Threads.cpp revision 823b18ef45f0593386d9d5d20fbf9a0379ad0ebb
1/* 2** 3** Copyright 2012, The Android Open Source Project 4** 5** Licensed under the Apache License, Version 2.0 (the "License"); 6** you may not use this file except in compliance with the License. 7** You may obtain a copy of the License at 8** 9** http://www.apache.org/licenses/LICENSE-2.0 10** 11** Unless required by applicable law or agreed to in writing, software 12** distributed under the License is distributed on an "AS IS" BASIS, 13** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. 14** See the License for the specific language governing permissions and 15** limitations under the License. 16*/ 17 18 19#define LOG_TAG "AudioFlinger" 20//#define LOG_NDEBUG 0 21#define ATRACE_TAG ATRACE_TAG_AUDIO 22 23#include "Configuration.h" 24#include <math.h> 25#include <fcntl.h> 26#include <linux/futex.h> 27#include <sys/stat.h> 28#include <sys/syscall.h> 29#include <cutils/properties.h> 30#include <media/AudioParameter.h> 31#include <media/AudioResamplerPublic.h> 32#include <utils/Log.h> 33#include <utils/Trace.h> 34 35#include <private/media/AudioTrackShared.h> 36#include <hardware/audio.h> 37#include <audio_effects/effect_ns.h> 38#include <audio_effects/effect_aec.h> 39#include <audio_utils/conversion.h> 40#include <audio_utils/primitives.h> 41#include <audio_utils/format.h> 42#include <audio_utils/minifloat.h> 43 44// NBAIO implementations 45#include <media/nbaio/AudioStreamInSource.h> 46#include <media/nbaio/AudioStreamOutSink.h> 47#include <media/nbaio/MonoPipe.h> 48#include <media/nbaio/MonoPipeReader.h> 49#include <media/nbaio/Pipe.h> 50#include <media/nbaio/PipeReader.h> 51#include <media/nbaio/SourceAudioBufferProvider.h> 52#include <mediautils/BatteryNotifier.h> 53 54#include <powermanager/PowerManager.h> 55 56#include "AudioFlinger.h" 57#include "AudioMixer.h" 58#include "BufferProviders.h" 59#include "FastMixer.h" 60#include "FastCapture.h" 61#include "ServiceUtilities.h" 62#include "mediautils/SchedulingPolicyService.h" 63 64#ifdef ADD_BATTERY_DATA 65#include <media/IMediaPlayerService.h> 66#include <media/IMediaDeathNotifier.h> 67#endif 68 69#ifdef DEBUG_CPU_USAGE 70#include <cpustats/CentralTendencyStatistics.h> 71#include <cpustats/ThreadCpuUsage.h> 72#endif 73 74#include "AutoPark.h" 75 76// ---------------------------------------------------------------------------- 77 78// Note: the following macro is used for extremely verbose logging message. In 79// order to run with ALOG_ASSERT turned on, we need to have LOG_NDEBUG set to 80// 0; but one side effect of this is to turn all LOGV's as well. Some messages 81// are so verbose that we want to suppress them even when we have ALOG_ASSERT 82// turned on. Do not uncomment the #def below unless you really know what you 83// are doing and want to see all of the extremely verbose messages. 84//#define VERY_VERY_VERBOSE_LOGGING 85#ifdef VERY_VERY_VERBOSE_LOGGING 86#define ALOGVV ALOGV 87#else 88#define ALOGVV(a...) do { } while(0) 89#endif 90 91// TODO: Move these macro/inlines to a header file. 92#define max(a, b) ((a) > (b) ? (a) : (b)) 93template <typename T> 94static inline T min(const T& a, const T& b) 95{ 96 return a < b ? a : b; 97} 98 99#ifndef ARRAY_SIZE 100#define ARRAY_SIZE(a) (sizeof(a) / sizeof(a[0])) 101#endif 102 103namespace android { 104 105// retry counts for buffer fill timeout 106// 50 * ~20msecs = 1 second 107static const int8_t kMaxTrackRetries = 50; 108static const int8_t kMaxTrackStartupRetries = 50; 109// allow less retry attempts on direct output thread. 110// direct outputs can be a scarce resource in audio hardware and should 111// be released as quickly as possible. 112static const int8_t kMaxTrackRetriesDirect = 2; 113// retry count before removing active track in case of underrun on offloaded thread: 114// we need to make sure that AudioTrack client has enough time to send large buffers 115//FIXME may be more appropriate if expressed in time units. Need to revise how underrun is handled 116// for offloaded tracks 117static const int8_t kMaxTrackRetriesOffload = 10; 118static const int8_t kMaxTrackStartupRetriesOffload = 100; 119 120 121// don't warn about blocked writes or record buffer overflows more often than this 122static const nsecs_t kWarningThrottleNs = seconds(5); 123 124// RecordThread loop sleep time upon application overrun or audio HAL read error 125static const int kRecordThreadSleepUs = 5000; 126 127// maximum time to wait in sendConfigEvent_l() for a status to be received 128static const nsecs_t kConfigEventTimeoutNs = seconds(2); 129 130// minimum sleep time for the mixer thread loop when tracks are active but in underrun 131static const uint32_t kMinThreadSleepTimeUs = 5000; 132// maximum divider applied to the active sleep time in the mixer thread loop 133static const uint32_t kMaxThreadSleepTimeShift = 2; 134 135// minimum normal sink buffer size, expressed in milliseconds rather than frames 136// FIXME This should be based on experimentally observed scheduling jitter 137static const uint32_t kMinNormalSinkBufferSizeMs = 20; 138// maximum normal sink buffer size 139static const uint32_t kMaxNormalSinkBufferSizeMs = 24; 140 141// minimum capture buffer size in milliseconds to _not_ need a fast capture thread 142// FIXME This should be based on experimentally observed scheduling jitter 143static const uint32_t kMinNormalCaptureBufferSizeMs = 12; 144 145// Offloaded output thread standby delay: allows track transition without going to standby 146static const nsecs_t kOffloadStandbyDelayNs = seconds(1); 147 148// Direct output thread minimum sleep time in idle or active(underrun) state 149static const nsecs_t kDirectMinSleepTimeUs = 10000; 150 151// Offloaded output bit rate in bits per second when unknown. 152// Used for sleep time calculation, so use a high default bitrate to be conservative on sleep time. 153static const uint32_t kOffloadDefaultBitRateBps = 1500000; 154 155 156// Whether to use fast mixer 157static const enum { 158 FastMixer_Never, // never initialize or use: for debugging only 159 FastMixer_Always, // always initialize and use, even if not needed: for debugging only 160 // normal mixer multiplier is 1 161 FastMixer_Static, // initialize if needed, then use all the time if initialized, 162 // multiplier is calculated based on min & max normal mixer buffer size 163 FastMixer_Dynamic, // initialize if needed, then use dynamically depending on track load, 164 // multiplier is calculated based on min & max normal mixer buffer size 165 // FIXME for FastMixer_Dynamic: 166 // Supporting this option will require fixing HALs that can't handle large writes. 167 // For example, one HAL implementation returns an error from a large write, 168 // and another HAL implementation corrupts memory, possibly in the sample rate converter. 169 // We could either fix the HAL implementations, or provide a wrapper that breaks 170 // up large writes into smaller ones, and the wrapper would need to deal with scheduler. 171} kUseFastMixer = FastMixer_Static; 172 173// Whether to use fast capture 174static const enum { 175 FastCapture_Never, // never initialize or use: for debugging only 176 FastCapture_Always, // always initialize and use, even if not needed: for debugging only 177 FastCapture_Static, // initialize if needed, then use all the time if initialized 178} kUseFastCapture = FastCapture_Static; 179 180// Priorities for requestPriority 181static const int kPriorityAudioApp = 2; 182static const int kPriorityFastMixer = 3; 183static const int kPriorityFastCapture = 3; 184 185// IAudioFlinger::createTrack() has an in/out parameter 'pFrameCount' for the total size of the 186// track buffer in shared memory. Zero on input means to use a default value. For fast tracks, 187// AudioFlinger derives the default from HAL buffer size and 'fast track multiplier'. 188 189// This is the default value, if not specified by property. 190static const int kFastTrackMultiplier = 2; 191 192// The minimum and maximum allowed values 193static const int kFastTrackMultiplierMin = 1; 194static const int kFastTrackMultiplierMax = 2; 195 196// The actual value to use, which can be specified per-device via property af.fast_track_multiplier. 197static int sFastTrackMultiplier = kFastTrackMultiplier; 198 199// See Thread::readOnlyHeap(). 200// Initially this heap is used to allocate client buffers for "fast" AudioRecord. 201// Eventually it will be the single buffer that FastCapture writes into via HAL read(), 202// and that all "fast" AudioRecord clients read from. In either case, the size can be small. 203static const size_t kRecordThreadReadOnlyHeapSize = 0x2000; 204 205// ---------------------------------------------------------------------------- 206 207static pthread_once_t sFastTrackMultiplierOnce = PTHREAD_ONCE_INIT; 208 209static void sFastTrackMultiplierInit() 210{ 211 char value[PROPERTY_VALUE_MAX]; 212 if (property_get("af.fast_track_multiplier", value, NULL) > 0) { 213 char *endptr; 214 unsigned long ul = strtoul(value, &endptr, 0); 215 if (*endptr == '\0' && kFastTrackMultiplierMin <= ul && ul <= kFastTrackMultiplierMax) { 216 sFastTrackMultiplier = (int) ul; 217 } 218 } 219} 220 221// ---------------------------------------------------------------------------- 222 223#ifdef ADD_BATTERY_DATA 224// To collect the amplifier usage 225static void addBatteryData(uint32_t params) { 226 sp<IMediaPlayerService> service = IMediaDeathNotifier::getMediaPlayerService(); 227 if (service == NULL) { 228 // it already logged 229 return; 230 } 231 232 service->addBatteryData(params); 233} 234#endif 235 236// Track the CLOCK_BOOTTIME versus CLOCK_MONOTONIC timebase offset 237struct { 238 // call when you acquire a partial wakelock 239 void acquire(const sp<IBinder> &wakeLockToken) { 240 pthread_mutex_lock(&mLock); 241 if (wakeLockToken.get() == nullptr) { 242 adjustTimebaseOffset(&mBoottimeOffset, ExtendedTimestamp::TIMEBASE_BOOTTIME); 243 } else { 244 if (mCount == 0) { 245 adjustTimebaseOffset(&mBoottimeOffset, ExtendedTimestamp::TIMEBASE_BOOTTIME); 246 } 247 ++mCount; 248 } 249 pthread_mutex_unlock(&mLock); 250 } 251 252 // call when you release a partial wakelock. 253 void release(const sp<IBinder> &wakeLockToken) { 254 if (wakeLockToken.get() == nullptr) { 255 return; 256 } 257 pthread_mutex_lock(&mLock); 258 if (--mCount < 0) { 259 ALOGE("negative wakelock count"); 260 mCount = 0; 261 } 262 pthread_mutex_unlock(&mLock); 263 } 264 265 // retrieves the boottime timebase offset from monotonic. 266 int64_t getBoottimeOffset() { 267 pthread_mutex_lock(&mLock); 268 int64_t boottimeOffset = mBoottimeOffset; 269 pthread_mutex_unlock(&mLock); 270 return boottimeOffset; 271 } 272 273 // Adjusts the timebase offset between TIMEBASE_MONOTONIC 274 // and the selected timebase. 275 // Currently only TIMEBASE_BOOTTIME is allowed. 276 // 277 // This only needs to be called upon acquiring the first partial wakelock 278 // after all other partial wakelocks are released. 279 // 280 // We do an empirical measurement of the offset rather than parsing 281 // /proc/timer_list since the latter is not a formal kernel ABI. 282 static void adjustTimebaseOffset(int64_t *offset, ExtendedTimestamp::Timebase timebase) { 283 int clockbase; 284 switch (timebase) { 285 case ExtendedTimestamp::TIMEBASE_BOOTTIME: 286 clockbase = SYSTEM_TIME_BOOTTIME; 287 break; 288 default: 289 LOG_ALWAYS_FATAL("invalid timebase %d", timebase); 290 break; 291 } 292 // try three times to get the clock offset, choose the one 293 // with the minimum gap in measurements. 294 const int tries = 3; 295 nsecs_t bestGap, measured; 296 for (int i = 0; i < tries; ++i) { 297 const nsecs_t tmono = systemTime(SYSTEM_TIME_MONOTONIC); 298 const nsecs_t tbase = systemTime(clockbase); 299 const nsecs_t tmono2 = systemTime(SYSTEM_TIME_MONOTONIC); 300 const nsecs_t gap = tmono2 - tmono; 301 if (i == 0 || gap < bestGap) { 302 bestGap = gap; 303 measured = tbase - ((tmono + tmono2) >> 1); 304 } 305 } 306 307 // to avoid micro-adjusting, we don't change the timebase 308 // unless it is significantly different. 309 // 310 // Assumption: It probably takes more than toleranceNs to 311 // suspend and resume the device. 312 static int64_t toleranceNs = 10000; // 10 us 313 if (llabs(*offset - measured) > toleranceNs) { 314 ALOGV("Adjusting timebase offset old: %lld new: %lld", 315 (long long)*offset, (long long)measured); 316 *offset = measured; 317 } 318 } 319 320 pthread_mutex_t mLock; 321 int32_t mCount; 322 int64_t mBoottimeOffset; 323} gBoottime = { PTHREAD_MUTEX_INITIALIZER, 0, 0 }; // static, so use POD initialization 324 325// ---------------------------------------------------------------------------- 326// CPU Stats 327// ---------------------------------------------------------------------------- 328 329class CpuStats { 330public: 331 CpuStats(); 332 void sample(const String8 &title); 333#ifdef DEBUG_CPU_USAGE 334private: 335 ThreadCpuUsage mCpuUsage; // instantaneous thread CPU usage in wall clock ns 336 CentralTendencyStatistics mWcStats; // statistics on thread CPU usage in wall clock ns 337 338 CentralTendencyStatistics mHzStats; // statistics on thread CPU usage in cycles 339 340 int mCpuNum; // thread's current CPU number 341 int mCpukHz; // frequency of thread's current CPU in kHz 342#endif 343}; 344 345CpuStats::CpuStats() 346#ifdef DEBUG_CPU_USAGE 347 : mCpuNum(-1), mCpukHz(-1) 348#endif 349{ 350} 351 352void CpuStats::sample(const String8 &title 353#ifndef DEBUG_CPU_USAGE 354 __unused 355#endif 356 ) { 357#ifdef DEBUG_CPU_USAGE 358 // get current thread's delta CPU time in wall clock ns 359 double wcNs; 360 bool valid = mCpuUsage.sampleAndEnable(wcNs); 361 362 // record sample for wall clock statistics 363 if (valid) { 364 mWcStats.sample(wcNs); 365 } 366 367 // get the current CPU number 368 int cpuNum = sched_getcpu(); 369 370 // get the current CPU frequency in kHz 371 int cpukHz = mCpuUsage.getCpukHz(cpuNum); 372 373 // check if either CPU number or frequency changed 374 if (cpuNum != mCpuNum || cpukHz != mCpukHz) { 375 mCpuNum = cpuNum; 376 mCpukHz = cpukHz; 377 // ignore sample for purposes of cycles 378 valid = false; 379 } 380 381 // if no change in CPU number or frequency, then record sample for cycle statistics 382 if (valid && mCpukHz > 0) { 383 double cycles = wcNs * cpukHz * 0.000001; 384 mHzStats.sample(cycles); 385 } 386 387 unsigned n = mWcStats.n(); 388 // mCpuUsage.elapsed() is expensive, so don't call it every loop 389 if ((n & 127) == 1) { 390 long long elapsed = mCpuUsage.elapsed(); 391 if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) { 392 double perLoop = elapsed / (double) n; 393 double perLoop100 = perLoop * 0.01; 394 double perLoop1k = perLoop * 0.001; 395 double mean = mWcStats.mean(); 396 double stddev = mWcStats.stddev(); 397 double minimum = mWcStats.minimum(); 398 double maximum = mWcStats.maximum(); 399 double meanCycles = mHzStats.mean(); 400 double stddevCycles = mHzStats.stddev(); 401 double minCycles = mHzStats.minimum(); 402 double maxCycles = mHzStats.maximum(); 403 mCpuUsage.resetElapsed(); 404 mWcStats.reset(); 405 mHzStats.reset(); 406 ALOGD("CPU usage for %s over past %.1f secs\n" 407 " (%u mixer loops at %.1f mean ms per loop):\n" 408 " us per mix loop: mean=%.0f stddev=%.0f min=%.0f max=%.0f\n" 409 " %% of wall: mean=%.1f stddev=%.1f min=%.1f max=%.1f\n" 410 " MHz: mean=%.1f, stddev=%.1f, min=%.1f max=%.1f", 411 title.string(), 412 elapsed * .000000001, n, perLoop * .000001, 413 mean * .001, 414 stddev * .001, 415 minimum * .001, 416 maximum * .001, 417 mean / perLoop100, 418 stddev / perLoop100, 419 minimum / perLoop100, 420 maximum / perLoop100, 421 meanCycles / perLoop1k, 422 stddevCycles / perLoop1k, 423 minCycles / perLoop1k, 424 maxCycles / perLoop1k); 425 426 } 427 } 428#endif 429}; 430 431// ---------------------------------------------------------------------------- 432// ThreadBase 433// ---------------------------------------------------------------------------- 434 435// static 436const char *AudioFlinger::ThreadBase::threadTypeToString(AudioFlinger::ThreadBase::type_t type) 437{ 438 switch (type) { 439 case MIXER: 440 return "MIXER"; 441 case DIRECT: 442 return "DIRECT"; 443 case DUPLICATING: 444 return "DUPLICATING"; 445 case RECORD: 446 return "RECORD"; 447 case OFFLOAD: 448 return "OFFLOAD"; 449 default: 450 return "unknown"; 451 } 452} 453 454String8 devicesToString(audio_devices_t devices) 455{ 456 static const struct mapping { 457 audio_devices_t mDevices; 458 const char * mString; 459 } mappingsOut[] = { 460 {AUDIO_DEVICE_OUT_EARPIECE, "EARPIECE"}, 461 {AUDIO_DEVICE_OUT_SPEAKER, "SPEAKER"}, 462 {AUDIO_DEVICE_OUT_WIRED_HEADSET, "WIRED_HEADSET"}, 463 {AUDIO_DEVICE_OUT_WIRED_HEADPHONE, "WIRED_HEADPHONE"}, 464 {AUDIO_DEVICE_OUT_BLUETOOTH_SCO, "BLUETOOTH_SCO"}, 465 {AUDIO_DEVICE_OUT_BLUETOOTH_SCO_HEADSET, "BLUETOOTH_SCO_HEADSET"}, 466 {AUDIO_DEVICE_OUT_BLUETOOTH_SCO_CARKIT, "BLUETOOTH_SCO_CARKIT"}, 467 {AUDIO_DEVICE_OUT_BLUETOOTH_A2DP, "BLUETOOTH_A2DP"}, 468 {AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES,"BLUETOOTH_A2DP_HEADPHONES"}, 469 {AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER, "BLUETOOTH_A2DP_SPEAKER"}, 470 {AUDIO_DEVICE_OUT_AUX_DIGITAL, "AUX_DIGITAL"}, 471 {AUDIO_DEVICE_OUT_HDMI, "HDMI"}, 472 {AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET,"ANLG_DOCK_HEADSET"}, 473 {AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET,"DGTL_DOCK_HEADSET"}, 474 {AUDIO_DEVICE_OUT_USB_ACCESSORY, "USB_ACCESSORY"}, 475 {AUDIO_DEVICE_OUT_USB_DEVICE, "USB_DEVICE"}, 476 {AUDIO_DEVICE_OUT_TELEPHONY_TX, "TELEPHONY_TX"}, 477 {AUDIO_DEVICE_OUT_LINE, "LINE"}, 478 {AUDIO_DEVICE_OUT_HDMI_ARC, "HDMI_ARC"}, 479 {AUDIO_DEVICE_OUT_SPDIF, "SPDIF"}, 480 {AUDIO_DEVICE_OUT_FM, "FM"}, 481 {AUDIO_DEVICE_OUT_AUX_LINE, "AUX_LINE"}, 482 {AUDIO_DEVICE_OUT_SPEAKER_SAFE, "SPEAKER_SAFE"}, 483 {AUDIO_DEVICE_OUT_IP, "IP"}, 484 {AUDIO_DEVICE_OUT_BUS, "BUS"}, 485 {AUDIO_DEVICE_NONE, "NONE"}, // must be last 486 }, mappingsIn[] = { 487 {AUDIO_DEVICE_IN_COMMUNICATION, "COMMUNICATION"}, 488 {AUDIO_DEVICE_IN_AMBIENT, "AMBIENT"}, 489 {AUDIO_DEVICE_IN_BUILTIN_MIC, "BUILTIN_MIC"}, 490 {AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET, "BLUETOOTH_SCO_HEADSET"}, 491 {AUDIO_DEVICE_IN_WIRED_HEADSET, "WIRED_HEADSET"}, 492 {AUDIO_DEVICE_IN_AUX_DIGITAL, "AUX_DIGITAL"}, 493 {AUDIO_DEVICE_IN_VOICE_CALL, "VOICE_CALL"}, 494 {AUDIO_DEVICE_IN_TELEPHONY_RX, "TELEPHONY_RX"}, 495 {AUDIO_DEVICE_IN_BACK_MIC, "BACK_MIC"}, 496 {AUDIO_DEVICE_IN_REMOTE_SUBMIX, "REMOTE_SUBMIX"}, 497 {AUDIO_DEVICE_IN_ANLG_DOCK_HEADSET, "ANLG_DOCK_HEADSET"}, 498 {AUDIO_DEVICE_IN_DGTL_DOCK_HEADSET, "DGTL_DOCK_HEADSET"}, 499 {AUDIO_DEVICE_IN_USB_ACCESSORY, "USB_ACCESSORY"}, 500 {AUDIO_DEVICE_IN_USB_DEVICE, "USB_DEVICE"}, 501 {AUDIO_DEVICE_IN_FM_TUNER, "FM_TUNER"}, 502 {AUDIO_DEVICE_IN_TV_TUNER, "TV_TUNER"}, 503 {AUDIO_DEVICE_IN_LINE, "LINE"}, 504 {AUDIO_DEVICE_IN_SPDIF, "SPDIF"}, 505 {AUDIO_DEVICE_IN_BLUETOOTH_A2DP, "BLUETOOTH_A2DP"}, 506 {AUDIO_DEVICE_IN_LOOPBACK, "LOOPBACK"}, 507 {AUDIO_DEVICE_IN_IP, "IP"}, 508 {AUDIO_DEVICE_IN_BUS, "BUS"}, 509 {AUDIO_DEVICE_NONE, "NONE"}, // must be last 510 }; 511 String8 result; 512 audio_devices_t allDevices = AUDIO_DEVICE_NONE; 513 const mapping *entry; 514 if (devices & AUDIO_DEVICE_BIT_IN) { 515 devices &= ~AUDIO_DEVICE_BIT_IN; 516 entry = mappingsIn; 517 } else { 518 entry = mappingsOut; 519 } 520 for ( ; entry->mDevices != AUDIO_DEVICE_NONE; entry++) { 521 allDevices = (audio_devices_t) (allDevices | entry->mDevices); 522 if (devices & entry->mDevices) { 523 if (!result.isEmpty()) { 524 result.append("|"); 525 } 526 result.append(entry->mString); 527 } 528 } 529 if (devices & ~allDevices) { 530 if (!result.isEmpty()) { 531 result.append("|"); 532 } 533 result.appendFormat("0x%X", devices & ~allDevices); 534 } 535 if (result.isEmpty()) { 536 result.append(entry->mString); 537 } 538 return result; 539} 540 541String8 inputFlagsToString(audio_input_flags_t flags) 542{ 543 static const struct mapping { 544 audio_input_flags_t mFlag; 545 const char * mString; 546 } mappings[] = { 547 {AUDIO_INPUT_FLAG_FAST, "FAST"}, 548 {AUDIO_INPUT_FLAG_HW_HOTWORD, "HW_HOTWORD"}, 549 {AUDIO_INPUT_FLAG_RAW, "RAW"}, 550 {AUDIO_INPUT_FLAG_SYNC, "SYNC"}, 551 {AUDIO_INPUT_FLAG_NONE, "NONE"}, // must be last 552 }; 553 String8 result; 554 audio_input_flags_t allFlags = AUDIO_INPUT_FLAG_NONE; 555 const mapping *entry; 556 for (entry = mappings; entry->mFlag != AUDIO_INPUT_FLAG_NONE; entry++) { 557 allFlags = (audio_input_flags_t) (allFlags | entry->mFlag); 558 if (flags & entry->mFlag) { 559 if (!result.isEmpty()) { 560 result.append("|"); 561 } 562 result.append(entry->mString); 563 } 564 } 565 if (flags & ~allFlags) { 566 if (!result.isEmpty()) { 567 result.append("|"); 568 } 569 result.appendFormat("0x%X", flags & ~allFlags); 570 } 571 if (result.isEmpty()) { 572 result.append(entry->mString); 573 } 574 return result; 575} 576 577String8 outputFlagsToString(audio_output_flags_t flags) 578{ 579 static const struct mapping { 580 audio_output_flags_t mFlag; 581 const char * mString; 582 } mappings[] = { 583 {AUDIO_OUTPUT_FLAG_DIRECT, "DIRECT"}, 584 {AUDIO_OUTPUT_FLAG_PRIMARY, "PRIMARY"}, 585 {AUDIO_OUTPUT_FLAG_FAST, "FAST"}, 586 {AUDIO_OUTPUT_FLAG_DEEP_BUFFER, "DEEP_BUFFER"}, 587 {AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD,"COMPRESS_OFFLOAD"}, 588 {AUDIO_OUTPUT_FLAG_NON_BLOCKING, "NON_BLOCKING"}, 589 {AUDIO_OUTPUT_FLAG_HW_AV_SYNC, "HW_AV_SYNC"}, 590 {AUDIO_OUTPUT_FLAG_RAW, "RAW"}, 591 {AUDIO_OUTPUT_FLAG_SYNC, "SYNC"}, 592 {AUDIO_OUTPUT_FLAG_IEC958_NONAUDIO, "IEC958_NONAUDIO"}, 593 {AUDIO_OUTPUT_FLAG_NONE, "NONE"}, // must be last 594 }; 595 String8 result; 596 audio_output_flags_t allFlags = AUDIO_OUTPUT_FLAG_NONE; 597 const mapping *entry; 598 for (entry = mappings; entry->mFlag != AUDIO_OUTPUT_FLAG_NONE; entry++) { 599 allFlags = (audio_output_flags_t) (allFlags | entry->mFlag); 600 if (flags & entry->mFlag) { 601 if (!result.isEmpty()) { 602 result.append("|"); 603 } 604 result.append(entry->mString); 605 } 606 } 607 if (flags & ~allFlags) { 608 if (!result.isEmpty()) { 609 result.append("|"); 610 } 611 result.appendFormat("0x%X", flags & ~allFlags); 612 } 613 if (result.isEmpty()) { 614 result.append(entry->mString); 615 } 616 return result; 617} 618 619const char *sourceToString(audio_source_t source) 620{ 621 switch (source) { 622 case AUDIO_SOURCE_DEFAULT: return "default"; 623 case AUDIO_SOURCE_MIC: return "mic"; 624 case AUDIO_SOURCE_VOICE_UPLINK: return "voice uplink"; 625 case AUDIO_SOURCE_VOICE_DOWNLINK: return "voice downlink"; 626 case AUDIO_SOURCE_VOICE_CALL: return "voice call"; 627 case AUDIO_SOURCE_CAMCORDER: return "camcorder"; 628 case AUDIO_SOURCE_VOICE_RECOGNITION: return "voice recognition"; 629 case AUDIO_SOURCE_VOICE_COMMUNICATION: return "voice communication"; 630 case AUDIO_SOURCE_REMOTE_SUBMIX: return "remote submix"; 631 case AUDIO_SOURCE_UNPROCESSED: return "unprocessed"; 632 case AUDIO_SOURCE_FM_TUNER: return "FM tuner"; 633 case AUDIO_SOURCE_HOTWORD: return "hotword"; 634 default: return "unknown"; 635 } 636} 637 638AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, 639 audio_devices_t outDevice, audio_devices_t inDevice, type_t type, bool systemReady) 640 : Thread(false /*canCallJava*/), 641 mType(type), 642 mAudioFlinger(audioFlinger), 643 // mSampleRate, mFrameCount, mChannelMask, mChannelCount, mFrameSize, mFormat, mBufferSize 644 // are set by PlaybackThread::readOutputParameters_l() or 645 // RecordThread::readInputParameters_l() 646 //FIXME: mStandby should be true here. Is this some kind of hack? 647 mStandby(false), mOutDevice(outDevice), mInDevice(inDevice), 648 mPrevOutDevice(AUDIO_DEVICE_NONE), mPrevInDevice(AUDIO_DEVICE_NONE), 649 mAudioSource(AUDIO_SOURCE_DEFAULT), mId(id), 650 // mName will be set by concrete (non-virtual) subclass 651 mDeathRecipient(new PMDeathRecipient(this)), 652 mSystemReady(systemReady), 653 mNotifiedBatteryStart(false) 654{ 655 memset(&mPatch, 0, sizeof(struct audio_patch)); 656} 657 658AudioFlinger::ThreadBase::~ThreadBase() 659{ 660 // mConfigEvents should be empty, but just in case it isn't, free the memory it owns 661 mConfigEvents.clear(); 662 663 // do not lock the mutex in destructor 664 releaseWakeLock_l(); 665 if (mPowerManager != 0) { 666 sp<IBinder> binder = IInterface::asBinder(mPowerManager); 667 binder->unlinkToDeath(mDeathRecipient); 668 } 669} 670 671status_t AudioFlinger::ThreadBase::readyToRun() 672{ 673 status_t status = initCheck(); 674 if (status == NO_ERROR) { 675 ALOGI("AudioFlinger's thread %p ready to run", this); 676 } else { 677 ALOGE("No working audio driver found."); 678 } 679 return status; 680} 681 682void AudioFlinger::ThreadBase::exit() 683{ 684 ALOGV("ThreadBase::exit"); 685 // do any cleanup required for exit to succeed 686 preExit(); 687 { 688 // This lock prevents the following race in thread (uniprocessor for illustration): 689 // if (!exitPending()) { 690 // // context switch from here to exit() 691 // // exit() calls requestExit(), what exitPending() observes 692 // // exit() calls signal(), which is dropped since no waiters 693 // // context switch back from exit() to here 694 // mWaitWorkCV.wait(...); 695 // // now thread is hung 696 // } 697 AutoMutex lock(mLock); 698 requestExit(); 699 mWaitWorkCV.broadcast(); 700 } 701 // When Thread::requestExitAndWait is made virtual and this method is renamed to 702 // "virtual status_t requestExitAndWait()", replace by "return Thread::requestExitAndWait();" 703 requestExitAndWait(); 704} 705 706status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs) 707{ 708 ALOGV("ThreadBase::setParameters() %s", keyValuePairs.string()); 709 Mutex::Autolock _l(mLock); 710 711 return sendSetParameterConfigEvent_l(keyValuePairs); 712} 713 714// sendConfigEvent_l() must be called with ThreadBase::mLock held 715// Can temporarily release the lock if waiting for a reply from processConfigEvents_l(). 