { 6411 ALOGV("active record track PAUSING -> ACTIVE"); 6412 recordTrack->mState = TrackBase::ACTIVE; 6413 } else { 6414 ALOGV("active record track state %d", recordTrack->mState); 6415 } 6416 return status; 6417 } 6418 6419 // TODO consider other ways of handling this, such as changing the state to :STARTING and 6420 // adding the track to mActiveTracks after returning from AudioSystem::startInput(), 6421 // or using a separate command thread 6422 recordTrack->mState = TrackBase::STARTING_1; 6423 mActiveTracks.add(recordTrack); 6424 mActiveTracksGen++; 6425 status_t status = NO_ERROR; 6426 if (recordTrack->isExternalTrack()) { 6427 mLock.unlock(); 6428 status = AudioSystem::startInput(mId, recordTrack->sessionId()); 6429 mLock.lock(); 6430 // FIXME should verify that recordTrack is still in mActiveTracks 6431 if (status != NO_ERROR) { 6432 mActiveTracks.remove(recordTrack); 6433 mActiveTracksGen++; 6434 recordTrack->clearSyncStartEvent(); 6435 ALOGV("RecordThread::start error %d", status); 6436 return status; 6437 } 6438 } 6439 // Catch up with current buffer indices if thread is already running. 6440 // This is what makes a new client discard all buffered data. If the track's mRsmpInFront 6441 // was initialized to some value closer to the thread's mRsmpInFront, then the track could 6442 // see previously buffered data before it called start(), but with greater risk of overrun. 6443 6444 recordTrack->mResamplerBufferProvider->reset(); 6445 // clear any converter state as new data will be discontinuous 6446 recordTrack->mRecordBufferConverter->reset(); 6447 recordTrack->mState = TrackBase::STARTING_2; 6448 // signal thread to start 6449 mWaitWorkCV.broadcast(); 6450 if (mActiveTracks.indexOf(recordTrack) < 0) { 6451 ALOGV("Record failed to start"); 6452 status = BAD_VALUE; 6453 goto startError; 6454 } 6455 return status; 6456 } 6457 6458startError: 6459 if (recordTrack->isExternalTrack()) { 6460 AudioSystem::stopInput(mId, recordTrack->sessionId()); 6461 } 6462 recordTrack->clearSyncStartEvent(); 6463 // FIXME I wonder why we do not reset the state here? 6464 return status; 6465} 6466 6467void AudioFlinger::RecordThread::syncStartEventCallback(const wp<SyncEvent>& event) 6468{ 6469 sp<SyncEvent> strongEvent = event.promote(); 6470 6471 if (strongEvent != 0) { 6472 sp<RefBase> ptr = strongEvent->cookie().promote(); 6473 if (ptr != 0) { 6474 RecordTrack *recordTrack = (RecordTrack *)ptr.get(); 6475 recordTrack->handleSyncStartEvent(strongEvent); 6476 } 6477 } 6478} 6479 6480bool AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 6481 ALOGV("RecordThread::stop"); 6482 AutoMutex _l(mLock); 6483 if (mActiveTracks.indexOf(recordTrack) != 0 || recordTrack->mState == TrackBase::PAUSING) { 6484 return false; 6485 } 6486 // note that threadLoop may still be processing the track at this point [without lock] 6487 recordTrack->mState = TrackBase::PAUSING; 6488 // do not wait for mStartStopCond if exiting 6489 if (exitPending()) { 6490 return true; 6491 } 6492 // FIXME incorrect usage of wait: no explicit predicate or loop 6493 mStartStopCond.wait(mLock); 6494 // if we have been restarted, recordTrack is in mActiveTracks here 6495 if (exitPending() || mActiveTracks.indexOf(recordTrack) != 0) { 6496 ALOGV("Record stopped OK"); 6497 return true; 6498 } 6499 return false; 6500} 6501 6502bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event __unused) const 6503{ 6504 return false; 6505} 6506 6507status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event __unused) 6508{ 6509#if 0 // This branch is currently dead code, but is preserved in case it will be needed in future 6510 if (!isValidSyncEvent(event)) { 6511 return BAD_VALUE; 6512 } 6513 6514 audio_session_t eventSession = event->triggerSession(); 6515 status_t ret = NAME_NOT_FOUND; 6516 6517 Mutex::Autolock _l(mLock); 6518 6519 for (size_t i = 0; i < mTracks.size(); i++) { 6520 sp<RecordTrack> track = mTracks[i]; 6521 if (eventSession == track->sessionId()) { 6522 (void) track->setSyncEvent(event); 6523 ret = NO_ERROR; 6524 } 6525 } 6526 return ret; 6527#else 6528 return BAD_VALUE; 6529#endif 6530} 6531 6532// destroyTrack_l() must be called with ThreadBase::mLock held 6533void AudioFlinger::RecordThread::destroyTrack_l(const sp<RecordTrack>& track) 6534{ 6535 track->terminate(); 6536 