Threads.cpp revision ea38ee7742e799b23bd8675f5801ef72f94de0f4
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::kMaxFastTracks) - 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::kMaxFastTracks); 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(audio_stream_type_t streamType) 2674{ 2675 ALOGV("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %zu", 2676 this, streamType, mTracks.size()); 2677 Mutex::Autolock _l(mLock); 2678 2679 size_t size = mTracks.size(); 2680 for (size_t i = 0; i < size; i++) { 2681 sp<Track> t = mTracks[i]; 2682 if (t->streamType() == streamType && t->isExternalTrack()) { 2683 t->invalidate(); 2684 } 2685 } 2686} 2687 2688status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 2689{ 2690 audio_session_t session = chain->sessionId(); 2691 int16_t* buffer = reinterpret_cast<int16_t*>(mEffectBufferEnabled 2692 ? mEffectBuffer : mSinkBuffer); 2693 bool ownsBuffer = false; 2694 2695 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 2696 if (session > AUDIO_SESSION_OUTPUT_MIX) { 2697 // Only one effect chain can be present in direct output thread and it uses 2698 // the sink buffer as input 2699 if (mType != DIRECT) { 2700 size_t numSamples = mNormalFrameCount * mChannelCount; 2701 buffer = new int16_t[numSamples]; 2702 memset(buffer, 0, numSamples * sizeof(int16_t)); 2703 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 2704 ownsBuffer = true; 2705 } 2706 2707 // Attach all tracks with same session ID to this chain. 2708 for (size_t i = 0; i < mTracks.size(); ++i) { 2709 sp<Track> track = mTracks[i]; 2710 if (session == track->sessionId()) { 2711 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), 2712 buffer); 2713 track->setMainBuffer(buffer); 2714 chain->incTrackCnt(); 2715 } 2716 } 2717 2718 // indicate all active tracks in the chain 2719 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 2720 sp<Track> track = mActiveTracks[i].promote(); 2721 if (track == 0) { 2722 continue; 2723 } 2724 if (session == track->sessionId()) { 2725 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 2726 chain->incActiveTrackCnt(); 2727 } 2728 } 2729 } 2730 chain->setThread(this); 2731 chain->setInBuffer(buffer, ownsBuffer); 2732 chain->setOutBuffer(reinterpret_cast<int16_t*>(mEffectBufferEnabled 2733 ? mEffectBuffer : mSinkBuffer)); 2734 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 2735 // chains list in order to be processed last as it contains output stage effects. 2736 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 2737 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 2738 // after track specific effects and before output stage. 2739 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 2740 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX. 2741 // Effect chain for other sessions are inserted at beginning of effect 2742 // chains list to be processed before output mix effects. Relative order between other 2743 // sessions is not important. 2744 static_assert(AUDIO_SESSION_OUTPUT_MIX == 0 && 2745 AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX, 2746 "audio_session_t constants misdefined"); 2747 size_t size = mEffectChains.size(); 2748 size_t i = 0; 2749 for (i = 0; i < size; i++) { 2750 if (mEffectChains[i]->sessionId() < session) { 2751 break; 2752 } 2753 } 2754 mEffectChains.insertAt(chain, i); 2755 checkSuspendOnAddEffectChain_l(chain); 2756 2757 return NO_ERROR; 2758} 2759 2760size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 2761{ 2762 audio_session_t session = chain->sessionId(); 2763 2764 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 2765 2766 for (size_t i = 0; i < mEffectChains.size(); i++) { 2767 if (chain == mEffectChains[i]) { 2768 mEffectChains.removeAt(i); 2769 // detach all active tracks from the chain 2770 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 2771 sp<Track> track = mActiveTracks[i].promote(); 2772 if (track == 0) { 2773 continue; 2774 } 2775 if (session == track->sessionId()) { 2776 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 2777 chain.get(), session); 2778 chain->decActiveTrackCnt(); 2779 } 2780 } 2781 2782 // detach all tracks with same session ID from this chain 2783 for (size_t i = 0; i < mTracks.size(); ++i) { 2784 sp<Track> track = mTracks[i]; 2785 if (session == track->sessionId()) { 2786 track->setMainBuffer(reinterpret_cast<int16_t*>(mSinkBuffer)); 2787 chain->decTrackCnt(); 2788 } 2789 } 2790 break; 2791 } 2792 } 2793 return mEffectChains.size(); 2794} 2795 2796status_t AudioFlinger::PlaybackThread::attachAuxEffect( 2797 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2798{ 2799 Mutex::Autolock _l(mLock); 2800 return attachAuxEffect_l(track, EffectId); 2801} 2802 2803status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 2804 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2805{ 2806 status_t status = NO_ERROR; 2807 2808 if (EffectId == 0) { 2809 track->setAuxBuffer(0, NULL); 2810 } else { 2811 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 2812 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 2813 if (effect != 0) { 2814 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 2815 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 2816 } else { 2817 status = INVALID_OPERATION; 2818 } 2819 } else { 2820 status = BAD_VALUE; 2821 } 2822 } 2823 return status; 2824} 2825 2826void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 2827{ 2828 for (size_t i = 0; i < mTracks.size(); ++i) { 2829 sp<Track> track = mTracks[i]; 2830 if (track->auxEffectId() == effectId) { 2831 attachAuxEffect_l(track, 0); 2832 } 2833 } 2834} 2835 2836bool AudioFlinger::PlaybackThread::threadLoop() 2837{ 2838 Vector< sp<Track> > tracksToRemove; 2839 2840 mStandbyTimeNs = systemTime(); 2841 2842 // MIXER 2843 nsecs_t lastWarning = 0; 2844 2845 // DUPLICATING 2846 // FIXME could this be made local to while loop? 2847 writeFrames = 0; 2848 2849 int lastGeneration = 0; 2850 2851 cacheParameters_l(); 2852 mSleepTimeUs = mIdleSleepTimeUs; 2853 2854 if (mType == MIXER) { 2855 sleepTimeShift = 0; 2856 } 2857 2858 CpuStats cpuStats; 2859 const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid())); 2860 2861 acquireWakeLock(); 2862 2863 // mNBLogWriter->log can only be called while thread mutex mLock is held. 2864 // So if you need to log when mutex is unlocked, set logString to a non-NULL string, 2865 // and then that string will be logged at the next convenient opportunity. 2866 const char *logString = NULL; 2867 2868 checkSilentMode_l(); 2869 2870 while (!exitPending()) 2871 { 2872 cpuStats.sample(myName); 2873 2874 Vector< sp<EffectChain> > effectChains; 2875 2876 { // scope for mLock 2877 2878 Mutex::Autolock _l(mLock); 2879 2880 processConfigEvents_l(); 2881 2882 if (logString != NULL) { 2883 mNBLogWriter->logTimestamp(); 2884 mNBLogWriter->log(logString); 2885 logString = NULL; 2886 } 2887 2888 // Gather the framesReleased counters for all active tracks, 2889 // and associate with the sink frames written out. We need 2890 // this to convert the sink timestamp to the track timestamp. 2891 if (mNormalSink != 0) { 2892 // Note: The DuplicatingThread may not have a mNormalSink. 2893 // We always fetch the timestamp here because often the downstream 2894 // sink will block whie writing. 2895 ExtendedTimestamp timestamp; // use private copy to fetch 2896 (void) mNormalSink->getTimestamp(timestamp); 2897 // copy over kernel info 2898 mTimestamp.mPosition[ExtendedTimestamp::LOCATION_KERNEL] = 2899 timestamp.mPosition[ExtendedTimestamp::LOCATION_KERNEL]; 2900 mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL] = 2901 timestamp.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL]; 2902 } 2903 // mFramesWritten for non-offloaded tracks are contiguous 2904 // even after standby() is called. This is useful for the track frame 2905 // to sink frame mapping. 2906 mTimestamp.mPosition[ExtendedTimestamp::LOCATION_SERVER] = mFramesWritten; 2907 mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_SERVER] = systemTime(); 2908 const size_t size = mActiveTracks.size(); 2909 for (size_t i = 0; i < size; ++i) { 2910 sp<Track> t = mActiveTracks[i].promote(); 2911 if (t != 0 && !t->isFastTrack()) { 2912 t->updateTrackFrameInfo( 2913 t->mAudioTrackServerProxy->framesReleased(), 2914 mFramesWritten, 2915 mTimestamp); 2916 } 2917 } 2918 2919 saveOutputTracks(); 2920 if (mSignalPending) { 2921 // A signal was raised while we were unlocked 2922 mSignalPending = false; 2923 } else if (waitingAsyncCallback_l()) { 2924 if (exitPending()) { 2925 break; 2926 } 2927 bool released = false; 2928 if (!keepWakeLock()) { 2929 releaseWakeLock_l(); 2930 released = true; 2931 } 2932 mWakeLockUids.clear(); 2933 mActiveTracksGeneration++; 2934 ALOGV("wait async completion"); 2935 mWaitWorkCV.wait(mLock); 2936 ALOGV("async completion/wake"); 2937 if (released) { 2938 acquireWakeLock_l(); 2939 } 2940 mStandbyTimeNs = systemTime() + mStandbyDelayNs; 2941 mSleepTimeUs = 0; 2942 2943 continue; 2944 } 2945 if ((!mActiveTracks.size() && systemTime() > mStandbyTimeNs) || 2946 isSuspended()) { 2947 // put audio hardware into standby after short delay 2948 if (shouldStandby_l()) { 2949 2950 threadLoop_standby(); 2951 2952 mStandby = true; 2953 } 2954 2955 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2956 // we're about to wait, flush the binder command buffer 2957 IPCThreadState::self()->flushCommands(); 2958 2959 clearOutputTracks(); 2960 2961 if (exitPending()) { 2962 break; 2963 } 2964 2965 releaseWakeLock_l(); 2966 mWakeLockUids.clear(); 2967 mActiveTracksGeneration++; 2968 // wait until we have something to do... 2969 ALOGV("%s going to sleep", myName.string()); 2970 mWaitWorkCV.wait(mLock); 2971 ALOGV("%s waking up", myName.string()); 2972 acquireWakeLock_l(); 2973 2974 mMixerStatus = MIXER_IDLE; 2975 mMixerStatusIgnoringFastTracks = MIXER_IDLE; 2976 mBytesWritten = 0; 2977 mBytesRemaining = 0; 2978 checkSilentMode_l(); 2979 2980 mStandbyTimeNs = systemTime() + mStandbyDelayNs; 2981 mSleepTimeUs = mIdleSleepTimeUs; 2982 if (mType == MIXER) { 2983 sleepTimeShift = 0; 2984 } 2985 2986 continue; 2987 } 2988 } 2989 // mMixerStatusIgnoringFastTracks is also updated internally 2990 mMixerStatus = prepareTracks_l(&tracksToRemove); 2991 2992 // compare with previously applied list 2993 if (lastGeneration != mActiveTracksGeneration) { 2994 // update wakelock 2995 updateWakeLockUids_l(mWakeLockUids); 2996 lastGeneration = mActiveTracksGeneration; 2997 } 2998 2999 // prevent any changes in effect chain list and in each effect chain 3000 // during mixing and effect process as the audio buffers could be deleted 3001 // or modified if an effect is created or deleted 3002 lockEffectChains_l(effectChains); 3003 } // mLock scope ends 3004 3005 if (mBytesRemaining == 0) { 3006 mCurrentWriteLength = 0; 3007 if (mMixerStatus == MIXER_TRACKS_READY) { 3008 // threadLoop_mix() sets mCurrentWriteLength 3009 threadLoop_mix(); 3010 } else if ((mMixerStatus != MIXER_DRAIN_TRACK) 3011 && (mMixerStatus != MIXER_DRAIN_ALL)) { 3012 // threadLoop_sleepTime sets mSleepTimeUs to 0 if data 3013 // must be written to HAL 3014 threadLoop_sleepTime(); 3015 if (mSleepTimeUs == 0) { 3016 mCurrentWriteLength = mSinkBufferSize; 3017 } 3018 } 3019 // Either threadLoop_mix() or threadLoop_sleepTime() should have set 3020 // mMixerBuffer with data if mMixerBufferValid is true and mSleepTimeUs == 0. 3021 // Merge mMixerBuffer data into mEffectBuffer (if any effects are valid) 3022 // or mSinkBuffer (if there are no effects). 3023 // 3024 // This is done pre-effects computation; if effects change to 3025 // support higher precision, this needs to move. 3026 // 3027 // mMixerBufferValid is only set true by MixerThread::prepareTracks_l(). 3028 // TODO use mSleepTimeUs == 0 as an additional condition. 3029 if (mMixerBufferValid) { 3030 void *buffer = mEffectBufferValid ? mEffectBuffer : mSinkBuffer; 3031 audio_format_t format = mEffectBufferValid ? mEffectBufferFormat : mFormat; 3032 3033 // mono blend occurs for mixer threads only (not direct or offloaded) 3034 // and is handled here if we're going directly to the sink. 3035 if (requireMonoBlend() && !mEffectBufferValid) { 3036 mono_blend(mMixerBuffer, mMixerBufferFormat, mChannelCount, mNormalFrameCount, 3037 true /*limit*/); 3038 } 3039 3040 memcpy_by_audio_format(buffer, format, mMixerBuffer, mMixerBufferFormat, 3041 mNormalFrameCount * mChannelCount); 3042 } 3043 3044 mBytesRemaining = mCurrentWriteLength; 3045 if (isSuspended()) { 3046 mSleepTimeUs = suspendSleepTimeUs(); 3047 // simulate write to HAL when suspended 3048 mBytesWritten += mSinkBufferSize; 3049 mFramesWritten += mSinkBufferSize / mFrameSize; 3050 mBytesRemaining = 0; 3051 } 3052 3053 // only process effects if we're going to write 3054 if (mSleepTimeUs == 0 && mType != OFFLOAD) { 3055 for (size_t i = 0; i < effectChains.size(); i ++) { 3056 effectChains[i]->process_l(); 3057 } 3058 } 3059 } 3060 // Process effect chains for offloaded thread even if no audio 3061 // was read from audio track: process only updates effect state 3062 // and thus does have to be synchronized with audio writes but may have 3063 // to be called while waiting for async write callback 3064 if (mType == OFFLOAD) { 3065 for (size_t i = 0; i < effectChains.size(); i ++) { 3066 effectChains[i]->process_l(); 3067 } 3068 } 3069 3070 // Only if the Effects buffer is enabled and there is data in the 3071 // Effects buffer (buffer valid), we need to 3072 // copy into the sink buffer. 3073 // TODO use mSleepTimeUs == 0 as an additional condition. 3074 if (mEffectBufferValid) { 3075 //ALOGV("writing effect buffer to sink buffer format %#x", mFormat); 3076 3077 if (requireMonoBlend()) { 3078 mono_blend(mEffectBuffer, mEffectBufferFormat, mChannelCount, mNormalFrameCount, 3079 true /*limit*/); 3080 } 3081 3082 memcpy_by_audio_format(mSinkBuffer, mFormat, mEffectBuffer, mEffectBufferFormat, 3083 mNormalFrameCount * mChannelCount); 3084 } 3085 3086 // enable changes in effect chain 3087 unlockEffectChains(effectChains); 3088 3089 if (!waitingAsyncCallback()) { 3090 // mSleepTimeUs == 0 means we must write to audio hardware 3091 if (mSleepTimeUs == 0) { 3092 ssize_t ret = 0; 3093 if (mBytesRemaining) { 3094 ret = threadLoop_write(); 3095 if (ret < 0) { 3096 mBytesRemaining = 0; 3097 } else { 3098 mBytesWritten += ret; 3099 mBytesRemaining -= ret; 3100 mFramesWritten += ret / mFrameSize; 3101 } 3102 } else if ((mMixerStatus == MIXER_DRAIN_TRACK) || 3103 (mMixerStatus == MIXER_DRAIN_ALL)) { 3104 threadLoop_drain(); 3105 } 3106 if (mType == MIXER && !mStandby) { 3107 // write blocked detection 3108 nsecs_t now = systemTime(); 3109 nsecs_t delta = now - mLastWriteTime; 3110 if (delta > maxPeriod) { 3111 mNumDelayedWrites++; 3112 if ((now - lastWarning) > kWarningThrottleNs) { 3113 ATRACE_NAME("underrun"); 3114 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 3115 (unsigned long long) ns2ms(delta), mNumDelayedWrites, this); 3116 lastWarning = now; 3117 } 3118 } 3119 3120 if (mThreadThrottle 3121 && mMixerStatus == MIXER_TRACKS_READY // we are mixing (active tracks) 3122 && ret > 0) { // we wrote something 3123 // Limit MixerThread data processing to no more than twice the 3124 // expected processing rate. 3125 // 3126 // This helps prevent underruns with NuPlayer and other applications 3127 // which may set up buffers that are close to the minimum size, or use 3128 // deep buffers, and rely on a double-buffering sleep strategy to fill. 3129 // 3130 // The throttle smooths out sudden large data drains from the device, 3131 // e.g. when it comes out of standby, which often causes problems with 3132 // (1) mixer threads without a fast mixer (which has its own warm-up) 3133 // (2) minimum buffer sized tracks (even if the track is full, 3134 // the app won't fill fast enough to handle the sudden draw). 3135 3136 const int32_t deltaMs = delta / 1000000; 3137 const int32_t throttleMs = mHalfBufferMs - deltaMs; 3138 if ((signed)mHalfBufferMs >= throttleMs && throttleMs > 0) { 3139 usleep(throttleMs * 1000); 3140 // notify of throttle start on verbose log 3141 ALOGV_IF(mThreadThrottleEndMs == mThreadThrottleTimeMs, 3142 "mixer(%p) throttle begin:" 3143 " ret(%zd) deltaMs(%d) requires sleep %d ms", 3144 this, ret, deltaMs, throttleMs); 3145 mThreadThrottleTimeMs += throttleMs; 3146 } else { 3147 uint32_t diff = mThreadThrottleTimeMs - mThreadThrottleEndMs; 3148 if (diff > 0) { 3149 // notify of throttle end on debug log 3150 ALOGD("mixer(%p) throttle end: throttle time(%u)", this, diff); 3151 mThreadThrottleEndMs = mThreadThrottleTimeMs; 3152 } 3153 } 3154 } 3155 } 3156 3157 } else { 3158 ATRACE_BEGIN("sleep"); 3159 if ((mType == OFFLOAD) && !audio_has_proportional_frames(mFormat)) { 3160 Mutex::Autolock _l(mLock); 3161 if (!mSignalPending && !exitPending()) { 3162 // If more than one buffer has been written to the audio HAL since exiting 3163 // standby or last flush, do not sleep more than one buffer duration 3164 // since last write and not less than kDirectMinSleepTimeUs. 3165 // Wake up if a command is received 3166 uint32_t timeoutUs = mSleepTimeUs; 3167 if (mBytesWritten >= (int64_t) mBufferSize) { 3168 nsecs_t now = systemTime(); 3169 uint32_t deltaUs = (uint32_t)((now - mLastWriteTime) / 1000); 3170 if (timeoutUs + deltaUs > mBufferDurationUs) { 3171 if (mBufferDurationUs > deltaUs) { 3172 timeoutUs = mBufferDurationUs - deltaUs; 3173 if (timeoutUs < kDirectMinSleepTimeUs) { 3174 timeoutUs = kDirectMinSleepTimeUs; 3175 } 3176 } else { 3177 timeoutUs = kDirectMinSleepTimeUs; 3178 } 3179 } 3180 } 3181 mWaitWorkCV.waitRelative(mLock, microseconds((nsecs_t)timeoutUs)); 3182 } 3183 } else { 3184 usleep(mSleepTimeUs); 3185 } 3186 ATRACE_END(); 3187 } 3188 } 3189 3190 // Finally let go of removed track(s), without the lock held 3191 // since we can't guarantee the destructors won't acquire that 3192 // same lock. This will also mutate and push a new fast mixer state. 3193 threadLoop_removeTracks(tracksToRemove); 3194 tracksToRemove.clear(); 3195 3196 // FIXME I don't understand the need for this here; 3197 // it was in the original code but maybe the 3198 // assignment in saveOutputTracks() makes this unnecessary? 3199 clearOutputTracks(); 3200 3201 // Effect chains will be actually deleted here if they were removed from 3202 // mEffectChains list during mixing or effects processing 3203 effectChains.clear(); 3204 3205 // FIXME Note that the above .clear() is no longer necessary since effectChains 3206 // is now local to this block, but will keep it for now (at least until merge done). 