716status_t AudioFlinger::ThreadBase::sendConfigEvent_l(sp<ConfigEvent>& event) 717{ 718 status_t status = NO_ERROR; 719 720 if (event->mRequiresSystemReady && !mSystemReady) { 721 event->mWaitStatus = false; 722 mPendingConfigEvents.add(event); 723 return status; 724 } 725 mConfigEvents.add(event); 726 ALOGV("sendConfigEvent_l() num events %zu event %d", mConfigEvents.size(), event->mType); 727 mWaitWorkCV.signal(); 728 mLock.unlock(); 729 { 730 Mutex::Autolock _l(event->mLock); 731 while (event->mWaitStatus) { 732 if (event->mCond.waitRelative(event->mLock, kConfigEventTimeoutNs) != NO_ERROR) { 733 event->mStatus = TIMED_OUT; 734 event->mWaitStatus = false; 735 } 736 } 737 status = event->mStatus; 738 } 739 mLock.lock(); 740 return status; 741} 742 743void AudioFlinger::ThreadBase::sendIoConfigEvent(audio_io_config_event event, pid_t pid) 744{ 745 Mutex::Autolock _l(mLock); 746 sendIoConfigEvent_l(event, pid); 747} 748 749// sendIoConfigEvent_l() must be called with ThreadBase::mLock held 750void AudioFlinger::ThreadBase::sendIoConfigEvent_l(audio_io_config_event event, pid_t pid) 751{ 752 sp<ConfigEvent> configEvent = (ConfigEvent *)new IoConfigEvent(event, pid); 753 sendConfigEvent_l(configEvent); 754} 755 756void AudioFlinger::ThreadBase::sendPrioConfigEvent(pid_t pid, pid_t tid, int32_t prio) 757{ 758 Mutex::Autolock _l(mLock); 759 sendPrioConfigEvent_l(pid, tid, prio); 760} 761 762// sendPrioConfigEvent_l() must be called with ThreadBase::mLock held 763void AudioFlinger::ThreadBase::sendPrioConfigEvent_l(pid_t pid, pid_t tid, int32_t prio) 764{ 765 sp<ConfigEvent> configEvent = (ConfigEvent *)new PrioConfigEvent(pid, tid, prio); 766 sendConfigEvent_l(configEvent); 767} 768 769// sendSetParameterConfigEvent_l() must be called with ThreadBase::mLock held 770status_t AudioFlinger::ThreadBase::sendSetParameterConfigEvent_l(const String8& keyValuePair) 771{ 772 sp<ConfigEvent> configEvent; 773 AudioParameter param(keyValuePair); 774 int value; 775 if (param.getInt(String8(AUDIO_PARAMETER_MONO_OUTPUT), value) == NO_ERROR) { 776 setMasterMono_l(value != 0); 777 if (param.size() == 1) { 778 return NO_ERROR; // should be a solo parameter - we don't pass down 779 } 780 param.remove(String8(AUDIO_PARAMETER_MONO_OUTPUT)); 781 configEvent = new SetParameterConfigEvent(param.toString()); 782 } else { 783 configEvent = new SetParameterConfigEvent(keyValuePair); 784 } 785 return sendConfigEvent_l(configEvent); 786} 787 788status_t AudioFlinger::ThreadBase::sendCreateAudioPatchConfigEvent( 789 const struct audio_patch *patch, 790 audio_patch_handle_t *handle) 791{ 792 Mutex::Autolock _l(mLock); 793 sp<ConfigEvent> configEvent = (ConfigEvent *)new CreateAudioPatchConfigEvent(*patch, *handle); 794 status_t status = sendConfigEvent_l(configEvent); 795 if (status == NO_ERROR) { 796 CreateAudioPatchConfigEventData *data = 797 (CreateAudioPatchConfigEventData *)configEvent->mData.get(); 798 *handle = data->mHandle; 799 } 800 return status; 801} 802 803status_t AudioFlinger::ThreadBase::sendReleaseAudioPatchConfigEvent( 804 const audio_patch_handle_t handle) 805{ 806 Mutex::Autolock _l(mLock); 807 sp<ConfigEvent> configEvent = (ConfigEvent *)new ReleaseAudioPatchConfigEvent(handle); 808 return sendConfigEvent_l(configEvent); 809} 810 811 812// post condition: mConfigEvents.isEmpty() 813void AudioFlinger::ThreadBase::processConfigEvents_l() 814{ 815 bool configChanged = false; 816 817 while (!mConfigEvents.isEmpty()) { 818 ALOGV("processConfigEvents_l() remaining events %zu", mConfigEvents.size()); 819 sp<ConfigEvent> event = mConfigEvents[0]; 820 mConfigEvents.removeAt(0); 821 switch (event->mType) { 822 case CFG_EVENT_PRIO: { 823 PrioConfigEventData *data = (PrioConfigEventData *)event->mData.get(); 824 // FIXME Need to understand why this has to be done asynchronously 825 int err = requestPriority(data->mPid, data->mTid, data->mPrio, 826 true /*asynchronous*/); 827 if (err != 0) { 828 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 829 data->mPrio, data->mPid, data->mTid, err); 830 } 831 } break; 832 case CFG_EVENT_IO: { 833 IoConfigEventData *data = (IoConfigEventData *)event->mData.get(); 834 ioConfigChanged(data->mEvent, data->mPid); 835 } break; 836 case CFG_EVENT_SET_PARAMETER: { 837 SetParameterConfigEventData *data = (SetParameterConfigEventData *)event->mData.get(); 838 if (checkForNewParameter_l(data->mKeyValuePairs, event->mStatus)) { 839 configChanged = true; 840 } 841 } break; 842 case CFG_EVENT_CREATE_AUDIO_PATCH: { 843 CreateAudioPatchConfigEventData *data = 844 (CreateAudioPatchConfigEventData *)event->mData.get(); 845 event->mStatus = createAudioPatch_l(&data->mPatch, &data->mHandle); 846 } break; 847 case CFG_EVENT_RELEASE_AUDIO_PATCH: { 848 ReleaseAudioPatchConfigEventData *data = 849 (ReleaseAudioPatchConfigEventData *)event->mData.get(); 850 event->mStatus = releaseAudioPatch_l(data->mHandle); 851 } break; 852 default: 853 ALOG_ASSERT(false, "processConfigEvents_l() unknown event type %d", event->mType); 854 break; 855 } 856 { 857 Mutex::Autolock _l(event->mLock); 858 if (event->mWaitStatus) { 859 event->mWaitStatus = false; 860 event->mCond.signal(); 861 } 862 } 863 ALOGV_IF(mConfigEvents.isEmpty(), "processConfigEvents_l() DONE thread %p", this); 864 } 865 866 if (configChanged) { 867 cacheParameters_l(); 868 } 869} 870 871String8 channelMaskToString(audio_channel_mask_t mask, bool output) { 872 String8 s; 873 const audio_channel_representation_t representation = 874 audio_channel_mask_get_representation(mask); 875 876 switch (representation) { 877 case AUDIO_CHANNEL_REPRESENTATION_POSITION: { 878 if (output) { 879 if (mask & AUDIO_CHANNEL_OUT_FRONT_LEFT) s.append("front-left, "); 880 if (mask & AUDIO_CHANNEL_OUT_FRONT_RIGHT) s.append("front-right, "); 881 if (mask & AUDIO_CHANNEL_OUT_FRONT_CENTER) s.append("front-center, "); 882 if (mask & AUDIO_CHANNEL_OUT_LOW_FREQUENCY) s.append("low freq, "); 883 if (mask & AUDIO_CHANNEL_OUT_BACK_LEFT) s.append("back-left, "); 884 if (mask & AUDIO_CHANNEL_OUT_BACK_RIGHT) s.append("back-right, "); 885 if (mask & AUDIO_CHANNEL_OUT_FRONT_LEFT_OF_CENTER) s.append("front-left-of-center, "); 886 if (mask & AUDIO_CHANNEL_OUT_FRONT_RIGHT_OF_CENTER) s.append("front-right-of-center, "); 887 if (mask & AUDIO_CHANNEL_OUT_BACK_CENTER) s.append("back-center, "); 888 if (mask & AUDIO_CHANNEL_OUT_SIDE_LEFT) s.append("side-left, "); 889 if (mask & AUDIO_CHANNEL_OUT_SIDE_RIGHT) s.append("side-right, "); 890 if (mask & AUDIO_CHANNEL_OUT_TOP_CENTER) s.append("top-center ,"); 891 if (mask & AUDIO_CHANNEL_OUT_TOP_FRONT_LEFT) s.append("top-front-left, "); 892 if (mask & AUDIO_CHANNEL_OUT_TOP_FRONT_CENTER) s.append("top-front-center, "); 893 if (mask & AUDIO_CHANNEL_OUT_TOP_FRONT_RIGHT) s.append("top-front-right, "); 894 if (mask & AUDIO_CHANNEL_OUT_TOP_BACK_LEFT) s.append("top-back-left, "); 895 if (mask & AUDIO_CHANNEL_OUT_TOP_BACK_CENTER) s.append("top-back-center, " ); 896 if (mask & AUDIO_CHANNEL_OUT_TOP_BACK_RIGHT) s.append("top-back-right, " ); 897 if (mask & ~AUDIO_CHANNEL_OUT_ALL) s.append("unknown, "); 898 } else { 899 if (mask & AUDIO_CHANNEL_IN_LEFT) s.append("left, "); 900 if (mask & AUDIO_CHANNEL_IN_RIGHT) s.append("right, "); 901 if (mask & AUDIO_CHANNEL_IN_FRONT) s.append("front, "); 902 if (mask & AUDIO_CHANNEL_IN_BACK) s.append("back, "); 903 if (mask & AUDIO_CHANNEL_IN_LEFT_PROCESSED) s.append("left-processed, "); 904 if (mask & AUDIO_CHANNEL_IN_RIGHT_PROCESSED) s.append("right-processed, "); 905 if (mask & AUDIO_CHANNEL_IN_FRONT_PROCESSED) s.append("front-processed, "); 906 if (mask & AUDIO_CHANNEL_IN_BACK_PROCESSED) s.append("back-processed, "); 907 if (mask & AUDIO_CHANNEL_IN_PRESSURE) s.append("pressure, "); 908 if (mask & AUDIO_CHANNEL_IN_X_AXIS) s.append("X, "); 909 if (mask & AUDIO_CHANNEL_IN_Y_AXIS) s.append("Y, "); 910 if (mask & AUDIO_CHANNEL_IN_Z_AXIS) s.append("Z, "); 911 if (mask & AUDIO_CHANNEL_IN_VOICE_UPLINK) s.append("voice-uplink, "); 912 if (mask & AUDIO_CHANNEL_IN_VOICE_DNLINK) s.append("voice-dnlink, "); 913 if (mask & ~AUDIO_CHANNEL_IN_ALL) s.append("unknown, "); 914 } 915 const int len = s.length(); 916 if (len > 2) { 917 (void) s.lockBuffer(len); // needed? 918 s.unlockBuffer(len - 2); // remove trailing ", " 919 } 920 return s; 921 } 922 case AUDIO_CHANNEL_REPRESENTATION_INDEX: 923 s.appendFormat("index mask, bits:%#x", audio_channel_mask_get_bits(mask)); 924 return s; 925 default: 926 s.appendFormat("unknown mask, representation:%d bits:%#x", 927 representation, audio_channel_mask_get_bits(mask)); 928 return s; 929 } 930} 931 932void AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args __unused) 933{ 934 const size_t SIZE = 256; 935 char buffer[SIZE]; 936 String8 result; 937 938 bool locked = AudioFlinger::dumpTryLock(mLock); 939 if (!locked) { 940 dprintf(fd, "thread %p may be deadlocked\n", this); 941 } 942 943 dprintf(fd, " Thread name: %s\n", mThreadName); 944 dprintf(fd, " I/O handle: %d\n", mId); 945 dprintf(fd, " TID: %d\n", getTid()); 946 dprintf(fd, " Standby: %s\n", mStandby ? "yes" : "no"); 947 dprintf(fd, " Sample rate: %u Hz\n", mSampleRate); 948 dprintf(fd, " HAL frame count: %zu\n", mFrameCount); 949 dprintf(fd, " HAL format: 0x%x (%s)\n", mHALFormat, formatToString(mHALFormat)); 950 dprintf(fd, " HAL buffer size: %zu bytes\n", mBufferSize); 951 dprintf(fd, " Channel count: %u\n", mChannelCount); 952 dprintf(fd, " Channel mask: 0x%08x (%s)\n", mChannelMask, 953 channelMaskToString(mChannelMask, mType != RECORD).string()); 954 dprintf(fd, " Processing format: 0x%x (%s)\n", mFormat, formatToString(mFormat)); 955 dprintf(fd, " Processing frame size: %zu bytes\n", mFrameSize); 956 dprintf(fd, " Pending config events:"); 957 size_t numConfig = mConfigEvents.size(); 958 if (numConfig) { 959 for (size_t i = 0; i < numConfig; i++) { 960 mConfigEvents[i]->dump(buffer, SIZE); 961 dprintf(fd, "\n %s", buffer); 962 } 963 dprintf(fd, "\n"); 964 } else { 965 dprintf(fd, " none\n"); 966 } 967 dprintf(fd, " Output device: %#x (%s)\n", mOutDevice, devicesToString(mOutDevice).string()); 968 dprintf(fd, " Input device: %#x (%s)\n", mInDevice, devicesToString(mInDevice).string()); 969 dprintf(fd, " Audio source: %d (%s)\n", mAudioSource, sourceToString(mAudioSource)); 970 971 if (locked) { 972 mLock.unlock(); 973 } 974} 975 976void AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args) 977{ 978 const size_t SIZE = 256; 979 char buffer[SIZE]; 980 String8 result; 981 982 size_t numEffectChains = mEffectChains.size(); 983 snprintf(buffer, SIZE, " %zu Effect Chains\n", numEffectChains); 984 write(fd, buffer, strlen(buffer)); 985 986 for (size_t i = 0; i < numEffectChains; ++i) { 987 sp<EffectChain> chain = mEffectChains[i]; 988 if (chain != 0) { 989 chain->dump(fd, args); 990 } 991 } 992} 993 994void AudioFlinger::ThreadBase::acquireWakeLock(int uid) 995{ 996 Mutex::Autolock _l(mLock); 997 acquireWakeLock_l(uid); 998} 999 1000String16 AudioFlinger::ThreadBase::getWakeLockTag() 1001{ 1002 switch (mType) { 1003 case MIXER: 1004 return String16("AudioMix"); 1005 case DIRECT: 1006 return String16("AudioDirectOut"); 1007 case DUPLICATING: 1008 return String16("AudioDup"); 1009 case RECORD: 1010 return String16("AudioIn"); 1011 case OFFLOAD: 1012 return String16("AudioOffload"); 1013 default: 1014 ALOG_ASSERT(false); 1015 return String16("AudioUnknown"); 1016 } 1017} 1018 1019void AudioFlinger::ThreadBase::acquireWakeLock_l(int uid) 1020{ 1021 getPowerManager_l(); 1022 if (mPowerManager != 0) { 1023 sp<IBinder> binder = new BBinder(); 1024 status_t status; 1025 if (uid >= 0) { 1026 status = mPowerManager->acquireWakeLockWithUid(POWERMANAGER_PARTIAL_WAKE_LOCK, 1027 binder, 1028 getWakeLockTag(), 1029 String16("audioserver"), 1030 uid, 1031 true /* FIXME force oneway contrary to .aidl */); 1032 } else { 1033 status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK, 1034 binder, 1035 getWakeLockTag(), 1036 String16("audioserver"), 1037 true /* FIXME force oneway contrary to .aidl */); 1038 } 1039 if (status == NO_ERROR) { 1040 mWakeLockToken = binder; 1041 } 1042 ALOGV("acquireWakeLock_l() %s status %d", mThreadName, status); 1043 } 1044 1045 if (!mNotifiedBatteryStart) { 1046 BatteryNotifier::getInstance().noteStartAudio(); 1047 mNotifiedBatteryStart = true; 1048 } 1049 gBoottime.acquire(mWakeLockToken); 1050 mTimestamp.mTimebaseOffset[ExtendedTimestamp::TIMEBASE_BOOTTIME] = 1051 gBoottime.getBoottimeOffset(); 1052} 1053 1054void AudioFlinger::ThreadBase::releaseWakeLock() 1055{ 1056 Mutex::Autolock _l(mLock); 1057 releaseWakeLock_l(); 1058} 1059 1060void AudioFlinger::ThreadBase::releaseWakeLock_l() 1061{ 1062 gBoottime.release(mWakeLockToken); 1063 if (mWakeLockToken != 0) { 1064 ALOGV("releaseWakeLock_l() %s", mThreadName); 1065 if (mPowerManager != 0) { 1066 mPowerManager->releaseWakeLock(mWakeLockToken, 0, 1067 true /* FIXME force oneway contrary to .aidl */); 1068 } 1069 mWakeLockToken.clear(); 1070 } 1071 1072 if (mNotifiedBatteryStart) { 1073 BatteryNotifier::getInstance().noteStopAudio(); 1074 mNotifiedBatteryStart = false; 1075 } 1076} 1077 1078void AudioFlinger::ThreadBase::updateWakeLockUids(const SortedVector<int> &uids) { 1079 Mutex::Autolock _l(mLock); 1080 updateWakeLockUids_l(uids); 1081} 1082 1083void AudioFlinger::ThreadBase::getPowerManager_l() { 1084 if (mSystemReady && mPowerManager == 0) { 1085 // use checkService() to avoid blocking if power service is not up yet 1086 sp<IBinder> binder = 1087 defaultServiceManager()->checkService(String16("power")); 1088 if (binder == 0) { 1089 ALOGW("Thread %s cannot connect to the power manager service", mThreadName); 1090 } else { 1091 mPowerManager = interface_cast<IPowerManager>(binder); 1092 binder->linkToDeath(mDeathRecipient); 1093 } 1094 } 1095} 1096 1097void AudioFlinger::ThreadBase::updateWakeLockUids_l(const SortedVector<int> &uids) { 1098 getPowerManager_l(); 1099 if (mWakeLockToken == NULL) { // token may be NULL if AudioFlinger::systemReady() not called. 1100 if (mSystemReady) { 1101 ALOGE("no wake lock to update, but system ready!"); 1102 } else { 1103 ALOGW("no wake lock to update, system not ready yet"); 1104 } 1105 return; 1106 } 1107 if (mPowerManager != 0) { 1108 sp<IBinder> binder = new BBinder(); 1109 status_t status; 1110 status = mPowerManager->updateWakeLockUids(mWakeLockToken, uids.size(), uids.array(), 1111 true /* FIXME force oneway contrary to .aidl */); 1112 ALOGV("updateWakeLockUids_l() %s status %d", mThreadName, status); 1113 } 1114} 1115 1116void AudioFlinger::ThreadBase::clearPowerManager() 1117{ 1118 Mutex::Autolock _l(mLock); 1119 releaseWakeLock_l(); 1120 mPowerManager.clear(); 1121} 1122 1123void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who __unused) 1124{ 1125 sp<ThreadBase> thread = mThread.promote(); 1126 if (thread != 0) { 1127 thread->clearPowerManager(); 1128 } 1129 ALOGW("power manager service died !!!"); 1130} 1131 1132void AudioFlinger::ThreadBase::setEffectSuspended( 1133 const effect_uuid_t *type, bool suspend, audio_session_t sessionId) 1134{ 1135 Mutex::Autolock _l(mLock); 1136 setEffectSuspended_l(type, suspend, sessionId); 1137} 1138 1139void AudioFlinger::ThreadBase::setEffectSuspended_l( 1140 const effect_uuid_t *type, bool suspend, audio_session_t sessionId) 1141{ 1142 sp<EffectChain> chain = getEffectChain_l(sessionId); 1143 if (chain != 0) { 1144 if (type != NULL) { 1145 chain->setEffectSuspended_l(type, suspend); 1146 } else { 1147 chain->setEffectSuspendedAll_l(suspend); 1148 } 1149 } 1150 1151 updateSuspendedSessions_l(type, suspend, sessionId); 1152} 1153 1154void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain) 1155{ 1156 ssize_t index = mSuspendedSessions.indexOfKey(chain->sessionId()); 1157 if (index < 0) { 1158 return; 1159 } 1160 1161 const KeyedVector <int, sp<SuspendedSessionDesc> >& sessionEffects = 1162 mSuspendedSessions.valueAt(index); 1163 1164 for (size_t i = 0; i < sessionEffects.size(); i++) { 1165 sp<SuspendedSessionDesc> desc = sessionEffects.valueAt(i); 1166 for (int j = 0; j < desc->mRefCount; j++) { 1167 if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) { 1168 chain->setEffectSuspendedAll_l(true); 1169 } else { 1170 ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x", 1171 desc->mType.timeLow); 1172 chain->setEffectSuspended_l(&desc->mType, true); 1173 } 1174 } 1175 } 1176} 1177 1178void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type, 1179 bool suspend, 1180 audio_session_t sessionId) 1181{ 1182 ssize_t index = mSuspendedSessions.indexOfKey(sessionId); 1183 1184 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects; 1185 1186 if (suspend) { 1187 if (index >= 0) { 1188 sessionEffects = mSuspendedSessions.valueAt(index); 1189 } else { 1190 mSuspendedSessions.add(sessionId, sessionEffects); 1191 } 1192 } else { 1193 if (index < 0) { 1194 return; 1195 } 1196 sessionEffects = mSuspendedSessions.valueAt(index); 1197 } 1198 1199 1200 int key = EffectChain::kKeyForSuspendAll; 1201 if (type != NULL) { 1202 key = type->timeLow; 1203 } 1204 index = sessionEffects.indexOfKey(key); 1205 1206 sp<SuspendedSessionDesc> desc; 1207 if (suspend) { 1208 if (index >= 0) { 1209 desc = sessionEffects.valueAt(index); 1210 } else { 1211 desc = new SuspendedSessionDesc(); 1212 if (type != NULL) { 1213 desc->mType = *type; 1214 } 1215 sessionEffects.add(key, desc); 1216 ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key); 1217 } 1218 desc->mRefCount++; 1219 } else { 1220 if (index < 0) { 1221 return; 1222 } 1223 desc = sessionEffects.valueAt(index); 1224 if (--desc->mRefCount == 0) { 1225 ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key); 1226 sessionEffects.removeItemsAt(index); 1227 if (sessionEffects.isEmpty()) { 1228 ALOGV("updateSuspendedSessions_l() restore removing session %d", 1229 sessionId); 1230 mSuspendedSessions.removeItem(sessionId); 1231 } 1232 } 1233 } 1234 if (!sessionEffects.isEmpty()) { 1235 mSuspendedSessions.replaceValueFor(sessionId, sessionEffects); 1236 } 1237} 1238 1239void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 1240 bool enabled, 1241 audio_session_t sessionId) 1242{ 1243 Mutex::Autolock _l(mLock); 1244 checkSuspendOnEffectEnabled_l(effect, enabled, sessionId); 1245} 1246 1247void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 1248 bool enabled, 1249 audio_session_t sessionId) 1250{ 1251 if (mType != RECORD) { 1252 // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on 1253 // another session. This gives the priority to well behaved effect control panels 1254 // and applications not using global effects. 1255 // Enabling post processing in AUDIO_SESSION_OUTPUT_STAGE session does not affect 1256 // global effects 1257 if ((sessionId != AUDIO_SESSION_OUTPUT_MIX) && (sessionId != AUDIO_SESSION_OUTPUT_STAGE)) { 1258 setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX); 1259 } 1260 } 1261 1262 sp<EffectChain> chain = getEffectChain_l(sessionId); 1263 if (chain != 0) { 1264 chain->checkSuspendOnEffectEnabled(effect, enabled); 1265 } 1266} 1267 1268// ThreadBase::createEffect_l() must be called with AudioFlinger::mLock held 1269sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 1270 const sp<AudioFlinger::Client>& client, 1271 const sp<IEffectClient>& effectClient, 1272 int32_t priority, 1273 audio_session_t sessionId, 1274 effect_descriptor_t *desc, 1275 int *enabled, 1276 status_t *status) 1277{ 1278 sp<EffectModule> effect; 1279 sp<EffectHandle> handle; 1280 status_t lStatus; 1281 sp<EffectChain> chain; 1282 bool chainCreated = false; 1283 bool effectCreated = false; 1284 bool effectRegistered = false; 1285 1286 lStatus = initCheck(); 1287 if (lStatus != NO_ERROR) { 1288 ALOGW("createEffect_l() Audio driver not initialized."); 1289 goto Exit; 1290 } 1291 1292 // Reject any effect on Direct output threads for now, since the format of 1293 // mSinkBuffer is not guaranteed to be compatible with effect processing (PCM 16 stereo). 1294 if (mType == DIRECT) { 1295 ALOGW("createEffect_l() Cannot add effect %s on Direct output type thread %s", 1296 desc->name, mThreadName); 1297 lStatus = BAD_VALUE; 1298 goto Exit; 1299 } 1300 1301 // Reject any effect on mixer or duplicating multichannel sinks. 1302 // TODO: fix both format and multichannel issues with effects. 1303 if ((mType == MIXER || mType == DUPLICATING) && mChannelCount != FCC_2) { 1304 ALOGW("createEffect_l() Cannot add effect %s for multichannel(%d) %s threads", 1305 desc->name, mChannelCount, mType == MIXER ? "MIXER" : "DUPLICATING"); 1306 lStatus = BAD_VALUE; 1307 goto Exit; 1308 } 1309 1310 // Allow global effects only on offloaded and mixer threads 1311 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 1312 switch (mType) { 1313 case MIXER: 1314 case OFFLOAD: 1315 break; 1316 case DIRECT: 1317 case DUPLICATING: 1318 case RECORD: 1319 default: 1320 ALOGW("createEffect_l() Cannot add global effect %s on thread %s", 1321 desc->name, mThreadName); 1322 lStatus = BAD_VALUE; 1323 goto Exit; 1324 } 1325 } 1326 1327 // Only Pre processor effects are allowed on input threads and only on input threads 1328 if ((mType == RECORD) != ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { 1329 ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d", 1330 desc->name, desc->flags, mType); 1331 lStatus = BAD_VALUE; 1332 goto Exit; 1333 } 1334 1335 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 1336 1337 { // scope for mLock 1338 Mutex::Autolock _l(mLock); 1339 1340 // check for existing effect chain with the requested audio session 1341 chain = getEffectChain_l(sessionId); 1342 if (chain == 0) { 1343 // create a new chain for this session 1344 ALOGV("createEffect_l() new effect chain for session %d", sessionId); 1345 chain = new EffectChain(this, sessionId); 1346 addEffectChain_l(chain); 1347 chain->setStrategy(getStrategyForSession_l(sessionId)); 1348 chainCreated = true; 1349 } else { 1350 effect = chain->getEffectFromDesc_l(desc); 1351 } 1352 1353 ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get()); 1354 1355 if (effect == 0) { 1356 audio_unique_id_t id = mAudioFlinger->nextUniqueId(AUDIO_UNIQUE_ID_USE_EFFECT); 1357 // Check CPU and memory usage 1358 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 1359 if (lStatus != NO_ERROR) { 1360 goto Exit; 1361 } 1362 effectRegistered = true; 1363 // create a new effect module if none present in the chain 1364 effect = new EffectModule(this, chain, desc, id, sessionId); 1365 lStatus = effect->status(); 1366 if (lStatus != NO_ERROR) { 1367 goto Exit; 1368 } 1369 effect->setOffloaded(mType == OFFLOAD, mId); 1370 1371 lStatus = chain->addEffect_l(effect); 1372 if (lStatus != NO_ERROR) { 1373 goto Exit; 1374 } 1375 effectCreated = true; 1376 1377 effect->setDevice(mOutDevice); 1378 effect->setDevice(mInDevice); 1379 effect->setMode(mAudioFlinger->getMode()); 1380 effect->setAudioSource(mAudioSource); 1381 } 1382 // create effect handle and connect it to effect module 1383 handle = new EffectHandle(effect, client, effectClient, priority); 1384 lStatus = handle->initCheck(); 1385 if (lStatus == OK) { 1386 lStatus = effect->addHandle(handle.get()); 1387 } 1388 if (enabled != NULL) { 1389 *enabled = (int)effect->isEnabled(); 1390 } 1391 } 1392 1393Exit: 1394 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 1395 Mutex::Autolock _l(mLock); 1396 if (effectCreated) { 1397 chain->removeEffect_l(effect); 1398 } 1399 if (effectRegistered) { 1400 AudioSystem::unregisterEffect(effect->id()); 1401 } 1402 if (chainCreated) { 1403 removeEffectChain_l(chain); 1404 } 1405 handle.clear(); 1406 } 1407 1408 *status = lStatus; 1409 return handle; 1410} 1411 1412sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect(audio_session_t sessionId, 1413 int effectId) 1414{ 1415 Mutex::Autolock _l(mLock); 1416 return getEffect_l(sessionId, effectId); 1417} 1418 1419sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(audio_session_t sessionId, 1420 int effectId) 1421{ 1422 sp<EffectChain> chain = getEffectChain_l(sessionId); 1423 return chain != 0 ? chain->getEffectFromId_l(effectId) : 0; 1424} 1425 1426// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 1427// PlaybackThread::mLock held 1428status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 1429{ 1430 // check for existing effect chain with the requested audio session 1431 audio_session_t sessionId = effect->sessionId(); 1432 sp<EffectChain> chain = getEffectChain_l(sessionId); 1433 bool chainCreated = false; 1434 1435 ALOGD_IF((mType == OFFLOAD) && !effect->isOffloadable(), 1436 "addEffect_l() on offloaded thread %p: effect %s does not support offload flags %x", 1437 this, effect->desc().name, effect->desc().flags); 1438 1439 if (chain == 0) { 1440 // create a new chain for this session 1441 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 1442 chain = new EffectChain(this, sessionId); 1443 addEffectChain_l(chain); 1444 chain->setStrategy(getStrategyForSession_l(sessionId)); 1445 chainCreated = true; 1446 } 1447 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 1448 1449 if (chain->getEffectFromId_l(effect->id()) != 0) { 1450 ALOGW("addEffect_l() %p effect %s already present in chain %p", 1451 this, effect->desc().name, chain.get()); 1452 return BAD_VALUE; 1453 } 1454 1455 effect->setOffloaded(mType == OFFLOAD, mId); 1456 1457 status_t status = chain->addEffect_l(effect); 1458 if (status != NO_ERROR) { 1459 if (chainCreated) { 1460 removeEffectChain_l(chain); 1461 } 1462 return status; 1463 } 1464 1465 effect->setDevice(mOutDevice); 1466 effect->setDevice(mInDevice); 1467 effect->setMode(mAudioFlinger->getMode()); 1468 effect->setAudioSource(mAudioSource); 1469 return NO_ERROR; 1470} 1471 1472void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 1473 1474 ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 1475 effect_descriptor_t desc = effect->desc(); 1476 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 1477 detachAuxEffect_l(effect->id()); 1478 } 1479 1480 sp<EffectChain> chain = effect->chain().promote(); 1481 if (chain != 0) { 1482 // remove effect chain if removing last effect 1483 if (chain->removeEffect_l(effect) == 0) { 1484 removeEffectChain_l(chain); 1485 } 1486 } else { 1487 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 1488 } 1489} 1490 1491void AudioFlinger::ThreadBase::lockEffectChains_l( 1492 Vector< sp<AudioFlinger::EffectChain> >& effectChains) 1493{ 1494 effectChains = mEffectChains; 1495 for (size_t i = 0; i < mEffectChains.size(); i++) { 1496 mEffectChains[i]->lock(); 1497 } 1498} 1499 1500void AudioFlinger::ThreadBase::unlockEffectChains( 1501 const Vector< sp<AudioFlinger::EffectChain> >& effectChains) 1502{ 1503 for (size_t i = 0; i < effectChains.size(); i++) { 1504 effectChains[i]->unlock(); 1505 } 1506} 1507 1508sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(audio_session_t sessionId) 1509{ 1510 Mutex::Autolock _l(mLock); 1511 return getEffectChain_l(sessionId); 1512} 1513 1514sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(audio_session_t sessionId) 1515 const 1516{ 1517 size_t size = mEffectChains.size(); 1518 for (size_t i = 0; i < size; i++) { 1519 if (mEffectChains[i]->sessionId() == sessionId) { 1520 return mEffectChains[i]; 1521 } 1522 } 1523 return 0; 1524} 1525 1526void AudioFlinger::ThreadBase::setMode(audio_mode_t mode) 1527{ 1528 Mutex::Autolock _l(mLock); 1529 size_t size = mEffectChains.size(); 1530 for (size_t i = 0; i < size; i++) { 1531 mEffectChains[i]->setMode_l(mode); 1532 } 1533} 1534 1535void AudioFlinger::ThreadBase::getAudioPortConfig(struct audio_port_config *config) 1536{ 1537 config->type = AUDIO_PORT_TYPE_MIX; 1538 config->ext.mix.handle = mId; 1539 config->sample_rate = mSampleRate; 1540 config->format = mFormat; 1541 config->channel_mask = mChannelMask; 1542 config->config_mask = AUDIO_PORT_CONFIG_SAMPLE_RATE|AUDIO_PORT_CONFIG_CHANNEL_MASK| 1543 AUDIO_PORT_CONFIG_FORMAT; 1544} 1545 1546void AudioFlinger::ThreadBase::systemReady() 1547{ 1548 Mutex::Autolock _l(mLock); 1549 if (mSystemReady) { 1550 return; 1551 } 1552 mSystemReady = true; 1553 1554 for (size_t i = 0; i < mPendingConfigEvents.size(); i++) { 1555 sendConfigEvent_l(mPendingConfigEvents.editItemAt(i)); 1556 } 1557 mPendingConfigEvents.clear(); 1558} 1559 1560 1561// ---------------------------------------------------------------------------- 1562// Playback 1563// ---------------------------------------------------------------------------- 1564 1565AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, 1566 AudioStreamOut* output, 1567 audio_io_handle_t id, 1568 audio_devices_t device, 1569 type_t type, 1570 bool systemReady, 1571 uint32_t bitRate) 1572 : ThreadBase(audioFlinger, id, device, AUDIO_DEVICE_NONE, type, systemReady), 1573 mNormalFrameCount(0), mSinkBuffer(NULL), 1574 mMixerBufferEnabled(AudioFlinger::kEnableExtendedPrecision), 1575 mMixerBuffer(NULL), 1576 mMixerBufferSize(0), 1577 mMixerBufferFormat(AUDIO_FORMAT_INVALID), 1578 mMixerBufferValid(false), 1579 mEffectBufferEnabled(AudioFlinger::kEnableExtendedPrecision), 1580 mEffectBuffer(NULL), 1581 mEffectBufferSize(0), 1582 mEffectBufferFormat(AUDIO_FORMAT_INVALID), 1583 mEffectBufferValid(false), 1584 mSuspended(0), mBytesWritten(0), 1585 mFramesWritten(0), 1586 mActiveTracksGeneration(0), 1587 // mStreamTypes[] initialized in constructor body 1588 mOutput(output), 1589 mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false), 1590 mMixerStatus(MIXER_IDLE), 1591 mMixerStatusIgnoringFastTracks(MIXER_IDLE), 1592 mStandbyDelayNs(AudioFlinger::mStandbyTimeInNsecs), 1593 mBytesRemaining(0), 1594 mCurrentWriteLength(0), 1595 mUseAsyncWrite(false), 1596 mWriteAckSequence(0), 1597 mDrainSequence(0), 1598 mSignalPending(false), 1599 mScreenState(AudioFlinger::mScreenState), 1600 // index 0 is reserved for normal mixer's submix 1601 mFastTrackAvailMask(((1 << FastMixerState::sMaxFastTracks) - 1) & ~1), 1602 mHwSupportsPause(false), mHwPaused(false), mFlushPending(false) 1603{ 1604 snprintf(mThreadName, kThreadNameLength, "AudioOut_%X", id); 1605 mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mThreadName); 1606 1607 // Assumes constructor is called by AudioFlinger with it's mLock held, but 1608 // it would be safer to explicitly pass initial masterVolume/masterMute as 1609 // parameter. 