track->mState = TrackBase::STOPPED; 6537 // active tracks are removed by threadLoop() 6538 if (mActiveTracks.indexOf(track) < 0) { 6539 removeTrack_l(track); 6540 } 6541} 6542 6543void AudioFlinger::RecordThread::removeTrack_l(const sp<RecordTrack>& track) 6544{ 6545 mTracks.remove(track); 6546 // need anything related to effects here? 6547 if (track->isFastTrack()) { 6548 ALOG_ASSERT(!mFastTrackAvail); 6549 mFastTrackAvail = true; 6550 } 6551} 6552 6553void AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 6554{ 6555 dumpInternals(fd, args); 6556 dumpTracks(fd, args); 6557 dumpEffectChains(fd, args); 6558} 6559 6560void AudioFlinger::RecordThread::dumpInternals(int fd, const Vector<String16>& args) 6561{ 6562 dprintf(fd, "\nInput thread %p:\n", this); 6563 6564 dumpBase(fd, args); 6565 6566 if (mActiveTracks.size() == 0) { 6567 dprintf(fd, " No active record clients\n"); 6568 } 6569 dprintf(fd, " Fast capture thread: %s\n", hasFastCapture() ? "yes" : "no"); 6570 dprintf(fd, " Fast track available: %s\n", mFastTrackAvail ? "yes" : "no"); 6571 6572 // Make a non-atomic copy of fast capture dump state so it won't change underneath us 6573 // while we are dumping it. It may be inconsistent, but it won't mutate! 6574 // This is a large object so we place it on the heap. 6575 // FIXME 25972958: Need an intelligent copy constructor that does not touch unused pages. 6576 const FastCaptureDumpState *copy = new FastCaptureDumpState(mFastCaptureDumpState); 6577 copy->dump(fd); 6578 delete copy; 6579} 6580 6581void AudioFlinger::RecordThread::dumpTracks(int fd, const Vector<String16>& args __unused) 6582{ 6583 const size_t SIZE = 256; 6584 char buffer[SIZE]; 6585 String8 result; 6586 6587 size_t numtracks = mTracks.size(); 6588 size_t numactive = mActiveTracks.size(); 6589 size_t numactiveseen = 0; 6590 dprintf(fd, " %zu Tracks", numtracks); 6591 if (numtracks) { 6592 dprintf(fd, " of which %zu are active\n", numactive); 6593 RecordTrack::appendDumpHeader(result); 6594 for (size_t i = 0; i < numtracks ; ++i) { 6595 sp<RecordTrack> track = mTracks[i]; 6596 if (track != 0) { 6597 bool active = mActiveTracks.indexOf(track) >= 0; 6598 if (active) { 6599 numactiveseen++; 6600 } 6601 track->dump(buffer, SIZE, active); 6602 result.append(buffer); 6603 } 6604 } 6605 } else { 6606 dprintf(fd, "\n"); 6607 } 6608 6609 if (numactiveseen != numactive) { 6610 snprintf(buffer, SIZE, " The following tracks are in the active list but" 6611 " not in the track list\n"); 6612 result.append(buffer); 6613 RecordTrack::appendDumpHeader(result); 6614 for (size_t i = 0; i < numactive; ++i) { 6615 sp<RecordTrack> track = mActiveTracks[i]; 6616 if (mTracks.indexOf(track) < 0) { 6617 track->dump(buffer, SIZE, true); 6618 result.append(buffer); 6619 } 6620 } 6621 6622 } 6623 write(fd, result.string(), result.size()); 6624} 6625 6626 6627void AudioFlinger::RecordThread::ResamplerBufferProvider::reset() 6628{ 6629 sp<ThreadBase> threadBase = mRecordTrack->mThread.promote(); 6630 RecordThread *recordThread = (RecordThread *) threadBase.get(); 6631 mRsmpInFront = recordThread->mRsmpInRear; 6632 mRsmpInUnrel = 0; 6633} 6634 6635void AudioFlinger::RecordThread::ResamplerBufferProvider::sync( 6636 size_t *framesAvailable, bool *hasOverrun) 6637{ 6638 sp<ThreadBase> threadBase = mRecordTrack->mThread.promote(); 6639 RecordThread *recordThread = (RecordThread *) threadBase.get(); 6640 const int32_t rear = recordThread->mRsmpInRear; 6641 const int32_t front = mRsmpInFront; 6642 const ssize_t filled = rear - front; 6643 6644 size_t framesIn; 6645 bool overrun = false; 6646 if (filled < 0) { 6647 // should not happen, but treat like a massive overrun and re-sync 6648 framesIn = 0; 6649 mRsmpInFront = rear; 6650 overrun = true; 6651 } else if ((size_t) filled <= recordThread->mRsmpInFrames) { 6652 framesIn = (size_t) filled; 6653 } else { 6654 // client is not keeping up with server, but give it latest data 6655 framesIn = recordThread->mRsmpInFrames; 6656 mRsmpInFront = /* front = */ rear - framesIn; 6657 overrun = true; 6658 } 6659 if (framesAvailable != NULL) { 6660 *framesAvailable = framesIn; 6661 } 6662 if (hasOverrun != NULL) { 6663 *hasOverrun = overrun; 6664 } 6665} 6666 6667// AudioBufferProvider interface 6668status_t AudioFlinger::RecordThread::ResamplerBufferProvider::getNextBuffer( 6669 AudioBufferProvider::Buffer* buffer) 6670{ 6671 sp<ThreadBase> threadBase = mRecordTrack->mThread.promote(); 6672 if (threadBase == 0) { 6673 buffer->frameCount = 0; 6674 buffer->raw = NULL; 6675 return NOT_ENOUGH_DATA; 6676 } 6677 RecordThread *recordThread = (RecordThread *) threadBase.get(); 6678 int32_t rear = recordThread->mRsmpInRear; 6679 int32_t front = mRsmpInFront; 6680 ssize_t filled = rear - front; 6681 // FIXME should not be P2 (don't want to increase latency) 6682 // FIXME if client not keeping up, discard 6683 LOG_ALWAYS_FATAL_IF(!