3207 } 3208 3209 threadLoop_exit(); 3210 3211 if (!mStandby) { 3212 threadLoop_standby(); 3213 mStandby = true; 3214 } 3215 3216 releaseWakeLock(); 3217 mWakeLockUids.clear(); 3218 mActiveTracksGeneration++; 3219 3220 ALOGV("Thread %p type %d exiting", this, mType); 3221 return false; 3222} 3223 3224// removeTracks_l() must be called with ThreadBase::mLock held 3225void AudioFlinger::PlaybackThread::removeTracks_l(const Vector< sp<Track> >& tracksToRemove) 3226{ 3227 size_t count = tracksToRemove.size(); 3228 if (count > 0) { 3229 for (size_t i=0 ; i<count ; i++) { 3230 const sp<Track>& track = tracksToRemove.itemAt(i); 3231 mActiveTracks.remove(track); 3232 mWakeLockUids.remove(track->uid()); 3233 mActiveTracksGeneration++; 3234 ALOGV("removeTracks_l removing track on session %d", track->sessionId()); 3235 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 3236 if (chain != 0) { 3237 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), 3238 track->sessionId()); 3239 chain->decActiveTrackCnt(); 3240 } 3241 if (track->isTerminated()) { 3242 removeTrack_l(track); 3243 } 3244 } 3245 } 3246 3247} 3248 3249status_t AudioFlinger::PlaybackThread::getTimestamp_l(AudioTimestamp& timestamp) 3250{ 3251 if (mNormalSink != 0) { 3252 ExtendedTimestamp ets; 3253 status_t status = mNormalSink->getTimestamp(ets); 3254 if (status == NO_ERROR) { 3255 status = ets.getBestTimestamp(×tamp); 3256 } 3257 return status; 3258 } 3259 if ((mType == OFFLOAD || mType == DIRECT) 3260 && mOutput != NULL && mOutput->stream->get_presentation_position) { 3261 uint64_t position64; 3262 int ret = mOutput->getPresentationPosition(&position64, ×tamp.mTime); 3263 if (ret == 0) { 3264 timestamp.mPosition = (uint32_t)position64; 3265 return NO_ERROR; 3266 } 3267 } 3268 return INVALID_OPERATION; 3269} 3270 3271status_t AudioFlinger::MixerThread::createAudioPatch_l(const struct audio_patch *patch, 3272 audio_patch_handle_t *handle) 3273{ 3274 AutoPark<FastMixer> park(mFastMixer); 3275 3276 status_t status = PlaybackThread::createAudioPatch_l(patch, handle); 3277 3278 return status; 3279} 3280 3281status_t AudioFlinger::PlaybackThread::createAudioPatch_l(const struct audio_patch *patch, 3282 audio_patch_handle_t *handle) 3283{ 3284 status_t status = NO_ERROR; 3285 3286 // store new device and send to effects 3287 audio_devices_t type = AUDIO_DEVICE_NONE; 3288 for (unsigned int i = 0; i < patch->num_sinks; i++) { 3289 type |= patch->sinks[i].ext.device.type; 3290 } 3291 3292#ifdef ADD_BATTERY_DATA 3293 // when changing the audio output device, call addBatteryData to notify 3294 // the change 3295 if (mOutDevice != type) { 3296 uint32_t params = 0; 3297 // check whether speaker is on 3298 if (type & AUDIO_DEVICE_OUT_SPEAKER) { 3299 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 3300 } 3301 3302 audio_devices_t deviceWithoutSpeaker 3303 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 3304 // check if any other device (except speaker) is on 3305 if (type & deviceWithoutSpeaker) { 3306 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 3307 } 3308 3309 if (params != 0) { 3310 addBatteryData(params); 3311 } 3312 } 3313#endif 3314 3315 for (size_t i = 0; i < mEffectChains.size(); i++) { 3316 mEffectChains[i]->setDevice_l(type); 3317 } 3318 3319 // mPrevOutDevice is the latest device set by createAudioPatch_l(). It is not set when 3320 // the thread is created so that the first patch creation triggers an ioConfigChanged callback 3321 bool configChanged = mPrevOutDevice != type; 3322 mOutDevice = type; 3323 mPatch = *patch; 3324 3325 if (mOutput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 3326 audio_hw_device_t *hwDevice = mOutput->audioHwDev->hwDevice(); 3327 status = hwDevice->create_audio_patch(hwDevice, 3328 patch->num_sources, 3329 patch->sources, 3330 patch->num_sinks, 3331 patch->sinks, 3332 handle); 3333 } else { 3334 char *address; 3335 if (strcmp(patch->sinks[0].ext.device.address, "") != 0) { 3336 //FIXME: we only support address on first sink with HAL version < 3.0 3337 address = audio_device_address_to_parameter( 3338 patch->sinks[0].ext.device.type, 3339 patch->sinks[0].ext.device.address); 3340 } else { 3341 address = (char *)calloc(1, 1); 3342 } 3343 AudioParameter param = AudioParameter(String8(address)); 3344 free(address); 3345 param.addInt(String8(AUDIO_PARAMETER_STREAM_ROUTING), (int)type); 3346 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3347 param.toString().string()); 3348 *handle = AUDIO_PATCH_HANDLE_NONE; 3349 } 3350 if (configChanged) { 3351 mPrevOutDevice = type; 3352 sendIoConfigEvent_l(AUDIO_OUTPUT_CONFIG_CHANGED); 3353 } 3354 return status; 3355} 3356 3357status_t AudioFlinger::MixerThread::releaseAudioPatch_l(const audio_patch_handle_t handle) 3358{ 3359 AutoPark<FastMixer> park(mFastMixer); 3360 3361 status_t status = PlaybackThread::releaseAudioPatch_l(handle); 3362 3363 return status; 3364} 3365 3366status_t AudioFlinger::PlaybackThread::releaseAudioPatch_l(const audio_patch_handle_t handle) 3367{ 3368 status_t status = NO_ERROR; 3369 3370 mOutDevice = AUDIO_DEVICE_NONE; 3371 3372 if (mOutput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 3373 audio_hw_device_t *hwDevice = mOutput->audioHwDev->hwDevice(); 3374 status = hwDevice->release_audio_patch(hwDevice, handle); 3375 } else { 3376 AudioParameter param; 3377 param.addInt(String8(AUDIO_PARAMETER_STREAM_ROUTING), 0); 3378 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3379 param.toString().string()); 3380 } 3381 return status; 3382} 3383 3384void AudioFlinger::PlaybackThread::addPatchTrack(const sp<PatchTrack>& track) 3385{ 3386 Mutex::Autolock _l(mLock); 3387 mTracks.add(track); 3388} 3389 3390void AudioFlinger::PlaybackThread::deletePatchTrack(const sp<PatchTrack>& track) 3391{ 3392 Mutex::Autolock _l(mLock); 3393 destroyTrack_l(track); 3394} 3395 3396void AudioFlinger::PlaybackThread::getAudioPortConfig(struct audio_port_config *config) 3397{ 3398 ThreadBase::getAudioPortConfig(config); 3399 config->role = AUDIO_PORT_ROLE_SOURCE; 3400 config->ext.mix.hw_module = mOutput->audioHwDev->handle(); 3401 config->ext.mix.usecase.stream = AUDIO_STREAM_DEFAULT; 3402} 3403 3404// ---------------------------------------------------------------------------- 3405 3406AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 3407 audio_io_handle_t id, audio_devices_t device, bool systemReady, type_t type) 3408 : PlaybackThread(audioFlinger, output, id, device, type, systemReady), 3409 // mAudioMixer below 3410 // mFastMixer below 3411 mFastMixerFutex(0), 3412 mMasterMono(false) 3413 // mOutputSink below 3414 // mPipeSink below 3415 // mNormalSink below 3416{ 3417 ALOGV("MixerThread() id=%d device=%#x type=%d", id, device, type); 3418 ALOGV("mSampleRate=%u, mChannelMask=%#x, mChannelCount=%u, mFormat=%d, mFrameSize=%zu, " 3419 "mFrameCount=%zu, mNormalFrameCount=%zu", 3420 mSampleRate, mChannelMask, mChannelCount, mFormat, mFrameSize, mFrameCount, 3421 mNormalFrameCount); 3422 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 3423 3424 if (type == DUPLICATING) { 3425 // The Duplicating thread uses the AudioMixer and delivers data to OutputTracks 3426 // (downstream MixerThreads) in DuplicatingThread::threadLoop_write(). 3427 // Do not create or use mFastMixer, mOutputSink, mPipeSink, or mNormalSink. 3428 return; 3429 } 3430 // create an NBAIO sink for the HAL output stream, and negotiate 3431 mOutputSink = new AudioStreamOutSink(output->stream); 3432 size_t numCounterOffers = 0; 3433 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount, mFormat)}; 3434#if !LOG_NDEBUG 3435 ssize_t index = 3436#else 3437 (void) 3438#endif 3439 mOutputSink->negotiate(offers, 1, NULL, numCounterOffers); 3440 ALOG_ASSERT(index == 0); 3441 3442 // initialize fast mixer depending on configuration 3443 bool initFastMixer; 3444 switch (kUseFastMixer) { 3445 case FastMixer_Never: 3446 initFastMixer = false; 3447 break; 3448 case FastMixer_Always: 3449 initFastMixer = true; 3450 break; 3451 case FastMixer_Static: 3452 case FastMixer_Dynamic: 3453 initFastMixer = mFrameCount < mNormalFrameCount; 3454 break; 3455 } 3456 if (initFastMixer) { 3457 audio_format_t fastMixerFormat; 3458 if (mMixerBufferEnabled && mEffectBufferEnabled) { 3459 fastMixerFormat = AUDIO_FORMAT_PCM_FLOAT; 3460 } else { 3461 fastMixerFormat = AUDIO_FORMAT_PCM_16_BIT; 3462 } 3463 if (mFormat != fastMixerFormat) { 3464 // change our Sink format to accept our intermediate precision 3465 mFormat = fastMixerFormat; 3466 free(mSinkBuffer); 3467 mFrameSize = mChannelCount * audio_bytes_per_sample(mFormat); 3468 const size_t sinkBufferSize = mNormalFrameCount * mFrameSize; 3469 (void)posix_memalign(&mSinkBuffer, 32, sinkBufferSize); 3470 } 3471 3472 // create a MonoPipe to connect our submix to FastMixer 3473 NBAIO_Format format = mOutputSink->format(); 3474#ifdef TEE_SINK 3475 NBAIO_Format origformat = format; 3476#endif 3477 // adjust format to match that of the Fast Mixer 3478 ALOGV("format changed from %d to %d", format.mFormat, fastMixerFormat); 3479 format.mFormat = fastMixerFormat; 3480 format.mFrameSize = audio_bytes_per_sample(format.mFormat) * format.mChannelCount; 3481 3482 // This pipe depth compensates for scheduling latency of the normal mixer thread. 3483 // When it wakes up after a maximum latency, it runs a few cycles quickly before 3484 // finally blocking. Note the pipe implementation rounds up the request to a power of 2. 3485 MonoPipe *monoPipe = new MonoPipe(mNormalFrameCount * 4, format, true /*writeCanBlock*/); 3486 const NBAIO_Format offers[1] = {format}; 3487 size_t numCounterOffers = 0; 3488#if !LOG_NDEBUG || defined(TEE_SINK) 3489 ssize_t index = 3490#else 3491 (void) 3492#endif 3493 monoPipe->negotiate(offers, 1, NULL, numCounterOffers); 3494 ALOG_ASSERT(index == 0); 3495 monoPipe->setAvgFrames((mScreenState & 1) ? 3496 (monoPipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 3497 mPipeSink = monoPipe; 3498 3499#ifdef TEE_SINK 3500 if (mTeeSinkOutputEnabled) { 3501 // create a Pipe to archive a copy of FastMixer's output for dumpsys 3502 Pipe *teeSink = new Pipe(mTeeSinkOutputFrames, origformat); 3503 const NBAIO_Format offers2[1] = {origformat}; 3504 numCounterOffers = 0; 3505 index = teeSink->negotiate(offers2, 1, NULL, numCounterOffers); 3506 ALOG_ASSERT(index == 0); 3507 mTeeSink = teeSink; 3508 PipeReader *teeSource = new PipeReader(*teeSink); 3509 numCounterOffers = 0; 3510 index = teeSource->negotiate(offers2, 1, NULL, numCounterOffers); 3511 ALOG_ASSERT(index == 0); 3512 mTeeSource = teeSource; 3513 } 3514#endif 3515 3516 // create fast mixer and configure it initially with just one fast track for our submix 3517 mFastMixer = new FastMixer(); 3518 FastMixerStateQueue *sq = mFastMixer->sq(); 3519#ifdef STATE_QUEUE_DUMP 3520 sq->setObserverDump(&mStateQueueObserverDump); 3521 sq->setMutatorDump(&mStateQueueMutatorDump); 3522#endif 3523 FastMixerState *state = sq->begin(); 3524 FastTrack *fastTrack = &state->mFastTracks[0]; 3525 // wrap the source side of the MonoPipe to make it an AudioBufferProvider 3526 fastTrack->mBufferProvider = new SourceAudioBufferProvider(new MonoPipeReader(monoPipe)); 3527 fastTrack->mVolumeProvider = NULL; 3528 fastTrack->mChannelMask = mChannelMask; // mPipeSink channel mask for audio to FastMixer 3529 fastTrack->mFormat = mFormat; // mPipeSink format for audio to FastMixer 3530 fastTrack->mGeneration++; 3531 state->mFastTracksGen++; 3532 state->mTrackMask = 1; 3533 // fast mixer will use the HAL output sink 3534 state->mOutputSink = mOutputSink.get(); 3535 state->mOutputSinkGen++; 3536 state->mFrameCount = mFrameCount; 3537 state->mCommand = FastMixerState::COLD_IDLE; 3538 // already done in constructor initialization list 3539 //mFastMixerFutex = 0; 3540 state->mColdFutexAddr = &mFastMixerFutex; 3541 state->mColdGen++; 3542 state->mDumpState = &mFastMixerDumpState; 3543#ifdef TEE_SINK 3544 state->mTeeSink = mTeeSink.get(); 3545#endif 3546 mFastMixerNBLogWriter = audioFlinger->newWriter_l(kFastMixerLogSize, "FastMixer"); 3547 state->mNBLogWriter = mFastMixerNBLogWriter.get(); 3548 sq->end(); 3549 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3550 3551 // start the fast mixer 3552 mFastMixer->run("FastMixer", PRIORITY_URGENT_AUDIO); 3553 pid_t tid = mFastMixer->getTid(); 3554 sendPrioConfigEvent(getpid_cached, tid, kPriorityFastMixer); 3555 3556#ifdef AUDIO_WATCHDOG 3557 // create and start the watchdog 3558 mAudioWatchdog = new AudioWatchdog(); 3559 mAudioWatchdog->setDump(&mAudioWatchdogDump); 3560 mAudioWatchdog->run("AudioWatchdog", PRIORITY_URGENT_AUDIO); 3561 tid = mAudioWatchdog->getTid(); 3562 sendPrioConfigEvent(getpid_cached, tid, kPriorityFastMixer); 3563#endif 3564 3565 } 3566 3567 switch (kUseFastMixer) { 3568 case FastMixer_Never: 3569 case FastMixer_Dynamic: 3570 mNormalSink = mOutputSink; 3571 break; 3572 case FastMixer_Always: 3573 mNormalSink = mPipeSink; 3574 break; 3575 case FastMixer_Static: 3576 mNormalSink = initFastMixer ? mPipeSink : mOutputSink; 3577 break; 3578 } 3579} 3580 3581AudioFlinger::MixerThread::~MixerThread() 3582{ 3583 if (mFastMixer != 0) { 3584 FastMixerStateQueue *sq = mFastMixer->sq(); 3585 FastMixerState *state = sq->begin(); 3586 if (state->mCommand == FastMixerState::COLD_IDLE) { 3587 int32_t old = android_atomic_inc(&mFastMixerFutex); 3588 if (old == -1) { 3589 (void) syscall(__NR_futex, &mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 3590 } 3591 } 3592 state->mCommand = FastMixerState::EXIT; 3593 sq->end(); 3594 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3595 mFastMixer->join(); 3596 // Though the fast mixer thread has exited, it's state queue is still valid. 3597 // We'll use that extract the final state which contains one remaining fast track 3598 // corresponding to our sub-mix. 3599 state = sq->begin(); 3600 ALOG_ASSERT(state->mTrackMask == 1); 3601 FastTrack *fastTrack = &state->mFastTracks[0]; 3602 ALOG_ASSERT(fastTrack->mBufferProvider != NULL); 3603 delete fastTrack->mBufferProvider; 3604 sq->end(false /*didModify*/); 3605 mFastMixer.clear(); 3606#ifdef AUDIO_WATCHDOG 3607 if (mAudioWatchdog != 0) { 3608 mAudioWatchdog->requestExit(); 3609 mAudioWatchdog->requestExitAndWait(); 3610 mAudioWatchdog.clear(); 3611 } 3612#endif 3613 } 3614 mAudioFlinger->unregisterWriter(mFastMixerNBLogWriter); 3615 delete mAudioMixer; 3616} 3617 3618 3619uint32_t AudioFlinger::MixerThread::correctLatency_l(uint32_t latency) const 3620{ 3621 if (mFastMixer != 0) { 3622 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 3623 latency += (pipe->getAvgFrames() * 1000) / mSampleRate; 3624 } 3625 return latency; 3626} 3627 3628 3629void AudioFlinger::MixerThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) 3630{ 3631 PlaybackThread::threadLoop_removeTracks(tracksToRemove); 3632} 3633 3634ssize_t AudioFlinger::MixerThread::threadLoop_write() 3635{ 3636 // FIXME we should only do one push per cycle; confirm this is true 3637 // Start the fast mixer if it's not already running 3638 if (mFastMixer != 0) { 3639 FastMixerStateQueue *sq = mFastMixer->sq(); 3640 FastMixerState *state = sq->begin(); 3641 if (state->mCommand != FastMixerState::MIX_WRITE && 3642 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)) { 3643 if (state->mCommand == FastMixerState::COLD_IDLE) { 3644 3645 // FIXME workaround for first HAL write being CPU bound on some devices 3646 ATRACE_BEGIN("write"); 3647 mOutput->write((char *)mSinkBuffer, 0); 3648 ATRACE_END(); 3649 3650 int32_t old = android_atomic_inc(&mFastMixerFutex); 3651 if (old == -1) { 3652 (void) syscall(__NR_futex, &mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 3653 } 3654#ifdef AUDIO_WATCHDOG 3655 if (mAudioWatchdog != 0) { 3656 mAudioWatchdog->resume(); 3657 } 3658#endif 3659 } 3660 state->mCommand = FastMixerState::MIX_WRITE; 3661#ifdef FAST_THREAD_STATISTICS 3662 mFastMixerDumpState.increaseSamplingN(mAudioFlinger->isLowRamDevice() ? 3663 FastThreadDumpState::kSamplingNforLowRamDevice : FastThreadDumpState::kSamplingN); 3664#endif 3665 sq->end(); 3666 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3667 if (kUseFastMixer == FastMixer_Dynamic) { 3668 mNormalSink = mPipeSink; 3669 } 3670 } else { 3671 sq->end(false /*didModify*/); 3672 } 3673 } 3674 return PlaybackThread::threadLoop_write(); 3675} 3676 3677void AudioFlinger::MixerThread::threadLoop_standby() 3678{ 3679 // Idle the fast mixer if it's currently running 3680 if (mFastMixer != 0) { 3681 FastMixerStateQueue *sq = mFastMixer->sq(); 3682 FastMixerState *state = sq->begin(); 3683 if (!(state->mCommand & FastMixerState::IDLE)) { 3684 state->mCommand = FastMixerState::COLD_IDLE; 3685 state->mColdFutexAddr = &mFastMixerFutex; 3686 state->mColdGen++; 3687 mFastMixerFutex = 0; 3688 sq->end(); 3689 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 3690 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3691 if (kUseFastMixer == FastMixer_Dynamic) { 3692 mNormalSink = mOutputSink; 3693 } 3694#ifdef AUDIO_WATCHDOG 3695 if (mAudioWatchdog != 0) { 3696 mAudioWatchdog->pause(); 3697 } 3698#endif 3699 } else { 3700 sq->end(false /*didModify*/); 3701 } 3702 } 3703 PlaybackThread::threadLoop_standby(); 3704} 3705 3706bool AudioFlinger::PlaybackThread::waitingAsyncCallback_l() 3707{ 3708 return false; 3709} 3710 3711bool AudioFlinger::PlaybackThread::shouldStandby_l() 3712{ 3713 return !mStandby; 3714} 3715 3716bool AudioFlinger::PlaybackThread::waitingAsyncCallback() 3717{ 3718 Mutex::Autolock _l(mLock); 3719 return waitingAsyncCallback_l(); 3720} 3721 3722// shared by MIXER and DIRECT, overridden by DUPLICATING 3723void AudioFlinger::PlaybackThread::threadLoop_standby() 3724{ 3725 ALOGV("Audio hardware entering standby, mixer %p, suspend count %d", this, mSuspended); 3726 mOutput->standby(); 3727 if (mUseAsyncWrite != 0) { 3728 // discard any pending drain or write ack by incrementing sequence 3729 mWriteAckSequence = (mWriteAckSequence + 2) & ~1; 3730 mDrainSequence = (mDrainSequence + 2) & ~1; 3731 ALOG_ASSERT(mCallbackThread != 0); 3732 mCallbackThread->setWriteBlocked(mWriteAckSequence); 3733 mCallbackThread->setDraining(mDrainSequence); 3734 } 3735 mHwPaused = false; 3736} 3737 3738void AudioFlinger::PlaybackThread::onAddNewTrack_l() 3739{ 3740 ALOGV("signal playback thread"); 3741 broadcast_l(); 3742} 3743 3744void AudioFlinger::MixerThread::threadLoop_mix() 3745{ 3746 // mix buffers... 3747 mAudioMixer->process(); 3748 mCurrentWriteLength = mSinkBufferSize; 3749 // increase sleep time progressively when application underrun condition clears. 3750 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 3751 // that a steady state of alternating ready/not ready conditions keeps the sleep time 3752 // such that we would underrun the audio HAL. 3753 if ((mSleepTimeUs == 0) && (sleepTimeShift > 0)) { 3754 sleepTimeShift--; 3755 } 3756 mSleepTimeUs = 0; 3757 mStandbyTimeNs = systemTime() + mStandbyDelayNs; 3758 //TODO: delay standby when effects have a tail 3759 3760} 3761 3762void AudioFlinger::MixerThread::threadLoop_sleepTime() 3763{ 3764 // If no tracks are ready, sleep once for the duration of an output 3765 // buffer size, then write 0s to the output 3766 if (mSleepTimeUs == 0) { 3767 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3768 mSleepTimeUs = mActiveSleepTimeUs >> sleepTimeShift; 3769 if (mSleepTimeUs < kMinThreadSleepTimeUs) { 3770 mSleepTimeUs = kMinThreadSleepTimeUs; 3771 } 3772 // reduce sleep time in case of consecutive application underruns to avoid 3773 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 3774 // duration we would end up writing less data than needed by the audio HAL if 3775 // the condition persists. 3776 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 3777 sleepTimeShift++; 3778 } 3779 } else { 3780 mSleepTimeUs = mIdleSleepTimeUs; 3781 } 3782 } else if (mBytesWritten != 0 || (mMixerStatus == MIXER_TRACKS_ENABLED)) { 3783 // clear out mMixerBuffer or mSinkBuffer, to ensure buffers are cleared 3784 // before effects processing or output. 3785 if (mMixerBufferValid) { 3786 memset(mMixerBuffer, 0, mMixerBufferSize); 3787 } else { 3788 memset(mSinkBuffer, 0, mSinkBufferSize); 3789 } 3790 mSleepTimeUs = 0; 3791 ALOGV_IF(mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED), 3792 "anticipated start"); 3793 } 3794 // TODO add standby time extension fct of effect tail 3795} 3796 3797// prepareTracks_l() must be called with ThreadBase::mLock held 3798AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 3799 Vector< sp<Track> > *tracksToRemove) 3800{ 3801 3802 mixer_state mixerStatus = MIXER_IDLE; 3803 // find out which tracks need to be processed 3804 size_t count = mActiveTracks.size(); 3805 size_t mixedTracks = 0; 3806 size_t tracksWithEffect = 0; 3807 // counts only _active_ fast tracks 3808 size_t fastTracks = 0; 3809 uint32_t resetMask = 0; // bit mask of fast tracks that need to be reset 3810 3811 float masterVolume = mMasterVolume; 3812 bool masterMute = mMasterMute; 3813 3814 if (masterMute) { 3815 masterVolume = 0; 3816 } 3817 // Delegate master volume control to effect in output mix effect chain if needed 3818 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 3819 if (chain != 0) { 3820 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 3821 chain->setVolume_l(&v, &v); 3822 masterVolume = (float)((v + (1 << 23)) >> 24); 3823 chain.clear(); 3824 } 3825 3826 // prepare a new state to push 3827 FastMixerStateQueue *sq = NULL; 3828 FastMixerState *state = NULL; 3829 bool didModify = false; 3830 FastMixerStateQueue::block_t block = FastMixerStateQueue::BLOCK_UNTIL_PUSHED; 3831 if (mFastMixer != 0) { 3832 sq = mFastMixer->sq(); 3833 state = sq->begin(); 3834 } 3835 3836 mMixerBufferValid = false; // mMixerBuffer has no valid data until appropriate tracks found. 3837 mEffectBufferValid = false; // mEffectBuffer has no valid data until tracks found. 3838 3839 for (size_t i=0 ; i<count ; i++) { 3840 const sp<Track> t = mActiveTracks[i].promote(); 3841 if (t == 0) { 3842 continue; 3843 } 3844 3845 // this const just means the local variable doesn't change 3846 Track* const track = t.get(); 3847 3848 // process fast tracks 3849 if (track->isFastTrack()) { 3850 3851 // It's theoretically possible (though unlikely) for a fast track to be created 3852 // and then removed within the same normal mix cycle. This is not a problem, as 3853 // the track never becomes active so it's fast mixer slot is never touched. 