1610 // 1611 // If the HAL we are using has support for master volume or master mute, 1612 // then do not attenuate or mute during mixing (just leave the volume at 1.0 1613 // and the mute set to false). 1614 mMasterVolume = audioFlinger->masterVolume_l(); 1615 mMasterMute = audioFlinger->masterMute_l(); 1616 if (mOutput && mOutput->audioHwDev) { 1617 if (mOutput->audioHwDev->canSetMasterVolume()) { 1618 mMasterVolume = 1.0; 1619 } 1620 1621 if (mOutput->audioHwDev->canSetMasterMute()) { 1622 mMasterMute = false; 1623 } 1624 } 1625 1626 readOutputParameters_l(); 1627 1628 // ++ operator does not compile 1629 for (audio_stream_type_t stream = AUDIO_STREAM_MIN; stream < AUDIO_STREAM_CNT; 1630 stream = (audio_stream_type_t) (stream + 1)) { 1631 mStreamTypes[stream].volume = mAudioFlinger->streamVolume_l(stream); 1632 mStreamTypes[stream].mute = mAudioFlinger->streamMute_l(stream); 1633 } 1634 1635 if (audio_has_proportional_frames(mFormat)) { 1636 mBufferDurationUs = (uint32_t)((mNormalFrameCount * 1000000LL) / mSampleRate); 1637 } else { 1638 bitRate = bitRate != 0 ? bitRate : kOffloadDefaultBitRateBps; 1639 mBufferDurationUs = (uint32_t)((mBufferSize * 8 * 1000000LL) / bitRate); 1640 } 1641} 1642 1643AudioFlinger::PlaybackThread::~PlaybackThread() 1644{ 1645 mAudioFlinger->unregisterWriter(mNBLogWriter); 1646 free(mSinkBuffer); 1647 free(mMixerBuffer); 1648 free(mEffectBuffer); 1649} 1650 1651void AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) 1652{ 1653 dumpInternals(fd, args); 1654 dumpTracks(fd, args); 1655 dumpEffectChains(fd, args); 1656} 1657 1658void AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args __unused) 1659{ 1660 const size_t SIZE = 256; 1661 char buffer[SIZE]; 1662 String8 result; 1663 1664 result.appendFormat(" Stream volumes in dB: "); 1665 for (int i = 0; i < AUDIO_STREAM_CNT; ++i) { 1666 const stream_type_t *st = &mStreamTypes[i]; 1667 if (i > 0) { 1668 result.appendFormat(", "); 1669 } 1670 result.appendFormat("%d:%.2g", i, 20.0 * log10(st->volume)); 1671 if (st->mute) { 1672 result.append("M"); 1673 } 1674 } 1675 result.append("\n"); 1676 write(fd, result.string(), result.length()); 1677 result.clear(); 1678 1679 // These values are "raw"; they will wrap around. See prepareTracks_l() for a better way. 1680 FastTrackUnderruns underruns = getFastTrackUnderruns(0); 1681 dprintf(fd, " Normal mixer raw underrun counters: partial=%u empty=%u\n", 1682 underruns.mBitFields.mPartial, underruns.mBitFields.mEmpty); 1683 1684 size_t numtracks = mTracks.size(); 1685 size_t numactive = mActiveTracks.size(); 1686 dprintf(fd, " %zu Tracks", numtracks); 1687 size_t numactiveseen = 0; 1688 if (numtracks) { 1689 dprintf(fd, " of which %zu are active\n", numactive); 1690 Track::appendDumpHeader(result); 1691 for (size_t i = 0; i < numtracks; ++i) { 1692 sp<Track> track = mTracks[i]; 1693 if (track != 0) { 1694 bool active = mActiveTracks.indexOf(track) >= 0; 1695 if (active) { 1696 numactiveseen++; 1697 } 1698 track->dump(buffer, SIZE, active); 1699 result.append(buffer); 1700 } 1701 } 1702 } else { 1703 result.append("\n"); 1704 } 1705 if (numactiveseen != numactive) { 1706 // some tracks in the active list were not in the tracks list 1707 snprintf(buffer, SIZE, " The following tracks are in the active list but" 1708 " not in the track list\n"); 1709 result.append(buffer); 1710 Track::appendDumpHeader(result); 1711 for (size_t i = 0; i < numactive; ++i) { 1712 sp<Track> track = mActiveTracks[i].promote(); 1713 if (track != 0 && mTracks.indexOf(track) < 0) { 1714 track->dump(buffer, SIZE, true); 1715 result.append(buffer); 1716 } 1717 } 1718 } 1719 1720 write(fd, result.string(), result.size()); 1721} 1722 1723void AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 1724{ 1725 dprintf(fd, "\nOutput thread %p type %d (%s):\n", this, type(), threadTypeToString(type())); 1726 1727 dumpBase(fd, args); 1728 1729 dprintf(fd, " Normal frame count: %zu\n", mNormalFrameCount); 1730 dprintf(fd, " Last write occurred (msecs): %llu\n", 1731 (unsigned long long) ns2ms(systemTime() - mLastWriteTime)); 1732 dprintf(fd, " Total writes: %d\n", mNumWrites); 1733 dprintf(fd, " Delayed writes: %d\n", mNumDelayedWrites); 1734 dprintf(fd, " Blocked in write: %s\n", mInWrite ? "yes" : "no"); 1735 dprintf(fd, " Suspend count: %d\n", mSuspended); 1736 dprintf(fd, " Sink buffer : %p\n", mSinkBuffer); 1737 dprintf(fd, " Mixer buffer: %p\n", mMixerBuffer); 1738 dprintf(fd, " Effect buffer: %p\n", mEffectBuffer); 1739 dprintf(fd, " Fast track availMask=%#x\n", mFastTrackAvailMask); 1740 dprintf(fd, " Standby delay ns=%lld\n", (long long)mStandbyDelayNs); 1741 AudioStreamOut *output = mOutput; 1742 audio_output_flags_t flags = output != NULL ? output->flags : AUDIO_OUTPUT_FLAG_NONE; 1743 String8 flagsAsString = outputFlagsToString(flags); 1744 dprintf(fd, " AudioStreamOut: %p flags %#x (%s)\n", output, flags, flagsAsString.string()); 1745} 1746 1747// Thread virtuals 1748 1749void AudioFlinger::PlaybackThread::onFirstRef() 1750{ 1751 run(mThreadName, ANDROID_PRIORITY_URGENT_AUDIO); 1752} 1753 1754// ThreadBase virtuals 1755void AudioFlinger::PlaybackThread::preExit() 1756{ 1757 ALOGV(" preExit()"); 1758 // FIXME this is using hard-coded strings but in the future, this functionality will be 1759 // converted to use audio HAL extensions required to support tunneling 1760 mOutput->stream->common.set_parameters(&mOutput->stream->common, "exiting=1"); 1761} 1762 1763// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 1764sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 1765 const sp<AudioFlinger::Client>& client, 1766 audio_stream_type_t streamType, 1767 uint32_t sampleRate, 1768 audio_format_t format, 1769 audio_channel_mask_t channelMask, 1770 size_t *pFrameCount, 1771 const sp<IMemory>& sharedBuffer, 1772 audio_session_t sessionId, 1773 IAudioFlinger::track_flags_t *flags, 1774 pid_t tid, 1775 int uid, 1776 status_t *status) 1777{ 1778 size_t frameCount = *pFrameCount; 1779 sp<Track> track; 1780 status_t lStatus; 1781 1782 // client expresses a preference for FAST, but we get the final say 1783 if (*flags & IAudioFlinger::TRACK_FAST) { 1784 if ( 1785 // PCM data 1786 audio_is_linear_pcm(format) && 1787 // TODO: extract as a data library function that checks that a computationally 1788 // expensive downmixer is not required: isFastOutputChannelConversion() 1789 (channelMask == mChannelMask || 1790 mChannelMask != AUDIO_CHANNEL_OUT_STEREO || 1791 (channelMask == AUDIO_CHANNEL_OUT_MONO 1792 /* && mChannelMask == AUDIO_CHANNEL_OUT_STEREO */)) && 1793 // hardware sample rate 1794 (sampleRate == mSampleRate) && 1795 // normal mixer has an associated fast mixer 1796 hasFastMixer() && 1797 // there are sufficient fast track slots available 1798 (mFastTrackAvailMask != 0) 1799 // FIXME test that MixerThread for this fast track has a capable output HAL 1800 // FIXME add a permission test also? 1801 ) { 1802 // static tracks can have any nonzero framecount, streaming tracks check against minimum. 1803 if (sharedBuffer == 0) { 1804 // read the fast track multiplier property the first time it is needed 1805 int ok = pthread_once(&sFastTrackMultiplierOnce, sFastTrackMultiplierInit); 1806 if (ok != 0) { 1807 ALOGE("%s pthread_once failed: %d", __func__, ok); 1808 } 1809 frameCount = max(frameCount, mFrameCount * sFastTrackMultiplier); // incl framecount 0 1810 } 1811 ALOGV("AUDIO_OUTPUT_FLAG_FAST accepted: frameCount=%zu mFrameCount=%zu", 1812 frameCount, mFrameCount); 1813 } else { 1814 ALOGV("AUDIO_OUTPUT_FLAG_FAST denied: sharedBuffer=%p frameCount=%zu " 1815 "mFrameCount=%zu format=%#x mFormat=%#x isLinear=%d channelMask=%#x " 1816 "sampleRate=%u mSampleRate=%u " 1817 "hasFastMixer=%d tid=%d fastTrackAvailMask=%#x", 1818 sharedBuffer.get(), frameCount, mFrameCount, format, mFormat, 1819 audio_is_linear_pcm(format), 1820 channelMask, sampleRate, mSampleRate, hasFastMixer(), tid, mFastTrackAvailMask); 1821 *flags &= ~IAudioFlinger::TRACK_FAST; 1822 } 1823 } 1824 // For normal PCM streaming tracks, update minimum frame count. 1825 // For compatibility with AudioTrack calculation, buffer depth is forced 1826 // to be at least 2 x the normal mixer frame count and cover audio hardware latency. 1827 // This is probably too conservative, but legacy application code may depend on it. 1828 // If you change this calculation, also review the start threshold which is related. 1829 if (!(*flags & IAudioFlinger::TRACK_FAST) 1830 && audio_has_proportional_frames(format) && sharedBuffer == 0) { 1831 // this must match AudioTrack.cpp calculateMinFrameCount(). 1832 // TODO: Move to a common library 1833 uint32_t latencyMs = mOutput->stream->get_latency(mOutput->stream); 1834 uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate); 1835 if (minBufCount < 2) { 1836 minBufCount = 2; 1837 } 1838 // For normal mixing tracks, if speed is > 1.0f (normal), AudioTrack 1839 // or the client should compute and pass in a larger buffer request. 1840 size_t minFrameCount = 1841 minBufCount * sourceFramesNeededWithTimestretch( 1842 sampleRate, mNormalFrameCount, 1843 mSampleRate, AUDIO_TIMESTRETCH_SPEED_NORMAL /*speed*/); 1844 if (frameCount < minFrameCount) { // including frameCount == 0 1845 frameCount = minFrameCount; 1846 } 1847 } 1848 *pFrameCount = frameCount; 1849 1850 switch (mType) { 1851 1852 case DIRECT: 1853 if (audio_is_linear_pcm(format)) { // TODO maybe use audio_has_proportional_frames()? 1854 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1855 ALOGE("createTrack_l() Bad parameter: sampleRate %u format %#x, channelMask 0x%08x " 1856 "for output %p with format %#x", 1857 sampleRate, format, channelMask, mOutput, mFormat); 1858 lStatus = BAD_VALUE; 1859 goto Exit; 1860 } 1861 } 1862 break; 1863 1864 case OFFLOAD: 1865 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1866 ALOGE("createTrack_l() Bad parameter: sampleRate %d format %#x, channelMask 0x%08x \"" 1867 "for output %p with format %#x", 1868 sampleRate, format, channelMask, mOutput, mFormat); 1869 lStatus = BAD_VALUE; 1870 goto Exit; 1871 } 1872 break; 1873 1874 default: 1875 if (!audio_is_linear_pcm(format)) { 1876 ALOGE("createTrack_l() Bad parameter: format %#x \"" 1877 "for output %p with format %#x", 1878 format, mOutput, mFormat); 1879 lStatus = BAD_VALUE; 1880 goto Exit; 1881 } 1882 if (sampleRate > mSampleRate * AUDIO_RESAMPLER_DOWN_RATIO_MAX) { 1883 ALOGE("Sample rate out of range: %u mSampleRate %u", sampleRate, mSampleRate); 1884 lStatus = BAD_VALUE; 1885 goto Exit; 1886 } 1887 break; 1888 1889 } 1890 1891 lStatus = initCheck(); 1892 if (lStatus != NO_ERROR) { 1893 ALOGE("createTrack_l() audio driver not initialized"); 1894 goto Exit; 1895 } 1896 1897 { // scope for mLock 1898 Mutex::Autolock _l(mLock); 1899 1900 // all tracks in same audio session must share the same routing strategy otherwise 1901 // conflicts will happen when tracks are moved from one output to another by audio policy 1902 // manager 1903 uint32_t strategy = AudioSystem::getStrategyForStream(streamType); 1904 for (size_t i = 0; i < mTracks.size(); ++i) { 1905 sp<Track> t = mTracks[i]; 1906 if (t != 0 && t->isExternalTrack()) { 1907 uint32_t actual = AudioSystem::getStrategyForStream(t->streamType()); 1908 if (sessionId == t->sessionId() && strategy != actual) { 1909 ALOGE("createTrack_l() mismatched strategy; expected %u but found %u", 1910 strategy, actual); 1911 lStatus = BAD_VALUE; 1912 goto Exit; 1913 } 1914 } 1915 } 1916 1917 track = new Track(this, client, streamType, sampleRate, format, 1918 channelMask, frameCount, NULL, sharedBuffer, 1919 sessionId, uid, *flags, TrackBase::TYPE_DEFAULT); 1920 1921 lStatus = track != 0 ? track->initCheck() : (status_t) NO_MEMORY; 1922 if (lStatus != NO_ERROR) { 1923 ALOGE("createTrack_l() initCheck failed %d; no control block?", lStatus); 1924 // track must be cleared from the caller as the caller has the AF lock 1925 goto Exit; 1926 } 1927 mTracks.add(track); 1928 1929 sp<EffectChain> chain = getEffectChain_l(sessionId); 1930 if (chain != 0) { 1931 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1932 track->setMainBuffer(chain->inBuffer()); 1933 chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType())); 1934 chain->incTrackCnt(); 1935 } 1936 1937 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 1938 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 1939 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 1940 // so ask activity manager to do this on our behalf 1941 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 1942 } 1943 } 1944 1945 lStatus = NO_ERROR; 1946 1947Exit: 1948 *status = lStatus; 1949 return track; 1950} 1951 1952uint32_t AudioFlinger::PlaybackThread::correctLatency_l(uint32_t latency) const 1953{ 1954 return latency; 1955} 1956 1957uint32_t AudioFlinger::PlaybackThread::latency() const 1958{ 1959 Mutex::Autolock _l(mLock); 1960 return latency_l(); 1961} 1962uint32_t AudioFlinger::PlaybackThread::latency_l() const 1963{ 1964 if (initCheck() == NO_ERROR) { 1965 return correctLatency_l(mOutput->stream->get_latency(mOutput->stream)); 1966 } else { 1967 return 0; 1968 } 1969} 1970 1971void AudioFlinger::PlaybackThread::setMasterVolume(float value) 1972{ 1973 Mutex::Autolock _l(mLock); 1974 // Don't apply master volume in SW if our HAL can do it for us. 1975 if (mOutput && mOutput->audioHwDev && 1976 mOutput->audioHwDev->canSetMasterVolume()) { 1977 mMasterVolume = 1.0; 1978 } else { 1979 mMasterVolume = value; 1980 } 1981} 1982 1983void AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1984{ 1985 Mutex::Autolock _l(mLock); 1986 // Don't apply master mute in SW if our HAL can do it for us. 1987 if (mOutput && mOutput->audioHwDev && 1988 mOutput->audioHwDev->canSetMasterMute()) { 1989 mMasterMute = false; 1990 } else { 1991 mMasterMute = muted; 1992 } 1993} 1994 1995void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) 1996{ 1997 Mutex::Autolock _l(mLock); 1998 mStreamTypes[stream].volume = value; 1999 broadcast_l(); 2000} 2001 2002void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) 2003{ 2004 Mutex::Autolock _l(mLock); 2005 mStreamTypes[stream].mute = muted; 2006 broadcast_l(); 2007} 2008 2009float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const 2010{ 2011 Mutex::Autolock _l(mLock); 2012 return mStreamTypes[stream].volume; 2013} 2014 2015// addTrack_l() must be called with ThreadBase::mLock held 2016status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 2017{ 2018 status_t status = ALREADY_EXISTS; 2019 2020 if (mActiveTracks.indexOf(track) < 0) { 2021 // the track is newly added, make sure it fills up all its 2022 // buffers before playing. This is to ensure the client will 2023 // effectively get the latency it requested. 2024 if (track->isExternalTrack()) { 2025 TrackBase::track_state state = track->mState; 2026 mLock.unlock(); 2027 status = AudioSystem::startOutput(mId, track->streamType(), 2028 track->sessionId()); 2029 mLock.lock(); 2030 // abort track was stopped/paused while we released the lock 2031 if (state != track->mState) { 2032 if (status == NO_ERROR) { 2033 mLock.unlock(); 2034 AudioSystem::stopOutput(mId, track->streamType(), 2035 track->sessionId()); 2036 mLock.lock(); 2037 } 2038 return INVALID_OPERATION; 2039 } 2040 // abort if start is rejected by audio policy manager 2041 if (status != NO_ERROR) { 2042 return PERMISSION_DENIED; 2043 } 2044#ifdef ADD_BATTERY_DATA 2045 // to track the speaker usage 2046 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 2047#endif 2048 } 2049 2050 // set retry count for buffer fill 2051 if (track->isOffloaded()) { 2052 track->mRetryCount = kMaxTrackStartupRetriesOffload; 2053 } else { 2054 track->mRetryCount = kMaxTrackStartupRetries; 2055 } 2056 2057 track->mFillingUpStatus = track->sharedBuffer() != 0 ? Track::FS_FILLED : Track::FS_FILLING; 2058 track->mResetDone = false; 2059 track->mPresentationCompleteFrames = 0; 2060 mActiveTracks.add(track); 2061 mWakeLockUids.add(track->uid()); 2062 mActiveTracksGeneration++; 2063 mLatestActiveTrack = track; 2064 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 2065 if (chain != 0) { 2066 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), 2067 track->sessionId()); 2068 chain->incActiveTrackCnt(); 2069 } 2070 2071 status = NO_ERROR; 2072 } 2073 2074 onAddNewTrack_l(); 2075 return status; 2076} 2077 2078bool AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 2079{ 2080 track->terminate(); 2081 // active tracks are removed by threadLoop() 2082 bool trackActive = (mActiveTracks.indexOf(track) >= 0); 2083 track->mState = TrackBase::STOPPED; 2084 if (!trackActive) { 2085 removeTrack_l(track); 2086 } else if (track->isFastTrack() || track->isOffloaded() || track->isDirect()) { 2087 track->mState = TrackBase::STOPPING_1; 2088 } 2089 2090 return trackActive; 2091} 2092 2093void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 2094{ 2095 track->triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 2096 mTracks.remove(track); 2097 deleteTrackName_l(track->name()); 2098 // redundant as track is about to be destroyed, for dumpsys only 2099 track->mName = -1; 2100 if (track->isFastTrack()) { 2101 int index = track->mFastIndex; 2102 ALOG_ASSERT(0 < index && index < (int)FastMixerState::sMaxFastTracks); 2103 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << index))); 2104 mFastTrackAvailMask |= 1 << index; 2105 // redundant as track is about to be destroyed, for dumpsys only 2106 track->mFastIndex = -1; 2107 } 2108 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 2109 if (chain != 0) { 2110 chain->decTrackCnt(); 2111 } 2112} 2113 2114void AudioFlinger::PlaybackThread::broadcast_l() 2115{ 2116 // Thread could be blocked waiting for async 2117 // so signal it to handle state changes immediately 2118 // If threadLoop is currently unlocked a signal of mWaitWorkCV will 2119 // be lost so we also flag to prevent it blocking on mWaitWorkCV 2120 mSignalPending = true; 2121 mWaitWorkCV.broadcast(); 2122} 2123 2124String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 2125{ 2126 Mutex::Autolock _l(mLock); 2127 if (initCheck() != NO_ERROR) { 2128 return String8(); 2129 } 2130 2131 char *s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 2132 const String8 out_s8(s); 2133 free(s); 2134 return out_s8; 2135} 2136 2137void AudioFlinger::PlaybackThread::ioConfigChanged(audio_io_config_event event, pid_t pid) { 2138 sp<AudioIoDescriptor> desc = new AudioIoDescriptor(); 2139 ALOGV("PlaybackThread::ioConfigChanged, thread %p, event %d", this, event); 2140 2141 desc->mIoHandle = mId; 2142 2143 switch (event) { 2144 case AUDIO_OUTPUT_OPENED: 2145 case AUDIO_OUTPUT_CONFIG_CHANGED: 2146 desc->mPatch = mPatch; 2147 desc->mChannelMask = mChannelMask; 2148 desc->mSamplingRate = mSampleRate; 2149 desc->mFormat = mFormat; 2150 desc->mFrameCount = mNormalFrameCount; // FIXME see 2151 // AudioFlinger::frameCount(audio_io_handle_t) 2152 desc->mFrameCountHAL = mFrameCount; 2153 desc->mLatency = latency_l(); 2154 break; 2155 2156 case AUDIO_OUTPUT_CLOSED: 2157 default: 2158 break; 2159 } 2160 mAudioFlinger->ioConfigChanged(event, desc, pid); 2161} 2162 2163void AudioFlinger::PlaybackThread::writeCallback() 2164{ 2165 ALOG_ASSERT(mCallbackThread != 0); 2166 mCallbackThread->resetWriteBlocked(); 2167} 2168 2169void AudioFlinger::PlaybackThread::drainCallback() 2170{ 2171 ALOG_ASSERT(mCallbackThread != 0); 2172 mCallbackThread->resetDraining(); 2173} 2174 2175void AudioFlinger::PlaybackThread::resetWriteBlocked(uint32_t sequence) 2176{ 2177 Mutex::Autolock _l(mLock); 2178 // reject out of sequence requests 2179 if ((mWriteAckSequence & 1) && (sequence == mWriteAckSequence)) { 2180 mWriteAckSequence &= ~1; 2181 mWaitWorkCV.signal(); 2182 } 2183} 2184 2185void AudioFlinger::PlaybackThread::resetDraining(uint32_t sequence) 2186{ 2187 Mutex::Autolock _l(mLock); 2188 // reject out of sequence requests 2189 if ((mDrainSequence & 1) && (sequence == mDrainSequence)) { 2190 mDrainSequence &= ~1; 2191 mWaitWorkCV.signal(); 2192 } 2193} 2194 2195// static 2196int AudioFlinger::PlaybackThread::asyncCallback(stream_callback_event_t event, 2197 void *param __unused, 2198 void *cookie) 2199{ 2200 AudioFlinger::PlaybackThread *me = (AudioFlinger::PlaybackThread *)cookie; 2201 ALOGV("asyncCallback() event %d", event); 2202 switch (event) { 2203 case STREAM_CBK_EVENT_WRITE_READY: 2204 me->writeCallback(); 2205 break; 2206 case STREAM_CBK_EVENT_DRAIN_READY: 2207 me->drainCallback(); 2208 break; 2209 default: 2210 ALOGW("asyncCallback() unknown event %d", event); 2211 break; 2212 } 2213 return 0; 2214} 2215 2216void AudioFlinger::PlaybackThread::readOutputParameters_l() 2217{ 2218 // unfortunately we have no way of recovering from errors here, hence the LOG_ALWAYS_FATAL 2219 mSampleRate = mOutput->getSampleRate(); 2220 mChannelMask = mOutput->getChannelMask(); 2221 if (!audio_is_output_channel(mChannelMask)) { 2222 LOG_ALWAYS_FATAL("HAL channel mask %#x not valid for output", mChannelMask); 2223 } 2224 if ((mType == MIXER || mType == DUPLICATING) 2225 && !isValidPcmSinkChannelMask(mChannelMask)) { 2226 LOG_ALWAYS_FATAL("HAL channel mask %#x not supported for mixed output", 2227 mChannelMask); 2228 } 2229 mChannelCount = audio_channel_count_from_out_mask(mChannelMask); 2230 2231 // Get actual HAL format. 2232 mHALFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 2233 // Get format from the shim, which will be different than the HAL format 2234 // if playing compressed audio over HDMI passthrough. 