(0 <= filled && (size_t) filled <= recordThread->mRsmpInFrames)); 6684 // 'filled' may be non-contiguous, so return only the first contiguous chunk 6685 front &= recordThread->mRsmpInFramesP2 - 1; 6686 size_t part1 = recordThread->mRsmpInFramesP2 - front; 6687 if (part1 > (size_t) filled) { 6688 part1 = filled; 6689 } 6690 size_t ask = buffer->frameCount; 6691 ALOG_ASSERT(ask > 0); 6692 if (part1 > ask) { 6693 part1 = ask; 6694 } 6695 if (part1 == 0) { 6696 // out of data is fine since the resampler will return a short-count. 6697 buffer->raw = NULL; 6698 buffer->frameCount = 0; 6699 mRsmpInUnrel = 0; 6700 return NOT_ENOUGH_DATA; 6701 } 6702 6703 buffer->raw = (uint8_t*)recordThread->mRsmpInBuffer + front * recordThread->mFrameSize; 6704 buffer->frameCount = part1; 6705 mRsmpInUnrel = part1; 6706 return NO_ERROR; 6707} 6708 6709// AudioBufferProvider interface 6710void AudioFlinger::RecordThread::ResamplerBufferProvider::releaseBuffer( 6711 AudioBufferProvider::Buffer* buffer) 6712{ 6713 size_t stepCount = buffer->frameCount; 6714 if (stepCount == 0) { 6715 return; 6716 } 6717 ALOG_ASSERT(stepCount <= mRsmpInUnrel); 6718 mRsmpInUnrel -= stepCount; 6719 mRsmpInFront += stepCount; 6720 buffer->raw = NULL; 6721 buffer->frameCount = 0; 6722} 6723 6724AudioFlinger::RecordThread::RecordBufferConverter::RecordBufferConverter( 6725 audio_channel_mask_t srcChannelMask, audio_format_t srcFormat, 6726 uint32_t srcSampleRate, 6727 audio_channel_mask_t dstChannelMask, audio_format_t dstFormat, 6728 uint32_t dstSampleRate) : 6729 mSrcChannelMask(AUDIO_CHANNEL_INVALID), // updateParameters will set following vars 6730 // mSrcFormat 6731 // mSrcSampleRate 6732 // mDstChannelMask 6733 // mDstFormat 6734 // mDstSampleRate 6735 // mSrcChannelCount 6736 // mDstChannelCount 6737 // mDstFrameSize 6738 mBuf(NULL), mBufFrames(0), mBufFrameSize(0), 6739 mResampler(NULL), 6740 mIsLegacyDownmix(false), 6741 mIsLegacyUpmix(false), 6742 mRequiresFloat(false), 6743 mInputConverterProvider(NULL) 6744{ 6745 (void)updateParameters(srcChannelMask, srcFormat, srcSampleRate, 6746 dstChannelMask, dstFormat, dstSampleRate); 6747} 6748 6749AudioFlinger::RecordThread::RecordBufferConverter::~RecordBufferConverter() { 6750 free(mBuf); 6751 delete mResampler; 6752 delete mInputConverterProvider; 6753} 6754 6755size_t AudioFlinger::RecordThread::RecordBufferConverter::convert(void *dst, 6756 AudioBufferProvider *provider, size_t frames) 6757{ 6758 if (mInputConverterProvider != NULL) { 6759 mInputConverterProvider->setBufferProvider(provider); 6760 provider = mInputConverterProvider; 6761 } 6762 6763 if (mResampler == NULL) { 6764 ALOGVV("NO RESAMPLING sampleRate:%u mSrcFormat:%#x mDstFormat:%#x", 6765 mSrcSampleRate, mSrcFormat, mDstFormat); 6766 6767 AudioBufferProvider::Buffer buffer; 6768 for (size_t i = frames; i > 0; ) { 6769 buffer.frameCount = i; 6770 status_t status = provider->getNextBuffer(&buffer); 6771 if (status != OK || buffer.frameCount == 0) { 6772 frames -= i; // cannot fill request. 6773 break; 6774 } 6775 // format convert to destination buffer 6776 convertNoResampler(dst, buffer.raw, buffer.frameCount); 6777 6778 dst = (int8_t*)dst + buffer.frameCount * mDstFrameSize; 6779 i -= buffer.frameCount; 6780 provider->releaseBuffer(&buffer); 6781 } 6782 } else { 6783 ALOGVV("RESAMPLING mSrcSampleRate:%u mDstSampleRate:%u mSrcFormat:%#x mDstFormat:%#x", 6784 mSrcSampleRate, mDstSampleRate, mSrcFormat, mDstFormat); 6785 6786 // reallocate buffer if needed 6787 if (mBufFrameSize != 0 && mBufFrames < frames) { 6788 free(mBuf); 6789 mBufFrames = frames; 6790 (void)posix_memalign(&mBuf, 32, mBufFrames * mBufFrameSize); 6791 } 6792 // resampler accumulates, but we only have one source track 6793 memset(mBuf, 0, frames * mBufFrameSize); 6794 frames = mResampler->resample((int32_t*)mBuf, frames, provider); 6795 // format convert to destination buffer 6796 convertResampler(dst, mBuf, frames); 6797 } 6798 return frames; 6799} 6800 6801status_t AudioFlinger::RecordThread::RecordBufferConverter::updateParameters( 6802 audio_channel_mask_t srcChannelMask, audio_format_t srcFormat, 6803 uint32_t srcSampleRate, 6804 audio_channel_mask_t dstChannelMask, audio_format_t dstFormat, 6805 uint32_t dstSampleRate) 6806{ 6807 // quick evaluation if there is any change. 