Threads.cpp revision 6d7b119a416c9f10288051e562f294365e5d954c
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() has an in/out parameter 'pFrameCount' for the total size of the 186// track buffer in shared memory. Zero on input means to use a default value. For fast tracks, 187// AudioFlinger derives the default from HAL buffer size and 'fast track multiplier'. 188 189// This is the default value, if not specified by property. 190static const int kFastTrackMultiplier = 2; 191 192// The minimum and maximum allowed values 193static const int kFastTrackMultiplierMin = 1; 194static const int kFastTrackMultiplierMax = 2; 195 196// The actual value to use, which can be specified per-device via property af.fast_track_multiplier. 197static int sFastTrackMultiplier = kFastTrackMultiplier; 198 199// See Thread::readOnlyHeap(). 200// Initially this heap is used to allocate client buffers for "fast" AudioRecord. 201// Eventually it will be the single buffer that FastCapture writes into via HAL read(), 202// and that all "fast" AudioRecord clients read from. In either case, the size can be small. 203static const size_t kRecordThreadReadOnlyHeapSize = 0x2000; 204 205// ---------------------------------------------------------------------------- 206 207static pthread_once_t sFastTrackMultiplierOnce = PTHREAD_ONCE_INIT; 208 209static void sFastTrackMultiplierInit() 210{ 211 char value[PROPERTY_VALUE_MAX]; 212 if (property_get("af.fast_track_multiplier", value, NULL) > 0) { 213 char *endptr; 214 unsigned long ul = strtoul(value, &endptr, 0); 215 if (*endptr == '\0' && kFastTrackMultiplierMin <= ul && ul <= kFastTrackMultiplierMax) { 216 sFastTrackMultiplier = (int) ul; 217 } 218 } 219} 220 221// ---------------------------------------------------------------------------- 222 223#ifdef ADD_BATTERY_DATA 224// To collect the amplifier usage 225static void addBatteryData(uint32_t params) { 226 sp<IMediaPlayerService> service = IMediaDeathNotifier::getMediaPlayerService(); 227 if (service == NULL) { 228 // it already logged 229 return; 230 } 231 232 service->addBatteryData(params); 233} 234#endif 235 236// Track the CLOCK_BOOTTIME versus CLOCK_MONOTONIC timebase offset 237struct { 238 // call when you acquire a partial wakelock 239 void acquire(const sp<IBinder> &wakeLockToken) { 240 pthread_mutex_lock(&mLock); 241 if (wakeLockToken.get() == nullptr) { 242 adjustTimebaseOffset(&mBoottimeOffset, ExtendedTimestamp::TIMEBASE_BOOTTIME); 243 } else { 244 if (mCount == 0) { 245 adjustTimebaseOffset(&mBoottimeOffset, ExtendedTimestamp::TIMEBASE_BOOTTIME); 246 } 247 ++mCount; 248 } 249 pthread_mutex_unlock(&mLock); 250 } 251 252 // call when you release a partial wakelock. 253 void release(const sp<IBinder> &wakeLockToken) { 254 if (wakeLockToken.get() == nullptr) { 255 return; 256 } 257 pthread_mutex_lock(&mLock); 258 if (--mCount < 0) { 259 ALOGE("negative wakelock count"); 260 mCount = 0; 261 } 262 pthread_mutex_unlock(&mLock); 263 } 264 265 // retrieves the boottime timebase offset from monotonic. 266 int64_t getBoottimeOffset() { 267 pthread_mutex_lock(&mLock); 268 int64_t boottimeOffset = mBoottimeOffset; 269 pthread_mutex_unlock(&mLock); 270 return boottimeOffset; 271 } 272 273 // Adjusts the timebase offset between TIMEBASE_MONOTONIC 274 // and the selected timebase. 275 // Currently only TIMEBASE_BOOTTIME is allowed. 276 // 277 // This only needs to be called upon acquiring the first partial wakelock 278 // after all other partial wakelocks are released. 279 // 280 // We do an empirical measurement of the offset rather than parsing 281 // /proc/timer_list since the latter is not a formal kernel ABI. 282 static void adjustTimebaseOffset(int64_t *offset, ExtendedTimestamp::Timebase timebase) { 283 int clockbase; 284 switch (timebase) { 285 case ExtendedTimestamp::TIMEBASE_BOOTTIME: 286 clockbase = SYSTEM_TIME_BOOTTIME; 287 break; 288 default: 289 LOG_ALWAYS_FATAL("invalid timebase %d", timebase); 290 break; 291 } 292 // try three times to get the clock offset, choose the one 293 // with the minimum gap in measurements. 294 const int tries = 3; 295 nsecs_t bestGap, measured; 296 for (int i = 0; i < tries; ++i) { 297 const nsecs_t tmono = systemTime(SYSTEM_TIME_MONOTONIC); 298 const nsecs_t tbase = systemTime(clockbase); 299 const nsecs_t tmono2 = systemTime(SYSTEM_TIME_MONOTONIC); 300 const nsecs_t gap = tmono2 - tmono; 301 if (i == 0 || gap < bestGap) { 302 bestGap = gap; 303 measured = tbase - ((tmono + tmono2) >> 1); 304 } 305 } 306 307 // to avoid micro-adjusting, we don't change the timebase 308 // unless it is significantly different. 309 // 310 // Assumption: It probably takes more than toleranceNs to 311 // suspend and resume the device. 312 static int64_t toleranceNs = 10000; // 10 us 313 if (llabs(*offset - measured) > toleranceNs) { 314 ALOGV("Adjusting timebase offset old: %lld new: %lld", 315 (long long)*offset, (long long)measured); 316 *offset = measured; 317 } 318 } 319 320 pthread_mutex_t mLock; 321 int32_t mCount; 322 int64_t mBoottimeOffset; 323} gBoottime = { PTHREAD_MUTEX_INITIALIZER, 0, 0 }; // static, so use POD initialization 324 325// ---------------------------------------------------------------------------- 326// CPU Stats 327// ---------------------------------------------------------------------------- 328 329class CpuStats { 330public: 331 CpuStats(); 332 void sample(const String8 &title); 333#ifdef DEBUG_CPU_USAGE 334private: 335 ThreadCpuUsage mCpuUsage; // instantaneous thread CPU usage in wall clock ns 336 CentralTendencyStatistics mWcStats; // statistics on thread CPU usage in wall clock ns 337 338 CentralTendencyStatistics mHzStats; // statistics on thread CPU usage in cycles 339 340 int mCpuNum; // thread's current CPU number 341 int mCpukHz; // frequency of thread's current CPU in kHz 342#endif 343}; 344 345CpuStats::CpuStats() 346#ifdef DEBUG_CPU_USAGE 347 : mCpuNum(-1), mCpukHz(-1) 348#endif 349{ 350} 351 352void CpuStats::sample(const String8 &title 353#ifndef DEBUG_CPU_USAGE 354 __unused 355#endif 356 ) { 357#ifdef DEBUG_CPU_USAGE 358 // get current thread's delta CPU time in wall clock ns 359 double wcNs; 360 bool valid = mCpuUsage.sampleAndEnable(wcNs); 361 362 // record sample for wall clock statistics 363 if (valid) { 364 mWcStats.sample(wcNs); 365 } 366 367 // get the current CPU number 368 int cpuNum = sched_getcpu(); 369 370 // get the current CPU frequency in kHz 371 int cpukHz = mCpuUsage.getCpukHz(cpuNum); 372 373 // check if either CPU number or frequency changed 374 if (cpuNum != mCpuNum || cpukHz != mCpukHz) { 375 mCpuNum = cpuNum; 376 mCpukHz = cpukHz; 377 // ignore sample for purposes of cycles 378 valid = false; 379 } 380 381 // if no change in CPU number or frequency, then record sample for cycle statistics 382 if (valid && mCpukHz > 0) { 383 double cycles = wcNs * cpukHz * 0.000001; 384 mHzStats.sample(cycles); 385 } 386 387 unsigned n = mWcStats.n(); 388 // mCpuUsage.elapsed() is expensive, so don't call it every loop 389 if ((n & 127) == 1) { 390 long long elapsed = mCpuUsage.elapsed(); 391 if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) { 392 double perLoop = elapsed / (double) n; 393 double perLoop100 = perLoop * 0.01; 394 double perLoop1k = perLoop * 0.001; 395 double mean = mWcStats.mean(); 396 double stddev = mWcStats.stddev(); 397 double minimum = mWcStats.minimum(); 398 double maximum = mWcStats.maximum(); 399 double meanCycles = mHzStats.mean(); 400 double stddevCycles = mHzStats.stddev(); 401 double minCycles = mHzStats.minimum(); 402 double maxCycles = mHzStats.maximum(); 403 mCpuUsage.resetElapsed(); 404 mWcStats.reset(); 405 mHzStats.reset(); 406 ALOGD("CPU usage for %s over past %.1f secs\n" 407 " (%u mixer loops at %.1f mean ms per loop):\n" 408 " us per mix loop: mean=%.0f stddev=%.0f min=%.0f max=%.0f\n" 409 " %% of wall: mean=%.1f stddev=%.1f min=%.1f max=%.1f\n" 410 " MHz: mean=%.1f, stddev=%.1f, min=%.1f max=%.1f", 411 title.string(), 412 elapsed * .000000001, n, perLoop * .000001, 413 mean * .001, 414 stddev * .001, 415 minimum * .001, 416 maximum * .001, 417 mean / perLoop100, 418 stddev / perLoop100, 419 minimum / perLoop100, 420 maximum / perLoop100, 421 meanCycles / perLoop1k, 422 stddevCycles / perLoop1k, 423 minCycles / perLoop1k, 424 maxCycles / perLoop1k); 425 426 } 427 } 428#endif 429}; 430 431// ---------------------------------------------------------------------------- 432// ThreadBase 433// ---------------------------------------------------------------------------- 434 435// static 436const char *AudioFlinger::ThreadBase::threadTypeToString(AudioFlinger::ThreadBase::type_t type) 437{ 438 switch (type) { 439 case MIXER: 440 return "MIXER"; 441 case DIRECT: 442 return "DIRECT"; 443 case DUPLICATING: 444 return "DUPLICATING"; 445 case RECORD: 446 return "RECORD"; 447 case OFFLOAD: 448 return "OFFLOAD"; 449 default: 450 return "unknown"; 451 } 452} 453 454String8 devicesToString(audio_devices_t devices) 455{ 456 static const struct mapping { 457 audio_devices_t mDevices; 458 const char * mString; 459 } mappingsOut[] = { 460 {AUDIO_DEVICE_OUT_EARPIECE, "EARPIECE"}, 461 {AUDIO_DEVICE_OUT_SPEAKER, "SPEAKER"}, 462 {AUDIO_DEVICE_OUT_WIRED_HEADSET, "WIRED_HEADSET"}, 463 {AUDIO_DEVICE_OUT_WIRED_HEADPHONE, "WIRED_HEADPHONE"}, 464 {AUDIO_DEVICE_OUT_BLUETOOTH_SCO, "BLUETOOTH_SCO"}, 465 {AUDIO_DEVICE_OUT_BLUETOOTH_SCO_HEADSET, "BLUETOOTH_SCO_HEADSET"}, 466 {AUDIO_DEVICE_OUT_BLUETOOTH_SCO_CARKIT, "BLUETOOTH_SCO_CARKIT"}, 467 {AUDIO_DEVICE_OUT_BLUETOOTH_A2DP, "BLUETOOTH_A2DP"}, 468 {AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES,"BLUETOOTH_A2DP_HEADPHONES"}, 469 {AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER, "BLUETOOTH_A2DP_SPEAKER"}, 470 {AUDIO_DEVICE_OUT_AUX_DIGITAL, "AUX_DIGITAL"}, 471 {AUDIO_DEVICE_OUT_HDMI, "HDMI"}, 472 {AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET,"ANLG_DOCK_HEADSET"}, 473 {AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET,"DGTL_DOCK_HEADSET"}, 474 {AUDIO_DEVICE_OUT_USB_ACCESSORY, "USB_ACCESSORY"}, 475 {AUDIO_DEVICE_OUT_USB_DEVICE, "USB_DEVICE"}, 476 {AUDIO_DEVICE_OUT_TELEPHONY_TX, "TELEPHONY_TX"}, 477 {AUDIO_DEVICE_OUT_LINE, "LINE"}, 478 {AUDIO_DEVICE_OUT_HDMI_ARC, "HDMI_ARC"}, 479 {AUDIO_DEVICE_OUT_SPDIF, "SPDIF"}, 480 {AUDIO_DEVICE_OUT_FM, "FM"}, 481 {AUDIO_DEVICE_OUT_AUX_LINE, "AUX_LINE"}, 482 {AUDIO_DEVICE_OUT_SPEAKER_SAFE, "SPEAKER_SAFE"}, 483 {AUDIO_DEVICE_OUT_IP, "IP"}, 484 {AUDIO_DEVICE_OUT_BUS, "BUS"}, 485 {AUDIO_DEVICE_NONE, "NONE"}, // must be last 486 }, mappingsIn[] = { 487 {AUDIO_DEVICE_IN_COMMUNICATION, "COMMUNICATION"}, 488 {AUDIO_DEVICE_IN_AMBIENT, "AMBIENT"}, 489 {AUDIO_DEVICE_IN_BUILTIN_MIC, "BUILTIN_MIC"}, 490 {AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET, "BLUETOOTH_SCO_HEADSET"}, 491 {AUDIO_DEVICE_IN_WIRED_HEADSET, "WIRED_HEADSET"}, 492 {AUDIO_DEVICE_IN_AUX_DIGITAL, "AUX_DIGITAL"}, 493 {AUDIO_DEVICE_IN_VOICE_CALL, "VOICE_CALL"}, 494 {AUDIO_DEVICE_IN_TELEPHONY_RX, "TELEPHONY_RX"}, 495 {AUDIO_DEVICE_IN_BACK_MIC, "BACK_MIC"}, 496 {AUDIO_DEVICE_IN_REMOTE_SUBMIX, "REMOTE_SUBMIX"}, 497 {AUDIO_DEVICE_IN_ANLG_DOCK_HEADSET, "ANLG_DOCK_HEADSET"}, 498 {AUDIO_DEVICE_IN_DGTL_DOCK_HEADSET, "DGTL_DOCK_HEADSET"}, 499 {AUDIO_DEVICE_IN_USB_ACCESSORY, "USB_ACCESSORY"}, 500 {AUDIO_DEVICE_IN_USB_DEVICE, "USB_DEVICE"}, 501 {AUDIO_DEVICE_IN_FM_TUNER, "FM_TUNER"}, 502 {AUDIO_DEVICE_IN_TV_TUNER, "TV_TUNER"}, 503 {AUDIO_DEVICE_IN_LINE, "LINE"}, 504 {AUDIO_DEVICE_IN_SPDIF, "SPDIF"}, 505 {AUDIO_DEVICE_IN_BLUETOOTH_A2DP, "BLUETOOTH_A2DP"}, 506 {AUDIO_DEVICE_IN_LOOPBACK, "LOOPBACK"}, 507 {AUDIO_DEVICE_IN_IP, "IP"}, 508 {AUDIO_DEVICE_IN_BUS, "BUS"}, 509 {AUDIO_DEVICE_NONE, "NONE"}, // must be last 510 }; 511 String8 result; 512 audio_devices_t allDevices = AUDIO_DEVICE_NONE; 513 const mapping *entry; 514 if (devices & AUDIO_DEVICE_BIT_IN) { 515 devices &= ~AUDIO_DEVICE_BIT_IN; 516 entry = mappingsIn; 517 } else { 518 entry = mappingsOut; 519 } 520 for ( ; entry->mDevices != AUDIO_DEVICE_NONE; entry++) { 521 allDevices = (audio_devices_t) (allDevices | entry->mDevices); 522 if (devices & entry->mDevices) { 523 if (!result.isEmpty()) { 524 result.append("|"); 525 } 526 result.append(entry->mString); 527 } 528 } 529 if (devices & ~allDevices) { 530 if (!result.isEmpty()) { 531 result.append("|"); 532 } 533 result.appendFormat("0x%X", devices & ~allDevices); 534 } 535 if (result.isEmpty()) { 536 result.append(entry->mString); 537 } 538 return result; 539} 540 541String8 inputFlagsToString(audio_input_flags_t flags) 542{ 543 static const struct mapping { 544 audio_input_flags_t mFlag; 545 const char * mString; 546 } mappings[] = { 547 {AUDIO_INPUT_FLAG_FAST, "FAST"}, 548 {AUDIO_INPUT_FLAG_HW_HOTWORD, "HW_HOTWORD"}, 549 {AUDIO_INPUT_FLAG_RAW, "RAW"}, 550 {AUDIO_INPUT_FLAG_SYNC, "SYNC"}, 551 {AUDIO_INPUT_FLAG_NONE, "NONE"}, // must be last 552 }; 553 String8 result; 554 audio_input_flags_t allFlags = AUDIO_INPUT_FLAG_NONE; 555 const mapping *entry; 556 for (entry = mappings; entry->mFlag != AUDIO_INPUT_FLAG_NONE; entry++) { 557 allFlags = (audio_input_flags_t) (allFlags | entry->mFlag); 558 if (flags & entry->mFlag) { 559 if (!result.isEmpty()) { 560 result.append("|"); 561 } 562 result.append(entry->mString); 563 } 564 } 565 if (flags & ~allFlags) { 566 if (!result.isEmpty()) { 567 result.append("|"); 568 } 569 result.appendFormat("0x%X", flags & ~allFlags); 570 } 571 if (result.isEmpty()) { 572 result.append(entry->mString); 573 } 574 return result; 575} 576 577String8 outputFlagsToString(audio_output_flags_t flags) 578{ 579 static const struct mapping { 580 audio_output_flags_t mFlag; 581 const char * mString; 582 } mappings[] = { 583 {AUDIO_OUTPUT_FLAG_DIRECT, "DIRECT"}, 584 {AUDIO_OUTPUT_FLAG_PRIMARY, "PRIMARY"}, 585 {AUDIO_OUTPUT_FLAG_FAST, "FAST"}, 586 {AUDIO_OUTPUT_FLAG_DEEP_BUFFER, "DEEP_BUFFER"}, 587 {AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD,"COMPRESS_OFFLOAD"}, 588 {AUDIO_OUTPUT_FLAG_NON_BLOCKING, "NON_BLOCKING"}, 589 {AUDIO_OUTPUT_FLAG_HW_AV_SYNC, "HW_AV_SYNC"}, 590 {AUDIO_OUTPUT_FLAG_RAW, "RAW"}, 591 {AUDIO_OUTPUT_FLAG_SYNC, "SYNC"}, 592 {AUDIO_OUTPUT_FLAG_IEC958_NONAUDIO, "IEC958_NONAUDIO"}, 593 {AUDIO_OUTPUT_FLAG_NONE, "NONE"}, // must be last 594 }; 595 String8 result; 596 audio_output_flags_t allFlags = AUDIO_OUTPUT_FLAG_NONE; 597 const mapping *entry; 598 for (entry = mappings; entry->mFlag != AUDIO_OUTPUT_FLAG_NONE; entry++) { 599 allFlags = (audio_output_flags_t) (allFlags | entry->mFlag); 600 if (flags & entry->mFlag) { 601 if (!result.isEmpty()) { 602 result.append("|"); 603 } 604 result.append(entry->mString); 605 } 606 } 607 if (flags & ~allFlags) { 608 if (!result.isEmpty()) { 609 result.append("|"); 610 } 611 result.appendFormat("0x%X", flags & ~allFlags); 612 } 613 if (result.isEmpty()) { 614 result.append(entry->mString); 615 } 616 return result; 617} 618 619const char *sourceToString(audio_source_t source) 620{ 621 switch (source) { 622 case AUDIO_SOURCE_DEFAULT: return "default"; 623 case AUDIO_SOURCE_MIC: return "mic"; 624 case AUDIO_SOURCE_VOICE_UPLINK: return "voice uplink"; 625 case AUDIO_SOURCE_VOICE_DOWNLINK: return "voice downlink"; 626 case AUDIO_SOURCE_VOICE_CALL: return "voice call"; 627 case AUDIO_SOURCE_CAMCORDER: return "camcorder"; 628 case AUDIO_SOURCE_VOICE_RECOGNITION: return "voice recognition"; 629 case AUDIO_SOURCE_VOICE_COMMUNICATION: return "voice communication"; 630 case AUDIO_SOURCE_REMOTE_SUBMIX: return "remote submix"; 631 case AUDIO_SOURCE_UNPROCESSED: return "unprocessed"; 632 case AUDIO_SOURCE_FM_TUNER: return "FM tuner"; 633 case AUDIO_SOURCE_HOTWORD: return "hotword"; 634 default: return "unknown"; 635 } 636} 637 638AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, 639 audio_devices_t outDevice, audio_devices_t inDevice, type_t type, bool systemReady) 640 : Thread(false /*canCallJava*/), 641 mType(type), 642 mAudioFlinger(audioFlinger), 643 // mSampleRate, mFrameCount, mChannelMask, mChannelCount, mFrameSize, mFormat, mBufferSize 644 // are set by PlaybackThread::readOutputParameters_l() or 645 // RecordThread::readInputParameters_l() 646 //FIXME: mStandby should be true here. Is this some kind of hack? 647 mStandby(false), mOutDevice(outDevice), mInDevice(inDevice), 648 mPrevOutDevice(AUDIO_DEVICE_NONE), mPrevInDevice(AUDIO_DEVICE_NONE), 649 mAudioSource(AUDIO_SOURCE_DEFAULT), mId(id), 650 // mName will be set by concrete (non-virtual) subclass 651 mDeathRecipient(new PMDeathRecipient(this)), 652 mSystemReady(systemReady), 653 mNotifiedBatteryStart(false) 654{ 655 memset(&mPatch, 0, sizeof(struct audio_patch)); 656} 657 658AudioFlinger::ThreadBase::~ThreadBase() 659{ 660 // mConfigEvents should be empty, but just in case it isn't, free the memory it owns 661 mConfigEvents.clear(); 662 663 // do not lock the mutex in destructor 664 releaseWakeLock_l(); 665 if (mPowerManager != 0) { 666 sp<IBinder> binder = IInterface::asBinder(mPowerManager); 667 binder->unlinkToDeath(mDeathRecipient); 668 } 669} 670 671status_t AudioFlinger::ThreadBase::readyToRun() 672{ 673 status_t status = initCheck(); 674 if (status == NO_ERROR) { 675 ALOGI("AudioFlinger's thread %p ready to run", this); 676 } else { 677 ALOGE("No working audio driver found."); 678 } 679 return status; 680} 681 682void AudioFlinger::ThreadBase::exit() 683{ 684 ALOGV("ThreadBase::exit"); 685 // do any cleanup required for exit to succeed 686 preExit(); 687 { 688 // This lock prevents the following race in thread (uniprocessor for illustration): 689 // if (!exitPending()) { 690 // // context switch from here to exit() 691 // // exit() calls requestExit(), what exitPending() observes 692 // // exit() calls signal(), which is dropped since no waiters 693 // // context switch back from exit() to here 694 // mWaitWorkCV.wait(...); 695 // // now thread is hung 696 // } 697 AutoMutex lock(mLock); 698 requestExit(); 699 mWaitWorkCV.broadcast(); 700 } 701 // When Thread::requestExitAndWait is made virtual and this method is renamed to 702 // "virtual status_t requestExitAndWait()", replace by "return Thread::requestExitAndWait();" 703 requestExitAndWait(); 704} 705 706status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs) 707{ 708 ALOGV("ThreadBase::setParameters() %s", keyValuePairs.string()); 709 Mutex::Autolock _l(mLock); 710 711 return sendSetParameterConfigEvent_l(keyValuePairs); 712} 713 714// sendConfigEvent_l() must be called with ThreadBase::mLock held 715// Can temporarily release the lock if waiting for a reply from processConfigEvents_l(). 716status_t AudioFlinger::ThreadBase::sendConfigEvent_l(sp<ConfigEvent>& event) 717{ 718 status_t status = NO_ERROR; 719 720 if (event->mRequiresSystemReady && !mSystemReady) { 721 event->mWaitStatus = false; 722 mPendingConfigEvents.add(event); 723 return status; 724 } 725 mConfigEvents.add(event); 726 ALOGV("sendConfigEvent_l() num events %zu event %d", mConfigEvents.size(), event->mType); 727 mWaitWorkCV.signal(); 728 mLock.unlock(); 729 { 730 Mutex::Autolock _l(event->mLock); 731 while (event->mWaitStatus) { 732 if (event->mCond.waitRelative(event->mLock, kConfigEventTimeoutNs) != NO_ERROR) { 733 event->mStatus = TIMED_OUT; 734 event->mWaitStatus = false; 735 } 736 } 737 status = event->mStatus; 738 } 739 mLock.lock(); 740 return status; 741} 742 743void AudioFlinger::ThreadBase::sendIoConfigEvent(audio_io_config_event event, pid_t pid) 744{ 745 Mutex::Autolock _l(mLock); 746 sendIoConfigEvent_l(event, pid); 747} 748 749// sendIoConfigEvent_l() must be called with ThreadBase::mLock held 750void AudioFlinger::ThreadBase::sendIoConfigEvent_l(audio_io_config_event event, pid_t pid) 751{ 752 sp<ConfigEvent> configEvent = (ConfigEvent *)new IoConfigEvent(event, pid); 753 sendConfigEvent_l(configEvent); 754} 755 756void AudioFlinger::ThreadBase::sendPrioConfigEvent(pid_t pid, pid_t tid, int32_t prio) 757{ 758 Mutex::Autolock _l(mLock); 759 sendPrioConfigEvent_l(pid, tid, prio); 760} 761 762// sendPrioConfigEvent_l() must be called with ThreadBase::mLock held 763void AudioFlinger::ThreadBase::sendPrioConfigEvent_l(pid_t pid, pid_t tid, int32_t prio) 764{ 765 sp<ConfigEvent> configEvent = (ConfigEvent *)new PrioConfigEvent(pid, tid, prio); 766 sendConfigEvent_l(configEvent); 767} 768 769// sendSetParameterConfigEvent_l() must be called with ThreadBase::mLock held 770status_t AudioFlinger::ThreadBase::sendSetParameterConfigEvent_l(const String8& keyValuePair) 771{ 772 sp<ConfigEvent> configEvent; 773 AudioParameter param(keyValuePair); 774 int value; 775 if (param.getInt(String8(AUDIO_PARAMETER_MONO_OUTPUT), value) == NO_ERROR) { 776 setMasterMono_l(value != 0); 777 if (param.size() == 1) { 778 return NO_ERROR; // should be a solo parameter - we don't pass down 779 } 780 param.remove(String8(AUDIO_PARAMETER_MONO_OUTPUT)); 781 configEvent = new SetParameterConfigEvent(param.toString()); 782 } else { 783 configEvent = new SetParameterConfigEvent(keyValuePair); 784 } 785 return sendConfigEvent_l(configEvent); 786} 787 788status_t AudioFlinger::ThreadBase::sendCreateAudioPatchConfigEvent( 789 const struct audio_patch *patch, 790 audio_patch_handle_t *handle) 791{ 792 Mutex::Autolock _l(mLock); 793 sp<ConfigEvent> configEvent = (ConfigEvent *)new CreateAudioPatchConfigEvent(*patch, *handle); 794 status_t status = sendConfigEvent_l(configEvent); 795 if (status == NO_ERROR) { 796 CreateAudioPatchConfigEventData *data = 797 (CreateAudioPatchConfigEventData *)configEvent->mData.get(); 798 *handle = data->mHandle; 799 } 800 return status; 801} 802 803status_t AudioFlinger::ThreadBase::sendReleaseAudioPatchConfigEvent( 804 const audio_patch_handle_t handle) 805{ 806 Mutex::Autolock _l(mLock); 807 sp<ConfigEvent> configEvent = (ConfigEvent *)new ReleaseAudioPatchConfigEvent(handle); 808 return sendConfigEvent_l(configEvent); 809} 810 811 812// post condition: mConfigEvents.isEmpty() 813void AudioFlinger::ThreadBase::processConfigEvents_l() 814{ 815 bool configChanged = false; 816 817 while (!mConfigEvents.isEmpty()) { 818 ALOGV("processConfigEvents_l() remaining events %zu", mConfigEvents.size()); 819 sp<ConfigEvent> event = mConfigEvents[0]; 820 mConfigEvents.removeAt(0); 821 switch (event->mType) { 822 case CFG_EVENT_PRIO: { 823 PrioConfigEventData *data = (PrioConfigEventData *)event->mData.get(); 824 // FIXME Need to understand why this has to be done asynchronously 825 int err = requestPriority(data->mPid, data->mTid, data->mPrio, 826 true /*asynchronous*/); 827 if (err != 0) { 828 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 829 data->mPrio, data->mPid, data->mTid, err); 830 } 831 } break; 832 case CFG_EVENT_IO: { 833 IoConfigEventData *data = (IoConfigEventData *)event->mData.get(); 834 ioConfigChanged(data->mEvent, data->mPid); 835 } break; 836 case CFG_EVENT_SET_PARAMETER: { 837 SetParameterConfigEventData *data = (SetParameterConfigEventData *)event->mData.get(); 838 if (checkForNewParameter_l(data->mKeyValuePairs, event->mStatus)) { 839 configChanged = true; 840 } 841 } break; 842 case CFG_EVENT_CREATE_AUDIO_PATCH: { 843 CreateAudioPatchConfigEventData *data = 844 (CreateAudioPatchConfigEventData *)event->mData.get(); 845 event->mStatus = createAudioPatch_l(&data->mPatch, &data->mHandle); 846 } break; 847 case CFG_EVENT_RELEASE_AUDIO_PATCH: { 848 ReleaseAudioPatchConfigEventData *data = 849 (ReleaseAudioPatchConfigEventData *)event->mData.get(); 850 event->mStatus = releaseAudioPatch_l(data->mHandle); 851 } break; 852 default: 853 ALOG_ASSERT(false, "processConfigEvents_l() unknown event type %d", event->mType); 854 break; 855 } 856 { 857 Mutex::Autolock _l(event->mLock); 858 if (event->mWaitStatus) { 859 event->mWaitStatus = false; 860 event->mCond.signal(); 861 } 862 } 863 ALOGV_IF(mConfigEvents.isEmpty(), "processConfigEvents_l() DONE thread %p", this); 864 } 865 866 if (configChanged) { 867 cacheParameters_l(); 868 } 869} 870 871String8 channelMaskToString(audio_channel_mask_t mask, bool output) { 872 String8 s; 873 const audio_channel_representation_t representation = 874 audio_channel_mask_get_representation(mask); 875 876 switch (representation) { 877 case AUDIO_CHANNEL_REPRESENTATION_POSITION: { 878 if (output) { 879 if (mask & AUDIO_CHANNEL_OUT_FRONT_LEFT) s.append("front-left, "); 880 if (mask & AUDIO_CHANNEL_OUT_FRONT_RIGHT) s.append("front-right, "); 881 if (mask & AUDIO_CHANNEL_OUT_FRONT_CENTER) s.append("front-center, "); 882 if (mask & AUDIO_CHANNEL_OUT_LOW_FREQUENCY) s.append("low freq, "); 883 if (mask & AUDIO_CHANNEL_OUT_BACK_LEFT) s.append("back-left, "); 884 if (mask & AUDIO_CHANNEL_OUT_BACK_RIGHT) s.append("back-right, "); 885 if (mask & AUDIO_CHANNEL_OUT_FRONT_LEFT_OF_CENTER) s.append("front-left-of-center, "); 886 if (mask & AUDIO_CHANNEL_OUT_FRONT_RIGHT_OF_CENTER) s.append("front-right-of-center, "); 887 if (mask & AUDIO_CHANNEL_OUT_BACK_CENTER) s.append("back-center, "); 888 if (mask & AUDIO_CHANNEL_OUT_SIDE_LEFT) s.append("side-left, "); 889 if (mask & AUDIO_CHANNEL_OUT_SIDE_RIGHT) s.append("side-right, "); 890 if (mask & AUDIO_CHANNEL_OUT_TOP_CENTER) s.append("top-center ,"); 891 if (mask & AUDIO_CHANNEL_OUT_TOP_FRONT_LEFT) s.append("top-front-left, "); 892 if (mask & AUDIO_CHANNEL_OUT_TOP_FRONT_CENTER) s.append("top-front-center, "); 893 if (mask & AUDIO_CHANNEL_OUT_TOP_FRONT_RIGHT) s.append("top-front-right, "); 894 if (mask & AUDIO_CHANNEL_OUT_TOP_BACK_LEFT) s.append("top-back-left, "); 895 if (mask & AUDIO_CHANNEL_OUT_TOP_BACK_CENTER) s.append("top-back-center, " ); 896 if (mask & AUDIO_CHANNEL_OUT_TOP_BACK_RIGHT) s.append("top-back-right, " ); 897 if (mask & ~AUDIO_CHANNEL_OUT_ALL) s.append("unknown, "); 898 } else { 899 if (mask & AUDIO_CHANNEL_IN_LEFT) s.append("left, "); 900 if (mask & AUDIO_CHANNEL_IN_RIGHT) s.append("right, "); 901 if (mask & AUDIO_CHANNEL_IN_FRONT) s.append("front, "); 902 if (mask & AUDIO_CHANNEL_IN_BACK) s.append("back, "); 903 if (mask & AUDIO_CHANNEL_IN_LEFT_PROCESSED) s.append("left-processed, "); 904 if (mask & AUDIO_CHANNEL_IN_RIGHT_PROCESSED) s.append("right-processed, "); 905 if (mask & AUDIO_CHANNEL_IN_FRONT_PROCESSED) s.append("front-processed, "); 906 if (mask & AUDIO_CHANNEL_IN_BACK_PROCESSED) s.append("back-processed, "); 907 if (mask & AUDIO_CHANNEL_IN_PRESSURE) s.append("pressure, "); 908 if (mask & AUDIO_CHANNEL_IN_X_AXIS) s.append("X, "); 909 if (mask & AUDIO_CHANNEL_IN_Y_AXIS) s.append("Y, "); 910 if (mask & AUDIO_CHANNEL_IN_Z_AXIS) s.append("Z, "); 911 if (mask & AUDIO_CHANNEL_IN_VOICE_UPLINK) s.append("voice-uplink, "); 912 if (mask & AUDIO_CHANNEL_IN_VOICE_DNLINK) s.append("voice-dnlink, "); 913 if (mask & ~AUDIO_CHANNEL_IN_ALL) s.append("unknown, "); 914 } 915 const int len = s.length(); 916 if (len > 2) { 917 (void) s.lockBuffer(len); // needed? 918 s.unlockBuffer(len - 2); // remove trailing ", " 919 } 920 return s; 921 } 922 case AUDIO_CHANNEL_REPRESENTATION_INDEX: 923 s.appendFormat("index mask, bits:%#x", audio_channel_mask_get_bits(mask)); 924 return s; 925 default: 926 s.appendFormat("unknown mask, representation:%d bits:%#x", 927 representation, audio_channel_mask_get_bits(mask)); 928 return s; 929 } 930} 931 932void AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args __unused) 933{ 934 const size_t SIZE = 256; 935 char buffer[SIZE]; 936 String8 result; 937 938 bool locked = AudioFlinger::dumpTryLock(mLock); 939 if (!locked) { 940 dprintf(fd, "thread %p may be deadlocked\n", this); 941 } 942 943 dprintf(fd, " Thread name: %s\n", mThreadName); 944 dprintf(fd, " I/O handle: %d\n", mId); 945 dprintf(fd, " TID: %d\n", getTid()); 946 dprintf(fd, " Standby: %s\n", mStandby ? "yes" : "no"); 947 dprintf(fd, " Sample rate: %u Hz\n", mSampleRate); 948 dprintf(fd, " HAL frame count: %zu\n", mFrameCount); 949 dprintf(fd, " HAL format: 0x%x (%s)\n", mHALFormat, formatToString(mHALFormat)); 950 dprintf(fd, " HAL buffer size: %zu bytes\n", mBufferSize); 951 dprintf(fd, " Channel count: %u\n", mChannelCount); 952 dprintf(fd, " Channel mask: 0x%08x (%s)\n", mChannelMask, 953 channelMaskToString(mChannelMask, mType != RECORD).string()); 954 dprintf(fd, " Processing format: 0x%x (%s)\n", mFormat, formatToString(mFormat)); 955 dprintf(fd, " Processing frame size: %zu bytes\n", mFrameSize); 956 dprintf(fd, " Pending config events:"); 957 size_t numConfig = mConfigEvents.size(); 958 if (numConfig) { 959 for (size_t i = 0; i < numConfig; i++) { 960 mConfigEvents[i]->dump(buffer, SIZE); 961 dprintf(fd, "\n %s", buffer); 962 } 963 dprintf(fd, "\n"); 964 } else { 965 dprintf(fd, " none\n"); 966 } 967 dprintf(fd, " Output device: %#x (%s)\n", mOutDevice, devicesToString(mOutDevice).string()); 968 dprintf(fd, " Input device: %#x (%s)\n", mInDevice, devicesToString(mInDevice).string()); 969 dprintf(fd, " Audio source: %d (%s)\n", mAudioSource, sourceToString(mAudioSource)); 970 971 if (locked) { 972 mLock.unlock(); 973 } 974} 975 976void AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args) 977{ 978 const size_t SIZE = 256; 979 char buffer[SIZE]; 980 String8 result; 981 982 size_t numEffectChains = mEffectChains.size(); 983 snprintf(buffer, SIZE, " %zu Effect Chains\n", numEffectChains); 984 write(fd, buffer, strlen(buffer)); 985 986 for (size_t i = 0; i < numEffectChains; ++i) { 987 sp<EffectChain> chain = mEffectChains[i]; 988 if (chain != 0) { 989 chain->dump(fd, args); 990 } 991 } 992} 993 994void AudioFlinger::ThreadBase::acquireWakeLock(int uid) 995{ 996 Mutex::Autolock _l(mLock); 997 acquireWakeLock_l(uid); 998} 999 1000String16 AudioFlinger::ThreadBase::getWakeLockTag() 1001{ 1002 switch (mType) { 1003 case MIXER: 1004 return String16("AudioMix"); 1005 case DIRECT: 1006 return String16("AudioDirectOut"); 1007 case DUPLICATING: 1008 return String16("AudioDup"); 1009 case RECORD: 1010 return String16("AudioIn"); 1011 case OFFLOAD: 1012 return String16("AudioOffload"); 1013 default: 1014 ALOG_ASSERT(false); 1015 return String16("AudioUnknown"); 1016 } 1017} 1018 1019void AudioFlinger::ThreadBase::acquireWakeLock_l(int uid) 1020{ 1021 getPowerManager_l(); 1022 if (mPowerManager != 0) { 1023 sp<IBinder> binder = new BBinder(); 1024 status_t status; 1025 if (uid >= 0) { 1026 status = mPowerManager->acquireWakeLockWithUid(POWERMANAGER_PARTIAL_WAKE_LOCK, 1027 binder, 1028 getWakeLockTag(), 1029 String16("audioserver"), 1030 uid, 1031 true /* FIXME force oneway contrary to .aidl */); 1032 } else { 1033 status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK, 1034 binder, 1035 getWakeLockTag(), 1036 String16("audioserver"), 1037 true /* FIXME force oneway contrary to .aidl */); 1038 } 1039 if (status == NO_ERROR) { 1040 mWakeLockToken = binder; 1041 } 1042 ALOGV("acquireWakeLock_l() %s status %d", mThreadName, status); 1043 } 1044 1045 if (!mNotifiedBatteryStart) { 1046 BatteryNotifier::getInstance().noteStartAudio(); 1047 mNotifiedBatteryStart = true; 1048 } 1049 gBoottime.acquire(mWakeLockToken); 1050 mTimestamp.mTimebaseOffset[ExtendedTimestamp::TIMEBASE_BOOTTIME] = 1051 gBoottime.getBoottimeOffset(); 1052} 1053 1054void AudioFlinger::ThreadBase::releaseWakeLock() 1055{ 1056 Mutex::Autolock _l(mLock); 1057 releaseWakeLock_l(); 1058} 1059 1060void AudioFlinger::ThreadBase::releaseWakeLock_l() 1061{ 1062 gBoottime.release(mWakeLockToken); 1063 if (mWakeLockToken != 0) { 1064 ALOGV("releaseWakeLock_l() %s", mThreadName); 1065 if (mPowerManager != 0) { 1066 mPowerManager->releaseWakeLock(mWakeLockToken, 0, 1067 true /* FIXME force oneway contrary to .aidl */); 1068 } 1069 mWakeLockToken.clear(); 1070 } 1071 1072 if (mNotifiedBatteryStart) { 1073 BatteryNotifier::getInstance().noteStopAudio(); 1074 mNotifiedBatteryStart = false; 1075 } 1076} 1077 1078void AudioFlinger::ThreadBase::updateWakeLockUids(const SortedVector<int> &uids) { 1079 Mutex::Autolock _l(mLock); 1080 updateWakeLockUids_l(uids); 1081} 1082 1083void AudioFlinger::ThreadBase::getPowerManager_l() { 1084 if (mSystemReady && mPowerManager == 0) { 1085 // use checkService() to avoid blocking if power service is not up yet 1086 sp<IBinder> binder = 1087 defaultServiceManager()->checkService(String16("power")); 1088 if (binder == 0) { 1089 ALOGW("Thread %s cannot connect to the power manager service", mThreadName); 1090 } else { 1091 mPowerManager = interface_cast<IPowerManager>(binder); 1092 binder->linkToDeath(mDeathRecipient); 1093 } 1094 } 1095} 1096 1097void AudioFlinger::ThreadBase::updateWakeLockUids_l(const SortedVector<int> &uids) { 1098 getPowerManager_l(); 1099 if (mWakeLockToken == NULL) { // token may be NULL if AudioFlinger::systemReady() not called. 