2235 mFormat = mOutput->getFormat(); 2236 if (!audio_is_valid_format(mFormat)) { 2237 LOG_ALWAYS_FATAL("HAL format %#x not valid for output", mFormat); 2238 } 2239 if ((mType == MIXER || mType == DUPLICATING) 2240 && !isValidPcmSinkFormat(mFormat)) { 2241 LOG_FATAL("HAL format %#x not supported for mixed output", 2242 mFormat); 2243 } 2244 mFrameSize = mOutput->getFrameSize(); 2245 mBufferSize = mOutput->stream->common.get_buffer_size(&mOutput->stream->common); 2246 mFrameCount = mBufferSize / mFrameSize; 2247 if (mFrameCount & 15) { 2248 ALOGW("HAL output buffer size is %zu frames but AudioMixer requires multiples of 16 frames", 2249 mFrameCount); 2250 } 2251 2252 if ((mOutput->flags & AUDIO_OUTPUT_FLAG_NON_BLOCKING) && 2253 (mOutput->stream->set_callback != NULL)) { 2254 if (mOutput->stream->set_callback(mOutput->stream, 2255 AudioFlinger::PlaybackThread::asyncCallback, this) == 0) { 2256 mUseAsyncWrite = true; 2257 mCallbackThread = new AudioFlinger::AsyncCallbackThread(this); 2258 } 2259 } 2260 2261 mHwSupportsPause = false; 2262 if (mOutput->flags & AUDIO_OUTPUT_FLAG_DIRECT) { 2263 if (mOutput->stream->pause != NULL) { 2264 if (mOutput->stream->resume != NULL) { 2265 mHwSupportsPause = true; 2266 } else { 2267 ALOGW("direct output implements pause but not resume"); 2268 } 2269 } else if (mOutput->stream->resume != NULL) { 2270 ALOGW("direct output implements resume but not pause"); 2271 } 2272 } 2273 if (!mHwSupportsPause && mOutput->flags & AUDIO_OUTPUT_FLAG_HW_AV_SYNC) { 2274 LOG_ALWAYS_FATAL("HW_AV_SYNC requested but HAL does not implement pause and resume"); 2275 } 2276 2277 if (mType == DUPLICATING && mMixerBufferEnabled && mEffectBufferEnabled) { 2278 // For best precision, we use float instead of the associated output 2279 // device format (typically PCM 16 bit). 2280 2281 mFormat = AUDIO_FORMAT_PCM_FLOAT; 2282 mFrameSize = mChannelCount * audio_bytes_per_sample(mFormat); 2283 mBufferSize = mFrameSize * mFrameCount; 2284 2285 // TODO: We currently use the associated output device channel mask and sample rate. 2286 // (1) Perhaps use the ORed channel mask of all downstream MixerThreads 2287 // (if a valid mask) to avoid premature downmix. 2288 // (2) Perhaps use the maximum sample rate of all downstream MixerThreads 2289 // instead of the output device sample rate to avoid loss of high frequency information. 2290 // This may need to be updated as MixerThread/OutputTracks are added and not here. 2291 } 2292 2293 // Calculate size of normal sink buffer relative to the HAL output buffer size 2294 double multiplier = 1.0; 2295 if (mType == MIXER && (kUseFastMixer == FastMixer_Static || 2296 kUseFastMixer == FastMixer_Dynamic)) { 2297 size_t minNormalFrameCount = (kMinNormalSinkBufferSizeMs * mSampleRate) / 1000; 2298 size_t maxNormalFrameCount = (kMaxNormalSinkBufferSizeMs * mSampleRate) / 1000; 2299 // round up minimum and round down maximum to nearest 16 frames to satisfy AudioMixer 2300 minNormalFrameCount = (minNormalFrameCount + 15) & ~15; 2301 maxNormalFrameCount = maxNormalFrameCount & ~15; 2302 if (maxNormalFrameCount < minNormalFrameCount) { 2303 maxNormalFrameCount = minNormalFrameCount; 2304 } 2305 multiplier = (double) minNormalFrameCount / (double) mFrameCount; 2306 if (multiplier <= 1.0) { 2307 multiplier = 1.0; 2308 } else if (multiplier <= 2.0) { 2309 if (2 * mFrameCount <= maxNormalFrameCount) { 2310 multiplier = 2.0; 2311 } else { 2312 multiplier = (double) maxNormalFrameCount / (double) mFrameCount; 2313 } 2314 } else { 2315 // prefer an even multiplier, for compatibility with doubling of fast tracks due to HAL 2316 // SRC (it would be unusual for the normal sink buffer size to not be a multiple of fast 2317 // track, but we sometimes have to do this to satisfy the maximum frame count 2318 // constraint) 2319 // FIXME this rounding up should not be done if no HAL SRC 2320 uint32_t truncMult = (uint32_t) multiplier; 2321 if ((truncMult & 1)) { 2322 if ((truncMult + 1) * mFrameCount <= maxNormalFrameCount) { 2323 ++truncMult; 2324 } 2325 } 2326 multiplier = (double) truncMult; 2327 } 2328 } 2329 mNormalFrameCount = multiplier * mFrameCount; 2330 // round up to nearest 16 frames to satisfy AudioMixer 2331 if (mType == MIXER || mType == DUPLICATING) { 2332 mNormalFrameCount = (mNormalFrameCount + 15) & ~15; 2333 } 2334 ALOGI("HAL output buffer size %zu frames, normal sink buffer size %zu frames", mFrameCount, 2335 mNormalFrameCount); 2336 2337 // Check if we want to throttle the processing to no more than 2x normal rate 2338 mThreadThrottle = property_get_bool("af.thread.throttle", true /* default_value */); 2339 mThreadThrottleTimeMs = 0; 2340 mThreadThrottleEndMs = 0; 2341 mHalfBufferMs = mNormalFrameCount * 1000 / (2 * mSampleRate); 2342 2343 // mSinkBuffer is the sink buffer. Size is always multiple-of-16 frames. 2344 // Originally this was int16_t[] array, need to remove legacy implications. 2345 free(mSinkBuffer); 2346 mSinkBuffer = NULL; 2347 // For sink buffer size, we use the frame size from the downstream sink to avoid problems 2348 // with non PCM formats for compressed music, e.g. AAC, and Offload threads. 2349 const size_t sinkBufferSize = mNormalFrameCount * mFrameSize; 2350 (void)posix_memalign(&mSinkBuffer, 32, sinkBufferSize); 2351 2352 // We resize the mMixerBuffer according to the requirements of the sink buffer which 2353 // drives the output. 2354 free(mMixerBuffer); 2355 mMixerBuffer = NULL; 2356 if (mMixerBufferEnabled) { 2357 mMixerBufferFormat = AUDIO_FORMAT_PCM_FLOAT; // also valid: AUDIO_FORMAT_PCM_16_BIT. 2358 mMixerBufferSize = mNormalFrameCount * mChannelCount 2359 * audio_bytes_per_sample(mMixerBufferFormat); 2360 (void)posix_memalign(&mMixerBuffer, 32, mMixerBufferSize); 2361 } 2362 free(mEffectBuffer); 2363 mEffectBuffer = NULL; 2364 if (mEffectBufferEnabled) { 2365 mEffectBufferFormat = AUDIO_FORMAT_PCM_16_BIT; // Note: Effects support 16b only 2366 mEffectBufferSize = mNormalFrameCount * mChannelCount 2367 * audio_bytes_per_sample(mEffectBufferFormat); 2368 (void)posix_memalign(&mEffectBuffer, 32, mEffectBufferSize); 2369 } 2370 2371 // force reconfiguration of effect chains and engines to take new buffer size and audio 2372 // parameters into account 2373 // Note that mLock is not held when readOutputParameters_l() is called from the constructor 2374 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 2375 // matter. 2376 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 2377 Vector< sp<EffectChain> > effectChains = mEffectChains; 2378 for (size_t i = 0; i < effectChains.size(); i ++) { 2379 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 2380 } 2381} 2382 2383 2384status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames) 2385{ 2386 if (halFrames == NULL || dspFrames == NULL) { 2387 return BAD_VALUE; 2388 } 2389 Mutex::Autolock _l(mLock); 2390 if (initCheck() != NO_ERROR) { 2391 return INVALID_OPERATION; 2392 } 2393 int64_t framesWritten = mBytesWritten / mFrameSize; 2394 *halFrames = framesWritten; 2395 2396 if (isSuspended()) { 2397 // return an estimation of rendered frames when the output is suspended 2398 size_t latencyFrames = (latency_l() * mSampleRate) / 1000; 2399 *dspFrames = (uint32_t) 2400 (framesWritten >= (int64_t)latencyFrames ? framesWritten - latencyFrames : 0); 2401 return NO_ERROR; 2402 } else { 2403 status_t status; 2404 uint32_t frames; 2405 status = mOutput->getRenderPosition(&frames); 2406 *dspFrames = (size_t)frames; 2407 return status; 2408 } 2409} 2410 2411uint32_t AudioFlinger::PlaybackThread::hasAudioSession(audio_session_t sessionId) const 2412{ 2413 Mutex::Autolock _l(mLock); 2414 uint32_t result = 0; 2415 if (getEffectChain_l(sessionId) != 0) { 2416 result = EFFECT_SESSION; 2417 } 2418 2419 for (size_t i = 0; i < mTracks.size(); ++i) { 2420 sp<Track> track = mTracks[i]; 2421 if (sessionId == track->sessionId() && !track->isInvalid()) { 2422 result |= TRACK_SESSION; 2423 break; 2424 } 2425 } 2426 2427 return result; 2428} 2429 2430uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(audio_session_t sessionId) 2431{ 2432 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 2433 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 2434 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 2435 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 2436 } 2437 for (size_t i = 0; i < mTracks.size(); i++) { 2438 sp<Track> track = mTracks[i]; 2439 if (sessionId == track->sessionId() && !track->isInvalid()) { 2440 return AudioSystem::getStrategyForStream(track->streamType()); 2441 } 2442 } 2443 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 2444} 2445 2446 2447AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 2448{ 2449 Mutex::Autolock _l(mLock); 2450 return mOutput; 2451} 2452 2453AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 2454{ 2455 Mutex::Autolock _l(mLock); 2456 AudioStreamOut *output = mOutput; 2457 mOutput = NULL; 2458 // FIXME FastMixer might also have a raw ptr to mOutputSink; 2459 // must push a NULL and wait for ack 2460 mOutputSink.clear(); 2461 mPipeSink.clear(); 2462 mNormalSink.clear(); 2463 return output; 2464} 2465 2466// this method must always be called either with ThreadBase mLock held or inside the thread loop 2467audio_stream_t* AudioFlinger::PlaybackThread::stream() const 2468{ 2469 if (mOutput == NULL) { 2470 return NULL; 2471 } 2472 return &mOutput->stream->common; 2473} 2474 2475uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() const 2476{ 2477 return (uint32_t)((uint32_t)((mNormalFrameCount * 1000) / mSampleRate) * 1000); 2478} 2479 2480status_t AudioFlinger::PlaybackThread::setSyncEvent(const sp<SyncEvent>& event) 2481{ 2482 if (!isValidSyncEvent(event)) { 2483 return BAD_VALUE; 2484 } 2485 2486 Mutex::Autolock _l(mLock); 2487 2488 for (size_t i = 0; i < mTracks.size(); ++i) { 2489 sp<Track> track = mTracks[i]; 2490 if (event->triggerSession() == track->sessionId()) { 2491 (void) track->setSyncEvent(event); 2492 return NO_ERROR; 2493 } 2494 } 2495 2496 return NAME_NOT_FOUND; 2497} 2498 2499bool AudioFlinger::PlaybackThread::isValidSyncEvent(const sp<SyncEvent>& event) const 2500{ 2501 return event->type() == AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE; 2502} 2503 2504void AudioFlinger::PlaybackThread::threadLoop_removeTracks( 2505 const Vector< sp<Track> >& tracksToRemove) 2506{ 2507 size_t count = tracksToRemove.size(); 2508 if (count > 0) { 2509 for (size_t i = 0 ; i < count ; i++) { 2510 const sp<Track>& track = tracksToRemove.itemAt(i); 2511 if (track->isExternalTrack()) { 2512 AudioSystem::stopOutput(mId, track->streamType(), 2513 track->sessionId()); 2514#ifdef ADD_BATTERY_DATA 2515 // to track the speaker usage 2516 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 2517#endif 2518 if (track->isTerminated()) { 2519 AudioSystem::releaseOutput(mId, track->streamType(), 2520 track->sessionId()); 2521 } 2522 } 2523 } 2524 } 2525} 2526 2527void AudioFlinger::PlaybackThread::checkSilentMode_l() 2528{ 2529 if (!mMasterMute) { 2530 char value[PROPERTY_VALUE_MAX]; 2531 if (mOutDevice == AUDIO_DEVICE_OUT_REMOTE_SUBMIX) { 2532 ALOGD("ro.audio.silent will be ignored for threads on AUDIO_DEVICE_OUT_REMOTE_SUBMIX"); 2533 return; 2534 } 2535 if (property_get("ro.audio.silent", value, "0") > 0) { 2536 char *endptr; 2537 unsigned long ul = strtoul(value, &endptr, 0); 2538 if (*endptr == '\0' && ul != 0) { 2539 ALOGD("Silence is golden"); 2540 // The setprop command will not allow a property to be changed after 2541 // the first time it is set, so we don't have to worry about un-muting. 2542 setMasterMute_l(true); 2543 } 2544 } 2545 } 2546} 2547 2548// shared by MIXER and DIRECT, overridden by DUPLICATING 2549ssize_t AudioFlinger::PlaybackThread::threadLoop_write() 2550{ 2551 // FIXME rewrite to reduce number of system calls 2552 mLastWriteTime = systemTime(); 2553 mInWrite = true; 2554 ssize_t bytesWritten; 2555 const size_t offset = mCurrentWriteLength - mBytesRemaining; 2556 2557 // If an NBAIO sink is present, use it to write the normal mixer's submix 2558 if (mNormalSink != 0) { 2559 2560 const size_t count = mBytesRemaining / mFrameSize; 2561 2562 ATRACE_BEGIN("write"); 2563 // update the setpoint when AudioFlinger::mScreenState changes 2564 uint32_t screenState = AudioFlinger::mScreenState; 2565 if (screenState != mScreenState) { 2566 mScreenState = screenState; 2567 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 2568 if (pipe != NULL) { 2569 pipe->setAvgFrames((mScreenState & 1) ? 2570 (pipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2571 } 2572 } 2573 ssize_t framesWritten = mNormalSink->write((char *)mSinkBuffer + offset, count); 2574 ATRACE_END(); 2575 if (framesWritten > 0) { 2576 bytesWritten = framesWritten * mFrameSize; 2577 } else { 2578 bytesWritten = framesWritten; 2579 } 2580 // otherwise use the HAL / AudioStreamOut directly 2581 } else { 2582 // Direct output and offload threads 2583 2584 if (mUseAsyncWrite) { 2585 ALOGW_IF(mWriteAckSequence & 1, "threadLoop_write(): out of sequence write request"); 2586 mWriteAckSequence += 2; 2587 mWriteAckSequence |= 1; 2588 ALOG_ASSERT(mCallbackThread != 0); 2589 mCallbackThread->setWriteBlocked(mWriteAckSequence); 2590 } 2591 // FIXME We should have an implementation of timestamps for direct output threads. 2592 // They are used e.g for multichannel PCM playback over HDMI. 2593 bytesWritten = mOutput->write((char *)mSinkBuffer + offset, mBytesRemaining); 2594 2595 if (mUseAsyncWrite && 2596 ((bytesWritten < 0) || (bytesWritten == (ssize_t)mBytesRemaining))) { 2597 // do not wait for async callback in case of error of full write 2598 mWriteAckSequence &= ~1; 2599 ALOG_ASSERT(mCallbackThread != 0); 2600 mCallbackThread->setWriteBlocked(mWriteAckSequence); 2601 } 2602 } 2603 2604 mNumWrites++; 2605 mInWrite = false; 2606 mStandby = false; 2607 return bytesWritten; 2608} 2609 2610void AudioFlinger::PlaybackThread::threadLoop_drain() 2611{ 2612 if (mOutput->stream->drain) { 2613 ALOGV("draining %s", (mMixerStatus == MIXER_DRAIN_TRACK) ? "early" : "full"); 2614 if (mUseAsyncWrite) { 2615 ALOGW_IF(mDrainSequence & 1, "threadLoop_drain(): out of sequence drain request"); 2616 mDrainSequence |= 1; 2617 ALOG_ASSERT(mCallbackThread != 0); 2618 mCallbackThread->setDraining(mDrainSequence); 2619 } 2620 mOutput->stream->drain(mOutput->stream, 2621 (mMixerStatus == MIXER_DRAIN_TRACK) ? AUDIO_DRAIN_EARLY_NOTIFY 2622 : AUDIO_DRAIN_ALL); 2623 } 2624} 2625 2626void AudioFlinger::PlaybackThread::threadLoop_exit() 2627{ 2628 { 2629 Mutex::Autolock _l(mLock); 2630 for (size_t i = 0; i < mTracks.size(); i++) { 2631 sp<Track> track = mTracks[i]; 2632 track->invalidate(); 2633 } 2634 } 2635} 2636 2637/* 2638The derived values that are cached: 2639 - mSinkBufferSize from frame count * frame size 2640 - mActiveSleepTimeUs from activeSleepTimeUs() 2641 - mIdleSleepTimeUs from idleSleepTimeUs() 2642 - mStandbyDelayNs from mActiveSleepTimeUs (DIRECT only) or forced to at least 2643 kDefaultStandbyTimeInNsecs when connected to an A2DP device. 2644 - maxPeriod from frame count and sample rate (MIXER only) 2645 2646The parameters that affect these derived values are: 2647 - frame count 2648 - frame size 2649 - sample rate 2650 - device type: A2DP or not 2651 - device latency 2652 - format: PCM or not 2653 - active sleep time 2654 - idle sleep time 2655*/ 2656 2657void AudioFlinger::PlaybackThread::cacheParameters_l() 2658{ 2659 mSinkBufferSize = mNormalFrameCount * mFrameSize; 2660 mActiveSleepTimeUs = activeSleepTimeUs(); 2661 mIdleSleepTimeUs = idleSleepTimeUs(); 2662 2663 // make sure standby delay is not too short when connected to an A2DP sink to avoid 2664 // truncating audio when going to standby. 2665 mStandbyDelayNs = AudioFlinger::mStandbyTimeInNsecs; 2666 if ((mOutDevice & AUDIO_DEVICE_OUT_ALL_A2DP) != 0) { 2667 if (mStandbyDelayNs < kDefaultStandbyTimeInNsecs) { 2668 mStandbyDelayNs = kDefaultStandbyTimeInNsecs; 2669 } 2670 } 2671} 2672 2673void AudioFlinger::PlaybackThread::invalidateTracks_l(audio_stream_type_t streamType) 2674{ 2675 ALOGV("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %zu", 2676 this, streamType, mTracks.size()); 2677 2678 size_t size = mTracks.size(); 2679 for (size_t i = 0; i < size; i++) { 2680 sp<Track> t = mTracks[i]; 2681 if (t->streamType() == streamType && t->isExternalTrack()) { 2682 t->invalidate(); 2683 } 2684 } 2685} 2686 2687void AudioFlinger::PlaybackThread::invalidateTracks(audio_stream_type_t streamType) 2688{ 2689 Mutex::Autolock _l(mLock); 2690 invalidateTracks_l(streamType); 2691} 2692 2693status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 2694{ 2695 audio_session_t session = chain->sessionId(); 2696 int16_t* buffer = reinterpret_cast<int16_t*>(mEffectBufferEnabled 2697 ? mEffectBuffer : mSinkBuffer); 2698 bool ownsBuffer = false; 2699 2700 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 2701 if (session > AUDIO_SESSION_OUTPUT_MIX) { 2702 // Only one effect chain can be present in direct output thread and it uses 2703 // the sink buffer as input 2704 if (mType != DIRECT) { 2705 size_t numSamples = mNormalFrameCount * mChannelCount; 2706 buffer = new int16_t[numSamples]; 2707 memset(buffer, 0, numSamples * sizeof(int16_t)); 2708 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 2709 ownsBuffer = true; 2710 } 2711 2712 // Attach all tracks with same session ID to this chain. 2713 for (size_t i = 0; i < mTracks.size(); ++i) { 2714 sp<Track> track = mTracks[i]; 2715 if (session == track->sessionId()) { 2716 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), 2717 buffer); 2718 track->setMainBuffer(buffer); 2719 chain->incTrackCnt(); 2720 } 2721 } 2722 2723 // indicate all active tracks in the chain 2724 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 2725 sp<Track> track = mActiveTracks[i].promote(); 2726 if (track == 0) { 2727 continue; 2728 } 2729 if (session == track->sessionId()) { 2730 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 2731 chain->incActiveTrackCnt(); 2732 } 2733 } 2734 } 2735 chain->setThread(this); 2736 chain->setInBuffer(buffer, ownsBuffer); 2737 chain->setOutBuffer(reinterpret_cast<int16_t*>(mEffectBufferEnabled 2738 ? mEffectBuffer : mSinkBuffer)); 2739 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 2740 // chains list in order to be processed last as it contains output stage effects. 2741 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 2742 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 2743 // after track specific effects and before output stage. 2744 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 2745 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX. 2746 // Effect chain for other sessions are inserted at beginning of effect 2747 // chains list to be processed before output mix effects. Relative order between other 2748 // sessions is not important. 2749 static_assert(AUDIO_SESSION_OUTPUT_MIX == 0 && 2750 AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX, 2751 "audio_session_t constants misdefined"); 2752 size_t size = mEffectChains.size(); 2753 size_t i = 0; 2754 for (i = 0; i < size; i++) { 2755 if (mEffectChains[i]->sessionId() < session) { 2756 break; 2757 } 2758 } 2759 mEffectChains.insertAt(chain, i); 2760 checkSuspendOnAddEffectChain_l(chain); 2761 2762 return NO_ERROR; 2763} 2764 2765size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 2766{ 2767 audio_session_t session = chain->sessionId(); 2768 2769 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 2770 2771 for (size_t i = 0; i < mEffectChains.size(); i++) { 2772 if (chain == mEffectChains[i]) { 2773 mEffectChains.removeAt(i); 2774 // detach all active tracks from the chain 2775 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 2776 sp<Track> track = mActiveTracks[i].promote(); 2777 if (track == 0) { 2778 continue; 2779 } 2780 if (session == track->sessionId()) { 2781 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 2782 chain.get(), session); 2783 chain->decActiveTrackCnt(); 2784 } 2785 } 2786 2787 // detach all tracks with same session ID from this chain 2788 for (size_t i = 0; i < mTracks.size(); ++i) { 2789 sp<Track> track = mTracks[i]; 2790 if (session == track->sessionId()) { 2791 track->setMainBuffer(reinterpret_cast<int16_t*>(mSinkBuffer)); 2792 chain->decTrackCnt(); 2793 } 2794 } 2795 break; 2796 } 2797 } 2798 return mEffectChains.size(); 2799} 2800 2801status_t AudioFlinger::PlaybackThread::attachAuxEffect( 2802 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2803{ 2804 Mutex::Autolock _l(mLock); 2805 return attachAuxEffect_l(track, EffectId); 2806} 2807 2808status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 2809 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2810{ 2811 status_t status = NO_ERROR; 2812 2813 if (EffectId == 0) { 2814 track->setAuxBuffer(0, NULL); 2815 } else { 2816 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 2817 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 2818 if (effect != 0) { 2819 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 2820 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 2821 } else { 2822 status = INVALID_OPERATION; 2823 } 2824 } else { 2825 status = BAD_VALUE; 2826 } 2827 } 2828 return status; 2829} 2830 2831void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 2832{ 2833 for (size_t i = 0; i < mTracks.size(); ++i) { 2834 sp<Track> track = mTracks[i]; 2835 if (track->auxEffectId() == effectId) { 2836 attachAuxEffect_l(track, 0); 2837 } 2838 } 2839} 2840 2841bool AudioFlinger::PlaybackThread::threadLoop() 2842{ 2843 Vector< sp<Track> > tracksToRemove; 2844 2845 mStandbyTimeNs = systemTime(); 2846 2847 // MIXER 2848 nsecs_t lastWarning = 0; 2849 2850 // DUPLICATING 2851 // FIXME could this be made local to while loop? 2852 writeFrames = 0; 2853 2854 int lastGeneration = 0; 2855 2856 cacheParameters_l(); 2857 mSleepTimeUs = mIdleSleepTimeUs; 2858 2859 if (mType == MIXER) { 2860 sleepTimeShift = 0; 2861 } 2862 2863 CpuStats cpuStats; 2864 const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid())); 2865 2866 acquireWakeLock(); 2867 2868 // mNBLogWriter->log can only be called while thread mutex mLock is held. 2869 // So if you need to log when mutex is unlocked, set logString to a non-NULL string, 2870 // and then that string will be logged at the next convenient opportunity. 2871 const char *logString = NULL; 2872 2873 checkSilentMode_l(); 2874 2875 while (!exitPending()) 2876 { 2877 cpuStats.sample(myName); 2878 2879 Vector< sp<EffectChain> > effectChains; 2880 2881 { // scope for mLock 2882 2883 Mutex::Autolock _l(mLock); 2884 2885 processConfigEvents_l(); 2886 2887 if (logString != NULL) { 2888 mNBLogWriter->logTimestamp(); 2889 mNBLogWriter->log(logString); 2890 logString = NULL; 2891 } 2892 2893 // Gather the framesReleased counters for all active tracks, 2894 // and associate with the sink frames written out. We need 2895 // this to convert the sink timestamp to the track timestamp. 2896 if (mNormalSink != 0) { 2897 // Note: The DuplicatingThread may not have a mNormalSink. 2898 // We always fetch the timestamp here because often the downstream 2899 // sink will block whie writing. 2900 ExtendedTimestamp timestamp; // use private copy to fetch 2901 (void) mNormalSink->getTimestamp(timestamp); 2902 // copy over kernel info 2903 mTimestamp.mPosition[ExtendedTimestamp::LOCATION_KERNEL] = 2904 timestamp.mPosition[ExtendedTimestamp::LOCATION_KERNEL]; 2905 mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL] = 2906 timestamp.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL]; 2907 } 2908 // mFramesWritten for non-offloaded tracks are contiguous 2909 // even after standby() is called. This is useful for the track frame 2910 // to sink frame mapping. 2911 mTimestamp.mPosition[ExtendedTimestamp::LOCATION_SERVER] = mFramesWritten; 2912 mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_SERVER] = systemTime(); 2913 const size_t size = mActiveTracks.size(); 2914 for (size_t i = 0; i < size; ++i) { 2915 sp<Track> t = mActiveTracks[i].promote(); 2916 if (t != 0 && !t->isFastTrack()) { 2917 t->updateTrackFrameInfo( 2918 t->mAudioTrackServerProxy->framesReleased(), 2919 mFramesWritten, 2920 mTimestamp); 2921 } 2922 } 2923 2924 saveOutputTracks(); 2925 if (mSignalPending) { 2926 // A signal was raised while we were unlocked 2927 mSignalPending = false; 2928 } else if (waitingAsyncCallback_l()) { 2929 if (exitPending()) { 2930 break; 2931 } 2932 bool released = false; 2933 if (!keepWakeLock()) { 2934 releaseWakeLock_l(); 2935 released = true; 2936 } 2937 mWakeLockUids.clear(); 2938 mActiveTracksGeneration++; 2939 ALOGV("wait async completion"); 2940 mWaitWorkCV.wait(mLock); 2941 ALOGV("async completion/wake"); 2942 if (released) { 2943 acquireWakeLock_l(); 2944 } 2945 mStandbyTimeNs = systemTime() + mStandbyDelayNs; 2946 mSleepTimeUs = 0; 2947 2948 continue; 2949 } 2950 if ((!mActiveTracks.size() && systemTime() > mStandbyTimeNs) || 2951 isSuspended()) { 2952 // put audio hardware into standby after short delay 2953 if (shouldStandby_l()) { 2954 2955 threadLoop_standby(); 2956 2957 mStandby = true; 2958 } 2959 2960 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2961 // we're about to wait, flush the binder command buffer 2962 IPCThreadState::self()->flushCommands(); 2963 2964 clearOutputTracks(); 2965 2966 if (exitPending()) { 2967 break; 2968 } 2969 2970 releaseWakeLock_l(); 2971 mWakeLockUids.clear(); 2972 mActiveTracksGeneration++; 2973 // wait until we have something to do... 2974 ALOGV("%s going to sleep", myName.string()); 2975 mWaitWorkCV.wait(mLock); 2976 ALOGV("%s waking up", myName.string()); 2977 acquireWakeLock_l(); 2978 2979 mMixerStatus = MIXER_IDLE; 2980 mMixerStatusIgnoringFastTracks = MIXER_IDLE; 2981 mBytesWritten = 0; 2982 mBytesRemaining = 0; 2983 checkSilentMode_l(); 2984 2985 mStandbyTimeNs = systemTime() + mStandbyDelayNs; 2986 mSleepTimeUs = mIdleSleepTimeUs; 2987 if (mType == MIXER) { 2988 sleepTimeShift = 0; 2989 } 2990 2991 continue; 2992 } 2993 } 2994 // mMixerStatusIgnoringFastTracks is also updated internally 2995 mMixerStatus = prepareTracks_l(&tracksToRemove); 2996 2997 // compare with previously applied list 2998 if (lastGeneration != mActiveTracksGeneration) { 2999 // update wakelock 3000 updateWakeLockUids_l(mWakeLockUids); 3001 lastGeneration = mActiveTracksGeneration; 3002 } 3003 3004 // prevent any changes in effect chain list and in each effect chain 3005 // during mixing and effect process as the audio buffers could be deleted 3006 // or modified if an effect is created or deleted 3007 lockEffectChains_l(effectChains); 3008 } // mLock scope ends 3009 3010 if (mBytesRemaining == 0) { 3011 mCurrentWriteLength = 0; 3012 if (mMixerStatus == MIXER_TRACKS_READY) { 3013 // threadLoop_mix() sets mCurrentWriteLength 3014 threadLoop_mix(); 3015 } else if ((mMixerStatus != MIXER_DRAIN_TRACK) 3016 && (mMixerStatus != MIXER_DRAIN_ALL)) { 3017 // threadLoop_sleepTime sets mSleepTimeUs to 0 if data 3018 // must be written to HAL 3019 threadLoop_sleepTime(); 3020 if (mSleepTimeUs == 0) { 3021 mCurrentWriteLength = mSinkBufferSize; 3022 } 3023 } 3024 // Either threadLoop_mix() or threadLoop_sleepTime() should have set 3025 // mMixerBuffer with data if mMixerBufferValid is true and mSleepTimeUs == 0. 