6808 if (mSrcFormat == srcFormat 6809 && mSrcChannelMask == srcChannelMask 6810 && mSrcSampleRate == srcSampleRate 6811 && mDstFormat == dstFormat 6812 && mDstChannelMask == dstChannelMask 6813 && mDstSampleRate == dstSampleRate) { 6814 return NO_ERROR; 6815 } 6816 6817 ALOGV("RecordBufferConverter updateParameters srcMask:%#x dstMask:%#x" 6818 " srcFormat:%#x dstFormat:%#x srcRate:%u dstRate:%u", 6819 srcChannelMask, dstChannelMask, srcFormat, dstFormat, srcSampleRate, dstSampleRate); 6820 const bool valid = 6821 audio_is_input_channel(srcChannelMask) 6822 && audio_is_input_channel(dstChannelMask) 6823 && audio_is_valid_format(srcFormat) && audio_is_linear_pcm(srcFormat) 6824 && audio_is_valid_format(dstFormat) && audio_is_linear_pcm(dstFormat) 6825 && (srcSampleRate <= dstSampleRate * AUDIO_RESAMPLER_DOWN_RATIO_MAX) 6826 ; // no upsampling checks for now 6827 if (!valid) { 6828 return BAD_VALUE; 6829 } 6830 6831 mSrcFormat = srcFormat; 6832 mSrcChannelMask = srcChannelMask; 6833 mSrcSampleRate = srcSampleRate; 6834 mDstFormat = dstFormat; 6835 mDstChannelMask = dstChannelMask; 6836 mDstSampleRate = dstSampleRate; 6837 6838 // compute derived parameters 6839 mSrcChannelCount = audio_channel_count_from_in_mask(srcChannelMask); 6840 mDstChannelCount = audio_channel_count_from_in_mask(dstChannelMask); 6841 mDstFrameSize = mDstChannelCount * audio_bytes_per_sample(mDstFormat); 6842 6843 // do we need to resample? 6844 delete mResampler; 6845 mResampler = NULL; 6846 if (mSrcSampleRate != mDstSampleRate) { 6847 mResampler = AudioResampler::create(AUDIO_FORMAT_PCM_FLOAT, 6848 mSrcChannelCount, mDstSampleRate); 6849 mResampler->setSampleRate(mSrcSampleRate); 6850 mResampler->setVolume(AudioMixer::UNITY_GAIN_FLOAT, AudioMixer::UNITY_GAIN_FLOAT); 6851 } 6852 6853 // are we running legacy channel conversion modes? 6854 mIsLegacyDownmix = (mSrcChannelMask == AUDIO_CHANNEL_IN_STEREO 6855 || mSrcChannelMask == AUDIO_CHANNEL_IN_FRONT_BACK) 6856 && mDstChannelMask == AUDIO_CHANNEL_IN_MONO; 6857 mIsLegacyUpmix = mSrcChannelMask == AUDIO_CHANNEL_IN_MONO 6858 && (mDstChannelMask == AUDIO_CHANNEL_IN_STEREO 6859 || mDstChannelMask == AUDIO_CHANNEL_IN_FRONT_BACK); 6860 6861 // do we need to process in float? 6862 mRequiresFloat = mResampler != NULL || mIsLegacyDownmix || mIsLegacyUpmix; 6863 6864 // do we need a staging buffer to convert for destination (we can still optimize this)? 6865 // we use mBufFrameSize > 0 to indicate both frame size as well as buffer necessity 6866 if (mResampler != NULL) { 6867 mBufFrameSize = max(mSrcChannelCount, FCC_2) 6868 * audio_bytes_per_sample(AUDIO_FORMAT_PCM_FLOAT); 6869 } else if (mIsLegacyUpmix || mIsLegacyDownmix) { // legacy modes always float 6870 mBufFrameSize = mDstChannelCount * audio_bytes_per_sample(AUDIO_FORMAT_PCM_FLOAT); 6871 } else if (mSrcChannelMask != mDstChannelMask && mDstFormat != mSrcFormat) { 6872 mBufFrameSize = mDstChannelCount * audio_bytes_per_sample(mSrcFormat); 6873 } else { 6874 mBufFrameSize = 0; 6875 } 6876 mBufFrames = 0; // force the buffer to be resized. 6877 6878 // do we need an input converter buffer provider to give us float? 6879 delete mInputConverterProvider; 6880 mInputConverterProvider = NULL; 6881 if (mRequiresFloat && mSrcFormat != AUDIO_FORMAT_PCM_FLOAT) { 6882 mInputConverterProvider = new ReformatBufferProvider( 6883 audio_channel_count_from_in_mask(mSrcChannelMask), 6884 mSrcFormat, 6885 AUDIO_FORMAT_PCM_FLOAT, 6886 256 /* provider buffer frame count */); 6887 } 6888 6889 // do we need a remixer to do channel mask conversion 6890 if (!mIsLegacyDownmix && !mIsLegacyUpmix && mSrcChannelMask != mDstChannelMask) { 6891 (void) memcpy_by_index_array_initialization_from_channel_mask( 6892 mIdxAry, ARRAY_SIZE(mIdxAry), mDstChannelMask, mSrcChannelMask); 6893 } 6894 return NO_ERROR; 6895} 6896 6897void AudioFlinger::RecordThread::RecordBufferConverter::convertNoResampler( 6898 void *dst, const void *src, size_t frames) 6899{ 6900 // src is native type unless there is legacy upmix or downmix, whereupon it is float. 