1100 if (mSystemReady) { 1101 ALOGE("no wake lock to update, but system ready!"); 1102 } else { 1103 ALOGW("no wake lock to update, system not ready yet"); 1104 } 1105 return; 1106 } 1107 if (mPowerManager != 0) { 1108 sp<IBinder> binder = new BBinder(); 1109 status_t status; 1110 status = mPowerManager->updateWakeLockUids(mWakeLockToken, uids.size(), uids.array(), 1111 true /* FIXME force oneway contrary to .aidl */); 1112 ALOGV("updateWakeLockUids_l() %s status %d", mThreadName, status); 1113 } 1114} 1115 1116void AudioFlinger::ThreadBase::clearPowerManager() 1117{ 1118 Mutex::Autolock _l(mLock); 1119 releaseWakeLock_l(); 1120 mPowerManager.clear(); 1121} 1122 1123void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who __unused) 1124{ 1125 sp<ThreadBase> thread = mThread.promote(); 1126 if (thread != 0) { 1127 thread->clearPowerManager(); 1128 } 1129 ALOGW("power manager service died !!!"); 1130} 1131 1132void AudioFlinger::ThreadBase::setEffectSuspended( 1133 const effect_uuid_t *type, bool suspend, audio_session_t sessionId) 1134{ 1135 Mutex::Autolock _l(mLock); 1136 setEffectSuspended_l(type, suspend, sessionId); 1137} 1138 1139void AudioFlinger::ThreadBase::setEffectSuspended_l( 1140 const effect_uuid_t *type, bool suspend, audio_session_t sessionId) 1141{ 1142 sp<EffectChain> chain = getEffectChain_l(sessionId); 1143 if (chain != 0) { 1144 if (type != NULL) { 1145 chain->setEffectSuspended_l(type, suspend); 1146 } else { 1147 chain->setEffectSuspendedAll_l(suspend); 1148 } 1149 } 1150 1151 updateSuspendedSessions_l(type, suspend, sessionId); 1152} 1153 1154void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain) 1155{ 1156 ssize_t index = mSuspendedSessions.indexOfKey(chain->sessionId()); 1157 if (index < 0) { 1158 return; 1159 } 1160 1161 const KeyedVector <int, sp<SuspendedSessionDesc> >& sessionEffects = 1162 mSuspendedSessions.valueAt(index); 1163 1164 for (size_t i = 0; i < sessionEffects.size(); i++) { 1165 sp<SuspendedSessionDesc> desc = sessionEffects.valueAt(i); 1166 for (int j = 0; j < desc->mRefCount; j++) { 1167 if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) { 1168 chain->setEffectSuspendedAll_l(true); 1169 } else { 1170 ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x", 1171 desc->mType.timeLow); 1172 chain->setEffectSuspended_l(&desc->mType, true); 1173 } 1174 } 1175 } 1176} 1177 1178void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type, 1179 bool suspend, 1180 audio_session_t sessionId) 1181{ 1182 ssize_t index = mSuspendedSessions.indexOfKey(sessionId); 1183 1184 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects; 1185 1186 if (suspend) { 1187 if (index >= 0) { 1188 sessionEffects = mSuspendedSessions.valueAt(index); 1189 } else { 1190 mSuspendedSessions.add(sessionId, sessionEffects); 1191 } 1192 } else { 1193 if (index < 0) { 1194 return; 1195 } 1196 sessionEffects = mSuspendedSessions.valueAt(index); 1197 } 1198 1199 1200 int key = EffectChain::kKeyForSuspendAll; 1201 if (type != NULL) { 1202 key = type->timeLow; 1203 } 1204 index = sessionEffects.indexOfKey(key); 1205 1206 sp<SuspendedSessionDesc> desc; 1207 if (suspend) { 1208 if (index >= 0) { 1209 desc = sessionEffects.valueAt(index); 1210 } else { 1211 desc = new SuspendedSessionDesc(); 1212 if (type != NULL) { 1213 desc->mType = *type; 1214 } 1215 sessionEffects.add(key, desc); 1216 ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key); 1217 } 1218 desc->mRefCount++; 1219 } else { 1220 if (index < 0) { 1221 return; 1222 } 1223 desc = sessionEffects.valueAt(index); 1224 if (--desc->mRefCount == 0) { 1225 ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key); 1226 sessionEffects.removeItemsAt(index); 1227 if (sessionEffects.isEmpty()) { 1228 ALOGV("updateSuspendedSessions_l() restore removing session %d", 1229 sessionId); 1230 mSuspendedSessions.removeItem(sessionId); 1231 } 1232 } 1233 } 1234 if (!sessionEffects.isEmpty()) { 1235 mSuspendedSessions.replaceValueFor(sessionId, sessionEffects); 1236 } 1237} 1238 1239void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 1240 bool enabled, 1241 audio_session_t sessionId) 1242{ 1243 Mutex::Autolock _l(mLock); 1244 checkSuspendOnEffectEnabled_l(effect, enabled, sessionId); 1245} 1246 1247void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 1248 bool enabled, 1249 audio_session_t sessionId) 1250{ 1251 if (mType != RECORD) { 1252 // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on 1253 // another session. This gives the priority to well behaved effect control panels 1254 // and applications not using global effects. 1255 // Enabling post processing in AUDIO_SESSION_OUTPUT_STAGE session does not affect 1256 // global effects 1257 if ((sessionId != AUDIO_SESSION_OUTPUT_MIX) && (sessionId != AUDIO_SESSION_OUTPUT_STAGE)) { 1258 setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX); 1259 } 1260 } 1261 1262 sp<EffectChain> chain = getEffectChain_l(sessionId); 1263 if (chain != 0) { 1264 chain->checkSuspendOnEffectEnabled(effect, enabled); 1265 } 1266} 1267 1268// ThreadBase::createEffect_l() must be called with AudioFlinger::mLock held 1269sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 1270 const sp<AudioFlinger::Client>& client, 1271 const sp<IEffectClient>& effectClient, 1272 int32_t priority, 1273 audio_session_t sessionId, 1274 effect_descriptor_t *desc, 1275 int *enabled, 1276 status_t *status) 1277{ 1278 sp<EffectModule> effect; 1279 sp<EffectHandle> handle; 1280 status_t lStatus; 1281 sp<EffectChain> chain; 1282 bool chainCreated = false; 1283 bool effectCreated = false; 1284 bool effectRegistered = false; 1285 1286 lStatus = initCheck(); 1287 if (lStatus != NO_ERROR) { 1288 ALOGW("createEffect_l() Audio driver not initialized."); 1289 goto Exit; 1290 } 1291 1292 // Reject any effect on Direct output threads for now, since the format of 1293 // mSinkBuffer is not guaranteed to be compatible with effect processing (PCM 16 stereo). 1294 if (mType == DIRECT) { 1295 ALOGW("createEffect_l() Cannot add effect %s on Direct output type thread %s", 1296 desc->name, mThreadName); 1297 lStatus = BAD_VALUE; 1298 goto Exit; 1299 } 1300 1301 // Reject any effect on mixer or duplicating multichannel sinks. 1302 // TODO: fix both format and multichannel issues with effects. 1303 if ((mType == MIXER || mType == DUPLICATING) && mChannelCount != FCC_2) { 1304 ALOGW("createEffect_l() Cannot add effect %s for multichannel(%d) %s threads", 1305 desc->name, mChannelCount, mType == MIXER ? "MIXER" : "DUPLICATING"); 1306 lStatus = BAD_VALUE; 1307 goto Exit; 1308 } 1309 1310 // Allow global effects only on offloaded and mixer threads 1311 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 1312 switch (mType) { 1313 case MIXER: 1314 case OFFLOAD: 1315 break; 1316 case DIRECT: 1317 case DUPLICATING: 1318 case RECORD: 1319 default: 1320 ALOGW("createEffect_l() Cannot add global effect %s on thread %s", 1321 desc->name, mThreadName); 1322 lStatus = BAD_VALUE; 1323 goto Exit; 1324 } 1325 } 1326 1327 // Only Pre processor effects are allowed on input threads and only on input threads 1328 if ((mType == RECORD) != ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { 1329 ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d", 1330 desc->name, desc->flags, mType); 1331 lStatus = BAD_VALUE; 1332 goto Exit; 1333 } 1334 1335 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 1336 1337 { // scope for mLock 1338 Mutex::Autolock _l(mLock); 1339 1340 // check for existing effect chain with the requested audio session 1341 chain = getEffectChain_l(sessionId); 1342 if (chain == 0) { 1343 // create a new chain for this session 1344 ALOGV("createEffect_l() new effect chain for session %d", sessionId); 1345 chain = new EffectChain(this, sessionId); 1346 addEffectChain_l(chain); 1347 chain->setStrategy(getStrategyForSession_l(sessionId)); 1348 chainCreated = true; 1349 } else { 1350 effect = chain->getEffectFromDesc_l(desc); 1351 } 1352 1353 ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get()); 1354 1355 if (effect == 0) { 1356 audio_unique_id_t id = mAudioFlinger->nextUniqueId(AUDIO_UNIQUE_ID_USE_EFFECT); 1357 // Check CPU and memory usage 1358 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 1359 if (lStatus != NO_ERROR) { 1360 goto Exit; 1361 } 1362 effectRegistered = true; 1363 // create a new effect module if none present in the chain 1364 effect = new EffectModule(this, chain, desc, id, sessionId); 1365 lStatus = effect->status(); 1366 if (lStatus != NO_ERROR) { 1367 goto Exit; 1368 } 1369 effect->setOffloaded(mType == OFFLOAD, mId); 1370 1371 lStatus = chain->addEffect_l(effect); 1372 if (lStatus != NO_ERROR) { 1373 goto Exit; 1374 } 1375 effectCreated = true; 1376 1377 effect->setDevice(mOutDevice); 1378 effect->setDevice(mInDevice); 1379 effect->setMode(mAudioFlinger->getMode()); 1380 effect->setAudioSource(mAudioSource); 1381 } 1382 // create effect handle and connect it to effect module 1383 handle = new EffectHandle(effect, client, effectClient, priority); 1384 lStatus = handle->initCheck(); 1385 if (lStatus == OK) { 1386 lStatus = effect->addHandle(handle.get()); 1387 } 1388 if (enabled != NULL) { 1389 *enabled = (int)effect->isEnabled(); 1390 } 1391 } 1392 1393Exit: 1394 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 1395 Mutex::Autolock _l(mLock); 1396 if (effectCreated) { 1397 chain->removeEffect_l(effect); 1398 } 1399 if (effectRegistered) { 1400 AudioSystem::unregisterEffect(effect->id()); 1401 } 1402 if (chainCreated) { 1403 removeEffectChain_l(chain); 1404 } 1405 handle.clear(); 1406 } 1407 1408 *status = lStatus; 1409 return handle; 1410} 1411 1412sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect(audio_session_t sessionId, 1413 int effectId) 1414{ 1415 Mutex::Autolock _l(mLock); 1416 return getEffect_l(sessionId, effectId); 1417} 1418 1419sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(audio_session_t sessionId, 1420 int effectId) 1421{ 1422 sp<EffectChain> chain = getEffectChain_l(sessionId); 1423 return chain != 0 ? chain->getEffectFromId_l(effectId) : 0; 1424} 1425 1426// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 1427// PlaybackThread::mLock held 1428status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 1429{ 1430 // check for existing effect chain with the requested audio session 1431 audio_session_t sessionId = effect->sessionId(); 1432 sp<EffectChain> chain = getEffectChain_l(sessionId); 1433 bool chainCreated = false; 1434 1435 ALOGD_IF((mType == OFFLOAD) && !effect->isOffloadable(), 1436 "addEffect_l() on offloaded thread %p: effect %s does not support offload flags %x", 1437 this, effect->desc().name, effect->desc().flags); 1438 1439 if (chain == 0) { 1440 // create a new chain for this session 1441 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 1442 chain = new EffectChain(this, sessionId); 1443 addEffectChain_l(chain); 1444 chain->setStrategy(getStrategyForSession_l(sessionId)); 1445 chainCreated = true; 1446 } 1447 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 1448 1449 if (chain->getEffectFromId_l(effect->id()) != 0) { 1450 ALOGW("addEffect_l() %p effect %s already present in chain %p", 1451 this, effect->desc().name, chain.get()); 1452 return BAD_VALUE; 1453 } 1454 1455 effect->setOffloaded(mType == OFFLOAD, mId); 1456 1457 status_t status = chain->addEffect_l(effect); 1458 if (status != NO_ERROR) { 1459 if (chainCreated) { 1460 removeEffectChain_l(chain); 1461 } 1462 return status; 1463 } 1464 1465 effect->setDevice(mOutDevice); 1466 effect->setDevice(mInDevice); 1467 effect->setMode(mAudioFlinger->getMode()); 1468 effect->setAudioSource(mAudioSource); 1469 return NO_ERROR; 1470} 1471 1472void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 1473 1474 ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 1475 effect_descriptor_t desc = effect->desc(); 1476 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 1477 detachAuxEffect_l(effect->id()); 1478 } 1479 1480 sp<EffectChain> chain = effect->chain().promote(); 1481 if (chain != 0) { 1482 // remove effect chain if removing last effect 1483 if (chain->removeEffect_l(effect) == 0) { 1484 removeEffectChain_l(chain); 1485 } 1486 } else { 1487 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 1488 } 1489} 1490 1491void AudioFlinger::ThreadBase::lockEffectChains_l( 1492 Vector< sp<AudioFlinger::EffectChain> >& effectChains) 1493{ 1494 effectChains = mEffectChains; 1495 for (size_t i = 0; i < mEffectChains.size(); i++) { 1496 mEffectChains[i]->lock(); 1497 } 1498} 1499 1500void AudioFlinger::ThreadBase::unlockEffectChains( 1501 const Vector< sp<AudioFlinger::EffectChain> >& effectChains) 1502{ 1503 for (size_t i = 0; i < effectChains.size(); i++) { 1504 effectChains[i]->unlock(); 1505 } 1506} 1507 1508sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(audio_session_t sessionId) 1509{ 1510 Mutex::Autolock _l(mLock); 1511 return getEffectChain_l(sessionId); 1512} 1513 1514sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(audio_session_t sessionId) 1515 const 1516{ 1517 size_t size = mEffectChains.size(); 1518 for (size_t i = 0; i < size; i++) { 1519 if (mEffectChains[i]->sessionId() == sessionId) { 1520 return mEffectChains[i]; 1521 } 1522 } 1523 return 0; 1524} 1525 1526void AudioFlinger::ThreadBase::setMode(audio_mode_t mode) 1527{ 1528 Mutex::Autolock _l(mLock); 1529 size_t size = mEffectChains.size(); 1530 for (size_t i = 0; i < size; i++) { 1531 mEffectChains[i]->setMode_l(mode); 1532 } 1533} 1534 1535void AudioFlinger::ThreadBase::getAudioPortConfig(struct audio_port_config *config) 1536{ 1537 config->type = AUDIO_PORT_TYPE_MIX; 1538 config->ext.mix.handle = mId; 1539 config->sample_rate = mSampleRate; 1540 config->format = mFormat; 1541 config->channel_mask = mChannelMask; 1542 config->config_mask = AUDIO_PORT_CONFIG_SAMPLE_RATE|AUDIO_PORT_CONFIG_CHANNEL_MASK| 1543 AUDIO_PORT_CONFIG_FORMAT; 1544} 1545 1546void AudioFlinger::ThreadBase::systemReady() 1547{ 1548 Mutex::Autolock _l(mLock); 1549 if (mSystemReady) { 1550 return; 1551 } 1552 mSystemReady = true; 1553 1554 for (size_t i = 0; i < mPendingConfigEvents.size(); i++) { 1555 sendConfigEvent_l(mPendingConfigEvents.editItemAt(i)); 1556 } 1557 mPendingConfigEvents.clear(); 1558} 1559 1560 1561// ---------------------------------------------------------------------------- 1562// Playback 1563// ---------------------------------------------------------------------------- 1564 1565AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, 1566 AudioStreamOut* output, 1567 audio_io_handle_t id, 1568 audio_devices_t device, 1569 type_t type, 1570 bool systemReady, 1571 uint32_t bitRate) 1572 : ThreadBase(audioFlinger, id, device, AUDIO_DEVICE_NONE, type, systemReady), 1573 mNormalFrameCount(0), mSinkBuffer(NULL), 1574 mMixerBufferEnabled(AudioFlinger::kEnableExtendedPrecision), 1575 mMixerBuffer(NULL), 1576 mMixerBufferSize(0), 1577 mMixerBufferFormat(AUDIO_FORMAT_INVALID), 1578 mMixerBufferValid(false), 1579 mEffectBufferEnabled(AudioFlinger::kEnableExtendedPrecision), 1580 mEffectBuffer(NULL), 1581 mEffectBufferSize(0), 1582 mEffectBufferFormat(AUDIO_FORMAT_INVALID), 1583 mEffectBufferValid(false), 1584 mSuspended(0), mBytesWritten(0), 1585 mFramesWritten(0), 1586 mActiveTracksGeneration(0), 1587 // mStreamTypes[] initialized in constructor body 1588 mOutput(output), 1589 mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false), 1590 mMixerStatus(MIXER_IDLE), 1591 mMixerStatusIgnoringFastTracks(MIXER_IDLE), 1592 mStandbyDelayNs(AudioFlinger::mStandbyTimeInNsecs), 1593 mBytesRemaining(0), 1594 mCurrentWriteLength(0), 1595 mUseAsyncWrite(false), 1596 mWriteAckSequence(0), 1597 mDrainSequence(0), 1598 mSignalPending(false), 1599 mScreenState(AudioFlinger::mScreenState), 1600 // index 0 is reserved for normal mixer's submix 1601 mFastTrackAvailMask(((1 << FastMixerState::sMaxFastTracks) - 1) & ~1), 1602 mHwSupportsPause(false), mHwPaused(false), mFlushPending(false) 1603{ 1604 snprintf(mThreadName, kThreadNameLength, "AudioOut_%X", id); 1605 mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mThreadName); 1606 1607 // Assumes constructor is called by AudioFlinger with it's mLock held, but 1608 // it would be safer to explicitly pass initial masterVolume/masterMute as 1609 // parameter. 1610 // 1611 // If the HAL we are using has support for master volume or master mute, 1612 // then do not attenuate or mute during mixing (just leave the volume at 1.0 1613 // and the mute set to false). 1614 mMasterVolume = audioFlinger->masterVolume_l(); 1615 mMasterMute = audioFlinger->masterMute_l(); 1616 if (mOutput && mOutput->audioHwDev) { 1617 if (mOutput->audioHwDev->canSetMasterVolume()) { 1618 mMasterVolume = 1.0; 1619 } 1620 1621 if (mOutput->audioHwDev->canSetMasterMute()) { 1622 mMasterMute = false; 1623 } 1624 } 1625 1626 readOutputParameters_l(); 1627 1628 // ++ operator does not compile 1629 for (audio_stream_type_t stream = AUDIO_STREAM_MIN; stream < AUDIO_STREAM_CNT; 1630 stream = (audio_stream_type_t) (stream + 1)) { 1631 mStreamTypes[stream].volume = mAudioFlinger->streamVolume_l(stream); 1632 mStreamTypes[stream].mute = mAudioFlinger->streamMute_l(stream); 1633 } 1634 1635 if (audio_has_proportional_frames(mFormat)) { 1636 mBufferDurationUs = (uint32_t)((mNormalFrameCount * 1000000LL) / mSampleRate); 1637 } else { 1638 bitRate = bitRate != 0 ? bitRate : kOffloadDefaultBitRateBps; 1639 mBufferDurationUs = (uint32_t)((mBufferSize * 8 * 1000000LL) / bitRate); 1640 } 1641} 1642 1643AudioFlinger::PlaybackThread::~PlaybackThread() 1644{ 1645 mAudioFlinger->unregisterWriter(mNBLogWriter); 1646 free(mSinkBuffer); 1647 free(mMixerBuffer); 1648 free(mEffectBuffer); 1649} 1650 1651void AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) 1652{ 1653 dumpInternals(fd, args); 1654 dumpTracks(fd, args); 1655 dumpEffectChains(fd, args); 1656} 1657 1658void AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args __unused) 1659{ 1660 const size_t SIZE = 256; 1661 char buffer[SIZE]; 1662 String8 result; 1663 1664 result.appendFormat(" Stream volumes in dB: "); 1665 for (int i = 0; i < AUDIO_STREAM_CNT; ++i) { 1666 const stream_type_t *st = &mStreamTypes[i]; 1667 if (i > 0) { 1668 result.appendFormat(", "); 1669 } 1670 result.appendFormat("%d:%.2g", i, 20.0 * log10(st->volume)); 1671 if (st->mute) { 1672 result.append("M"); 1673 } 1674 } 1675 result.append("\n"); 1676 write(fd, result.string(), result.length()); 1677 result.clear(); 1678 1679 // These values are "raw"; they will wrap around. See prepareTracks_l() for a better way. 1680 FastTrackUnderruns underruns = getFastTrackUnderruns(0); 1681 dprintf(fd, " Normal mixer raw underrun counters: partial=%u empty=%u\n", 1682 underruns.mBitFields.mPartial, underruns.mBitFields.mEmpty); 1683 1684 size_t numtracks = mTracks.size(); 1685 size_t numactive = mActiveTracks.size(); 1686 dprintf(fd, " %zu Tracks", numtracks); 1687 size_t numactiveseen = 0; 1688 if (numtracks) { 1689 dprintf(fd, " of which %zu are active\n", numactive); 1690 Track::appendDumpHeader(result); 1691 for (size_t i = 0; i < numtracks; ++i) { 1692 sp<Track> track = mTracks[i]; 1693 if (track != 0) { 1694 bool active = mActiveTracks.indexOf(track) >= 0; 1695 if (active) { 1696 numactiveseen++; 1697 } 1698 track->dump(buffer, SIZE, active); 1699 result.append(buffer); 1700 } 1701 } 1702 } else { 1703 result.append("\n"); 1704 } 1705 if (numactiveseen != numactive) { 1706 // some tracks in the active list were not in the tracks list 1707 snprintf(buffer, SIZE, " The following tracks are in the active list but" 1708 " not in the track list\n"); 1709 result.append(buffer); 1710 Track::appendDumpHeader(result); 1711 for (size_t i = 0; i < numactive; ++i) { 1712 sp<Track> track = mActiveTracks[i].promote(); 1713 if (track != 0 && mTracks.indexOf(track) < 0) { 1714 track->dump(buffer, SIZE, true); 1715 result.append(buffer); 1716 } 1717 } 1718 } 1719 1720 write(fd, result.string(), result.size()); 1721} 1722 1723void AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 1724{ 1725 dprintf(fd, "\nOutput thread %p type %d (%s):\n", this, type(), threadTypeToString(type())); 1726 1727 dumpBase(fd, args); 1728 1729 dprintf(fd, " Normal frame count: %zu\n", mNormalFrameCount); 1730 dprintf(fd, " Last write occurred (msecs): %llu\n", 1731 (unsigned long long) ns2ms(systemTime() - mLastWriteTime)); 1732 dprintf(fd, " Total writes: %d\n", mNumWrites); 1733 dprintf(fd, " Delayed writes: %d\n", mNumDelayedWrites); 1734 dprintf(fd, " Blocked in write: %s\n", mInWrite ? "yes" : "no"); 1735 dprintf(fd, " Suspend count: %d\n", mSuspended); 1736 dprintf(fd, " Sink buffer : %p\n", mSinkBuffer); 1737 dprintf(fd, " Mixer buffer: %p\n", mMixerBuffer); 1738 dprintf(fd, " Effect buffer: %p\n", mEffectBuffer); 1739 dprintf(fd, " Fast track availMask=%#x\n", mFastTrackAvailMask); 1740 dprintf(fd, " Standby delay ns=%lld\n", (long long)mStandbyDelayNs); 1741 AudioStreamOut *output = mOutput; 1742 audio_output_flags_t flags = output != NULL ? output->flags : AUDIO_OUTPUT_FLAG_NONE; 1743 String8 flagsAsString = outputFlagsToString(flags); 1744 dprintf(fd, " AudioStreamOut: %p flags %#x (%s)\n", output, flags, flagsAsString.string()); 1745} 1746 1747// Thread virtuals 1748 1749void AudioFlinger::PlaybackThread::onFirstRef() 1750{ 1751 run(mThreadName, ANDROID_PRIORITY_URGENT_AUDIO); 1752} 1753 1754// ThreadBase virtuals 1755void AudioFlinger::PlaybackThread::preExit() 1756{ 1757 ALOGV(" preExit()"); 1758 // FIXME this is using hard-coded strings but in the future, this functionality will be 1759 // converted to use audio HAL extensions required to support tunneling 1760 mOutput->stream->common.set_parameters(&mOutput->stream->common, "exiting=1"); 1761} 1762 1763// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 1764sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 1765 const sp<AudioFlinger::Client>& client, 1766 audio_stream_type_t streamType, 1767 uint32_t sampleRate, 1768 audio_format_t format, 1769 audio_channel_mask_t channelMask, 1770 size_t *pFrameCount, 1771 const sp<IMemory>& sharedBuffer, 1772 audio_session_t sessionId, 1773 IAudioFlinger::track_flags_t *flags, 1774 pid_t tid, 1775 int uid, 1776 status_t *status) 1777{ 1778 size_t frameCount = *pFrameCount; 1779 sp<Track> track; 1780 status_t lStatus; 1781 1782 // client expresses a preference for FAST, but we get the final say 1783 if (*flags & IAudioFlinger::TRACK_FAST) { 1784 if ( 1785 // PCM data 1786 audio_is_linear_pcm(format) && 1787 // TODO: extract as a data library function that checks that a computationally 1788 // expensive downmixer is not required: isFastOutputChannelConversion() 1789 (channelMask == mChannelMask || 1790 mChannelMask != AUDIO_CHANNEL_OUT_STEREO || 1791 (channelMask == AUDIO_CHANNEL_OUT_MONO 1792 /* && mChannelMask == AUDIO_CHANNEL_OUT_STEREO */)) && 1793 // hardware sample rate 1794 (sampleRate == mSampleRate) && 1795 // normal mixer has an associated fast mixer 1796 hasFastMixer() && 1797 // there are sufficient fast track slots available 1798 (mFastTrackAvailMask != 0) 1799 // FIXME test that MixerThread for this fast track has a capable output HAL 1800 // FIXME add a permission test also? 1801 ) { 1802 // static tracks can have any nonzero framecount, streaming tracks check against minimum. 1803 if (sharedBuffer == 0) { 1804 // read the fast track multiplier property the first time it is needed 1805 int ok = pthread_once(&sFastTrackMultiplierOnce, sFastTrackMultiplierInit); 1806 if (ok != 0) { 1807 ALOGE("%s pthread_once failed: %d", __func__, ok); 1808 } 1809 frameCount = max(frameCount, mFrameCount * sFastTrackMultiplier); // incl framecount 0 1810 } 1811 ALOGV("AUDIO_OUTPUT_FLAG_FAST accepted: frameCount=%zu mFrameCount=%zu", 1812 frameCount, mFrameCount); 1813 } else { 1814 ALOGV("AUDIO_OUTPUT_FLAG_FAST denied: sharedBuffer=%p frameCount=%zu " 1815 "mFrameCount=%zu format=%#x mFormat=%#x isLinear=%d channelMask=%#x " 1816 "sampleRate=%u mSampleRate=%u " 1817 "hasFastMixer=%d tid=%d fastTrackAvailMask=%#x", 1818 sharedBuffer.get(), frameCount, mFrameCount, format, mFormat, 1819 audio_is_linear_pcm(format), 1820 channelMask, sampleRate, mSampleRate, hasFastMixer(), tid, mFastTrackAvailMask); 1821 *flags &= ~IAudioFlinger::TRACK_FAST; 1822 } 1823 } 1824 // For normal PCM streaming tracks, update minimum frame count. 1825 // For compatibility with AudioTrack calculation, buffer depth is forced 1826 // to be at least 2 x the normal mixer frame count and cover audio hardware latency. 1827 // This is probably too conservative, but legacy application code may depend on it. 1828 // If you change this calculation, also review the start threshold which is related. 1829 if (!(*flags & IAudioFlinger::TRACK_FAST) 1830 && audio_has_proportional_frames(format) && sharedBuffer == 0) { 1831 // this must match AudioTrack.cpp calculateMinFrameCount(). 1832 // TODO: Move to a common library 1833 uint32_t latencyMs = mOutput->stream->get_latency(mOutput->stream); 1834 uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate); 1835 if (minBufCount < 2) { 1836 minBufCount = 2; 1837 } 1838 // For normal mixing tracks, if speed is > 1.0f (normal), AudioTrack 1839 // or the client should compute and pass in a larger buffer request. 1840 size_t minFrameCount = 1841 minBufCount * sourceFramesNeededWithTimestretch( 1842 sampleRate, mNormalFrameCount, 1843 mSampleRate, AUDIO_TIMESTRETCH_SPEED_NORMAL /*speed*/); 1844 if (frameCount < minFrameCount) { // including frameCount == 0 1845 frameCount = minFrameCount; 1846 } 1847 } 1848 *pFrameCount = frameCount; 1849 1850 switch (mType) { 1851 1852 case DIRECT: 1853 if (audio_is_linear_pcm(format)) { // TODO maybe use audio_has_proportional_frames()? 1854 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1855 ALOGE("createTrack_l() Bad parameter: sampleRate %u format %#x, channelMask 0x%08x " 1856 "for output %p with format %#x", 1857 sampleRate, format, channelMask, mOutput, mFormat); 1858 lStatus = BAD_VALUE; 1859 goto Exit; 1860 } 1861 } 1862 break; 1863 1864 case OFFLOAD: 1865 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1866 ALOGE("createTrack_l() Bad parameter: sampleRate %d format %#x, channelMask 0x%08x \"" 1867 "for output %p with format %#x", 1868 sampleRate, format, channelMask, mOutput, mFormat); 1869 lStatus = BAD_VALUE; 1870 goto Exit; 1871 } 1872 break; 1873 1874 default: 1875 if (!audio_is_linear_pcm(format)) { 1876 ALOGE("createTrack_l() Bad parameter: format %#x \"" 1877 "for output %p with format %#x", 1878 format, mOutput, mFormat); 1879 lStatus = BAD_VALUE; 1880 goto Exit; 1881 } 1882 if (sampleRate > mSampleRate * AUDIO_RESAMPLER_DOWN_RATIO_MAX) { 1883 ALOGE("Sample rate out of range: %u mSampleRate %u", sampleRate, mSampleRate); 1884 lStatus = BAD_VALUE; 1885 goto Exit; 1886 } 1887 break; 1888 1889 } 1890 1891 lStatus = initCheck(); 1892 if (lStatus != NO_ERROR) { 1893 ALOGE("createTrack_l() audio driver not initialized"); 1894 goto Exit; 1895 } 1896 1897 { // scope for mLock 1898 Mutex::Autolock _l(mLock); 1899 1900 // all tracks in same audio session must share the same routing strategy otherwise 1901 // conflicts will happen when tracks are moved from one output to another by audio policy 1902 // manager 1903 uint32_t strategy = AudioSystem::getStrategyForStream(streamType); 1904 for (size_t i = 0; i < mTracks.size(); ++i) { 1905 sp<Track> t = mTracks[i]; 1906 if (t != 0 && t->isExternalTrack()) { 1907 uint32_t actual = AudioSystem::getStrategyForStream(t->streamType()); 1908 if (sessionId == t->sessionId() && strategy != actual) { 1909 ALOGE("createTrack_l() mismatched strategy; expected %u but found %u", 1910 strategy, actual); 1911 lStatus = BAD_VALUE; 1912 goto Exit; 1913 } 1914 } 1915 } 1916 1917 track = new Track(this, client, streamType, sampleRate, format, 1918 channelMask, frameCount, NULL, sharedBuffer, 1919 sessionId, uid, *flags, TrackBase::TYPE_DEFAULT); 1920 1921 lStatus = track != 0 ? track->initCheck() : (status_t) NO_MEMORY; 1922 if (lStatus != NO_ERROR) { 1923 ALOGE("createTrack_l() initCheck failed %d; no control block?", lStatus); 1924 // track must be cleared from the caller as the caller has the AF lock 1925 goto Exit; 1926 } 1927 mTracks.add(track); 1928 1929 sp<EffectChain> chain = getEffectChain_l(sessionId); 1930 if (chain != 0) { 1931 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1932 track->setMainBuffer(chain->inBuffer()); 1933 chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType())); 1934 chain->incTrackCnt(); 1935 } 1936 1937 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 1938 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 1939 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 1940 // so ask activity manager to do this on our behalf 1941 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 1942 } 1943 } 1944 1945 lStatus = NO_ERROR; 1946 1947Exit: 1948 *status = lStatus; 1949 return track; 1950} 1951 1952uint32_t AudioFlinger::PlaybackThread::correctLatency_l(uint32_t latency) const 1953{ 1954 return latency; 1955} 1956 1957uint32_t AudioFlinger::PlaybackThread::latency() const 1958{ 1959 Mutex::Autolock _l(mLock); 1960 return latency_l(); 1961} 1962uint32_t AudioFlinger::PlaybackThread::latency_l() const 1963{ 1964 if (initCheck() == NO_ERROR) { 1965 return correctLatency_l(mOutput->stream->get_latency(mOutput->stream)); 1966 } else { 1967 return 0; 1968 } 1969} 1970 1971void AudioFlinger::PlaybackThread::setMasterVolume(float value) 1972{ 1973 Mutex::Autolock _l(mLock); 1974 // Don't apply master volume in SW if our HAL can do it for us. 1975 if (mOutput && mOutput->audioHwDev && 1976 mOutput->audioHwDev->canSetMasterVolume()) { 1977 mMasterVolume = 1.0; 1978 } else { 1979 mMasterVolume = value; 1980 } 1981} 1982 1983void AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1984{ 1985 Mutex::Autolock _l(mLock); 1986 // Don't apply master mute in SW if our HAL can do it for us. 1987 if (mOutput && mOutput->audioHwDev && 1988 mOutput->audioHwDev->canSetMasterMute()) { 1989 mMasterMute = false; 1990 } else { 1991 mMasterMute = muted; 1992 } 1993} 1994 1995void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) 1996{ 1997 Mutex::Autolock _l(mLock); 1998 mStreamTypes[stream].volume = value; 1999 broadcast_l(); 2000} 2001 2002void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) 2003{ 2004 Mutex::Autolock _l(mLock); 2005 mStreamTypes[stream].mute = muted; 2006 broadcast_l(); 2007} 2008 2009float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const 2010{ 2011 Mutex::Autolock _l(mLock); 2012 return mStreamTypes[stream].volume; 2013} 2014 2015// addTrack_l() must be called with ThreadBase::mLock held 2016status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 2017{ 2018 status_t status = ALREADY_EXISTS; 2019 2020 if (mActiveTracks.indexOf(track) < 0) { 2021 // the track is newly added, make sure it fills up all its 2022 // buffers before playing. This is to ensure the client will 2023 // effectively get the latency it requested. 