3026 // Merge mMixerBuffer data into mEffectBuffer (if any effects are valid) 3027 // or mSinkBuffer (if there are no effects). 3028 // 3029 // This is done pre-effects computation; if effects change to 3030 // support higher precision, this needs to move. 3031 // 3032 // mMixerBufferValid is only set true by MixerThread::prepareTracks_l(). 3033 // TODO use mSleepTimeUs == 0 as an additional condition. 3034 if (mMixerBufferValid) { 3035 void *buffer = mEffectBufferValid ? mEffectBuffer : mSinkBuffer; 3036 audio_format_t format = mEffectBufferValid ? mEffectBufferFormat : mFormat; 3037 3038 // mono blend occurs for mixer threads only (not direct or offloaded) 3039 // and is handled here if we're going directly to the sink. 3040 if (requireMonoBlend() && !mEffectBufferValid) { 3041 mono_blend(mMixerBuffer, mMixerBufferFormat, mChannelCount, mNormalFrameCount, 3042 true /*limit*/); 3043 } 3044 3045 memcpy_by_audio_format(buffer, format, mMixerBuffer, mMixerBufferFormat, 3046 mNormalFrameCount * mChannelCount); 3047 } 3048 3049 mBytesRemaining = mCurrentWriteLength; 3050 if (isSuspended()) { 3051 mSleepTimeUs = suspendSleepTimeUs(); 3052 // simulate write to HAL when suspended 3053 mBytesWritten += mSinkBufferSize; 3054 mFramesWritten += mSinkBufferSize / mFrameSize; 3055 mBytesRemaining = 0; 3056 } 3057 3058 // only process effects if we're going to write 3059 if (mSleepTimeUs == 0 && mType != OFFLOAD) { 3060 for (size_t i = 0; i < effectChains.size(); i ++) { 3061 effectChains[i]->process_l(); 3062 } 3063 } 3064 } 3065 // Process effect chains for offloaded thread even if no audio 3066 // was read from audio track: process only updates effect state 3067 // and thus does have to be synchronized with audio writes but may have 3068 // to be called while waiting for async write callback 3069 if (mType == OFFLOAD) { 3070 for (size_t i = 0; i < effectChains.size(); i ++) { 3071 effectChains[i]->process_l(); 3072 } 3073 } 3074 3075 // Only if the Effects buffer is enabled and there is data in the 3076 // Effects buffer (buffer valid), we need to 3077 // copy into the sink buffer. 3078 // TODO use mSleepTimeUs == 0 as an additional condition. 3079 if (mEffectBufferValid) { 3080 //ALOGV("writing effect buffer to sink buffer format %#x", mFormat); 3081 3082 if (requireMonoBlend()) { 3083 mono_blend(mEffectBuffer, mEffectBufferFormat, mChannelCount, mNormalFrameCount, 3084 true /*limit*/); 3085 } 3086 3087 memcpy_by_audio_format(mSinkBuffer, mFormat, mEffectBuffer, mEffectBufferFormat, 3088 mNormalFrameCount * mChannelCount); 3089 } 3090 3091 // enable changes in effect chain 3092 unlockEffectChains(effectChains); 3093 3094 if (!waitingAsyncCallback()) { 3095 // mSleepTimeUs == 0 means we must write to audio hardware 3096 if (mSleepTimeUs == 0) { 3097 ssize_t ret = 0; 3098 if (mBytesRemaining) { 3099 ret = threadLoop_write(); 3100 if (ret < 0) { 3101 mBytesRemaining = 0; 3102 } else { 3103 mBytesWritten += ret; 3104 mBytesRemaining -= ret; 3105 mFramesWritten += ret / mFrameSize; 3106 } 3107 } else if ((mMixerStatus == MIXER_DRAIN_TRACK) || 3108 (mMixerStatus == MIXER_DRAIN_ALL)) { 3109 threadLoop_drain(); 3110 } 3111 if (mType == MIXER && !mStandby) { 3112 // write blocked detection 3113 nsecs_t now = systemTime(); 3114 nsecs_t delta = now - mLastWriteTime; 3115 if (delta > maxPeriod) { 3116 mNumDelayedWrites++; 3117 if ((now - lastWarning) > kWarningThrottleNs) { 3118 ATRACE_NAME("underrun"); 3119 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 3120 (unsigned long long) ns2ms(delta), mNumDelayedWrites, this); 3121 lastWarning = now; 3122 } 3123 } 3124 3125 if (mThreadThrottle 3126 && mMixerStatus == MIXER_TRACKS_READY // we are mixing (active tracks) 3127 && ret > 0) { // we wrote something 3128 // Limit MixerThread data processing to no more than twice the 3129 // expected processing rate. 3130 // 3131 // This helps prevent underruns with NuPlayer and other applications 3132 // which may set up buffers that are close to the minimum size, or use 3133 // deep buffers, and rely on a double-buffering sleep strategy to fill. 3134 // 3135 // The throttle smooths out sudden large data drains from the device, 3136 // e.g. when it comes out of standby, which often causes problems with 3137 // (1) mixer threads without a fast mixer (which has its own warm-up) 3138 // (2) minimum buffer sized tracks (even if the track is full, 3139 // the app won't fill fast enough to handle the sudden draw). 3140 3141 const int32_t deltaMs = delta / 1000000; 3142 const int32_t throttleMs = mHalfBufferMs - deltaMs; 3143 if ((signed)mHalfBufferMs >= throttleMs && throttleMs > 0) { 3144 usleep(throttleMs * 1000); 3145 // notify of throttle start on verbose log 3146 ALOGV_IF(mThreadThrottleEndMs == mThreadThrottleTimeMs, 3147 "mixer(%p) throttle begin:" 3148 " ret(%zd) deltaMs(%d) requires sleep %d ms", 3149 this, ret, deltaMs, throttleMs); 3150 mThreadThrottleTimeMs += throttleMs; 3151 } else { 3152 uint32_t diff = mThreadThrottleTimeMs - mThreadThrottleEndMs; 3153 if (diff > 0) { 3154 // notify of throttle end on debug log 3155 // but prevent spamming for bluetooth 3156 ALOGD_IF(!audio_is_a2dp_out_device(outDevice()), 3157 "mixer(%p) throttle end: throttle time(%u)", this, diff); 3158 mThreadThrottleEndMs = mThreadThrottleTimeMs; 3159 } 3160 } 3161 } 3162 } 3163 3164 } else { 3165 ATRACE_BEGIN("sleep"); 3166 if ((mType == OFFLOAD) && !audio_has_proportional_frames(mFormat)) { 3167 Mutex::Autolock _l(mLock); 3168 if (!mSignalPending && !exitPending()) { 3169 // If more than one buffer has been written to the audio HAL since exiting 3170 // standby or last flush, do not sleep more than one buffer duration 3171 // since last write and not less than kDirectMinSleepTimeUs. 3172 // Wake up if a command is received 3173 uint32_t timeoutUs = mSleepTimeUs; 3174 if (mBytesWritten >= (int64_t) mBufferSize) { 3175 nsecs_t now = systemTime(); 3176 uint32_t deltaUs = (uint32_t)((now - mLastWriteTime) / 1000); 3177 if (timeoutUs + deltaUs > mBufferDurationUs) { 3178 if (mBufferDurationUs > deltaUs) { 3179 timeoutUs = mBufferDurationUs - deltaUs; 3180 if (timeoutUs < kDirectMinSleepTimeUs) { 3181 timeoutUs = kDirectMinSleepTimeUs; 3182 } 3183 } else { 3184 timeoutUs = kDirectMinSleepTimeUs; 3185 } 3186 } 3187 } 3188 mWaitWorkCV.waitRelative(mLock, microseconds((nsecs_t)timeoutUs)); 3189 } 3190 } else { 3191 usleep(mSleepTimeUs); 3192 } 3193 ATRACE_END(); 3194 } 3195 } 3196 3197 // Finally let go of removed track(s), without the lock held 3198 // since we can't guarantee the destructors won't acquire that 3199 // same lock. This will also mutate and push a new fast mixer state. 3200 threadLoop_removeTracks(tracksToRemove); 3201 tracksToRemove.clear(); 3202 3203 // FIXME I don't understand the need for this here; 3204 // it was in the original code but maybe the 3205 // assignment in saveOutputTracks() makes this unnecessary? 3206 clearOutputTracks(); 3207 3208 // Effect chains will be actually deleted here if they were removed from 3209 // mEffectChains list during mixing or effects processing 3210 effectChains.clear(); 3211 3212 // FIXME Note that the above .clear() is no longer necessary since effectChains 3213 // is now local to this block, but will keep it for now (at least until merge done). 3214 } 3215 3216 threadLoop_exit(); 3217 3218 if (!mStandby) { 3219 threadLoop_standby(); 3220 mStandby = true; 3221 } 3222 3223 releaseWakeLock(); 3224 mWakeLockUids.clear(); 3225 mActiveTracksGeneration++; 3226 3227 ALOGV("Thread %p type %d exiting", this, mType); 3228 return false; 3229} 3230 3231// removeTracks_l() must be called with ThreadBase::mLock held 3232void AudioFlinger::PlaybackThread::removeTracks_l(const Vector< sp<Track> >& tracksToRemove) 3233{ 3234 size_t count = tracksToRemove.size(); 3235 if (count > 0) { 3236 for (size_t i=0 ; i<count ; i++) { 3237 const sp<Track>& track = tracksToRemove.itemAt(i); 3238 mActiveTracks.remove(track); 3239 mWakeLockUids.remove(track->uid()); 3240 mActiveTracksGeneration++; 3241 ALOGV("removeTracks_l removing track on session %d", track->sessionId()); 3242 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 3243 if (chain != 0) { 3244 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), 3245 track->sessionId()); 3246 chain->decActiveTrackCnt(); 3247 } 3248 if (track->isTerminated()) { 3249 removeTrack_l(track); 3250 } 3251 } 3252 } 3253 3254} 3255 3256status_t AudioFlinger::PlaybackThread::getTimestamp_l(AudioTimestamp& timestamp) 3257{ 3258 if (mNormalSink != 0) { 3259 ExtendedTimestamp ets; 3260 status_t status = mNormalSink->getTimestamp(ets); 3261 if (status == NO_ERROR) { 3262 status = ets.getBestTimestamp(×tamp); 3263 } 3264 return status; 3265 } 3266 if ((mType == OFFLOAD || mType == DIRECT) 3267 && mOutput != NULL && mOutput->stream->get_presentation_position) { 3268 uint64_t position64; 3269 int ret = mOutput->getPresentationPosition(&position64, ×tamp.mTime); 3270 if (ret == 0) { 3271 timestamp.mPosition = (uint32_t)position64; 3272 return NO_ERROR; 3273 } 3274 } 3275 return INVALID_OPERATION; 3276} 3277 3278status_t AudioFlinger::MixerThread::createAudioPatch_l(const struct audio_patch *patch, 3279 audio_patch_handle_t *handle) 3280{ 3281 AutoPark<FastMixer> park(mFastMixer); 3282 3283 status_t status = PlaybackThread::createAudioPatch_l(patch, handle); 3284 3285 return status; 3286} 3287 3288status_t AudioFlinger::PlaybackThread::createAudioPatch_l(const struct audio_patch *patch, 3289 audio_patch_handle_t *handle) 3290{ 3291 status_t status = NO_ERROR; 3292 3293 // store new device and send to effects 3294 audio_devices_t type = AUDIO_DEVICE_NONE; 3295 for (unsigned int i = 0; i < patch->num_sinks; i++) { 3296 type |= patch->sinks[i].ext.device.type; 3297 } 3298 3299#ifdef ADD_BATTERY_DATA 3300 // when changing the audio output device, call addBatteryData to notify 3301 // the change 3302 if (mOutDevice != type) { 3303 uint32_t params = 0; 3304 // check whether speaker is on 3305 if (type & AUDIO_DEVICE_OUT_SPEAKER) { 3306 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 3307 } 3308 3309 audio_devices_t deviceWithoutSpeaker 3310 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 3311 // check if any other device (except speaker) is on 3312 if (type & deviceWithoutSpeaker) { 3313 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 3314 } 3315 3316 if (params != 0) { 3317 addBatteryData(params); 3318 } 3319 } 3320#endif 3321 3322 for (size_t i = 0; i < mEffectChains.size(); i++) { 3323 mEffectChains[i]->setDevice_l(type); 3324 } 3325 3326 // mPrevOutDevice is the latest device set by createAudioPatch_l(). It is not set when 3327 // the thread is created so that the first patch creation triggers an ioConfigChanged callback 3328 bool configChanged = mPrevOutDevice != type; 3329 mOutDevice = type; 3330 mPatch = *patch; 3331 3332 if (mOutput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 3333 audio_hw_device_t *hwDevice = mOutput->audioHwDev->hwDevice(); 3334 status = hwDevice->create_audio_patch(hwDevice, 3335 patch->num_sources, 3336 patch->sources, 3337 patch->num_sinks, 3338 patch->sinks, 3339 handle); 3340 } else { 3341 char *address; 3342 if (strcmp(patch->sinks[0].ext.device.address, "") != 0) { 3343 //FIXME: we only support address on first sink with HAL version < 3.0 3344 address = audio_device_address_to_parameter( 3345 patch->sinks[0].ext.device.type, 3346 patch->sinks[0].ext.device.address); 3347 } else { 3348 address = (char *)calloc(1, 1); 3349 } 3350 AudioParameter param = AudioParameter(String8(address)); 3351 free(address); 3352 param.addInt(String8(AUDIO_PARAMETER_STREAM_ROUTING), (int)type); 3353 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3354 param.toString().string()); 3355 *handle = AUDIO_PATCH_HANDLE_NONE; 3356 } 3357 if (configChanged) { 3358 mPrevOutDevice = type; 3359 sendIoConfigEvent_l(AUDIO_OUTPUT_CONFIG_CHANGED); 3360 } 3361 return status; 3362} 3363 3364status_t AudioFlinger::MixerThread::releaseAudioPatch_l(const audio_patch_handle_t handle) 3365{ 3366 AutoPark<FastMixer> park(mFastMixer); 3367 3368 status_t status = PlaybackThread::releaseAudioPatch_l(handle); 3369 3370 return status; 3371} 3372 3373status_t AudioFlinger::PlaybackThread::releaseAudioPatch_l(const audio_patch_handle_t handle) 3374{ 3375 status_t status = NO_ERROR; 3376 3377 mOutDevice = AUDIO_DEVICE_NONE; 3378 3379 if (mOutput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 3380 audio_hw_device_t *hwDevice = mOutput->audioHwDev->hwDevice(); 3381 status = hwDevice->release_audio_patch(hwDevice, handle); 3382 } else { 3383 AudioParameter param; 3384 param.addInt(String8(AUDIO_PARAMETER_STREAM_ROUTING), 0); 3385 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3386 param.toString().string()); 3387 } 3388 return status; 3389} 3390 3391void AudioFlinger::PlaybackThread::addPatchTrack(const sp<PatchTrack>& track) 3392{ 3393 Mutex::Autolock _l(mLock); 3394 mTracks.add(track); 3395} 3396 3397void AudioFlinger::PlaybackThread::deletePatchTrack(const sp<PatchTrack>& track) 3398{ 3399 Mutex::Autolock _l(mLock); 3400 destroyTrack_l(track); 3401} 3402 3403void AudioFlinger::PlaybackThread::getAudioPortConfig(struct audio_port_config *config) 3404{ 3405 ThreadBase::getAudioPortConfig(config); 3406 config->role = AUDIO_PORT_ROLE_SOURCE; 3407 config->ext.mix.hw_module = mOutput->audioHwDev->handle(); 3408 config->ext.mix.usecase.stream = AUDIO_STREAM_DEFAULT; 3409} 3410 3411// ---------------------------------------------------------------------------- 3412 3413AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 3414 audio_io_handle_t id, audio_devices_t device, bool systemReady, type_t type) 3415 : PlaybackThread(audioFlinger, output, id, device, type, systemReady), 3416 // mAudioMixer below 3417 // mFastMixer below 3418 mFastMixerFutex(0), 3419 mMasterMono(false) 3420 // mOutputSink below 3421 // mPipeSink below 3422 // mNormalSink below 3423{ 3424 ALOGV("MixerThread() id=%d device=%#x type=%d", id, device, type); 3425 ALOGV("mSampleRate=%u, mChannelMask=%#x, mChannelCount=%u, mFormat=%d, mFrameSize=%zu, " 3426 "mFrameCount=%zu, mNormalFrameCount=%zu", 3427 mSampleRate, mChannelMask, mChannelCount, mFormat, mFrameSize, mFrameCount, 3428 mNormalFrameCount); 3429 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 3430 3431 if (type == DUPLICATING) { 3432 // The Duplicating thread uses the AudioMixer and delivers data to OutputTracks 3433 // (downstream MixerThreads) in DuplicatingThread::threadLoop_write(). 3434 // Do not create or use mFastMixer, mOutputSink, mPipeSink, or mNormalSink. 3435 return; 3436 } 3437 // create an NBAIO sink for the HAL output stream, and negotiate 3438 mOutputSink = new AudioStreamOutSink(output->stream); 3439 size_t numCounterOffers = 0; 3440 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount, mFormat)}; 3441#if !LOG_NDEBUG 3442 ssize_t index = 3443#else 3444 (void) 3445#endif 3446 mOutputSink->negotiate(offers, 1, NULL, numCounterOffers); 3447 ALOG_ASSERT(index == 0); 3448 3449 // initialize fast mixer depending on configuration 3450 bool initFastMixer; 3451 switch (kUseFastMixer) { 3452 case FastMixer_Never: 3453 initFastMixer = false; 3454 break; 3455 case FastMixer_Always: 3456 initFastMixer = true; 3457 break; 3458 case FastMixer_Static: 3459 case FastMixer_Dynamic: 3460 initFastMixer = mFrameCount < mNormalFrameCount; 3461 break; 3462 } 3463 if (initFastMixer) { 3464 audio_format_t fastMixerFormat; 3465 if (mMixerBufferEnabled && mEffectBufferEnabled) { 3466 fastMixerFormat = AUDIO_FORMAT_PCM_FLOAT; 3467 } else { 3468 fastMixerFormat = AUDIO_FORMAT_PCM_16_BIT; 3469 } 3470 if (mFormat != fastMixerFormat) { 3471 // change our Sink format to accept our intermediate precision 3472 mFormat = fastMixerFormat; 3473 free(mSinkBuffer); 3474 mFrameSize = mChannelCount * audio_bytes_per_sample(mFormat); 3475 const size_t sinkBufferSize = mNormalFrameCount * mFrameSize; 3476 (void)posix_memalign(&mSinkBuffer, 32, sinkBufferSize); 3477 } 3478 3479 // create a MonoPipe to connect our submix to FastMixer 3480 NBAIO_Format format = mOutputSink->format(); 3481#ifdef TEE_SINK 3482 NBAIO_Format origformat = format; 3483#endif 3484 // adjust format to match that of the Fast Mixer 3485 ALOGV("format changed from %d to %d", format.mFormat, fastMixerFormat); 3486 format.mFormat = fastMixerFormat; 3487 format.mFrameSize = audio_bytes_per_sample(format.mFormat) * format.mChannelCount; 3488 3489 // This pipe depth compensates for scheduling latency of the normal mixer thread. 3490 // When it wakes up after a maximum latency, it runs a few cycles quickly before 3491 // finally blocking. Note the pipe implementation rounds up the request to a power of 2. 3492 MonoPipe *monoPipe = new MonoPipe(mNormalFrameCount * 4, format, true /*writeCanBlock*/); 3493 const NBAIO_Format offers[1] = {format}; 3494 size_t numCounterOffers = 0; 3495#if !LOG_NDEBUG || defined(TEE_SINK) 3496 ssize_t index = 3497#else 3498 (void) 3499#endif 3500 monoPipe->negotiate(offers, 1, NULL, numCounterOffers); 3501 ALOG_ASSERT(index == 0); 3502 monoPipe->setAvgFrames((mScreenState & 1) ? 3503 (monoPipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 3504 mPipeSink = monoPipe; 3505 3506#ifdef TEE_SINK 3507 if (mTeeSinkOutputEnabled) { 3508 // create a Pipe to archive a copy of FastMixer's output for dumpsys 3509 Pipe *teeSink = new Pipe(mTeeSinkOutputFrames, origformat); 3510 const NBAIO_Format offers2[1] = {origformat}; 3511 numCounterOffers = 0; 3512 index = teeSink->negotiate(offers2, 1, NULL, numCounterOffers); 3513 ALOG_ASSERT(index == 0); 3514 mTeeSink = teeSink; 3515 PipeReader *teeSource = new PipeReader(*teeSink); 3516 numCounterOffers = 0; 3517 index = teeSource->negotiate(offers2, 1, NULL, numCounterOffers); 3518 ALOG_ASSERT(index == 0); 3519 mTeeSource = teeSource; 3520 } 3521#endif 3522 3523 // create fast mixer and configure it initially with just one fast track for our submix 3524 mFastMixer = new FastMixer(); 3525 FastMixerStateQueue *sq = mFastMixer->sq(); 3526#ifdef STATE_QUEUE_DUMP 3527 sq->setObserverDump(&mStateQueueObserverDump); 3528 sq->setMutatorDump(&mStateQueueMutatorDump); 3529#endif 3530 FastMixerState *state = sq->begin(); 3531 FastTrack *fastTrack = &state->mFastTracks[0]; 3532 // wrap the source side of the MonoPipe to make it an AudioBufferProvider 3533 fastTrack->mBufferProvider = new SourceAudioBufferProvider(new MonoPipeReader(monoPipe)); 3534 fastTrack->mVolumeProvider = NULL; 3535 fastTrack->mChannelMask = mChannelMask; // mPipeSink channel mask for audio to FastMixer 3536 fastTrack->mFormat = mFormat; // mPipeSink format for audio to FastMixer 3537 fastTrack->mGeneration++; 3538 state->mFastTracksGen++; 3539 state->mTrackMask = 1; 3540 // fast mixer will use the HAL output sink 3541 state->mOutputSink = mOutputSink.get(); 3542 state->mOutputSinkGen++; 3543 state->mFrameCount = mFrameCount; 3544 state->mCommand = FastMixerState::COLD_IDLE; 3545 // already done in constructor initialization list 3546 //mFastMixerFutex = 0; 3547 state->mColdFutexAddr = &mFastMixerFutex; 3548 state->mColdGen++; 3549 state->mDumpState = &mFastMixerDumpState; 3550#ifdef TEE_SINK 3551 state->mTeeSink = mTeeSink.get(); 3552#endif 3553 mFastMixerNBLogWriter = audioFlinger->newWriter_l(kFastMixerLogSize, "FastMixer"); 3554 state->mNBLogWriter = mFastMixerNBLogWriter.get(); 3555 sq->end(); 3556 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3557 3558 // start the fast mixer 3559 mFastMixer->run("FastMixer", PRIORITY_URGENT_AUDIO); 3560 pid_t tid = mFastMixer->getTid(); 3561 sendPrioConfigEvent(getpid_cached, tid, kPriorityFastMixer); 3562 3563#ifdef AUDIO_WATCHDOG 3564 // create and start the watchdog 3565 mAudioWatchdog = new AudioWatchdog(); 3566 mAudioWatchdog->setDump(&mAudioWatchdogDump); 3567 mAudioWatchdog->run("AudioWatchdog", PRIORITY_URGENT_AUDIO); 3568 tid = mAudioWatchdog->getTid(); 3569 sendPrioConfigEvent(getpid_cached, tid, kPriorityFastMixer); 3570#endif 3571 3572 } 3573 3574 switch (kUseFastMixer) { 3575 case FastMixer_Never: 3576 case FastMixer_Dynamic: 3577 mNormalSink = mOutputSink; 3578 break; 3579 case FastMixer_Always: 3580 mNormalSink = mPipeSink; 3581 break; 3582 case FastMixer_Static: 3583 mNormalSink = initFastMixer ? mPipeSink : mOutputSink; 3584 break; 3585 } 3586} 3587 3588AudioFlinger::MixerThread::~MixerThread() 3589{ 3590 if (mFastMixer != 0) { 3591 FastMixerStateQueue *sq = mFastMixer->sq(); 3592 FastMixerState *state = sq->begin(); 3593 if (state->mCommand == FastMixerState::COLD_IDLE) { 3594 int32_t old = android_atomic_inc(&mFastMixerFutex); 3595 if (old == -1) { 3596 (void) syscall(__NR_futex, &mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 3597 } 3598 } 3599 state->mCommand = FastMixerState::EXIT; 3600 sq->end(); 3601 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3602 mFastMixer->join(); 3603 // Though the fast mixer thread has exited, it's state queue is still valid. 3604 // We'll use that extract the final state which contains one remaining fast track 3605 // corresponding to our sub-mix. 3606 state = sq->begin(); 3607 ALOG_ASSERT(state->mTrackMask == 1); 3608 FastTrack *fastTrack = &state->mFastTracks[0]; 3609 ALOG_ASSERT(fastTrack->mBufferProvider != NULL); 3610 delete fastTrack->mBufferProvider; 3611 sq->end(false /*didModify*/); 3612 mFastMixer.clear(); 3613#ifdef AUDIO_WATCHDOG 3614 if (mAudioWatchdog != 0) { 3615 mAudioWatchdog->requestExit(); 3616 mAudioWatchdog->requestExitAndWait(); 3617 mAudioWatchdog.clear(); 3618 } 3619#endif 3620 } 3621 mAudioFlinger->unregisterWriter(mFastMixerNBLogWriter); 3622 delete mAudioMixer; 3623} 3624 3625 3626uint32_t AudioFlinger::MixerThread::correctLatency_l(uint32_t latency) const 3627{ 3628 if (mFastMixer != 0) { 3629 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 3630 latency += (pipe->getAvgFrames() * 1000) / mSampleRate; 3631 } 3632 return latency; 3633} 3634 3635 3636void AudioFlinger::MixerThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) 3637{ 3638 PlaybackThread::threadLoop_removeTracks(tracksToRemove); 3639} 3640 3641ssize_t AudioFlinger::MixerThread::threadLoop_write() 3642{ 3643 // FIXME we should only do one push per cycle; confirm this is true 3644 // Start the fast mixer if it's not already running 3645 if (mFastMixer != 0) { 3646 FastMixerStateQueue *sq = mFastMixer->sq(); 3647 FastMixerState *state = sq->begin(); 3648 if (state->mCommand != FastMixerState::MIX_WRITE && 3649 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)) { 3650 if (state->mCommand == FastMixerState::COLD_IDLE) { 3651 3652 // FIXME workaround for first HAL write being CPU bound on some devices 3653 ATRACE_BEGIN("write"); 3654 mOutput->write((char *)mSinkBuffer, 0); 3655 ATRACE_END(); 3656 3657 int32_t old = android_atomic_inc(&mFastMixerFutex); 3658 if (old == -1) { 3659 (void) syscall(__NR_futex, &mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 3660 } 3661#ifdef AUDIO_WATCHDOG 3662 if (mAudioWatchdog != 0) { 3663 mAudioWatchdog->resume(); 3664 } 3665#endif 3666 } 3667 state->mCommand = FastMixerState::MIX_WRITE; 3668#ifdef FAST_THREAD_STATISTICS 3669 mFastMixerDumpState.increaseSamplingN(mAudioFlinger->isLowRamDevice() ? 3670 FastThreadDumpState::kSamplingNforLowRamDevice : FastThreadDumpState::kSamplingN); 3671#endif 3672 sq->end(); 3673 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3674 if (kUseFastMixer == FastMixer_Dynamic) { 3675 mNormalSink = mPipeSink; 3676 } 3677 } else { 3678 sq->end(false /*didModify*/); 3679 } 3680 } 3681 return PlaybackThread::threadLoop_write(); 3682} 3683 3684void AudioFlinger::MixerThread::threadLoop_standby() 3685{ 3686 // Idle the fast mixer if it's currently running 3687 if (mFastMixer != 0) { 3688 FastMixerStateQueue *sq = mFastMixer->sq(); 3689 FastMixerState *state = sq->begin(); 3690 if (!(state->mCommand & FastMixerState::IDLE)) { 3691 state->mCommand = FastMixerState::COLD_IDLE; 3692 state->mColdFutexAddr = &mFastMixerFutex; 3693 state->mColdGen++; 3694 mFastMixerFutex = 0; 3695 sq->end(); 3696 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 3697 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3698 if (kUseFastMixer == FastMixer_Dynamic) { 3699 mNormalSink = mOutputSink; 3700 } 3701#ifdef AUDIO_WATCHDOG 3702 if (mAudioWatchdog != 0) { 3703 mAudioWatchdog->pause(); 3704 } 3705#endif 3706 } else { 3707 sq->end(false /*didModify*/); 3708 } 3709 } 3710 PlaybackThread::threadLoop_standby(); 3711} 3712 3713bool AudioFlinger::PlaybackThread::waitingAsyncCallback_l() 3714{ 3715 return false; 3716} 3717 3718bool AudioFlinger::PlaybackThread::shouldStandby_l() 3719{ 3720 return !mStandby; 3721} 3722 3723bool AudioFlinger::PlaybackThread::waitingAsyncCallback() 3724{ 3725 Mutex::Autolock _l(mLock); 3726 return waitingAsyncCallback_l(); 3727} 3728 3729// shared by MIXER and DIRECT, overridden by DUPLICATING 3730void AudioFlinger::PlaybackThread::threadLoop_standby() 3731{ 3732 ALOGV("Audio hardware entering standby, mixer %p, suspend count %d", this, mSuspended); 3733 mOutput->standby(); 3734 if (mUseAsyncWrite != 0) { 3735 // discard any pending drain or write ack by incrementing sequence 3736 mWriteAckSequence = (mWriteAckSequence + 2) & ~1; 3737 mDrainSequence = (mDrainSequence + 2) & ~1; 3738 ALOG_ASSERT(mCallbackThread != 0); 3739 mCallbackThread->setWriteBlocked(mWriteAckSequence); 3740 mCallbackThread->setDraining(mDrainSequence); 3741 } 3742 mHwPaused = false; 3743} 3744 3745void AudioFlinger::PlaybackThread::onAddNewTrack_l() 3746{ 3747 ALOGV("signal playback thread"); 3748 broadcast_l(); 3749} 3750 3751void AudioFlinger::MixerThread::threadLoop_mix() 3752{ 3753 // mix buffers... 3754 mAudioMixer->process(); 3755 mCurrentWriteLength = mSinkBufferSize; 3756 // increase sleep time progressively when application underrun condition clears. 