6901 if (mBufFrameSize != 0 && mBufFrames < frames) { 6902 free(mBuf); 6903 mBufFrames = frames; 6904 (void)posix_memalign(&mBuf, 32, mBufFrames * mBufFrameSize); 6905 } 6906 // do we need to do legacy upmix and downmix? 6907 if (mIsLegacyUpmix || mIsLegacyDownmix) { 6908 void *dstBuf = mBuf != NULL ? mBuf : dst; 6909 if (mIsLegacyUpmix) { 6910 upmix_to_stereo_float_from_mono_float((float *)dstBuf, 6911 (const float *)src, frames); 6912 } else /*mIsLegacyDownmix */ { 6913 downmix_to_mono_float_from_stereo_float((float *)dstBuf, 6914 (const float *)src, frames); 6915 } 6916 if (mBuf != NULL) { 6917 memcpy_by_audio_format(dst, mDstFormat, mBuf, AUDIO_FORMAT_PCM_FLOAT, 6918 frames * mDstChannelCount); 6919 } 6920 return; 6921 } 6922 // do we need to do channel mask conversion? 6923 if (mSrcChannelMask != mDstChannelMask) { 6924 void *dstBuf = mBuf != NULL ? mBuf : dst; 6925 memcpy_by_index_array(dstBuf, mDstChannelCount, 6926 src, mSrcChannelCount, mIdxAry, audio_bytes_per_sample(mSrcFormat), frames); 6927 if (dstBuf == dst) { 6928 return; // format is the same 6929 } 6930 } 6931 // convert to destination buffer 6932 const void *convertBuf = mBuf != NULL ? mBuf : src; 6933 memcpy_by_audio_format(dst, mDstFormat, convertBuf, mSrcFormat, 6934 frames * mDstChannelCount); 6935} 6936 6937void AudioFlinger::RecordThread::RecordBufferConverter::convertResampler( 6938 void *dst, /*not-a-const*/ void *src, size_t frames) 6939{ 6940 // src buffer format is ALWAYS float when entering this routine 6941 if (mIsLegacyUpmix) { 6942 ; // mono to stereo already handled by resampler 6943 } else if (mIsLegacyDownmix 6944 || (mSrcChannelMask == mDstChannelMask && mSrcChannelCount == 1)) { 6945 // the resampler outputs stereo for mono input channel (a feature?) 6946 // must convert to mono 6947 downmix_to_mono_float_from_stereo_float((float *)src, 6948 (const float *)src, frames); 6949 } else if (mSrcChannelMask != mDstChannelMask) { 6950 // convert to mono channel again for channel mask conversion (could be skipped 6951 // with further optimization). 6952 if (mSrcChannelCount == 1) { 6953 downmix_to_mono_float_from_stereo_float((float *)src, 6954 (const float *)src, frames); 6955 } 6956 // convert to destination format (in place, OK as float is larger than other types) 6957 if (mDstFormat != AUDIO_FORMAT_PCM_FLOAT) { 6958 memcpy_by_audio_format(src, mDstFormat, src, AUDIO_FORMAT_PCM_FLOAT, 6959 frames * mSrcChannelCount); 6960 } 6961 // channel convert and save to dst 6962 memcpy_by_index_array(dst, mDstChannelCount, 6963 src, mSrcChannelCount, mIdxAry, audio_bytes_per_sample(mDstFormat), frames); 6964 return; 6965 } 6966 // convert to destination format and save to dst 6967 memcpy_by_audio_format(dst, mDstFormat, src, AUDIO_FORMAT_PCM_FLOAT, 6968 frames * mDstChannelCount); 6969} 6970 6971bool AudioFlinger::RecordThread::checkForNewParameter_l(const String8& keyValuePair, 6972 status_t& status) 6973{ 6974 bool reconfig = false; 6975 6976 status = NO_ERROR; 6977 6978 audio_format_t reqFormat = mFormat; 6979 uint32_t samplingRate = mSampleRate; 6980 // TODO this may change if we want to support capture from HDMI PCM multi channel (e.g on TVs). 6981 audio_channel_mask_t channelMask = audio_channel_in_mask_from_count(mChannelCount); 6982 6983 AudioParameter param = AudioParameter(keyValuePair); 6984 int value; 6985 6986 // scope for AutoPark extends to end of method 6987 AutoPark<FastCapture> park(mFastCapture); 6988 6989 // TODO Investigate when this code runs. Check with audio policy when a sample rate and 6990 // channel count change can be requested. Do we mandate the first client defines the 6991 // HAL sampling rate and channel count or do we allow changes on the fly? 6992 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 6993 samplingRate = value; 6994 reconfig = true; 6995 } 6996 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 6997 if (!audio_is_linear_pcm((audio_format_t) value)) { 6998 status = BAD_VALUE; 6999 } else { 7000 reqFormat = (audio_format_t) value; 7001 reconfig = true; 7002 } 7003 } 7004 