2024 if (track->isExternalTrack()) { 2025 TrackBase::track_state state = track->mState; 2026 mLock.unlock(); 2027 status = AudioSystem::startOutput(mId, track->streamType(), 2028 track->sessionId()); 2029 mLock.lock(); 2030 // abort track was stopped/paused while we released the lock 2031 if (state != track->mState) { 2032 if (status == NO_ERROR) { 2033 mLock.unlock(); 2034 AudioSystem::stopOutput(mId, track->streamType(), 2035 track->sessionId()); 2036 mLock.lock(); 2037 } 2038 return INVALID_OPERATION; 2039 } 2040 // abort if start is rejected by audio policy manager 2041 if (status != NO_ERROR) { 2042 return PERMISSION_DENIED; 2043 } 2044#ifdef ADD_BATTERY_DATA 2045 // to track the speaker usage 2046 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 2047#endif 2048 } 2049 2050 // set retry count for buffer fill 2051 if (track->isOffloaded()) { 2052 track->mRetryCount = kMaxTrackStartupRetriesOffload; 2053 } else { 2054 track->mRetryCount = kMaxTrackStartupRetries; 2055 } 2056 2057 track->mFillingUpStatus = track->sharedBuffer() != 0 ? Track::FS_FILLED : Track::FS_FILLING; 2058 track->mResetDone = false; 2059 track->mPresentationCompleteFrames = 0; 2060 mActiveTracks.add(track); 2061 mWakeLockUids.add(track->uid()); 2062 mActiveTracksGeneration++; 2063 mLatestActiveTrack = track; 2064 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 2065 if (chain != 0) { 2066 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), 2067 track->sessionId()); 2068 chain->incActiveTrackCnt(); 2069 } 2070 2071 status = NO_ERROR; 2072 } 2073 2074 onAddNewTrack_l(); 2075 return status; 2076} 2077 2078bool AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 2079{ 2080 track->terminate(); 2081 // active tracks are removed by threadLoop() 2082 bool trackActive = (mActiveTracks.indexOf(track) >= 0); 2083 track->mState = TrackBase::STOPPED; 2084 if (!trackActive) { 2085 removeTrack_l(track); 2086 } else if (track->isFastTrack() || track->isOffloaded() || track->isDirect()) { 2087 track->mState = TrackBase::STOPPING_1; 2088 } 2089 2090 return trackActive; 2091} 2092 2093void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 2094{ 2095 track->triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 2096 mTracks.remove(track); 2097 deleteTrackName_l(track->name()); 2098 // redundant as track is about to be destroyed, for dumpsys only 2099 track->mName = -1; 2100 if (track->isFastTrack()) { 2101 int index = track->mFastIndex; 2102 ALOG_ASSERT(0 < index && index < (int)FastMixerState::sMaxFastTracks); 2103 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << index))); 2104 mFastTrackAvailMask |= 1 << index; 2105 // redundant as track is about to be destroyed, for dumpsys only 2106 track->mFastIndex = -1; 2107 } 2108 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 2109 if (chain != 0) { 2110 chain->decTrackCnt(); 2111 } 2112} 2113 2114void AudioFlinger::PlaybackThread::broadcast_l() 2115{ 2116 // Thread could be blocked waiting for async 2117 // so signal it to handle state changes immediately 2118 // If threadLoop is currently unlocked a signal of mWaitWorkCV will 2119 // be lost so we also flag to prevent it blocking on mWaitWorkCV 2120 mSignalPending = true; 2121 mWaitWorkCV.broadcast(); 2122} 2123 2124String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 2125{ 2126 Mutex::Autolock _l(mLock); 2127 if (initCheck() != NO_ERROR) { 2128 return String8(); 2129 } 2130 2131 char *s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 2132 const String8 out_s8(s); 2133 free(s); 2134 return out_s8; 2135} 2136 2137void AudioFlinger::PlaybackThread::ioConfigChanged(audio_io_config_event event, pid_t pid) { 2138 sp<AudioIoDescriptor> desc = new AudioIoDescriptor(); 2139 ALOGV("PlaybackThread::ioConfigChanged, thread %p, event %d", this, event); 2140 2141 desc->mIoHandle = mId; 2142 2143 switch (event) { 2144 case AUDIO_OUTPUT_OPENED: 2145 case AUDIO_OUTPUT_CONFIG_CHANGED: 2146 desc->mPatch = mPatch; 2147 desc->mChannelMask = mChannelMask; 2148 desc->mSamplingRate = mSampleRate; 2149 desc->mFormat = mFormat; 2150 desc->mFrameCount = mNormalFrameCount; // FIXME see 2151 // AudioFlinger::frameCount(audio_io_handle_t) 2152 desc->mFrameCountHAL = mFrameCount; 2153 desc->mLatency = latency_l(); 2154 break; 2155 2156 case AUDIO_OUTPUT_CLOSED: 2157 default: 2158 break; 2159 } 2160 mAudioFlinger->ioConfigChanged(event, desc, pid); 2161} 2162 2163void AudioFlinger::PlaybackThread::writeCallback() 2164{ 2165 ALOG_ASSERT(mCallbackThread != 0); 2166 mCallbackThread->resetWriteBlocked(); 2167} 2168 2169void AudioFlinger::PlaybackThread::drainCallback() 2170{ 2171 ALOG_ASSERT(mCallbackThread != 0); 2172 mCallbackThread->resetDraining(); 2173} 2174 2175void AudioFlinger::PlaybackThread::resetWriteBlocked(uint32_t sequence) 2176{ 2177 Mutex::Autolock _l(mLock); 2178 // reject out of sequence requests 2179 if ((mWriteAckSequence & 1) && (sequence == mWriteAckSequence)) { 2180 mWriteAckSequence &= ~1; 2181 mWaitWorkCV.signal(); 2182 } 2183} 2184 2185void AudioFlinger::PlaybackThread::resetDraining(uint32_t sequence) 2186{ 2187 Mutex::Autolock _l(mLock); 2188 // reject out of sequence requests 2189 if ((mDrainSequence & 1) && (sequence == mDrainSequence)) { 2190 mDrainSequence &= ~1; 2191 mWaitWorkCV.signal(); 2192 } 2193} 2194 2195// static 2196int AudioFlinger::PlaybackThread::asyncCallback(stream_callback_event_t event, 2197 void *param __unused, 2198 void *cookie) 2199{ 2200 AudioFlinger::PlaybackThread *me = (AudioFlinger::PlaybackThread *)cookie; 2201 ALOGV("asyncCallback() event %d", event); 2202 switch (event) { 2203 case STREAM_CBK_EVENT_WRITE_READY: 2204 me->writeCallback(); 2205 break; 2206 case STREAM_CBK_EVENT_DRAIN_READY: 2207 me->drainCallback(); 2208 break; 2209 default: 2210 ALOGW("asyncCallback() unknown event %d", event); 2211 break; 2212 } 2213 return 0; 2214} 2215 2216void AudioFlinger::PlaybackThread::readOutputParameters_l() 2217{ 2218 // unfortunately we have no way of recovering from errors here, hence the LOG_ALWAYS_FATAL 2219 mSampleRate = mOutput->getSampleRate(); 2220 mChannelMask = mOutput->getChannelMask(); 2221 if (!audio_is_output_channel(mChannelMask)) { 2222 LOG_ALWAYS_FATAL("HAL channel mask %#x not valid for output", mChannelMask); 2223 } 2224 if ((mType == MIXER || mType == DUPLICATING) 2225 && !isValidPcmSinkChannelMask(mChannelMask)) { 2226 LOG_ALWAYS_FATAL("HAL channel mask %#x not supported for mixed output", 2227 mChannelMask); 2228 } 2229 mChannelCount = audio_channel_count_from_out_mask(mChannelMask); 2230 2231 // Get actual HAL format. 2232 mHALFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 2233 // Get format from the shim, which will be different than the HAL format 2234 // if playing compressed audio over HDMI passthrough. 2235 mFormat = mOutput->getFormat(); 2236 if (!audio_is_valid_format(mFormat)) { 2237 LOG_ALWAYS_FATAL("HAL format %#x not valid for output", mFormat); 2238 } 2239 if ((mType == MIXER || mType == DUPLICATING) 2240 && !isValidPcmSinkFormat(mFormat)) { 2241 LOG_FATAL("HAL format %#x not supported for mixed output", 2242 mFormat); 2243 } 2244 mFrameSize = mOutput->getFrameSize(); 2245 mBufferSize = mOutput->stream->common.get_buffer_size(&mOutput->stream->common); 2246 mFrameCount = mBufferSize / mFrameSize; 2247 if (mFrameCount & 15) { 2248 ALOGW("HAL output buffer size is %zu frames but AudioMixer requires multiples of 16 frames", 2249 mFrameCount); 2250 } 2251 2252 if ((mOutput->flags & AUDIO_OUTPUT_FLAG_NON_BLOCKING) && 2253 (mOutput->stream->set_callback != NULL)) { 2254 if (mOutput->stream->set_callback(mOutput->stream, 2255 AudioFlinger::PlaybackThread::asyncCallback, this) == 0) { 2256 mUseAsyncWrite = true; 2257 mCallbackThread = new AudioFlinger::AsyncCallbackThread(this); 2258 } 2259 } 2260 2261 mHwSupportsPause = false; 2262 if (mOutput->flags & AUDIO_OUTPUT_FLAG_DIRECT) { 2263 if (mOutput->stream->pause != NULL) { 2264 if (mOutput->stream->resume != NULL) { 2265 mHwSupportsPause = true; 2266 } else { 2267 ALOGW("direct output implements pause but not resume"); 2268 } 2269 } else if (mOutput->stream->resume != NULL) { 2270 ALOGW("direct output implements resume but not pause"); 2271 } 2272 } 2273 if (!mHwSupportsPause && mOutput->flags & AUDIO_OUTPUT_FLAG_HW_AV_SYNC) { 2274 LOG_ALWAYS_FATAL("HW_AV_SYNC requested but HAL does not implement pause and resume"); 2275 } 2276 2277 if (mType == DUPLICATING && mMixerBufferEnabled && mEffectBufferEnabled) { 2278 // For best precision, we use float instead of the associated output 2279 // device format (typically PCM 16 bit). 2280 2281 mFormat = AUDIO_FORMAT_PCM_FLOAT; 2282 mFrameSize = mChannelCount * audio_bytes_per_sample(mFormat); 2283 mBufferSize = mFrameSize * mFrameCount; 2284 2285 // TODO: We currently use the associated output device channel mask and sample rate. 2286 // (1) Perhaps use the ORed channel mask of all downstream MixerThreads 2287 // (if a valid mask) to avoid premature downmix. 2288 // (2) Perhaps use the maximum sample rate of all downstream MixerThreads 2289 // instead of the output device sample rate to avoid loss of high frequency information. 2290 // This may need to be updated as MixerThread/OutputTracks are added and not here. 2291 } 2292 2293 // Calculate size of normal sink buffer relative to the HAL output buffer size 2294 double multiplier = 1.0; 2295 if (mType == MIXER && (kUseFastMixer == FastMixer_Static || 2296 kUseFastMixer == FastMixer_Dynamic)) { 2297 size_t minNormalFrameCount = (kMinNormalSinkBufferSizeMs * mSampleRate) / 1000; 2298 size_t maxNormalFrameCount = (kMaxNormalSinkBufferSizeMs * mSampleRate) / 1000; 2299 // round up minimum and round down maximum to nearest 16 frames to satisfy AudioMixer 2300 minNormalFrameCount = (minNormalFrameCount + 15) & ~15; 2301 maxNormalFrameCount = maxNormalFrameCount & ~15; 2302 if (maxNormalFrameCount < minNormalFrameCount) { 2303 maxNormalFrameCount = minNormalFrameCount; 2304 } 2305 multiplier = (double) minNormalFrameCount / (double) mFrameCount; 2306 if (multiplier <= 1.0) { 2307 multiplier = 1.0; 2308 } else if (multiplier <= 2.0) { 2309 if (2 * mFrameCount <= maxNormalFrameCount) { 2310 multiplier = 2.0; 2311 } else { 2312 multiplier = (double) maxNormalFrameCount / (double) mFrameCount; 2313 } 2314 } else { 2315 // prefer an even multiplier, for compatibility with doubling of fast tracks due to HAL 2316 // SRC (it would be unusual for the normal sink buffer size to not be a multiple of fast 2317 // track, but we sometimes have to do this to satisfy the maximum frame count 2318 // constraint) 2319 // FIXME this rounding up should not be done if no HAL SRC 2320 uint32_t truncMult = (uint32_t) multiplier; 2321 if ((truncMult & 1)) { 2322 if ((truncMult + 1) * mFrameCount <= maxNormalFrameCount) { 2323 ++truncMult; 2324 } 2325 } 2326 multiplier = (double) truncMult; 2327 } 2328 } 2329 mNormalFrameCount = multiplier * mFrameCount; 2330 // round up to nearest 16 frames to satisfy AudioMixer 2331 if (mType == MIXER || mType == DUPLICATING) { 2332 mNormalFrameCount = (mNormalFrameCount + 15) & ~15; 2333 } 2334 ALOGI("HAL output buffer size %zu frames, normal sink buffer size %zu frames", mFrameCount, 2335 mNormalFrameCount); 2336 2337 // Check if we want to throttle the processing to no more than 2x normal rate 2338 mThreadThrottle = property_get_bool("af.thread.throttle", true /* default_value */); 2339 mThreadThrottleTimeMs = 0; 2340 mThreadThrottleEndMs = 0; 2341 mHalfBufferMs = mNormalFrameCount * 1000 / (2 * mSampleRate); 2342 2343 // mSinkBuffer is the sink buffer. Size is always multiple-of-16 frames. 2344 // Originally this was int16_t[] array, need to remove legacy implications. 2345 free(mSinkBuffer); 2346 mSinkBuffer = NULL; 2347 // For sink buffer size, we use the frame size from the downstream sink to avoid problems 2348 // with non PCM formats for compressed music, e.g. AAC, and Offload threads. 2349 const size_t sinkBufferSize = mNormalFrameCount * mFrameSize; 2350 (void)posix_memalign(&mSinkBuffer, 32, sinkBufferSize); 2351 2352 // We resize the mMixerBuffer according to the requirements of the sink buffer which 2353 // drives the output. 2354 free(mMixerBuffer); 2355 mMixerBuffer = NULL; 2356 if (mMixerBufferEnabled) { 2357 mMixerBufferFormat = AUDIO_FORMAT_PCM_FLOAT; // also valid: AUDIO_FORMAT_PCM_16_BIT. 2358 mMixerBufferSize = mNormalFrameCount * mChannelCount 2359 * audio_bytes_per_sample(mMixerBufferFormat); 2360 (void)posix_memalign(&mMixerBuffer, 32, mMixerBufferSize); 2361 } 2362 free(mEffectBuffer); 2363 mEffectBuffer = NULL; 2364 if (mEffectBufferEnabled) { 2365 mEffectBufferFormat = AUDIO_FORMAT_PCM_16_BIT; // Note: Effects support 16b only 2366 mEffectBufferSize = mNormalFrameCount * mChannelCount 2367 * audio_bytes_per_sample(mEffectBufferFormat); 2368 (void)posix_memalign(&mEffectBuffer, 32, mEffectBufferSize); 2369 } 2370 2371 // force reconfiguration of effect chains and engines to take new buffer size and audio 2372 // parameters into account 2373 // Note that mLock is not held when readOutputParameters_l() is called from the constructor 2374 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 2375 // matter. 2376 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 2377 Vector< sp<EffectChain> > effectChains = mEffectChains; 2378 for (size_t i = 0; i < effectChains.size(); i ++) { 2379 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 2380 } 2381} 2382 2383 2384status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames) 2385{ 2386 if (halFrames == NULL || dspFrames == NULL) { 2387 return BAD_VALUE; 2388 } 2389 Mutex::Autolock _l(mLock); 2390 if (initCheck() != NO_ERROR) { 2391 return INVALID_OPERATION; 2392 } 2393 int64_t framesWritten = mBytesWritten / mFrameSize; 2394 *halFrames = framesWritten; 2395 2396 if (isSuspended()) { 2397 // return an estimation of rendered frames when the output is suspended 2398 size_t latencyFrames = (latency_l() * mSampleRate) / 1000; 2399 *dspFrames = (uint32_t) 2400 (framesWritten >= (int64_t)latencyFrames ? framesWritten - latencyFrames : 0); 2401 return NO_ERROR; 2402 } else { 2403 status_t status; 2404 uint32_t frames; 2405 status = mOutput->getRenderPosition(&frames); 2406 *dspFrames = (size_t)frames; 2407 return status; 2408 } 2409} 2410 2411uint32_t AudioFlinger::PlaybackThread::hasAudioSession(audio_session_t sessionId) const 2412{ 2413 Mutex::Autolock _l(mLock); 2414 uint32_t result = 0; 2415 if (getEffectChain_l(sessionId) != 0) { 2416 result = EFFECT_SESSION; 2417 } 2418 2419 for (size_t i = 0; i < mTracks.size(); ++i) { 2420 sp<Track> track = mTracks[i]; 2421 if (sessionId == track->sessionId() && !track->isInvalid()) { 2422 result |= TRACK_SESSION; 2423 break; 2424 } 2425 } 2426 2427 return result; 2428} 2429 2430uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(audio_session_t sessionId) 2431{ 2432 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 2433 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 2434 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 2435 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 2436 } 2437 for (size_t i = 0; i < mTracks.size(); i++) { 2438 sp<Track> track = mTracks[i]; 2439 if (sessionId == track->sessionId() && !track->isInvalid()) { 2440 return AudioSystem::getStrategyForStream(track->streamType()); 2441 } 2442 } 2443 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 2444} 2445 2446 2447AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 2448{ 2449 Mutex::Autolock _l(mLock); 2450 return mOutput; 2451} 2452 2453AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 2454{ 2455 Mutex::Autolock _l(mLock); 2456 AudioStreamOut *output = mOutput; 2457 mOutput = NULL; 2458 // FIXME FastMixer might also have a raw ptr to mOutputSink; 2459 // must push a NULL and wait for ack 2460 mOutputSink.clear(); 2461 mPipeSink.clear(); 2462 mNormalSink.clear(); 2463 return output; 2464} 2465 2466// this method must always be called either with ThreadBase mLock held or inside the thread loop 2467audio_stream_t* AudioFlinger::PlaybackThread::stream() const 2468{ 2469 if (mOutput == NULL) { 2470 return NULL; 2471 } 2472 return &mOutput->stream->common; 2473} 2474 2475uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() const 2476{ 2477 return (uint32_t)((uint32_t)((mNormalFrameCount * 1000) / mSampleRate) * 1000); 2478} 2479 2480status_t AudioFlinger::PlaybackThread::setSyncEvent(const sp<SyncEvent>& event) 2481{ 2482 if (!isValidSyncEvent(event)) { 2483 return BAD_VALUE; 2484 } 2485 2486 Mutex::Autolock _l(mLock); 2487 2488 for (size_t i = 0; i < mTracks.size(); ++i) { 2489 sp<Track> track = mTracks[i]; 2490 if (event->triggerSession() == track->sessionId()) { 2491 (void) track->setSyncEvent(event); 2492 return NO_ERROR; 2493 } 2494 } 2495 2496 return NAME_NOT_FOUND; 2497} 2498 2499bool AudioFlinger::PlaybackThread::isValidSyncEvent(const sp<SyncEvent>& event) const 2500{ 2501 return event->type() == AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE; 2502} 2503 2504void AudioFlinger::PlaybackThread::threadLoop_removeTracks( 2505 const Vector< sp<Track> >& tracksToRemove) 2506{ 2507 size_t count = tracksToRemove.size(); 2508 if (count > 0) { 2509 for (size_t i = 0 ; i < count ; i++) { 2510 const sp<Track>& track = tracksToRemove.itemAt(i); 2511 if (track->isExternalTrack()) { 2512 AudioSystem::stopOutput(mId, track->streamType(), 2513 track->sessionId()); 2514#ifdef ADD_BATTERY_DATA 2515 // to track the speaker usage 2516 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 2517#endif 2518 if (track->isTerminated()) { 2519 AudioSystem::releaseOutput(mId, track->streamType(), 2520 track->sessionId()); 2521 } 2522 } 2523 } 2524 } 2525} 2526 2527void AudioFlinger::PlaybackThread::checkSilentMode_l() 2528{ 2529 if (!mMasterMute) { 2530 char value[PROPERTY_VALUE_MAX]; 2531 if (mOutDevice == AUDIO_DEVICE_OUT_REMOTE_SUBMIX) { 2532 ALOGD("ro.audio.silent will be ignored for threads on AUDIO_DEVICE_OUT_REMOTE_SUBMIX"); 2533 return; 2534 } 2535 if (property_get("ro.audio.silent", value, "0") > 0) { 2536 char *endptr; 2537 unsigned long ul = strtoul(value, &endptr, 0); 2538 if (*endptr == '\0' && ul != 0) { 2539 ALOGD("Silence is golden"); 2540 // The setprop command will not allow a property to be changed after 2541 // the first time it is set, so we don't have to worry about un-muting. 2542 setMasterMute_l(true); 2543 } 2544 } 2545 } 2546} 2547 2548// shared by MIXER and DIRECT, overridden by DUPLICATING 2549ssize_t AudioFlinger::PlaybackThread::threadLoop_write() 2550{ 2551 // FIXME rewrite to reduce number of system calls 2552 mLastWriteTime = systemTime(); 2553 mInWrite = true; 2554 ssize_t bytesWritten; 2555 const size_t offset = mCurrentWriteLength - mBytesRemaining; 2556 2557 // If an NBAIO sink is present, use it to write the normal mixer's submix 2558 if (mNormalSink != 0) { 2559 2560 const size_t count = mBytesRemaining / mFrameSize; 2561 2562 ATRACE_BEGIN("write"); 2563 // update the setpoint when AudioFlinger::mScreenState changes 2564 uint32_t screenState = AudioFlinger::mScreenState; 2565 if (screenState != mScreenState) { 2566 mScreenState = screenState; 2567 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 2568 if (pipe != NULL) { 2569 pipe->setAvgFrames((mScreenState & 1) ? 2570 (pipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2571 } 2572 } 2573 ssize_t framesWritten = mNormalSink->write((char *)mSinkBuffer + offset, count); 2574 ATRACE_END(); 2575 if (framesWritten > 0) { 2576 bytesWritten = framesWritten * mFrameSize; 2577 } else { 2578 bytesWritten = framesWritten; 2579 } 2580 // otherwise use the HAL / AudioStreamOut directly 2581 } else { 2582 // Direct output and offload threads 2583 2584 if (mUseAsyncWrite) { 2585 ALOGW_IF(mWriteAckSequence & 1, "threadLoop_write(): out of sequence write request"); 2586 mWriteAckSequence += 2; 2587 mWriteAckSequence |= 1; 2588 ALOG_ASSERT(mCallbackThread != 0); 2589 mCallbackThread->setWriteBlocked(mWriteAckSequence); 2590 } 2591 // FIXME We should have an implementation of timestamps for direct output threads. 2592 // They are used e.g for multichannel PCM playback over HDMI. 2593 bytesWritten = mOutput->write((char *)mSinkBuffer + offset, mBytesRemaining); 2594 2595 if (mUseAsyncWrite && 2596 ((bytesWritten < 0) || (bytesWritten == (ssize_t)mBytesRemaining))) { 2597 // do not wait for async callback in case of error of full write 2598 mWriteAckSequence &= ~1; 2599 ALOG_ASSERT(mCallbackThread != 0); 2600 mCallbackThread->setWriteBlocked(mWriteAckSequence); 2601 } 2602 } 2603 2604 mNumWrites++; 2605 mInWrite = false; 2606 mStandby = false; 2607 return bytesWritten; 2608} 2609 2610void AudioFlinger::PlaybackThread::threadLoop_drain() 2611{ 2612 if (mOutput->stream->drain) { 2613 ALOGV("draining %s", (mMixerStatus == MIXER_DRAIN_TRACK) ? "early" : "full"); 2614 if (mUseAsyncWrite) { 2615 ALOGW_IF(mDrainSequence & 1, "threadLoop_drain(): out of sequence drain request"); 2616 mDrainSequence |= 1; 2617 ALOG_ASSERT(mCallbackThread != 0); 2618 mCallbackThread->setDraining(mDrainSequence); 2619 } 2620 mOutput->stream->drain(mOutput->stream, 2621 (mMixerStatus == MIXER_DRAIN_TRACK) ? AUDIO_DRAIN_EARLY_NOTIFY 2622 : AUDIO_DRAIN_ALL); 2623 } 2624} 2625 2626void AudioFlinger::PlaybackThread::threadLoop_exit() 2627{ 2628 { 2629 Mutex::Autolock _l(mLock); 2630 for (size_t i = 0; i < mTracks.size(); i++) { 2631 sp<Track> track = mTracks[i]; 2632 track->invalidate(); 2633 } 2634 } 2635} 2636 2637/* 2638The derived values that are cached: 2639 - mSinkBufferSize from frame count * frame size 2640 - mActiveSleepTimeUs from activeSleepTimeUs() 2641 - mIdleSleepTimeUs from idleSleepTimeUs() 2642 - mStandbyDelayNs from mActiveSleepTimeUs (DIRECT only) or forced to at least 2643 kDefaultStandbyTimeInNsecs when connected to an A2DP device. 2644 - maxPeriod from frame count and sample rate (MIXER only) 2645 2646The parameters that affect these derived values are: 2647 - frame count 2648 - frame size 2649 - sample rate 2650 - device type: A2DP or not 2651 - device latency 2652 - format: PCM or not 2653 - active sleep time 2654 - idle sleep time 2655*/ 2656 2657void AudioFlinger::PlaybackThread::cacheParameters_l() 2658{ 2659 mSinkBufferSize = mNormalFrameCount * mFrameSize; 2660 mActiveSleepTimeUs = activeSleepTimeUs(); 2661 mIdleSleepTimeUs = idleSleepTimeUs(); 2662 2663 // make sure standby delay is not too short when connected to an A2DP sink to avoid 2664 // truncating audio when going to standby. 2665 mStandbyDelayNs = AudioFlinger::mStandbyTimeInNsecs; 2666 if ((mOutDevice & AUDIO_DEVICE_OUT_ALL_A2DP) != 0) { 2667 if (mStandbyDelayNs < kDefaultStandbyTimeInNsecs) { 2668 mStandbyDelayNs = kDefaultStandbyTimeInNsecs; 2669 } 2670 } 2671} 2672 2673void AudioFlinger::PlaybackThread::invalidateTracks_l(audio_stream_type_t streamType) 2674{ 2675 ALOGV("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %zu", 2676 this, streamType, mTracks.size()); 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 2687void AudioFlinger::PlaybackThread::invalidateTracks(audio_stream_type_t streamType) 2688{ 2689 Mutex::Autolock _l(mLock); 2690 invalidateTracks_l(streamType); 2691} 2692 2693status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 2694{ 2695 audio_session_t session = chain->sessionId(); 2696 int16_t* buffer = reinterpret_cast<int16_t*>(mEffectBufferEnabled 2697 ? mEffectBuffer : mSinkBuffer); 2698 bool ownsBuffer = false; 2699 2700 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 2701 if (session > AUDIO_SESSION_OUTPUT_MIX) { 2702 // Only one effect chain can be present in direct output thread and it uses 2703 // the sink buffer as input 2704 if (mType != DIRECT) { 2705 size_t numSamples = mNormalFrameCount * mChannelCount; 2706 buffer = new int16_t[numSamples]; 2707 memset(buffer, 0, numSamples * sizeof(int16_t)); 2708 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 2709 ownsBuffer = true; 2710 } 2711 2712 // Attach all tracks with same session ID to this chain. 2713 for (size_t i = 0; i < mTracks.size(); ++i) { 2714 sp<Track> track = mTracks[i]; 2715 if (session == track->sessionId()) { 2716 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), 2717 buffer); 2718 track->setMainBuffer(buffer); 2719 chain->incTrackCnt(); 2720 } 2721 } 2722 2723 // indicate all active tracks in the chain 2724 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 2725 sp<Track> track = mActiveTracks[i].promote(); 2726 if (track == 0) { 2727 continue; 2728 } 2729 if (session == track->sessionId()) { 2730 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 2731 chain->incActiveTrackCnt(); 2732 } 2733 } 2734 } 2735 chain->setThread(this); 2736 chain->setInBuffer(buffer, ownsBuffer); 2737 chain->setOutBuffer(reinterpret_cast<int16_t*>(mEffectBufferEnabled 2738 ? mEffectBuffer : mSinkBuffer)); 2739 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 2740 // chains list in order to be processed last as it contains output stage effects. 2741 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 2742 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 2743 // after track specific effects and before output stage. 2744 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 2745 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX. 2746 // Effect chain for other sessions are inserted at beginning of effect 2747 // chains list to be processed before output mix effects. Relative order between other 2748 // sessions is not important. 2749 static_assert(AUDIO_SESSION_OUTPUT_MIX == 0 && 2750 AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX, 2751 "audio_session_t constants misdefined"); 2752 size_t size = mEffectChains.size(); 2753 size_t i = 0; 2754 for (i = 0; i < size; i++) { 2755 if (mEffectChains[i]->sessionId() < session) { 2756 break; 2757 } 2758 } 2759 mEffectChains.insertAt(chain, i); 2760 checkSuspendOnAddEffectChain_l(chain); 2761 2762 return NO_ERROR; 2763} 2764 2765size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 2766{ 2767 audio_session_t session = chain->sessionId(); 2768 2769 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 2770 2771 for (size_t i = 0; i < mEffectChains.size(); i++) { 2772 if (chain == mEffectChains[i]) { 2773 mEffectChains.removeAt(i); 2774 // detach all active tracks from the chain 2775 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 2776 sp<Track> track = mActiveTracks[i].promote(); 2777 if (track == 0) { 2778 continue; 2779 } 2780 if (session == track->sessionId()) { 2781 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 2782 chain.get(), session); 2783 chain->decActiveTrackCnt(); 2784 } 2785 } 2786 2787 // detach all tracks with same session ID from this chain 2788 for (size_t i = 0; i < mTracks.size(); ++i) { 2789 sp<Track> track = mTracks[i]; 2790 if (session == track->sessionId()) { 2791 track->setMainBuffer(reinterpret_cast<int16_t*>(mSinkBuffer)); 2792 chain->decTrackCnt(); 2793 } 2794 } 2795 break; 2796 } 2797 } 2798 return mEffectChains.size(); 2799} 2800 2801status_t AudioFlinger::PlaybackThread::attachAuxEffect( 2802 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2803{ 2804 Mutex::Autolock _l(mLock); 2805 return attachAuxEffect_l(track, EffectId); 2806} 2807 2808status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 2809 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2810{ 2811 status_t status = NO_ERROR; 2812 2813 if (EffectId == 0) { 2814 track->setAuxBuffer(0, NULL); 2815 } else { 2816 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 2817 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 2818 if (effect != 0) { 2819 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 2820 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 2821 } else { 2822 status = INVALID_OPERATION; 2823 } 2824 } else { 2825 status = BAD_VALUE; 2826 } 2827 } 2828 return status; 2829} 2830 2831void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 2832{ 2833 for (size_t i = 0; i < mTracks.size(); ++i) { 2834 sp<Track> track = mTracks[i]; 2835 if (track->auxEffectId() == effectId) { 2836 attachAuxEffect_l(track, 0); 2837 } 2838 } 2839} 2840 2841bool AudioFlinger::PlaybackThread::threadLoop() 2842{ 2843 Vector< sp<Track> > tracksToRemove; 2844 2845 mStandbyTimeNs = systemTime(); 2846 2847 // MIXER 2848 nsecs_t lastWarning = 0; 2849 2850 // DUPLICATING 2851 // FIXME could this be made local to while loop? 2852 writeFrames = 0; 2853 2854 int lastGeneration = 0; 2855 2856 cacheParameters_l(); 2857 mSleepTimeUs = mIdleSleepTimeUs; 2858 2859 if (mType == MIXER) { 2860 sleepTimeShift = 0; 2861 } 2862 2863 CpuStats cpuStats; 2864 const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid())); 2865 2866 acquireWakeLock(); 2867 2868 // mNBLogWriter->log can only be called while thread mutex mLock is held. 2869 // So if you need to log when mutex is unlocked, set logString to a non-NULL string, 2870 // and then that string will be logged at the next convenient opportunity. 2871 const char *logString = NULL; 2872 2873 checkSilentMode_l(); 2874 2875 while (!exitPending()) 2876 { 2877 cpuStats.sample(myName); 2878 2879 Vector< sp<EffectChain> > effectChains; 2880 2881 { // scope for mLock 2882 2883 Mutex::Autolock _l(mLock); 2884 2885 processConfigEvents_l(); 2886 2887 if (logString != NULL) { 2888 mNBLogWriter->logTimestamp(); 2889 mNBLogWriter->log(logString); 2890 logString = NULL; 2891 } 2892 2893 // Gather the framesReleased counters for all active tracks, 2894 // and associate with the sink frames written out. We need 2895 // this to convert the sink timestamp to the track timestamp. 2896 if (mNormalSink != 0) { 2897 // Note: The DuplicatingThread may not have a mNormalSink. 2898 // We always fetch the timestamp here because often the downstream 2899 // sink will block whie writing. 2900 ExtendedTimestamp timestamp; // use private copy to fetch 2901 (void) mNormalSink->getTimestamp(timestamp); 2902 2903 // We keep track of the last valid kernel position in case we are in underrun 2904 // and the normal mixer period is the same as the fast mixer period, or there 2905 // is some error from the HAL. 2906 if (mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL] >= 0) { 2907 mTimestamp.mPosition[ExtendedTimestamp::LOCATION_KERNEL_LASTKERNELOK] = 2908 mTimestamp.mPosition[ExtendedTimestamp::LOCATION_KERNEL]; 2909 mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL_LASTKERNELOK] = 2910 mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL]; 2911 2912 mTimestamp.mPosition[ExtendedTimestamp::LOCATION_SERVER_LASTKERNELOK] = 2913 mTimestamp.mPosition[ExtendedTimestamp::LOCATION_SERVER]; 2914 mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_SERVER_LASTKERNELOK] = 2915 mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_SERVER]; 2916 } else { 2917 ALOGV("getTimestamp error - no valid kernel position"); 2918 } 2919 2920 // copy over kernel info 2921 mTimestamp.mPosition[ExtendedTimestamp::LOCATION_KERNEL] = 2922 timestamp.mPosition[ExtendedTimestamp::LOCATION_KERNEL]; 2923 mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL] = 2924 timestamp.