3854 // The converse, of removing an (active) track and then creating a new track 3855 // at the identical fast mixer slot within the same normal mix cycle, 3856 // is impossible because the slot isn't marked available until the end of each cycle. 3857 int j = track->mFastIndex; 3858 ALOG_ASSERT(0 < j && j < (int)FastMixerState::kMaxFastTracks); 3859 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << j))); 3860 FastTrack *fastTrack = &state->mFastTracks[j]; 3861 3862 // Determine whether the track is currently in underrun condition, 3863 // and whether it had a recent underrun. 3864 FastTrackDump *ftDump = &mFastMixerDumpState.mTracks[j]; 3865 FastTrackUnderruns underruns = ftDump->mUnderruns; 3866 uint32_t recentFull = (underruns.mBitFields.mFull - 3867 track->mObservedUnderruns.mBitFields.mFull) & UNDERRUN_MASK; 3868 uint32_t recentPartial = (underruns.mBitFields.mPartial - 3869 track->mObservedUnderruns.mBitFields.mPartial) & UNDERRUN_MASK; 3870 uint32_t recentEmpty = (underruns.mBitFields.mEmpty - 3871 track->mObservedUnderruns.mBitFields.mEmpty) & UNDERRUN_MASK; 3872 uint32_t recentUnderruns = recentPartial + recentEmpty; 3873 track->mObservedUnderruns = underruns; 3874 // don't count underruns that occur while stopping or pausing 3875 // or stopped which can occur when flush() is called while active 3876 if (!(track->isStopping() || track->isPausing() || track->isStopped()) && 3877 recentUnderruns > 0) { 3878 // FIXME fast mixer will pull & mix partial buffers, but we count as a full underrun 3879 track->mAudioTrackServerProxy->tallyUnderrunFrames(recentUnderruns * mFrameCount); 3880 } else { 3881 track->mAudioTrackServerProxy->tallyUnderrunFrames(0); 3882 } 3883 3884 // This is similar to the state machine for normal tracks, 3885 // with a few modifications for fast tracks. 3886 bool isActive = true; 3887 switch (track->mState) { 3888 case TrackBase::STOPPING_1: 3889 // track stays active in STOPPING_1 state until first underrun 3890 if (recentUnderruns > 0 || track->isTerminated()) { 3891 track->mState = TrackBase::STOPPING_2; 3892 } 3893 break; 3894 case TrackBase::PAUSING: 3895 // ramp down is not yet implemented 3896 track->setPaused(); 3897 break; 3898 case TrackBase::RESUMING: 3899 // ramp up is not yet implemented 3900 track->mState = TrackBase::ACTIVE; 3901 break; 3902 case TrackBase::ACTIVE: 3903 if (recentFull > 0 || recentPartial > 0) { 3904 // track has provided at least some frames recently: reset retry count 3905 track->mRetryCount = kMaxTrackRetries; 3906 } 3907 if (recentUnderruns == 0) { 3908 // no recent underruns: stay active 3909 break; 3910 } 3911 // there has recently been an underrun of some kind 3912 if (track->sharedBuffer() == 0) { 3913 // were any of the recent underruns "empty" (no frames available)? 3914 if (recentEmpty == 0) { 3915 // no, then ignore the partial underruns as they are allowed indefinitely 3916 break; 3917 } 3918 // there has recently been an "empty" underrun: decrement the retry counter 3919 if (--(track->mRetryCount) > 0) { 3920 break; 3921 } 3922 // indicate to client process that the track was disabled because of underrun; 3923 // it will then automatically call start() when data is available 3924 track->disable(); 3925 // remove from active list, but state remains ACTIVE [confusing but true] 3926 isActive = false; 3927 break; 3928 } 3929 // fall through 3930 case TrackBase::STOPPING_2: 3931 case TrackBase::PAUSED: 3932 case TrackBase::STOPPED: 3933 case TrackBase::FLUSHED: // flush() while active 3934 // Check for presentation complete if track is inactive 3935 // We have consumed all the buffers of this track. 3936 // This would be incomplete if we auto-paused on underrun 3937 { 3938 size_t audioHALFrames = 3939 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 3940 int64_t framesWritten = mBytesWritten / mFrameSize; 3941 if (!(mStandby || track->presentationComplete(framesWritten, audioHALFrames))) { 3942 // track stays in active list until presentation is complete 3943 break; 3944 } 3945 } 3946 if (track->isStopping_2()) { 3947 track->mState = TrackBase::STOPPED; 3948 } 3949 if (track->isStopped()) { 3950 // Can't reset directly, as fast mixer is still polling this track 3951 // track->reset(); 3952 // So instead mark this track as needing to be reset after push with ack 3953 resetMask |= 1 << i; 3954 } 3955 isActive = false; 3956 break; 3957 case TrackBase::IDLE: 3958 default: 3959 LOG_ALWAYS_FATAL("unexpected track state %d", track->mState); 3960 } 3961 3962 if (isActive) { 3963 // was it previously inactive? 3964 if (!(state->mTrackMask & (1 << j))) { 3965 ExtendedAudioBufferProvider *eabp = track; 3966 VolumeProvider *vp = track; 3967 fastTrack->mBufferProvider = eabp; 3968 fastTrack->mVolumeProvider = vp; 3969 fastTrack->mChannelMask = track->mChannelMask; 3970 fastTrack->mFormat = track->mFormat; 3971 fastTrack->mGeneration++; 3972 state->mTrackMask |= 1 << j; 3973 didModify = true; 3974 // no acknowledgement required for newly active tracks 3975 } 3976 // cache the combined master volume and stream type volume for fast mixer; this 3977 // lacks any synchronization or barrier so VolumeProvider may read a stale value 3978 track->mCachedVolume = masterVolume * mStreamTypes[track->streamType()].volume; 3979 ++fastTracks; 3980 } else { 3981 // was it previously active? 3982 if (state->mTrackMask & (1 << j)) { 3983 fastTrack->mBufferProvider = NULL; 3984 fastTrack->mGeneration++; 3985 state->mTrackMask &= ~(1 << j); 3986 didModify = true; 3987 // If any fast tracks were removed, we must wait for acknowledgement 3988 // because we're about to decrement the last sp<> on those tracks. 3989 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3990 } else { 3991 LOG_ALWAYS_FATAL("fast track %d should have been active; " 3992 "mState=%d, mTrackMask=%#x, recentUnderruns=%u, isShared=%d", 3993 j, track->mState, state->mTrackMask, recentUnderruns, 3994 track->sharedBuffer() != 0); 3995 } 3996 tracksToRemove->add(track); 3997 // Avoids a misleading display in dumpsys 3998 track->mObservedUnderruns.mBitFields.mMostRecent = UNDERRUN_FULL; 3999 } 4000 continue; 4001 } 4002 4003 { // local variable scope to avoid goto warning 4004 4005 audio_track_cblk_t* cblk = track->cblk(); 4006 4007 // The first time a track is added we wait 4008 // for all its buffers to be filled before processing it 4009 int name = track->name(); 4010 // make sure that we have enough frames to mix one full buffer. 4011 // enforce this condition only once to enable draining the buffer in case the client 4012 // app does not call stop() and relies on underrun to stop: 4013 // hence the test on (mMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 4014 // during last round 4015 size_t desiredFrames; 4016 const uint32_t sampleRate = track->mAudioTrackServerProxy->getSampleRate(); 4017 AudioPlaybackRate playbackRate = track->mAudioTrackServerProxy->getPlaybackRate(); 4018 4019 desiredFrames = sourceFramesNeededWithTimestretch( 4020 sampleRate, mNormalFrameCount, mSampleRate, playbackRate.mSpeed); 4021 // TODO: ONLY USED FOR LEGACY RESAMPLERS, remove when they are removed. 4022 // add frames already consumed but not yet released by the resampler 4023 // because mAudioTrackServerProxy->framesReady() will include these frames 4024 desiredFrames += mAudioMixer->getUnreleasedFrames(track->name()); 4025 4026 uint32_t minFrames = 1; 4027 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() && 4028 (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) { 4029 minFrames = desiredFrames; 4030 } 4031 4032 size_t framesReady = track->framesReady(); 4033 if (ATRACE_ENABLED()) { 4034 // I wish we had formatted trace names 4035 char traceName[16]; 4036 strcpy(traceName, "nRdy"); 4037 int name = track->name(); 4038 if (AudioMixer::TRACK0 <= name && 4039 name < (int) (AudioMixer::TRACK0 + AudioMixer::MAX_NUM_TRACKS)) { 4040 name -= AudioMixer::TRACK0; 4041 traceName[4] = (name / 10) + '0'; 4042 traceName[5] = (name % 10) + '0'; 4043 } else { 4044 traceName[4] = '?'; 4045 traceName[5] = '?'; 4046 } 4047 traceName[6] = '\0'; 4048 ATRACE_INT(traceName, framesReady); 4049 } 4050 if ((framesReady >= minFrames) && track->isReady() && 4051 !track->isPaused() && !track->isTerminated()) 4052 { 4053 ALOGVV("track %d s=%08x [OK] on thread %p", name, cblk->mServer, this); 4054 4055 mixedTracks++; 4056 4057 // track->mainBuffer() != mSinkBuffer or mMixerBuffer means 4058 // there is an effect chain connected to the track 4059 chain.clear(); 4060 if (track->mainBuffer() != mSinkBuffer && 4061 track->mainBuffer() != mMixerBuffer) { 4062 if (mEffectBufferEnabled) { 4063 mEffectBufferValid = true; // Later can set directly. 4064 } 4065 chain = getEffectChain_l(track->sessionId()); 4066 // Delegate volume control to effect in track effect chain if needed 4067 if (chain != 0) { 4068 tracksWithEffect++; 4069 } else { 4070 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on " 4071 "session %d", 4072 name, track->sessionId()); 4073 } 4074 } 4075 4076 4077 int param = AudioMixer::VOLUME; 4078 if (track->mFillingUpStatus == Track::FS_FILLED) { 4079 // no ramp for the first volume setting 4080 track->mFillingUpStatus = Track::FS_ACTIVE; 4081 if (track->mState == TrackBase::RESUMING) { 4082 track->mState = TrackBase::ACTIVE; 4083 param = AudioMixer::RAMP_VOLUME; 4084 } 4085 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 4086 // FIXME should not make a decision based on mServer 4087 } else if (cblk->mServer != 0) { 4088 // If the track is stopped before the first frame was mixed, 4089 // do not apply ramp 4090 param = AudioMixer::RAMP_VOLUME; 4091 } 4092 4093 // compute volume for this track 4094 uint32_t vl, vr; // in U8.24 integer format 4095 float vlf, vrf, vaf; // in [0.0, 1.0] float format 4096 if (track->isPausing() || mStreamTypes[track->streamType()].mute) { 4097 vl = vr = 0; 4098 vlf = vrf = vaf = 0.; 4099 if (track->isPausing()) { 4100 track->setPaused(); 4101 } 4102 } else { 4103 4104 // read original volumes with volume control 4105 float typeVolume = mStreamTypes[track->streamType()].volume; 4106 float v = masterVolume * typeVolume; 4107 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 4108 gain_minifloat_packed_t vlr = proxy->getVolumeLR(); 4109 vlf = float_from_gain(gain_minifloat_unpack_left(vlr)); 4110 vrf = float_from_gain(gain_minifloat_unpack_right(vlr)); 4111 // track volumes come from shared memory, so can't be trusted and must be clamped 4112 if (vlf > GAIN_FLOAT_UNITY) { 4113 ALOGV("Track left volume out of range: %.3g", vlf); 4114 vlf = GAIN_FLOAT_UNITY; 4115 } 4116 if (vrf > GAIN_FLOAT_UNITY) { 4117 ALOGV("Track right volume out of range: %.3g", vrf); 4118 vrf = GAIN_FLOAT_UNITY; 4119 } 4120 // now apply the master volume and stream type volume 4121 vlf *= v; 4122 vrf *= v; 4123 // assuming master volume and stream type volume each go up to 1.0, 4124 // then derive vl and vr as U8.24 versions for the effect chain 4125 const float scaleto8_24 = MAX_GAIN_INT * MAX_GAIN_INT; 4126 vl = (uint32_t) (scaleto8_24 * vlf); 4127 vr = (uint32_t) (scaleto8_24 * vrf); 4128 // vl and vr are now in U8.24 format 4129 uint16_t sendLevel = proxy->getSendLevel_U4_12(); 4130 // send level comes from shared memory and so may be corrupt 4131 if (sendLevel > MAX_GAIN_INT) { 4132 ALOGV("Track send level out of range: %04X", sendLevel); 4133 sendLevel = MAX_GAIN_INT; 4134 } 4135 // vaf is represented as [0.0, 1.0] float by rescaling sendLevel 4136 vaf = v * sendLevel * (1. / MAX_GAIN_INT); 4137 } 4138 4139 // Delegate volume control to effect in track effect chain if needed 4140 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 4141 // Do not ramp volume if volume is controlled by effect 4142 param = AudioMixer::VOLUME; 4143 // Update remaining floating point volume levels 4144 vlf = (float)vl / (1 << 24); 4145 vrf = (float)vr / (1 << 24); 4146 track->mHasVolumeController = true; 4147 } else { 4148 // force no volume ramp when volume controller was just disabled or removed 4149 // from effect chain to avoid volume spike 4150 if (track->mHasVolumeController) { 4151 param = AudioMixer::VOLUME; 4152 } 4153 track->mHasVolumeController = false; 4154 } 4155 4156 // XXX: these things DON'T need to be done each time 4157 mAudioMixer->setBufferProvider(name, track); 4158 mAudioMixer->enable(name); 4159 4160 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, &vlf); 4161 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, &vrf); 4162 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, &vaf); 4163 mAudioMixer->setParameter( 4164 name, 4165 AudioMixer::TRACK, 4166 AudioMixer::FORMAT, (void *)track->format()); 4167 mAudioMixer->setParameter( 4168 name, 4169 AudioMixer::TRACK, 4170 AudioMixer::CHANNEL_MASK, (void *)(uintptr_t)track->channelMask()); 4171 mAudioMixer->setParameter( 4172 name, 4173 AudioMixer::TRACK, 4174 AudioMixer::MIXER_CHANNEL_MASK, (void *)(uintptr_t)mChannelMask); 4175 // limit track sample rate to 2 x output sample rate, which changes at re-configuration 4176 uint32_t maxSampleRate = mSampleRate * AUDIO_RESAMPLER_DOWN_RATIO_MAX; 4177 uint32_t reqSampleRate = track->mAudioTrackServerProxy->getSampleRate(); 4178 if (reqSampleRate == 0) { 4179 reqSampleRate = mSampleRate; 4180 } else if (reqSampleRate > maxSampleRate) { 4181 reqSampleRate = maxSampleRate; 4182 } 4183 mAudioMixer->setParameter( 4184 name, 4185 AudioMixer::RESAMPLE, 4186 AudioMixer::SAMPLE_RATE, 4187 (void *)(uintptr_t)reqSampleRate); 4188 4189 AudioPlaybackRate playbackRate = track->mAudioTrackServerProxy->getPlaybackRate(); 4190 mAudioMixer->setParameter( 4191 name, 4192 AudioMixer::TIMESTRETCH, 4193 AudioMixer::PLAYBACK_RATE, 4194 &playbackRate); 4195 4196 /* 4197 * Select the appropriate output buffer for the track. 4198 * 4199 * Tracks with effects go into their own effects chain buffer 4200 * and from there into either mEffectBuffer or mSinkBuffer. 4201 * 4202 * Other tracks can use mMixerBuffer for higher precision 4203 * channel accumulation. If this buffer is enabled 4204 * (mMixerBufferEnabled true), then selected tracks will accumulate 4205 * into it. 4206 * 4207 */ 4208 if (mMixerBufferEnabled 4209 && (track->mainBuffer() == mSinkBuffer 4210 || track->mainBuffer() == mMixerBuffer)) { 4211 mAudioMixer->setParameter( 4212 name, 4213 AudioMixer::TRACK, 4214 AudioMixer::MIXER_FORMAT, (void *)mMixerBufferFormat); 4215 mAudioMixer->setParameter( 4216 name, 4217 AudioMixer::TRACK, 4218 AudioMixer::MAIN_BUFFER, (void *)mMixerBuffer); 4219 // TODO: override track->mainBuffer()? 4220 mMixerBufferValid = true; 4221 } else { 4222 mAudioMixer->setParameter( 4223 name, 4224 AudioMixer::TRACK, 4225 AudioMixer::MIXER_FORMAT, (void *)AUDIO_FORMAT_PCM_16_BIT); 4226 mAudioMixer->setParameter( 4227 name, 4228 AudioMixer::TRACK, 4229 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 4230 } 4231 mAudioMixer->setParameter( 4232 name, 4233 AudioMixer::TRACK, 4234 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 4235 4236 // reset retry count 4237 track->mRetryCount = kMaxTrackRetries; 4238 4239 // If one track is ready, set the mixer ready if: 4240 // - the mixer was not ready during previous round OR 4241 // - no other track is not ready 4242 if (mMixerStatusIgnoringFastTracks != MIXER_TRACKS_READY || 4243 mixerStatus != MIXER_TRACKS_ENABLED) { 4244 mixerStatus = MIXER_TRACKS_READY; 4245 } 4246 } else { 4247 if (framesReady < desiredFrames && !track->isStopped() && !track->isPaused()) { 4248 ALOGV("track(%p) underrun, framesReady(%zu) < framesDesired(%zd)", 4249 track, framesReady, desiredFrames); 4250 track->mAudioTrackServerProxy->tallyUnderrunFrames(desiredFrames); 4251 } else { 4252 track->mAudioTrackServerProxy->tallyUnderrunFrames(0); 4253 } 4254 4255 // clear effect chain input buffer if an active track underruns to avoid sending 4256 // previous audio buffer again to effects 4257 chain = getEffectChain_l(track->sessionId()); 4258 if (chain != 0) { 4259 chain->clearInputBuffer(); 4260 } 4261 4262 ALOGVV("track %d s=%08x [NOT READY] on thread %p", name, cblk->mServer, this); 4263 if ((track->sharedBuffer() != 0) || track->isTerminated() || 4264 track->isStopped() || track->isPaused()) { 4265 // We have consumed all the buffers of this track. 4266 // Remove it from the list of active tracks. 4267 // TODO: use actual buffer filling status instead of latency when available from 4268 // audio HAL 4269 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 4270 int64_t framesWritten = mBytesWritten / mFrameSize; 4271 if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) { 4272 if (track->isStopped()) { 4273 track->reset(); 4274 } 4275 tracksToRemove->add(track); 4276 } 4277 } else { 4278 // No buffers for this track. Give it a few chances to 4279 // fill a buffer, then remove it from active list. 4280 if (--(track->mRetryCount) <= 0) { 4281 ALOGI("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 4282 tracksToRemove->add(track); 4283 // indicate to client process that the track was disabled because of underrun; 4284 // it will then automatically call start() when data is available 4285 track->disable(); 4286 // If one track is not ready, mark the mixer also not ready if: 4287 // - the mixer was ready during previous round OR 4288 // - no other track is ready 4289 } else if (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY || 4290 mixerStatus != MIXER_TRACKS_READY) { 4291 mixerStatus = MIXER_TRACKS_ENABLED; 4292 } 4293 } 4294 mAudioMixer->disable(name); 4295 } 4296 4297 } // local variable scope to avoid goto warning 4298 4299 } 4300 4301 // Push the new FastMixer state if necessary 4302 bool pauseAudioWatchdog = false; 4303 if (didModify) { 4304 state->mFastTracksGen++; 4305 // if the fast mixer was active, but now there are no fast tracks, then put it in cold idle 4306 if (kUseFastMixer == FastMixer_Dynamic && 4307 state->mCommand == FastMixerState::MIX_WRITE && state->mTrackMask <= 1) { 4308 state->mCommand = FastMixerState::COLD_IDLE; 4309 state->mColdFutexAddr = &mFastMixerFutex; 4310 state->mColdGen++; 4311 mFastMixerFutex = 0; 4312 if (kUseFastMixer == FastMixer_Dynamic) { 4313 mNormalSink = mOutputSink; 4314 } 4315 // If we go into cold idle, need to wait for acknowledgement 4316 // so that fast mixer stops doing I/O. 4317 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 4318 pauseAudioWatchdog = true; 4319 } 4320 } 4321 if (sq != NULL) { 4322 sq->end(didModify); 4323 sq->push(block); 4324 } 4325#ifdef AUDIO_WATCHDOG 4326 if (pauseAudioWatchdog && mAudioWatchdog != 0) { 4327 mAudioWatchdog->pause(); 4328 } 4329#endif 4330 4331 // Now perform the deferred reset on fast tracks that have stopped 4332 while (resetMask != 0) { 4333 size_t i = __builtin_ctz(resetMask); 4334 ALOG_ASSERT(i < count); 4335 resetMask &= ~(1 << i); 4336 sp<Track> t = mActiveTracks[i].promote(); 4337 if (t == 0) { 4338 continue; 4339 } 4340 Track* track = t.get(); 4341 ALOG_ASSERT(track->isFastTrack() && track->isStopped()); 4342 track->reset(); 4343 } 4344 4345 // remove all the tracks that need to be... 4346 removeTracks_l(*tracksToRemove); 4347 4348 if (getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX) != 0) { 4349 mEffectBufferValid = true; 4350 } 4351 4352 if (mEffectBufferValid) { 4353 // as long as there are effects we should clear the effects buffer, to avoid 4354 // passing a non-clean buffer to the effect chain 4355 memset(mEffectBuffer, 0, mEffectBufferSize); 4356 } 4357 // sink or mix buffer must be cleared if all tracks are connected to an 4358 // effect chain as in this case the mixer will not write to the sink or mix buffer 4359 // and track effects will accumulate into it 4360 if ((mBytesRemaining == 0) && ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 4361 (mixedTracks == 0 && fastTracks > 0))) { 4362 // FIXME as a performance optimization, should remember previous zero status 4363 if (mMixerBufferValid) { 4364 memset(mMixerBuffer, 0, mMixerBufferSize); 4365 // TODO: In testing, mSinkBuffer below need not be cleared because 4366 // the PlaybackThread::threadLoop() copies mMixerBuffer into mSinkBuffer 4367 // after mixing. 4368 // 4369 // To enforce this guarantee: 4370 // ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 4371 // (mixedTracks == 0 && fastTracks > 0)) 4372 // must imply MIXER_TRACKS_READY. 4373 // Later, we may clear buffers regardless, and skip much of this logic. 