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 7005 audio_channel_mask_t mask = (audio_channel_mask_t) value; 7006 if (!audio_is_input_channel(mask) || 7007 audio_channel_count_from_in_mask(mask) > FCC_8) { 7008 status = BAD_VALUE; 7009 } else { 7010 channelMask = mask; 7011 reconfig = true; 7012 } 7013 } 7014 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 7015 // do not accept frame count changes if tracks are open as the track buffer 7016 // size depends on frame count and correct behavior would not be guaranteed 7017 // if frame count is changed after track creation 7018 if (mActiveTracks.size() > 0) { 7019 status = INVALID_OPERATION; 7020 } else { 7021 reconfig = true; 7022 } 7023 } 7024 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 7025 // forward device change to effects that have requested to be 7026 // aware of attached audio device. 7027 for (size_t i = 0; i < mEffectChains.size(); i++) { 7028 mEffectChains[i]->setDevice_l(value); 7029 } 7030 7031 // store input device and output device but do not forward output device to audio HAL. 7032 // Note that status is ignored by the caller for output device 7033 // (see AudioFlinger::setParameters() 7034 if (audio_is_output_devices(value)) { 7035 mOutDevice = value; 7036 status = BAD_VALUE; 7037 } else { 7038 mInDevice = value; 7039 if (value != AUDIO_DEVICE_NONE) { 7040 mPrevInDevice = value; 7041 } 7042 // disable AEC and NS if the device is a BT SCO headset supporting those 7043 // pre processings 7044 if (mTracks.size() > 0) { 7045 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 7046 mAudioFlinger->btNrecIsOff(); 7047 for (size_t i = 0; i < mTracks.size(); i++) { 7048 sp<RecordTrack> track = mTracks[i]; 7049 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId()); 7050 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId()); 7051 } 7052 } 7053 } 7054 } 7055 if (param.getInt(String8(AudioParameter::keyInputSource), value) == NO_ERROR && 7056 mAudioSource != (audio_source_t)value) { 7057 // forward device change to effects that have requested to be 7058 // aware of attached audio device. 7059 for (size_t i = 0; i < mEffectChains.size(); i++) { 7060 mEffectChains[i]->setAudioSource_l((audio_source_t)value); 7061 } 7062 mAudioSource = (audio_source_t)value; 7063 } 7064 7065 if (status == NO_ERROR) { 7066 status = mInput->stream->common.set_parameters(&mInput->stream->common, 7067 keyValuePair.string()); 7068 if (status == INVALID_OPERATION) { 7069 inputStandBy(); 7070 status = mInput->stream->common.set_parameters(&mInput->stream->common, 7071 keyValuePair.string()); 7072 } 7073 if (reconfig) { 7074 if (status == BAD_VALUE && 7075 audio_is_linear_pcm(mInput->stream->common.get_format(&mInput->stream->common)) && 7076 audio_is_linear_pcm(reqFormat) && 7077 (mInput->stream->common.get_sample_rate(&mInput->stream->common) 7078 <= (AUDIO_RESAMPLER_DOWN_RATIO_MAX * samplingRate)) && 7079 audio_channel_count_from_in_mask( 7080 mInput->stream->common.get_channels(&mInput->stream->common)) <= FCC_8) { 7081 status = NO_ERROR; 7082 } 7083 if (status == NO_ERROR) { 7084 readInputParameters_l(); 7085 sendIoConfigEvent_l(AUDIO_INPUT_CONFIG_CHANGED); 7086 } 7087 } 7088 } 7089 7090 return reconfig; 7091} 7092 7093String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 7094{ 7095 Mutex::Autolock _l(mLock); 7096 if (initCheck() != NO_ERROR) { 7097 return String8(); 7098 } 7099 7100 char *s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 7101 const String8 out_s8(s); 7102 free(s); 7103 return out_s8; 7104} 7105 7106void AudioFlinger::RecordThread::ioConfigChanged(audio_io_config_event event, pid_t pid) { 7107 sp<AudioIoDescriptor> desc = new AudioIoDescriptor(); 7108 7109 desc->mIoHandle = mId; 7110 7111 switch (event) { 7112 case AUDIO_INPUT_OPENED: 7113 case AUDIO_INPUT_CONFIG_CHANGED: 7114 desc->mPatch = mPatch; 7115 desc->mChannelMask = mChannelMask; 7116 desc->mSamplingRate = mSampleRate; 7117 desc->mFormat = mFormat; 7118 desc->mFrameCount = mFrameCount; 7119 desc->mLatency = 0; 7120 break; 7121 7122 case AUDIO_INPUT_CLOSED: 7123 default: 7124 break; 7125 } 7126 mAudioFlinger->ioConfigChanged(event, desc, pid); 7127} 7128 7129void AudioFlinger::RecordThread::readInputParameters_l() 7130{ 7131 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 7132 