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL]; 2925 } 2926 // mFramesWritten for non-offloaded tracks are contiguous 2927 // even after standby() is called. This is useful for the track frame 2928 // to sink frame mapping. 2929 mTimestamp.mPosition[ExtendedTimestamp::LOCATION_SERVER] = mFramesWritten; 2930 mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_SERVER] = systemTime(); 2931 const size_t size = mActiveTracks.size(); 2932 for (size_t i = 0; i < size; ++i) { 2933 sp<Track> t = mActiveTracks[i].promote(); 2934 if (t != 0 && !t->isFastTrack()) { 2935 t->updateTrackFrameInfo( 2936 t->mAudioTrackServerProxy->framesReleased(), 2937 mFramesWritten, 2938 mTimestamp); 2939 } 2940 } 2941 2942 saveOutputTracks(); 2943 if (mSignalPending) { 2944 // A signal was raised while we were unlocked 2945 mSignalPending = false; 2946 } else if (waitingAsyncCallback_l()) { 2947 if (exitPending()) { 2948 break; 2949 } 2950 bool released = false; 2951 if (!keepWakeLock()) { 2952 releaseWakeLock_l(); 2953 released = true; 2954 } 2955 mWakeLockUids.clear(); 2956 mActiveTracksGeneration++; 2957 ALOGV("wait async completion"); 2958 mWaitWorkCV.wait(mLock); 2959 ALOGV("async completion/wake"); 2960 if (released) { 2961 acquireWakeLock_l(); 2962 } 2963 mStandbyTimeNs = systemTime() + mStandbyDelayNs; 2964 mSleepTimeUs = 0; 2965 2966 continue; 2967 } 2968 if ((!mActiveTracks.size() && systemTime() > mStandbyTimeNs) || 2969 isSuspended()) { 2970 // put audio hardware into standby after short delay 2971 if (shouldStandby_l()) { 2972 2973 threadLoop_standby(); 2974 2975 mStandby = true; 2976 } 2977 2978 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2979 // we're about to wait, flush the binder command buffer 2980 IPCThreadState::self()->flushCommands(); 2981 2982 clearOutputTracks(); 2983 2984 if (exitPending()) { 2985 break; 2986 } 2987 2988 releaseWakeLock_l(); 2989 mWakeLockUids.clear(); 2990 mActiveTracksGeneration++; 2991 // wait until we have something to do... 2992 ALOGV("%s going to sleep", myName.string()); 2993 mWaitWorkCV.wait(mLock); 2994 ALOGV("%s waking up", myName.string()); 2995 acquireWakeLock_l(); 2996 2997 mMixerStatus = MIXER_IDLE; 2998 mMixerStatusIgnoringFastTracks = MIXER_IDLE; 2999 mBytesWritten = 0; 3000 mBytesRemaining = 0; 3001 checkSilentMode_l(); 3002 3003 mStandbyTimeNs = systemTime() + mStandbyDelayNs; 3004 mSleepTimeUs = mIdleSleepTimeUs; 3005 if (mType == MIXER) { 3006 sleepTimeShift = 0; 3007 } 3008 3009 continue; 3010 } 3011 } 3012 // mMixerStatusIgnoringFastTracks is also updated internally 3013 mMixerStatus = prepareTracks_l(&tracksToRemove); 3014 3015 // compare with previously applied list 3016 if (lastGeneration != mActiveTracksGeneration) { 3017 // update wakelock 3018 updateWakeLockUids_l(mWakeLockUids); 3019 lastGeneration = mActiveTracksGeneration; 3020 } 3021 3022 // prevent any changes in effect chain list and in each effect chain 3023 // during mixing and effect process as the audio buffers could be deleted 3024 // or modified if an effect is created or deleted 3025 lockEffectChains_l(effectChains); 3026 } // mLock scope ends 3027 3028 if (mBytesRemaining == 0) { 3029 mCurrentWriteLength = 0; 3030 if (mMixerStatus == MIXER_TRACKS_READY) { 3031 // threadLoop_mix() sets mCurrentWriteLength 3032 threadLoop_mix(); 3033 } else if ((mMixerStatus != MIXER_DRAIN_TRACK) 3034 && (mMixerStatus != MIXER_DRAIN_ALL)) { 3035 // threadLoop_sleepTime sets mSleepTimeUs to 0 if data 3036 // must be written to HAL 3037 threadLoop_sleepTime(); 3038 if (mSleepTimeUs == 0) { 3039 mCurrentWriteLength = mSinkBufferSize; 3040 } 3041 } 3042 // Either threadLoop_mix() or threadLoop_sleepTime() should have set 3043 // mMixerBuffer with data if mMixerBufferValid is true and mSleepTimeUs == 0. 3044 // Merge mMixerBuffer data into mEffectBuffer (if any effects are valid) 3045 // or mSinkBuffer (if there are no effects). 3046 // 3047 // This is done pre-effects computation; if effects change to 3048 // support higher precision, this needs to move. 3049 // 3050 // mMixerBufferValid is only set true by MixerThread::prepareTracks_l(). 3051 // TODO use mSleepTimeUs == 0 as an additional condition. 3052 if (mMixerBufferValid) { 3053 void *buffer = mEffectBufferValid ? mEffectBuffer : mSinkBuffer; 3054 audio_format_t format = mEffectBufferValid ? mEffectBufferFormat : mFormat; 3055 3056 // mono blend occurs for mixer threads only (not direct or offloaded) 3057 // and is handled here if we're going directly to the sink. 3058 if (requireMonoBlend() && !mEffectBufferValid) { 3059 mono_blend(mMixerBuffer, mMixerBufferFormat, mChannelCount, mNormalFrameCount, 3060 true /*limit*/); 3061 } 3062 3063 memcpy_by_audio_format(buffer, format, mMixerBuffer, mMixerBufferFormat, 3064 mNormalFrameCount * mChannelCount); 3065 } 3066 3067 mBytesRemaining = mCurrentWriteLength; 3068 if (isSuspended()) { 3069 mSleepTimeUs = suspendSleepTimeUs(); 3070 // simulate write to HAL when suspended 3071 mBytesWritten += mSinkBufferSize; 3072 mFramesWritten += mSinkBufferSize / mFrameSize; 3073 mBytesRemaining = 0; 3074 } 3075 3076 // only process effects if we're going to write 3077 if (mSleepTimeUs == 0 && mType != OFFLOAD) { 3078 for (size_t i = 0; i < effectChains.size(); i ++) { 3079 effectChains[i]->process_l(); 3080 } 3081 } 3082 } 3083 // Process effect chains for offloaded thread even if no audio 3084 // was read from audio track: process only updates effect state 3085 // and thus does have to be synchronized with audio writes but may have 3086 // to be called while waiting for async write callback 3087 if (mType == OFFLOAD) { 3088 for (size_t i = 0; i < effectChains.size(); i ++) { 3089 effectChains[i]->process_l(); 3090 } 3091 } 3092 3093 // Only if the Effects buffer is enabled and there is data in the 3094 // Effects buffer (buffer valid), we need to 3095 // copy into the sink buffer. 3096 // TODO use mSleepTimeUs == 0 as an additional condition. 3097 if (mEffectBufferValid) { 3098 //ALOGV("writing effect buffer to sink buffer format %#x", mFormat); 3099 3100 if (requireMonoBlend()) { 3101 mono_blend(mEffectBuffer, mEffectBufferFormat, mChannelCount, mNormalFrameCount, 3102 true /*limit*/); 3103 } 3104 3105 memcpy_by_audio_format(mSinkBuffer, mFormat, mEffectBuffer, mEffectBufferFormat, 3106 mNormalFrameCount * mChannelCount); 3107 } 3108 3109 // enable changes in effect chain 3110 unlockEffectChains(effectChains); 3111 3112 if (!waitingAsyncCallback()) { 3113 // mSleepTimeUs == 0 means we must write to audio hardware 3114 if (mSleepTimeUs == 0) { 3115 ssize_t ret = 0; 3116 if (mBytesRemaining) { 3117 ret = threadLoop_write(); 3118 if (ret < 0) { 3119 mBytesRemaining = 0; 3120 } else { 3121 mBytesWritten += ret; 3122 mBytesRemaining -= ret; 3123 mFramesWritten += ret / mFrameSize; 3124 } 3125 } else if ((mMixerStatus == MIXER_DRAIN_TRACK) || 3126 (mMixerStatus == MIXER_DRAIN_ALL)) { 3127 threadLoop_drain(); 3128 } 3129 if (mType == MIXER && !mStandby) { 3130 // write blocked detection 3131 nsecs_t now = systemTime(); 3132 nsecs_t delta = now - mLastWriteTime; 3133 if (delta > maxPeriod) { 3134 mNumDelayedWrites++; 3135 if ((now - lastWarning) > kWarningThrottleNs) { 3136 ATRACE_NAME("underrun"); 3137 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 3138 (unsigned long long) ns2ms(delta), mNumDelayedWrites, this); 3139 lastWarning = now; 3140 } 3141 } 3142 3143 if (mThreadThrottle 3144 && mMixerStatus == MIXER_TRACKS_READY // we are mixing (active tracks) 3145 && ret > 0) { // we wrote something 3146 // Limit MixerThread data processing to no more than twice the 3147 // expected processing rate. 3148 // 3149 // This helps prevent underruns with NuPlayer and other applications 3150 // which may set up buffers that are close to the minimum size, or use 3151 // deep buffers, and rely on a double-buffering sleep strategy to fill. 3152 // 3153 // The throttle smooths out sudden large data drains from the device, 3154 // e.g. when it comes out of standby, which often causes problems with 3155 // (1) mixer threads without a fast mixer (which has its own warm-up) 3156 // (2) minimum buffer sized tracks (even if the track is full, 3157 // the app won't fill fast enough to handle the sudden draw). 3158 3159 const int32_t deltaMs = delta / 1000000; 3160 const int32_t throttleMs = mHalfBufferMs - deltaMs; 3161 if ((signed)mHalfBufferMs >= throttleMs && throttleMs > 0) { 3162 usleep(throttleMs * 1000); 3163 // notify of throttle start on verbose log 3164 ALOGV_IF(mThreadThrottleEndMs == mThreadThrottleTimeMs, 3165 "mixer(%p) throttle begin:" 3166 " ret(%zd) deltaMs(%d) requires sleep %d ms", 3167 this, ret, deltaMs, throttleMs); 3168 mThreadThrottleTimeMs += throttleMs; 3169 } else { 3170 uint32_t diff = mThreadThrottleTimeMs - mThreadThrottleEndMs; 3171 if (diff > 0) { 3172 // notify of throttle end on debug log 3173 // but prevent spamming for bluetooth 3174 ALOGD_IF(!audio_is_a2dp_out_device(outDevice()), 3175 "mixer(%p) throttle end: throttle time(%u)", this, diff); 3176 mThreadThrottleEndMs = mThreadThrottleTimeMs; 3177 } 3178 } 3179 } 3180 } 3181 3182 } else { 3183 ATRACE_BEGIN("sleep"); 3184 if ((mType == OFFLOAD) && !audio_has_proportional_frames(mFormat)) { 3185 Mutex::Autolock _l(mLock); 3186 if (!mSignalPending && !exitPending()) { 3187 // If more than one buffer has been written to the audio HAL since exiting 3188 // standby or last flush, do not sleep more than one buffer duration 3189 // since last write and not less than kDirectMinSleepTimeUs. 3190 // Wake up if a command is received 3191 uint32_t timeoutUs = mSleepTimeUs; 3192 if (mBytesWritten >= (int64_t) mBufferSize) { 3193 nsecs_t now = systemTime(); 3194 uint32_t deltaUs = (uint32_t)((now - mLastWriteTime) / 1000); 3195 if (timeoutUs + deltaUs > mBufferDurationUs) { 3196 if (mBufferDurationUs > deltaUs) { 3197 timeoutUs = mBufferDurationUs - deltaUs; 3198 if (timeoutUs < kDirectMinSleepTimeUs) { 3199 timeoutUs = kDirectMinSleepTimeUs; 3200 } 3201 } else { 3202 timeoutUs = kDirectMinSleepTimeUs; 3203 } 3204 } 3205 } 3206 mWaitWorkCV.waitRelative(mLock, microseconds((nsecs_t)timeoutUs)); 3207 } 3208 } else { 3209 usleep(mSleepTimeUs); 3210 } 3211 ATRACE_END(); 3212 } 3213 } 3214 3215 // Finally let go of removed track(s), without the lock held 3216 // since we can't guarantee the destructors won't acquire that 3217 // same lock. This will also mutate and push a new fast mixer state. 3218 threadLoop_removeTracks(tracksToRemove); 3219 tracksToRemove.clear(); 3220 3221 // FIXME I don't understand the need for this here; 3222 // it was in the original code but maybe the 3223 // assignment in saveOutputTracks() makes this unnecessary? 3224 clearOutputTracks(); 3225 3226 // Effect chains will be actually deleted here if they were removed from 3227 // mEffectChains list during mixing or effects processing 3228 effectChains.clear(); 3229 3230 // FIXME Note that the above .clear() is no longer necessary since effectChains 3231 // is now local to this block, but will keep it for now (at least until merge done). 3232 } 3233 3234 threadLoop_exit(); 3235 3236 if (!mStandby) { 3237 threadLoop_standby(); 3238 mStandby = true; 3239 } 3240 3241 releaseWakeLock(); 3242 mWakeLockUids.clear(); 3243 mActiveTracksGeneration++; 3244 3245 ALOGV("Thread %p type %d exiting", this, mType); 3246 return false; 3247} 3248 3249// removeTracks_l() must be called with ThreadBase::mLock held 3250void AudioFlinger::PlaybackThread::removeTracks_l(const Vector< sp<Track> >& tracksToRemove) 3251{ 3252 size_t count = tracksToRemove.size(); 3253 if (count > 0) { 3254 for (size_t i=0 ; i<count ; i++) { 3255 const sp<Track>& track = tracksToRemove.itemAt(i); 3256 mActiveTracks.remove(track); 3257 mWakeLockUids.remove(track->uid()); 3258 mActiveTracksGeneration++; 3259 ALOGV("removeTracks_l removing track on session %d", track->sessionId()); 3260 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 3261 if (chain != 0) { 3262 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), 3263 track->sessionId()); 3264 chain->decActiveTrackCnt(); 3265 } 3266 if (track->isTerminated()) { 3267 removeTrack_l(track); 3268 } 3269 } 3270 } 3271 3272} 3273 3274status_t AudioFlinger::PlaybackThread::getTimestamp_l(AudioTimestamp& timestamp) 3275{ 3276 if (mNormalSink != 0) { 3277 ExtendedTimestamp ets; 3278 status_t status = mNormalSink->getTimestamp(ets); 3279 if (status == NO_ERROR) { 3280 status = ets.getBestTimestamp(×tamp); 3281 } 3282 return status; 3283 } 3284 if ((mType == OFFLOAD || mType == DIRECT) 3285 && mOutput != NULL && mOutput->stream->get_presentation_position) { 3286 uint64_t position64; 3287 int ret = mOutput->getPresentationPosition(&position64, ×tamp.mTime); 3288 if (ret == 0) { 3289 timestamp.mPosition = (uint32_t)position64; 3290 return NO_ERROR; 3291 } 3292 } 3293 return INVALID_OPERATION; 3294} 3295 3296status_t AudioFlinger::MixerThread::createAudioPatch_l(const struct audio_patch *patch, 3297 audio_patch_handle_t *handle) 3298{ 3299 AutoPark<FastMixer> park(mFastMixer); 3300 3301 status_t status = PlaybackThread::createAudioPatch_l(patch, handle); 3302 3303 return status; 3304} 3305 3306status_t AudioFlinger::PlaybackThread::createAudioPatch_l(const struct audio_patch *patch, 3307 audio_patch_handle_t *handle) 3308{ 3309 status_t status = NO_ERROR; 3310 3311 // store new device and send to effects 3312 audio_devices_t type = AUDIO_DEVICE_NONE; 3313 for (unsigned int i = 0; i < patch->num_sinks; i++) { 3314 type |= patch->sinks[i].ext.device.type; 3315 } 3316 3317#ifdef ADD_BATTERY_DATA 3318 // when changing the audio output device, call addBatteryData to notify 3319 // the change 3320 if (mOutDevice != type) { 3321 uint32_t params = 0; 3322 // check whether speaker is on 3323 if (type & AUDIO_DEVICE_OUT_SPEAKER) { 3324 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 3325 } 3326 3327 audio_devices_t deviceWithoutSpeaker 3328 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 3329 // check if any other device (except speaker) is on 3330 if (type & deviceWithoutSpeaker) { 3331 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 3332 } 3333 3334 if (params != 0) { 3335 addBatteryData(params); 3336 } 3337 } 3338#endif 3339 3340 for (size_t i = 0; i < mEffectChains.size(); i++) { 3341 mEffectChains[i]->setDevice_l(type); 3342 } 3343 3344 // mPrevOutDevice is the latest device set by createAudioPatch_l(). It is not set when 3345 // the thread is created so that the first patch creation triggers an ioConfigChanged callback 3346 bool configChanged = mPrevOutDevice != type; 3347 mOutDevice = type; 3348 mPatch = *patch; 3349 3350 if (mOutput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 3351 audio_hw_device_t *hwDevice = mOutput->audioHwDev->hwDevice(); 3352 status = hwDevice->create_audio_patch(hwDevice, 3353 patch->num_sources, 3354 patch->sources, 3355 patch->num_sinks, 3356 patch->sinks, 3357 handle); 3358 } else { 3359 char *address; 3360 if (strcmp(patch->sinks[0].ext.device.address, "") != 0) { 3361 //FIXME: we only support address on first sink with HAL version < 3.0 3362 address = audio_device_address_to_parameter( 3363 patch->sinks[0].ext.device.type, 3364 patch->sinks[0].ext.device.address); 3365 } else { 3366 address = (char *)calloc(1, 1); 3367 } 3368 AudioParameter param = AudioParameter(String8(address)); 3369 free(address); 3370 param.addInt(String8(AUDIO_PARAMETER_STREAM_ROUTING), (int)type); 3371 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3372 param.toString().string()); 3373 *handle = AUDIO_PATCH_HANDLE_NONE; 3374 } 3375 if (configChanged) { 3376 mPrevOutDevice = type; 3377 sendIoConfigEvent_l(AUDIO_OUTPUT_CONFIG_CHANGED); 3378 } 3379 return status; 3380} 3381 3382status_t AudioFlinger::MixerThread::releaseAudioPatch_l(const audio_patch_handle_t handle) 3383{ 3384 AutoPark<FastMixer> park(mFastMixer); 3385 3386 status_t status = PlaybackThread::releaseAudioPatch_l(handle); 3387 3388 return status; 3389} 3390 3391status_t AudioFlinger::PlaybackThread::releaseAudioPatch_l(const audio_patch_handle_t handle) 3392{ 3393 status_t status = NO_ERROR; 3394 3395 mOutDevice = AUDIO_DEVICE_NONE; 3396 3397 if (mOutput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 3398 audio_hw_device_t *hwDevice = mOutput->audioHwDev->hwDevice(); 3399 status = hwDevice->release_audio_patch(hwDevice, handle); 3400 } else { 3401 AudioParameter param; 3402 param.addInt(String8(AUDIO_PARAMETER_STREAM_ROUTING), 0); 3403 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3404 param.toString().string()); 3405 } 3406 return status; 3407} 3408 3409void AudioFlinger::PlaybackThread::addPatchTrack(const sp<PatchTrack>& track) 3410{ 3411 Mutex::Autolock _l(mLock); 3412 mTracks.add(track); 3413} 3414 3415void AudioFlinger::PlaybackThread::deletePatchTrack(const sp<PatchTrack>& track) 3416{ 3417 Mutex::Autolock _l(mLock); 3418 destroyTrack_l(track); 3419} 3420 3421void AudioFlinger::PlaybackThread::getAudioPortConfig(struct audio_port_config *config) 3422{ 3423 ThreadBase::getAudioPortConfig(config); 3424 config->role = AUDIO_PORT_ROLE_SOURCE; 3425 config->ext.mix.hw_module = mOutput->audioHwDev->handle(); 3426 config->ext.mix.usecase.stream = AUDIO_STREAM_DEFAULT; 3427} 3428 3429// ---------------------------------------------------------------------------- 3430 3431AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 3432 audio_io_handle_t id, audio_devices_t device, bool systemReady, type_t type) 3433 : PlaybackThread(audioFlinger, output, id, device, type, systemReady), 3434 // mAudioMixer below 3435 // mFastMixer below 3436 mFastMixerFutex(0), 3437 mMasterMono(false) 3438 // mOutputSink below 3439 // mPipeSink below 3440 // mNormalSink below 3441{ 3442 ALOGV("MixerThread() id=%d device=%#x type=%d", id, device, type); 3443 ALOGV("mSampleRate=%u, mChannelMask=%#x, mChannelCount=%u, mFormat=%d, mFrameSize=%zu, " 3444 "mFrameCount=%zu, mNormalFrameCount=%zu", 3445 mSampleRate, mChannelMask, mChannelCount, mFormat, mFrameSize, mFrameCount, 3446 mNormalFrameCount); 3447 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 3448 3449 if (type == DUPLICATING) { 3450 // The Duplicating thread uses the AudioMixer and delivers data to OutputTracks 3451 // (downstream MixerThreads) in DuplicatingThread::threadLoop_write(). 3452 // Do not create or use mFastMixer, mOutputSink, mPipeSink, or mNormalSink. 3453 return; 3454 } 3455 // create an NBAIO sink for the HAL output stream, and negotiate 3456 mOutputSink = new AudioStreamOutSink(output->stream); 3457 size_t numCounterOffers = 0; 3458 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount, mFormat)}; 3459#if !LOG_NDEBUG 3460 ssize_t index = 3461#else 3462 (void) 3463#endif 3464 mOutputSink->negotiate(offers, 1, NULL, numCounterOffers); 3465 ALOG_ASSERT(index == 0); 3466 3467 // initialize fast mixer depending on configuration 3468 bool initFastMixer; 3469 switch (kUseFastMixer) { 3470 case FastMixer_Never: 3471 initFastMixer = false; 3472 break; 3473 case FastMixer_Always: 3474 initFastMixer = true; 3475 break; 3476 case FastMixer_Static: 3477 case FastMixer_Dynamic: 3478 initFastMixer = mFrameCount < mNormalFrameCount; 3479 break; 3480 } 3481 if (initFastMixer) { 3482 audio_format_t fastMixerFormat; 3483 if (mMixerBufferEnabled && mEffectBufferEnabled) { 3484 fastMixerFormat = AUDIO_FORMAT_PCM_FLOAT; 3485 } else { 3486 fastMixerFormat = AUDIO_FORMAT_PCM_16_BIT; 3487 } 3488 if (mFormat != fastMixerFormat) { 3489 // change our Sink format to accept our intermediate precision 3490 mFormat = fastMixerFormat; 3491 free(mSinkBuffer); 3492 mFrameSize = mChannelCount * audio_bytes_per_sample(mFormat); 3493 const size_t sinkBufferSize = mNormalFrameCount * mFrameSize; 3494 (void)posix_memalign(&mSinkBuffer, 32, sinkBufferSize); 3495 } 3496 3497 // create a MonoPipe to connect our submix to FastMixer 3498 NBAIO_Format format = mOutputSink->format(); 3499#ifdef TEE_SINK 3500 NBAIO_Format origformat = format; 3501#endif 3502 // adjust format to match that of the Fast Mixer 3503 ALOGV("format changed from %d to %d", format.mFormat, fastMixerFormat); 3504 format.mFormat = fastMixerFormat; 3505 format.mFrameSize = audio_bytes_per_sample(format.mFormat) * format.mChannelCount; 3506 3507 // This pipe depth compensates for scheduling latency of the normal mixer thread. 3508 // When it wakes up after a maximum latency, it runs a few cycles quickly before 3509 // finally blocking. Note the pipe implementation rounds up the request to a power of 2. 3510 MonoPipe *monoPipe = new MonoPipe(mNormalFrameCount * 4, format, true /*writeCanBlock*/); 3511 const NBAIO_Format offers[1] = {format}; 3512 size_t numCounterOffers = 0; 3513#if !LOG_NDEBUG || defined(TEE_SINK) 3514 ssize_t index = 3515#else 3516 (void) 3517#endif 3518 monoPipe->negotiate(offers, 1, NULL, numCounterOffers); 3519 ALOG_ASSERT(index == 0); 3520 monoPipe->setAvgFrames((mScreenState & 1) ? 3521 (monoPipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 3522 mPipeSink = monoPipe; 3523 3524#ifdef TEE_SINK 3525 if (mTeeSinkOutputEnabled) { 3526 // create a Pipe to archive a copy of FastMixer's output for dumpsys 3527 Pipe *teeSink = new Pipe(mTeeSinkOutputFrames, origformat); 3528 const NBAIO_Format offers2[1] = {origformat}; 3529 numCounterOffers = 0; 3530 index = teeSink->negotiate(offers2, 1, NULL, numCounterOffers); 3531 ALOG_ASSERT(index == 0); 3532 mTeeSink = teeSink; 3533 PipeReader *teeSource = new PipeReader(*teeSink); 3534 numCounterOffers = 0; 3535 index = teeSource->negotiate(offers2, 1, NULL, numCounterOffers); 3536 ALOG_ASSERT(index == 0); 3537 mTeeSource = teeSource; 3538 } 3539#endif 3540 3541 // create fast mixer and configure it initially with just one fast track for our submix 3542 mFastMixer = new FastMixer(); 3543 FastMixerStateQueue *sq = mFastMixer->sq(); 3544#ifdef STATE_QUEUE_DUMP 3545 sq->setObserverDump(&mStateQueueObserverDump); 3546 sq->setMutatorDump(&mStateQueueMutatorDump); 3547#endif 3548 FastMixerState *state = sq->begin(); 3549 FastTrack *fastTrack = &state->mFastTracks[0]; 3550 // wrap the source side of the MonoPipe to make it an AudioBufferProvider 3551 fastTrack->mBufferProvider = new SourceAudioBufferProvider(new MonoPipeReader(monoPipe)); 3552 fastTrack->mVolumeProvider = NULL; 3553 fastTrack->mChannelMask = mChannelMask; // mPipeSink channel mask for audio to FastMixer 3554 fastTrack->mFormat = mFormat; // mPipeSink format for audio to FastMixer 3555 fastTrack->mGeneration++; 3556 state->mFastTracksGen++; 3557 state->mTrackMask = 1; 3558 // fast mixer will use the HAL output sink 3559 state->mOutputSink = mOutputSink.get(); 3560 state->mOutputSinkGen++; 3561 state->mFrameCount = mFrameCount; 3562 state->mCommand = FastMixerState::COLD_IDLE; 3563 // already done in constructor initialization list 3564 //mFastMixerFutex = 0; 3565 state->mColdFutexAddr = &mFastMixerFutex; 3566 state->mColdGen++; 3567 state->mDumpState = &mFastMixerDumpState; 3568#ifdef TEE_SINK 3569 state->mTeeSink = mTeeSink.get(); 3570#endif 3571 mFastMixerNBLogWriter = audioFlinger->newWriter_l(kFastMixerLogSize, "FastMixer"); 3572 state->mNBLogWriter = mFastMixerNBLogWriter.get(); 3573 sq->end(); 3574 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3575 3576 // start the fast mixer 3577 mFastMixer->run("FastMixer", PRIORITY_URGENT_AUDIO); 3578 pid_t tid = mFastMixer->getTid(); 3579 sendPrioConfigEvent(getpid_cached, tid, kPriorityFastMixer); 3580 3581#ifdef AUDIO_WATCHDOG 3582 // create and start the watchdog 3583 mAudioWatchdog = new AudioWatchdog(); 3584 mAudioWatchdog->setDump(&mAudioWatchdogDump); 3585 mAudioWatchdog->run("AudioWatchdog", PRIORITY_URGENT_AUDIO); 3586 tid = mAudioWatchdog->getTid(); 3587 sendPrioConfigEvent(getpid_cached, tid, kPriorityFastMixer); 3588#endif 3589 3590 } 3591 3592 switch (kUseFastMixer) { 3593 case FastMixer_Never: 3594 case FastMixer_Dynamic: 3595 mNormalSink = mOutputSink; 3596 break; 3597 case FastMixer_Always: 3598 mNormalSink = mPipeSink; 3599 break; 3600 case FastMixer_Static: 3601 mNormalSink = initFastMixer ? mPipeSink : mOutputSink; 3602 break; 3603 } 3604} 3605 3606AudioFlinger::MixerThread::~MixerThread() 3607{ 3608 if (mFastMixer != 0) { 3609 FastMixerStateQueue *sq = mFastMixer->sq(); 3610 FastMixerState *state = sq->begin(); 3611 if (state->mCommand == FastMixerState::COLD_IDLE) { 3612 int32_t old = android_atomic_inc(&mFastMixerFutex); 3613 if (old == -1) { 3614 (void) syscall(__NR_futex, &mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 3615 } 3616 } 3617 state->mCommand = FastMixerState::EXIT; 3618 sq->end(); 3619 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3620 mFastMixer->join(); 3621 // Though the fast mixer thread has exited, it's state queue is still valid. 3622 // We'll use that extract the final state which contains one remaining fast track 3623 // corresponding to our sub-mix. 3624 state = sq->begin(); 3625 ALOG_ASSERT(state->mTrackMask == 1); 3626 FastTrack *fastTrack = &state->mFastTracks[0]; 3627 ALOG_ASSERT(fastTrack->mBufferProvider != NULL); 3628 delete fastTrack->mBufferProvider; 3629 sq->end(false /*didModify*/); 3630 mFastMixer.clear(); 3631#ifdef AUDIO_WATCHDOG 3632 if (mAudioWatchdog != 0) { 3633 mAudioWatchdog->requestExit(); 3634 mAudioWatchdog->requestExitAndWait(); 3635 mAudioWatchdog.clear(); 3636 } 3637#endif 3638 } 3639 mAudioFlinger->unregisterWriter(mFastMixerNBLogWriter); 3640 delete mAudioMixer; 3641} 3642 3643 3644uint32_t AudioFlinger::MixerThread::correctLatency_l(uint32_t latency) const 3645{ 3646 if (mFastMixer != 0) { 3647 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 3648 latency += (pipe->getAvgFrames() * 1000) / mSampleRate; 3649 } 3650 return latency; 3651} 3652 3653 3654void AudioFlinger::MixerThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) 3655{ 3656 PlaybackThread::threadLoop_removeTracks(tracksToRemove); 3657} 3658 3659ssize_t AudioFlinger::MixerThread::threadLoop_write() 3660{ 3661 // FIXME we should only do one push per cycle; confirm this is true 3662 // Start the fast mixer if it's not already running 3663 if (mFastMixer != 0) { 3664 FastMixerStateQueue *sq = mFastMixer->sq(); 3665 FastMixerState *state = sq->begin(); 3666 if (state->mCommand != FastMixerState::MIX_WRITE && 3667 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)) { 3668 if (state->mCommand == FastMixerState::COLD_IDLE) { 3669 3670 // FIXME workaround for first HAL write being CPU bound on some devices 3671 ATRACE_BEGIN("write"); 3672 mOutput->write((char *)mSinkBuffer, 0); 3673 ATRACE_END(); 3674 3675 int32_t old = android_atomic_inc(&mFastMixerFutex); 3676 if (old == -1) { 3677 (void) syscall(__NR_futex, &mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 3678 } 3679#ifdef AUDIO_WATCHDOG 3680 if (mAudioWatchdog != 0) { 3681 mAudioWatchdog->resume(); 3682 } 3683#endif 3684 } 3685 state->mCommand = FastMixerState::MIX_WRITE; 3686#ifdef FAST_THREAD_STATISTICS 3687 mFastMixerDumpState.increaseSamplingN(mAudioFlinger->isLowRamDevice() ? 3688 FastThreadDumpState::kSamplingNforLowRamDevice : FastThreadDumpState::kSamplingN); 3689#endif 3690 sq->end(); 3691 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3692 if (kUseFastMixer == FastMixer_Dynamic) { 3693 mNormalSink = mPipeSink; 3694 } 3695 } else { 3696 sq->end(false /*didModify*/); 3697 } 3698 } 3699 return PlaybackThread::threadLoop_write(); 3700} 3701 3702void AudioFlinger::MixerThread::threadLoop_standby() 3703{ 3704 // Idle the fast mixer if it's currently running 3705 if (mFastMixer != 0) { 3706 FastMixerStateQueue *sq = mFastMixer->sq(); 3707 FastMixerState *state = sq->begin(); 3708 if (!(state->mCommand & FastMixerState::IDLE)) { 3709 state->mCommand = FastMixerState::COLD_IDLE; 3710 state->mColdFutexAddr = &mFastMixerFutex; 3711 state->mColdGen++; 3712 mFastMixerFutex = 0; 3713 sq->end(); 3714 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 3715 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3716 if (kUseFastMixer == FastMixer_Dynamic) { 3717 mNormalSink = mOutputSink; 3718 } 3719#ifdef AUDIO_WATCHDOG 3720 if (mAudioWatchdog != 0) { 3721 mAudioWatchdog->pause(); 3722 } 3723#endif 3724 } else { 3725 sq->end(false /*didModify*/); 3726 } 3727 } 3728 PlaybackThread::threadLoop_standby(); 3729} 3730 3731bool AudioFlinger::PlaybackThread::waitingAsyncCallback_l() 3732{ 3733 return false; 3734} 3735 3736bool AudioFlinger::PlaybackThread::shouldStandby_l() 3737{ 3738 return !mStandby; 3739} 3740 3741bool AudioFlinger::PlaybackThread::waitingAsyncCallback() 3742{ 3743 Mutex::Autolock _l(mLock); 3744 return waitingAsyncCallback_l(); 3745} 3746 3747// shared by MIXER and DIRECT, overridden by DUPLICATING 3748void AudioFlinger::PlaybackThread::threadLoop_standby() 3749{ 3750 ALOGV("Audio hardware entering standby, mixer %p, suspend count %d", this, mSuspended); 3751 mOutput->standby(); 3752 if (mUseAsyncWrite != 0) { 3753 // discard any pending drain or write ack by incrementing sequence 3754 mWriteAckSequence = (mWriteAckSequence + 2) & ~1; 3755 mDrainSequence = (mDrainSequence + 2) & ~1; 3756 ALOG_ASSERT(mCallbackThread != 0); 3757 mCallbackThread->setWriteBlocked(mWriteAckSequence); 3758 mCallbackThread->setDraining(mDrainSequence); 3759 } 3760 mHwPaused = false; 3761} 3762 3763void AudioFlinger::PlaybackThread::onAddNewTrack_l() 3764{ 3765 ALOGV("signal playback thread"); 3766 broadcast_l(); 3767} 3768 3769void AudioFlinger::MixerThread::threadLoop_mix() 3770{ 3771 // mix buffers... 3772 mAudioMixer->process(); 3773 mCurrentWriteLength = mSinkBufferSize; 3774 // increase sleep time progressively when application underrun condition clears. 3775 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 3776 // that a steady state of alternating ready/not ready conditions keeps the sleep time 3777 // such that we would underrun the audio HAL. 3778 if ((mSleepTimeUs == 0) && (sleepTimeShift > 0)) { 3779 sleepTimeShift--; 3780 } 3781 mSleepTimeUs = 0; 3782 mStandbyTimeNs = systemTime() + mStandbyDelayNs; 3783 //TODO: delay standby when effects have a tail 3784 3785} 3786 3787void AudioFlinger::MixerThread::threadLoop_sleepTime() 3788{ 3789 // If no tracks are ready, sleep once for the duration of an output 3790 // buffer size, then write 0s to the output 3791 if (mSleepTimeUs == 0) { 3792 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3793 mSleepTimeUs = mActiveSleepTimeUs >> sleepTimeShift; 3794 if (mSleepTimeUs < kMinThreadSleepTimeUs) { 3795 mSleepTimeUs = kMinThreadSleepTimeUs; 3796 } 3797 // reduce sleep time in case of consecutive application underruns to avoid 3798 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 3799 // duration we would end up writing less data than needed by the audio HAL if 3800 // the condition persists. 3801 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 3802 sleepTimeShift++; 3803 } 3804 } else { 3805 mSleepTimeUs = mIdleSleepTimeUs; 3806 } 3807 } else if (mBytesWritten != 0 || (mMixerStatus == MIXER_TRACKS_ENABLED)) { 3808 // clear out mMixerBuffer or mSinkBuffer, to ensure buffers are cleared 3809 // before effects processing or output. 