3757 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 3758 // that a steady state of alternating ready/not ready conditions keeps the sleep time 3759 // such that we would underrun the audio HAL. 3760 if ((mSleepTimeUs == 0) && (sleepTimeShift > 0)) { 3761 sleepTimeShift--; 3762 } 3763 mSleepTimeUs = 0; 3764 mStandbyTimeNs = systemTime() + mStandbyDelayNs; 3765 //TODO: delay standby when effects have a tail 3766 3767} 3768 3769void AudioFlinger::MixerThread::threadLoop_sleepTime() 3770{ 3771 // If no tracks are ready, sleep once for the duration of an output 3772 // buffer size, then write 0s to the output 3773 if (mSleepTimeUs == 0) { 3774 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3775 mSleepTimeUs = mActiveSleepTimeUs >> sleepTimeShift; 3776 if (mSleepTimeUs < kMinThreadSleepTimeUs) { 3777 mSleepTimeUs = kMinThreadSleepTimeUs; 3778 } 3779 // reduce sleep time in case of consecutive application underruns to avoid 3780 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 3781 // duration we would end up writing less data than needed by the audio HAL if 3782 // the condition persists. 3783 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 3784 sleepTimeShift++; 3785 } 3786 } else { 3787 mSleepTimeUs = mIdleSleepTimeUs; 3788 } 3789 } else if (mBytesWritten != 0 || (mMixerStatus == MIXER_TRACKS_ENABLED)) { 3790 // clear out mMixerBuffer or mSinkBuffer, to ensure buffers are cleared 3791 // before effects processing or output. 3792 if (mMixerBufferValid) { 3793 memset(mMixerBuffer, 0, mMixerBufferSize); 3794 } else { 3795 memset(mSinkBuffer, 0, mSinkBufferSize); 3796 } 3797 mSleepTimeUs = 0; 3798 ALOGV_IF(mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED), 3799 "anticipated start"); 3800 } 3801 // TODO add standby time extension fct of effect tail 3802} 3803 3804// prepareTracks_l() must be called with ThreadBase::mLock held 3805AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 3806 Vector< sp<Track> > *tracksToRemove) 3807{ 3808 3809 mixer_state mixerStatus = MIXER_IDLE; 3810 // find out which tracks need to be processed 3811 size_t count = mActiveTracks.size(); 3812 size_t mixedTracks = 0; 3813 size_t tracksWithEffect = 0; 3814 // counts only _active_ fast tracks 3815 size_t fastTracks = 0; 3816 uint32_t resetMask = 0; // bit mask of fast tracks that need to be reset 3817 3818 float masterVolume = mMasterVolume; 3819 bool masterMute = mMasterMute; 3820 3821 if (masterMute) { 3822 masterVolume = 0; 3823 } 3824 // Delegate master volume control to effect in output mix effect chain if needed 3825 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 3826 if (chain != 0) { 3827 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 3828 chain->setVolume_l(&v, &v); 3829 masterVolume = (float)((v + (1 << 23)) >> 24); 3830 chain.clear(); 3831 } 3832 3833 // prepare a new state to push 3834 FastMixerStateQueue *sq = NULL; 3835 FastMixerState *state = NULL; 3836 bool didModify = false; 3837 FastMixerStateQueue::block_t block = FastMixerStateQueue::BLOCK_UNTIL_PUSHED; 3838 if (mFastMixer != 0) { 3839 sq = mFastMixer->sq(); 3840 state = sq->begin(); 3841 } 3842 3843 mMixerBufferValid = false; // mMixerBuffer has no valid data until appropriate tracks found. 3844 mEffectBufferValid = false; // mEffectBuffer has no valid data until tracks found. 3845 3846 for (size_t i=0 ; i<count ; i++) { 3847 const sp<Track> t = mActiveTracks[i].promote(); 3848 if (t == 0) { 3849 continue; 3850 } 3851 3852 // this const just means the local variable doesn't change 3853 Track* const track = t.get(); 3854 3855 // process fast tracks 3856 if (track->isFastTrack()) { 3857 3858 // It's theoretically possible (though unlikely) for a fast track to be created 3859 // and then removed within the same normal mix cycle. This is not a problem, as 3860 // the track never becomes active so it's fast mixer slot is never touched. 3861 // The converse, of removing an (active) track and then creating a new track 3862 // at the identical fast mixer slot within the same normal mix cycle, 3863 // is impossible because the slot isn't marked available until the end of each cycle. 3864 int j = track->mFastIndex; 3865 ALOG_ASSERT(0 < j && j < (int)FastMixerState::sMaxFastTracks); 3866 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << j))); 3867 FastTrack *fastTrack = &state->mFastTracks[j]; 3868 3869 // Determine whether the track is currently in underrun condition, 3870 // and whether it had a recent underrun. 3871 FastTrackDump *ftDump = &mFastMixerDumpState.mTracks[j]; 3872 FastTrackUnderruns underruns = ftDump->mUnderruns; 3873 uint32_t recentFull = (underruns.mBitFields.mFull - 3874 track->mObservedUnderruns.mBitFields.mFull) & UNDERRUN_MASK; 3875 uint32_t recentPartial = (underruns.mBitFields.mPartial - 3876 track->mObservedUnderruns.mBitFields.mPartial) & UNDERRUN_MASK; 3877 uint32_t recentEmpty = (underruns.mBitFields.mEmpty - 3878 track->mObservedUnderruns.mBitFields.mEmpty) & UNDERRUN_MASK; 3879 uint32_t recentUnderruns = recentPartial + recentEmpty; 3880 track->mObservedUnderruns = underruns; 3881 // don't count underruns that occur while stopping or pausing 3882 // or stopped which can occur when flush() is called while active 3883 if (!(track->isStopping() || track->isPausing() || track->isStopped()) && 3884 recentUnderruns > 0) { 3885 // FIXME fast mixer will pull & mix partial buffers, but we count as a full underrun 3886 track->mAudioTrackServerProxy->tallyUnderrunFrames(recentUnderruns * mFrameCount); 3887 } else { 3888 track->mAudioTrackServerProxy->tallyUnderrunFrames(0); 3889 } 3890 3891 // This is similar to the state machine for normal tracks, 3892 // with a few modifications for fast tracks. 3893 bool isActive = true; 3894 switch (track->mState) { 3895 case TrackBase::STOPPING_1: 3896 // track stays active in STOPPING_1 state until first underrun 3897 if (recentUnderruns > 0 || track->isTerminated()) { 3898 track->mState = TrackBase::STOPPING_2; 3899 } 3900 break; 3901 case TrackBase::PAUSING: 3902 // ramp down is not yet implemented 3903 track->setPaused(); 3904 break; 3905 case TrackBase::RESUMING: 3906 // ramp up is not yet implemented 3907 track->mState = TrackBase::ACTIVE; 3908 break; 3909 case TrackBase::ACTIVE: 3910 if (recentFull > 0 || recentPartial > 0) { 3911 // track has provided at least some frames recently: reset retry count 3912 track->mRetryCount = kMaxTrackRetries; 3913 } 3914 if (recentUnderruns == 0) { 3915 // no recent underruns: stay active 3916 break; 3917 } 3918 // there has recently been an underrun of some kind 3919 if (track->sharedBuffer() == 0) { 3920 // were any of the recent underruns "empty" (no frames available)? 3921 if (recentEmpty == 0) { 3922 // no, then ignore the partial underruns as they are allowed indefinitely 3923 break; 3924 } 3925 // there has recently been an "empty" underrun: decrement the retry counter 3926 if (--(track->mRetryCount) > 0) { 3927 break; 3928 } 3929 // indicate to client process that the track was disabled because of underrun; 3930 // it will then automatically call start() when data is available 3931 track->disable(); 3932 // remove from active list, but state remains ACTIVE [confusing but true] 3933 isActive = false; 3934 break; 3935 } 3936 // fall through 3937 case TrackBase::STOPPING_2: 3938 case TrackBase::PAUSED: 3939 case TrackBase::STOPPED: 3940 case TrackBase::FLUSHED: // flush() while active 3941 // Check for presentation complete if track is inactive 3942 // We have consumed all the buffers of this track. 3943 // This would be incomplete if we auto-paused on underrun 3944 { 3945 size_t audioHALFrames = 3946 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 3947 int64_t framesWritten = mBytesWritten / mFrameSize; 3948 if (!(mStandby || track->presentationComplete(framesWritten, audioHALFrames))) { 3949 // track stays in active list until presentation is complete 3950 break; 3951 } 3952 } 3953 if (track->isStopping_2()) { 3954 track->mState = TrackBase::STOPPED; 3955 } 3956 if (track->isStopped()) { 3957 // Can't reset directly, as fast mixer is still polling this track 3958 // track->reset(); 3959 // So instead mark this track as needing to be reset after push with ack 3960 resetMask |= 1 << i; 3961 } 3962 isActive = false; 3963 break; 3964 case TrackBase::IDLE: 3965 default: 3966 LOG_ALWAYS_FATAL("unexpected track state %d", track->mState); 3967 } 3968 3969 if (isActive) { 3970 // was it previously inactive? 3971 if (!(state->mTrackMask & (1 << j))) { 3972 ExtendedAudioBufferProvider *eabp = track; 3973 VolumeProvider *vp = track; 3974 fastTrack->mBufferProvider = eabp; 3975 fastTrack->mVolumeProvider = vp; 3976 fastTrack->mChannelMask = track->mChannelMask; 3977 fastTrack->mFormat = track->mFormat; 3978 fastTrack->mGeneration++; 3979 state->mTrackMask |= 1 << j; 3980 didModify = true; 3981 // no acknowledgement required for newly active tracks 3982 } 3983 // cache the combined master volume and stream type volume for fast mixer; this 3984 // lacks any synchronization or barrier so VolumeProvider may read a stale value 3985 track->mCachedVolume = masterVolume * mStreamTypes[track->streamType()].volume; 3986 ++fastTracks; 3987 } else { 3988 // was it previously active? 3989 if (state->mTrackMask & (1 << j)) { 3990 fastTrack->mBufferProvider = NULL; 3991 fastTrack->mGeneration++; 3992 state->mTrackMask &= ~(1 << j); 3993 didModify = true; 3994 // If any fast tracks were removed, we must wait for acknowledgement 3995 // because we're about to decrement the last sp<> on those tracks. 3996 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3997 } else { 3998 LOG_ALWAYS_FATAL("fast track %d should have been active; " 3999 "mState=%d, mTrackMask=%#x, recentUnderruns=%u, isShared=%d", 4000 j, track->mState, state->mTrackMask, recentUnderruns, 4001 track->sharedBuffer() != 0); 4002 } 4003 tracksToRemove->add(track); 4004 // Avoids a misleading display in dumpsys 4005 track->mObservedUnderruns.mBitFields.mMostRecent = UNDERRUN_FULL; 4006 } 4007 continue; 4008 } 4009 4010 { // local variable scope to avoid goto warning 4011 4012 audio_track_cblk_t* cblk = track->cblk(); 4013 4014 // The first time a track is added we wait 4015 // for all its buffers to be filled before processing it 4016 int name = track->name(); 4017 // make sure that we have enough frames to mix one full buffer. 4018 // enforce this condition only once to enable draining the buffer in case the client 4019 // app does not call stop() and relies on underrun to stop: 4020 // hence the test on (mMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 4021 // during last round 4022 size_t desiredFrames; 4023 const uint32_t sampleRate = track->mAudioTrackServerProxy->getSampleRate(); 4024 AudioPlaybackRate playbackRate = track->mAudioTrackServerProxy->getPlaybackRate(); 4025 4026 desiredFrames = sourceFramesNeededWithTimestretch( 4027 sampleRate, mNormalFrameCount, mSampleRate, playbackRate.mSpeed); 4028 // TODO: ONLY USED FOR LEGACY RESAMPLERS, remove when they are removed. 4029 // add frames already consumed but not yet released by the resampler 4030 // because mAudioTrackServerProxy->framesReady() will include these frames 4031 desiredFrames += mAudioMixer->getUnreleasedFrames(track->name()); 4032 4033 uint32_t minFrames = 1; 4034 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() && 4035 (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) { 4036 minFrames = desiredFrames; 4037 } 4038 4039 size_t framesReady = track->framesReady(); 4040 if (ATRACE_ENABLED()) { 4041 // I wish we had formatted trace names 4042 char traceName[16]; 4043 strcpy(traceName, "nRdy"); 4044 int name = track->name(); 4045 if (AudioMixer::TRACK0 <= name && 4046 name < (int) (AudioMixer::TRACK0 + AudioMixer::MAX_NUM_TRACKS)) { 4047 name -= AudioMixer::TRACK0; 4048 traceName[4] = (name / 10) + '0'; 4049 traceName[5] = (name % 10) + '0'; 4050 } else { 4051 traceName[4] = '?'; 4052 traceName[5] = '?'; 4053 } 4054 traceName[6] = '\0'; 4055 ATRACE_INT(traceName, framesReady); 4056 } 4057 if ((framesReady >= minFrames) && track->isReady() && 4058 !track->isPaused() && !track->isTerminated()) 4059 { 4060 ALOGVV("track %d s=%08x [OK] on thread %p", name, cblk->mServer, this); 4061 4062 mixedTracks++; 4063 4064 // track->mainBuffer() != mSinkBuffer or mMixerBuffer means 4065 // there is an effect chain connected to the track 4066 chain.clear(); 4067 if (track->mainBuffer() != mSinkBuffer && 4068 track->mainBuffer() != mMixerBuffer) { 4069 if (mEffectBufferEnabled) { 4070 mEffectBufferValid = true; // Later can set directly. 4071 } 4072 chain = getEffectChain_l(track->sessionId()); 4073 // Delegate volume control to effect in track effect chain if needed 4074 if (chain != 0) { 4075 tracksWithEffect++; 4076 } else { 4077 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on " 4078 "session %d", 4079 name, track->sessionId()); 4080 } 4081 } 4082 4083 4084 int param = AudioMixer::VOLUME; 4085 if (track->mFillingUpStatus == Track::FS_FILLED) { 4086 // no ramp for the first volume setting 4087 track->mFillingUpStatus = Track::FS_ACTIVE; 4088 if (track->mState == TrackBase::RESUMING) { 4089 track->mState = TrackBase::ACTIVE; 4090 param = AudioMixer::RAMP_VOLUME; 4091 } 4092 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 4093 // FIXME should not make a decision based on mServer 4094 } else if (cblk->mServer != 0) { 4095 // If the track is stopped before the first frame was mixed, 4096 // do not apply ramp 4097 param = AudioMixer::RAMP_VOLUME; 4098 } 4099 4100 // compute volume for this track 4101 uint32_t vl, vr; // in U8.24 integer format 4102 float vlf, vrf, vaf; // in [0.0, 1.0] float format 4103 if (track->isPausing() || mStreamTypes[track->streamType()].mute) { 4104 vl = vr = 0; 4105 vlf = vrf = vaf = 0.; 4106 if (track->isPausing()) { 4107 track->setPaused(); 4108 } 4109 } else { 4110 4111 // read original volumes with volume control 4112 float typeVolume = mStreamTypes[track->streamType()].volume; 4113 float v = masterVolume * typeVolume; 4114 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 4115 gain_minifloat_packed_t vlr = proxy->getVolumeLR(); 4116 vlf = float_from_gain(gain_minifloat_unpack_left(vlr)); 4117 vrf = float_from_gain(gain_minifloat_unpack_right(vlr)); 4118 // track volumes come from shared memory, so can't be trusted and must be clamped 4119 if (vlf > GAIN_FLOAT_UNITY) { 4120 ALOGV("Track left volume out of range: %.3g", vlf); 4121 vlf = GAIN_FLOAT_UNITY; 4122 } 4123 if (vrf > GAIN_FLOAT_UNITY) { 4124 ALOGV("Track right volume out of range: %.3g", vrf); 4125 vrf = GAIN_FLOAT_UNITY; 4126 } 4127 // now apply the master volume and stream type volume 4128 vlf *= v; 4129 vrf *= v; 4130 // assuming master volume and stream type volume each go up to 1.0, 4131 // then derive vl and vr as U8.24 versions for the effect chain 4132 const float scaleto8_24 = MAX_GAIN_INT * MAX_GAIN_INT; 4133 vl = (uint32_t) (scaleto8_24 * vlf); 4134 vr = (uint32_t) (scaleto8_24 * vrf); 4135 // vl and vr are now in U8.24 format 4136 uint16_t sendLevel = proxy->getSendLevel_U4_12(); 4137 // send level comes from shared memory and so may be corrupt 4138 if (sendLevel > MAX_GAIN_INT) { 4139 ALOGV("Track send level out of range: %04X", sendLevel); 4140 sendLevel = MAX_GAIN_INT; 4141 } 4142 // vaf is represented as [0.0, 1.0] float by rescaling sendLevel 4143 vaf = v * sendLevel * (1. / MAX_GAIN_INT); 4144 } 4145 4146 // Delegate volume control to effect in track effect chain if needed 4147 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 4148 // Do not ramp volume if volume is controlled by effect 4149 param = AudioMixer::VOLUME; 4150 // Update remaining floating point volume levels 4151 vlf = (float)vl / (1 << 24); 4152 vrf = (float)vr / (1 << 24); 4153 track->mHasVolumeController = true; 4154 } else { 4155 // force no volume ramp when volume controller was just disabled or removed 4156 // from effect chain to avoid volume spike 4157 if (track->mHasVolumeController) { 4158 param = AudioMixer::VOLUME; 4159 } 4160 track->mHasVolumeController = false; 4161 } 4162 4163 // XXX: these things DON'T need to be done each time 4164 mAudioMixer->setBufferProvider(name, track); 4165 mAudioMixer->enable(name); 4166 4167 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, &vlf); 4168 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, &vrf); 4169 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, &vaf); 4170 mAudioMixer->setParameter( 4171 name, 4172 AudioMixer::TRACK, 4173 AudioMixer::FORMAT, (void *)track->format()); 4174 mAudioMixer->setParameter( 4175 name, 4176 AudioMixer::TRACK, 4177 AudioMixer::CHANNEL_MASK, (void *)(uintptr_t)track->channelMask()); 4178 mAudioMixer->setParameter( 4179 name, 4180 AudioMixer::TRACK, 4181 AudioMixer::MIXER_CHANNEL_MASK, (void *)(uintptr_t)mChannelMask); 4182 // limit track sample rate to 2 x output sample rate, which changes at re-configuration 4183 uint32_t maxSampleRate = mSampleRate * AUDIO_RESAMPLER_DOWN_RATIO_MAX; 4184 uint32_t reqSampleRate = track->mAudioTrackServerProxy->getSampleRate(); 4185 if (reqSampleRate == 0) { 4186 reqSampleRate = mSampleRate; 4187 } else if (reqSampleRate > maxSampleRate) { 4188 reqSampleRate = maxSampleRate; 4189 } 4190 mAudioMixer->setParameter( 4191 name, 4192 AudioMixer::RESAMPLE, 4193 AudioMixer::SAMPLE_RATE, 4194 (void *)(uintptr_t)reqSampleRate); 4195 4196 AudioPlaybackRate playbackRate = track->mAudioTrackServerProxy->getPlaybackRate(); 4197 mAudioMixer->setParameter( 4198 name, 4199 AudioMixer::TIMESTRETCH, 4200 AudioMixer::PLAYBACK_RATE, 4201 &playbackRate); 4202 4203 /* 4204 * Select the appropriate output buffer for the track. 4205 * 4206 * Tracks with effects go into their own effects chain buffer 4207 * and from there into either mEffectBuffer or mSinkBuffer. 4208 * 4209 * Other tracks can use mMixerBuffer for higher precision 4210 * channel accumulation. If this buffer is enabled 4211 * (mMixerBufferEnabled true), then selected tracks will accumulate 4212 * into it. 4213 * 4214 */ 4215 if (mMixerBufferEnabled 4216 && (track->mainBuffer() == mSinkBuffer 4217 || track->mainBuffer() == mMixerBuffer)) { 4218 mAudioMixer->setParameter( 4219 name, 4220 AudioMixer::TRACK, 4221 AudioMixer::MIXER_FORMAT, (void *)mMixerBufferFormat); 4222 mAudioMixer->setParameter( 4223 name, 4224 AudioMixer::TRACK, 4225 AudioMixer::MAIN_BUFFER, (void *)mMixerBuffer); 4226 // TODO: override track->mainBuffer()? 4227 mMixerBufferValid = true; 4228 } else { 4229 mAudioMixer->setParameter( 4230 name, 4231 AudioMixer::TRACK, 4232 AudioMixer::MIXER_FORMAT, (void *)AUDIO_FORMAT_PCM_16_BIT); 4233 mAudioMixer->setParameter( 4234 name, 4235 AudioMixer::TRACK, 4236 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 4237 } 4238 mAudioMixer->setParameter( 4239 name, 4240 AudioMixer::TRACK, 4241 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 4242 4243 // reset retry count 4244 track->mRetryCount = kMaxTrackRetries; 4245 4246 // If one track is ready, set the mixer ready if: 4247 // - the mixer was not ready during previous round OR 4248 // - no other track is not ready 4249 if (mMixerStatusIgnoringFastTracks != MIXER_TRACKS_READY || 4250 mixerStatus != MIXER_TRACKS_ENABLED) { 4251 mixerStatus = MIXER_TRACKS_READY; 4252 } 4253 } else { 4254 if (framesReady < desiredFrames && !track->isStopped() && !track->isPaused()) { 4255 ALOGV("track(%p) underrun, framesReady(%zu) < framesDesired(%zd)", 4256 track, framesReady, desiredFrames); 4257 track->mAudioTrackServerProxy->tallyUnderrunFrames(desiredFrames); 4258 } else { 4259 track->mAudioTrackServerProxy->tallyUnderrunFrames(0); 4260 } 4261 4262 // clear effect chain input buffer if an active track underruns to avoid sending 4263 // previous audio buffer again to effects 4264 chain = getEffectChain_l(track->sessionId()); 4265 if (chain != 0) { 4266 chain->clearInputBuffer(); 4267 } 4268 4269 ALOGVV("track %d s=%08x [NOT READY] on thread %p", name, cblk->mServer, this); 4270 if ((track->sharedBuffer() != 0) || track->isTerminated() || 4271 track->isStopped() || track->isPaused()) { 4272 // We have consumed all the buffers of this track. 4273 // Remove it from the list of active tracks. 4274 // TODO: use actual buffer filling status instead of latency when available from 4275 // audio HAL 4276 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 4277 int64_t framesWritten = mBytesWritten / mFrameSize; 4278 if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) { 4279 if (track->isStopped()) { 4280 track->reset(); 4281 } 4282 tracksToRemove->add(track); 4283 } 4284 } else { 4285 // No buffers for this track. Give it a few chances to 4286 // fill a buffer, then remove it from active list. 4287 if (--(track->mRetryCount) <= 0) { 4288 ALOGI("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 4289 tracksToRemove->add(track); 4290 // indicate to client process that the track was disabled because of underrun; 4291 // it will then automatically call start() when data is available 4292 track->disable(); 4293 // If one track is not ready, mark the mixer also not ready if: 4294 // - the mixer was ready during previous round OR 4295 // - no other track is ready 4296 } else if (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY || 4297 mixerStatus != MIXER_TRACKS_READY) { 4298 mixerStatus = MIXER_TRACKS_ENABLED; 4299 } 4300 } 4301 mAudioMixer->disable(name); 4302 } 4303 4304 } // local variable scope to avoid goto warning 4305 4306 } 4307 4308 // Push the new FastMixer state if necessary 4309 bool pauseAudioWatchdog = false; 4310 if (didModify) { 4311 state->mFastTracksGen++; 4312 // if the fast mixer was active, but now there are no fast tracks, then put it in cold idle 4313 if (kUseFastMixer == FastMixer_Dynamic && 4314 state->mCommand == FastMixerState::MIX_WRITE && state->mTrackMask <= 1) { 4315 state->mCommand = FastMixerState::COLD_IDLE; 4316 state->mColdFutexAddr = &mFastMixerFutex; 4317 state->mColdGen++; 4318 mFastMixerFutex = 0; 4319 if (kUseFastMixer == FastMixer_Dynamic) { 4320 mNormalSink = mOutputSink; 4321 } 4322 // If we go into cold idle, need to wait for acknowledgement 4323 // so that fast mixer stops doing I/O. 4324 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 4325 pauseAudioWatchdog = true; 4326 } 4327 } 4328 if (sq != NULL) { 4329 sq->end(didModify); 4330 sq->push(block); 4331 } 4332#ifdef AUDIO_WATCHDOG 4333 if (pauseAudioWatchdog && mAudioWatchdog != 0) { 4334 mAudioWatchdog->pause(); 4335 } 4336#endif 4337 4338 // Now perform the deferred reset on fast tracks that have stopped 4339 while (resetMask != 0) { 4340 size_t i = __builtin_ctz(resetMask); 4341 ALOG_ASSERT(i < count); 4342 resetMask &= ~(1 << i); 4343 sp<Track> t = mActiveTracks[i].promote(); 4344 if (t == 0) { 4345 continue; 4346 } 4347 Track* track = t.get(); 4348 ALOG_ASSERT(track->isFastTrack() && track->isStopped()); 4349 track->reset(); 4350 } 4351 4352 // remove all the tracks that need to be... 4353 removeTracks_l(*tracksToRemove); 4354 4355 if (getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX) != 0) { 4356 mEffectBufferValid = true; 4357 } 4358 4359 if (mEffectBufferValid) { 4360 // as long as there are effects we should clear the effects buffer, to avoid 4361 // passing a non-clean buffer to the effect chain 4362 memset(mEffectBuffer, 0, mEffectBufferSize); 4363 } 4364 // sink or mix buffer must be cleared if all tracks are connected to an 4365 // effect chain as in this case the mixer will not write to the sink or mix buffer 4366 // and track effects will accumulate into it 4367 if ((mBytesRemaining == 0) && ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 4368 (mixedTracks == 0 && fastTracks > 0))) { 4369 // FIXME as a performance optimization, should remember previous zero status 4370 if (mMixerBufferValid) { 4371 memset(mMixerBuffer, 0, mMixerBufferSize); 4372 // TODO: In testing, mSinkBuffer below need not be cleared because 4373 // the PlaybackThread::threadLoop() copies mMixerBuffer into mSinkBuffer 4374 // after mixing. 4375 // 4376 // To enforce this guarantee: 4377 // ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 4378 // (mixedTracks == 0 && fastTracks > 0)) 4379 // must imply MIXER_TRACKS_READY. 4380 // Later, we may clear buffers regardless, and skip much of this logic. 4381 } 4382 // FIXME as a performance optimization, should remember previous zero status 4383 memset(mSinkBuffer, 0, mNormalFrameCount * mFrameSize); 4384 } 4385 4386 // if any fast tracks, then status is ready 4387 mMixerStatusIgnoringFastTracks = mixerStatus; 4388 if (fastTracks > 0) { 4389 mixerStatus = MIXER_TRACKS_READY; 4390 } 4391 return mixerStatus; 4392} 4393 4394// getTrackName_l() must be called with ThreadBase::mLock held 4395int AudioFlinger::MixerThread::getTrackName_l(audio_channel_mask_t channelMask, 4396 audio_format_t format, audio_session_t sessionId) 4397{ 4398 return mAudioMixer->getTrackName(channelMask, format, sessionId); 4399} 4400 4401// deleteTrackName_l() must be called with ThreadBase::mLock held 4402void AudioFlinger::MixerThread::deleteTrackName_l(int name) 4403{ 4404 ALOGV("remove track (%d) and delete from mixer", name); 4405 mAudioMixer->deleteTrackName(name); 4406} 4407 4408// checkForNewParameter_l() must be called with ThreadBase::mLock held 4409bool AudioFlinger::MixerThread::checkForNewParameter_l(const String8& keyValuePair, 4410 status_t& status) 4411{ 4412 bool reconfig = false; 4413 bool a2dpDeviceChanged = false; 4414 4415 status = NO_ERROR; 4416 4417 AutoPark<FastMixer> park(mFastMixer); 4418 4419 AudioParameter param = AudioParameter(keyValuePair); 4420 int value; 4421 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 4422 reconfig = true; 4423 } 4424 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 4425 if (!isValidPcmSinkFormat((audio_format_t) value)) { 4426 status = BAD_VALUE; 4427 } else { 4428 // no need to save value, since it's constant 4429 reconfig = true; 4430 } 4431 } 4432 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 4433 if (!isValidPcmSinkChannelMask((audio_channel_mask_t) value)) { 4434 status = BAD_VALUE; 4435 } else { 4436 // no need to save value, since it's constant 4437 reconfig = true; 4438 } 4439 } 4440 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 4441 // do not accept frame count changes if tracks are open as the track buffer 4442 // size depends on frame count and correct behavior would not be guaranteed 4443 // if frame count is changed after track creation 4444 if (!mTracks.isEmpty()) { 4445 status = INVALID_OPERATION; 4446 } else { 4447 reconfig = true; 4448 } 4449 } 4450 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 4451#ifdef ADD_BATTERY_DATA 4452 // when changing the audio output device, call addBatteryData to notify 4453 // the change 4454 if (mOutDevice != value) { 4455 uint32_t params = 0; 4456 // check whether speaker is on 4457 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 4458 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 4459 } 4460 4461 audio_devices_t deviceWithoutSpeaker 4462 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 4463 // check if any other device (except speaker) is on 4464 if (value & deviceWithoutSpeaker) { 4465 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 4466 } 4467 4468 if (params != 0) { 4469 addBatteryData(params); 4470 } 4471 } 4472#endif 4473 4474 // forward device change to effects that have requested to be 4475 // aware of attached audio device. 