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 7133 mChannelCount = audio_channel_count_from_in_mask(mChannelMask); 7134 if (mChannelCount > FCC_8) { 7135 ALOGE("HAL channel count %d > %d", mChannelCount, FCC_8); 7136 } 7137 mHALFormat = mInput->stream->common.get_format(&mInput->stream->common); 7138 mFormat = mHALFormat; 7139 if (!audio_is_linear_pcm(mFormat)) { 7140 ALOGE("HAL format %#x is not linear pcm", mFormat); 7141 } 7142 mFrameSize = audio_stream_in_frame_size(mInput->stream); 7143 mBufferSize = mInput->stream->common.get_buffer_size(&mInput->stream->common); 7144 mFrameCount = mBufferSize / mFrameSize; 7145 // This is the formula for calculating the temporary buffer size. 7146 // With 7 HAL buffers, we can guarantee ability to down-sample the input by ratio of 6:1 to 7147 // 1 full output buffer, regardless of the alignment of the available input. 7148 // The value is somewhat arbitrary, and could probably be even larger. 7149 // A larger value should allow more old data to be read after a track calls start(), 7150 // without increasing latency. 7151 // 7152 // Note this is independent of the maximum downsampling ratio permitted for capture. 7153 mRsmpInFrames = mFrameCount * 7; 7154 mRsmpInFramesP2 = roundup(mRsmpInFrames); 7155 free(mRsmpInBuffer); 7156 mRsmpInBuffer = NULL; 7157 7158 // TODO optimize audio capture buffer sizes ... 7159 // Here we calculate the size of the sliding buffer used as a source 7160 // for resampling. mRsmpInFramesP2 is currently roundup(mFrameCount * 7). 7161 // For current HAL frame counts, this is usually 2048 = 40 ms. It would 7162 // be better to have it derived from the pipe depth in the long term. 7163 // The current value is higher than necessary. However it should not add to latency. 7164 7165 // Over-allocate beyond mRsmpInFramesP2 to permit a HAL read past end of buffer 7166 size_t bufferSize = (mRsmpInFramesP2 + mFrameCount - 1) * mFrameSize; 7167 (void)posix_memalign(&mRsmpInBuffer, 32, bufferSize); 7168 memset(mRsmpInBuffer, 0, bufferSize); // if posix_memalign fails, will segv here. 7169 7170 // AudioRecord mSampleRate and mChannelCount are constant due to AudioRecord API constraints. 7171 // But if thread's mSampleRate or mChannelCount changes, how will that affect active tracks? 7172} 7173 7174uint32_t AudioFlinger::RecordThread::getInputFramesLost() 7175{ 7176 Mutex::Autolock _l(mLock); 7177 if (initCheck() != NO_ERROR) { 7178 return 0; 7179 } 7180 7181 return mInput->stream->get_input_frames_lost(mInput->stream); 7182} 7183 7184uint32_t AudioFlinger::RecordThread::hasAudioSession(audio_session_t sessionId) const 7185{ 7186 Mutex::Autolock _l(mLock); 7187 uint32_t result = 0; 7188 if (getEffectChain_l(sessionId) != 0) { 7189 result = EFFECT_SESSION; 7190 } 7191 7192 for (size_t i = 0; i < mTracks.size(); ++i) { 7193 if (sessionId == mTracks[i]->sessionId()) { 7194 result |= TRACK_SESSION; 7195 break; 7196 } 7197 } 7198 7199 return result; 7200} 7201 7202KeyedVector<audio_session_t, bool> AudioFlinger::RecordThread::sessionIds() const 7203{ 7204 KeyedVector<audio_session_t, bool> ids; 7205 Mutex::Autolock _l(mLock); 7206 for (size_t j = 0; j < mTracks.size(); ++j) { 7207 sp<RecordThread::RecordTrack> track = mTracks[j]; 7208 audio_session_t sessionId = track->sessionId(); 7209 if (ids.indexOfKey(sessionId) < 0) { 7210 ids.add(sessionId, true); 7211 } 7212 } 7213 return ids; 7214} 7215 7216AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 7217{ 7218 Mutex::Autolock _l(mLock); 7219 AudioStreamIn *input = mInput; 7220 mInput = NULL; 7221 return input; 7222} 7223 7224// this method must always be called either with ThreadBase mLock held or inside the thread loop 7225audio_stream_t* AudioFlinger::RecordThread::stream() const 7226{ 7227 if (mInput == NULL) { 7228 return NULL; 7229 } 7230 return &mInput->stream->common; 7231} 7232 7233status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 7234{ 7235 // only one chain per input thread 7236 if (mEffectChains.size() != 0) { 7237 ALOGW("addEffectChain_l() already one chain %p on thread %p", chain.get(), this); 7238 return INVALID_OPERATION; 7239 } 7240 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 7241 chain->setThread(this); 7242 chain->setInBuffer(NULL); 7243 chain->setOutBuffer(NULL); 7244 7245 checkSuspendOnAddEffectChain_l(chain); 7246 7247 // make sure enabled pre processing effects state is communicated to the HAL as we 7248 // just moved them to a new input stream. 