3810 if (mMixerBufferValid) { 3811 memset(mMixerBuffer, 0, mMixerBufferSize); 3812 } else { 3813 memset(mSinkBuffer, 0, mSinkBufferSize); 3814 } 3815 mSleepTimeUs = 0; 3816 ALOGV_IF(mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED), 3817 "anticipated start"); 3818 } 3819 // TODO add standby time extension fct of effect tail 3820} 3821 3822// prepareTracks_l() must be called with ThreadBase::mLock held 3823AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 3824 Vector< sp<Track> > *tracksToRemove) 3825{ 3826 3827 mixer_state mixerStatus = MIXER_IDLE; 3828 // find out which tracks need to be processed 3829 size_t count = mActiveTracks.size(); 3830 size_t mixedTracks = 0; 3831 size_t tracksWithEffect = 0; 3832 // counts only _active_ fast tracks 3833 size_t fastTracks = 0; 3834 uint32_t resetMask = 0; // bit mask of fast tracks that need to be reset 3835 3836 float masterVolume = mMasterVolume; 3837 bool masterMute = mMasterMute; 3838 3839 if (masterMute) { 3840 masterVolume = 0; 3841 } 3842 // Delegate master volume control to effect in output mix effect chain if needed 3843 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 3844 if (chain != 0) { 3845 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 3846 chain->setVolume_l(&v, &v); 3847 masterVolume = (float)((v + (1 << 23)) >> 24); 3848 chain.clear(); 3849 } 3850 3851 // prepare a new state to push 3852 FastMixerStateQueue *sq = NULL; 3853 FastMixerState *state = NULL; 3854 bool didModify = false; 3855 FastMixerStateQueue::block_t block = FastMixerStateQueue::BLOCK_UNTIL_PUSHED; 3856 if (mFastMixer != 0) { 3857 sq = mFastMixer->sq(); 3858 state = sq->begin(); 3859 } 3860 3861 mMixerBufferValid = false; // mMixerBuffer has no valid data until appropriate tracks found. 3862 mEffectBufferValid = false; // mEffectBuffer has no valid data until tracks found. 3863 3864 for (size_t i=0 ; i<count ; i++) { 3865 const sp<Track> t = mActiveTracks[i].promote(); 3866 if (t == 0) { 3867 continue; 3868 } 3869 3870 // this const just means the local variable doesn't change 3871 Track* const track = t.get(); 3872 3873 // process fast tracks 3874 if (track->isFastTrack()) { 3875 3876 // It's theoretically possible (though unlikely) for a fast track to be created 3877 // and then removed within the same normal mix cycle. This is not a problem, as 3878 // the track never becomes active so it's fast mixer slot is never touched. 3879 // The converse, of removing an (active) track and then creating a new track 3880 // at the identical fast mixer slot within the same normal mix cycle, 3881 // is impossible because the slot isn't marked available until the end of each cycle. 3882 int j = track->mFastIndex; 3883 ALOG_ASSERT(0 < j && j < (int)FastMixerState::sMaxFastTracks); 3884 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << j))); 3885 FastTrack *fastTrack = &state->mFastTracks[j]; 3886 3887 // Determine whether the track is currently in underrun condition, 3888 // and whether it had a recent underrun. 3889 FastTrackDump *ftDump = &mFastMixerDumpState.mTracks[j]; 3890 FastTrackUnderruns underruns = ftDump->mUnderruns; 3891 uint32_t recentFull = (underruns.mBitFields.mFull - 3892 track->mObservedUnderruns.mBitFields.mFull) & UNDERRUN_MASK; 3893 uint32_t recentPartial = (underruns.mBitFields.mPartial - 3894 track->mObservedUnderruns.mBitFields.mPartial) & UNDERRUN_MASK; 3895 uint32_t recentEmpty = (underruns.mBitFields.mEmpty - 3896 track->mObservedUnderruns.mBitFields.mEmpty) & UNDERRUN_MASK; 3897 uint32_t recentUnderruns = recentPartial + recentEmpty; 3898 track->mObservedUnderruns = underruns; 3899 // don't count underruns that occur while stopping or pausing 3900 // or stopped which can occur when flush() is called while active 3901 if (!(track->isStopping() || track->isPausing() || track->isStopped()) && 3902 recentUnderruns > 0) { 3903 // FIXME fast mixer will pull & mix partial buffers, but we count as a full underrun 3904 track->mAudioTrackServerProxy->tallyUnderrunFrames(recentUnderruns * mFrameCount); 3905 } else { 3906 track->mAudioTrackServerProxy->tallyUnderrunFrames(0); 3907 } 3908 3909 // This is similar to the state machine for normal tracks, 3910 // with a few modifications for fast tracks. 3911 bool isActive = true; 3912 switch (track->mState) { 3913 case TrackBase::STOPPING_1: 3914 // track stays active in STOPPING_1 state until first underrun 3915 if (recentUnderruns > 0 || track->isTerminated()) { 3916 track->mState = TrackBase::STOPPING_2; 3917 } 3918 break; 3919 case TrackBase::PAUSING: 3920 // ramp down is not yet implemented 3921 track->setPaused(); 3922 break; 3923 case TrackBase::RESUMING: 3924 // ramp up is not yet implemented 3925 track->mState = TrackBase::ACTIVE; 3926 break; 3927 case TrackBase::ACTIVE: 3928 if (recentFull > 0 || recentPartial > 0) { 3929 // track has provided at least some frames recently: reset retry count 3930 track->mRetryCount = kMaxTrackRetries; 3931 } 3932 if (recentUnderruns == 0) { 3933 // no recent underruns: stay active 3934 break; 3935 } 3936 // there has recently been an underrun of some kind 3937 if (track->sharedBuffer() == 0) { 3938 // were any of the recent underruns "empty" (no frames available)? 3939 if (recentEmpty == 0) { 3940 // no, then ignore the partial underruns as they are allowed indefinitely 3941 break; 3942 } 3943 // there has recently been an "empty" underrun: decrement the retry counter 3944 if (--(track->mRetryCount) > 0) { 3945 break; 3946 } 3947 // indicate to client process that the track was disabled because of underrun; 3948 // it will then automatically call start() when data is available 3949 track->disable(); 3950 // remove from active list, but state remains ACTIVE [confusing but true] 3951 isActive = false; 3952 break; 3953 } 3954 // fall through 3955 case TrackBase::STOPPING_2: 3956 case TrackBase::PAUSED: 3957 case TrackBase::STOPPED: 3958 case TrackBase::FLUSHED: // flush() while active 3959 // Check for presentation complete if track is inactive 3960 // We have consumed all the buffers of this track. 3961 // This would be incomplete if we auto-paused on underrun 3962 { 3963 size_t audioHALFrames = 3964 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 3965 int64_t framesWritten = mBytesWritten / mFrameSize; 3966 if (!(mStandby || track->presentationComplete(framesWritten, audioHALFrames))) { 3967 // track stays in active list until presentation is complete 3968 break; 3969 } 3970 } 3971 if (track->isStopping_2()) { 3972 track->mState = TrackBase::STOPPED; 3973 } 3974 if (track->isStopped()) { 3975 // Can't reset directly, as fast mixer is still polling this track 3976 // track->reset(); 3977 // So instead mark this track as needing to be reset after push with ack 3978 resetMask |= 1 << i; 3979 } 3980 isActive = false; 3981 break; 3982 case TrackBase::IDLE: 3983 default: 3984 LOG_ALWAYS_FATAL("unexpected track state %d", track->mState); 3985 } 3986 3987 if (isActive) { 3988 // was it previously inactive? 3989 if (!(state->mTrackMask & (1 << j))) { 3990 ExtendedAudioBufferProvider *eabp = track; 3991 VolumeProvider *vp = track; 3992 fastTrack->mBufferProvider = eabp; 3993 fastTrack->mVolumeProvider = vp; 3994 fastTrack->mChannelMask = track->mChannelMask; 3995 fastTrack->mFormat = track->mFormat; 3996 fastTrack->mGeneration++; 3997 state->mTrackMask |= 1 << j; 3998 didModify = true; 3999 // no acknowledgement required for newly active tracks 4000 } 4001 // cache the combined master volume and stream type volume for fast mixer; this 4002 // lacks any synchronization or barrier so VolumeProvider may read a stale value 4003 track->mCachedVolume = masterVolume * mStreamTypes[track->streamType()].volume; 4004 ++fastTracks; 4005 } else { 4006 // was it previously active? 4007 if (state->mTrackMask & (1 << j)) { 4008 fastTrack->mBufferProvider = NULL; 4009 fastTrack->mGeneration++; 4010 state->mTrackMask &= ~(1 << j); 4011 didModify = true; 4012 // If any fast tracks were removed, we must wait for acknowledgement 4013 // because we're about to decrement the last sp<> on those tracks. 4014 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 4015 } else { 4016 LOG_ALWAYS_FATAL("fast track %d should have been active; " 4017 "mState=%d, mTrackMask=%#x, recentUnderruns=%u, isShared=%d", 4018 j, track->mState, state->mTrackMask, recentUnderruns, 4019 track->sharedBuffer() != 0); 4020 } 4021 tracksToRemove->add(track); 4022 // Avoids a misleading display in dumpsys 4023 track->mObservedUnderruns.mBitFields.mMostRecent = UNDERRUN_FULL; 4024 } 4025 continue; 4026 } 4027 4028 { // local variable scope to avoid goto warning 4029 4030 audio_track_cblk_t* cblk = track->cblk(); 4031 4032 // The first time a track is added we wait 4033 // for all its buffers to be filled before processing it 4034 int name = track->name(); 4035 // make sure that we have enough frames to mix one full buffer. 4036 // enforce this condition only once to enable draining the buffer in case the client 4037 // app does not call stop() and relies on underrun to stop: 4038 // hence the test on (mMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 4039 // during last round 4040 size_t desiredFrames; 4041 const uint32_t sampleRate = track->mAudioTrackServerProxy->getSampleRate(); 4042 AudioPlaybackRate playbackRate = track->mAudioTrackServerProxy->getPlaybackRate(); 4043 4044 desiredFrames = sourceFramesNeededWithTimestretch( 4045 sampleRate, mNormalFrameCount, mSampleRate, playbackRate.mSpeed); 4046 // TODO: ONLY USED FOR LEGACY RESAMPLERS, remove when they are removed. 4047 // add frames already consumed but not yet released by the resampler 4048 // because mAudioTrackServerProxy->framesReady() will include these frames 4049 desiredFrames += mAudioMixer->getUnreleasedFrames(track->name()); 4050 4051 uint32_t minFrames = 1; 4052 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() && 4053 (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) { 4054 minFrames = desiredFrames; 4055 } 4056 4057 size_t framesReady = track->framesReady(); 4058 if (ATRACE_ENABLED()) { 4059 // I wish we had formatted trace names 4060 char traceName[16]; 4061 strcpy(traceName, "nRdy"); 4062 int name = track->name(); 4063 if (AudioMixer::TRACK0 <= name && 4064 name < (int) (AudioMixer::TRACK0 + AudioMixer::MAX_NUM_TRACKS)) { 4065 name -= AudioMixer::TRACK0; 4066 traceName[4] = (name / 10) + '0'; 4067 traceName[5] = (name % 10) + '0'; 4068 } else { 4069 traceName[4] = '?'; 4070 traceName[5] = '?'; 4071 } 4072 traceName[6] = '\0'; 4073 ATRACE_INT(traceName, framesReady); 4074 } 4075 if ((framesReady >= minFrames) && track->isReady() && 4076 !track->isPaused() && !track->isTerminated()) 4077 { 4078 ALOGVV("track %d s=%08x [OK] on thread %p", name, cblk->mServer, this); 4079 4080 mixedTracks++; 4081 4082 // track->mainBuffer() != mSinkBuffer or mMixerBuffer means 4083 // there is an effect chain connected to the track 4084 chain.clear(); 4085 if (track->mainBuffer() != mSinkBuffer && 4086 track->mainBuffer() != mMixerBuffer) { 4087 if (mEffectBufferEnabled) { 4088 mEffectBufferValid = true; // Later can set directly. 4089 } 4090 chain = getEffectChain_l(track->sessionId()); 4091 // Delegate volume control to effect in track effect chain if needed 4092 if (chain != 0) { 4093 tracksWithEffect++; 4094 } else { 4095 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on " 4096 "session %d", 4097 name, track->sessionId()); 4098 } 4099 } 4100 4101 4102 int param = AudioMixer::VOLUME; 4103 if (track->mFillingUpStatus == Track::FS_FILLED) { 4104 // no ramp for the first volume setting 4105 track->mFillingUpStatus = Track::FS_ACTIVE; 4106 if (track->mState == TrackBase::RESUMING) { 4107 track->mState = TrackBase::ACTIVE; 4108 param = AudioMixer::RAMP_VOLUME; 4109 } 4110 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 4111 // FIXME should not make a decision based on mServer 4112 } else if (cblk->mServer != 0) { 4113 // If the track is stopped before the first frame was mixed, 4114 // do not apply ramp 4115 param = AudioMixer::RAMP_VOLUME; 4116 } 4117 4118 // compute volume for this track 4119 uint32_t vl, vr; // in U8.24 integer format 4120 float vlf, vrf, vaf; // in [0.0, 1.0] float format 4121 if (track->isPausing() || mStreamTypes[track->streamType()].mute) { 4122 vl = vr = 0; 4123 vlf = vrf = vaf = 0.; 4124 if (track->isPausing()) { 4125 track->setPaused(); 4126 } 4127 } else { 4128 4129 // read original volumes with volume control 4130 float typeVolume = mStreamTypes[track->streamType()].volume; 4131 float v = masterVolume * typeVolume; 4132 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 4133 gain_minifloat_packed_t vlr = proxy->getVolumeLR(); 4134 vlf = float_from_gain(gain_minifloat_unpack_left(vlr)); 4135 vrf = float_from_gain(gain_minifloat_unpack_right(vlr)); 4136 // track volumes come from shared memory, so can't be trusted and must be clamped 4137 if (vlf > GAIN_FLOAT_UNITY) { 4138 ALOGV("Track left volume out of range: %.3g", vlf); 4139 vlf = GAIN_FLOAT_UNITY; 4140 } 4141 if (vrf > GAIN_FLOAT_UNITY) { 4142 ALOGV("Track right volume out of range: %.3g", vrf); 4143 vrf = GAIN_FLOAT_UNITY; 4144 } 4145 // now apply the master volume and stream type volume 4146 vlf *= v; 4147 vrf *= v; 4148 // assuming master volume and stream type volume each go up to 1.0, 4149 // then derive vl and vr as U8.24 versions for the effect chain 4150 const float scaleto8_24 = MAX_GAIN_INT * MAX_GAIN_INT; 4151 vl = (uint32_t) (scaleto8_24 * vlf); 4152 vr = (uint32_t) (scaleto8_24 * vrf); 4153 // vl and vr are now in U8.24 format 4154 uint16_t sendLevel = proxy->getSendLevel_U4_12(); 4155 // send level comes from shared memory and so may be corrupt 4156 if (sendLevel > MAX_GAIN_INT) { 4157 ALOGV("Track send level out of range: %04X", sendLevel); 4158 sendLevel = MAX_GAIN_INT; 4159 } 4160 // vaf is represented as [0.0, 1.0] float by rescaling sendLevel 4161 vaf = v * sendLevel * (1. / MAX_GAIN_INT); 4162 } 4163 4164 // Delegate volume control to effect in track effect chain if needed 4165 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 4166 // Do not ramp volume if volume is controlled by effect 4167 param = AudioMixer::VOLUME; 4168 // Update remaining floating point volume levels 4169 vlf = (float)vl / (1 << 24); 4170 vrf = (float)vr / (1 << 24); 4171 track->mHasVolumeController = true; 4172 } else { 4173 // force no volume ramp when volume controller was just disabled or removed 4174 // from effect chain to avoid volume spike 4175 if (track->mHasVolumeController) { 4176 param = AudioMixer::VOLUME; 4177 } 4178 track->mHasVolumeController = false; 4179 } 4180 4181 // XXX: these things DON'T need to be done each time 4182 mAudioMixer->setBufferProvider(name, track); 4183 mAudioMixer->enable(name); 4184 4185 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, &vlf); 4186 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, &vrf); 4187 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, &vaf); 4188 mAudioMixer->setParameter( 4189 name, 4190 AudioMixer::TRACK, 4191 AudioMixer::FORMAT, (void *)track->format()); 4192 mAudioMixer->setParameter( 4193 name, 4194 AudioMixer::TRACK, 4195 AudioMixer::CHANNEL_MASK, (void *)(uintptr_t)track->channelMask()); 4196 mAudioMixer->setParameter( 4197 name, 4198 AudioMixer::TRACK, 4199 AudioMixer::MIXER_CHANNEL_MASK, (void *)(uintptr_t)mChannelMask); 4200 // limit track sample rate to 2 x output sample rate, which changes at re-configuration 4201 uint32_t maxSampleRate = mSampleRate * AUDIO_RESAMPLER_DOWN_RATIO_MAX; 4202 uint32_t reqSampleRate = track->mAudioTrackServerProxy->getSampleRate(); 4203 if (reqSampleRate == 0) { 4204 reqSampleRate = mSampleRate; 4205 } else if (reqSampleRate > maxSampleRate) { 4206 reqSampleRate = maxSampleRate; 4207 } 4208 mAudioMixer->setParameter( 4209 name, 4210 AudioMixer::RESAMPLE, 4211 AudioMixer::SAMPLE_RATE, 4212 (void *)(uintptr_t)reqSampleRate); 4213 4214 AudioPlaybackRate playbackRate = track->mAudioTrackServerProxy->getPlaybackRate(); 4215 mAudioMixer->setParameter( 4216 name, 4217 AudioMixer::TIMESTRETCH, 4218 AudioMixer::PLAYBACK_RATE, 4219 &playbackRate); 4220 4221 /* 4222 * Select the appropriate output buffer for the track. 4223 * 4224 * Tracks with effects go into their own effects chain buffer 4225 * and from there into either mEffectBuffer or mSinkBuffer. 4226 * 4227 * Other tracks can use mMixerBuffer for higher precision 4228 * channel accumulation. If this buffer is enabled 4229 * (mMixerBufferEnabled true), then selected tracks will accumulate 4230 * into it. 4231 * 4232 */ 4233 if (mMixerBufferEnabled 4234 && (track->mainBuffer() == mSinkBuffer 4235 || track->mainBuffer() == mMixerBuffer)) { 4236 mAudioMixer->setParameter( 4237 name, 4238 AudioMixer::TRACK, 4239 AudioMixer::MIXER_FORMAT, (void *)mMixerBufferFormat); 4240 mAudioMixer->setParameter( 4241 name, 4242 AudioMixer::TRACK, 4243 AudioMixer::MAIN_BUFFER, (void *)mMixerBuffer); 4244 // TODO: override track->mainBuffer()? 4245 mMixerBufferValid = true; 4246 } else { 4247 mAudioMixer->setParameter( 4248 name, 4249 AudioMixer::TRACK, 4250 AudioMixer::MIXER_FORMAT, (void *)AUDIO_FORMAT_PCM_16_BIT); 4251 mAudioMixer->setParameter( 4252 name, 4253 AudioMixer::TRACK, 4254 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 4255 } 4256 mAudioMixer->setParameter( 4257 name, 4258 AudioMixer::TRACK, 4259 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 4260 4261 // reset retry count 4262 track->mRetryCount = kMaxTrackRetries; 4263 4264 // If one track is ready, set the mixer ready if: 4265 // - the mixer was not ready during previous round OR 4266 // - no other track is not ready 4267 if (mMixerStatusIgnoringFastTracks != MIXER_TRACKS_READY || 4268 mixerStatus != MIXER_TRACKS_ENABLED) { 4269 mixerStatus = MIXER_TRACKS_READY; 4270 } 4271 } else { 4272 if (framesReady < desiredFrames && !track->isStopped() && !track->isPaused()) { 4273 ALOGV("track(%p) underrun, framesReady(%zu) < framesDesired(%zd)", 4274 track, framesReady, desiredFrames); 4275 track->mAudioTrackServerProxy->tallyUnderrunFrames(desiredFrames); 4276 } else { 4277 track->mAudioTrackServerProxy->tallyUnderrunFrames(0); 4278 } 4279 4280 // clear effect chain input buffer if an active track underruns to avoid sending 4281 // previous audio buffer again to effects 4282 chain = getEffectChain_l(track->sessionId()); 4283 if (chain != 0) { 4284 chain->clearInputBuffer(); 4285 } 4286 4287 ALOGVV("track %d s=%08x [NOT READY] on thread %p", name, cblk->mServer, this); 4288 if ((track->sharedBuffer() != 0) || track->isTerminated() || 4289 track->isStopped() || track->isPaused()) { 4290 // We have consumed all the buffers of this track. 4291 // Remove it from the list of active tracks. 4292 // TODO: use actual buffer filling status instead of latency when available from 4293 // audio HAL 4294 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 4295 int64_t framesWritten = mBytesWritten / mFrameSize; 4296 if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) { 4297 if (track->isStopped()) { 4298 track->reset(); 4299 } 4300 tracksToRemove->add(track); 4301 } 4302 } else { 4303 // No buffers for this track. Give it a few chances to 4304 // fill a buffer, then remove it from active list. 4305 if (--(track->mRetryCount) <= 0) { 4306 ALOGI("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 4307 tracksToRemove->add(track); 4308 // indicate to client process that the track was disabled because of underrun; 4309 // it will then automatically call start() when data is available 4310 track->disable(); 4311 // If one track is not ready, mark the mixer also not ready if: 4312 // - the mixer was ready during previous round OR 4313 // - no other track is ready 4314 } else if (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY || 4315 mixerStatus != MIXER_TRACKS_READY) { 4316 mixerStatus = MIXER_TRACKS_ENABLED; 4317 } 4318 } 4319 mAudioMixer->disable(name); 4320 } 4321 4322 } // local variable scope to avoid goto warning 4323 4324 } 4325 4326 // Push the new FastMixer state if necessary 4327 bool pauseAudioWatchdog = false; 4328 if (didModify) { 4329 state->mFastTracksGen++; 4330 // if the fast mixer was active, but now there are no fast tracks, then put it in cold idle 4331 if (kUseFastMixer == FastMixer_Dynamic && 4332 state->mCommand == FastMixerState::MIX_WRITE && state->mTrackMask <= 1) { 4333 state->mCommand = FastMixerState::COLD_IDLE; 4334 state->mColdFutexAddr = &mFastMixerFutex; 4335 state->mColdGen++; 4336 mFastMixerFutex = 0; 4337 if (kUseFastMixer == FastMixer_Dynamic) { 4338 mNormalSink = mOutputSink; 4339 } 4340 // If we go into cold idle, need to wait for acknowledgement 4341 // so that fast mixer stops doing I/O. 4342 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 4343 pauseAudioWatchdog = true; 4344 } 4345 } 4346 if (sq != NULL) { 4347 sq->end(didModify); 4348 sq->push(block); 4349 } 4350#ifdef AUDIO_WATCHDOG 4351 if (pauseAudioWatchdog && mAudioWatchdog != 0) { 4352 mAudioWatchdog->pause(); 4353 } 4354#endif 4355 4356 // Now perform the deferred reset on fast tracks that have stopped 4357 while (resetMask != 0) { 4358 size_t i = __builtin_ctz(resetMask); 4359 ALOG_ASSERT(i < count); 4360 resetMask &= ~(1 << i); 4361 sp<Track> t = mActiveTracks[i].promote(); 4362 if (t == 0) { 4363 continue; 4364 } 4365 Track* track = t.get(); 4366 ALOG_ASSERT(track->isFastTrack() && track->isStopped()); 4367 track->reset(); 4368 } 4369 4370 // remove all the tracks that need to be... 4371 removeTracks_l(*tracksToRemove); 4372 4373 if (getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX) != 0) { 4374 mEffectBufferValid = true; 4375 } 4376 4377 if (mEffectBufferValid) { 4378 // as long as there are effects we should clear the effects buffer, to avoid 4379 // passing a non-clean buffer to the effect chain 4380 memset(mEffectBuffer, 0, mEffectBufferSize); 4381 } 4382 // sink or mix buffer must be cleared if all tracks are connected to an 4383 // effect chain as in this case the mixer will not write to the sink or mix buffer 4384 // and track effects will accumulate into it 4385 if ((mBytesRemaining == 0) && ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 4386 (mixedTracks == 0 && fastTracks > 0))) { 4387 // FIXME as a performance optimization, should remember previous zero status 4388 if (mMixerBufferValid) { 4389 memset(mMixerBuffer, 0, mMixerBufferSize); 4390 // TODO: In testing, mSinkBuffer below need not be cleared because 4391 // the PlaybackThread::threadLoop() copies mMixerBuffer into mSinkBuffer 4392 // after mixing. 4393 // 4394 // To enforce this guarantee: 4395 // ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 4396 // (mixedTracks == 0 && fastTracks > 0)) 4397 // must imply MIXER_TRACKS_READY. 4398 // Later, we may clear buffers regardless, and skip much of this logic. 4399 } 4400 // FIXME as a performance optimization, should remember previous zero status 4401 memset(mSinkBuffer, 0, mNormalFrameCount * mFrameSize); 4402 } 4403 4404 // if any fast tracks, then status is ready 4405 mMixerStatusIgnoringFastTracks = mixerStatus; 4406 if (fastTracks > 0) { 4407 mixerStatus = MIXER_TRACKS_READY; 4408 } 4409 return mixerStatus; 4410} 4411 4412// getTrackName_l() must be called with ThreadBase::mLock held 4413int AudioFlinger::MixerThread::getTrackName_l(audio_channel_mask_t channelMask, 4414 audio_format_t format, audio_session_t sessionId) 4415{ 4416 return mAudioMixer->getTrackName(channelMask, format, sessionId); 4417} 4418 4419// deleteTrackName_l() must be called with ThreadBase::mLock held 4420void AudioFlinger::MixerThread::deleteTrackName_l(int name) 4421{ 4422 ALOGV("remove track (%d) and delete from mixer", name); 4423 mAudioMixer->deleteTrackName(name); 4424} 4425 4426// checkForNewParameter_l() must be called with ThreadBase::mLock held 4427bool AudioFlinger::MixerThread::checkForNewParameter_l(const String8& keyValuePair, 4428 status_t& status) 4429{ 4430 bool reconfig = false; 4431 bool a2dpDeviceChanged = false; 4432 4433 status = NO_ERROR; 4434 4435 AutoPark<FastMixer> park(mFastMixer); 4436 4437 AudioParameter param = AudioParameter(keyValuePair); 4438 int value; 4439 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 4440 reconfig = true; 4441 } 4442 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 4443 if (!isValidPcmSinkFormat((audio_format_t) value)) { 4444 status = BAD_VALUE; 4445 } else { 4446 // no need to save value, since it's constant 4447 reconfig = true; 4448 } 4449 } 4450 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 4451 if (!isValidPcmSinkChannelMask((audio_channel_mask_t) value)) { 4452 status = BAD_VALUE; 4453 } else { 4454 // no need to save value, since it's constant 4455 reconfig = true; 4456 } 4457 } 4458 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 4459 // do not accept frame count changes if tracks are open as the track buffer 4460 // size depends on frame count and correct behavior would not be guaranteed 4461 // if frame count is changed after track creation 4462 if (!mTracks.isEmpty()) { 4463 status = INVALID_OPERATION; 4464 } else { 4465 reconfig = true; 4466 } 4467 } 4468 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 4469#ifdef ADD_BATTERY_DATA 4470 // when changing the audio output device, call addBatteryData to notify 4471 // the change 4472 if (mOutDevice != value) { 4473 uint32_t params = 0; 4474 // check whether speaker is on 4475 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 4476 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 4477 } 4478 4479 audio_devices_t deviceWithoutSpeaker 4480 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 4481 // check if any other device (except speaker) is on 4482 if (value & deviceWithoutSpeaker) { 4483 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 4484 } 4485 4486 if (params != 0) { 4487 addBatteryData(params); 4488 } 4489 } 4490#endif 4491 4492 // forward device change to effects that have requested to be 4493 // aware of attached audio device. 4494 if (value != AUDIO_DEVICE_NONE) { 4495 a2dpDeviceChanged = 4496 (mOutDevice & AUDIO_DEVICE_OUT_ALL_A2DP) != (value & AUDIO_DEVICE_OUT_ALL_A2DP); 4497 mOutDevice = value; 4498 for (size_t i = 0; i < mEffectChains.size(); i++) { 4499 mEffectChains[i]->setDevice_l(mOutDevice); 4500 } 4501 } 4502 } 4503 4504 if (status == NO_ERROR) { 4505 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 4506 keyValuePair.string()); 4507 if (!mStandby && status == INVALID_OPERATION) { 4508 mOutput->standby(); 4509 mStandby = true; 4510 mBytesWritten = 0; 4511 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 4512 keyValuePair.string()); 4513 } 4514 if (status == NO_ERROR && reconfig) { 4515 readOutputParameters_l(); 4516 delete mAudioMixer; 4517 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 4518 for (size_t i = 0; i < mTracks.size() ; i++) { 4519 int name = getTrackName_l(mTracks[i]->mChannelMask, 4520 mTracks[i]->mFormat, mTracks[i]->mSessionId); 4521 if (name < 0) { 4522 break; 4523 } 4524 mTracks[i]->mName = name; 4525 } 4526 sendIoConfigEvent_l(AUDIO_OUTPUT_CONFIG_CHANGED); 4527 } 4528 } 4529 4530 return reconfig || a2dpDeviceChanged; 4531} 4532 4533 4534void AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 4535{ 4536 PlaybackThread::dumpInternals(fd, args); 4537 dprintf(fd, " Thread throttle time (msecs): %u\n", mThreadThrottleTimeMs); 4538 dprintf(fd, " AudioMixer tracks: 0x%08x\n", mAudioMixer->trackNames()); 4539 dprintf(fd, " Master mono: %s\n", mMasterMono ? "on" : "off"); 4540 4541 // Make a non-atomic copy of fast mixer dump state so it won't change underneath us 4542 // while we are dumping it. It may be inconsistent, but it won't mutate! 4543 // This is a large object so we place it on the heap. 4544 // FIXME 25972958: Need an intelligent copy constructor that does not touch unused pages. 4545 const FastMixerDumpState *copy = new FastMixerDumpState(mFastMixerDumpState); 4546 copy->dump(fd); 4547 delete copy; 4548 4549#ifdef STATE_QUEUE_DUMP 4550 // Similar for state queue 4551 StateQueueObserverDump observerCopy = mStateQueueObserverDump; 4552 observerCopy.dump(fd); 4553 StateQueueMutatorDump mutatorCopy = mStateQueueMutatorDump; 4554 mutatorCopy.dump(fd); 4555#endif 4556 4557#ifdef TEE_SINK 4558 // Write the tee output to a .wav file 4559 dumpTee(fd, mTeeSource, mId); 4560#endif 4561 4562#ifdef AUDIO_WATCHDOG 4563 if (mAudioWatchdog != 0) { 4564 // Make a non-atomic copy of audio watchdog dump so it won't change underneath us 4565 AudioWatchdogDump wdCopy = mAudioWatchdogDump; 4566 wdCopy.dump(fd); 4567 } 4568#endif 4569} 4570 4571uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const 4572{ 4573 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000) / 2; 4574} 4575 4576uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() const 4577{ 4578 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000); 4579} 4580 4581void AudioFlinger::MixerThread::cacheParameters_l() 4582{ 4583 PlaybackThread::cacheParameters_l(); 4584 4585 // FIXME: Relaxed timing because of a certain device that can't meet latency 4586 // Should be reduced to 2x after the vendor fixes the driver issue 4587 // increase threshold again due to low power audio mode. The way this warning 4588 // threshold is calculated and its usefulness should be reconsidered anyway. 4589 maxPeriod = seconds(mNormalFrameCount) / mSampleRate * 15; 4590} 4591 4592// ---------------------------------------------------------------------------- 4593 4594AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 4595 AudioStreamOut* output, audio_io_handle_t id, audio_devices_t device, bool systemReady, 4596 uint32_t bitRate) 4597 : PlaybackThread(audioFlinger, output, id, device, DIRECT, systemReady, bitRate) 4598 // mLeftVolFloat, mRightVolFloat 4599{ 4600} 4601 4602AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 4603 AudioStreamOut* output, audio_io_handle_t id, uint32_t device, 4604 ThreadBase::type_t type, bool systemReady, uint32_t bitRate) 4605 : PlaybackThread(audioFlinger, output, id, device, type, systemReady, bitRate) 4606 // mLeftVolFloat, mRightVolFloat 4607{ 4608} 4609 4610AudioFlinger::DirectOutputThread::~DirectOutputThread() 4611{ 4612} 4613 4614void AudioFlinger::DirectOutputThread::processVolume_l(Track *track, bool lastTrack) 4615{ 4616 float left, right; 4617 4618 if (mMasterMute || mStreamTypes[track->streamType()].mute) { 4619 left = right = 0; 4620 } else { 4621 float typeVolume = mStreamTypes[track->streamType()].volume; 4622 float v = mMasterVolume * typeVolume; 4623 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 4624 gain_minifloat_packed_t vlr = proxy->getVolumeLR(); 4625 left = float_from_gain(gain_minifloat_unpack_left(vlr)); 4626 if (left > GAIN_FLOAT_UNITY) { 4627 left = GAIN_FLOAT_UNITY; 4628 } 4629 left *= v; 4630 right = float_from_gain(gain_minifloat_unpack_right(vlr)); 4631 if (right > GAIN_FLOAT_UNITY) { 4632 right = GAIN_FLOAT_UNITY; 4633 } 4634 right *= v; 4635 } 4636 4637 if (lastTrack) { 4638 if (left != mLeftVolFloat || right != mRightVolFloat) { 4639 mLeftVolFloat = left; 4640 mRightVolFloat = right; 4641 4642 // Convert volumes from float to 8.24 4643 uint32_t vl = (uint32_t)(left * (1 << 24)); 4644 uint32_t vr = (uint32_t)(right * (1 << 24)); 4645 4646 // Delegate volume control to effect in track effect chain if needed 4647 // only one effect chain can be present on DirectOutputThread, so if 4648 // there is one, the track is connected to it 4649 if (!mEffectChains.isEmpty()) { 4650 mEffectChains[0]->setVolume_l(&vl, &vr); 4651 left = (float)vl / (1 << 24); 4652 right = (float)vr / (1 << 24); 4653 } 4654 if (mOutput->stream->set_volume) { 4655 mOutput->stream->set_volume(mOutput->stream, left, right); 4656 } 4657 } 4658 } 4659} 4660 4661void AudioFlinger::DirectOutputThread::onAddNewTrack_l() 4662{ 4663 sp<Track> previousTrack = mPreviousTrack.promote(); 4664 sp<Track> latestTrack = mLatestActiveTrack.promote(); 4665 4666 if (previousTrack != 0 && latestTrack != 0) { 4667 if (mType == DIRECT) { 4668 if (previousTrack.get() != latestTrack.get()) { 4669 mFlushPending = true; 4670 } 4671 } else /* mType == OFFLOAD */ { 4672 if (previousTrack->sessionId() != latestTrack->sessionId()) { 4673 mFlushPending = true; 4674 } 4675 } 4676 } 4677 PlaybackThread::onAddNewTrack_l(); 4678} 4679 4680AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l( 4681 Vector< sp<Track> > *tracksToRemove 4682) 4683{ 4684 size_t count = mActiveTracks.size(); 4685 mixer_state mixerStatus = MIXER_IDLE; 4686 bool doHwPause = false; 4687 bool doHwResume = false; 4688 4689 // find out which tracks need to be processed 4690 for (size_t i = 0; i < count; i++) { 4691 sp<Track> t = mActiveTracks[i].promote(); 4692 // The track died recently 4693 if (t == 0) { 4694 continue; 4695 } 4696 4697 if (t->isInvalid()) { 4698 ALOGW("An invalidated track shouldn't be in active list"); 4699 tracksToRemove->add(t); 4700 continue; 4701 } 4702 4703 Track* const track = t.get(); 4704#ifdef VERY_VERY_VERBOSE_LOGGING 4705 audio_track_cblk_t* cblk = track->cblk(); 4706#endif 4707 // Only consider last track started for volume and mixer state control. 