4374 } 4375 // FIXME as a performance optimization, should remember previous zero status 4376 memset(mSinkBuffer, 0, mNormalFrameCount * mFrameSize); 4377 } 4378 4379 // if any fast tracks, then status is ready 4380 mMixerStatusIgnoringFastTracks = mixerStatus; 4381 if (fastTracks > 0) { 4382 mixerStatus = MIXER_TRACKS_READY; 4383 } 4384 return mixerStatus; 4385} 4386 4387// getTrackName_l() must be called with ThreadBase::mLock held 4388int AudioFlinger::MixerThread::getTrackName_l(audio_channel_mask_t channelMask, 4389 audio_format_t format, audio_session_t sessionId) 4390{ 4391 return mAudioMixer->getTrackName(channelMask, format, sessionId); 4392} 4393 4394// deleteTrackName_l() must be called with ThreadBase::mLock held 4395void AudioFlinger::MixerThread::deleteTrackName_l(int name) 4396{ 4397 ALOGV("remove track (%d) and delete from mixer", name); 4398 mAudioMixer->deleteTrackName(name); 4399} 4400 4401// checkForNewParameter_l() must be called with ThreadBase::mLock held 4402bool AudioFlinger::MixerThread::checkForNewParameter_l(const String8& keyValuePair, 4403 status_t& status) 4404{ 4405 bool reconfig = false; 4406 bool a2dpDeviceChanged = false; 4407 4408 status = NO_ERROR; 4409 4410 AutoPark<FastMixer> park(mFastMixer); 4411 4412 AudioParameter param = AudioParameter(keyValuePair); 4413 int value; 4414 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 4415 reconfig = true; 4416 } 4417 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 4418 if (!isValidPcmSinkFormat((audio_format_t) value)) { 4419 status = BAD_VALUE; 4420 } else { 4421 // no need to save value, since it's constant 4422 reconfig = true; 4423 } 4424 } 4425 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 4426 if (!isValidPcmSinkChannelMask((audio_channel_mask_t) value)) { 4427 status = BAD_VALUE; 4428 } else { 4429 // no need to save value, since it's constant 4430 reconfig = true; 4431 } 4432 } 4433 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 4434 // do not accept frame count changes if tracks are open as the track buffer 4435 // size depends on frame count and correct behavior would not be guaranteed 4436 // if frame count is changed after track creation 4437 if (!mTracks.isEmpty()) { 4438 status = INVALID_OPERATION; 4439 } else { 4440 reconfig = true; 4441 } 4442 } 4443 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 4444#ifdef ADD_BATTERY_DATA 4445 // when changing the audio output device, call addBatteryData to notify 4446 // the change 4447 if (mOutDevice != value) { 4448 uint32_t params = 0; 4449 // check whether speaker is on 4450 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 4451 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 4452 } 4453 4454 audio_devices_t deviceWithoutSpeaker 4455 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 4456 // check if any other device (except speaker) is on 4457 if (value & deviceWithoutSpeaker) { 4458 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 4459 } 4460 4461 if (params != 0) { 4462 addBatteryData(params); 4463 } 4464 } 4465#endif 4466 4467 // forward device change to effects that have requested to be 4468 // aware of attached audio device. 4469 if (value != AUDIO_DEVICE_NONE) { 4470 a2dpDeviceChanged = 4471 (mOutDevice & AUDIO_DEVICE_OUT_ALL_A2DP) != (value & AUDIO_DEVICE_OUT_ALL_A2DP); 4472 mOutDevice = value; 4473 for (size_t i = 0; i < mEffectChains.size(); i++) { 4474 mEffectChains[i]->setDevice_l(mOutDevice); 4475 } 4476 } 4477 } 4478 4479 if (status == NO_ERROR) { 4480 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 4481 keyValuePair.string()); 4482 if (!mStandby && status == INVALID_OPERATION) { 4483 mOutput->standby(); 4484 mStandby = true; 4485 mBytesWritten = 0; 4486 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 4487 keyValuePair.string()); 4488 } 4489 if (status == NO_ERROR && reconfig) { 4490 readOutputParameters_l(); 4491 delete mAudioMixer; 4492 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 4493 for (size_t i = 0; i < mTracks.size() ; i++) { 4494 int name = getTrackName_l(mTracks[i]->mChannelMask, 4495 mTracks[i]->mFormat, mTracks[i]->mSessionId); 4496 if (name < 0) { 4497 break; 4498 } 4499 mTracks[i]->mName = name; 4500 } 4501 sendIoConfigEvent_l(AUDIO_OUTPUT_CONFIG_CHANGED); 4502 } 4503 } 4504 4505 return reconfig || a2dpDeviceChanged; 4506} 4507 4508 4509void AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 4510{ 4511 PlaybackThread::dumpInternals(fd, args); 4512 dprintf(fd, " Thread throttle time (msecs): %u\n", mThreadThrottleTimeMs); 4513 dprintf(fd, " AudioMixer tracks: 0x%08x\n", mAudioMixer->trackNames()); 4514 dprintf(fd, " Master mono: %s\n", mMasterMono ? "on" : "off"); 4515 4516 // Make a non-atomic copy of fast mixer dump state so it won't change underneath us 4517 // while we are dumping it. It may be inconsistent, but it won't mutate! 4518 // This is a large object so we place it on the heap. 4519 // FIXME 25972958: Need an intelligent copy constructor that does not touch unused pages. 4520 const FastMixerDumpState *copy = new FastMixerDumpState(mFastMixerDumpState); 4521 copy->dump(fd); 4522 delete copy; 4523 4524#ifdef STATE_QUEUE_DUMP 4525 // Similar for state queue 4526 StateQueueObserverDump observerCopy = mStateQueueObserverDump; 4527 observerCopy.dump(fd); 4528 StateQueueMutatorDump mutatorCopy = mStateQueueMutatorDump; 4529 mutatorCopy.dump(fd); 4530#endif 4531 4532#ifdef TEE_SINK 4533 // Write the tee output to a .wav file 4534 dumpTee(fd, mTeeSource, mId); 4535#endif 4536 4537#ifdef AUDIO_WATCHDOG 4538 if (mAudioWatchdog != 0) { 4539 // Make a non-atomic copy of audio watchdog dump so it won't change underneath us 4540 AudioWatchdogDump wdCopy = mAudioWatchdogDump; 4541 wdCopy.dump(fd); 4542 } 4543#endif 4544} 4545 4546uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const 4547{ 4548 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000) / 2; 4549} 4550 4551uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() const 4552{ 4553 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000); 4554} 4555 4556void AudioFlinger::MixerThread::cacheParameters_l() 4557{ 4558 PlaybackThread::cacheParameters_l(); 4559 4560 // FIXME: Relaxed timing because of a certain device that can't meet latency 4561 // Should be reduced to 2x after the vendor fixes the driver issue 4562 // increase threshold again due to low power audio mode. The way this warning 4563 // threshold is calculated and its usefulness should be reconsidered anyway. 4564 maxPeriod = seconds(mNormalFrameCount) / mSampleRate * 15; 4565} 4566 4567// ---------------------------------------------------------------------------- 4568 4569AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 4570 AudioStreamOut* output, audio_io_handle_t id, audio_devices_t device, bool systemReady, 4571 uint32_t bitRate) 4572 : PlaybackThread(audioFlinger, output, id, device, DIRECT, systemReady, bitRate) 4573 // mLeftVolFloat, mRightVolFloat 4574{ 4575} 4576 4577AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 4578 AudioStreamOut* output, audio_io_handle_t id, uint32_t device, 4579 ThreadBase::type_t type, bool systemReady, uint32_t bitRate) 4580 : PlaybackThread(audioFlinger, output, id, device, type, systemReady, bitRate) 4581 // mLeftVolFloat, mRightVolFloat 4582{ 4583} 4584 4585AudioFlinger::DirectOutputThread::~DirectOutputThread() 4586{ 4587} 4588 4589void AudioFlinger::DirectOutputThread::processVolume_l(Track *track, bool lastTrack) 4590{ 4591 float left, right; 4592 4593 if (mMasterMute || mStreamTypes[track->streamType()].mute) { 4594 left = right = 0; 4595 } else { 4596 float typeVolume = mStreamTypes[track->streamType()].volume; 4597 float v = mMasterVolume * typeVolume; 4598 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 4599 gain_minifloat_packed_t vlr = proxy->getVolumeLR(); 4600 left = float_from_gain(gain_minifloat_unpack_left(vlr)); 4601 if (left > GAIN_FLOAT_UNITY) { 4602 left = GAIN_FLOAT_UNITY; 4603 } 4604 left *= v; 4605 right = float_from_gain(gain_minifloat_unpack_right(vlr)); 4606 if (right > GAIN_FLOAT_UNITY) { 4607 right = GAIN_FLOAT_UNITY; 4608 } 4609 right *= v; 4610 } 4611 4612 if (lastTrack) { 4613 if (left != mLeftVolFloat || right != mRightVolFloat) { 4614 mLeftVolFloat = left; 4615 mRightVolFloat = right; 4616 4617 // Convert volumes from float to 8.24 4618 uint32_t vl = (uint32_t)(left * (1 << 24)); 4619 uint32_t vr = (uint32_t)(right * (1 << 24)); 4620 4621 // Delegate volume control to effect in track effect chain if needed 4622 // only one effect chain can be present on DirectOutputThread, so if 4623 // there is one, the track is connected to it 4624 if (!mEffectChains.isEmpty()) { 4625 mEffectChains[0]->setVolume_l(&vl, &vr); 4626 left = (float)vl / (1 << 24); 4627 right = (float)vr / (1 << 24); 4628 } 4629 if (mOutput->stream->set_volume) { 4630 mOutput->stream->set_volume(mOutput->stream, left, right); 4631 } 4632 } 4633 } 4634} 4635 4636void AudioFlinger::DirectOutputThread::onAddNewTrack_l() 4637{ 4638 sp<Track> previousTrack = mPreviousTrack.promote(); 4639 sp<Track> latestTrack = mLatestActiveTrack.promote(); 4640 4641 if (previousTrack != 0 && latestTrack != 0) { 4642 if (mType == DIRECT) { 4643 if (previousTrack.get() != latestTrack.get()) { 4644 mFlushPending = true; 4645 } 4646 } else /* mType == OFFLOAD */ { 4647 if (previousTrack->sessionId() != latestTrack->sessionId()) { 4648 mFlushPending = true; 4649 } 4650 } 4651 } 4652 PlaybackThread::onAddNewTrack_l(); 4653} 4654 4655AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l( 4656 Vector< sp<Track> > *tracksToRemove 4657) 4658{ 4659 size_t count = mActiveTracks.size(); 4660 mixer_state mixerStatus = MIXER_IDLE; 4661 bool doHwPause = false; 4662 bool doHwResume = false; 4663 4664 // find out which tracks need to be processed 4665 for (size_t i = 0; i < count; i++) { 4666 sp<Track> t = mActiveTracks[i].promote(); 4667 // The track died recently 4668 if (t == 0) { 4669 continue; 4670 } 4671 4672 if (t->isInvalid()) { 4673 ALOGW("An invalidated track shouldn't be in active list"); 4674 tracksToRemove->add(t); 4675 continue; 4676 } 4677 4678 Track* const track = t.get(); 4679#ifdef VERY_VERY_VERBOSE_LOGGING 4680 audio_track_cblk_t* cblk = track->cblk(); 4681#endif 4682 // Only consider last track started for volume and mixer state control. 4683 // In theory an older track could underrun and restart after the new one starts 4684 // but as we only care about the transition phase between two tracks on a 4685 // direct output, it is not a problem to ignore the underrun case. 4686 sp<Track> l = mLatestActiveTrack.promote(); 4687 bool last = l.get() == track; 4688 4689 if (track->isPausing()) { 4690 track->setPaused(); 4691 if (mHwSupportsPause && last && !mHwPaused) { 4692 doHwPause = true; 4693 mHwPaused = true; 4694 } 4695 tracksToRemove->add(track); 4696 } else if (track->isFlushPending()) { 4697 track->flushAck(); 4698 if (last) { 4699 mFlushPending = true; 4700 } 4701 } else if (track->isResumePending()) { 4702 track->resumeAck(); 4703 if (last && mHwPaused) { 4704 doHwResume = true; 4705 mHwPaused = false; 4706 } 4707 } 4708 4709 // The first time a track is added we wait 4710 // for all its buffers to be filled before processing it. 4711 // Allow draining the buffer in case the client 4712 // app does not call stop() and relies on underrun to stop: 4713 // hence the test on (track->mRetryCount > 1). 4714 // If retryCount<=1 then track is about to underrun and be removed. 4715 // Do not use a high threshold for compressed audio. 4716 uint32_t minFrames; 4717 if ((track->sharedBuffer() == 0) && !track->isStopping_1() && !track->isPausing() 4718 && (track->mRetryCount > 1) && audio_has_proportional_frames(mFormat)) { 4719 minFrames = mNormalFrameCount; 4720 } else { 4721 minFrames = 1; 4722 } 4723 4724 if ((track->framesReady() >= minFrames) && track->isReady() && !track->isPaused() && 4725 !track->isStopping_2() && !track->isStopped()) 4726 { 4727 ALOGVV("track %d s=%08x [OK]", track->name(), cblk->mServer); 4728 4729 if (track->mFillingUpStatus == Track::FS_FILLED) { 4730 track->mFillingUpStatus = Track::FS_ACTIVE; 4731 // make sure processVolume_l() will apply new volume even if 0 4732 mLeftVolFloat = mRightVolFloat = -1.0; 4733 if (!mHwSupportsPause) { 4734 track->resumeAck(); 4735 } 4736 } 4737 4738 // compute volume for this track 4739 processVolume_l(track, last); 4740 if (last) { 4741 sp<Track> previousTrack = mPreviousTrack.promote(); 4742 if (previousTrack != 0) { 4743 if (track != previousTrack.get()) { 4744 // Flush any data still being written from last track 4745 mBytesRemaining = 0; 4746 // Invalidate previous track to force a seek when resuming. 4747 previousTrack->invalidate(); 4748 } 4749 } 4750 mPreviousTrack = track; 4751 4752 // reset retry count 4753 track->mRetryCount = kMaxTrackRetriesDirect; 4754 mActiveTrack = t; 4755 mixerStatus = MIXER_TRACKS_READY; 4756 if (mHwPaused) { 4757 doHwResume = true; 4758 mHwPaused = false; 4759 } 4760 } 4761 } else { 4762 // clear effect chain input buffer if the last active track started underruns 4763 // to avoid sending previous audio buffer again to effects 4764 if (!mEffectChains.isEmpty() && last) { 4765 mEffectChains[0]->clearInputBuffer(); 4766 } 4767 if (track->isStopping_1()) { 4768 track->mState = TrackBase::STOPPING_2; 4769 if (last && mHwPaused) { 4770 doHwResume = true; 4771 mHwPaused = false; 4772 } 4773 } 4774 if ((track->sharedBuffer() != 0) || track->isStopped() || 4775 track->isStopping_2() || track->isPaused()) { 4776 // We have consumed all the buffers of this track. 4777 // Remove it from the list of active tracks. 4778 size_t audioHALFrames; 4779 if (audio_has_proportional_frames(mFormat)) { 4780 audioHALFrames = (latency_l() * mSampleRate) / 1000; 4781 } else { 4782 audioHALFrames = 0; 4783 } 4784 4785 int64_t framesWritten = mBytesWritten / mFrameSize; 4786 if (mStandby || !last || 4787 track->presentationComplete(framesWritten, audioHALFrames)) { 4788 if (track->isStopping_2()) { 4789 track->mState = TrackBase::STOPPED; 4790 } 4791 if (track->isStopped()) { 4792 track->reset(); 4793 } 4794 tracksToRemove->add(track); 4795 } 4796 } else { 4797 // No buffers for this track. Give it a few chances to 4798 // fill a buffer, then remove it from active list. 4799 // Only consider last track started for mixer state control 4800 if (--(track->mRetryCount) <= 0) { 4801 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 4802 tracksToRemove->add(track); 4803 // indicate to client process that the track was disabled because of underrun; 4804 // it will then automatically call start() when data is available 4805 track->disable(); 4806 } else if (last) { 4807 ALOGW("pause because of UNDERRUN, framesReady = %zu," 4808 "minFrames = %u, mFormat = %#x", 4809 track->framesReady(), minFrames, mFormat); 4810 mixerStatus = MIXER_TRACKS_ENABLED; 4811 if (mHwSupportsPause && !mHwPaused && !mStandby) { 4812 doHwPause = true; 4813 mHwPaused = true; 4814 } 4815 } 4816 } 4817 } 4818 } 4819 4820 // if an active track did not command a flush, check for pending flush on stopped tracks 4821 if (!mFlushPending) { 4822 for (size_t i = 0; i < mTracks.size(); i++) { 4823 if (mTracks[i]->isFlushPending()) { 4824 mTracks[i]->flushAck(); 4825 mFlushPending = true; 4826 } 4827 } 4828 } 4829 4830 // make sure the pause/flush/resume sequence is executed in the right order. 4831 // If a flush is pending and a track is active but the HW is not paused, force a HW pause 4832 // before flush and then resume HW. This can happen in case of pause/flush/resume 4833 // if resume is received before pause is executed. 4834 if (mHwSupportsPause && !mStandby && 4835 (doHwPause || (mFlushPending && !mHwPaused && (count != 0)))) { 4836 mOutput->stream->pause(mOutput->stream); 4837 } 4838 if (mFlushPending) { 4839 flushHw_l(); 4840 } 4841 if (mHwSupportsPause && !mStandby && doHwResume) { 4842 mOutput->stream->resume(mOutput->stream); 4843 } 4844 // remove all the tracks that need to be... 4845 removeTracks_l(*tracksToRemove); 4846 4847 return mixerStatus; 4848} 4849 4850void AudioFlinger::DirectOutputThread::threadLoop_mix() 4851{ 4852 size_t frameCount = mFrameCount; 4853 int8_t *curBuf = (int8_t *)mSinkBuffer; 4854 // output audio to hardware 4855 while (frameCount) { 4856 AudioBufferProvider::Buffer buffer; 4857 buffer.frameCount = frameCount; 4858 status_t status = mActiveTrack->getNextBuffer(&buffer); 4859 if (status != NO_ERROR || buffer.raw == NULL) { 4860 // no need to pad with 0 for compressed audio 4861 if (audio_has_proportional_frames(mFormat)) { 4862 memset(curBuf, 0, frameCount * mFrameSize); 4863 } 4864 break; 4865 } 4866 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 4867 frameCount -= buffer.frameCount; 4868 curBuf += buffer.frameCount * mFrameSize; 4869 mActiveTrack->releaseBuffer(&buffer); 4870 } 4871 mCurrentWriteLength = curBuf - (int8_t *)mSinkBuffer; 4872 mSleepTimeUs = 0; 4873 mStandbyTimeNs = systemTime() + mStandbyDelayNs; 4874 mActiveTrack.clear(); 4875} 4876 4877void AudioFlinger::DirectOutputThread::threadLoop_sleepTime() 4878{ 4879 // do not write to HAL when paused 4880 if (mHwPaused || (usesHwAvSync() && mStandby)) { 4881 mSleepTimeUs = mIdleSleepTimeUs; 4882 return; 4883 } 4884 if (mSleepTimeUs == 0) { 4885 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4886 // For compressed offload, use faster sleep time when underruning until more than an 4887 // entire buffer was written to the audio HAL 4888 if (!audio_has_proportional_frames(mFormat) && 4889 (mType == OFFLOAD) && (mBytesWritten < (int64_t) mBufferSize)) { 4890 mSleepTimeUs = kDirectMinSleepTimeUs; 4891 } else { 4892 mSleepTimeUs = mActiveSleepTimeUs; 4893 } 4894 } else { 4895 mSleepTimeUs = mIdleSleepTimeUs; 4896 } 4897 } else if (mBytesWritten != 0 && audio_has_proportional_frames(mFormat)) { 4898 memset(mSinkBuffer, 0, mFrameCount * mFrameSize); 4899 mSleepTimeUs = 0; 4900 } 4901} 4902 4903void AudioFlinger::DirectOutputThread::threadLoop_exit() 4904{ 4905 { 4906 Mutex::Autolock _l(mLock); 4907 for (size_t i = 0; i < mTracks.size(); i++) { 4908 if (mTracks[i]->isFlushPending()) { 4909 mTracks[i]->flushAck(); 4910 mFlushPending = true; 4911 } 4912 } 4913 if (mFlushPending) { 4914 flushHw_l(); 4915 } 4916 } 4917 PlaybackThread::threadLoop_exit(); 4918} 4919 4920// must be called with thread mutex locked 4921bool AudioFlinger::DirectOutputThread::shouldStandby_l() 4922{ 4923 bool trackPaused = false; 4924 bool trackStopped = false; 4925 4926 if ((mType == DIRECT) && audio_is_linear_pcm(mFormat) && !usesHwAvSync()) { 4927 return !mStandby; 4928 } 4929 4930 // do not put the HAL in standby when paused. AwesomePlayer clear the offloaded AudioTrack 4931 // after a timeout and we will enter standby then. 4932 if (mTracks.size() > 0) { 4933 trackPaused = mTracks[mTracks.size() - 1]->isPaused(); 4934 trackStopped = mTracks[mTracks.size() - 1]->isStopped() || 4935 mTracks[mTracks.size() - 1]->mState == TrackBase::IDLE; 4936 } 4937 4938 return !mStandby && !(trackPaused || (mHwPaused && !trackStopped)); 4939} 4940 4941// getTrackName_l() must be called with ThreadBase::mLock held 4942int AudioFlinger::DirectOutputThread::getTrackName_l(audio_channel_mask_t channelMask __unused, 4943 audio_format_t format __unused, audio_session_t sessionId __unused) 4944{ 4945 return 0; 4946} 4947 4948// deleteTrackName_l() must be called with ThreadBase::mLock held 4949void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name __unused) 4950{ 4951} 4952 4953// checkForNewParameter_l() must be called with ThreadBase::mLock held 4954bool AudioFlinger::DirectOutputThread::checkForNewParameter_l(const String8& keyValuePair, 4955 status_t& status) 4956{ 4957 bool reconfig = false; 4958 bool a2dpDeviceChanged = false; 4959 4960 status = NO_ERROR; 4961 4962 AudioParameter param = AudioParameter(keyValuePair); 4963 int value; 4964 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 4965 // forward device change to effects that have requested to be 4966 // aware of attached audio device. 