7249 chain->syncHalEffectsState(); 7250 7251 mEffectChains.add(chain); 7252 7253 return NO_ERROR; 7254} 7255 7256size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 7257{ 7258 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 7259 ALOGW_IF(mEffectChains.size() != 1, 7260 "removeEffectChain_l() %p invalid chain size %zu on thread %p", 7261 chain.get(), mEffectChains.size(), this); 7262 if (mEffectChains.size() == 1) { 7263 mEffectChains.removeAt(0); 7264 } 7265 return 0; 7266} 7267 7268status_t AudioFlinger::RecordThread::createAudioPatch_l(const struct audio_patch *patch, 7269 audio_patch_handle_t *handle) 7270{ 7271 status_t status = NO_ERROR; 7272 7273 // store new device and send to effects 7274 mInDevice = patch->sources[0].ext.device.type; 7275 mPatch = *patch; 7276 for (size_t i = 0; i < mEffectChains.size(); i++) { 7277 mEffectChains[i]->setDevice_l(mInDevice); 7278 } 7279 7280 // disable AEC and NS if the device is a BT SCO headset supporting those 7281 // pre processings 7282 if (mTracks.size() > 0) { 7283 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 7284 mAudioFlinger->btNrecIsOff(); 7285 for (size_t i = 0; i < mTracks.size(); i++) { 7286 sp<RecordTrack> track = mTracks[i]; 7287 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId()); 7288 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId()); 7289 } 7290 } 7291 7292 // store new source and send to effects 7293 if (mAudioSource != patch->sinks[0].ext.mix.usecase.source) { 7294 mAudioSource = patch->sinks[0].ext.mix.usecase.source; 7295 for (size_t i = 0; i < mEffectChains.size(); i++) { 7296 mEffectChains[i]->setAudioSource_l(mAudioSource); 7297 } 7298 } 7299 7300 if (mInput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 7301 audio_hw_device_t *hwDevice = mInput->audioHwDev->hwDevice(); 7302 status = hwDevice->create_audio_patch(hwDevice, 7303 patch->num_sources, 7304 patch->sources, 7305 patch->num_sinks, 7306 patch->sinks, 7307 handle); 7308 } else { 7309 char *address; 7310 if (strcmp(patch->sources[0].ext.device.address, "") != 0) { 7311 address = audio_device_address_to_parameter( 7312 patch->sources[0].ext.device.type, 7313 patch->sources[0].ext.device.address); 7314 } else { 7315 address = (char *)calloc(1, 1); 7316 } 7317 AudioParameter param = AudioParameter(String8(address)); 7318 free(address); 7319 param.addInt(String8(AUDIO_PARAMETER_STREAM_ROUTING), 7320 (int)patch->sources[0].ext.device.type); 7321 param.addInt(String8(AUDIO_PARAMETER_STREAM_INPUT_SOURCE), 7322 (int)patch->sinks[0].ext.mix.usecase.source); 7323 status = mInput->stream->common.set_parameters(&mInput->stream->common, 7324 param.toString().string()); 7325 *handle = AUDIO_PATCH_HANDLE_NONE; 7326 } 7327 7328 if (mInDevice != mPrevInDevice) { 7329 sendIoConfigEvent_l(AUDIO_INPUT_CONFIG_CHANGED); 7330 mPrevInDevice = mInDevice; 7331 } 7332 7333 return status; 7334} 7335 7336status_t AudioFlinger::RecordThread::releaseAudioPatch_l(const audio_patch_handle_t handle) 7337{ 7338 status_t status = NO_ERROR; 7339 7340 mInDevice = AUDIO_DEVICE_NONE; 7341 7342 if (mInput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 7343 audio_hw_device_t *hwDevice = mInput->audioHwDev->hwDevice(); 7344 status = hwDevice->release_audio_patch(hwDevice, handle); 7345 } else { 7346 AudioParameter param; 7347 param.addInt(String8(AUDIO_PARAMETER_STREAM_ROUTING), 0); 7348 status = mInput->stream->common.set_parameters(&mInput->stream->common, 7349 param.toString().string()); 7350 } 7351 return status; 7352} 7353 7354void AudioFlinger::RecordThread::addPatchRecord(const sp<PatchRecord>& record) 7355{ 7356 Mutex::Autolock _l(mLock); 7357 mTracks.add(record); 7358} 7359 7360void AudioFlinger::RecordThread::deletePatchRecord(const sp<PatchRecord>& record) 7361{ 7362 Mutex::Autolock _l(mLock); 7363 destroyTrack_l(record); 7364} 7365 7366void AudioFlinger::RecordThread::getAudioPortConfig(struct audio_port_config *config) 7367{ 7368 ThreadBase::getAudioPortConfig(config); 7369 config->role = AUDIO_PORT_ROLE_SINK; 7370 config->ext.mix.hw_module = mInput->audioHwDev->handle(); 7371 config->ext.mix.usecase.source = mAudioSource; 7372} 7373 7374} // namespace android 7375