4708 // In theory an older track could underrun and restart after the new one starts 4709 // but as we only care about the transition phase between two tracks on a 4710 // direct output, it is not a problem to ignore the underrun case. 4711 sp<Track> l = mLatestActiveTrack.promote(); 4712 bool last = l.get() == track; 4713 4714 if (track->isPausing()) { 4715 track->setPaused(); 4716 if (mHwSupportsPause && last && !mHwPaused) { 4717 doHwPause = true; 4718 mHwPaused = true; 4719 } 4720 tracksToRemove->add(track); 4721 } else if (track->isFlushPending()) { 4722 track->flushAck(); 4723 if (last) { 4724 mFlushPending = true; 4725 } 4726 } else if (track->isResumePending()) { 4727 track->resumeAck(); 4728 if (last && mHwPaused) { 4729 doHwResume = true; 4730 mHwPaused = false; 4731 } 4732 } 4733 4734 // The first time a track is added we wait 4735 // for all its buffers to be filled before processing it. 4736 // Allow draining the buffer in case the client 4737 // app does not call stop() and relies on underrun to stop: 4738 // hence the test on (track->mRetryCount > 1). 4739 // If retryCount<=1 then track is about to underrun and be removed. 4740 // Do not use a high threshold for compressed audio. 4741 uint32_t minFrames; 4742 if ((track->sharedBuffer() == 0) && !track->isStopping_1() && !track->isPausing() 4743 && (track->mRetryCount > 1) && audio_has_proportional_frames(mFormat)) { 4744 minFrames = mNormalFrameCount; 4745 } else { 4746 minFrames = 1; 4747 } 4748 4749 if ((track->framesReady() >= minFrames) && track->isReady() && !track->isPaused() && 4750 !track->isStopping_2() && !track->isStopped()) 4751 { 4752 ALOGVV("track %d s=%08x [OK]", track->name(), cblk->mServer); 4753 4754 if (track->mFillingUpStatus == Track::FS_FILLED) { 4755 track->mFillingUpStatus = Track::FS_ACTIVE; 4756 // make sure processVolume_l() will apply new volume even if 0 4757 mLeftVolFloat = mRightVolFloat = -1.0; 4758 if (!mHwSupportsPause) { 4759 track->resumeAck(); 4760 } 4761 } 4762 4763 // compute volume for this track 4764 processVolume_l(track, last); 4765 if (last) { 4766 sp<Track> previousTrack = mPreviousTrack.promote(); 4767 if (previousTrack != 0) { 4768 if (track != previousTrack.get()) { 4769 // Flush any data still being written from last track 4770 mBytesRemaining = 0; 4771 // Invalidate previous track to force a seek when resuming. 4772 previousTrack->invalidate(); 4773 } 4774 } 4775 mPreviousTrack = track; 4776 4777 // reset retry count 4778 track->mRetryCount = kMaxTrackRetriesDirect; 4779 mActiveTrack = t; 4780 mixerStatus = MIXER_TRACKS_READY; 4781 if (mHwPaused) { 4782 doHwResume = true; 4783 mHwPaused = false; 4784 } 4785 } 4786 } else { 4787 // clear effect chain input buffer if the last active track started underruns 4788 // to avoid sending previous audio buffer again to effects 4789 if (!mEffectChains.isEmpty() && last) { 4790 mEffectChains[0]->clearInputBuffer(); 4791 } 4792 if (track->isStopping_1()) { 4793 track->mState = TrackBase::STOPPING_2; 4794 if (last && mHwPaused) { 4795 doHwResume = true; 4796 mHwPaused = false; 4797 } 4798 } 4799 if ((track->sharedBuffer() != 0) || track->isStopped() || 4800 track->isStopping_2() || track->isPaused()) { 4801 // We have consumed all the buffers of this track. 4802 // Remove it from the list of active tracks. 4803 size_t audioHALFrames; 4804 if (audio_has_proportional_frames(mFormat)) { 4805 audioHALFrames = (latency_l() * mSampleRate) / 1000; 4806 } else { 4807 audioHALFrames = 0; 4808 } 4809 4810 int64_t framesWritten = mBytesWritten / mFrameSize; 4811 if (mStandby || !last || 4812 track->presentationComplete(framesWritten, audioHALFrames)) { 4813 if (track->isStopping_2()) { 4814 track->mState = TrackBase::STOPPED; 4815 } 4816 if (track->isStopped()) { 4817 track->reset(); 4818 } 4819 tracksToRemove->add(track); 4820 } 4821 } else { 4822 // No buffers for this track. Give it a few chances to 4823 // fill a buffer, then remove it from active list. 4824 // Only consider last track started for mixer state control 4825 if (--(track->mRetryCount) <= 0) { 4826 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 4827 tracksToRemove->add(track); 4828 // indicate to client process that the track was disabled because of underrun; 4829 // it will then automatically call start() when data is available 4830 track->disable(); 4831 } else if (last) { 4832 ALOGW("pause because of UNDERRUN, framesReady = %zu," 4833 "minFrames = %u, mFormat = %#x", 4834 track->framesReady(), minFrames, mFormat); 4835 mixerStatus = MIXER_TRACKS_ENABLED; 4836 if (mHwSupportsPause && !mHwPaused && !mStandby) { 4837 doHwPause = true; 4838 mHwPaused = true; 4839 } 4840 } 4841 } 4842 } 4843 } 4844 4845 // if an active track did not command a flush, check for pending flush on stopped tracks 4846 if (!mFlushPending) { 4847 for (size_t i = 0; i < mTracks.size(); i++) { 4848 if (mTracks[i]->isFlushPending()) { 4849 mTracks[i]->flushAck(); 4850 mFlushPending = true; 4851 } 4852 } 4853 } 4854 4855 // make sure the pause/flush/resume sequence is executed in the right order. 4856 // If a flush is pending and a track is active but the HW is not paused, force a HW pause 4857 // before flush and then resume HW. This can happen in case of pause/flush/resume 4858 // if resume is received before pause is executed. 4859 if (mHwSupportsPause && !mStandby && 4860 (doHwPause || (mFlushPending && !mHwPaused && (count != 0)))) { 4861 mOutput->stream->pause(mOutput->stream); 4862 } 4863 if (mFlushPending) { 4864 flushHw_l(); 4865 } 4866 if (mHwSupportsPause && !mStandby && doHwResume) { 4867 mOutput->stream->resume(mOutput->stream); 4868 } 4869 // remove all the tracks that need to be... 4870 removeTracks_l(*tracksToRemove); 4871 4872 return mixerStatus; 4873} 4874 4875void AudioFlinger::DirectOutputThread::threadLoop_mix() 4876{ 4877 size_t frameCount = mFrameCount; 4878 int8_t *curBuf = (int8_t *)mSinkBuffer; 4879 // output audio to hardware 4880 while (frameCount) { 4881 AudioBufferProvider::Buffer buffer; 4882 buffer.frameCount = frameCount; 4883 status_t status = mActiveTrack->getNextBuffer(&buffer); 4884 if (status != NO_ERROR || buffer.raw == NULL) { 4885 // no need to pad with 0 for compressed audio 4886 if (audio_has_proportional_frames(mFormat)) { 4887 memset(curBuf, 0, frameCount * mFrameSize); 4888 } 4889 break; 4890 } 4891 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 4892 frameCount -= buffer.frameCount; 4893 curBuf += buffer.frameCount * mFrameSize; 4894 mActiveTrack->releaseBuffer(&buffer); 4895 } 4896 mCurrentWriteLength = curBuf - (int8_t *)mSinkBuffer; 4897 mSleepTimeUs = 0; 4898 mStandbyTimeNs = systemTime() + mStandbyDelayNs; 4899 mActiveTrack.clear(); 4900} 4901 4902void AudioFlinger::DirectOutputThread::threadLoop_sleepTime() 4903{ 4904 // do not write to HAL when paused 4905 if (mHwPaused || (usesHwAvSync() && mStandby)) { 4906 mSleepTimeUs = mIdleSleepTimeUs; 4907 return; 4908 } 4909 if (mSleepTimeUs == 0) { 4910 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4911 // For compressed offload, use faster sleep time when underruning until more than an 4912 // entire buffer was written to the audio HAL 4913 if (!audio_has_proportional_frames(mFormat) && 4914 (mType == OFFLOAD) && (mBytesWritten < (int64_t) mBufferSize)) { 4915 mSleepTimeUs = kDirectMinSleepTimeUs; 4916 } else { 4917 mSleepTimeUs = mActiveSleepTimeUs; 4918 } 4919 } else { 4920 mSleepTimeUs = mIdleSleepTimeUs; 4921 } 4922 } else if (mBytesWritten != 0 && audio_has_proportional_frames(mFormat)) { 4923 memset(mSinkBuffer, 0, mFrameCount * mFrameSize); 4924 mSleepTimeUs = 0; 4925 } 4926} 4927 4928void AudioFlinger::DirectOutputThread::threadLoop_exit() 4929{ 4930 { 4931 Mutex::Autolock _l(mLock); 4932 for (size_t i = 0; i < mTracks.size(); i++) { 4933 if (mTracks[i]->isFlushPending()) { 4934 mTracks[i]->flushAck(); 4935 mFlushPending = true; 4936 } 4937 } 4938 if (mFlushPending) { 4939 flushHw_l(); 4940 } 4941 } 4942 PlaybackThread::threadLoop_exit(); 4943} 4944 4945// must be called with thread mutex locked 4946bool AudioFlinger::DirectOutputThread::shouldStandby_l() 4947{ 4948 bool trackPaused = false; 4949 bool trackStopped = false; 4950 4951 if ((mType == DIRECT) && audio_is_linear_pcm(mFormat) && !usesHwAvSync()) { 4952 return !mStandby; 4953 } 4954 4955 // do not put the HAL in standby when paused. AwesomePlayer clear the offloaded AudioTrack 4956 // after a timeout and we will enter standby then. 4957 if (mTracks.size() > 0) { 4958 trackPaused = mTracks[mTracks.size() - 1]->isPaused(); 4959 trackStopped = mTracks[mTracks.size() - 1]->isStopped() || 4960 mTracks[mTracks.size() - 1]->mState == TrackBase::IDLE; 4961 } 4962 4963 return !mStandby && !(trackPaused || (mHwPaused && !trackStopped)); 4964} 4965 4966// getTrackName_l() must be called with ThreadBase::mLock held 4967int AudioFlinger::DirectOutputThread::getTrackName_l(audio_channel_mask_t channelMask __unused, 4968 audio_format_t format __unused, audio_session_t sessionId __unused) 4969{ 4970 return 0; 4971} 4972 4973// deleteTrackName_l() must be called with ThreadBase::mLock held 4974void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name __unused) 4975{ 4976} 4977 4978// checkForNewParameter_l() must be called with ThreadBase::mLock held 4979bool AudioFlinger::DirectOutputThread::checkForNewParameter_l(const String8& keyValuePair, 4980 status_t& status) 4981{ 4982 bool reconfig = false; 4983 bool a2dpDeviceChanged = false; 4984 4985 status = NO_ERROR; 4986 4987 AudioParameter param = AudioParameter(keyValuePair); 4988 int value; 4989 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 4990 // forward device change to effects that have requested to be 4991 // aware of attached audio device. 4992 if (value != AUDIO_DEVICE_NONE) { 4993 a2dpDeviceChanged = 4994 (mOutDevice & AUDIO_DEVICE_OUT_ALL_A2DP) != (value & AUDIO_DEVICE_OUT_ALL_A2DP); 4995 mOutDevice = value; 4996 for (size_t i = 0; i < mEffectChains.size(); i++) { 4997 mEffectChains[i]->setDevice_l(mOutDevice); 4998 } 4999 } 5000 } 5001 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 5002 // do not accept frame count changes if tracks are open as the track buffer 5003 // size depends on frame count and correct behavior would not be garantied 5004 // if frame count is changed after track creation 5005 if (!mTracks.isEmpty()) { 5006 status = INVALID_OPERATION; 5007 } else { 5008 reconfig = true; 5009 } 5010 } 5011 if (status == NO_ERROR) { 5012 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 5013 keyValuePair.string()); 5014 if (!mStandby && status == INVALID_OPERATION) { 5015 mOutput->standby(); 5016 mStandby = true; 5017 mBytesWritten = 0; 5018 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 5019 keyValuePair.string()); 5020 } 5021 if (status == NO_ERROR && reconfig) { 5022 readOutputParameters_l(); 5023 sendIoConfigEvent_l(AUDIO_OUTPUT_CONFIG_CHANGED); 5024 } 5025 } 5026 5027 return reconfig || a2dpDeviceChanged; 5028} 5029 5030uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() const 5031{ 5032 uint32_t time; 5033 if (audio_has_proportional_frames(mFormat)) { 5034 time = PlaybackThread::activeSleepTimeUs(); 5035 } else { 5036 time = kDirectMinSleepTimeUs; 5037 } 5038 return time; 5039} 5040 5041uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() const 5042{ 5043 uint32_t time; 5044 if (audio_has_proportional_frames(mFormat)) { 5045 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 5046 } else { 5047 time = kDirectMinSleepTimeUs; 5048 } 5049 return time; 5050} 5051 5052uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() const 5053{ 5054 uint32_t time; 5055 if (audio_has_proportional_frames(mFormat)) { 5056 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 5057 } else { 5058 time = kDirectMinSleepTimeUs; 5059 } 5060 return time; 5061} 5062 5063void AudioFlinger::DirectOutputThread::cacheParameters_l() 5064{ 5065 PlaybackThread::cacheParameters_l(); 5066 5067 // use shorter standby delay as on normal output to release 5068 // hardware resources as soon as possible 5069 // no delay on outputs with HW A/V sync 5070 if (usesHwAvSync()) { 5071 mStandbyDelayNs = 0; 5072 } else if ((mType == OFFLOAD) && !audio_has_proportional_frames(mFormat)) { 5073 mStandbyDelayNs = kOffloadStandbyDelayNs; 5074 } else { 5075 mStandbyDelayNs = microseconds(mActiveSleepTimeUs*2); 5076 } 5077} 5078 5079void AudioFlinger::DirectOutputThread::flushHw_l() 5080{ 5081 mOutput->flush(); 5082 mHwPaused = false; 5083 mFlushPending = false; 5084} 5085 5086// ---------------------------------------------------------------------------- 5087 5088AudioFlinger::AsyncCallbackThread::AsyncCallbackThread( 5089 const wp<AudioFlinger::PlaybackThread>& playbackThread) 5090 : Thread(false /*canCallJava*/), 5091 mPlaybackThread(playbackThread), 5092 mWriteAckSequence(0), 5093 mDrainSequence(0) 5094{ 5095} 5096 5097AudioFlinger::AsyncCallbackThread::~AsyncCallbackThread() 5098{ 5099} 5100 5101void AudioFlinger::AsyncCallbackThread::onFirstRef() 5102{ 5103 run("Offload Cbk", ANDROID_PRIORITY_URGENT_AUDIO); 5104} 5105 5106bool AudioFlinger::AsyncCallbackThread::threadLoop() 5107{ 5108 while (!exitPending()) { 5109 uint32_t writeAckSequence; 5110 uint32_t drainSequence; 5111 5112 { 5113 Mutex::Autolock _l(mLock); 5114 while (!((mWriteAckSequence & 1) || 5115 (mDrainSequence & 1) || 5116 exitPending())) { 5117 mWaitWorkCV.wait(mLock); 5118 } 5119 5120 if (exitPending()) { 5121 break; 5122 } 5123 ALOGV("AsyncCallbackThread mWriteAckSequence %d mDrainSequence %d", 5124 mWriteAckSequence, mDrainSequence); 5125 writeAckSequence = mWriteAckSequence; 5126 mWriteAckSequence &= ~1; 5127 drainSequence = mDrainSequence; 5128 mDrainSequence &= ~1; 5129 } 5130 { 5131 sp<AudioFlinger::PlaybackThread> playbackThread = mPlaybackThread.promote(); 5132 if (playbackThread != 0) { 5133 if (writeAckSequence & 1) { 5134 playbackThread->resetWriteBlocked(writeAckSequence >> 1); 5135 } 5136 if (drainSequence & 1) { 5137 playbackThread->resetDraining(drainSequence >> 1); 5138 } 5139 } 5140 } 5141 } 5142 return false; 5143} 5144 5145void AudioFlinger::AsyncCallbackThread::exit() 5146{ 5147 ALOGV("AsyncCallbackThread::exit"); 5148 Mutex::Autolock _l(mLock); 5149 requestExit(); 5150 mWaitWorkCV.broadcast(); 5151} 5152 5153void AudioFlinger::AsyncCallbackThread::setWriteBlocked(uint32_t sequence) 5154{ 5155 Mutex::Autolock _l(mLock); 5156 // bit 0 is cleared 5157 mWriteAckSequence = sequence << 1; 5158} 5159 5160void AudioFlinger::AsyncCallbackThread::resetWriteBlocked() 5161{ 5162 Mutex::Autolock _l(mLock); 5163 // ignore unexpected callbacks 5164 if (mWriteAckSequence & 2) { 5165 mWriteAckSequence |= 1; 5166 mWaitWorkCV.signal(); 5167 } 5168} 5169 5170void AudioFlinger::AsyncCallbackThread::setDraining(uint32_t sequence) 5171{ 5172 Mutex::Autolock _l(mLock); 5173 // bit 0 is cleared 5174 mDrainSequence = sequence << 1; 5175} 5176 5177void AudioFlinger::AsyncCallbackThread::resetDraining() 5178{ 5179 Mutex::Autolock _l(mLock); 5180 // ignore unexpected callbacks 5181 if (mDrainSequence & 2) { 5182 mDrainSequence |= 1; 5183 mWaitWorkCV.signal(); 5184 } 5185} 5186 5187 5188// ---------------------------------------------------------------------------- 5189AudioFlinger::OffloadThread::OffloadThread(const sp<AudioFlinger>& audioFlinger, 5190 AudioStreamOut* output, audio_io_handle_t id, uint32_t device, bool systemReady, 5191 uint32_t bitRate) 5192 : DirectOutputThread(audioFlinger, output, id, device, OFFLOAD, systemReady, bitRate), 5193 mPausedWriteLength(0), mPausedBytesRemaining(0), mKeepWakeLock(true) 5194{ 5195 //FIXME: mStandby should be set to true by ThreadBase constructor 5196 mStandby = true; 5197 mKeepWakeLock = property_get_bool("ro.audio.offload_wakelock", true /* default_value */); 5198} 5199 5200void AudioFlinger::OffloadThread::threadLoop_exit() 5201{ 5202 if (mFlushPending || mHwPaused) { 5203 // If a flush is pending or track was paused, just discard buffered data 5204 flushHw_l(); 5205 } else { 5206 mMixerStatus = MIXER_DRAIN_ALL; 5207 threadLoop_drain(); 5208 } 5209 if (mUseAsyncWrite) { 5210 ALOG_ASSERT(mCallbackThread != 0); 5211 mCallbackThread->exit(); 5212 } 5213 PlaybackThread::threadLoop_exit(); 5214} 5215 5216AudioFlinger::PlaybackThread::mixer_state AudioFlinger::OffloadThread::prepareTracks_l( 5217 Vector< sp<Track> > *tracksToRemove 5218) 5219{ 5220 size_t count = mActiveTracks.size(); 5221 5222 mixer_state mixerStatus = MIXER_IDLE; 5223 bool doHwPause = false; 5224 bool doHwResume = false; 5225 5226 ALOGV("OffloadThread::prepareTracks_l active tracks %zu", count); 5227 5228 // find out which tracks need to be processed 5229 for (size_t i = 0; i < count; i++) { 5230 sp<Track> t = mActiveTracks[i].promote(); 5231 // The track died recently 5232 if (t == 0) { 5233 continue; 5234 } 5235 Track* const track = t.get(); 5236#ifdef VERY_VERY_VERBOSE_LOGGING 5237 audio_track_cblk_t* cblk = track->cblk(); 5238#endif 5239 // Only consider last track started for volume and mixer state control. 5240 // In theory an older track could underrun and restart after the new one starts 5241 // but as we only care about the transition phase between two tracks on a 5242 // direct output, it is not a problem to ignore the underrun case. 5243 sp<Track> l = mLatestActiveTrack.promote(); 5244 bool last = l.get() == track; 5245 5246 if (track->isInvalid()) { 5247 ALOGW("An invalidated track shouldn't be in active list"); 5248 tracksToRemove->add(track); 5249 continue; 5250 } 5251 5252 if (track->mState == TrackBase::IDLE) { 5253 ALOGW("An idle track shouldn't be in active list"); 5254 continue; 5255 } 5256 5257 if (track->isPausing()) { 5258 track->setPaused(); 5259 if (last) { 5260 if (mHwSupportsPause && !mHwPaused) { 5261 doHwPause = true; 5262 mHwPaused = true; 5263 } 5264 // If we were part way through writing the mixbuffer to 5265 // the HAL we must save this until we resume 5266 // BUG - this will be wrong if a different track is made active, 5267 // in that case we want to discard the pending data in the 5268 // mixbuffer and tell the client to present it again when the 5269 // track is resumed 5270 mPausedWriteLength = mCurrentWriteLength; 5271 mPausedBytesRemaining = mBytesRemaining; 5272 mBytesRemaining = 0; // stop writing 5273 } 5274 tracksToRemove->add(track); 5275 } else if (track->isFlushPending()) { 5276 track->mRetryCount = kMaxTrackRetriesOffload; 5277 track->flushAck(); 5278 if (last) { 5279 mFlushPending = true; 5280 } 5281 } else if (track->isResumePending()){ 5282 track->resumeAck(); 5283 if (last) { 5284 if (mPausedBytesRemaining) { 5285 // Need to continue write that was interrupted 5286 mCurrentWriteLength = mPausedWriteLength; 5287 mBytesRemaining = mPausedBytesRemaining; 5288 mPausedBytesRemaining = 0; 5289 } 5290 if (mHwPaused) { 5291 doHwResume = true; 5292 mHwPaused = false; 5293 // threadLoop_mix() will handle the case that we need to 5294 // resume an interrupted write 5295 } 5296 // enable write to audio HAL 5297 mSleepTimeUs = 0; 5298 5299 // Do not handle new data in this iteration even if track->framesReady() 5300 mixerStatus = MIXER_TRACKS_ENABLED; 5301 } 5302 } else if (track->framesReady() && track->isReady() && 5303 !track->isPaused() && !track->isTerminated() && !track->isStopping_2()) { 5304 ALOGVV("OffloadThread: track %d s=%08x [OK]", track->name(), cblk->mServer); 5305 if (track->mFillingUpStatus == Track::FS_FILLED) { 5306 track->mFillingUpStatus = Track::FS_ACTIVE; 5307 // make sure processVolume_l() will apply new volume even if 0 5308 mLeftVolFloat = mRightVolFloat = -1.0; 5309 } 5310 5311 if (last) { 5312 sp<Track> previousTrack = mPreviousTrack.promote(); 5313 if (previousTrack != 0) { 5314 if (track != previousTrack.get()) { 5315 // Flush any data still being written from last track 5316 mBytesRemaining = 0; 5317 if (mPausedBytesRemaining) { 5318 // Last track was paused so we also need to flush saved 5319 // mixbuffer state and invalidate track so that it will 5320 // re-submit that unwritten data when it is next resumed 5321 mPausedBytesRemaining = 0; 5322 // Invalidate is a bit drastic - would be more efficient 5323 // to have a flag to tell client that some of the 5324 // previously written data was lost 5325 previousTrack->invalidate(); 5326 } 5327 // flush data already sent to the DSP if changing audio session as audio 5328 // comes from a different source. Also invalidate previous track to force a 5329 // seek when resuming. 5330 if (previousTrack->sessionId() != track->sessionId()) { 5331 previousTrack->invalidate(); 5332 } 5333 } 5334 } 5335 mPreviousTrack = track; 5336 // reset retry count 5337 track->mRetryCount = kMaxTrackRetriesOffload; 5338 mActiveTrack = t; 5339 mixerStatus = MIXER_TRACKS_READY; 5340 } 5341 } else { 5342 ALOGVV("OffloadThread: track %d s=%08x [NOT READY]", track->name(), cblk->mServer); 5343 if (track->isStopping_1()) { 5344 // Hardware buffer can hold a large amount of audio so we must 5345 // wait for all current track's data to drain before we say 5346 // that the track is stopped. 5347 if (mBytesRemaining == 0) { 5348 // Only start draining when all data in mixbuffer 5349 // has been written 5350 ALOGV("OffloadThread: underrun and STOPPING_1 -> draining, STOPPING_2"); 5351 track->mState = TrackBase::STOPPING_2; // so presentation completes after drain 5352 // do not drain if no data was ever sent to HAL (mStandby == true) 5353 if (last && !mStandby) { 5354 // do not modify drain sequence if we are already draining. This happens 5355 // when resuming from pause after drain. 5356 if ((mDrainSequence & 1) == 0) { 5357 mSleepTimeUs = 0; 5358 mStandbyTimeNs = systemTime() + mStandbyDelayNs; 5359 mixerStatus = MIXER_DRAIN_TRACK; 5360 mDrainSequence += 2; 5361 } 5362 if (mHwPaused) { 5363 // It is possible to move from PAUSED to STOPPING_1 without 5364 // a resume so we must ensure hardware is running 5365 doHwResume = true; 5366 mHwPaused = false; 5367 } 5368 } 5369 } 5370 } else if (track->isStopping_2()) { 5371 // Drain has completed or we are in standby, signal presentation complete 5372 if (!(mDrainSequence & 1) || !last || mStandby) { 5373 track->mState = TrackBase::STOPPED; 5374 size_t audioHALFrames = 5375 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 5376 int64_t framesWritten = 5377 mBytesWritten / mOutput->getFrameSize(); 5378 track->presentationComplete(framesWritten, audioHALFrames); 5379 track->reset(); 5380 tracksToRemove->add(track); 5381 } 5382 } else { 5383 // No buffers for this track. Give it a few chances to 5384 // fill a buffer, then remove it from active list. 5385 if (--(track->mRetryCount) <= 0) { 5386 ALOGV("OffloadThread: BUFFER TIMEOUT: remove(%d) from active list", 5387 track->name()); 5388 tracksToRemove->add(track); 5389 // indicate to client process that the track was disabled because of underrun; 5390 // it will then automatically call start() when data is available 5391 track->disable(); 5392 } else if (last){ 5393 mixerStatus = MIXER_TRACKS_ENABLED; 5394 } 5395 } 5396 } 5397 // compute volume for this track 5398 processVolume_l(track, last); 5399 } 5400 5401 // make sure the pause/flush/resume sequence is executed in the right order. 5402 // If a flush is pending and a track is active but the HW is not paused, force a HW pause 5403 // before flush and then resume HW. This can happen in case of pause/flush/resume 5404 // if resume is received before pause is executed. 5405 if (!mStandby && (doHwPause || (mFlushPending && !mHwPaused && (count != 0)))) { 5406 mOutput->stream->pause(mOutput->stream); 5407 } 5408 if (mFlushPending) { 5409 flushHw_l(); 5410 } 5411 if (!mStandby && doHwResume) { 5412 mOutput->stream->resume(mOutput->stream); 5413 } 5414 5415 // remove all the tracks that need to be... 5416 removeTracks_l(*tracksToRemove); 5417 5418 return mixerStatus; 5419} 5420 5421// must be called with thread mutex locked 5422bool AudioFlinger::OffloadThread::waitingAsyncCallback_l() 5423{ 5424 ALOGVV("waitingAsyncCallback_l mWriteAckSequence %d mDrainSequence %d", 5425 mWriteAckSequence, mDrainSequence); 5426 if (mUseAsyncWrite && ((mWriteAckSequence & 1) || (mDrainSequence & 1))) { 5427 return true; 5428 } 5429 return false; 5430} 5431 5432bool AudioFlinger::OffloadThread::waitingAsyncCallback() 5433{ 5434 Mutex::Autolock _l(mLock); 5435 return waitingAsyncCallback_l(); 5436} 5437 5438void AudioFlinger::OffloadThread::flushHw_l() 5439{ 5440 DirectOutputThread::flushHw_l(); 5441 // Flush anything still waiting in the mixbuffer 5442 mCurrentWriteLength = 0; 5443 mBytesRemaining = 0; 5444 mPausedWriteLength = 0; 5445 mPausedBytesRemaining = 0; 5446 // reset bytes written count to reflect that DSP buffers are empty after flush. 5447 mBytesWritten = 0; 5448 5449 if (mUseAsyncWrite) { 5450 // discard any pending drain or write ack by incrementing sequence 5451 mWriteAckSequence = (mWriteAckSequence + 2) & ~1; 5452 mDrainSequence = (mDrainSequence + 2) & ~1; 5453 ALOG_ASSERT(mCallbackThread != 0); 5454 mCallbackThread->setWriteBlocked(mWriteAckSequence); 5455 mCallbackThread->setDraining(mDrainSequence); 5456 } 5457} 5458 5459uint32_t AudioFlinger::OffloadThread::activeSleepTimeUs() const 5460{ 5461 uint32_t time; 5462 if (audio_has_proportional_frames(mFormat)) { 5463 time = PlaybackThread::activeSleepTimeUs(); 5464 } else { 5465 // sleep time is half the duration of an audio HAL buffer. 5466 // Note: This can be problematic in case of underrun with variable bit rate and 5467 // current rate is much less than initial rate. 5468 time = (uint32_t)max(kDirectMinSleepTimeUs, mBufferDurationUs / 2); 5469 } 5470 return time; 5471} 5472 5473void AudioFlinger::OffloadThread::invalidateTracks(audio_stream_type_t streamType) 5474{ 5475 Mutex::Autolock _l(mLock); 5476 mFlushPending = true; 5477 PlaybackThread::invalidateTracks_l(streamType); 5478} 5479 5480// ---------------------------------------------------------------------------- 5481 5482AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 5483 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id, bool systemReady) 5484 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->outDevice(), 5485 systemReady, DUPLICATING), 5486 mWaitTimeMs(UINT_MAX) 5487{ 5488 addOutputTrack(mainThread); 5489} 5490 5491AudioFlinger::DuplicatingThread::~DuplicatingThread() 5492{ 5493 for (size_t i = 0; i < mOutputTracks.size(); i++) { 5494 mOutputTracks[i]->destroy(); 5495 } 5496} 5497 5498void AudioFlinger::DuplicatingThread::threadLoop_mix() 5499{ 5500 // mix buffers... 5501 if (outputsReady(outputTracks)) { 5502 mAudioMixer->process(); 5503 } else { 5504 if (mMixerBufferValid) { 5505 memset(mMixerBuffer, 0, mMixerBufferSize); 5506 } else { 5507 memset(mSinkBuffer, 0, mSinkBufferSize); 5508 } 5509 } 5510 mSleepTimeUs = 0; 5511 writeFrames = mNormalFrameCount; 5512 mCurrentWriteLength = mSinkBufferSize; 5513 mStandbyTimeNs = systemTime() + mStandbyDelayNs; 5514} 5515 5516void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() 5517{ 5518 if (mSleepTimeUs == 0) { 5519 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 5520 mSleepTimeUs = mActiveSleepTimeUs; 5521 } else { 5522 mSleepTimeUs = mIdleSleepTimeUs; 5523 } 5524 } else if (mBytesWritten != 0) { 5525 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 5526 writeFrames = mNormalFrameCount; 5527 memset(mSinkBuffer, 0, mSinkBufferSize); 5528 } else { 5529 // flush remaining overflow buffers in output tracks 5530 writeFrames = 0; 5531 } 5532 mSleepTimeUs = 0; 5533 } 5534} 5535 5536ssize_t AudioFlinger::DuplicatingThread::threadLoop_write() 5537{ 5538 for (size_t i = 0; i < outputTracks.size(); i++) { 5539 outputTracks[i]->write(mSinkBuffer, writeFrames); 5540 } 5541 mStandby = false; 5542 return (ssize_t)mSinkBufferSize; 5543} 5544 5545void AudioFlinger::DuplicatingThread::threadLoop_standby() 5546{ 5547 // DuplicatingThread implements standby by stopping all tracks 5548 for (size_t i = 0; i < outputTracks.size(); i++) { 5549 outputTracks[i]->stop(); 5550 } 5551} 5552 5553void AudioFlinger::DuplicatingThread::saveOutputTracks() 5554{ 5555 outputTracks = mOutputTracks; 5556} 5557 5558void AudioFlinger::DuplicatingThread::clearOutputTracks() 5559{ 5560 outputTracks.clear(); 5561} 5562 5563void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 5564{ 5565 Mutex::Autolock _l(mLock); 5566 // The downstream MixerThread consumes thread->frameCount() amount of frames per mix pass. 5567 // Adjust for thread->sampleRate() to determine minimum buffer frame count. 5568 // Then triple buffer because Threads do not run synchronously and may not be clock locked. 5569 const size_t frameCount = 5570 3 * sourceFramesNeeded(mSampleRate, thread->frameCount(), thread->sampleRate()); 5571 // TODO: Consider asynchronous sample rate conversion to handle clock disparity 5572 // from different OutputTracks and their associated MixerThreads (e.g. one may 5573 // nearly empty and the other may be dropping data). 