4967 if (value != AUDIO_DEVICE_NONE) { 4968 a2dpDeviceChanged = 4969 (mOutDevice & AUDIO_DEVICE_OUT_ALL_A2DP) != (value & AUDIO_DEVICE_OUT_ALL_A2DP); 4970 mOutDevice = value; 4971 for (size_t i = 0; i < mEffectChains.size(); i++) { 4972 mEffectChains[i]->setDevice_l(mOutDevice); 4973 } 4974 } 4975 } 4976 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 4977 // do not accept frame count changes if tracks are open as the track buffer 4978 // size depends on frame count and correct behavior would not be garantied 4979 // if frame count is changed after track creation 4980 if (!mTracks.isEmpty()) { 4981 status = INVALID_OPERATION; 4982 } else { 4983 reconfig = true; 4984 } 4985 } 4986 if (status == NO_ERROR) { 4987 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 4988 keyValuePair.string()); 4989 if (!mStandby && status == INVALID_OPERATION) { 4990 mOutput->standby(); 4991 mStandby = true; 4992 mBytesWritten = 0; 4993 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 4994 keyValuePair.string()); 4995 } 4996 if (status == NO_ERROR && reconfig) { 4997 readOutputParameters_l(); 4998 sendIoConfigEvent_l(AUDIO_OUTPUT_CONFIG_CHANGED); 4999 } 5000 } 5001 5002 return reconfig || a2dpDeviceChanged; 5003} 5004 5005uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() const 5006{ 5007 uint32_t time; 5008 if (audio_has_proportional_frames(mFormat)) { 5009 time = PlaybackThread::activeSleepTimeUs(); 5010 } else { 5011 time = kDirectMinSleepTimeUs; 5012 } 5013 return time; 5014} 5015 5016uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() const 5017{ 5018 uint32_t time; 5019 if (audio_has_proportional_frames(mFormat)) { 5020 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 5021 } else { 5022 time = kDirectMinSleepTimeUs; 5023 } 5024 return time; 5025} 5026 5027uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() const 5028{ 5029 uint32_t time; 5030 if (audio_has_proportional_frames(mFormat)) { 5031 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 5032 } else { 5033 time = kDirectMinSleepTimeUs; 5034 } 5035 return time; 5036} 5037 5038void AudioFlinger::DirectOutputThread::cacheParameters_l() 5039{ 5040 PlaybackThread::cacheParameters_l(); 5041 5042 // use shorter standby delay as on normal output to release 5043 // hardware resources as soon as possible 5044 // no delay on outputs with HW A/V sync 5045 if (usesHwAvSync()) { 5046 mStandbyDelayNs = 0; 5047 } else if ((mType == OFFLOAD) && !audio_has_proportional_frames(mFormat)) { 5048 mStandbyDelayNs = kOffloadStandbyDelayNs; 5049 } else { 5050 mStandbyDelayNs = microseconds(mActiveSleepTimeUs*2); 5051 } 5052} 5053 5054void AudioFlinger::DirectOutputThread::flushHw_l() 5055{ 5056 mOutput->flush(); 5057 mHwPaused = false; 5058 mFlushPending = false; 5059} 5060 5061// ---------------------------------------------------------------------------- 5062 5063AudioFlinger::AsyncCallbackThread::AsyncCallbackThread( 5064 const wp<AudioFlinger::PlaybackThread>& playbackThread) 5065 : Thread(false /*canCallJava*/), 5066 mPlaybackThread(playbackThread), 5067 mWriteAckSequence(0), 5068 mDrainSequence(0) 5069{ 5070} 5071 5072AudioFlinger::AsyncCallbackThread::~AsyncCallbackThread() 5073{ 5074} 5075 5076void AudioFlinger::AsyncCallbackThread::onFirstRef() 5077{ 5078 run("Offload Cbk", ANDROID_PRIORITY_URGENT_AUDIO); 5079} 5080 5081bool AudioFlinger::AsyncCallbackThread::threadLoop() 5082{ 5083 while (!exitPending()) { 5084 uint32_t writeAckSequence; 5085 uint32_t drainSequence; 5086 5087 { 5088 Mutex::Autolock _l(mLock); 5089 while (!((mWriteAckSequence & 1) || 5090 (mDrainSequence & 1) || 5091 exitPending())) { 5092 mWaitWorkCV.wait(mLock); 5093 } 5094 5095 if (exitPending()) { 5096 break; 5097 } 5098 ALOGV("AsyncCallbackThread mWriteAckSequence %d mDrainSequence %d", 5099 mWriteAckSequence, mDrainSequence); 5100 writeAckSequence = mWriteAckSequence; 5101 mWriteAckSequence &= ~1; 5102 drainSequence = mDrainSequence; 5103 mDrainSequence &= ~1; 5104 } 5105 { 5106 sp<AudioFlinger::PlaybackThread> playbackThread = mPlaybackThread.promote(); 5107 if (playbackThread != 0) { 5108 if (writeAckSequence & 1) { 5109 playbackThread->resetWriteBlocked(writeAckSequence >> 1); 5110 } 5111 if (drainSequence & 1) { 5112 playbackThread->resetDraining(drainSequence >> 1); 5113 } 5114 } 5115 } 5116 } 5117 return false; 5118} 5119 5120void AudioFlinger::AsyncCallbackThread::exit() 5121{ 5122 ALOGV("AsyncCallbackThread::exit"); 5123 Mutex::Autolock _l(mLock); 5124 requestExit(); 5125 mWaitWorkCV.broadcast(); 5126} 5127 5128void AudioFlinger::AsyncCallbackThread::setWriteBlocked(uint32_t sequence) 5129{ 5130 Mutex::Autolock _l(mLock); 5131 // bit 0 is cleared 5132 mWriteAckSequence = sequence << 1; 5133} 5134 5135void AudioFlinger::AsyncCallbackThread::resetWriteBlocked() 5136{ 5137 Mutex::Autolock _l(mLock); 5138 // ignore unexpected callbacks 5139 if (mWriteAckSequence & 2) { 5140 mWriteAckSequence |= 1; 5141 mWaitWorkCV.signal(); 5142 } 5143} 5144 5145void AudioFlinger::AsyncCallbackThread::setDraining(uint32_t sequence) 5146{ 5147 Mutex::Autolock _l(mLock); 5148 // bit 0 is cleared 5149 mDrainSequence = sequence << 1; 5150} 5151 5152void AudioFlinger::AsyncCallbackThread::resetDraining() 5153{ 5154 Mutex::Autolock _l(mLock); 5155 // ignore unexpected callbacks 5156 if (mDrainSequence & 2) { 5157 mDrainSequence |= 1; 5158 mWaitWorkCV.signal(); 5159 } 5160} 5161 5162 5163// ---------------------------------------------------------------------------- 5164AudioFlinger::OffloadThread::OffloadThread(const sp<AudioFlinger>& audioFlinger, 5165 AudioStreamOut* output, audio_io_handle_t id, uint32_t device, bool systemReady, 5166 uint32_t bitRate) 5167 : DirectOutputThread(audioFlinger, output, id, device, OFFLOAD, systemReady, bitRate), 5168 mPausedWriteLength(0), mPausedBytesRemaining(0), mKeepWakeLock(true) 5169{ 5170 //FIXME: mStandby should be set to true by ThreadBase constructor 5171 mStandby = true; 5172 mKeepWakeLock = property_get_bool("ro.audio.offload_wakelock", true /* default_value */); 5173} 5174 5175void AudioFlinger::OffloadThread::threadLoop_exit() 5176{ 5177 if (mFlushPending || mHwPaused) { 5178 // If a flush is pending or track was paused, just discard buffered data 5179 flushHw_l(); 5180 } else { 5181 mMixerStatus = MIXER_DRAIN_ALL; 5182 threadLoop_drain(); 5183 } 5184 if (mUseAsyncWrite) { 5185 ALOG_ASSERT(mCallbackThread != 0); 5186 mCallbackThread->exit(); 5187 } 5188 PlaybackThread::threadLoop_exit(); 5189} 5190 5191AudioFlinger::PlaybackThread::mixer_state AudioFlinger::OffloadThread::prepareTracks_l( 5192 Vector< sp<Track> > *tracksToRemove 5193) 5194{ 5195 size_t count = mActiveTracks.size(); 5196 5197 mixer_state mixerStatus = MIXER_IDLE; 5198 bool doHwPause = false; 5199 bool doHwResume = false; 5200 5201 ALOGV("OffloadThread::prepareTracks_l active tracks %zu", count); 5202 5203 // find out which tracks need to be processed 5204 for (size_t i = 0; i < count; i++) { 5205 sp<Track> t = mActiveTracks[i].promote(); 5206 // The track died recently 5207 if (t == 0) { 5208 continue; 5209 } 5210 Track* const track = t.get(); 5211#ifdef VERY_VERY_VERBOSE_LOGGING 5212 audio_track_cblk_t* cblk = track->cblk(); 5213#endif 5214 // Only consider last track started for volume and mixer state control. 5215 // In theory an older track could underrun and restart after the new one starts 5216 // but as we only care about the transition phase between two tracks on a 5217 // direct output, it is not a problem to ignore the underrun case. 5218 sp<Track> l = mLatestActiveTrack.promote(); 5219 bool last = l.get() == track; 5220 5221 if (track->isInvalid()) { 5222 ALOGW("An invalidated track shouldn't be in active list"); 5223 tracksToRemove->add(track); 5224 continue; 5225 } 5226 5227 if (track->mState == TrackBase::IDLE) { 5228 ALOGW("An idle track shouldn't be in active list"); 5229 continue; 5230 } 5231 5232 if (track->isPausing()) { 5233 track->setPaused(); 5234 if (last) { 5235 if (mHwSupportsPause && !mHwPaused) { 5236 doHwPause = true; 5237 mHwPaused = true; 5238 } 5239 // If we were part way through writing the mixbuffer to 5240 // the HAL we must save this until we resume 5241 // BUG - this will be wrong if a different track is made active, 5242 // in that case we want to discard the pending data in the 5243 // mixbuffer and tell the client to present it again when the 5244 // track is resumed 5245 mPausedWriteLength = mCurrentWriteLength; 5246 mPausedBytesRemaining = mBytesRemaining; 5247 mBytesRemaining = 0; // stop writing 5248 } 5249 tracksToRemove->add(track); 5250 } else if (track->isFlushPending()) { 5251 track->mRetryCount = kMaxTrackRetriesOffload; 5252 track->flushAck(); 5253 if (last) { 5254 mFlushPending = true; 5255 } 5256 } else if (track->isResumePending()){ 5257 track->resumeAck(); 5258 if (last) { 5259 if (mPausedBytesRemaining) { 5260 // Need to continue write that was interrupted 5261 mCurrentWriteLength = mPausedWriteLength; 5262 mBytesRemaining = mPausedBytesRemaining; 5263 mPausedBytesRemaining = 0; 5264 } 5265 if (mHwPaused) { 5266 doHwResume = true; 5267 mHwPaused = false; 5268 // threadLoop_mix() will handle the case that we need to 5269 // resume an interrupted write 5270 } 5271 // enable write to audio HAL 5272 mSleepTimeUs = 0; 5273 5274 // Do not handle new data in this iteration even if track->framesReady() 5275 mixerStatus = MIXER_TRACKS_ENABLED; 5276 } 5277 } else if (track->framesReady() && track->isReady() && 5278 !track->isPaused() && !track->isTerminated() && !track->isStopping_2()) { 5279 ALOGVV("OffloadThread: track %d s=%08x [OK]", track->name(), cblk->mServer); 5280 if (track->mFillingUpStatus == Track::FS_FILLED) { 5281 track->mFillingUpStatus = Track::FS_ACTIVE; 5282 // make sure processVolume_l() will apply new volume even if 0 5283 mLeftVolFloat = mRightVolFloat = -1.0; 5284 } 5285 5286 if (last) { 5287 sp<Track> previousTrack = mPreviousTrack.promote(); 5288 if (previousTrack != 0) { 5289 if (track != previousTrack.get()) { 5290 // Flush any data still being written from last track 5291 mBytesRemaining = 0; 5292 if (mPausedBytesRemaining) { 5293 // Last track was paused so we also need to flush saved 5294 // mixbuffer state and invalidate track so that it will 5295 // re-submit that unwritten data when it is next resumed 5296 mPausedBytesRemaining = 0; 5297 // Invalidate is a bit drastic - would be more efficient 5298 // to have a flag to tell client that some of the 5299 // previously written data was lost 5300 previousTrack->invalidate(); 5301 } 5302 // flush data already sent to the DSP if changing audio session as audio 5303 // comes from a different source. Also invalidate previous track to force a 5304 // seek when resuming. 5305 if (previousTrack->sessionId() != track->sessionId()) { 5306 previousTrack->invalidate(); 5307 } 5308 } 5309 } 5310 mPreviousTrack = track; 5311 // reset retry count 5312 track->mRetryCount = kMaxTrackRetriesOffload; 5313 mActiveTrack = t; 5314 mixerStatus = MIXER_TRACKS_READY; 5315 } 5316 } else { 5317 ALOGVV("OffloadThread: track %d s=%08x [NOT READY]", track->name(), cblk->mServer); 5318 if (track->isStopping_1()) { 5319 // Hardware buffer can hold a large amount of audio so we must 5320 // wait for all current track's data to drain before we say 5321 // that the track is stopped. 5322 if (mBytesRemaining == 0) { 5323 // Only start draining when all data in mixbuffer 5324 // has been written 5325 ALOGV("OffloadThread: underrun and STOPPING_1 -> draining, STOPPING_2"); 5326 track->mState = TrackBase::STOPPING_2; // so presentation completes after drain 5327 // do not drain if no data was ever sent to HAL (mStandby == true) 5328 if (last && !mStandby) { 5329 // do not modify drain sequence if we are already draining. This happens 5330 // when resuming from pause after drain. 5331 if ((mDrainSequence & 1) == 0) { 5332 mSleepTimeUs = 0; 5333 mStandbyTimeNs = systemTime() + mStandbyDelayNs; 5334 mixerStatus = MIXER_DRAIN_TRACK; 5335 mDrainSequence += 2; 5336 } 5337 if (mHwPaused) { 5338 // It is possible to move from PAUSED to STOPPING_1 without 5339 // a resume so we must ensure hardware is running 5340 doHwResume = true; 5341 mHwPaused = false; 5342 } 5343 } 5344 } 5345 } else if (track->isStopping_2()) { 5346 // Drain has completed or we are in standby, signal presentation complete 5347 if (!(mDrainSequence & 1) || !last || mStandby) { 5348 track->mState = TrackBase::STOPPED; 5349 size_t audioHALFrames = 5350 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 5351 int64_t framesWritten = 5352 mBytesWritten / mOutput->getFrameSize(); 5353 track->presentationComplete(framesWritten, audioHALFrames); 5354 track->reset(); 5355 tracksToRemove->add(track); 5356 } 5357 } else { 5358 // No buffers for this track. Give it a few chances to 5359 // fill a buffer, then remove it from active list. 5360 if (--(track->mRetryCount) <= 0) { 5361 ALOGV("OffloadThread: BUFFER TIMEOUT: remove(%d) from active list", 5362 track->name()); 5363 tracksToRemove->add(track); 5364 // indicate to client process that the track was disabled because of underrun; 5365 // it will then automatically call start() when data is available 5366 track->disable(); 5367 } else if (last){ 5368 mixerStatus = MIXER_TRACKS_ENABLED; 5369 } 5370 } 5371 } 5372 // compute volume for this track 5373 processVolume_l(track, last); 5374 } 5375 5376 // make sure the pause/flush/resume sequence is executed in the right order. 5377 // If a flush is pending and a track is active but the HW is not paused, force a HW pause 5378 // before flush and then resume HW. This can happen in case of pause/flush/resume 5379 // if resume is received before pause is executed. 5380 if (!mStandby && (doHwPause || (mFlushPending && !mHwPaused && (count != 0)))) { 5381 mOutput->stream->pause(mOutput->stream); 5382 } 5383 if (mFlushPending) { 5384 flushHw_l(); 5385 } 5386 if (!mStandby && doHwResume) { 5387 mOutput->stream->resume(mOutput->stream); 5388 } 5389 5390 // remove all the tracks that need to be... 5391 removeTracks_l(*tracksToRemove); 5392 5393 return mixerStatus; 5394} 5395 5396// must be called with thread mutex locked 5397bool AudioFlinger::OffloadThread::waitingAsyncCallback_l() 5398{ 5399 ALOGVV("waitingAsyncCallback_l mWriteAckSequence %d mDrainSequence %d", 5400 mWriteAckSequence, mDrainSequence); 5401 if (mUseAsyncWrite && ((mWriteAckSequence & 1) || (mDrainSequence & 1))) { 5402 return true; 5403 } 5404 return false; 5405} 5406 5407bool AudioFlinger::OffloadThread::waitingAsyncCallback() 5408{ 5409 Mutex::Autolock _l(mLock); 5410 return waitingAsyncCallback_l(); 5411} 5412 5413void AudioFlinger::OffloadThread::flushHw_l() 5414{ 5415 DirectOutputThread::flushHw_l(); 5416 // Flush anything still waiting in the mixbuffer 5417 mCurrentWriteLength = 0; 5418 mBytesRemaining = 0; 5419 mPausedWriteLength = 0; 5420 mPausedBytesRemaining = 0; 5421 // reset bytes written count to reflect that DSP buffers are empty after flush. 5422 mBytesWritten = 0; 5423 5424 if (mUseAsyncWrite) { 5425 // discard any pending drain or write ack by incrementing sequence 5426 mWriteAckSequence = (mWriteAckSequence + 2) & ~1; 5427 mDrainSequence = (mDrainSequence + 2) & ~1; 5428 ALOG_ASSERT(mCallbackThread != 0); 5429 mCallbackThread->setWriteBlocked(mWriteAckSequence); 5430 mCallbackThread->setDraining(mDrainSequence); 5431 } 5432} 5433 5434uint32_t AudioFlinger::OffloadThread::activeSleepTimeUs() const 5435{ 5436 uint32_t time; 5437 if (audio_has_proportional_frames(mFormat)) { 5438 time = PlaybackThread::activeSleepTimeUs(); 5439 } else { 5440 // sleep time is half the duration of an audio HAL buffer. 5441 // Note: This can be problematic in case of underrun with variable bit rate and 5442 // current rate is much less than initial rate. 5443 time = (uint32_t)max(kDirectMinSleepTimeUs, mBufferDurationUs / 2); 5444 } 5445 return time; 5446} 5447 5448// ---------------------------------------------------------------------------- 5449 5450AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 5451 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id, bool systemReady) 5452 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->outDevice(), 5453 systemReady, DUPLICATING), 5454 mWaitTimeMs(UINT_MAX) 5455{ 5456 addOutputTrack(mainThread); 5457} 5458 5459AudioFlinger::DuplicatingThread::~DuplicatingThread() 5460{ 5461 for (size_t i = 0; i < mOutputTracks.size(); i++) { 5462 mOutputTracks[i]->destroy(); 5463 } 5464} 5465 5466void AudioFlinger::DuplicatingThread::threadLoop_mix() 5467{ 5468 // mix buffers... 5469 if (outputsReady(outputTracks)) { 5470 mAudioMixer->process(); 5471 } else { 5472 if (mMixerBufferValid) { 5473 memset(mMixerBuffer, 0, mMixerBufferSize); 5474 } else { 5475 memset(mSinkBuffer, 0, mSinkBufferSize); 5476 } 5477 } 5478 mSleepTimeUs = 0; 5479 writeFrames = mNormalFrameCount; 5480 mCurrentWriteLength = mSinkBufferSize; 5481 mStandbyTimeNs = systemTime() + mStandbyDelayNs; 5482} 5483 5484void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() 5485{ 5486 if (mSleepTimeUs == 0) { 5487 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 5488 mSleepTimeUs = mActiveSleepTimeUs; 5489 } else { 5490 mSleepTimeUs = mIdleSleepTimeUs; 5491 } 5492 } else if (mBytesWritten != 0) { 5493 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 5494 writeFrames = mNormalFrameCount; 5495 memset(mSinkBuffer, 0, mSinkBufferSize); 5496 } else { 5497 // flush remaining overflow buffers in output tracks 5498 writeFrames = 0; 5499 } 5500 mSleepTimeUs = 0; 5501 } 5502} 5503 5504ssize_t AudioFlinger::DuplicatingThread::threadLoop_write() 5505{ 5506 for (size_t i = 0; i < outputTracks.size(); i++) { 5507 outputTracks[i]->write(mSinkBuffer, writeFrames); 5508 } 5509 mStandby = false; 5510 return (ssize_t)mSinkBufferSize; 5511} 5512 5513void AudioFlinger::DuplicatingThread::threadLoop_standby() 5514{ 5515 // DuplicatingThread implements standby by stopping all tracks 5516 for (size_t i = 0; i < outputTracks.size(); i++) { 5517 outputTracks[i]->stop(); 5518 } 5519} 5520 5521void AudioFlinger::DuplicatingThread::saveOutputTracks() 5522{ 5523 outputTracks = mOutputTracks; 5524} 5525 5526void AudioFlinger::DuplicatingThread::clearOutputTracks() 5527{ 5528 outputTracks.clear(); 5529} 5530 5531void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 5532{ 5533 Mutex::Autolock _l(mLock); 5534 // The downstream MixerThread consumes thread->frameCount() amount of frames per mix pass. 5535 // Adjust for thread->sampleRate() to determine minimum buffer frame count. 5536 // Then triple buffer because Threads do not run synchronously and may not be clock locked. 5537 const size_t frameCount = 5538 3 * sourceFramesNeeded(mSampleRate, thread->frameCount(), thread->sampleRate()); 5539 // TODO: Consider asynchronous sample rate conversion to handle clock disparity 5540 // from different OutputTracks and their associated MixerThreads (e.g. one may 5541 // nearly empty and the other may be dropping data). 