4476 if (value != AUDIO_DEVICE_NONE) { 4477 a2dpDeviceChanged = 4478 (mOutDevice & AUDIO_DEVICE_OUT_ALL_A2DP) != (value & AUDIO_DEVICE_OUT_ALL_A2DP); 4479 mOutDevice = value; 4480 for (size_t i = 0; i < mEffectChains.size(); i++) { 4481 mEffectChains[i]->setDevice_l(mOutDevice); 4482 } 4483 } 4484 } 4485 4486 if (status == NO_ERROR) { 4487 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 4488 keyValuePair.string()); 4489 if (!mStandby && status == INVALID_OPERATION) { 4490 mOutput->standby(); 4491 mStandby = true; 4492 mBytesWritten = 0; 4493 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 4494 keyValuePair.string()); 4495 } 4496 if (status == NO_ERROR && reconfig) { 4497 readOutputParameters_l(); 4498 delete mAudioMixer; 4499 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 4500 for (size_t i = 0; i < mTracks.size() ; i++) { 4501 int name = getTrackName_l(mTracks[i]->mChannelMask, 4502 mTracks[i]->mFormat, mTracks[i]->mSessionId); 4503 if (name < 0) { 4504 break; 4505 } 4506 mTracks[i]->mName = name; 4507 } 4508 sendIoConfigEvent_l(AUDIO_OUTPUT_CONFIG_CHANGED); 4509 } 4510 } 4511 4512 return reconfig || a2dpDeviceChanged; 4513} 4514 4515 4516void AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 4517{ 4518 PlaybackThread::dumpInternals(fd, args); 4519 dprintf(fd, " Thread throttle time (msecs): %u\n", mThreadThrottleTimeMs); 4520 dprintf(fd, " AudioMixer tracks: 0x%08x\n", mAudioMixer->trackNames()); 4521 dprintf(fd, " Master mono: %s\n", mMasterMono ? "on" : "off"); 4522 4523 // Make a non-atomic copy of fast mixer dump state so it won't change underneath us 4524 // while we are dumping it. It may be inconsistent, but it won't mutate! 4525 // This is a large object so we place it on the heap. 4526 // FIXME 25972958: Need an intelligent copy constructor that does not touch unused pages. 4527 const FastMixerDumpState *copy = new FastMixerDumpState(mFastMixerDumpState); 4528 copy->dump(fd); 4529 delete copy; 4530 4531#ifdef STATE_QUEUE_DUMP 4532 // Similar for state queue 4533 StateQueueObserverDump observerCopy = mStateQueueObserverDump; 4534 observerCopy.dump(fd); 4535 StateQueueMutatorDump mutatorCopy = mStateQueueMutatorDump; 4536 mutatorCopy.dump(fd); 4537#endif 4538 4539#ifdef TEE_SINK 4540 // Write the tee output to a .wav file 4541 dumpTee(fd, mTeeSource, mId); 4542#endif 4543 4544#ifdef AUDIO_WATCHDOG 4545 if (mAudioWatchdog != 0) { 4546 // Make a non-atomic copy of audio watchdog dump so it won't change underneath us 4547 AudioWatchdogDump wdCopy = mAudioWatchdogDump; 4548 wdCopy.dump(fd); 4549 } 4550#endif 4551} 4552 4553uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const 4554{ 4555 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000) / 2; 4556} 4557 4558uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() const 4559{ 4560 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000); 4561} 4562 4563void AudioFlinger::MixerThread::cacheParameters_l() 4564{ 4565 PlaybackThread::cacheParameters_l(); 4566 4567 // FIXME: Relaxed timing because of a certain device that can't meet latency 4568 // Should be reduced to 2x after the vendor fixes the driver issue 4569 // increase threshold again due to low power audio mode. The way this warning 4570 // threshold is calculated and its usefulness should be reconsidered anyway. 4571 maxPeriod = seconds(mNormalFrameCount) / mSampleRate * 15; 4572} 4573 4574// ---------------------------------------------------------------------------- 4575 4576AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 4577 AudioStreamOut* output, audio_io_handle_t id, audio_devices_t device, bool systemReady, 4578 uint32_t bitRate) 4579 : PlaybackThread(audioFlinger, output, id, device, DIRECT, systemReady, bitRate) 4580 // mLeftVolFloat, mRightVolFloat 4581{ 4582} 4583 4584AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 4585 AudioStreamOut* output, audio_io_handle_t id, uint32_t device, 4586 ThreadBase::type_t type, bool systemReady, uint32_t bitRate) 4587 : PlaybackThread(audioFlinger, output, id, device, type, systemReady, bitRate) 4588 // mLeftVolFloat, mRightVolFloat 4589{ 4590} 4591 4592AudioFlinger::DirectOutputThread::~DirectOutputThread() 4593{ 4594} 4595 4596void AudioFlinger::DirectOutputThread::processVolume_l(Track *track, bool lastTrack) 4597{ 4598 float left, right; 4599 4600 if (mMasterMute || mStreamTypes[track->streamType()].mute) { 4601 left = right = 0; 4602 } else { 4603 float typeVolume = mStreamTypes[track->streamType()].volume; 4604 float v = mMasterVolume * typeVolume; 4605 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 4606 gain_minifloat_packed_t vlr = proxy->getVolumeLR(); 4607 left = float_from_gain(gain_minifloat_unpack_left(vlr)); 4608 if (left > GAIN_FLOAT_UNITY) { 4609 left = GAIN_FLOAT_UNITY; 4610 } 4611 left *= v; 4612 right = float_from_gain(gain_minifloat_unpack_right(vlr)); 4613 if (right > GAIN_FLOAT_UNITY) { 4614 right = GAIN_FLOAT_UNITY; 4615 } 4616 right *= v; 4617 } 4618 4619 if (lastTrack) { 4620 if (left != mLeftVolFloat || right != mRightVolFloat) { 4621 mLeftVolFloat = left; 4622 mRightVolFloat = right; 4623 4624 // Convert volumes from float to 8.24 4625 uint32_t vl = (uint32_t)(left * (1 << 24)); 4626 uint32_t vr = (uint32_t)(right * (1 << 24)); 4627 4628 // Delegate volume control to effect in track effect chain if needed 4629 // only one effect chain can be present on DirectOutputThread, so if 4630 // there is one, the track is connected to it 4631 if (!mEffectChains.isEmpty()) { 4632 mEffectChains[0]->setVolume_l(&vl, &vr); 4633 left = (float)vl / (1 << 24); 4634 right = (float)vr / (1 << 24); 4635 } 4636 if (mOutput->stream->set_volume) { 4637 mOutput->stream->set_volume(mOutput->stream, left, right); 4638 } 4639 } 4640 } 4641} 4642 4643void AudioFlinger::DirectOutputThread::onAddNewTrack_l() 4644{ 4645 sp<Track> previousTrack = mPreviousTrack.promote(); 4646 sp<Track> latestTrack = mLatestActiveTrack.promote(); 4647 4648 if (previousTrack != 0 && latestTrack != 0) { 4649 if (mType == DIRECT) { 4650 if (previousTrack.get() != latestTrack.get()) { 4651 mFlushPending = true; 4652 } 4653 } else /* mType == OFFLOAD */ { 4654 if (previousTrack->sessionId() != latestTrack->sessionId()) { 4655 mFlushPending = true; 4656 } 4657 } 4658 } 4659 PlaybackThread::onAddNewTrack_l(); 4660} 4661 4662AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l( 4663 Vector< sp<Track> > *tracksToRemove 4664) 4665{ 4666 size_t count = mActiveTracks.size(); 4667 mixer_state mixerStatus = MIXER_IDLE; 4668 bool doHwPause = false; 4669 bool doHwResume = false; 4670 4671 // find out which tracks need to be processed 4672 for (size_t i = 0; i < count; i++) { 4673 sp<Track> t = mActiveTracks[i].promote(); 4674 // The track died recently 4675 if (t == 0) { 4676 continue; 4677 } 4678 4679 if (t->isInvalid()) { 4680 ALOGW("An invalidated track shouldn't be in active list"); 4681 tracksToRemove->add(t); 4682 continue; 4683 } 4684 4685 Track* const track = t.get(); 4686#ifdef VERY_VERY_VERBOSE_LOGGING 4687 audio_track_cblk_t* cblk = track->cblk(); 4688#endif 4689 // Only consider last track started for volume and mixer state control. 4690 // In theory an older track could underrun and restart after the new one starts 4691 // but as we only care about the transition phase between two tracks on a 4692 // direct output, it is not a problem to ignore the underrun case. 4693 sp<Track> l = mLatestActiveTrack.promote(); 4694 bool last = l.get() == track; 4695 4696 if (track->isPausing()) { 4697 track->setPaused(); 4698 if (mHwSupportsPause && last && !mHwPaused) { 4699 doHwPause = true; 4700 mHwPaused = true; 4701 } 4702 tracksToRemove->add(track); 4703 } else if (track->isFlushPending()) { 4704 track->flushAck(); 4705 if (last) { 4706 mFlushPending = true; 4707 } 4708 } else if (track->isResumePending()) { 4709 track->resumeAck(); 4710 if (last && mHwPaused) { 4711 doHwResume = true; 4712 mHwPaused = false; 4713 } 4714 } 4715 4716 // The first time a track is added we wait 4717 // for all its buffers to be filled before processing it. 4718 // Allow draining the buffer in case the client 4719 // app does not call stop() and relies on underrun to stop: 4720 // hence the test on (track->mRetryCount > 1). 4721 // If retryCount<=1 then track is about to underrun and be removed. 4722 // Do not use a high threshold for compressed audio. 4723 uint32_t minFrames; 4724 if ((track->sharedBuffer() == 0) && !track->isStopping_1() && !track->isPausing() 4725 && (track->mRetryCount > 1) && audio_has_proportional_frames(mFormat)) { 4726 minFrames = mNormalFrameCount; 4727 } else { 4728 minFrames = 1; 4729 } 4730 4731 if ((track->framesReady() >= minFrames) && track->isReady() && !track->isPaused() && 4732 !track->isStopping_2() && !track->isStopped()) 4733 { 4734 ALOGVV("track %d s=%08x [OK]", track->name(), cblk->mServer); 4735 4736 if (track->mFillingUpStatus == Track::FS_FILLED) { 4737 track->mFillingUpStatus = Track::FS_ACTIVE; 4738 // make sure processVolume_l() will apply new volume even if 0 4739 mLeftVolFloat = mRightVolFloat = -1.0; 4740 if (!mHwSupportsPause) { 4741 track->resumeAck(); 4742 } 4743 } 4744 4745 // compute volume for this track 4746 processVolume_l(track, last); 4747 if (last) { 4748 sp<Track> previousTrack = mPreviousTrack.promote(); 4749 if (previousTrack != 0) { 4750 if (track != previousTrack.get()) { 4751 // Flush any data still being written from last track 4752 mBytesRemaining = 0; 4753 // Invalidate previous track to force a seek when resuming. 4754 previousTrack->invalidate(); 4755 } 4756 } 4757 mPreviousTrack = track; 4758 4759 // reset retry count 4760 track->mRetryCount = kMaxTrackRetriesDirect; 4761 mActiveTrack = t; 4762 mixerStatus = MIXER_TRACKS_READY; 4763 if (mHwPaused) { 4764 doHwResume = true; 4765 mHwPaused = false; 4766 } 4767 } 4768 } else { 4769 // clear effect chain input buffer if the last active track started underruns 4770 // to avoid sending previous audio buffer again to effects 4771 if (!mEffectChains.isEmpty() && last) { 4772 mEffectChains[0]->clearInputBuffer(); 4773 } 4774 if (track->isStopping_1()) { 4775 track->mState = TrackBase::STOPPING_2; 4776 if (last && mHwPaused) { 4777 doHwResume = true; 4778 mHwPaused = false; 4779 } 4780 } 4781 if ((track->sharedBuffer() != 0) || track->isStopped() || 4782 track->isStopping_2() || track->isPaused()) { 4783 // We have consumed all the buffers of this track. 4784 // Remove it from the list of active tracks. 4785 size_t audioHALFrames; 4786 if (audio_has_proportional_frames(mFormat)) { 4787 audioHALFrames = (latency_l() * mSampleRate) / 1000; 4788 } else { 4789 audioHALFrames = 0; 4790 } 4791 4792 int64_t framesWritten = mBytesWritten / mFrameSize; 4793 if (mStandby || !last || 4794 track->presentationComplete(framesWritten, audioHALFrames)) { 4795 if (track->isStopping_2()) { 4796 track->mState = TrackBase::STOPPED; 4797 } 4798 if (track->isStopped()) { 4799 track->reset(); 4800 } 4801 tracksToRemove->add(track); 4802 } 4803 } else { 4804 // No buffers for this track. Give it a few chances to 4805 // fill a buffer, then remove it from active list. 4806 // Only consider last track started for mixer state control 4807 if (--(track->mRetryCount) <= 0) { 4808 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 4809 tracksToRemove->add(track); 4810 // indicate to client process that the track was disabled because of underrun; 4811 // it will then automatically call start() when data is available 4812 track->disable(); 4813 } else if (last) { 4814 ALOGW("pause because of UNDERRUN, framesReady = %zu," 4815 "minFrames = %u, mFormat = %#x", 4816 track->framesReady(), minFrames, mFormat); 4817 mixerStatus = MIXER_TRACKS_ENABLED; 4818 if (mHwSupportsPause && !mHwPaused && !mStandby) { 4819 doHwPause = true; 4820 mHwPaused = true; 4821 } 4822 } 4823 } 4824 } 4825 } 4826 4827 // if an active track did not command a flush, check for pending flush on stopped tracks 4828 if (!mFlushPending) { 4829 for (size_t i = 0; i < mTracks.size(); i++) { 4830 if (mTracks[i]->isFlushPending()) { 4831 mTracks[i]->flushAck(); 4832 mFlushPending = true; 4833 } 4834 } 4835 } 4836 4837 // make sure the pause/flush/resume sequence is executed in the right order. 4838 // If a flush is pending and a track is active but the HW is not paused, force a HW pause 4839 // before flush and then resume HW. This can happen in case of pause/flush/resume 4840 // if resume is received before pause is executed. 4841 if (mHwSupportsPause && !mStandby && 4842 (doHwPause || (mFlushPending && !mHwPaused && (count != 0)))) { 4843 mOutput->stream->pause(mOutput->stream); 4844 } 4845 if (mFlushPending) { 4846 flushHw_l(); 4847 } 4848 if (mHwSupportsPause && !mStandby && doHwResume) { 4849 mOutput->stream->resume(mOutput->stream); 4850 } 4851 // remove all the tracks that need to be... 4852 removeTracks_l(*tracksToRemove); 4853 4854 return mixerStatus; 4855} 4856 4857void AudioFlinger::DirectOutputThread::threadLoop_mix() 4858{ 4859 size_t frameCount = mFrameCount; 4860 int8_t *curBuf = (int8_t *)mSinkBuffer; 4861 // output audio to hardware 4862 while (frameCount) { 4863 AudioBufferProvider::Buffer buffer; 4864 buffer.frameCount = frameCount; 4865 status_t status = mActiveTrack->getNextBuffer(&buffer); 4866 if (status != NO_ERROR || buffer.raw == NULL) { 4867 // no need to pad with 0 for compressed audio 4868 if (audio_has_proportional_frames(mFormat)) { 4869 memset(curBuf, 0, frameCount * mFrameSize); 4870 } 4871 break; 4872 } 4873 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 4874 frameCount -= buffer.frameCount; 4875 curBuf += buffer.frameCount * mFrameSize; 4876 mActiveTrack->releaseBuffer(&buffer); 4877 } 4878 mCurrentWriteLength = curBuf - (int8_t *)mSinkBuffer; 4879 mSleepTimeUs = 0; 4880 mStandbyTimeNs = systemTime() + mStandbyDelayNs; 4881 mActiveTrack.clear(); 4882} 4883 4884void AudioFlinger::DirectOutputThread::threadLoop_sleepTime() 4885{ 4886 // do not write to HAL when paused 4887 if (mHwPaused || (usesHwAvSync() && mStandby)) { 4888 mSleepTimeUs = mIdleSleepTimeUs; 4889 return; 4890 } 4891 if (mSleepTimeUs == 0) { 4892 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4893 // For compressed offload, use faster sleep time when underruning until more than an 4894 // entire buffer was written to the audio HAL 4895 if (!audio_has_proportional_frames(mFormat) && 4896 (mType == OFFLOAD) && (mBytesWritten < (int64_t) mBufferSize)) { 4897 mSleepTimeUs = kDirectMinSleepTimeUs; 4898 } else { 4899 mSleepTimeUs = mActiveSleepTimeUs; 4900 } 4901 } else { 4902 mSleepTimeUs = mIdleSleepTimeUs; 4903 } 4904 } else if (mBytesWritten != 0 && audio_has_proportional_frames(mFormat)) { 4905 memset(mSinkBuffer, 0, mFrameCount * mFrameSize); 4906 mSleepTimeUs = 0; 4907 } 4908} 4909 4910void AudioFlinger::DirectOutputThread::threadLoop_exit() 4911{ 4912 { 4913 Mutex::Autolock _l(mLock); 4914 for (size_t i = 0; i < mTracks.size(); i++) { 4915 if (mTracks[i]->isFlushPending()) { 4916 mTracks[i]->flushAck(); 4917 mFlushPending = true; 4918 } 4919 } 4920 if (mFlushPending) { 4921 flushHw_l(); 4922 } 4923 } 4924 PlaybackThread::threadLoop_exit(); 4925} 4926 4927// must be called with thread mutex locked 4928bool AudioFlinger::DirectOutputThread::shouldStandby_l() 4929{ 4930 bool trackPaused = false; 4931 bool trackStopped = false; 4932 4933 if ((mType == DIRECT) && audio_is_linear_pcm(mFormat) && !usesHwAvSync()) { 4934 return !mStandby; 4935 } 4936 4937 // do not put the HAL in standby when paused. AwesomePlayer clear the offloaded AudioTrack 4938 // after a timeout and we will enter standby then. 4939 if (mTracks.size() > 0) { 4940 trackPaused = mTracks[mTracks.size() - 1]->isPaused(); 4941 trackStopped = mTracks[mTracks.size() - 1]->isStopped() || 4942 mTracks[mTracks.size() - 1]->mState == TrackBase::IDLE; 4943 } 4944 4945 return !mStandby && !(trackPaused || (mHwPaused && !trackStopped)); 4946} 4947 4948// getTrackName_l() must be called with ThreadBase::mLock held 4949int AudioFlinger::DirectOutputThread::getTrackName_l(audio_channel_mask_t channelMask __unused, 4950 audio_format_t format __unused, audio_session_t sessionId __unused) 4951{ 4952 return 0; 4953} 4954 4955// deleteTrackName_l() must be called with ThreadBase::mLock held 4956void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name __unused) 4957{ 4958} 4959 4960// checkForNewParameter_l() must be called with ThreadBase::mLock held 4961bool AudioFlinger::DirectOutputThread::checkForNewParameter_l(const String8& keyValuePair, 4962 status_t& status) 4963{ 4964 bool reconfig = false; 4965 bool a2dpDeviceChanged = false; 4966 4967 status = NO_ERROR; 4968 4969 AudioParameter param = AudioParameter(keyValuePair); 4970 int value; 4971 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 4972 // forward device change to effects that have requested to be 4973 // aware of attached audio device. 4974 if (value != AUDIO_DEVICE_NONE) { 4975 a2dpDeviceChanged = 4976 (mOutDevice & AUDIO_DEVICE_OUT_ALL_A2DP) != (value & AUDIO_DEVICE_OUT_ALL_A2DP); 4977 mOutDevice = value; 4978 for (size_t i = 0; i < mEffectChains.size(); i++) { 4979 mEffectChains[i]->setDevice_l(mOutDevice); 4980 } 4981 } 4982 } 4983 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 4984 // do not accept frame count changes if tracks are open as the track buffer 4985 // size depends on frame count and correct behavior would not be garantied 4986 // if frame count is changed after track creation 4987 if (!mTracks.isEmpty()) { 4988 status = INVALID_OPERATION; 4989 } else { 4990 reconfig = true; 4991 } 4992 } 4993 if (status == NO_ERROR) { 4994 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 4995 keyValuePair.string()); 4996 if (!mStandby && status == INVALID_OPERATION) { 4997 mOutput->standby(); 4998 mStandby = true; 4999 mBytesWritten = 0; 5000 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 5001 keyValuePair.string()); 5002 } 5003 if (status == NO_ERROR && reconfig) { 5004 readOutputParameters_l(); 5005 sendIoConfigEvent_l(AUDIO_OUTPUT_CONFIG_CHANGED); 5006 } 5007 } 5008 5009 return reconfig || a2dpDeviceChanged; 5010} 5011 5012uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() const 5013{ 5014 uint32_t time; 5015 if (audio_has_proportional_frames(mFormat)) { 5016 time = PlaybackThread::activeSleepTimeUs(); 5017 } else { 5018 time = kDirectMinSleepTimeUs; 5019 } 5020 return time; 5021} 5022 5023uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() const 5024{ 5025 uint32_t time; 5026 if (audio_has_proportional_frames(mFormat)) { 5027 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 5028 } else { 5029 time = kDirectMinSleepTimeUs; 5030 } 5031 return time; 5032} 5033 5034uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() const 5035{ 5036 uint32_t time; 5037 if (audio_has_proportional_frames(mFormat)) { 5038 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 5039 } else { 5040 time = kDirectMinSleepTimeUs; 5041 } 5042 return time; 5043} 5044 5045void AudioFlinger::DirectOutputThread::cacheParameters_l() 5046{ 5047 PlaybackThread::cacheParameters_l(); 5048 5049 // use shorter standby delay as on normal output to release 5050 // hardware resources as soon as possible 5051 // no delay on outputs with HW A/V sync 5052 if (usesHwAvSync()) { 5053 mStandbyDelayNs = 0; 5054 } else if ((mType == OFFLOAD) && !audio_has_proportional_frames(mFormat)) { 5055 mStandbyDelayNs = kOffloadStandbyDelayNs; 5056 } else { 5057 mStandbyDelayNs = microseconds(mActiveSleepTimeUs*2); 5058 } 5059} 5060 5061void AudioFlinger::DirectOutputThread::flushHw_l() 5062{ 5063 mOutput->flush(); 5064 mHwPaused = false; 5065 mFlushPending = false; 5066} 5067 5068// ---------------------------------------------------------------------------- 5069 5070AudioFlinger::AsyncCallbackThread::AsyncCallbackThread( 5071 const wp<AudioFlinger::PlaybackThread>& playbackThread) 5072 : Thread(false /*canCallJava*/), 5073 mPlaybackThread(playbackThread), 5074 mWriteAckSequence(0), 5075 mDrainSequence(0) 5076{ 5077} 5078 5079AudioFlinger::AsyncCallbackThread::~AsyncCallbackThread() 5080{ 5081} 5082 5083void AudioFlinger::AsyncCallbackThread::onFirstRef() 5084{ 5085 run("Offload Cbk", ANDROID_PRIORITY_URGENT_AUDIO); 5086} 5087 5088bool AudioFlinger::AsyncCallbackThread::threadLoop() 5089{ 5090 while (!exitPending()) { 5091 uint32_t writeAckSequence; 5092 uint32_t drainSequence; 5093 5094 { 5095 Mutex::Autolock _l(mLock); 5096 while (!((mWriteAckSequence & 1) || 5097 (mDrainSequence & 1) || 5098 exitPending())) { 5099 mWaitWorkCV.wait(mLock); 5100 } 5101 5102 if (exitPending()) { 5103 break; 5104 } 5105 ALOGV("AsyncCallbackThread mWriteAckSequence %d mDrainSequence %d", 5106 mWriteAckSequence, mDrainSequence); 5107 writeAckSequence = mWriteAckSequence; 5108 mWriteAckSequence &= ~1; 5109 drainSequence = mDrainSequence; 5110 mDrainSequence &= ~1; 5111 } 5112 { 5113 sp<AudioFlinger::PlaybackThread> playbackThread = mPlaybackThread.promote(); 5114 if (playbackThread != 0) { 5115 if (writeAckSequence & 1) { 5116 playbackThread->resetWriteBlocked(writeAckSequence >> 1); 5117 } 5118 if (drainSequence & 1) { 5119 playbackThread->resetDraining(drainSequence >> 1); 5120 } 5121 } 5122 } 5123 } 5124 return false; 5125} 5126 5127void AudioFlinger::AsyncCallbackThread::exit() 5128{ 5129 ALOGV("AsyncCallbackThread::exit"); 5130 Mutex::Autolock _l(mLock); 5131 requestExit(); 5132 mWaitWorkCV.broadcast(); 5133} 5134 5135void AudioFlinger::AsyncCallbackThread::setWriteBlocked(uint32_t sequence) 5136{ 5137 Mutex::Autolock _l(mLock); 5138 // bit 0 is cleared 5139 mWriteAckSequence = sequence << 1; 5140} 5141 5142void AudioFlinger::AsyncCallbackThread::resetWriteBlocked() 5143{ 5144 Mutex::Autolock _l(mLock); 5145 // ignore unexpected callbacks 5146 if (mWriteAckSequence & 2) { 5147 mWriteAckSequence |= 1; 5148 mWaitWorkCV.signal(); 5149 } 5150} 5151 5152void AudioFlinger::AsyncCallbackThread::setDraining(uint32_t sequence) 5153{ 5154 Mutex::Autolock _l(mLock); 5155 // bit 0 is cleared 5156 mDrainSequence = sequence << 1; 5157} 5158 5159void AudioFlinger::AsyncCallbackThread::resetDraining() 5160{ 5161 Mutex::Autolock _l(mLock); 5162 // ignore unexpected callbacks 5163 if (mDrainSequence & 2) { 5164 mDrainSequence |= 1; 5165 mWaitWorkCV.signal(); 5166 } 5167} 5168 5169 5170// ---------------------------------------------------------------------------- 5171AudioFlinger::OffloadThread::OffloadThread(const sp<AudioFlinger>& audioFlinger, 5172 AudioStreamOut* output, audio_io_handle_t id, uint32_t device, bool systemReady, 5173 uint32_t bitRate) 5174 : DirectOutputThread(audioFlinger, output, id, device, OFFLOAD, systemReady, bitRate), 5175 mPausedWriteLength(0), mPausedBytesRemaining(0), mKeepWakeLock(true) 5176{ 5177 //FIXME: mStandby should be set to true by ThreadBase constructor 5178 mStandby = true; 5179 mKeepWakeLock = property_get_bool("ro.audio.offload_wakelock", true /* default_value */); 5180} 5181 5182void AudioFlinger::OffloadThread::threadLoop_exit() 5183{ 5184 if (mFlushPending || mHwPaused) { 5185 // If a flush is pending or track was paused, just discard buffered data 5186 flushHw_l(); 5187 } else { 5188 mMixerStatus = MIXER_DRAIN_ALL; 5189 threadLoop_drain(); 5190 } 5191 if (mUseAsyncWrite) { 5192 ALOG_ASSERT(mCallbackThread != 0); 5193 mCallbackThread->exit(); 5194 } 5195 PlaybackThread::threadLoop_exit(); 5196} 5197 5198AudioFlinger::PlaybackThread::mixer_state AudioFlinger::OffloadThread::prepareTracks_l( 5199 Vector< sp<Track> > *tracksToRemove 5200) 5201{ 5202 size_t count = mActiveTracks.size(); 5203 5204 mixer_state mixerStatus = MIXER_IDLE; 5205 bool doHwPause = false; 5206 bool doHwResume = false; 5207 5208 ALOGV("OffloadThread::prepareTracks_l active tracks %zu", count); 5209 5210 // find out which tracks need to be processed 5211 for (size_t i = 0; i < count; i++) { 5212 sp<Track> t = mActiveTracks[i].promote(); 5213 // The track died recently 5214 if (t == 0) { 5215 continue; 5216 } 5217 Track* const track = t.get(); 5218#ifdef VERY_VERY_VERBOSE_LOGGING 5219 audio_track_cblk_t* cblk = track->cblk(); 5220#endif 5221 // Only consider last track started for volume and mixer state control. 