5574 5575 sp<OutputTrack> outputTrack = new OutputTrack(thread, 5576 this, 5577 mSampleRate, 5578 mFormat, 5579 mChannelMask, 5580 frameCount, 5581 IPCThreadState::self()->getCallingUid()); 5582 if (outputTrack->cblk() != NULL) { 5583 thread->setStreamVolume(AUDIO_STREAM_PATCH, 1.0f); 5584 mOutputTracks.add(outputTrack); 5585 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack.get(), thread); 5586 updateWaitTime_l(); 5587 } 5588} 5589 5590void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 5591{ 5592 Mutex::Autolock _l(mLock); 5593 for (size_t i = 0; i < mOutputTracks.size(); i++) { 5594 if (mOutputTracks[i]->thread() == thread) { 5595 mOutputTracks[i]->destroy(); 5596 mOutputTracks.removeAt(i); 5597 updateWaitTime_l(); 5598 if (thread->getOutput() == mOutput) { 5599 mOutput = NULL; 5600 } 5601 return; 5602 } 5603 } 5604 ALOGV("removeOutputTrack(): unknown thread: %p", thread); 5605} 5606 5607// caller must hold mLock 5608void AudioFlinger::DuplicatingThread::updateWaitTime_l() 5609{ 5610 mWaitTimeMs = UINT_MAX; 5611 for (size_t i = 0; i < mOutputTracks.size(); i++) { 5612 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 5613 if (strong != 0) { 5614 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 5615 if (waitTimeMs < mWaitTimeMs) { 5616 mWaitTimeMs = waitTimeMs; 5617 } 5618 } 5619 } 5620} 5621 5622 5623bool AudioFlinger::DuplicatingThread::outputsReady( 5624 const SortedVector< sp<OutputTrack> > &outputTracks) 5625{ 5626 for (size_t i = 0; i < outputTracks.size(); i++) { 5627 sp<ThreadBase> thread = outputTracks[i]->thread().promote(); 5628 if (thread == 0) { 5629 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", 5630 outputTracks[i].get()); 5631 return false; 5632 } 5633 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 5634 // see note at standby() declaration 5635 if (playbackThread->standby() && !playbackThread->isSuspended()) { 5636 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), 5637 thread.get()); 5638 return false; 5639 } 5640 } 5641 return true; 5642} 5643 5644uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() const 5645{ 5646 return (mWaitTimeMs * 1000) / 2; 5647} 5648 5649void AudioFlinger::DuplicatingThread::cacheParameters_l() 5650{ 5651 // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first 5652 updateWaitTime_l(); 5653 5654 MixerThread::cacheParameters_l(); 5655} 5656 5657// ---------------------------------------------------------------------------- 5658// Record 5659// ---------------------------------------------------------------------------- 5660 5661AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 5662 AudioStreamIn *input, 5663 audio_io_handle_t id, 5664 audio_devices_t outDevice, 5665 audio_devices_t inDevice, 5666 bool systemReady 5667#ifdef TEE_SINK 5668 , const sp<NBAIO_Sink>& teeSink 5669#endif 5670 ) : 5671 ThreadBase(audioFlinger, id, outDevice, inDevice, RECORD, systemReady), 5672 mInput(input), mActiveTracksGen(0), mRsmpInBuffer(NULL), 5673 // mRsmpInFrames and mRsmpInFramesP2 are set by readInputParameters_l() 5674 mRsmpInRear(0) 5675#ifdef TEE_SINK 5676 , mTeeSink(teeSink) 5677#endif 5678 , mReadOnlyHeap(new MemoryDealer(kRecordThreadReadOnlyHeapSize, 5679 "RecordThreadRO", MemoryHeapBase::READ_ONLY)) 5680 // mFastCapture below 5681 , mFastCaptureFutex(0) 5682 // mInputSource 5683 // mPipeSink 5684 // mPipeSource 5685 , mPipeFramesP2(0) 5686 // mPipeMemory 5687 // mFastCaptureNBLogWriter 5688 , mFastTrackAvail(false) 5689{ 5690 snprintf(mThreadName, kThreadNameLength, "AudioIn_%X", id); 5691 mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mThreadName); 5692 5693 readInputParameters_l(); 5694 5695 // create an NBAIO source for the HAL input stream, and negotiate 5696 mInputSource = new AudioStreamInSource(input->stream); 5697 size_t numCounterOffers = 0; 5698 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount, mFormat)}; 5699#if !LOG_NDEBUG 5700 ssize_t index = 5701#else 5702 (void) 5703#endif 5704 mInputSource->negotiate(offers, 1, NULL, numCounterOffers); 5705 ALOG_ASSERT(index == 0); 5706 5707 // initialize fast capture depending on configuration 5708 bool initFastCapture; 5709 switch (kUseFastCapture) { 5710 case FastCapture_Never: 5711 initFastCapture = false; 5712 break; 5713 case FastCapture_Always: 5714 initFastCapture = true; 5715 break; 5716 case FastCapture_Static: 5717 initFastCapture = (mFrameCount * 1000) / mSampleRate < kMinNormalCaptureBufferSizeMs; 5718 break; 5719 // case FastCapture_Dynamic: 5720 } 5721 5722 if (initFastCapture) { 5723 // create a Pipe for FastCapture to write to, and for us and fast tracks to read from 5724 NBAIO_Format format = mInputSource->format(); 5725 size_t pipeFramesP2 = roundup(mSampleRate / 25); // double-buffering of 20 ms each 5726 size_t pipeSize = pipeFramesP2 * Format_frameSize(format); 5727 void *pipeBuffer; 5728 const sp<MemoryDealer> roHeap(readOnlyHeap()); 5729 sp<IMemory> pipeMemory; 5730 if ((roHeap == 0) || 5731 (pipeMemory = roHeap->allocate(pipeSize)) == 0 || 5732 (pipeBuffer = pipeMemory->pointer()) == NULL) { 5733 ALOGE("not enough memory for pipe buffer size=%zu", pipeSize); 5734 goto failed; 5735 } 5736 // pipe will be shared directly with fast clients, so clear to avoid leaking old information 5737 memset(pipeBuffer, 0, pipeSize); 5738 Pipe *pipe = new Pipe(pipeFramesP2, format, pipeBuffer); 5739 const NBAIO_Format offers[1] = {format}; 5740 size_t numCounterOffers = 0; 5741 ssize_t index = pipe->negotiate(offers, 1, NULL, numCounterOffers); 5742 ALOG_ASSERT(index == 0); 5743 mPipeSink = pipe; 5744 PipeReader *pipeReader = new PipeReader(*pipe); 5745 numCounterOffers = 0; 5746 index = pipeReader->negotiate(offers, 1, NULL, numCounterOffers); 5747 ALOG_ASSERT(index == 0); 5748 mPipeSource = pipeReader; 5749 mPipeFramesP2 = pipeFramesP2; 5750 mPipeMemory = pipeMemory; 5751 5752 // create fast capture 5753 mFastCapture = new FastCapture(); 5754 FastCaptureStateQueue *sq = mFastCapture->sq(); 5755#ifdef STATE_QUEUE_DUMP 5756 // FIXME 5757#endif 5758 FastCaptureState *state = sq->begin(); 5759 state->mCblk = NULL; 5760 state->mInputSource = mInputSource.get(); 5761 state->mInputSourceGen++; 5762 state->mPipeSink = pipe; 5763 state->mPipeSinkGen++; 5764 state->mFrameCount = mFrameCount; 5765 state->mCommand = FastCaptureState::COLD_IDLE; 5766 // already done in constructor initialization list 5767 //mFastCaptureFutex = 0; 5768 state->mColdFutexAddr = &mFastCaptureFutex; 5769 state->mColdGen++; 5770 state->mDumpState = &mFastCaptureDumpState; 5771#ifdef TEE_SINK 5772 // FIXME 5773#endif 5774 mFastCaptureNBLogWriter = audioFlinger->newWriter_l(kFastCaptureLogSize, "FastCapture"); 5775 state->mNBLogWriter = mFastCaptureNBLogWriter.get(); 5776 sq->end(); 5777 sq->push(FastCaptureStateQueue::BLOCK_UNTIL_PUSHED); 5778 5779 // start the fast capture 5780 mFastCapture->run("FastCapture", ANDROID_PRIORITY_URGENT_AUDIO); 5781 pid_t tid = mFastCapture->getTid(); 5782 sendPrioConfigEvent(getpid_cached, tid, kPriorityFastCapture); 5783#ifdef AUDIO_WATCHDOG 5784 // FIXME 5785#endif 5786 5787 mFastTrackAvail = true; 5788 } 5789failed: ; 5790 5791 // FIXME mNormalSource 5792} 5793 5794AudioFlinger::RecordThread::~RecordThread() 5795{ 5796 if (mFastCapture != 0) { 5797 FastCaptureStateQueue *sq = mFastCapture->sq(); 5798 FastCaptureState *state = sq->begin(); 5799 if (state->mCommand == FastCaptureState::COLD_IDLE) { 5800 int32_t old = android_atomic_inc(&mFastCaptureFutex); 5801 if (old == -1) { 5802 (void) syscall(__NR_futex, &mFastCaptureFutex, FUTEX_WAKE_PRIVATE, 1); 5803 } 5804 } 5805 state->mCommand = FastCaptureState::EXIT; 5806 sq->end(); 5807 sq->push(FastCaptureStateQueue::BLOCK_UNTIL_PUSHED); 5808 mFastCapture->join(); 5809 mFastCapture.clear(); 5810 } 5811 mAudioFlinger->unregisterWriter(mFastCaptureNBLogWriter); 5812 mAudioFlinger->unregisterWriter(mNBLogWriter); 5813 free(mRsmpInBuffer); 5814} 5815 5816void AudioFlinger::RecordThread::onFirstRef() 5817{ 5818 run(mThreadName, PRIORITY_URGENT_AUDIO); 5819} 5820 5821bool AudioFlinger::RecordThread::threadLoop() 5822{ 5823 nsecs_t lastWarning = 0; 5824 5825 inputStandBy(); 5826 5827reacquire_wakelock: 5828 sp<RecordTrack> activeTrack; 5829 int activeTracksGen; 5830 { 5831 Mutex::Autolock _l(mLock); 5832 size_t size = mActiveTracks.size(); 5833 activeTracksGen = mActiveTracksGen; 5834 if (size > 0) { 5835 // FIXME an arbitrary choice 5836 activeTrack = mActiveTracks[0]; 5837 acquireWakeLock_l(activeTrack->uid()); 5838 if (size > 1) { 5839 SortedVector<int> tmp; 5840 for (size_t i = 0; i < size; i++) { 5841 tmp.add(mActiveTracks[i]->uid()); 5842 } 5843 updateWakeLockUids_l(tmp); 5844 } 5845 } else { 5846 acquireWakeLock_l(-1); 5847 } 5848 } 5849 5850 // used to request a deferred sleep, to be executed later while mutex is unlocked 5851 uint32_t sleepUs = 0; 5852 5853 // loop while there is work to do 5854 for (;;) { 5855 Vector< sp<EffectChain> > effectChains; 5856 5857 // sleep with mutex unlocked 5858 if (sleepUs > 0) { 5859 ATRACE_BEGIN("sleep"); 5860 usleep(sleepUs); 5861 ATRACE_END(); 5862 sleepUs = 0; 5863 } 5864 5865 // activeTracks accumulates a copy of a subset of mActiveTracks 5866 Vector< sp<RecordTrack> > activeTracks; 5867 5868 // reference to the (first and only) active fast track 5869 sp<RecordTrack> fastTrack; 5870 5871 // reference to a fast track which is about to be removed 5872 sp<RecordTrack> fastTrackToRemove; 5873 5874 { // scope for mLock 5875 Mutex::Autolock _l(mLock); 5876 5877 processConfigEvents_l(); 5878 5879 // check exitPending here because checkForNewParameters_l() and 5880 // checkForNewParameters_l() can temporarily release mLock 5881 if (exitPending()) { 5882 break; 5883 } 5884 5885 // if no active track(s), then standby and release wakelock 5886 size_t size = mActiveTracks.size(); 5887 if (size == 0) { 5888 standbyIfNotAlreadyInStandby(); 5889 // exitPending() can't become true here 5890 releaseWakeLock_l(); 5891 ALOGV("RecordThread: loop stopping"); 5892 // go to sleep 5893 mWaitWorkCV.wait(mLock); 5894 ALOGV("RecordThread: loop starting"); 5895 goto reacquire_wakelock; 5896 } 5897 5898 if (mActiveTracksGen != activeTracksGen) { 5899 activeTracksGen = mActiveTracksGen; 5900 SortedVector<int> tmp; 5901 for (size_t i = 0; i < size; i++) { 5902 tmp.add(mActiveTracks[i]->uid()); 5903 } 5904 updateWakeLockUids_l(tmp); 5905 } 5906 5907 bool doBroadcast = false; 5908 for (size_t i = 0; i < size; ) { 5909 5910 activeTrack = mActiveTracks[i]; 5911 if (activeTrack->isTerminated()) { 5912 if (activeTrack->isFastTrack()) { 5913 ALOG_ASSERT(fastTrackToRemove == 0); 5914 fastTrackToRemove = activeTrack; 5915 } 5916 removeTrack_l(activeTrack); 5917 mActiveTracks.remove(activeTrack); 5918 mActiveTracksGen++; 5919 size--; 5920 continue; 5921 } 5922 5923 TrackBase::track_state activeTrackState = activeTrack->mState; 5924 switch (activeTrackState) { 5925 5926 case TrackBase::PAUSING: 5927 mActiveTracks.remove(activeTrack); 5928 mActiveTracksGen++; 5929 doBroadcast = true; 5930 size--; 5931 continue; 5932 5933 case TrackBase::STARTING_1: 5934 sleepUs = 10000; 5935 i++; 5936 continue; 5937 5938 case TrackBase::STARTING_2: 5939 doBroadcast = true; 5940 mStandby = false; 5941 activeTrack->mState = TrackBase::ACTIVE; 5942 break; 5943 5944 case TrackBase::ACTIVE: 5945 break; 5946 5947 case TrackBase::IDLE: 5948 i++; 5949 continue; 5950 5951 default: 5952 LOG_ALWAYS_FATAL("Unexpected activeTrackState %d", activeTrackState); 5953 } 5954 5955 activeTracks.add(activeTrack); 5956 i++; 5957 5958 if (activeTrack->isFastTrack()) { 5959 ALOG_ASSERT(!mFastTrackAvail); 5960 ALOG_ASSERT(fastTrack == 0); 5961 fastTrack = activeTrack; 5962 } 5963 } 5964 if (doBroadcast) { 5965 mStartStopCond.broadcast(); 5966 } 5967 5968 // sleep if there are no active tracks to process 5969 if (activeTracks.size() == 0) { 5970 if (sleepUs == 0) { 5971 sleepUs = kRecordThreadSleepUs; 5972 } 5973 continue; 5974 } 5975 sleepUs = 0; 5976 5977 lockEffectChains_l(effectChains); 5978 } 5979 5980 // thread mutex is now unlocked, mActiveTracks unknown, activeTracks.size() > 0 5981 5982 size_t size = effectChains.size(); 5983 for (size_t i = 0; i < size; i++) { 5984 // thread mutex is not locked, but effect chain is locked 5985 effectChains[i]->process_l(); 5986 } 5987 5988 // Push a new fast capture state if fast capture is not already running, or cblk change 5989 if (mFastCapture != 0) { 5990 FastCaptureStateQueue *sq = mFastCapture->sq(); 5991 FastCaptureState *state = sq->begin(); 5992 bool didModify = false; 5993 FastCaptureStateQueue::block_t block = FastCaptureStateQueue::BLOCK_UNTIL_PUSHED; 5994 if (state->mCommand != FastCaptureState::READ_WRITE /* FIXME && 5995 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)*/) { 5996 if (state->mCommand == FastCaptureState::COLD_IDLE) { 5997 int32_t old = android_atomic_inc(&mFastCaptureFutex); 5998 if (old == -1) { 5999 (void) syscall(__NR_futex, &mFastCaptureFutex, FUTEX_WAKE_PRIVATE, 1); 6000 } 6001 } 6002 state->mCommand = FastCaptureState::READ_WRITE; 6003#if 0 // FIXME 6004 mFastCaptureDumpState.increaseSamplingN(mAudioFlinger->isLowRamDevice() ? 6005 FastThreadDumpState::kSamplingNforLowRamDevice : 6006 FastThreadDumpState::kSamplingN); 6007#endif 6008 didModify = true; 6009 } 6010 audio_track_cblk_t *cblkOld = state->mCblk; 6011 audio_track_cblk_t *cblkNew = fastTrack != 0 ? fastTrack->cblk() : NULL; 6012 if (cblkNew != cblkOld) { 6013 state->mCblk = cblkNew; 6014 // block until acked if removing a fast track 6015 if (cblkOld != NULL) { 6016 block = FastCaptureStateQueue::BLOCK_UNTIL_ACKED; 6017 } 6018 didModify = true; 6019 } 6020 sq->end(didModify); 6021 if (didModify) { 6022 sq->push(block); 6023#if 0 6024 if (kUseFastCapture == FastCapture_Dynamic) { 6025 mNormalSource = mPipeSource; 6026 } 6027#endif 6028 } 6029 } 6030 6031 // now run the fast track destructor with thread mutex unlocked 6032 fastTrackToRemove.clear(); 6033 6034 // Read from HAL to keep up with fastest client if multiple active tracks, not slowest one. 6035 // Only the client(s) that are too slow will overrun. But if even the fastest client is too 6036 // slow, then this RecordThread will overrun by not calling HAL read often enough. 6037 // If destination is non-contiguous, first read past the nominal end of buffer, then 6038 // copy to the right place. Permitted because mRsmpInBuffer was over-allocated. 6039 6040 int32_t rear = mRsmpInRear & (mRsmpInFramesP2 - 1); 6041 ssize_t framesRead; 6042 6043 // If an NBAIO source is present, use it to read the normal capture's data 6044 if (mPipeSource != 0) { 6045 size_t framesToRead = mBufferSize / mFrameSize; 6046 framesRead = mPipeSource->read((uint8_t*)mRsmpInBuffer + rear * mFrameSize, 6047 framesToRead); 6048 if (framesRead == 0) { 6049 // since pipe is non-blocking, simulate blocking input 6050 sleepUs = (framesToRead * 1000000LL) / mSampleRate; 6051 } 6052 // otherwise use the HAL / AudioStreamIn directly 6053 } else { 6054 ATRACE_BEGIN("read"); 6055 ssize_t bytesRead = mInput->stream->read(mInput->stream, 6056 (uint8_t*)mRsmpInBuffer + rear * mFrameSize, mBufferSize); 6057 ATRACE_END(); 6058 if (bytesRead < 0) { 6059 framesRead = bytesRead; 6060 } else { 6061 framesRead = bytesRead / mFrameSize; 6062 } 6063 } 6064 6065 // Update server timestamp with server stats 6066 // systemTime() is optional if the hardware supports timestamps. 6067 mTimestamp.mPosition[ExtendedTimestamp::LOCATION_SERVER] += framesRead; 6068 mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_SERVER] = systemTime(); 6069 6070 // Update server timestamp with kernel stats 6071 if (mInput->stream->get_capture_position != nullptr) { 6072 int64_t position, time; 6073 int ret = mInput->stream->get_capture_position(mInput->stream, &position, &time); 6074 if (ret == NO_ERROR) { 6075 mTimestamp.mPosition[ExtendedTimestamp::LOCATION_KERNEL] = position; 6076 mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL] = time; 6077 // Note: In general record buffers should tend to be empty in 6078 // a properly running pipeline. 6079 // 6080 // Also, it is not advantageous to call get_presentation_position during the read 6081 // as the read obtains a lock, preventing the timestamp call from executing. 6082 } 6083 } 6084 // Use this to track timestamp information 6085 // ALOGD("%s", mTimestamp.toString().c_str()); 6086 6087 if (framesRead < 0 || (framesRead == 0 && mPipeSource == 0)) { 6088 ALOGE("read failed: framesRead=%zd", framesRead); 6089 // Force input into standby so that it tries to recover at next read attempt 6090 inputStandBy(); 6091 sleepUs = kRecordThreadSleepUs; 6092 } 6093 if (framesRead <= 0) { 6094 goto unlock; 6095 } 6096 ALOG_ASSERT(framesRead > 0); 6097 6098 if (mTeeSink != 0) { 6099 (void) mTeeSink->write((uint8_t*)mRsmpInBuffer + rear * mFrameSize, framesRead); 6100 } 6101 // If destination is non-contiguous, we now correct for reading past end of buffer. 6102 { 6103 size_t part1 = mRsmpInFramesP2 - rear; 6104 if ((size_t) framesRead > part1) { 6105 memcpy(mRsmpInBuffer, (uint8_t*)mRsmpInBuffer + mRsmpInFramesP2 * mFrameSize, 6106 (framesRead - part1) * mFrameSize); 6107 } 6108 } 6109 rear = mRsmpInRear += framesRead; 6110 6111 size = activeTracks.size(); 6112 // loop over each active track 6113 for (size_t i = 0; i < size; i++) { 6114 activeTrack = activeTracks[i]; 6115 6116 // skip fast tracks, as those are handled directly by FastCapture 6117 if (activeTrack->isFastTrack()) { 6118 continue; 6119 } 6120 6121 // TODO: This code probably should be moved to RecordTrack. 6122 // TODO: Update the activeTrack buffer converter in case of reconfigure. 6123 6124 enum { 6125 OVERRUN_UNKNOWN, 6126 OVERRUN_TRUE, 6127 OVERRUN_FALSE 6128 } overrun = OVERRUN_UNKNOWN; 6129 6130 // loop over getNextBuffer to handle circular sink 6131 for (;;) { 6132 6133 activeTrack->mSink.frameCount = ~0; 6134 status_t status = activeTrack->getNextBuffer(&activeTrack->mSink); 6135 size_t framesOut = activeTrack->mSink.frameCount; 6136 LOG_ALWAYS_FATAL_IF((status == OK) != (framesOut > 0)); 6137 6138 // check available frames and handle overrun conditions 6139 // if the record track isn't draining fast enough. 6140 bool hasOverrun; 6141 size_t framesIn; 6142 activeTrack->mResamplerBufferProvider->sync(&framesIn, &hasOverrun); 6143 if (hasOverrun) { 6144 overrun = OVERRUN_TRUE; 6145 } 6146 if (framesOut == 0 || framesIn == 0) { 6147 break; 6148 } 6149 6150 // Don't allow framesOut to be larger than what is possible with resampling 6151 // from framesIn. 6152 // This isn't strictly necessary but helps limit buffer resizing in 6153 // RecordBufferConverter. TODO: remove when no longer needed. 6154 framesOut = min(framesOut, 6155 destinationFramesPossible( 6156 framesIn, mSampleRate, activeTrack->mSampleRate)); 6157 // process frames from the RecordThread buffer provider to the RecordTrack buffer 6158 framesOut = activeTrack->mRecordBufferConverter->convert( 6159 activeTrack->mSink.raw, activeTrack->mResamplerBufferProvider, framesOut); 6160 6161 if (framesOut > 0 && (overrun == OVERRUN_UNKNOWN)) { 6162 overrun = OVERRUN_FALSE; 6163 } 6164 6165 if (activeTrack->mFramesToDrop == 0) { 6166 if (framesOut > 0) { 6167 activeTrack->mSink.frameCount = framesOut; 6168 activeTrack->releaseBuffer(&activeTrack->mSink); 6169 } 6170 } else { 6171 // FIXME could do a partial drop of framesOut 6172 if (activeTrack->mFramesToDrop > 0) { 6173 activeTrack->mFramesToDrop -= framesOut; 6174 if (activeTrack->mFramesToDrop <= 0) { 6175 activeTrack->clearSyncStartEvent(); 6176 } 6177 } else { 6178 activeTrack->mFramesToDrop += framesOut; 6179 if (activeTrack->mFramesToDrop >= 0 || activeTrack->mSyncStartEvent == 0 || 6180 activeTrack->mSyncStartEvent->isCancelled()) { 6181 ALOGW("Synced record %s, session %d, trigger session %d", 6182 (activeTrack->mFramesToDrop >= 0) ? "timed out" : "cancelled", 6183 activeTrack->sessionId(), 6184 (activeTrack->mSyncStartEvent != 0) ? 6185 activeTrack->mSyncStartEvent->triggerSession() : 6186 AUDIO_SESSION_NONE); 6187 activeTrack->clearSyncStartEvent(); 6188 } 6189 } 6190 } 6191 6192 if (framesOut == 0) { 6193 break; 6194 } 6195 } 6196 6197 switch (overrun) { 6198 case OVERRUN_TRUE: 6199 // client isn't retrieving buffers fast enough 6200 if (!activeTrack->setOverflow()) { 6201 nsecs_t now = systemTime(); 6202 // FIXME should lastWarning per track? 6203 if ((now - lastWarning) > kWarningThrottleNs) { 6204 ALOGW("RecordThread: buffer overflow"); 6205 lastWarning = now; 6206 } 6207 } 6208 break; 6209 case OVERRUN_FALSE: 6210 activeTrack->clearOverflow(); 6211 break; 6212 case OVERRUN_UNKNOWN: 6213 break; 6214 } 6215 6216 // update frame information and push timestamp out 6217 activeTrack->updateTrackFrameInfo( 6218 activeTrack->mServerProxy->framesReleased(), 6219 mTimestamp.mPosition[ExtendedTimestamp::LOCATION_SERVER], 6220 mSampleRate, mTimestamp); 6221 } 6222 6223unlock: 6224 // enable changes in effect chain 6225 unlockEffectChains(effectChains); 6226 // effectChains doesn't need to be cleared, since it is cleared by destructor at scope end 6227 } 6228 6229 standbyIfNotAlreadyInStandby(); 6230 6231 { 6232 Mutex::Autolock _l(mLock); 6233 for (size_t i = 0; i < mTracks.size(); i++) { 6234 sp<RecordTrack> track = mTracks[i]; 6235 track->invalidate(); 6236 } 6237 mActiveTracks.clear(); 6238 mActiveTracksGen++; 6239 mStartStopCond.broadcast(); 6240 } 6241 6242 releaseWakeLock(); 6243 6244 ALOGV("RecordThread %p exiting", this); 6245 return false; 6246} 6247 6248void AudioFlinger::RecordThread::standbyIfNotAlreadyInStandby() 6249{ 6250 if (!mStandby) { 6251 inputStandBy(); 6252 mStandby = true; 6253 } 6254} 6255 6256void AudioFlinger::RecordThread::inputStandBy() 6257{ 6258 // Idle the fast capture if it's currently running 6259 if (mFastCapture != 0) { 6260 FastCaptureStateQueue *sq = mFastCapture->sq(); 6261 FastCaptureState *state = sq->begin(); 6262 if (!(state->mCommand & FastCaptureState::IDLE)) { 6263 state->mCommand = FastCaptureState::COLD_IDLE; 6264 state->mColdFutexAddr = &mFastCaptureFutex; 6265 state->mColdGen++; 6266 mFastCaptureFutex = 0; 6267 sq->end(); 6268 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 6269 sq->push(FastCaptureStateQueue::BLOCK_UNTIL_ACKED); 6270#if 0 6271 if (kUseFastCapture == FastCapture_Dynamic) { 6272 // FIXME 6273 } 6274#endif 6275#ifdef AUDIO_WATCHDOG 6276 // FIXME 6277#endif 6278 } else { 6279 sq->end(false /*didModify*/); 6280 } 6281 } 6282 mInput->stream->common.standby(&mInput->stream->common); 6283} 6284 6285// RecordThread::createRecordTrack_l() must be called with AudioFlinger::mLock held 6286sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 6287 const sp<AudioFlinger::Client>& client, 6288 uint32_t sampleRate, 6289 audio_format_t format, 6290 audio_channel_mask_t channelMask, 6291 size_t *pFrameCount, 6292 audio_session_t sessionId, 6293 size_t *notificationFrames, 6294 int uid, 6295 IAudioFlinger::track_flags_t *flags, 6296 pid_t tid, 6297 status_t *status) 6298{ 6299 size_t frameCount = *pFrameCount; 6300 sp<RecordTrack> track; 6301 status_t lStatus; 6302 6303 // client expresses a preference for FAST, but we get the final say 6304 if (*flags & IAudioFlinger::TRACK_FAST) { 6305 if ( 6306 // we formerly checked for a callback handler (non-0 tid), 6307 // but that is no longer required for TRANSFER_OBTAIN mode 6308 // 6309 // frame count is not specified, or is exactly the pipe depth 6310 ((frameCount == 0) || (frameCount == mPipeFramesP2)) && 6311 // PCM data 6312 audio_is_linear_pcm(format) && 6313 // hardware format 6314 (format == mFormat) && 6315 // hardware channel mask 6316 (channelMask == mChannelMask) && 6317 // hardware sample rate 6318 (sampleRate == mSampleRate) && 6319 // record thread has an associated fast capture 6320 hasFastCapture() && 6321 // there are sufficient fast track slots available 6322 mFastTrackAvail 6323 ) { 6324 ALOGV("AUDIO_INPUT_FLAG_FAST accepted: frameCount=%zu mFrameCount=%zu", 6325 frameCount, mFrameCount); 6326 } else { 6327 ALOGV("AUDIO_INPUT_FLAG_FAST denied: frameCount=%zu mFrameCount=%zu mPipeFramesP2=%zu " 6328 "format=%#x isLinear=%d channelMask=%#x sampleRate=%u mSampleRate=%u " 6329 "hasFastCapture=%d tid=%d mFastTrackAvail=%d", 6330 frameCount, mFrameCount, mPipeFramesP2, 6331 format, audio_is_linear_pcm(format), channelMask, sampleRate, mSampleRate, 6332 hasFastCapture(), tid, mFastTrackAvail); 6333 *flags &= ~IAudioFlinger::TRACK_FAST; 6334 } 6335 } 6336 6337 // compute track buffer size in frames, and suggest the notification frame count 6338 if (*flags & IAudioFlinger::TRACK_FAST) { 6339 // fast track: frame count is exactly the pipe depth 6340 frameCount = mPipeFramesP2; 6341 // ignore requested notificationFrames, and always notify exactly once every HAL buffer 6342 *notificationFrames = mFrameCount; 6343 } else { 6344 // not fast track: max notification period is resampled equivalent of one HAL buffer time 6345 // or 20 ms if there is a fast capture 6346 // TODO This could be a roundupRatio inline, and const 6347 size_t maxNotificationFrames = ((int64_t) (hasFastCapture() ? mSampleRate/50 : mFrameCount) 6348 * sampleRate + mSampleRate - 1) / mSampleRate; 6349 // minimum number of notification periods is at least kMinNotifications, 6350 // and at least kMinMs rounded up to a whole notification period (minNotificationsByMs) 6351 static const size_t kMinNotifications = 3; 6352 static const uint32_t kMinMs = 30; 6353 // TODO This could be a roundupRatio inline 6354 const size_t minFramesByMs = (sampleRate * kMinMs + 1000 - 1) / 1000; 6355 // TODO This could be a roundupRatio inline 6356 const size_t minNotificationsByMs = (minFramesByMs + maxNotificationFrames - 1) / 6357 maxNotificationFrames; 6358 const size_t minFrameCount = maxNotificationFrames * 6359 max(kMinNotifications, minNotificationsByMs); 6360 frameCount = max(frameCount, minFrameCount); 6361 if (*notificationFrames == 0 || *notificationFrames > maxNotificationFrames) { 6362 *notificationFrames = maxNotificationFrames; 6363 } 6364 } 6365 *pFrameCount = frameCount; 6366 6367 lStatus = initCheck(); 6368 if (lStatus != NO_ERROR) { 6369 ALOGE("createRecordTrack_l() audio driver not initialized"); 6370 goto Exit; 6371 } 6372 6373 { // scope for mLock 6374 Mutex::Autolock _l(mLock); 6375 6376 track = new RecordTrack(this, client, sampleRate, 6377 format, channelMask, frameCount, NULL, sessionId, uid, 6378 *flags, TrackBase::TYPE_DEFAULT); 6379 6380 lStatus = track->initCheck(); 6381 if (lStatus != NO_ERROR) { 6382 ALOGE("createRecordTrack_l() initCheck failed %d; no control block?", lStatus); 6383 // track must be cleared from the caller as the caller has the AF lock 6384 goto Exit; 6385 } 6386 mTracks.add(track); 6387 6388 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 6389 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 6390 mAudioFlinger->btNrecIsOff(); 6391 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 6392 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 6393 6394 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 6395 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 6396 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 6397 // so ask activity manager to do this on our behalf 6398 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 6399 } 6400 } 6401 6402 lStatus = NO_ERROR; 6403 6404Exit: 6405 *status = lStatus; 6406 return track; 6407} 6408 6409status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, 6410 AudioSystem::sync_event_t event, 6411 audio_session_t triggerSession) 6412{ 6413 ALOGV("RecordThread::start event %d, triggerSession %d", event, triggerSession); 6414 sp<ThreadBase> strongMe = this; 6415 status_t status = NO_ERROR; 6416 6417 if (event == AudioSystem::SYNC_EVENT_NONE) { 6418 recordTrack->clearSyncStartEvent(); 6419 } else if (event != AudioSystem::SYNC_EVENT_SAME) { 6420 recordTrack->mSyncStartEvent = mAudioFlinger->createSyncEvent(event, 6421 triggerSession, 6422 recordTrack->sessionId(), 6423 syncStartEventCallback, 6424 recordTrack); 6425 // Sync event can be cancelled by the trigger session if the track is not in a 6426 // compatible state in which case we start record immediately 6427 if (recordTrack->mSyncStartEvent->isCancelled()) { 6428 recordTrack->clearSyncStartEvent(); 6429 } else { 6430 // do not wait for the event for more than AudioSystem::kSyncRecordStartTimeOutMs 6431 recordTrack->mFramesToDrop = - 6432 ((AudioSystem::kSyncRecordStartTimeOutMs * recordTrack->mSampleRate) / 1000); 6433 } 6434 } 6435 6436 { 6437 // This section is a rendezvous between binder thread executing start() and RecordThread 6438 AutoMutex lock(mLock); 6439 if (mActiveTracks.indexOf(recordTrack) >= 0) { 6440 if (recordTrack->mState == TrackBase::PAUSING) { 6441 ALOGV("active record track PAUSING -> ACTIVE"); 6442 recordTrack->mState = TrackBase::ACTIVE; 6443 } else { 6444 ALOGV("active record track state %d", recordTrack->mState); 6445 } 6446 return status; 6447 } 6448 6449 // TODO consider other ways of handling this, such as changing the state to :STARTING and 6450 // adding the track to mActiveTracks after returning from AudioSystem::startInput(), 6451 // or using a separate command thread 6452 recordTrack->mState = TrackBase::STARTING_1; 6453 mActiveTracks.add(recordTrack); 6454 mActiveTracksGen++; 6455 status_t status = NO_ERROR; 6456 if (recordTrack->isExternalTrack()) { 6457 mLock.unlock(); 6458 status = AudioSystem::startInput(mId, recordTrack->sessionId()); 6459 mLock.lock(); 6460 // FIXME should verify that recordTrack is still in mActiveTracks 6461 if (status != NO_ERROR) { 6462 mActiveTracks.remove(recordTrack); 6463 mActiveTracksGen++; 6464 recordTrack->clearSyncStartEvent(); 6465 ALOGV("RecordThread::start error %d", status); 6466 return status; 6467 } 6468 } 6469 // Catch up with current buffer indices if thread is already running. 6470 // This is what makes a new client discard all buffered data. If the track's mRsmpInFront 6471 // was initialized to some value closer to the thread's mRsmpInFront, then the track could 6472 // see previously buffered data before it called start(), but with greater risk of overrun. 