5542 5543 sp<OutputTrack> outputTrack = new OutputTrack(thread, 5544 this, 5545 mSampleRate, 5546 mFormat, 5547 mChannelMask, 5548 frameCount, 5549 IPCThreadState::self()->getCallingUid()); 5550 if (outputTrack->cblk() != NULL) { 5551 thread->setStreamVolume(AUDIO_STREAM_PATCH, 1.0f); 5552 mOutputTracks.add(outputTrack); 5553 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack.get(), thread); 5554 updateWaitTime_l(); 5555 } 5556} 5557 5558void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 5559{ 5560 Mutex::Autolock _l(mLock); 5561 for (size_t i = 0; i < mOutputTracks.size(); i++) { 5562 if (mOutputTracks[i]->thread() == thread) { 5563 mOutputTracks[i]->destroy(); 5564 mOutputTracks.removeAt(i); 5565 updateWaitTime_l(); 5566 if (thread->getOutput() == mOutput) { 5567 mOutput = NULL; 5568 } 5569 return; 5570 } 5571 } 5572 ALOGV("removeOutputTrack(): unknown thread: %p", thread); 5573} 5574 5575// caller must hold mLock 5576void AudioFlinger::DuplicatingThread::updateWaitTime_l() 5577{ 5578 mWaitTimeMs = UINT_MAX; 5579 for (size_t i = 0; i < mOutputTracks.size(); i++) { 5580 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 5581 if (strong != 0) { 5582 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 5583 if (waitTimeMs < mWaitTimeMs) { 5584 mWaitTimeMs = waitTimeMs; 5585 } 5586 } 5587 } 5588} 5589 5590 5591bool AudioFlinger::DuplicatingThread::outputsReady( 5592 const SortedVector< sp<OutputTrack> > &outputTracks) 5593{ 5594 for (size_t i = 0; i < outputTracks.size(); i++) { 5595 sp<ThreadBase> thread = outputTracks[i]->thread().promote(); 5596 if (thread == 0) { 5597 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", 5598 outputTracks[i].get()); 5599 return false; 5600 } 5601 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 5602 // see note at standby() declaration 5603 if (playbackThread->standby() && !playbackThread->isSuspended()) { 5604 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), 5605 thread.get()); 5606 return false; 5607 } 5608 } 5609 return true; 5610} 5611 5612uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() const 5613{ 5614 return (mWaitTimeMs * 1000) / 2; 5615} 5616 5617void AudioFlinger::DuplicatingThread::cacheParameters_l() 5618{ 5619 // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first 5620 updateWaitTime_l(); 5621 5622 MixerThread::cacheParameters_l(); 5623} 5624 5625// ---------------------------------------------------------------------------- 5626// Record 5627// ---------------------------------------------------------------------------- 5628 5629AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 5630 AudioStreamIn *input, 5631 audio_io_handle_t id, 5632 audio_devices_t outDevice, 5633 audio_devices_t inDevice, 5634 bool systemReady 5635#ifdef TEE_SINK 5636 , const sp<NBAIO_Sink>& teeSink 5637#endif 5638 ) : 5639 ThreadBase(audioFlinger, id, outDevice, inDevice, RECORD, systemReady), 5640 mInput(input), mActiveTracksGen(0), mRsmpInBuffer(NULL), 5641 // mRsmpInFrames and mRsmpInFramesP2 are set by readInputParameters_l() 5642 mRsmpInRear(0) 5643#ifdef TEE_SINK 5644 , mTeeSink(teeSink) 5645#endif 5646 , mReadOnlyHeap(new MemoryDealer(kRecordThreadReadOnlyHeapSize, 5647 "RecordThreadRO", MemoryHeapBase::READ_ONLY)) 5648 // mFastCapture below 5649 , mFastCaptureFutex(0) 5650 // mInputSource 5651 // mPipeSink 5652 // mPipeSource 5653 , mPipeFramesP2(0) 5654 // mPipeMemory 5655 // mFastCaptureNBLogWriter 5656 , mFastTrackAvail(false) 5657{ 5658 snprintf(mThreadName, kThreadNameLength, "AudioIn_%X", id); 5659 mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mThreadName); 5660 5661 readInputParameters_l(); 5662 5663 // create an NBAIO source for the HAL input stream, and negotiate 5664 mInputSource = new AudioStreamInSource(input->stream); 5665 size_t numCounterOffers = 0; 5666 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount, mFormat)}; 5667#if !LOG_NDEBUG 5668 ssize_t index = 5669#else 5670 (void) 5671#endif 5672 mInputSource->negotiate(offers, 1, NULL, numCounterOffers); 5673 ALOG_ASSERT(index == 0); 5674 5675 // initialize fast capture depending on configuration 5676 bool initFastCapture; 5677 switch (kUseFastCapture) { 5678 case FastCapture_Never: 5679 initFastCapture = false; 5680 break; 5681 case FastCapture_Always: 5682 initFastCapture = true; 5683 break; 5684 case FastCapture_Static: 5685 initFastCapture = (mFrameCount * 1000) / mSampleRate < kMinNormalCaptureBufferSizeMs; 5686 break; 5687 // case FastCapture_Dynamic: 5688 } 5689 5690 if (initFastCapture) { 5691 // create a Pipe for FastCapture to write to, and for us and fast tracks to read from 5692 NBAIO_Format format = mInputSource->format(); 5693 size_t pipeFramesP2 = roundup(mSampleRate / 25); // double-buffering of 20 ms each 5694 size_t pipeSize = pipeFramesP2 * Format_frameSize(format); 5695 void *pipeBuffer; 5696 const sp<MemoryDealer> roHeap(readOnlyHeap()); 5697 sp<IMemory> pipeMemory; 5698 if ((roHeap == 0) || 5699 (pipeMemory = roHeap->allocate(pipeSize)) == 0 || 5700 (pipeBuffer = pipeMemory->pointer()) == NULL) { 5701 ALOGE("not enough memory for pipe buffer size=%zu", pipeSize); 5702 goto failed; 5703 } 5704 // pipe will be shared directly with fast clients, so clear to avoid leaking old information 5705 memset(pipeBuffer, 0, pipeSize); 5706 Pipe *pipe = new Pipe(pipeFramesP2, format, pipeBuffer); 5707 const NBAIO_Format offers[1] = {format}; 5708 size_t numCounterOffers = 0; 5709 ssize_t index = pipe->negotiate(offers, 1, NULL, numCounterOffers); 5710 ALOG_ASSERT(index == 0); 5711 mPipeSink = pipe; 5712 PipeReader *pipeReader = new PipeReader(*pipe); 5713 numCounterOffers = 0; 5714 index = pipeReader->negotiate(offers, 1, NULL, numCounterOffers); 5715 ALOG_ASSERT(index == 0); 5716 mPipeSource = pipeReader; 5717 mPipeFramesP2 = pipeFramesP2; 5718 mPipeMemory = pipeMemory; 5719 5720 // create fast capture 5721 mFastCapture = new FastCapture(); 5722 FastCaptureStateQueue *sq = mFastCapture->sq(); 5723#ifdef STATE_QUEUE_DUMP 5724 // FIXME 5725#endif 5726 FastCaptureState *state = sq->begin(); 5727 state->mCblk = NULL; 5728 state->mInputSource = mInputSource.get(); 5729 state->mInputSourceGen++; 5730 state->mPipeSink = pipe; 5731 state->mPipeSinkGen++; 5732 state->mFrameCount = mFrameCount; 5733 state->mCommand = FastCaptureState::COLD_IDLE; 5734 // already done in constructor initialization list 5735 //mFastCaptureFutex = 0; 5736 state->mColdFutexAddr = &mFastCaptureFutex; 5737 state->mColdGen++; 5738 state->mDumpState = &mFastCaptureDumpState; 5739#ifdef TEE_SINK 5740 // FIXME 5741#endif 5742 mFastCaptureNBLogWriter = audioFlinger->newWriter_l(kFastCaptureLogSize, "FastCapture"); 5743 state->mNBLogWriter = mFastCaptureNBLogWriter.get(); 5744 sq->end(); 5745 sq->push(FastCaptureStateQueue::BLOCK_UNTIL_PUSHED); 5746 5747 // start the fast capture 5748 mFastCapture->run("FastCapture", ANDROID_PRIORITY_URGENT_AUDIO); 5749 pid_t tid = mFastCapture->getTid(); 5750 sendPrioConfigEvent(getpid_cached, tid, kPriorityFastCapture); 5751#ifdef AUDIO_WATCHDOG 5752 // FIXME 5753#endif 5754 5755 mFastTrackAvail = true; 5756 } 5757failed: ; 5758 5759 // FIXME mNormalSource 5760} 5761 5762AudioFlinger::RecordThread::~RecordThread() 5763{ 5764 if (mFastCapture != 0) { 5765 FastCaptureStateQueue *sq = mFastCapture->sq(); 5766 FastCaptureState *state = sq->begin(); 5767 if (state->mCommand == FastCaptureState::COLD_IDLE) { 5768 int32_t old = android_atomic_inc(&mFastCaptureFutex); 5769 if (old == -1) { 5770 (void) syscall(__NR_futex, &mFastCaptureFutex, FUTEX_WAKE_PRIVATE, 1); 5771 } 5772 } 5773 state->mCommand = FastCaptureState::EXIT; 5774 sq->end(); 5775 sq->push(FastCaptureStateQueue::BLOCK_UNTIL_PUSHED); 5776 mFastCapture->join(); 5777 mFastCapture.clear(); 5778 } 5779 mAudioFlinger->unregisterWriter(mFastCaptureNBLogWriter); 5780 mAudioFlinger->unregisterWriter(mNBLogWriter); 5781 free(mRsmpInBuffer); 5782} 5783 5784void AudioFlinger::RecordThread::onFirstRef() 5785{ 5786 run(mThreadName, PRIORITY_URGENT_AUDIO); 5787} 5788 5789bool AudioFlinger::RecordThread::threadLoop() 5790{ 5791 nsecs_t lastWarning = 0; 5792 5793 inputStandBy(); 5794 5795reacquire_wakelock: 5796 sp<RecordTrack> activeTrack; 5797 int activeTracksGen; 5798 { 5799 Mutex::Autolock _l(mLock); 5800 size_t size = mActiveTracks.size(); 5801 activeTracksGen = mActiveTracksGen; 5802 if (size > 0) { 5803 // FIXME an arbitrary choice 5804 activeTrack = mActiveTracks[0]; 5805 acquireWakeLock_l(activeTrack->uid()); 5806 if (size > 1) { 5807 SortedVector<int> tmp; 5808 for (size_t i = 0; i < size; i++) { 5809 tmp.add(mActiveTracks[i]->uid()); 5810 } 5811 updateWakeLockUids_l(tmp); 5812 } 5813 } else { 5814 acquireWakeLock_l(-1); 5815 } 5816 } 5817 5818 // used to request a deferred sleep, to be executed later while mutex is unlocked 5819 uint32_t sleepUs = 0; 5820 5821 // loop while there is work to do 5822 for (;;) { 5823 Vector< sp<EffectChain> > effectChains; 5824 5825 // sleep with mutex unlocked 5826 if (sleepUs > 0) { 5827 ATRACE_BEGIN("sleep"); 5828 usleep(sleepUs); 5829 ATRACE_END(); 5830 sleepUs = 0; 5831 } 5832 5833 // activeTracks accumulates a copy of a subset of mActiveTracks 5834 Vector< sp<RecordTrack> > activeTracks; 5835 5836 // reference to the (first and only) active fast track 5837 sp<RecordTrack> fastTrack; 5838 5839 // reference to a fast track which is about to be removed 5840 sp<RecordTrack> fastTrackToRemove; 5841 5842 { // scope for mLock 5843 Mutex::Autolock _l(mLock); 5844 5845 processConfigEvents_l(); 5846 5847 // check exitPending here because checkForNewParameters_l() and 5848 // checkForNewParameters_l() can temporarily release mLock 5849 if (exitPending()) { 5850 break; 5851 } 5852 5853 // if no active track(s), then standby and release wakelock 5854 size_t size = mActiveTracks.size(); 5855 if (size == 0) { 5856 standbyIfNotAlreadyInStandby(); 5857 // exitPending() can't become true here 5858 releaseWakeLock_l(); 5859 ALOGV("RecordThread: loop stopping"); 5860 // go to sleep 5861 mWaitWorkCV.wait(mLock); 5862 ALOGV("RecordThread: loop starting"); 5863 goto reacquire_wakelock; 5864 } 5865 5866 if (mActiveTracksGen != activeTracksGen) { 5867 activeTracksGen = mActiveTracksGen; 5868 SortedVector<int> tmp; 5869 for (size_t i = 0; i < size; i++) { 5870 tmp.add(mActiveTracks[i]->uid()); 5871 } 5872 updateWakeLockUids_l(tmp); 5873 } 5874 5875 bool doBroadcast = false; 5876 for (size_t i = 0; i < size; ) { 5877 5878 activeTrack = mActiveTracks[i]; 5879 if (activeTrack->isTerminated()) { 5880 if (activeTrack->isFastTrack()) { 5881 ALOG_ASSERT(fastTrackToRemove == 0); 5882 fastTrackToRemove = activeTrack; 5883 } 5884 removeTrack_l(activeTrack); 5885 mActiveTracks.remove(activeTrack); 5886 mActiveTracksGen++; 5887 size--; 5888 continue; 5889 } 5890 5891 TrackBase::track_state activeTrackState = activeTrack->mState; 5892 switch (activeTrackState) { 5893 5894 case TrackBase::PAUSING: 5895 mActiveTracks.remove(activeTrack); 5896 mActiveTracksGen++; 5897 doBroadcast = true; 5898 size--; 5899 continue; 5900 5901 case TrackBase::STARTING_1: 5902 sleepUs = 10000; 5903 i++; 5904 continue; 5905 5906 case TrackBase::STARTING_2: 5907 doBroadcast = true; 5908 mStandby = false; 5909 activeTrack->mState = TrackBase::ACTIVE; 5910 break; 5911 5912 case TrackBase::ACTIVE: 5913 break; 5914 5915 case TrackBase::IDLE: 5916 i++; 5917 continue; 5918 5919 default: 5920 LOG_ALWAYS_FATAL("Unexpected activeTrackState %d", activeTrackState); 5921 } 5922 5923 activeTracks.add(activeTrack); 5924 i++; 5925 5926 if (activeTrack->isFastTrack()) { 5927 ALOG_ASSERT(!mFastTrackAvail); 5928 ALOG_ASSERT(fastTrack == 0); 5929 fastTrack = activeTrack; 5930 } 5931 } 5932 if (doBroadcast) { 5933 mStartStopCond.broadcast(); 5934 } 5935 5936 // sleep if there are no active tracks to process 5937 if (activeTracks.size() == 0) { 5938 if (sleepUs == 0) { 5939 sleepUs = kRecordThreadSleepUs; 5940 } 5941 continue; 5942 } 5943 sleepUs = 0; 5944 5945 lockEffectChains_l(effectChains); 5946 } 5947 5948 // thread mutex is now unlocked, mActiveTracks unknown, activeTracks.size() > 0 5949 5950 size_t size = effectChains.size(); 5951 for (size_t i = 0; i < size; i++) { 5952 // thread mutex is not locked, but effect chain is locked 5953 effectChains[i]->process_l(); 5954 } 5955 5956 // Push a new fast capture state if fast capture is not already running, or cblk change 5957 if (mFastCapture != 0) { 5958 FastCaptureStateQueue *sq = mFastCapture->sq(); 5959 FastCaptureState *state = sq->begin(); 5960 bool didModify = false; 5961 FastCaptureStateQueue::block_t block = FastCaptureStateQueue::BLOCK_UNTIL_PUSHED; 5962 if (state->mCommand != FastCaptureState::READ_WRITE /* FIXME && 5963 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)*/) { 5964 if (state->mCommand == FastCaptureState::COLD_IDLE) { 5965 int32_t old = android_atomic_inc(&mFastCaptureFutex); 5966 if (old == -1) { 5967 (void) syscall(__NR_futex, &mFastCaptureFutex, FUTEX_WAKE_PRIVATE, 1); 5968 } 5969 } 5970 state->mCommand = FastCaptureState::READ_WRITE; 5971#if 0 // FIXME 5972 mFastCaptureDumpState.increaseSamplingN(mAudioFlinger->isLowRamDevice() ? 5973 FastThreadDumpState::kSamplingNforLowRamDevice : 5974 FastThreadDumpState::kSamplingN); 5975#endif 5976 didModify = true; 5977 } 5978 audio_track_cblk_t *cblkOld = state->mCblk; 5979 audio_track_cblk_t *cblkNew = fastTrack != 0 ? fastTrack->cblk() : NULL; 5980 if (cblkNew != cblkOld) { 5981 state->mCblk = cblkNew; 5982 // block until acked if removing a fast track 5983 if (cblkOld != NULL) { 5984 block = FastCaptureStateQueue::BLOCK_UNTIL_ACKED; 5985 } 5986 didModify = true; 5987 } 5988 sq->end(didModify); 5989 if (didModify) { 5990 sq->push(block); 5991#if 0 5992 if (kUseFastCapture == FastCapture_Dynamic) { 5993 mNormalSource = mPipeSource; 5994 } 5995#endif 5996 } 5997 } 5998 5999 // now run the fast track destructor with thread mutex unlocked 6000 fastTrackToRemove.clear(); 6001 6002 // Read from HAL to keep up with fastest client if multiple active tracks, not slowest one. 6003 // Only the client(s) that are too slow will overrun. But if even the fastest client is too 6004 // slow, then this RecordThread will overrun by not calling HAL read often enough. 6005 // If destination is non-contiguous, first read past the nominal end of buffer, then 6006 // copy to the right place. Permitted because mRsmpInBuffer was over-allocated. 6007 6008 int32_t rear = mRsmpInRear & (mRsmpInFramesP2 - 1); 6009 ssize_t framesRead; 6010 6011 // If an NBAIO source is present, use it to read the normal capture's data 6012 if (mPipeSource != 0) { 6013 size_t framesToRead = mBufferSize / mFrameSize; 6014 framesRead = mPipeSource->read((uint8_t*)mRsmpInBuffer + rear * mFrameSize, 6015 framesToRead); 6016 if (framesRead == 0) { 6017 // since pipe is non-blocking, simulate blocking input 6018 sleepUs = (framesToRead * 1000000LL) / mSampleRate; 6019 } 6020 // otherwise use the HAL / AudioStreamIn directly 6021 } else { 6022 ATRACE_BEGIN("read"); 6023 ssize_t bytesRead = mInput->stream->read(mInput->stream, 6024 (uint8_t*)mRsmpInBuffer + rear * mFrameSize, mBufferSize); 6025 ATRACE_END(); 6026 if (bytesRead < 0) { 6027 framesRead = bytesRead; 6028 } else { 6029 framesRead = bytesRead / mFrameSize; 6030 } 6031 } 6032 6033 // Update server timestamp with server stats 6034 // systemTime() is optional if the hardware supports timestamps. 6035 mTimestamp.mPosition[ExtendedTimestamp::LOCATION_SERVER] += framesRead; 6036 mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_SERVER] = systemTime(); 6037 6038 // Update server timestamp with kernel stats 6039 if (mInput->stream->get_capture_position != nullptr) { 6040 int64_t position, time; 6041 int ret = mInput->stream->get_capture_position(mInput->stream, &position, &time); 6042 if (ret == NO_ERROR) { 6043 mTimestamp.mPosition[ExtendedTimestamp::LOCATION_KERNEL] = position; 6044 mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL] = time; 6045 // Note: In general record buffers should tend to be empty in 6046 // a properly running pipeline. 6047 // 6048 // Also, it is not advantageous to call get_presentation_position during the read 6049 // as the read obtains a lock, preventing the timestamp call from executing. 6050 } 6051 } 6052 // Use this to track timestamp information 6053 // ALOGD("%s", mTimestamp.toString().c_str()); 6054 6055 if (framesRead < 0 || (framesRead == 0 && mPipeSource == 0)) { 6056 ALOGE("read failed: framesRead=%zd", framesRead); 6057 // Force input into standby so that it tries to recover at next read attempt 6058 inputStandBy(); 6059 sleepUs = kRecordThreadSleepUs; 6060 } 6061 if (framesRead <= 0) { 6062 goto unlock; 6063 } 6064 ALOG_ASSERT(framesRead > 0); 6065 6066 if (mTeeSink != 0) { 6067 (void) mTeeSink->write((uint8_t*)mRsmpInBuffer + rear * mFrameSize, framesRead); 6068 } 6069 // If destination is non-contiguous, we now correct for reading past end of buffer. 6070 { 6071 size_t part1 = mRsmpInFramesP2 - rear; 6072 if ((size_t) framesRead > part1) { 6073 memcpy(mRsmpInBuffer, (uint8_t*)mRsmpInBuffer + mRsmpInFramesP2 * mFrameSize, 6074 (framesRead - part1) * mFrameSize); 6075 } 6076 } 6077 rear = mRsmpInRear += framesRead; 6078 6079 size = activeTracks.size(); 6080 // loop over each active track 6081 for (size_t i = 0; i < size; i++) { 6082 activeTrack = activeTracks[i]; 6083 6084 // skip fast tracks, as those are handled directly by FastCapture 6085 if (activeTrack->isFastTrack()) { 6086 continue; 6087 } 6088 6089 // TODO: This code probably should be moved to RecordTrack. 6090 // TODO: Update the activeTrack buffer converter in case of reconfigure. 6091 6092 enum { 6093 OVERRUN_UNKNOWN, 6094 OVERRUN_TRUE, 6095 OVERRUN_FALSE 6096 } overrun = OVERRUN_UNKNOWN; 6097 6098 // loop over getNextBuffer to handle circular sink 6099 for (;;) { 6100 6101 activeTrack->mSink.frameCount = ~0; 6102 status_t status = activeTrack->getNextBuffer(&activeTrack->mSink); 6103 size_t framesOut = activeTrack->mSink.frameCount; 6104 LOG_ALWAYS_FATAL_IF((status == OK) != (framesOut > 0)); 6105 6106 // check available frames and handle overrun conditions 6107 // if the record track isn't draining fast enough. 6108 bool hasOverrun; 6109 size_t framesIn; 6110 activeTrack->mResamplerBufferProvider->sync(&framesIn, &hasOverrun); 6111 if (hasOverrun) { 6112 overrun = OVERRUN_TRUE; 6113 } 6114 if (framesOut == 0 || framesIn == 0) { 6115 break; 6116 } 6117 6118 // Don't allow framesOut to be larger than what is possible with resampling 6119 // from framesIn. 6120 // This isn't strictly necessary but helps limit buffer resizing in 6121 // RecordBufferConverter. TODO: remove when no longer needed. 6122 framesOut = min(framesOut, 6123 destinationFramesPossible( 6124 framesIn, mSampleRate, activeTrack->mSampleRate)); 6125 // process frames from the RecordThread buffer provider to the RecordTrack buffer 6126 framesOut = activeTrack->mRecordBufferConverter->convert( 6127 activeTrack->mSink.raw, activeTrack->mResamplerBufferProvider, framesOut); 6128 6129 if (framesOut > 0 && (overrun == OVERRUN_UNKNOWN)) { 6130 overrun = OVERRUN_FALSE; 6131 } 6132 6133 if (activeTrack->mFramesToDrop == 0) { 6134 if (framesOut > 0) { 6135 activeTrack->mSink.frameCount = framesOut; 6136 activeTrack->releaseBuffer(&activeTrack->mSink); 6137 } 6138 } else { 6139 // FIXME could do a partial drop of framesOut 6140 if (activeTrack->mFramesToDrop > 0) { 6141 activeTrack->mFramesToDrop -= framesOut; 6142 if (activeTrack->mFramesToDrop <= 0) { 6143 activeTrack->clearSyncStartEvent(); 6144 } 6145 } else { 6146 activeTrack->mFramesToDrop += framesOut; 6147 if (activeTrack->mFramesToDrop >= 0 || activeTrack->mSyncStartEvent == 0 || 6148 activeTrack->mSyncStartEvent->isCancelled()) { 6149 ALOGW("Synced record %s, session %d, trigger session %d", 6150 (activeTrack->mFramesToDrop >= 0) ? "timed out" : "cancelled", 6151 activeTrack->sessionId(), 6152 (activeTrack->mSyncStartEvent != 0) ? 6153 activeTrack->mSyncStartEvent->triggerSession() : 6154 AUDIO_SESSION_NONE); 6155 activeTrack->clearSyncStartEvent(); 6156 } 6157 } 6158 } 6159 6160 if (framesOut == 0) { 6161 break; 6162 } 6163 } 6164 6165 switch (overrun) { 6166 case OVERRUN_TRUE: 6167 // client isn't retrieving buffers fast enough 6168 if (!activeTrack->setOverflow()) { 6169 nsecs_t now = systemTime(); 6170 // FIXME should lastWarning per track? 6171 if ((now - lastWarning) > kWarningThrottleNs) { 6172 ALOGW("RecordThread: buffer overflow"); 6173 lastWarning = now; 6174 } 6175 } 6176 break; 6177 case OVERRUN_FALSE: 6178 activeTrack->clearOverflow(); 6179 break; 6180 case OVERRUN_UNKNOWN: 6181 break; 6182 } 6183 6184 // update frame information and push timestamp out 6185 activeTrack->updateTrackFrameInfo( 6186 activeTrack->mServerProxy->framesReleased(), 6187 mTimestamp.mPosition[ExtendedTimestamp::LOCATION_SERVER], 6188 mSampleRate, mTimestamp); 6189 } 6190 6191unlock: 6192 // enable changes in effect chain 6193 unlockEffectChains(effectChains); 6194 // effectChains doesn't need to be cleared, since it is cleared by destructor at scope end 6195 } 6196 6197 standbyIfNotAlreadyInStandby(); 6198 6199 { 6200 Mutex::Autolock _l(mLock); 6201 for (size_t i = 0; i < mTracks.size(); i++) { 6202 sp<RecordTrack> track = mTracks[i]; 6203 track->invalidate(); 6204 } 6205 mActiveTracks.clear(); 6206 mActiveTracksGen++; 6207 mStartStopCond.broadcast(); 6208 } 6209 6210 releaseWakeLock(); 6211 6212 ALOGV("RecordThread %p exiting", this); 6213 return false; 6214} 6215 6216void AudioFlinger::RecordThread::standbyIfNotAlreadyInStandby() 6217{ 6218 if (!mStandby) { 6219 inputStandBy(); 6220 mStandby = true; 6221 } 6222} 6223 6224void AudioFlinger::RecordThread::inputStandBy() 6225{ 6226 // Idle the fast capture if it's currently running 6227 if (mFastCapture != 0) { 6228 FastCaptureStateQueue *sq = mFastCapture->sq(); 6229 FastCaptureState *state = sq->begin(); 6230 if (!(state->mCommand & FastCaptureState::IDLE)) { 6231 state->mCommand = FastCaptureState::COLD_IDLE; 6232 state->mColdFutexAddr = &mFastCaptureFutex; 6233 state->mColdGen++; 6234 mFastCaptureFutex = 0; 6235 sq->end(); 6236 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 6237 sq->push(FastCaptureStateQueue::BLOCK_UNTIL_ACKED); 6238#if 0 6239 if (kUseFastCapture == FastCapture_Dynamic) { 6240 // FIXME 6241 } 6242#endif 6243#ifdef AUDIO_WATCHDOG 6244 // FIXME 6245#endif 6246 } else { 6247 sq->end(false /*didModify*/); 6248 } 6249 } 6250 mInput->stream->common.standby(&mInput->stream->common); 6251} 6252 6253// RecordThread::createRecordTrack_l() must be called with AudioFlinger::mLock held 6254sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 6255 const sp<AudioFlinger::Client>& client, 6256 uint32_t sampleRate, 6257 audio_format_t format, 6258 audio_channel_mask_t channelMask, 6259 size_t *pFrameCount, 6260 audio_session_t sessionId, 6261 size_t *notificationFrames, 6262 int uid, 6263 IAudioFlinger::track_flags_t *flags, 6264 pid_t tid, 6265 status_t *status) 6266{ 6267 size_t frameCount = *pFrameCount; 6268 sp<RecordTrack> track; 6269 status_t lStatus; 6270 6271 // client expresses a preference for FAST, but we get the final say 6272 if (*flags & IAudioFlinger::TRACK_FAST) { 6273 if ( 6274 // we formerly checked for a callback handler (non-0 tid), 6275 // but that is no longer required for TRANSFER_OBTAIN mode 6276 // 6277 // frame count is not specified, or is exactly the pipe depth 6278 ((frameCount == 0) || (frameCount == mPipeFramesP2)) && 6279 // PCM data 6280 audio_is_linear_pcm(format) && 6281 // hardware format 6282 (format == mFormat) && 6283 // hardware channel mask 6284 (channelMask == mChannelMask) && 6285 // hardware sample rate 6286 (sampleRate == mSampleRate) && 6287 // record thread has an associated fast capture 6288 hasFastCapture() && 6289 // there are sufficient fast track slots available 6290 mFastTrackAvail 6291 ) { 6292 ALOGV("AUDIO_INPUT_FLAG_FAST accepted: frameCount=%zu mFrameCount=%zu", 6293 frameCount, mFrameCount); 6294 } else { 6295 ALOGV("AUDIO_INPUT_FLAG_FAST denied: frameCount=%zu mFrameCount=%zu mPipeFramesP2=%zu " 6296 "format=%#x isLinear=%d channelMask=%#x sampleRate=%u mSampleRate=%u " 6297 "hasFastCapture=%d tid=%d mFastTrackAvail=%d", 6298 frameCount, mFrameCount, mPipeFramesP2, 6299 format, audio_is_linear_pcm(format), channelMask, sampleRate, mSampleRate, 6300 hasFastCapture(), tid, mFastTrackAvail); 6301 *flags &= ~IAudioFlinger::TRACK_FAST; 6302 } 6303 } 6304 6305 // compute track buffer size in frames, and suggest the notification frame count 6306 if (*flags & IAudioFlinger::TRACK_FAST) { 6307 // fast track: frame count is exactly the pipe depth 6308 frameCount = mPipeFramesP2; 6309 // ignore requested notificationFrames, and always notify exactly once every HAL buffer 6310 *notificationFrames = mFrameCount; 6311 } else { 6312 // not fast track: max notification period is resampled equivalent of one HAL buffer time 6313 // or 20 ms if there is a fast capture 6314 // TODO This could be a roundupRatio inline, and const 6315 size_t maxNotificationFrames = ((int64_t) (hasFastCapture() ? mSampleRate/50 : mFrameCount) 6316 * sampleRate + mSampleRate - 1) / mSampleRate; 6317 // minimum number of notification periods is at least kMinNotifications, 6318 // and at least kMinMs rounded up to a whole notification period (minNotificationsByMs) 6319 static const size_t kMinNotifications = 3; 6320 static const uint32_t kMinMs = 30; 6321 // TODO This could be a roundupRatio inline 6322 const size_t minFramesByMs = (sampleRate * kMinMs + 1000 - 1) / 1000; 6323 // TODO This could be a roundupRatio inline 6324 const size_t minNotificationsByMs = (minFramesByMs + maxNotificationFrames - 1) / 6325 maxNotificationFrames; 6326 const size_t minFrameCount = maxNotificationFrames * 6327 max(kMinNotifications, minNotificationsByMs); 6328 frameCount = max(frameCount, minFrameCount); 6329 if (*notificationFrames == 0 || *notificationFrames > maxNotificationFrames) { 6330 *notificationFrames = maxNotificationFrames; 6331 } 6332 } 6333 *pFrameCount = frameCount; 6334 6335 lStatus = initCheck(); 6336 if (lStatus != NO_ERROR) { 6337 ALOGE("createRecordTrack_l() audio driver not initialized"); 6338 goto Exit; 6339 } 6340 6341 { // scope for mLock 6342 Mutex::Autolock _l(mLock); 6343 6344 track = new RecordTrack(this, client, sampleRate, 6345 format, channelMask, frameCount, NULL, sessionId, uid, 6346 *flags, TrackBase::TYPE_DEFAULT); 6347 6348 lStatus = track->initCheck(); 6349 if (lStatus != NO_ERROR) { 6350 ALOGE("createRecordTrack_l() initCheck failed %d; no control block?", lStatus); 6351 // track must be cleared from the caller as the caller has the AF lock 6352 goto Exit; 6353 } 6354 mTracks.add(track); 6355 6356 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 6357 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 6358 mAudioFlinger->btNrecIsOff(); 6359 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 6360 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 6361 6362 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 6363 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 6364 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 6365 // so ask activity manager to do this on our behalf 6366 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 6367 } 6368 } 6369 6370 lStatus = NO_ERROR; 6371 6372Exit: 6373 *status = lStatus; 6374 return track; 6375} 6376 6377status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, 6378 AudioSystem::sync_event_t event, 6379 audio_session_t triggerSession) 6380{ 6381 ALOGV("RecordThread::start event %d, triggerSession %d", event, triggerSession); 6382 sp<ThreadBase> strongMe = this; 6383 status_t status = NO_ERROR; 6384 6385 if (event == AudioSystem::SYNC_EVENT_NONE) { 6386 recordTrack->clearSyncStartEvent(); 6387 } else if (event != AudioSystem::SYNC_EVENT_SAME) { 6388 recordTrack->mSyncStartEvent = mAudioFlinger->createSyncEvent(event, 6389 triggerSession, 6390 recordTrack->sessionId(), 6391 syncStartEventCallback, 6392 recordTrack); 6393 // Sync event can be cancelled by the trigger session if the track is not in a 6394 // compatible state in which case we start record immediately 6395 if (recordTrack->mSyncStartEvent->isCancelled()) { 6396 recordTrack->clearSyncStartEvent(); 6397 } else { 6398 // do not wait for the event for more than AudioSystem::kSyncRecordStartTimeOutMs 6399 recordTrack->mFramesToDrop = - 6400 ((AudioSystem::kSyncRecordStartTimeOutMs * recordTrack->mSampleRate) / 1000); 6401 } 6402 } 6403 6404 { 6405 // This section is a rendezvous between binder thread executing start() and RecordThread 6406 AutoMutex lock(mLock); 6407 if (mActiveTracks.indexOf(recordTrack) >= 0) { 6408 if (recordTrack->mState == TrackBase::PAUSING) { 6409 ALOGV("active record track PAUSING -> ACTIVE"); 6410 recordTrack->mState = TrackBase::ACTIVE; 6411 } else { 6412 ALOGV("active record track state %d", recordTrack->mState); 6413 } 6414 return status; 6415 } 6416 6417 // TODO consider other ways of handling this, such as changing the state to :STARTING and 6418 // adding the track to mActiveTracks after returning from AudioSystem::startInput(), 6419 // or using a separate command thread 6420 recordTrack->mState = TrackBase::STARTING_1; 6421 mActiveTracks.add(recordTrack); 6422 mActiveTracksGen++; 6423 status_t status = NO_ERROR; 6424 if (recordTrack->isExternalTrack()) { 6425 mLock.unlock(); 6426 status = AudioSystem::startInput(mId, recordTrack->sessionId()); 6427 mLock.lock(); 6428 // FIXME should verify that recordTrack is still in mActiveTracks 6429 if (status != NO_ERROR) { 6430 mActiveTracks.remove(recordTrack); 6431 mActiveTracksGen++; 6432 recordTrack->clearSyncStartEvent(); 6433 ALOGV("RecordThread::start error %d", status); 6434 return status; 6435 } 6436 } 6437 // Catch up with current buffer indices if thread is already running. 6438 // This is what makes a new client discard all buffered data. If the track's mRsmpInFront 6439 // was initialized to some value closer to the thread's mRsmpInFront, then the track could 6440 // see previously buffered data before it called start(), but with greater risk of overrun. 6441 6442 recordTrack->mResamplerBufferProvider->reset(); 6443 // clear any converter state as new data will be discontinuous 6444 recordTrack->mRecordBufferConverter->reset(); 6445 recordTrack->mState = TrackBase::STARTING_2; 6446 // signal thread to start 6447 mWaitWorkCV.broadcast(); 6448 if (mActiveTracks.indexOf(recordTrack) < 0) { 6449 ALOGV("Record failed to start"); 6450 status = BAD_VALUE; 6451 goto startError; 6452 } 6453 return status; 6454 } 6455 6456startError: 6457 if (recordTrack->isExternalTrack()) { 6458 AudioSystem::stopInput(mId, recordTrack->sessionId()); 6459 } 6460 recordTrack->clearSyncStartEvent(); 6461 // FIXME I wonder why we do not reset the state here? 6462 return status; 6463} 6464 6465void AudioFlinger::RecordThread::syncStartEventCallback(const wp<SyncEvent>& event) 6466{ 6467 sp<SyncEvent> strongEvent = event.promote(); 6468 6469 if (strongEvent != 0) { 6470 sp<RefBase> ptr = strongEvent->cookie().promote(); 6471 if (ptr != 0) { 6472 RecordTrack *recordTrack = (RecordTrack *)ptr.get(); 6473 recordTrack->handleSyncStartEvent(strongEvent); 6474 } 6475 } 6476} 6477 6478bool AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 6479 ALOGV("RecordThread::stop"); 6480 AutoMutex _l(mLock); 6481 if (mActiveTracks.indexOf(recordTrack) != 0 || recordTrack->mState == TrackBase::PAUSING) { 6482 return false; 6483 } 6484 // note that threadLoop may still be processing the track at this point [without lock] 6485 recordTrack->mState = TrackBase::PAUSING; 6486 // do not wait for mStartStopCond if exiting 6487 if (exitPending()) { 6488 return true; 6489 } 6490 // FIXME incorrect usage of wait: no explicit predicate or loop 6491 mStartStopCond.wait(mLock); 6492 // if we have been restarted, recordTrack is in mActiveTracks here 6493 if (exitPending() || mActiveTracks.indexOf(recordTrack) != 0) { 6494 ALOGV("Record stopped OK"); 6495 return true; 6496 } 6497 return false; 6498} 6499 6500bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event __unused) const 6501{ 6502 return false; 6503} 6504 6505status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event __unused) 6506{ 6507#if 0 // This branch is currently dead code, but is preserved in case it will be needed in future 6508 if (!isValidSyncEvent(event)) { 6509 return BAD_VALUE; 6510 } 6511 6512 audio_session_t eventSession = event->triggerSession(); 6513 status_t ret = NAME_NOT_FOUND; 6514 6515 Mutex::Autolock _l(mLock); 6516 6517 for (size_t i = 0; i < mTracks.size(); i++) { 6518 sp<RecordTrack> track = mTracks[i]; 6519 if (eventSession == track->sessionId()) { 6520 (void) track->setSyncEvent(event); 6521 ret = NO_ERROR; 6522 } 6523 } 6524 return ret; 6525#else 6526 return BAD_VALUE; 6527#endif 6528} 6529 6530// destroyTrack_l() must be called with ThreadBase::mLock held 6531void AudioFlinger::RecordThread::destroyTrack_l(const sp<RecordTrack>& track) 6532{ 6533 track->terminate(); 6534 track->mState = TrackBase::STOPPED; 6535 // active tracks are removed by threadLoop() 6536 if (mActiveTracks.indexOf(track) < 0) { 6537 removeTrack_l(track); 6538 } 6539} 6540 6541void AudioFlinger::RecordThread::removeTrack_l(const sp<RecordTrack>& track) 6542{ 6543 mTracks.remove(track); 6544 // need anything related to effects here? 6545 if (track->isFastTrack()) { 6546 ALOG_ASSERT(!mFastTrackAvail); 6547 mFastTrackAvail = true; 6548 } 6549} 6550 6551void AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 6552{ 6553 dumpInternals(fd, args); 6554 dumpTracks(fd, args); 6555 dumpEffectChains(fd, args); 6556} 6557 6558void AudioFlinger::RecordThread::dumpInternals(int fd, const Vector<String16>& args) 6559{ 6560 dprintf(fd, "\nInput thread %p:\n", this); 6561 6562 dumpBase(fd, args); 6563 6564 if (mActiveTracks.size() == 0) { 6565 dprintf(fd, " No active record clients\n"); 6566 } 6567 dprintf(fd, " Fast capture thread: %s\n", hasFastCapture() ? "yes" : "no"); 6568 dprintf(fd, " Fast track available: %s\n", mFastTrackAvail ? "yes" : "no"); 6569 6570 // Make a non-atomic copy of fast capture dump state so it won't change underneath us 6571 // while we are dumping it. It may be inconsistent, but it won't mutate! 6572 // This is a large object so we place it on the heap. 6573 // FIXME 25972958: Need an intelligent copy constructor that does not touch unused pages. 6574 const FastCaptureDumpState *copy = new FastCaptureDumpState(mFastCaptureDumpState); 6575 copy->dump(fd); 6576 delete copy; 6577} 6578 6579void AudioFlinger::RecordThread::dumpTracks(int fd, const Vector<String16>& args __unused) 6580{ 6581 const size_t SIZE = 256; 6582 char buffer[SIZE]; 6583 String8 result; 6584 6585 size_t numtracks = mTracks.size(); 6586 size_t numactive = mActiveTracks.size(); 6587 size_t numactiveseen = 0; 6588 dprintf(fd, " %zu Tracks", numtracks); 6589 if (numtracks) { 6590 dprintf(fd, " of which %zu are active\n", numactive); 6591 RecordTrack::appendDumpHeader(result); 6592 for (size_t i = 0; i < numtracks ; ++i) { 6593 sp<RecordTrack> track = mTracks[i]; 6594 if (track != 0) { 6595 bool active = mActiveTracks.indexOf(track) >= 0; 6596 if (active) { 6597 numactiveseen++; 6598 } 6599 track->dump(buffer, SIZE, active); 6600 result.append(buffer); 6601 } 6602 } 6603 } else { 6604 dprintf(fd, "\n"); 6605 } 6606 6607 if (numactiveseen != numactive) { 6608 snprintf(buffer, SIZE, " The following tracks are in the active list but" 6609 " not in the track list\n"); 6610 result.append(buffer); 6611 RecordTrack::appendDumpHeader(result); 6612 for (size_t i = 0; i < numactive; ++i) { 6613 sp<RecordTrack> track = mActiveTracks[i]; 6614 if (mTracks.indexOf(track) < 0) { 6615 track->dump(buffer, SIZE, true); 6616 result.append(buffer); 6617 } 6618 } 6619 6620 } 6621 write(fd, result.string(), result.size()); 6622} 6623 6624 6625void AudioFlinger::RecordThread::ResamplerBufferProvider::reset() 6626{ 6627 sp<ThreadBase> threadBase = mRecordTrack->mThread.promote(); 6628 RecordThread *recordThread = (RecordThread *) threadBase.get(); 6629 mRsmpInFront = recordThread->mRsmpInRear; 6630 mRsmpInUnrel = 0; 6631} 6632 6633void AudioFlinger::RecordThread::ResamplerBufferProvider::sync( 6634 size_t *framesAvailable, bool *hasOverrun) 6635{ 6636 sp<ThreadBase> threadBase = mRecordTrack->mThread.promote(); 6637 RecordThread *recordThread = (RecordThread *) threadBase.get(); 6638 const int32_t rear = recordThread->mRsmpInRear; 6639 const int32_t front = mRsmpInFront; 6640 const ssize_t filled = rear - front; 6641 6642 size_t framesIn; 6643 bool overrun = false; 6644 if (filled < 0) { 6645 // should not happen, but treat like a massive overrun and re-sync 6646 framesIn = 0; 6647 mRsmpInFront = rear; 6648 overrun = true; 6649 } else if ((size_t) filled <= recordThread->mRsmpInFrames) { 6650 framesIn = (size_t) filled; 6651 } else { 6652 // client is not keeping up with server, but give it latest data 6653 framesIn = recordThread->mRsmpInFrames; 6654 mRsmpInFront = /* front = */ rear - framesIn; 6655 overrun = true; 6656 } 6657 if (framesAvailable != NULL) { 6658 *framesAvailable = framesIn; 6659 } 6660 if (hasOverrun != NULL) { 6661 *hasOverrun = overrun; 6662 } 6663} 6664 6665// AudioBufferProvider interface 6666status_t AudioFlinger::RecordThread::ResamplerBufferProvider::getNextBuffer( 6667 AudioBufferProvider::Buffer* buffer) 6668{ 6669 sp<ThreadBase> threadBase = mRecordTrack->mThread.promote(); 6670 if (threadBase == 0) { 6671 buffer->frameCount = 0; 6672 buffer->raw = NULL; 6673 return NOT_ENOUGH_DATA; 6674 } 6675 RecordThread *recordThread = (RecordThread *) threadBase.get(); 6676 int32_t rear = recordThread->mRsmpInRear; 6677 int32_t front = mRsmpInFront; 6678 ssize_t filled = rear - front; 6679 // FIXME should not be P2 (don't want to increase latency) 6680 // FIXME if client not keeping up, discard 6681 LOG_ALWAYS_FATAL_IF(!(0 <= filled && (size_t) filled <= recordThread->mRsmpInFrames)); 6682 // 'filled' may be non-contiguous, so return only the first contiguous chunk 6683 front &= recordThread->mRsmpInFramesP2 - 1; 6684 size_t part1 = recordThread->mRsmpInFramesP2 - front; 6685 if (part1 > (size_t) filled) { 6686 part1 = filled; 6687 } 6688 size_t ask = buffer->frameCount; 6689 ALOG_ASSERT(ask > 0); 6690 if (part1 > ask) { 6691 part1 = ask; 6692 } 6693 if (part1 == 0) { 6694 // out of data is fine since the resampler will return a short-count. 6695 buffer->raw = NULL; 6696 buffer->frameCount = 0; 6697 mRsmpInUnrel = 0; 6698 return NOT_ENOUGH_DATA; 6699 } 6700 6701 buffer->raw = (uint8_t*)recordThread->mRsmpInBuffer + front * recordThread->mFrameSize; 6702 buffer->frameCount = part1; 6703 mRsmpInUnrel = part1; 6704 return NO_ERROR; 6705} 6706 6707// AudioBufferProvider interface 6708void AudioFlinger::RecordThread::ResamplerBufferProvider::releaseBuffer( 6709 AudioBufferProvider::Buffer* buffer) 6710{ 6711 size_t stepCount = buffer->frameCount; 6712 if (stepCount == 0) { 6713 return; 6714 } 6715 ALOG_ASSERT(stepCount <= mRsmpInUnrel); 6716 mRsmpInUnrel -= stepCount; 6717 mRsmpInFront += stepCount; 6718 buffer->raw = NULL; 6719 buffer->frameCount = 0; 6720} 6721 6722AudioFlinger::RecordThread::RecordBufferConverter::RecordBufferConverter( 6723 audio_channel_mask_t srcChannelMask, audio_format_t srcFormat, 6724 uint32_t srcSampleRate, 6725 audio_channel_mask_t dstChannelMask, audio_format_t dstFormat, 6726 uint32_t dstSampleRate) : 6727 mSrcChannelMask(AUDIO_CHANNEL_INVALID), // updateParameters will set following vars 6728 // mSrcFormat 6729 // mSrcSampleRate 6730 // mDstChannelMask 6731 // mDstFormat 6732 // mDstSampleRate 6733 // mSrcChannelCount 6734 // mDstChannelCount 6735 // mDstFrameSize 6736 mBuf(NULL), mBufFrames(0), mBufFrameSize(0), 6737 mResampler(NULL), 6738 mIsLegacyDownmix(false), 6739 mIsLegacyUpmix(false), 6740 mRequiresFloat(false), 6741 mInputConverterProvider(NULL) 6742{ 6743 (void)updateParameters(srcChannelMask, srcFormat, srcSampleRate, 6744 dstChannelMask, dstFormat, dstSampleRate); 6745} 6746 6747AudioFlinger::RecordThread::RecordBufferConverter::~RecordBufferConverter() { 6748 free(mBuf); 6749 delete mResampler; 6750 delete mInputConverterProvider; 6751} 6752 6753size_t AudioFlinger::RecordThread::RecordBufferConverter::convert(void *dst, 6754 AudioBufferProvider *provider, size_t frames) 6755{ 6756 if (mInputConverterProvider != NULL) { 6757 mInputConverterProvider->setBufferProvider(provider); 6758 provider = mInputConverterProvider; 6759 } 6760 6761 if (mResampler == NULL) { 6762 ALOGVV("NO RESAMPLING sampleRate:%u mSrcFormat:%#x mDstFormat:%#x", 6763 mSrcSampleRate, mSrcFormat, mDstFormat); 6764 6765 AudioBufferProvider::Buffer buffer; 6766 for (size_t i = frames; i > 0; ) { 6767 buffer.frameCount = i; 6768 status_t status = provider->getNextBuffer(&buffer); 6769 if (status != OK || buffer.frameCount == 0) { 6770 frames -= i; // cannot fill request. 6771 break; 6772 } 6773 // format convert to destination buffer 6774 convertNoResampler(dst, buffer.raw, buffer.frameCount); 6775 6776 dst = (int8_t*)dst + buffer.frameCount * mDstFrameSize; 6777 i -= buffer.frameCount; 6778 provider->releaseBuffer(&buffer); 6779 } 6780 } else { 6781 ALOGVV("RESAMPLING mSrcSampleRate:%u mDstSampleRate:%u mSrcFormat:%#x mDstFormat:%#x", 6782 mSrcSampleRate, mDstSampleRate, mSrcFormat, mDstFormat); 6783 6784 // reallocate buffer if needed 6785 if (mBufFrameSize != 0 && mBufFrames < frames) { 6786 free(mBuf); 6787 mBufFrames = frames; 6788 (void)posix_memalign(&mBuf, 32, mBufFrames * mBufFrameSize); 6789 } 6790 // resampler accumulates, but we only have one source track 6791 memset(mBuf, 0, frames * mBufFrameSize); 6792 frames = mResampler->resample((int32_t*)mBuf, frames, provider); 6793 // format convert to destination buffer 6794 convertResampler(dst, mBuf, frames); 6795 } 6796 return frames; 6797} 6798 6799status_t AudioFlinger::RecordThread::RecordBufferConverter::updateParameters( 6800 audio_channel_mask_t srcChannelMask, audio_format_t srcFormat, 6801 uint32_t srcSampleRate, 6802 audio_channel_mask_t dstChannelMask, audio_format_t dstFormat, 6803 uint32_t dstSampleRate) 6804{ 6805 // quick evaluation if there is any change. 