6473 6474 recordTrack->mResamplerBufferProvider->reset(); 6475 // clear any converter state as new data will be discontinuous 6476 recordTrack->mRecordBufferConverter->reset(); 6477 recordTrack->mState = TrackBase::STARTING_2; 6478 // signal thread to start 6479 mWaitWorkCV.broadcast(); 6480 if (mActiveTracks.indexOf(recordTrack) < 0) { 6481 ALOGV("Record failed to start"); 6482 status = BAD_VALUE; 6483 goto startError; 6484 } 6485 return status; 6486 } 6487 6488startError: 6489 if (recordTrack->isExternalTrack()) { 6490 AudioSystem::stopInput(mId, recordTrack->sessionId()); 6491 } 6492 recordTrack->clearSyncStartEvent(); 6493 // FIXME I wonder why we do not reset the state here? 6494 return status; 6495} 6496 6497void AudioFlinger::RecordThread::syncStartEventCallback(const wp<SyncEvent>& event) 6498{ 6499 sp<SyncEvent> strongEvent = event.promote(); 6500 6501 if (strongEvent != 0) { 6502 sp<RefBase> ptr = strongEvent->cookie().promote(); 6503 if (ptr != 0) { 6504 RecordTrack *recordTrack = (RecordTrack *)ptr.get(); 6505 recordTrack->handleSyncStartEvent(strongEvent); 6506 } 6507 } 6508} 6509 6510bool AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 6511 ALOGV("RecordThread::stop"); 6512 AutoMutex _l(mLock); 6513 if (mActiveTracks.indexOf(recordTrack) != 0 || recordTrack->mState == TrackBase::PAUSING) { 6514 return false; 6515 } 6516 // note that threadLoop may still be processing the track at this point [without lock] 6517 recordTrack->mState = TrackBase::PAUSING; 6518 // do not wait for mStartStopCond if exiting 6519 if (exitPending()) { 6520 return true; 6521 } 6522 // FIXME incorrect usage of wait: no explicit predicate or loop 6523 mStartStopCond.wait(mLock); 6524 // if we have been restarted, recordTrack is in mActiveTracks here 6525 if (exitPending() || mActiveTracks.indexOf(recordTrack) != 0) { 6526 ALOGV("Record stopped OK"); 6527 return true; 6528 } 6529 return false; 6530} 6531 6532bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event __unused) const 6533{ 6534 return false; 6535} 6536 6537status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event __unused) 6538{ 6539#if 0 // This branch is currently dead code, but is preserved in case it will be needed in future 6540 if (!isValidSyncEvent(event)) { 6541 return BAD_VALUE; 6542 } 6543 6544 audio_session_t eventSession = event->triggerSession(); 6545 status_t ret = NAME_NOT_FOUND; 6546 6547 Mutex::Autolock _l(mLock); 6548 6549 for (size_t i = 0; i < mTracks.size(); i++) { 6550 sp<RecordTrack> track = mTracks[i]; 6551 if (eventSession == track->sessionId()) { 6552 (void) track->setSyncEvent(event); 6553 ret = NO_ERROR; 6554 } 6555 } 6556 return ret; 6557#else 6558 return BAD_VALUE; 6559#endif 6560} 6561 6562// destroyTrack_l() must be called with ThreadBase::mLock held 6563void AudioFlinger::RecordThread::destroyTrack_l(const sp<RecordTrack>& track) 6564{ 6565 track->terminate(); 6566 track->mState = TrackBase::STOPPED; 6567 // active tracks are removed by threadLoop() 6568 if (mActiveTracks.indexOf(track) < 0) { 6569 removeTrack_l(track); 6570 } 6571} 6572 6573void AudioFlinger::RecordThread::removeTrack_l(const sp<RecordTrack>& track) 6574{ 6575 mTracks.remove(track); 6576 // need anything related to effects here? 6577 if (track->isFastTrack()) { 6578 ALOG_ASSERT(!mFastTrackAvail); 6579 mFastTrackAvail = true; 6580 } 6581} 6582 6583void AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 6584{ 6585 dumpInternals(fd, args); 6586 dumpTracks(fd, args); 6587 dumpEffectChains(fd, args); 6588} 6589 6590void AudioFlinger::RecordThread::dumpInternals(int fd, const Vector<String16>& args) 6591{ 6592 dprintf(fd, "\nInput thread %p:\n", this); 6593 6594 dumpBase(fd, args); 6595 6596 if (mActiveTracks.size() == 0) { 6597 dprintf(fd, " No active record clients\n"); 6598 } 6599 dprintf(fd, " Fast capture thread: %s\n", hasFastCapture() ? "yes" : "no"); 6600 dprintf(fd, " Fast track available: %s\n", mFastTrackAvail ? "yes" : "no"); 6601 6602 // Make a non-atomic copy of fast capture dump state so it won't change underneath us 6603 // while we are dumping it. It may be inconsistent, but it won't mutate! 6604 // This is a large object so we place it on the heap. 6605 // FIXME 25972958: Need an intelligent copy constructor that does not touch unused pages. 6606 const FastCaptureDumpState *copy = new FastCaptureDumpState(mFastCaptureDumpState); 6607 copy->dump(fd); 6608 delete copy; 6609} 6610 6611void AudioFlinger::RecordThread::dumpTracks(int fd, const Vector<String16>& args __unused) 6612{ 6613 const size_t SIZE = 256; 6614 char buffer[SIZE]; 6615 String8 result; 6616 6617 size_t numtracks = mTracks.size(); 6618 size_t numactive = mActiveTracks.size(); 6619 size_t numactiveseen = 0; 6620 dprintf(fd, " %zu Tracks", numtracks); 6621 if (numtracks) { 6622 dprintf(fd, " of which %zu are active\n", numactive); 6623 RecordTrack::appendDumpHeader(result); 6624 for (size_t i = 0; i < numtracks ; ++i) { 6625 sp<RecordTrack> track = mTracks[i]; 6626 if (track != 0) { 6627 bool active = mActiveTracks.indexOf(track) >= 0; 6628 if (active) { 6629 numactiveseen++; 6630 } 6631 track->dump(buffer, SIZE, active); 6632 result.append(buffer); 6633 } 6634 } 6635 } else { 6636 dprintf(fd, "\n"); 6637 } 6638 6639 if (numactiveseen != numactive) { 6640 snprintf(buffer, SIZE, " The following tracks are in the active list but" 6641 " not in the track list\n"); 6642 result.append(buffer); 6643 RecordTrack::appendDumpHeader(result); 6644 for (size_t i = 0; i < numactive; ++i) { 6645 sp<RecordTrack> track = mActiveTracks[i]; 6646 if (mTracks.indexOf(track) < 0) { 6647 track->dump(buffer, SIZE, true); 6648 result.append(buffer); 6649 } 6650 } 6651 6652 } 6653 write(fd, result.string(), result.size()); 6654} 6655 6656 6657void AudioFlinger::RecordThread::ResamplerBufferProvider::reset() 6658{ 6659 sp<ThreadBase> threadBase = mRecordTrack->mThread.promote(); 6660 RecordThread *recordThread = (RecordThread *) threadBase.get(); 6661 mRsmpInFront = recordThread->mRsmpInRear; 6662 mRsmpInUnrel = 0; 6663} 6664 6665void AudioFlinger::RecordThread::ResamplerBufferProvider::sync( 6666 size_t *framesAvailable, bool *hasOverrun) 6667{ 6668 sp<ThreadBase> threadBase = mRecordTrack->mThread.promote(); 6669 RecordThread *recordThread = (RecordThread *) threadBase.get(); 6670 const int32_t rear = recordThread->mRsmpInRear; 6671 const int32_t front = mRsmpInFront; 6672 const ssize_t filled = rear - front; 6673 6674 size_t framesIn; 6675 bool overrun = false; 6676 if (filled < 0) { 6677 // should not happen, but treat like a massive overrun and re-sync 6678 framesIn = 0; 6679 mRsmpInFront = rear; 6680 overrun = true; 6681 } else if ((size_t) filled <= recordThread->mRsmpInFrames) { 6682 framesIn = (size_t) filled; 6683 } else { 6684 // client is not keeping up with server, but give it latest data 6685 framesIn = recordThread->mRsmpInFrames; 6686 mRsmpInFront = /* front = */ rear - framesIn; 6687 overrun = true; 6688 } 6689 if (framesAvailable != NULL) { 6690 *framesAvailable = framesIn; 6691 } 6692 if (hasOverrun != NULL) { 6693 *hasOverrun = overrun; 6694 } 6695} 6696 6697// AudioBufferProvider interface 6698status_t AudioFlinger::RecordThread::ResamplerBufferProvider::getNextBuffer( 6699 AudioBufferProvider::Buffer* buffer) 6700{ 6701 sp<ThreadBase> threadBase = mRecordTrack->mThread.promote(); 6702 if (threadBase == 0) { 6703 buffer->frameCount = 0; 6704 buffer->raw = NULL; 6705 return NOT_ENOUGH_DATA; 6706 } 6707 RecordThread *recordThread = (RecordThread *) threadBase.get(); 6708 int32_t rear = recordThread->mRsmpInRear; 6709 int32_t front = mRsmpInFront; 6710 ssize_t filled = rear - front; 6711 // FIXME should not be P2 (don't want to increase latency) 6712 // FIXME if client not keeping up, discard 6713 LOG_ALWAYS_FATAL_IF(!(0 <= filled && (size_t) filled <= recordThread->mRsmpInFrames)); 6714 // 'filled' may be non-contiguous, so return only the first contiguous chunk 6715 front &= recordThread->mRsmpInFramesP2 - 1; 6716 size_t part1 = recordThread->mRsmpInFramesP2 - front; 6717 if (part1 > (size_t) filled) { 6718 part1 = filled; 6719 } 6720 size_t ask = buffer->frameCount; 6721 ALOG_ASSERT(ask > 0); 6722 if (part1 > ask) { 6723 part1 = ask; 6724 } 6725 if (part1 == 0) { 6726 // out of data is fine since the resampler will return a short-count. 6727 buffer->raw = NULL; 6728 buffer->frameCount = 0; 6729 mRsmpInUnrel = 0; 6730 return NOT_ENOUGH_DATA; 6731 } 6732 6733 buffer->raw = (uint8_t*)recordThread->mRsmpInBuffer + front * recordThread->mFrameSize; 6734 buffer->frameCount = part1; 6735 mRsmpInUnrel = part1; 6736 return NO_ERROR; 6737} 6738 6739// AudioBufferProvider interface 6740void AudioFlinger::RecordThread::ResamplerBufferProvider::releaseBuffer( 6741 AudioBufferProvider::Buffer* buffer) 6742{ 6743 size_t stepCount = buffer->frameCount; 6744 if (stepCount == 0) { 6745 return; 6746 } 6747 ALOG_ASSERT(stepCount <= mRsmpInUnrel); 6748 mRsmpInUnrel -= stepCount; 6749 mRsmpInFront += stepCount; 6750 buffer->raw = NULL; 6751 buffer->frameCount = 0; 6752} 6753 6754AudioFlinger::RecordThread::RecordBufferConverter::RecordBufferConverter( 6755 audio_channel_mask_t srcChannelMask, audio_format_t srcFormat, 6756 uint32_t srcSampleRate, 6757 audio_channel_mask_t dstChannelMask, audio_format_t dstFormat, 6758 uint32_t dstSampleRate) : 6759 mSrcChannelMask(AUDIO_CHANNEL_INVALID), // updateParameters will set following vars 6760 // mSrcFormat 6761 // mSrcSampleRate 6762 // mDstChannelMask 6763 // mDstFormat 6764 // mDstSampleRate 6765 // mSrcChannelCount 6766 // mDstChannelCount 6767 // mDstFrameSize 6768 mBuf(NULL), mBufFrames(0), mBufFrameSize(0), 6769 mResampler(NULL), 6770 mIsLegacyDownmix(false), 6771 mIsLegacyUpmix(false), 6772 mRequiresFloat(false), 6773 mInputConverterProvider(NULL) 6774{ 6775 (void)updateParameters(srcChannelMask, srcFormat, srcSampleRate, 6776 dstChannelMask, dstFormat, dstSampleRate); 6777} 6778 6779AudioFlinger::RecordThread::RecordBufferConverter::~RecordBufferConverter() { 6780 free(mBuf); 6781 delete mResampler; 6782 delete mInputConverterProvider; 6783} 6784 6785size_t AudioFlinger::RecordThread::RecordBufferConverter::convert(void *dst, 6786 AudioBufferProvider *provider, size_t frames) 6787{ 6788 if (mInputConverterProvider != NULL) { 6789 mInputConverterProvider->setBufferProvider(provider); 6790 provider = mInputConverterProvider; 6791 } 6792 6793 if (mResampler == NULL) { 6794 ALOGVV("NO RESAMPLING sampleRate:%u mSrcFormat:%#x mDstFormat:%#x", 6795 mSrcSampleRate, mSrcFormat, mDstFormat); 6796 6797 AudioBufferProvider::Buffer buffer; 6798 for (size_t i = frames; i > 0; ) { 6799 buffer.frameCount = i; 6800 status_t status = provider->getNextBuffer(&buffer); 6801 if (status != OK || buffer.frameCount == 0) { 6802 frames -= i; // cannot fill request. 6803 break; 6804 } 6805 // format convert to destination buffer 6806 convertNoResampler(dst, buffer.raw, buffer.frameCount); 6807 6808 dst = (int8_t*)dst + buffer.frameCount * mDstFrameSize; 6809 i -= buffer.frameCount; 6810 provider->releaseBuffer(&buffer); 6811 } 6812 } else { 6813 ALOGVV("RESAMPLING mSrcSampleRate:%u mDstSampleRate:%u mSrcFormat:%#x mDstFormat:%#x", 6814 mSrcSampleRate, mDstSampleRate, mSrcFormat, mDstFormat); 6815 6816 // reallocate buffer if needed 6817 if (mBufFrameSize != 0 && mBufFrames < frames) { 6818 free(mBuf); 6819 mBufFrames = frames; 6820 (void)posix_memalign(&mBuf, 32, mBufFrames * mBufFrameSize); 6821 } 6822 // resampler accumulates, but we only have one source track 6823 memset(mBuf, 0, frames * mBufFrameSize); 6824 frames = mResampler->resample((int32_t*)mBuf, frames, provider); 6825 // format convert to destination buffer 6826 convertResampler(dst, mBuf, frames); 6827 } 6828 return frames; 6829} 6830 6831status_t AudioFlinger::RecordThread::RecordBufferConverter::updateParameters( 6832 audio_channel_mask_t srcChannelMask, audio_format_t srcFormat, 6833 uint32_t srcSampleRate, 6834 audio_channel_mask_t dstChannelMask, audio_format_t dstFormat, 6835 uint32_t dstSampleRate) 6836{ 6837 // quick evaluation if there is any change. 6838 if (mSrcFormat == srcFormat 6839 && mSrcChannelMask == srcChannelMask 6840 && mSrcSampleRate == srcSampleRate 6841 && mDstFormat == dstFormat 6842 && mDstChannelMask == dstChannelMask 6843 && mDstSampleRate == dstSampleRate) { 6844 return NO_ERROR; 6845 } 6846 6847 ALOGV("RecordBufferConverter updateParameters srcMask:%#x dstMask:%#x" 6848 " srcFormat:%#x dstFormat:%#x srcRate:%u dstRate:%u", 6849 srcChannelMask, dstChannelMask, srcFormat, dstFormat, srcSampleRate, dstSampleRate); 6850 const bool valid = 6851 audio_is_input_channel(srcChannelMask) 6852 && audio_is_input_channel(dstChannelMask) 6853 && audio_is_valid_format(srcFormat) && audio_is_linear_pcm(srcFormat) 6854 && audio_is_valid_format(dstFormat) && audio_is_linear_pcm(dstFormat) 6855 && (srcSampleRate <= dstSampleRate * AUDIO_RESAMPLER_DOWN_RATIO_MAX) 6856 ; // no upsampling checks for now 6857 if (!valid) { 6858 return BAD_VALUE; 6859 } 6860 6861 mSrcFormat = srcFormat; 6862 mSrcChannelMask = srcChannelMask; 6863 mSrcSampleRate = srcSampleRate; 6864 mDstFormat = dstFormat; 6865 mDstChannelMask = dstChannelMask; 6866 mDstSampleRate = dstSampleRate; 6867 6868 // compute derived parameters 6869 mSrcChannelCount = audio_channel_count_from_in_mask(srcChannelMask); 6870 mDstChannelCount = audio_channel_count_from_in_mask(dstChannelMask); 6871 mDstFrameSize = mDstChannelCount * audio_bytes_per_sample(mDstFormat); 6872 6873 // do we need to resample? 6874 delete mResampler; 6875 mResampler = NULL; 6876 if (mSrcSampleRate != mDstSampleRate) { 6877 mResampler = AudioResampler::create(AUDIO_FORMAT_PCM_FLOAT, 6878 mSrcChannelCount, mDstSampleRate); 6879 mResampler->setSampleRate(mSrcSampleRate); 6880 mResampler->setVolume(AudioMixer::UNITY_GAIN_FLOAT, AudioMixer::UNITY_GAIN_FLOAT); 6881 } 6882 6883 // are we running legacy channel conversion modes? 6884 mIsLegacyDownmix = (mSrcChannelMask == AUDIO_CHANNEL_IN_STEREO 6885 || mSrcChannelMask == AUDIO_CHANNEL_IN_FRONT_BACK) 6886 && mDstChannelMask == AUDIO_CHANNEL_IN_MONO; 6887 mIsLegacyUpmix = mSrcChannelMask == AUDIO_CHANNEL_IN_MONO 6888 && (mDstChannelMask == AUDIO_CHANNEL_IN_STEREO 6889 || mDstChannelMask == AUDIO_CHANNEL_IN_FRONT_BACK); 6890 6891 // do we need to process in float? 6892 mRequiresFloat = mResampler != NULL || mIsLegacyDownmix || mIsLegacyUpmix; 6893 6894 // do we need a staging buffer to convert for destination (we can still optimize this)? 6895 // we use mBufFrameSize > 0 to indicate both frame size as well as buffer necessity 6896 if (mResampler != NULL) { 6897 mBufFrameSize = max(mSrcChannelCount, FCC_2) 6898 * audio_bytes_per_sample(AUDIO_FORMAT_PCM_FLOAT); 6899 } else if (mIsLegacyUpmix || mIsLegacyDownmix) { // legacy modes always float 6900 mBufFrameSize = mDstChannelCount * audio_bytes_per_sample(AUDIO_FORMAT_PCM_FLOAT); 6901 } else if (mSrcChannelMask != mDstChannelMask && mDstFormat != mSrcFormat) { 6902 mBufFrameSize = mDstChannelCount * audio_bytes_per_sample(mSrcFormat); 6903 } else { 6904 mBufFrameSize = 0; 6905 } 6906 mBufFrames = 0; // force the buffer to be resized. 6907 6908 // do we need an input converter buffer provider to give us float? 6909 delete mInputConverterProvider; 6910 mInputConverterProvider = NULL; 6911 if (mRequiresFloat && mSrcFormat != AUDIO_FORMAT_PCM_FLOAT) { 6912 mInputConverterProvider = new ReformatBufferProvider( 6913 audio_channel_count_from_in_mask(mSrcChannelMask), 6914 mSrcFormat, 6915 AUDIO_FORMAT_PCM_FLOAT, 6916 256 /* provider buffer frame count */); 6917 } 6918 6919 // do we need a remixer to do channel mask conversion 6920 if (!mIsLegacyDownmix && !mIsLegacyUpmix && mSrcChannelMask != mDstChannelMask) { 6921 (void) memcpy_by_index_array_initialization_from_channel_mask( 6922 mIdxAry, ARRAY_SIZE(mIdxAry), mDstChannelMask, mSrcChannelMask); 6923 } 6924 return NO_ERROR; 6925} 6926 6927void AudioFlinger::RecordThread::RecordBufferConverter::convertNoResampler( 6928 void *dst, const void *src, size_t frames) 6929{ 6930 // src is native type unless there is legacy upmix or downmix, whereupon it is float. 6931 if (mBufFrameSize != 0 && mBufFrames < frames) { 6932 free(mBuf); 6933 mBufFrames = frames; 6934 (void)posix_memalign(&mBuf, 32, mBufFrames * mBufFrameSize); 6935 } 6936 // do we need to do legacy upmix and downmix? 6937 if (mIsLegacyUpmix || mIsLegacyDownmix) { 6938 void *dstBuf = mBuf != NULL ? mBuf : dst; 6939 if (mIsLegacyUpmix) { 6940 upmix_to_stereo_float_from_mono_float((float *)dstBuf, 6941 (const float *)src, frames); 6942 } else /*mIsLegacyDownmix */ { 6943 downmix_to_mono_float_from_stereo_float((float *)dstBuf, 6944 (const float *)src, frames); 6945 } 6946 if (mBuf != NULL) { 6947 memcpy_by_audio_format(dst, mDstFormat, mBuf, AUDIO_FORMAT_PCM_FLOAT, 6948 frames * mDstChannelCount); 6949 } 6950 return; 6951 } 6952 // do we need to do channel mask conversion? 6953 if (mSrcChannelMask != mDstChannelMask) { 6954 void *dstBuf = mBuf != NULL ? mBuf : dst; 6955 memcpy_by_index_array(dstBuf, mDstChannelCount, 6956 src, mSrcChannelCount, mIdxAry, audio_bytes_per_sample(mSrcFormat), frames); 6957 if (dstBuf == dst) { 6958 return; // format is the same 6959 } 6960 } 6961 // convert to destination buffer 6962 const void *convertBuf = mBuf != NULL ? mBuf : src; 6963 memcpy_by_audio_format(dst, mDstFormat, convertBuf, mSrcFormat, 6964 frames * mDstChannelCount); 6965} 6966 6967void AudioFlinger::RecordThread::RecordBufferConverter::convertResampler( 6968 void *dst, /*not-a-const*/ void *src, size_t frames) 6969{ 6970 // src buffer format is ALWAYS float when entering this routine 6971 if (mIsLegacyUpmix) { 6972 ; // mono to stereo already handled by resampler 6973 } else if (mIsLegacyDownmix 6974 || (mSrcChannelMask == mDstChannelMask && mSrcChannelCount == 1)) { 6975 // the resampler outputs stereo for mono input channel (a feature?) 6976 // must convert to mono 6977 downmix_to_mono_float_from_stereo_float((float *)src, 6978 (const float *)src, frames); 6979 } else if (mSrcChannelMask != mDstChannelMask) { 6980 // convert to mono channel again for channel mask conversion (could be skipped 6981 // with further optimization). 6982 if (mSrcChannelCount == 1) { 6983 downmix_to_mono_float_from_stereo_float((float *)src, 6984 (const float *)src, frames); 6985 } 6986 // convert to destination format (in place, OK as float is larger than other types) 6987 if (mDstFormat != AUDIO_FORMAT_PCM_FLOAT) { 6988 memcpy_by_audio_format(src, mDstFormat, src, AUDIO_FORMAT_PCM_FLOAT, 6989 frames * mSrcChannelCount); 6990 } 6991 // channel convert and save to dst 6992 memcpy_by_index_array(dst, mDstChannelCount, 6993 src, mSrcChannelCount, mIdxAry, audio_bytes_per_sample(mDstFormat), frames); 6994 return; 6995 } 6996 // convert to destination format and save to dst 6997 memcpy_by_audio_format(dst, mDstFormat, src, AUDIO_FORMAT_PCM_FLOAT, 6998 frames * mDstChannelCount); 6999} 7000 7001bool AudioFlinger::RecordThread::checkForNewParameter_l(const String8& keyValuePair, 7002 status_t& status) 7003{ 7004 bool reconfig = false; 7005 7006 status = NO_ERROR; 7007 7008 audio_format_t reqFormat = mFormat; 7009 uint32_t samplingRate = mSampleRate; 7010 // TODO this may change if we want to support capture from HDMI PCM multi channel (e.g on TVs). 7011 audio_channel_mask_t channelMask = audio_channel_in_mask_from_count(mChannelCount); 7012 7013 AudioParameter param = AudioParameter(keyValuePair); 7014 int value; 7015 7016 // scope for AutoPark extends to end of method 7017 AutoPark<FastCapture> park(mFastCapture); 7018 7019 // TODO Investigate when this code runs. Check with audio policy when a sample rate and 7020 // channel count change can be requested. Do we mandate the first client defines the 7021 // HAL sampling rate and channel count or do we allow changes on the fly? 7022 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 7023 samplingRate = value; 7024 reconfig = true; 7025 } 7026 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 7027 if (!audio_is_linear_pcm((audio_format_t) value)) { 7028 status = BAD_VALUE; 7029 } else { 7030 reqFormat = (audio_format_t) value; 7031 reconfig = true; 7032 } 7033 } 7034 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 7035 audio_channel_mask_t mask = (audio_channel_mask_t) value; 7036 if (!audio_is_input_channel(mask) || 7037 audio_channel_count_from_in_mask(mask) > FCC_8) { 7038 status = BAD_VALUE; 7039 } else { 7040 channelMask = mask; 7041 reconfig = true; 7042 } 7043 } 7044 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 7045 // do not accept frame count changes if tracks are open as the track buffer 7046 // size depends on frame count and correct behavior would not be guaranteed 7047 // if frame count is changed after track creation 7048 if (mActiveTracks.size() > 0) { 7049 status = INVALID_OPERATION; 7050 } else { 7051 reconfig = true; 7052 } 7053 } 7054 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 7055 // forward device change to effects that have requested to be 7056 // aware of attached audio device. 7057 for (size_t i = 0; i < mEffectChains.size(); i++) { 7058 mEffectChains[i]->setDevice_l(value); 7059 } 7060 7061 // store input device and output device but do not forward output device to audio HAL. 7062 // Note that status is ignored by the caller for output device 7063 // (see AudioFlinger::setParameters() 7064 if (audio_is_output_devices(value)) { 7065 mOutDevice = value; 7066 status = BAD_VALUE; 7067 } else { 7068 mInDevice = value; 7069 if (value != AUDIO_DEVICE_NONE) { 7070 mPrevInDevice = value; 7071 } 7072 // disable AEC and NS if the device is a BT SCO headset supporting those 7073 // pre processings 7074 if (mTracks.size() > 0) { 7075 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 7076 mAudioFlinger->btNrecIsOff(); 7077 for (size_t i = 0; i < mTracks.size(); i++) { 7078 sp<RecordTrack> track = mTracks[i]; 7079 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId()); 7080 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId()); 7081 } 7082 } 7083 } 7084 } 7085 if (param.getInt(String8(AudioParameter::keyInputSource), value) == NO_ERROR && 7086 mAudioSource != (audio_source_t)value) { 7087 // forward device change to effects that have requested to be 7088 // aware of attached audio device. 7089 for (size_t i = 0; i < mEffectChains.size(); i++) { 7090 mEffectChains[i]->setAudioSource_l((audio_source_t)value); 7091 } 7092 mAudioSource = (audio_source_t)value; 7093 } 7094 7095 if (status == NO_ERROR) { 7096 status = mInput->stream->common.set_parameters(&mInput->stream->common, 7097 keyValuePair.string()); 7098 if (status == INVALID_OPERATION) { 7099 inputStandBy(); 7100 status = mInput->stream->common.set_parameters(&mInput->stream->common, 7101 keyValuePair.string()); 7102 } 7103 if (reconfig) { 7104 if (status == BAD_VALUE && 7105 audio_is_linear_pcm(mInput->stream->common.get_format(&mInput->stream->common)) && 7106 audio_is_linear_pcm(reqFormat) && 7107 (mInput->stream->common.get_sample_rate(&mInput->stream->common) 7108 <= (AUDIO_RESAMPLER_DOWN_RATIO_MAX * samplingRate)) && 7109 audio_channel_count_from_in_mask( 7110 mInput->stream->common.get_channels(&mInput->stream->common)) <= FCC_8) { 7111 status = NO_ERROR; 7112 } 7113 if (status == NO_ERROR) { 7114 readInputParameters_l(); 7115 sendIoConfigEvent_l(AUDIO_INPUT_CONFIG_CHANGED); 7116 } 7117 } 7118 } 7119 7120 return reconfig; 7121} 7122 7123String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 7124{ 7125 Mutex::Autolock _l(mLock); 7126 if (initCheck() != NO_ERROR) { 7127 return String8(); 7128 } 7129 7130 char *s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 7131 const String8 out_s8(s); 7132 free(s); 7133 return out_s8; 7134} 7135 7136void AudioFlinger::RecordThread::ioConfigChanged(audio_io_config_event event, pid_t pid) { 7137 sp<AudioIoDescriptor> desc = new AudioIoDescriptor(); 7138 7139 desc->mIoHandle = mId; 7140 7141 switch (event) { 7142 case AUDIO_INPUT_OPENED: 7143 case AUDIO_INPUT_CONFIG_CHANGED: 7144 desc->mPatch = mPatch; 7145 desc->mChannelMask = mChannelMask; 7146 desc->mSamplingRate = mSampleRate; 7147 desc->mFormat = mFormat; 7148 desc->mFrameCount = mFrameCount; 7149 desc->mFrameCountHAL = mFrameCount; 7150 desc->mLatency = 0; 7151 break; 7152 7153 case AUDIO_INPUT_CLOSED: 7154 default: 7155 break; 7156 } 7157 mAudioFlinger->ioConfigChanged(event, desc, pid); 7158} 7159 7160void AudioFlinger::RecordThread::readInputParameters_l() 7161{ 7162 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 7163 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 7164 mChannelCount = audio_channel_count_from_in_mask(mChannelMask); 7165 if (mChannelCount > FCC_8) { 7166 ALOGE("HAL channel count %d > %d", mChannelCount, FCC_8); 7167 } 7168 mHALFormat = mInput->stream->common.get_format(&mInput->stream->common); 7169 mFormat = mHALFormat; 7170 if (!audio_is_linear_pcm(mFormat)) { 7171 ALOGE("HAL format %#x is not linear pcm", mFormat); 7172 } 7173 mFrameSize = audio_stream_in_frame_size(mInput->stream); 7174 mBufferSize = mInput->stream->common.get_buffer_size(&mInput->stream->common); 7175 mFrameCount = mBufferSize / mFrameSize; 7176 // This is the formula for calculating the temporary buffer size. 7177 // With 7 HAL buffers, we can guarantee ability to down-sample the input by ratio of 6:1 to 7178 // 1 full output buffer, regardless of the alignment of the available input. 7179 // The value is somewhat arbitrary, and could probably be even larger. 7180 // A larger value should allow more old data to be read after a track calls start(), 7181 // without increasing latency. 7182 // 7183 // Note this is independent of the maximum downsampling ratio permitted for capture. 7184 mRsmpInFrames = mFrameCount * 7; 7185 mRsmpInFramesP2 = roundup(mRsmpInFrames); 7186 free(mRsmpInBuffer); 7187 mRsmpInBuffer = NULL; 7188 7189 // TODO optimize audio capture buffer sizes ... 7190 // Here we calculate the size of the sliding buffer used as a source 7191 // for resampling. mRsmpInFramesP2 is currently roundup(mFrameCount * 7). 7192 // For current HAL frame counts, this is usually 2048 = 40 ms. It would 7193 // be better to have it derived from the pipe depth in the long term. 7194 // The current value is higher than necessary. However it should not add to latency. 7195 7196 // Over-allocate beyond mRsmpInFramesP2 to permit a HAL read past end of buffer 7197 size_t bufferSize = (mRsmpInFramesP2 + mFrameCount - 1) * mFrameSize; 7198 (void)posix_memalign(&mRsmpInBuffer, 32, bufferSize); 7199 memset(mRsmpInBuffer, 0, bufferSize); // if posix_memalign fails, will segv here. 7200 7201 // AudioRecord mSampleRate and mChannelCount are constant due to AudioRecord API constraints. 7202 // But if thread's mSampleRate or mChannelCount changes, how will that affect active tracks? 7203} 7204 7205uint32_t AudioFlinger::RecordThread::getInputFramesLost() 7206{ 7207 Mutex::Autolock _l(mLock); 7208 if (initCheck() != NO_ERROR) { 7209 return 0; 7210 } 7211 7212 return mInput->stream->get_input_frames_lost(mInput->stream); 7213} 7214 7215uint32_t AudioFlinger::RecordThread::hasAudioSession(audio_session_t sessionId) const 7216{ 7217 Mutex::Autolock _l(mLock); 7218 uint32_t result = 0; 7219 if (getEffectChain_l(sessionId) != 0) { 7220 result = EFFECT_SESSION; 7221 } 7222 7223 for (size_t i = 0; i < mTracks.size(); ++i) { 7224 if (sessionId == mTracks[i]->sessionId()) { 7225 result |= TRACK_SESSION; 7226 break; 7227 } 7228 } 7229 7230 return result; 7231} 7232 7233KeyedVector<audio_session_t, bool> AudioFlinger::RecordThread::sessionIds() const 7234{ 7235 KeyedVector<audio_session_t, bool> ids; 7236 Mutex::Autolock _l(mLock); 7237 for (size_t j = 0; j < mTracks.size(); ++j) { 7238 sp<RecordThread::RecordTrack> track = mTracks[j]; 7239 audio_session_t sessionId = track->sessionId(); 7240 if (ids.indexOfKey(sessionId) < 0) { 7241 ids.add(sessionId, true); 7242 } 7243 } 7244 return ids; 7245} 7246 7247AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 7248{ 7249 Mutex::Autolock _l(mLock); 7250 AudioStreamIn *input = mInput; 7251 mInput = NULL; 7252 return input; 7253} 7254 7255// this method must always be called either with ThreadBase mLock held or inside the thread loop 7256audio_stream_t* AudioFlinger::RecordThread::stream() const 7257{ 7258 if (mInput == NULL) { 7259 return NULL; 7260 } 7261 return &mInput->stream->common; 7262} 7263 7264status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 7265{ 7266 // only one chain per input thread 7267 if (mEffectChains.size() != 0) { 7268 ALOGW("addEffectChain_l() already one chain %p on thread %p", chain.get(), this); 7269 return INVALID_OPERATION; 7270 } 7271 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 7272 chain->setThread(this); 7273 chain->setInBuffer(NULL); 7274 chain->setOutBuffer(NULL); 7275 7276 checkSuspendOnAddEffectChain_l(chain); 7277 7278 // make sure enabled pre processing effects state is communicated to the HAL as we 7279 // just moved them to a new input stream. 7280 chain->syncHalEffectsState(); 7281 7282 mEffectChains.add(chain); 7283 7284 return NO_ERROR; 7285} 7286 7287size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 7288{ 7289 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 7290 ALOGW_IF(mEffectChains.size() != 1, 7291 "removeEffectChain_l() %p invalid chain size %zu on thread %p", 7292 chain.get(), mEffectChains.size(), this); 7293 if (mEffectChains.size() == 1) { 7294 mEffectChains.removeAt(0); 7295 } 7296 return 0; 7297} 7298 7299status_t AudioFlinger::RecordThread::createAudioPatch_l(const struct audio_patch *patch, 7300 audio_patch_handle_t *handle) 7301{ 7302 status_t status = NO_ERROR; 7303 7304 // store new device and send to effects 7305 mInDevice = patch->sources[0].ext.device.type; 7306 mPatch = *patch; 7307 for (size_t i = 0; i < mEffectChains.size(); i++) { 7308 mEffectChains[i]->setDevice_l(mInDevice); 7309 } 7310 7311 // disable AEC and NS if the device is a BT SCO headset supporting those 7312 // pre processings 7313 if (mTracks.size() > 0) { 7314 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 7315 mAudioFlinger->btNrecIsOff(); 7316 for (size_t i = 0; i < mTracks.size(); i++) { 7317 sp<RecordTrack> track = mTracks[i]; 7318 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId()); 7319 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId()); 7320 } 7321 } 7322 7323 // store new source and send to effects 7324 if (mAudioSource != patch->sinks[0].ext.mix.usecase.source) { 7325 mAudioSource = patch->sinks[0].ext.mix.usecase.source; 7326 for (size_t i = 0; i < mEffectChains.size(); i++) { 7327 mEffectChains[i]->setAudioSource_l(mAudioSource); 7328 } 7329 } 7330 7331 if (mInput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 7332 audio_hw_device_t *hwDevice = mInput->audioHwDev->hwDevice(); 7333 status = hwDevice->create_audio_patch(hwDevice, 7334 patch->num_sources, 7335 patch->sources, 7336 patch->num_sinks, 7337 patch->sinks, 7338 handle); 7339 } else { 7340 char *address; 7341 if (strcmp(patch->sources[0].ext.device.address, "") != 0) { 7342 address = audio_device_address_to_parameter( 7343 patch->sources[0].ext.device.type, 7344 patch->sources[0].ext.device.address); 7345 } else { 7346 address = (char *)calloc(1, 1); 7347 } 7348 AudioParameter param = AudioParameter(String8(address)); 7349 free(address); 7350 param.addInt(String8(AUDIO_PARAMETER_STREAM_ROUTING), 7351 (int)patch->sources[0].ext.device.type); 7352 param.addInt(String8(AUDIO_PARAMETER_STREAM_INPUT_SOURCE), 7353 (int)patch->sinks[0].ext.mix.usecase.source); 7354 status = mInput->stream->common.set_parameters(&mInput->stream->common, 7355 param.toString().string()); 7356 *handle = AUDIO_PATCH_HANDLE_NONE; 7357 } 7358 7359 if (mInDevice != mPrevInDevice) { 7360 sendIoConfigEvent_l(AUDIO_INPUT_CONFIG_CHANGED); 7361 mPrevInDevice = mInDevice; 7362 } 7363 7364 return status; 7365} 7366 7367status_t AudioFlinger::RecordThread::releaseAudioPatch_l(const audio_patch_handle_t handle) 7368{ 7369 status_t status = NO_ERROR; 7370 7371 mInDevice = AUDIO_DEVICE_NONE; 7372 7373 if (mInput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 7374 audio_hw_device_t *hwDevice = mInput->audioHwDev->hwDevice(); 7375 status = hwDevice->release_audio_patch(hwDevice, handle); 7376 } else { 7377 AudioParameter param; 7378 param.addInt(String8(AUDIO_PARAMETER_STREAM_ROUTING), 0); 7379 status = mInput->stream->common.set_parameters(&mInput->stream->common, 7380 param.toString().string()); 7381 } 7382 return status; 7383} 7384 7385void AudioFlinger::RecordThread::addPatchRecord(const sp<PatchRecord>& record) 7386{ 7387 Mutex::Autolock _l(mLock); 7388 mTracks.add(record); 7389} 7390 7391void AudioFlinger::RecordThread::deletePatchRecord(const sp<PatchRecord>& record) 7392{ 7393 Mutex::Autolock _l(mLock); 7394 destroyTrack_l(record); 7395} 7396 7397void AudioFlinger::RecordThread::getAudioPortConfig(struct audio_port_config *config) 7398{ 7399 ThreadBase::getAudioPortConfig(config); 7400 config->role = AUDIO_PORT_ROLE_SINK; 7401 config->ext.mix.hw_module = mInput->audioHwDev->handle(); 7402 config->ext.mix.usecase.source = mAudioSource; 7403} 7404 7405} // namespace android 7406