6806 if (mSrcFormat == srcFormat 6807 && mSrcChannelMask == srcChannelMask 6808 && mSrcSampleRate == srcSampleRate 6809 && mDstFormat == dstFormat 6810 && mDstChannelMask == dstChannelMask 6811 && mDstSampleRate == dstSampleRate) { 6812 return NO_ERROR; 6813 } 6814 6815 ALOGV("RecordBufferConverter updateParameters srcMask:%#x dstMask:%#x" 6816 " srcFormat:%#x dstFormat:%#x srcRate:%u dstRate:%u", 6817 srcChannelMask, dstChannelMask, srcFormat, dstFormat, srcSampleRate, dstSampleRate); 6818 const bool valid = 6819 audio_is_input_channel(srcChannelMask) 6820 && audio_is_input_channel(dstChannelMask) 6821 && audio_is_valid_format(srcFormat) && audio_is_linear_pcm(srcFormat) 6822 && audio_is_valid_format(dstFormat) && audio_is_linear_pcm(dstFormat) 6823 && (srcSampleRate <= dstSampleRate * AUDIO_RESAMPLER_DOWN_RATIO_MAX) 6824 ; // no upsampling checks for now 6825 if (!valid) { 6826 return BAD_VALUE; 6827 } 6828 6829 mSrcFormat = srcFormat; 6830 mSrcChannelMask = srcChannelMask; 6831 mSrcSampleRate = srcSampleRate; 6832 mDstFormat = dstFormat; 6833 mDstChannelMask = dstChannelMask; 6834 mDstSampleRate = dstSampleRate; 6835 6836 // compute derived parameters 6837 mSrcChannelCount = audio_channel_count_from_in_mask(srcChannelMask); 6838 mDstChannelCount = audio_channel_count_from_in_mask(dstChannelMask); 6839 mDstFrameSize = mDstChannelCount * audio_bytes_per_sample(mDstFormat); 6840 6841 // do we need to resample? 6842 delete mResampler; 6843 mResampler = NULL; 6844 if (mSrcSampleRate != mDstSampleRate) { 6845 mResampler = AudioResampler::create(AUDIO_FORMAT_PCM_FLOAT, 6846 mSrcChannelCount, mDstSampleRate); 6847 mResampler->setSampleRate(mSrcSampleRate); 6848 mResampler->setVolume(AudioMixer::UNITY_GAIN_FLOAT, AudioMixer::UNITY_GAIN_FLOAT); 6849 } 6850 6851 // are we running legacy channel conversion modes? 6852 mIsLegacyDownmix = (mSrcChannelMask == AUDIO_CHANNEL_IN_STEREO 6853 || mSrcChannelMask == AUDIO_CHANNEL_IN_FRONT_BACK) 6854 && mDstChannelMask == AUDIO_CHANNEL_IN_MONO; 6855 mIsLegacyUpmix = mSrcChannelMask == AUDIO_CHANNEL_IN_MONO 6856 && (mDstChannelMask == AUDIO_CHANNEL_IN_STEREO 6857 || mDstChannelMask == AUDIO_CHANNEL_IN_FRONT_BACK); 6858 6859 // do we need to process in float? 6860 mRequiresFloat = mResampler != NULL || mIsLegacyDownmix || mIsLegacyUpmix; 6861 6862 // do we need a staging buffer to convert for destination (we can still optimize this)? 6863 // we use mBufFrameSize > 0 to indicate both frame size as well as buffer necessity 6864 if (mResampler != NULL) { 6865 mBufFrameSize = max(mSrcChannelCount, FCC_2) 6866 * audio_bytes_per_sample(AUDIO_FORMAT_PCM_FLOAT); 6867 } else if (mIsLegacyUpmix || mIsLegacyDownmix) { // legacy modes always float 6868 mBufFrameSize = mDstChannelCount * audio_bytes_per_sample(AUDIO_FORMAT_PCM_FLOAT); 6869 } else if (mSrcChannelMask != mDstChannelMask && mDstFormat != mSrcFormat) { 6870 mBufFrameSize = mDstChannelCount * audio_bytes_per_sample(mSrcFormat); 6871 } else { 6872 mBufFrameSize = 0; 6873 } 6874 mBufFrames = 0; // force the buffer to be resized. 6875 6876 // do we need an input converter buffer provider to give us float? 6877 delete mInputConverterProvider; 6878 mInputConverterProvider = NULL; 6879 if (mRequiresFloat && mSrcFormat != AUDIO_FORMAT_PCM_FLOAT) { 6880 mInputConverterProvider = new ReformatBufferProvider( 6881 audio_channel_count_from_in_mask(mSrcChannelMask), 6882 mSrcFormat, 6883 AUDIO_FORMAT_PCM_FLOAT, 6884 256 /* provider buffer frame count */); 6885 } 6886 6887 // do we need a remixer to do channel mask conversion 6888 if (!mIsLegacyDownmix && !mIsLegacyUpmix && mSrcChannelMask != mDstChannelMask) { 6889 (void) memcpy_by_index_array_initialization_from_channel_mask( 6890 mIdxAry, ARRAY_SIZE(mIdxAry), mDstChannelMask, mSrcChannelMask); 6891 } 6892 return NO_ERROR; 6893} 6894 6895void AudioFlinger::RecordThread::RecordBufferConverter::convertNoResampler( 6896 void *dst, const void *src, size_t frames) 6897{ 6898 // src is native type unless there is legacy upmix or downmix, whereupon it is float. 6899 if (mBufFrameSize != 0 && mBufFrames < frames) { 6900 free(mBuf); 6901 mBufFrames = frames; 6902 (void)posix_memalign(&mBuf, 32, mBufFrames * mBufFrameSize); 6903 } 6904 // do we need to do legacy upmix and downmix? 6905 if (mIsLegacyUpmix || mIsLegacyDownmix) { 6906 void *dstBuf = mBuf != NULL ? mBuf : dst; 6907 if (mIsLegacyUpmix) { 6908 upmix_to_stereo_float_from_mono_float((float *)dstBuf, 6909 (const float *)src, frames); 6910 } else /*mIsLegacyDownmix */ { 6911 downmix_to_mono_float_from_stereo_float((float *)dstBuf, 6912 (const float *)src, frames); 6913 } 6914 if (mBuf != NULL) { 6915 memcpy_by_audio_format(dst, mDstFormat, mBuf, AUDIO_FORMAT_PCM_FLOAT, 6916 frames * mDstChannelCount); 6917 } 6918 return; 6919 } 6920 // do we need to do channel mask conversion? 6921 if (mSrcChannelMask != mDstChannelMask) { 6922 void *dstBuf = mBuf != NULL ? mBuf : dst; 6923 memcpy_by_index_array(dstBuf, mDstChannelCount, 6924 src, mSrcChannelCount, mIdxAry, audio_bytes_per_sample(mSrcFormat), frames); 6925 if (dstBuf == dst) { 6926 return; // format is the same 6927 } 6928 } 6929 // convert to destination buffer 6930 const void *convertBuf = mBuf != NULL ? mBuf : src; 6931 memcpy_by_audio_format(dst, mDstFormat, convertBuf, mSrcFormat, 6932 frames * mDstChannelCount); 6933} 6934 6935void AudioFlinger::RecordThread::RecordBufferConverter::convertResampler( 6936 void *dst, /*not-a-const*/ void *src, size_t frames) 6937{ 6938 // src buffer format is ALWAYS float when entering this routine 6939 if (mIsLegacyUpmix) { 6940 ; // mono to stereo already handled by resampler 6941 } else if (mIsLegacyDownmix 6942 || (mSrcChannelMask == mDstChannelMask && mSrcChannelCount == 1)) { 6943 // the resampler outputs stereo for mono input channel (a feature?) 6944 // must convert to mono 6945 downmix_to_mono_float_from_stereo_float((float *)src, 6946 (const float *)src, frames); 6947 } else if (mSrcChannelMask != mDstChannelMask) { 6948 // convert to mono channel again for channel mask conversion (could be skipped 6949 // with further optimization). 6950 if (mSrcChannelCount == 1) { 6951 downmix_to_mono_float_from_stereo_float((float *)src, 6952 (const float *)src, frames); 6953 } 6954 // convert to destination format (in place, OK as float is larger than other types) 6955 if (mDstFormat != AUDIO_FORMAT_PCM_FLOAT) { 6956 memcpy_by_audio_format(src, mDstFormat, src, AUDIO_FORMAT_PCM_FLOAT, 6957 frames * mSrcChannelCount); 6958 } 6959 // channel convert and save to dst 6960 memcpy_by_index_array(dst, mDstChannelCount, 6961 src, mSrcChannelCount, mIdxAry, audio_bytes_per_sample(mDstFormat), frames); 6962 return; 6963 } 6964 // convert to destination format and save to dst 6965 memcpy_by_audio_format(dst, mDstFormat, src, AUDIO_FORMAT_PCM_FLOAT, 6966 frames * mDstChannelCount); 6967} 6968 6969bool AudioFlinger::RecordThread::checkForNewParameter_l(const String8& keyValuePair, 6970 status_t& status) 6971{ 6972 bool reconfig = false; 6973 6974 status = NO_ERROR; 6975 6976 audio_format_t reqFormat = mFormat; 6977 uint32_t samplingRate = mSampleRate; 6978 // TODO this may change if we want to support capture from HDMI PCM multi channel (e.g on TVs). 6979 audio_channel_mask_t channelMask = audio_channel_in_mask_from_count(mChannelCount); 6980 6981 AudioParameter param = AudioParameter(keyValuePair); 6982 int value; 6983 6984 // scope for AutoPark extends to end of method 6985 AutoPark<FastCapture> park(mFastCapture); 6986 6987 // TODO Investigate when this code runs. Check with audio policy when a sample rate and 6988 // channel count change can be requested. Do we mandate the first client defines the 6989 // HAL sampling rate and channel count or do we allow changes on the fly? 6990 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 6991 samplingRate = value; 6992 reconfig = true; 6993 } 6994 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 6995 if (!audio_is_linear_pcm((audio_format_t) value)) { 6996 status = BAD_VALUE; 6997 } else { 6998 reqFormat = (audio_format_t) value; 6999 reconfig = true; 7000 } 7001 } 7002 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 7003 audio_channel_mask_t mask = (audio_channel_mask_t) value; 7004 if (!audio_is_input_channel(mask) || 7005 audio_channel_count_from_in_mask(mask) > FCC_8) { 7006 status = BAD_VALUE; 7007 } else { 7008 channelMask = mask; 7009 reconfig = true; 7010 } 7011 } 7012 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 7013 // do not accept frame count changes if tracks are open as the track buffer 7014 // size depends on frame count and correct behavior would not be guaranteed 7015 // if frame count is changed after track creation 7016 if (mActiveTracks.size() > 0) { 7017 status = INVALID_OPERATION; 7018 } else { 7019 reconfig = true; 7020 } 7021 } 7022 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 7023 // forward device change to effects that have requested to be 7024 // aware of attached audio device. 7025 for (size_t i = 0; i < mEffectChains.size(); i++) { 7026 mEffectChains[i]->setDevice_l(value); 7027 } 7028 7029 // store input device and output device but do not forward output device to audio HAL. 7030 // Note that status is ignored by the caller for output device 7031 // (see AudioFlinger::setParameters() 7032 if (audio_is_output_devices(value)) { 7033 mOutDevice = value; 7034 status = BAD_VALUE; 7035 } else { 7036 mInDevice = value; 7037 if (value != AUDIO_DEVICE_NONE) { 7038 mPrevInDevice = value; 7039 } 7040 // disable AEC and NS if the device is a BT SCO headset supporting those 7041 // pre processings 7042 if (mTracks.size() > 0) { 7043 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 7044 mAudioFlinger->btNrecIsOff(); 7045 for (size_t i = 0; i < mTracks.size(); i++) { 7046 sp<RecordTrack> track = mTracks[i]; 7047 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId()); 7048 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId()); 7049 } 7050 } 7051 } 7052 } 7053 if (param.getInt(String8(AudioParameter::keyInputSource), value) == NO_ERROR && 7054 mAudioSource != (audio_source_t)value) { 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]->setAudioSource_l((audio_source_t)value); 7059 } 7060 mAudioSource = (audio_source_t)value; 7061 } 7062 7063 if (status == NO_ERROR) { 7064 status = mInput->stream->common.set_parameters(&mInput->stream->common, 7065 keyValuePair.string()); 7066 if (status == INVALID_OPERATION) { 7067 inputStandBy(); 7068 status = mInput->stream->common.set_parameters(&mInput->stream->common, 7069 keyValuePair.string()); 7070 } 7071 if (reconfig) { 7072 if (status == BAD_VALUE && 7073 audio_is_linear_pcm(mInput->stream->common.get_format(&mInput->stream->common)) && 7074 audio_is_linear_pcm(reqFormat) && 7075 (mInput->stream->common.get_sample_rate(&mInput->stream->common) 7076 <= (AUDIO_RESAMPLER_DOWN_RATIO_MAX * samplingRate)) && 7077 audio_channel_count_from_in_mask( 7078 mInput->stream->common.get_channels(&mInput->stream->common)) <= FCC_8) { 7079 status = NO_ERROR; 7080 } 7081 if (status == NO_ERROR) { 7082 readInputParameters_l(); 7083 sendIoConfigEvent_l(AUDIO_INPUT_CONFIG_CHANGED); 7084 } 7085 } 7086 } 7087 7088 return reconfig; 7089} 7090 7091String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 7092{ 7093 Mutex::Autolock _l(mLock); 7094 if (initCheck() != NO_ERROR) { 7095 return String8(); 7096 } 7097 7098 char *s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 7099 const String8 out_s8(s); 7100 free(s); 7101 return out_s8; 7102} 7103 7104void AudioFlinger::RecordThread::ioConfigChanged(audio_io_config_event event, pid_t pid) { 7105 sp<AudioIoDescriptor> desc = new AudioIoDescriptor(); 7106 7107 desc->mIoHandle = mId; 7108 7109 switch (event) { 7110 case AUDIO_INPUT_OPENED: 7111 case AUDIO_INPUT_CONFIG_CHANGED: 7112 desc->mPatch = mPatch; 7113 desc->mChannelMask = mChannelMask; 7114 desc->mSamplingRate = mSampleRate; 7115 desc->mFormat = mFormat; 7116 desc->mFrameCount = mFrameCount; 7117 desc->mFrameCountHAL = mFrameCount; 7118 desc->mLatency = 0; 7119 break; 7120 7121 case AUDIO_INPUT_CLOSED: 7122 default: 7123 break; 7124 } 7125 mAudioFlinger->ioConfigChanged(event, desc, pid); 7126} 7127 7128void AudioFlinger::RecordThread::readInputParameters_l() 7129{ 7130 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 7131 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 7132 mChannelCount = audio_channel_count_from_in_mask(mChannelMask); 7133 if (mChannelCount > FCC_8) { 7134 ALOGE("HAL channel count %d > %d", mChannelCount, FCC_8); 7135 } 7136 mHALFormat = mInput->stream->common.get_format(&mInput->stream->common); 7137 mFormat = mHALFormat; 7138 if (!audio_is_linear_pcm(mFormat)) { 7139 ALOGE("HAL format %#x is not linear pcm", mFormat); 7140 } 7141 mFrameSize = audio_stream_in_frame_size(mInput->stream); 7142 mBufferSize = mInput->stream->common.get_buffer_size(&mInput->stream->common); 7143 mFrameCount = mBufferSize / mFrameSize; 7144 // This is the formula for calculating the temporary buffer size. 7145 // With 7 HAL buffers, we can guarantee ability to down-sample the input by ratio of 6:1 to 7146 // 1 full output buffer, regardless of the alignment of the available input. 7147 // The value is somewhat arbitrary, and could probably be even larger. 7148 // A larger value should allow more old data to be read after a track calls start(), 7149 // without increasing latency. 7150 // 7151 // Note this is independent of the maximum downsampling ratio permitted for capture. 7152 mRsmpInFrames = mFrameCount * 7; 7153 mRsmpInFramesP2 = roundup(mRsmpInFrames); 7154 free(mRsmpInBuffer); 7155 mRsmpInBuffer = NULL; 7156 7157 // TODO optimize audio capture buffer sizes ... 7158 // Here we calculate the size of the sliding buffer used as a source 7159 // for resampling. mRsmpInFramesP2 is currently roundup(mFrameCount * 7). 7160 // For current HAL frame counts, this is usually 2048 = 40 ms. It would 7161 // be better to have it derived from the pipe depth in the long term. 7162 // The current value is higher than necessary. However it should not add to latency. 7163 7164 // Over-allocate beyond mRsmpInFramesP2 to permit a HAL read past end of buffer 7165 size_t bufferSize = (mRsmpInFramesP2 + mFrameCount - 1) * mFrameSize; 7166 (void)posix_memalign(&mRsmpInBuffer, 32, bufferSize); 7167 memset(mRsmpInBuffer, 0, bufferSize); // if posix_memalign fails, will segv here. 7168 7169 // AudioRecord mSampleRate and mChannelCount are constant due to AudioRecord API constraints. 7170 // But if thread's mSampleRate or mChannelCount changes, how will that affect active tracks? 7171} 7172 7173uint32_t AudioFlinger::RecordThread::getInputFramesLost() 7174{ 7175 Mutex::Autolock _l(mLock); 7176 if (initCheck() != NO_ERROR) { 7177 return 0; 7178 } 7179 7180 return mInput->stream->get_input_frames_lost(mInput->stream); 7181} 7182 7183uint32_t AudioFlinger::RecordThread::hasAudioSession(audio_session_t sessionId) const 7184{ 7185 Mutex::Autolock _l(mLock); 7186 uint32_t result = 0; 7187 if (getEffectChain_l(sessionId) != 0) { 7188 result = EFFECT_SESSION; 7189 } 7190 7191 for (size_t i = 0; i < mTracks.size(); ++i) { 7192 if (sessionId == mTracks[i]->sessionId()) { 7193 result |= TRACK_SESSION; 7194 break; 7195 } 7196 } 7197 7198 return result; 7199} 7200 7201KeyedVector<audio_session_t, bool> AudioFlinger::RecordThread::sessionIds() const 7202{ 7203 KeyedVector<audio_session_t, bool> ids; 7204 Mutex::Autolock _l(mLock); 7205 for (size_t j = 0; j < mTracks.size(); ++j) { 7206 sp<RecordThread::RecordTrack> track = mTracks[j]; 7207 audio_session_t sessionId = track->sessionId(); 7208 if (ids.indexOfKey(sessionId) < 0) { 7209 ids.add(sessionId, true); 7210 } 7211 } 7212 return ids; 7213} 7214 7215AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 7216{ 7217 Mutex::Autolock _l(mLock); 7218 AudioStreamIn *input = mInput; 7219 mInput = NULL; 7220 return input; 7221} 7222 7223// this method must always be called either with ThreadBase mLock held or inside the thread loop 7224audio_stream_t* AudioFlinger::RecordThread::stream() const 7225{ 7226 if (mInput == NULL) { 7227 return NULL; 7228 } 7229 return &mInput->stream->common; 7230} 7231 7232status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 7233{ 7234 // only one chain per input thread 7235 if (mEffectChains.size() != 0) { 7236 ALOGW("addEffectChain_l() already one chain %p on thread %p", chain.get(), this); 7237 return INVALID_OPERATION; 7238 } 7239 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 7240 chain->setThread(this); 7241 chain->setInBuffer(NULL); 7242 chain->setOutBuffer(NULL); 7243 7244 checkSuspendOnAddEffectChain_l(chain); 7245 7246 // make sure enabled pre processing effects state is communicated to the HAL as we 7247 // just moved them to a new input stream. 7248 chain->syncHalEffectsState(); 7249 7250 mEffectChains.add(chain); 7251 7252 return NO_ERROR; 7253} 7254 7255size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 7256{ 7257 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 7258 ALOGW_IF(mEffectChains.size() != 1, 7259 "removeEffectChain_l() %p invalid chain size %zu on thread %p", 7260 chain.get(), mEffectChains.size(), this); 7261 if (mEffectChains.size() == 1) { 7262 mEffectChains.removeAt(0); 7263 } 7264 return 0; 7265} 7266 7267status_t AudioFlinger::RecordThread::createAudioPatch_l(const struct audio_patch *patch, 7268 audio_patch_handle_t *handle) 7269{ 7270 status_t status = NO_ERROR; 7271 7272 // store new device and send to effects 7273 mInDevice = patch->sources[0].ext.device.type; 7274 mPatch = *patch; 7275 for (size_t i = 0; i < mEffectChains.size(); i++) { 7276 mEffectChains[i]->setDevice_l(mInDevice); 7277 } 7278 7279 // disable AEC and NS if the device is a BT SCO headset supporting those 7280 // pre processings 7281 if (mTracks.size() > 0) { 7282 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 7283 mAudioFlinger->btNrecIsOff(); 7284 for (size_t i = 0; i < mTracks.size(); i++) { 7285 sp<RecordTrack> track = mTracks[i]; 7286 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId()); 7287 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId()); 7288 } 7289 } 7290 7291 // store new source and send to effects 7292 if (mAudioSource != patch->sinks[0].ext.mix.usecase.source) { 7293 mAudioSource = patch->sinks[0].ext.mix.usecase.source; 7294 for (size_t i = 0; i < mEffectChains.size(); i++) { 7295 mEffectChains[i]->setAudioSource_l(mAudioSource); 7296 } 7297 } 7298 7299 if (mInput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 7300 audio_hw_device_t *hwDevice = mInput->audioHwDev->hwDevice(); 7301 status = hwDevice->create_audio_patch(hwDevice, 7302 patch->num_sources, 7303 patch->sources, 7304 patch->num_sinks, 7305 patch->sinks, 7306 handle); 7307 } else { 7308 char *address; 7309 if (strcmp(patch->sources[0].ext.device.address, "") != 0) { 7310 address = audio_device_address_to_parameter( 7311 patch->sources[0].ext.device.type, 7312 patch->sources[0].ext.device.address); 7313 } else { 7314 address = (char *)calloc(1, 1); 7315 } 7316 AudioParameter param = AudioParameter(String8(address)); 7317 free(address); 7318 param.addInt(String8(AUDIO_PARAMETER_STREAM_ROUTING), 7319 (int)patch->sources[0].ext.device.type); 7320 param.addInt(String8(AUDIO_PARAMETER_STREAM_INPUT_SOURCE), 7321 (int)patch->sinks[0].ext.mix.usecase.source); 7322 status = mInput->stream->common.set_parameters(&mInput->stream->common, 7323 param.toString().string()); 7324 *handle = AUDIO_PATCH_HANDLE_NONE; 7325 } 7326 7327 if (mInDevice != mPrevInDevice) { 7328 sendIoConfigEvent_l(AUDIO_INPUT_CONFIG_CHANGED); 7329 mPrevInDevice = mInDevice; 7330 } 7331 7332 return status; 7333} 7334 7335status_t AudioFlinger::RecordThread::releaseAudioPatch_l(const audio_patch_handle_t handle) 7336{ 7337 status_t status = NO_ERROR; 7338 7339 mInDevice = AUDIO_DEVICE_NONE; 7340 7341 if (mInput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 7342 audio_hw_device_t *hwDevice = mInput->audioHwDev->hwDevice(); 7343 status = hwDevice->release_audio_patch(hwDevice, handle); 7344 } else { 7345 AudioParameter param; 7346 param.addInt(String8(AUDIO_PARAMETER_STREAM_ROUTING), 0); 7347 status = mInput->stream->common.set_parameters(&mInput->stream->common, 7348 param.toString().string()); 7349 } 7350 return status; 7351} 7352 7353void AudioFlinger::RecordThread::addPatchRecord(const sp<PatchRecord>& record) 7354{ 7355 Mutex::Autolock _l(mLock); 7356 mTracks.add(record); 7357} 7358 7359void AudioFlinger::RecordThread::deletePatchRecord(const sp<PatchRecord>& record) 7360{ 7361 Mutex::Autolock _l(mLock); 7362 destroyTrack_l(record); 7363} 7364 7365void AudioFlinger::RecordThread::getAudioPortConfig(struct audio_port_config *config) 7366{ 7367 ThreadBase::getAudioPortConfig(config); 7368 config->role = AUDIO_PORT_ROLE_SINK; 7369 config->ext.mix.hw_module = mInput->audioHwDev->handle(); 7370 config->ext.mix.usecase.source = mAudioSource; 7371} 7372 7373} // namespace android 7374