5222 // In theory an older track could underrun and restart after the new one starts 5223 // but as we only care about the transition phase between two tracks on a 5224 // direct output, it is not a problem to ignore the underrun case. 5225 sp<Track> l = mLatestActiveTrack.promote(); 5226 bool last = l.get() == track; 5227 5228 if (track->isInvalid()) { 5229 ALOGW("An invalidated track shouldn't be in active list"); 5230 tracksToRemove->add(track); 5231 continue; 5232 } 5233 5234 if (track->mState == TrackBase::IDLE) { 5235 ALOGW("An idle track shouldn't be in active list"); 5236 continue; 5237 } 5238 5239 if (track->isPausing()) { 5240 track->setPaused(); 5241 if (last) { 5242 if (mHwSupportsPause && !mHwPaused) { 5243 doHwPause = true; 5244 mHwPaused = true; 5245 } 5246 // If we were part way through writing the mixbuffer to 5247 // the HAL we must save this until we resume 5248 // BUG - this will be wrong if a different track is made active, 5249 // in that case we want to discard the pending data in the 5250 // mixbuffer and tell the client to present it again when the 5251 // track is resumed 5252 mPausedWriteLength = mCurrentWriteLength; 5253 mPausedBytesRemaining = mBytesRemaining; 5254 mBytesRemaining = 0; // stop writing 5255 } 5256 tracksToRemove->add(track); 5257 } else if (track->isFlushPending()) { 5258 track->mRetryCount = kMaxTrackRetriesOffload; 5259 track->flushAck(); 5260 if (last) { 5261 mFlushPending = true; 5262 } 5263 } else if (track->isResumePending()){ 5264 track->resumeAck(); 5265 if (last) { 5266 if (mPausedBytesRemaining) { 5267 // Need to continue write that was interrupted 5268 mCurrentWriteLength = mPausedWriteLength; 5269 mBytesRemaining = mPausedBytesRemaining; 5270 mPausedBytesRemaining = 0; 5271 } 5272 if (mHwPaused) { 5273 doHwResume = true; 5274 mHwPaused = false; 5275 // threadLoop_mix() will handle the case that we need to 5276 // resume an interrupted write 5277 } 5278 // enable write to audio HAL 5279 mSleepTimeUs = 0; 5280 5281 // Do not handle new data in this iteration even if track->framesReady() 5282 mixerStatus = MIXER_TRACKS_ENABLED; 5283 } 5284 } else if (track->framesReady() && track->isReady() && 5285 !track->isPaused() && !track->isTerminated() && !track->isStopping_2()) { 5286 ALOGVV("OffloadThread: track %d s=%08x [OK]", track->name(), cblk->mServer); 5287 if (track->mFillingUpStatus == Track::FS_FILLED) { 5288 track->mFillingUpStatus = Track::FS_ACTIVE; 5289 // make sure processVolume_l() will apply new volume even if 0 5290 mLeftVolFloat = mRightVolFloat = -1.0; 5291 } 5292 5293 if (last) { 5294 sp<Track> previousTrack = mPreviousTrack.promote(); 5295 if (previousTrack != 0) { 5296 if (track != previousTrack.get()) { 5297 // Flush any data still being written from last track 5298 mBytesRemaining = 0; 5299 if (mPausedBytesRemaining) { 5300 // Last track was paused so we also need to flush saved 5301 // mixbuffer state and invalidate track so that it will 5302 // re-submit that unwritten data when it is next resumed 5303 mPausedBytesRemaining = 0; 5304 // Invalidate is a bit drastic - would be more efficient 5305 // to have a flag to tell client that some of the 5306 // previously written data was lost 5307 previousTrack->invalidate(); 5308 } 5309 // flush data already sent to the DSP if changing audio session as audio 5310 // comes from a different source. Also invalidate previous track to force a 5311 // seek when resuming. 5312 if (previousTrack->sessionId() != track->sessionId()) { 5313 previousTrack->invalidate(); 5314 } 5315 } 5316 } 5317 mPreviousTrack = track; 5318 // reset retry count 5319 track->mRetryCount = kMaxTrackRetriesOffload; 5320 mActiveTrack = t; 5321 mixerStatus = MIXER_TRACKS_READY; 5322 } 5323 } else { 5324 ALOGVV("OffloadThread: track %d s=%08x [NOT READY]", track->name(), cblk->mServer); 5325 if (track->isStopping_1()) { 5326 // Hardware buffer can hold a large amount of audio so we must 5327 // wait for all current track's data to drain before we say 5328 // that the track is stopped. 5329 if (mBytesRemaining == 0) { 5330 // Only start draining when all data in mixbuffer 5331 // has been written 5332 ALOGV("OffloadThread: underrun and STOPPING_1 -> draining, STOPPING_2"); 5333 track->mState = TrackBase::STOPPING_2; // so presentation completes after drain 5334 // do not drain if no data was ever sent to HAL (mStandby == true) 5335 if (last && !mStandby) { 5336 // do not modify drain sequence if we are already draining. This happens 5337 // when resuming from pause after drain. 5338 if ((mDrainSequence & 1) == 0) { 5339 mSleepTimeUs = 0; 5340 mStandbyTimeNs = systemTime() + mStandbyDelayNs; 5341 mixerStatus = MIXER_DRAIN_TRACK; 5342 mDrainSequence += 2; 5343 } 5344 if (mHwPaused) { 5345 // It is possible to move from PAUSED to STOPPING_1 without 5346 // a resume so we must ensure hardware is running 5347 doHwResume = true; 5348 mHwPaused = false; 5349 } 5350 } 5351 } 5352 } else if (track->isStopping_2()) { 5353 // Drain has completed or we are in standby, signal presentation complete 5354 if (!(mDrainSequence & 1) || !last || mStandby) { 5355 track->mState = TrackBase::STOPPED; 5356 size_t audioHALFrames = 5357 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 5358 int64_t framesWritten = 5359 mBytesWritten / mOutput->getFrameSize(); 5360 track->presentationComplete(framesWritten, audioHALFrames); 5361 track->reset(); 5362 tracksToRemove->add(track); 5363 } 5364 } else { 5365 // No buffers for this track. Give it a few chances to 5366 // fill a buffer, then remove it from active list. 5367 if (--(track->mRetryCount) <= 0) { 5368 ALOGV("OffloadThread: BUFFER TIMEOUT: remove(%d) from active list", 5369 track->name()); 5370 tracksToRemove->add(track); 5371 // indicate to client process that the track was disabled because of underrun; 5372 // it will then automatically call start() when data is available 5373 track->disable(); 5374 } else if (last){ 5375 mixerStatus = MIXER_TRACKS_ENABLED; 5376 } 5377 } 5378 } 5379 // compute volume for this track 5380 processVolume_l(track, last); 5381 } 5382 5383 // make sure the pause/flush/resume sequence is executed in the right order. 5384 // If a flush is pending and a track is active but the HW is not paused, force a HW pause 5385 // before flush and then resume HW. This can happen in case of pause/flush/resume 5386 // if resume is received before pause is executed. 5387 if (!mStandby && (doHwPause || (mFlushPending && !mHwPaused && (count != 0)))) { 5388 mOutput->stream->pause(mOutput->stream); 5389 } 5390 if (mFlushPending) { 5391 flushHw_l(); 5392 } 5393 if (!mStandby && doHwResume) { 5394 mOutput->stream->resume(mOutput->stream); 5395 } 5396 5397 // remove all the tracks that need to be... 5398 removeTracks_l(*tracksToRemove); 5399 5400 return mixerStatus; 5401} 5402 5403// must be called with thread mutex locked 5404bool AudioFlinger::OffloadThread::waitingAsyncCallback_l() 5405{ 5406 ALOGVV("waitingAsyncCallback_l mWriteAckSequence %d mDrainSequence %d", 5407 mWriteAckSequence, mDrainSequence); 5408 if (mUseAsyncWrite && ((mWriteAckSequence & 1) || (mDrainSequence & 1))) { 5409 return true; 5410 } 5411 return false; 5412} 5413 5414bool AudioFlinger::OffloadThread::waitingAsyncCallback() 5415{ 5416 Mutex::Autolock _l(mLock); 5417 return waitingAsyncCallback_l(); 5418} 5419 5420void AudioFlinger::OffloadThread::flushHw_l() 5421{ 5422 DirectOutputThread::flushHw_l(); 5423 // Flush anything still waiting in the mixbuffer 5424 mCurrentWriteLength = 0; 5425 mBytesRemaining = 0; 5426 mPausedWriteLength = 0; 5427 mPausedBytesRemaining = 0; 5428 // reset bytes written count to reflect that DSP buffers are empty after flush. 5429 mBytesWritten = 0; 5430 5431 if (mUseAsyncWrite) { 5432 // discard any pending drain or write ack by incrementing sequence 5433 mWriteAckSequence = (mWriteAckSequence + 2) & ~1; 5434 mDrainSequence = (mDrainSequence + 2) & ~1; 5435 ALOG_ASSERT(mCallbackThread != 0); 5436 mCallbackThread->setWriteBlocked(mWriteAckSequence); 5437 mCallbackThread->setDraining(mDrainSequence); 5438 } 5439} 5440 5441uint32_t AudioFlinger::OffloadThread::activeSleepTimeUs() const 5442{ 5443 uint32_t time; 5444 if (audio_has_proportional_frames(mFormat)) { 5445 time = PlaybackThread::activeSleepTimeUs(); 5446 } else { 5447 // sleep time is half the duration of an audio HAL buffer. 5448 // Note: This can be problematic in case of underrun with variable bit rate and 5449 // current rate is much less than initial rate. 5450 time = (uint32_t)max(kDirectMinSleepTimeUs, mBufferDurationUs / 2); 5451 } 5452 return time; 5453} 5454 5455void AudioFlinger::OffloadThread::invalidateTracks(audio_stream_type_t streamType) 5456{ 5457 Mutex::Autolock _l(mLock); 5458 mFlushPending = true; 5459 PlaybackThread::invalidateTracks_l(streamType); 5460} 5461 5462// ---------------------------------------------------------------------------- 5463 5464AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 5465 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id, bool systemReady) 5466 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->outDevice(), 5467 systemReady, DUPLICATING), 5468 mWaitTimeMs(UINT_MAX) 5469{ 5470 addOutputTrack(mainThread); 5471} 5472 5473AudioFlinger::DuplicatingThread::~DuplicatingThread() 5474{ 5475 for (size_t i = 0; i < mOutputTracks.size(); i++) { 5476 mOutputTracks[i]->destroy(); 5477 } 5478} 5479 5480void AudioFlinger::DuplicatingThread::threadLoop_mix() 5481{ 5482 // mix buffers... 5483 if (outputsReady(outputTracks)) { 5484 mAudioMixer->process(); 5485 } else { 5486 if (mMixerBufferValid) { 5487 memset(mMixerBuffer, 0, mMixerBufferSize); 5488 } else { 5489 memset(mSinkBuffer, 0, mSinkBufferSize); 5490 } 5491 } 5492 mSleepTimeUs = 0; 5493 writeFrames = mNormalFrameCount; 5494 mCurrentWriteLength = mSinkBufferSize; 5495 mStandbyTimeNs = systemTime() + mStandbyDelayNs; 5496} 5497 5498void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() 5499{ 5500 if (mSleepTimeUs == 0) { 5501 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 5502 mSleepTimeUs = mActiveSleepTimeUs; 5503 } else { 5504 mSleepTimeUs = mIdleSleepTimeUs; 5505 } 5506 } else if (mBytesWritten != 0) { 5507 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 5508 writeFrames = mNormalFrameCount; 5509 memset(mSinkBuffer, 0, mSinkBufferSize); 5510 } else { 5511 // flush remaining overflow buffers in output tracks 5512 writeFrames = 0; 5513 } 5514 mSleepTimeUs = 0; 5515 } 5516} 5517 5518ssize_t AudioFlinger::DuplicatingThread::threadLoop_write() 5519{ 5520 for (size_t i = 0; i < outputTracks.size(); i++) { 5521 outputTracks[i]->write(mSinkBuffer, writeFrames); 5522 } 5523 mStandby = false; 5524 return (ssize_t)mSinkBufferSize; 5525} 5526 5527void AudioFlinger::DuplicatingThread::threadLoop_standby() 5528{ 5529 // DuplicatingThread implements standby by stopping all tracks 5530 for (size_t i = 0; i < outputTracks.size(); i++) { 5531 outputTracks[i]->stop(); 5532 } 5533} 5534 5535void AudioFlinger::DuplicatingThread::saveOutputTracks() 5536{ 5537 outputTracks = mOutputTracks; 5538} 5539 5540void AudioFlinger::DuplicatingThread::clearOutputTracks() 5541{ 5542 outputTracks.clear(); 5543} 5544 5545void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 5546{ 5547 Mutex::Autolock _l(mLock); 5548 // The downstream MixerThread consumes thread->frameCount() amount of frames per mix pass. 5549 // Adjust for thread->sampleRate() to determine minimum buffer frame count. 5550 // Then triple buffer because Threads do not run synchronously and may not be clock locked. 5551 const size_t frameCount = 5552 3 * sourceFramesNeeded(mSampleRate, thread->frameCount(), thread->sampleRate()); 5553 // TODO: Consider asynchronous sample rate conversion to handle clock disparity 5554 // from different OutputTracks and their associated MixerThreads (e.g. one may 5555 // nearly empty and the other may be dropping data). 5556 5557 sp<OutputTrack> outputTrack = new OutputTrack(thread, 5558 this, 5559 mSampleRate, 5560 mFormat, 5561 mChannelMask, 5562 frameCount, 5563 IPCThreadState::self()->getCallingUid()); 5564 if (outputTrack->cblk() != NULL) { 5565 thread->setStreamVolume(AUDIO_STREAM_PATCH, 1.0f); 5566 mOutputTracks.add(outputTrack); 5567 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack.get(), thread); 5568 updateWaitTime_l(); 5569 } 5570} 5571 5572void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 5573{ 5574 Mutex::Autolock _l(mLock); 5575 for (size_t i = 0; i < mOutputTracks.size(); i++) { 5576 if (mOutputTracks[i]->thread() == thread) { 5577 mOutputTracks[i]->destroy(); 5578 mOutputTracks.removeAt(i); 5579 updateWaitTime_l(); 5580 if (thread->getOutput() == mOutput) { 5581 mOutput = NULL; 5582 } 5583 return; 5584 } 5585 } 5586 ALOGV("removeOutputTrack(): unknown thread: %p", thread); 5587} 5588 5589// caller must hold mLock 5590void AudioFlinger::DuplicatingThread::updateWaitTime_l() 5591{ 5592 mWaitTimeMs = UINT_MAX; 5593 for (size_t i = 0; i < mOutputTracks.size(); i++) { 5594 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 5595 if (strong != 0) { 5596 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 5597 if (waitTimeMs < mWaitTimeMs) { 5598 mWaitTimeMs = waitTimeMs; 5599 } 5600 } 5601 } 5602} 5603 5604 5605bool AudioFlinger::DuplicatingThread::outputsReady( 5606 const SortedVector< sp<OutputTrack> > &outputTracks) 5607{ 5608 for (size_t i = 0; i < outputTracks.size(); i++) { 5609 sp<ThreadBase> thread = outputTracks[i]->thread().promote(); 5610 if (thread == 0) { 5611 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", 5612 outputTracks[i].get()); 5613 return false; 5614 } 5615 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 5616 // see note at standby() declaration 5617 if (playbackThread->standby() && !playbackThread->isSuspended()) { 5618 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), 5619 thread.get()); 5620 return false; 5621 } 5622 } 5623 return true; 5624} 5625 5626uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() const 5627{ 5628 return (mWaitTimeMs * 1000) / 2; 5629} 5630 5631void AudioFlinger::DuplicatingThread::cacheParameters_l() 5632{ 5633 // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first 5634 updateWaitTime_l(); 5635 5636 MixerThread::cacheParameters_l(); 5637} 5638 5639// ---------------------------------------------------------------------------- 5640// Record 5641// ---------------------------------------------------------------------------- 5642 5643AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 5644 AudioStreamIn *input, 5645 audio_io_handle_t id, 5646 audio_devices_t outDevice, 5647 audio_devices_t inDevice, 5648 bool systemReady 5649#ifdef TEE_SINK 5650 , const sp<NBAIO_Sink>& teeSink 5651#endif 5652 ) : 5653 ThreadBase(audioFlinger, id, outDevice, inDevice, RECORD, systemReady), 5654 mInput(input), mActiveTracksGen(0), mRsmpInBuffer(NULL), 5655 // mRsmpInFrames and mRsmpInFramesP2 are set by readInputParameters_l() 5656 mRsmpInRear(0) 5657#ifdef TEE_SINK 5658 , mTeeSink(teeSink) 5659#endif 5660 , mReadOnlyHeap(new MemoryDealer(kRecordThreadReadOnlyHeapSize, 5661 "RecordThreadRO", MemoryHeapBase::READ_ONLY)) 5662 // mFastCapture below 5663 , mFastCaptureFutex(0) 5664 // mInputSource 5665 // mPipeSink 5666 // mPipeSource 5667 , mPipeFramesP2(0) 5668 // mPipeMemory 5669 // mFastCaptureNBLogWriter 5670 , mFastTrackAvail(false) 5671{ 5672 snprintf(mThreadName, kThreadNameLength, "AudioIn_%X", id); 5673 mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mThreadName); 5674 5675 readInputParameters_l(); 5676 5677 // create an NBAIO source for the HAL input stream, and negotiate 5678 mInputSource = new AudioStreamInSource(input->stream); 5679 size_t numCounterOffers = 0; 5680 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount, mFormat)}; 5681#if !LOG_NDEBUG 5682 ssize_t index = 5683#else 5684 (void) 5685#endif 5686 mInputSource->negotiate(offers, 1, NULL, numCounterOffers); 5687 ALOG_ASSERT(index == 0); 5688 5689 // initialize fast capture depending on configuration 5690 bool initFastCapture; 5691 switch (kUseFastCapture) { 5692 case FastCapture_Never: 5693 initFastCapture = false; 5694 break; 5695 case FastCapture_Always: 5696 initFastCapture = true; 5697 break; 5698 case FastCapture_Static: 5699 initFastCapture = (mFrameCount * 1000) / mSampleRate < kMinNormalCaptureBufferSizeMs; 5700 break; 5701 // case FastCapture_Dynamic: 5702 } 5703 5704 if (initFastCapture) { 5705 // create a Pipe for FastCapture to write to, and for us and fast tracks to read from 5706 NBAIO_Format format = mInputSource->format(); 5707 size_t pipeFramesP2 = roundup(mSampleRate / 25); // double-buffering of 20 ms each 5708 size_t pipeSize = pipeFramesP2 * Format_frameSize(format); 5709 void *pipeBuffer; 5710 const sp<MemoryDealer> roHeap(readOnlyHeap()); 5711 sp<IMemory> pipeMemory; 5712 if ((roHeap == 0) || 5713 (pipeMemory = roHeap->allocate(pipeSize)) == 0 || 5714 (pipeBuffer = pipeMemory->pointer()) == NULL) { 5715 ALOGE("not enough memory for pipe buffer size=%zu", pipeSize); 5716 goto failed; 5717 } 5718 // pipe will be shared directly with fast clients, so clear to avoid leaking old information 5719 memset(pipeBuffer, 0, pipeSize); 5720 Pipe *pipe = new Pipe(pipeFramesP2, format, pipeBuffer); 5721 const NBAIO_Format offers[1] = {format}; 5722 size_t numCounterOffers = 0; 5723 ssize_t index = pipe->negotiate(offers, 1, NULL, numCounterOffers); 5724 ALOG_ASSERT(index == 0); 5725 mPipeSink = pipe; 5726 PipeReader *pipeReader = new PipeReader(*pipe); 5727 numCounterOffers = 0; 5728 index = pipeReader->negotiate(offers, 1, NULL, numCounterOffers); 5729 ALOG_ASSERT(index == 0); 5730 mPipeSource = pipeReader; 5731 mPipeFramesP2 = pipeFramesP2; 5732 mPipeMemory = pipeMemory; 5733 5734 // create fast capture 5735 mFastCapture = new FastCapture(); 5736 FastCaptureStateQueue *sq = mFastCapture->sq(); 5737#ifdef STATE_QUEUE_DUMP 5738 // FIXME 5739#endif 5740 FastCaptureState *state = sq->begin(); 5741 state->mCblk = NULL; 5742 state->mInputSource = mInputSource.get(); 5743 state->mInputSourceGen++; 5744 state->mPipeSink = pipe; 5745 state->mPipeSinkGen++; 5746 state->mFrameCount = mFrameCount; 5747 state->mCommand = FastCaptureState::COLD_IDLE; 5748 // already done in constructor initialization list 5749 //mFastCaptureFutex = 0; 5750 state->mColdFutexAddr = &mFastCaptureFutex; 5751 state->mColdGen++; 5752 state->mDumpState = &mFastCaptureDumpState; 5753#ifdef TEE_SINK 5754 // FIXME 5755#endif 5756 mFastCaptureNBLogWriter = audioFlinger->newWriter_l(kFastCaptureLogSize, "FastCapture"); 5757 state->mNBLogWriter = mFastCaptureNBLogWriter.get(); 5758 sq->end(); 5759 sq->push(FastCaptureStateQueue::BLOCK_UNTIL_PUSHED); 5760 5761 // start the fast capture 5762 mFastCapture->run("FastCapture", ANDROID_PRIORITY_URGENT_AUDIO); 5763 pid_t tid = mFastCapture->getTid(); 5764 sendPrioConfigEvent(getpid_cached, tid, kPriorityFastCapture); 5765#ifdef AUDIO_WATCHDOG 5766 // FIXME 5767#endif 5768 5769 mFastTrackAvail = true; 5770 } 5771failed: ; 5772 5773 // FIXME mNormalSource 5774} 5775 5776AudioFlinger::RecordThread::~RecordThread() 5777{ 5778 if (mFastCapture != 0) { 5779 FastCaptureStateQueue *sq = mFastCapture->sq(); 5780 FastCaptureState *state = sq->begin(); 5781 if (state->mCommand == FastCaptureState::COLD_IDLE) { 5782 int32_t old = android_atomic_inc(&mFastCaptureFutex); 5783 if (old == -1) { 5784 (void) syscall(__NR_futex, &mFastCaptureFutex, FUTEX_WAKE_PRIVATE, 1); 5785 } 5786 } 5787 state->mCommand = FastCaptureState::EXIT; 5788 sq->end(); 5789 sq->push(FastCaptureStateQueue::BLOCK_UNTIL_PUSHED); 5790 mFastCapture->join(); 5791 mFastCapture.clear(); 5792 } 5793 mAudioFlinger->unregisterWriter(mFastCaptureNBLogWriter); 5794 mAudioFlinger->unregisterWriter(mNBLogWriter); 5795 free(mRsmpInBuffer); 5796} 5797 5798void AudioFlinger::RecordThread::onFirstRef() 5799{ 5800 run(mThreadName, PRIORITY_URGENT_AUDIO); 5801} 5802 5803bool AudioFlinger::RecordThread::threadLoop() 5804{ 5805 nsecs_t lastWarning = 0; 5806 5807 inputStandBy(); 5808 5809reacquire_wakelock: 5810 sp<RecordTrack> activeTrack; 5811 int activeTracksGen; 5812 { 5813 Mutex::Autolock _l(mLock); 5814 size_t size = mActiveTracks.size(); 5815 activeTracksGen = mActiveTracksGen; 5816 if (size > 0) { 5817 // FIXME an arbitrary choice 5818 activeTrack = mActiveTracks[0]; 5819 acquireWakeLock_l(activeTrack->uid()); 5820 if (size > 1) { 5821 SortedVector<int> tmp; 5822 for (size_t i = 0; i < size; i++) { 5823 tmp.add(mActiveTracks[i]->uid()); 5824 } 5825 updateWakeLockUids_l(tmp); 5826 } 5827 } else { 5828 acquireWakeLock_l(-1); 5829 } 5830 } 5831 5832 // used to request a deferred sleep, to be executed later while mutex is unlocked 5833 uint32_t sleepUs = 0; 5834 5835 // loop while there is work to do 5836 for (;;) { 5837 Vector< sp<EffectChain> > effectChains; 5838 5839 // sleep with mutex unlocked 5840 if (sleepUs > 0) { 5841 ATRACE_BEGIN("sleep"); 5842 usleep(sleepUs); 5843 ATRACE_END(); 5844 sleepUs = 0; 5845 } 5846 5847 // activeTracks accumulates a copy of a subset of mActiveTracks 5848 Vector< sp<RecordTrack> > activeTracks; 5849 5850 // reference to the (first and only) active fast track 5851 sp<RecordTrack> fastTrack; 5852 5853 // reference to a fast track which is about to be removed 5854 sp<RecordTrack> fastTrackToRemove; 5855 5856 { // scope for mLock 5857 Mutex::Autolock _l(mLock); 5858 5859 processConfigEvents_l(); 5860 5861 // check exitPending here because checkForNewParameters_l() and 5862 // checkForNewParameters_l() can temporarily release mLock 5863 if (exitPending()) { 5864 break; 5865 } 5866 5867 // if no active track(s), then standby and release wakelock 5868 size_t size = mActiveTracks.size(); 5869 if (size == 0) { 5870 standbyIfNotAlreadyInStandby(); 5871 // exitPending() can't become true here 5872 releaseWakeLock_l(); 5873 ALOGV("RecordThread: loop stopping"); 5874 // go to sleep 5875 mWaitWorkCV.wait(mLock); 5876 ALOGV("RecordThread: loop starting"); 5877 goto reacquire_wakelock; 5878 } 5879 5880 if (mActiveTracksGen != activeTracksGen) { 5881 activeTracksGen = mActiveTracksGen; 5882 SortedVector<int> tmp; 5883 for (size_t i = 0; i < size; i++) { 5884 tmp.add(mActiveTracks[i]->uid()); 5885 } 5886 updateWakeLockUids_l(tmp); 5887 } 5888 5889 bool doBroadcast = false; 5890 for (size_t i = 0; i < size; ) { 5891 5892 activeTrack = mActiveTracks[i]; 5893 if (activeTrack->isTerminated()) { 5894 if (activeTrack->isFastTrack()) { 5895 ALOG_ASSERT(fastTrackToRemove == 0); 5896 fastTrackToRemove = activeTrack; 5897 } 5898 removeTrack_l(activeTrack); 5899 mActiveTracks.remove(activeTrack); 5900 mActiveTracksGen++; 5901 size--; 5902 continue; 5903 } 5904 5905 TrackBase::track_state activeTrackState = activeTrack->mState; 5906 switch (activeTrackState) { 5907 5908 case TrackBase::PAUSING: 5909 mActiveTracks.remove(activeTrack); 5910 mActiveTracksGen++; 5911 doBroadcast = true; 5912 size--; 5913 continue; 5914 5915 case TrackBase::STARTING_1: 5916 sleepUs = 10000; 5917 i++; 5918 continue; 5919 5920 case TrackBase::STARTING_2: 5921 doBroadcast = true; 5922 mStandby = false; 5923 activeTrack->mState = TrackBase::ACTIVE; 5924 break; 5925 5926 case TrackBase::ACTIVE: 5927 break; 5928 5929 case TrackBase::IDLE: 5930 i++; 5931 continue; 5932 5933 default: 5934 LOG_ALWAYS_FATAL("Unexpected activeTrackState %d", activeTrackState); 5935 } 5936 5937 activeTracks.add(activeTrack); 5938 i++; 5939 5940 if (activeTrack->isFastTrack()) { 5941 ALOG_ASSERT(!mFastTrackAvail); 5942 ALOG_ASSERT(fastTrack == 0); 5943 fastTrack = activeTrack; 5944 } 5945 } 5946 if (doBroadcast) { 5947 mStartStopCond.broadcast(); 5948 } 5949 5950 // sleep if there are no active tracks to process 5951 if (activeTracks.size() == 0) { 5952 if (sleepUs == 0) { 5953 sleepUs = kRecordThreadSleepUs; 5954 } 5955 continue; 5956 } 5957 sleepUs = 0; 5958 5959 lockEffectChains_l(effectChains); 5960 } 5961 5962 // thread mutex is now unlocked, mActiveTracks unknown, activeTracks.size() > 0 5963 5964 size_t size = effectChains.size(); 5965 for (size_t i = 0; i < size; i++) { 5966 // thread mutex is not locked, but effect chain is locked 5967 effectChains[i]->process_l(); 5968 } 5969 5970 // Push a new fast capture state if fast capture is not already running, or cblk change 5971 if (mFastCapture != 0) { 5972 FastCaptureStateQueue *sq = mFastCapture->sq(); 5973 FastCaptureState *state = sq->begin(); 5974 bool didModify = false; 5975 FastCaptureStateQueue::block_t block = FastCaptureStateQueue::BLOCK_UNTIL_PUSHED; 5976 if (state->mCommand != FastCaptureState::READ_WRITE /* FIXME && 5977 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)*/) { 5978 if (state->mCommand == FastCaptureState::COLD_IDLE) { 5979 int32_t old = android_atomic_inc(&mFastCaptureFutex); 5980 if (old == -1) { 5981 (void) syscall(__NR_futex, &mFastCaptureFutex, FUTEX_WAKE_PRIVATE, 1); 5982 } 5983 } 5984 state->mCommand = FastCaptureState::READ_WRITE; 5985#if 0 // FIXME 5986 mFastCaptureDumpState.increaseSamplingN(mAudioFlinger->isLowRamDevice() ? 5987 FastThreadDumpState::kSamplingNforLowRamDevice : 5988 FastThreadDumpState::kSamplingN); 5989#endif 5990 didModify = true; 5991 } 5992 audio_track_cblk_t *cblkOld = state->mCblk; 5993 audio_track_cblk_t *cblkNew = fastTrack != 0 ? fastTrack->cblk() : NULL; 5994 if (cblkNew != cblkOld) { 5995 state->mCblk = cblkNew; 5996 // block until acked if removing a fast track 5997 if (cblkOld != NULL) { 5998 block = FastCaptureStateQueue::BLOCK_UNTIL_ACKED; 5999 } 6000 didModify = true; 6001 } 6002 sq->end(didModify); 6003 if (didModify) { 6004 sq->push(block); 6005#if 0 6006 if (kUseFastCapture == FastCapture_Dynamic) { 6007 mNormalSource = mPipeSource; 6008 } 6009#endif 6010 } 6011 } 6012 6013 // now run the fast track destructor with thread mutex unlocked 6014 fastTrackToRemove.clear(); 6015 6016 // Read from HAL to keep up with fastest client if multiple active tracks, not slowest one. 6017 // Only the client(s) that are too slow will overrun. But if even the fastest client is too 6018 // slow, then this RecordThread will overrun by not calling HAL read often enough. 6019 // If destination is non-contiguous, first read past the nominal end of buffer, then 6020 // copy to the right place. Permitted because mRsmpInBuffer was over-allocated. 6021 6022 int32_t rear = mRsmpInRear & (mRsmpInFramesP2 - 1); 6023 ssize_t framesRead; 6024 6025 // If an NBAIO source is present, use it to read the normal capture's data 6026 if (mPipeSource != 0) { 6027 size_t framesToRead = mBufferSize / mFrameSize; 6028 framesRead = mPipeSource->read((uint8_t*)mRsmpInBuffer + rear * mFrameSize, 6029 framesToRead); 6030 if (framesRead == 0) { 6031 // since pipe is non-blocking, simulate blocking input 6032 sleepUs = (framesToRead * 1000000LL) / mSampleRate; 6033 } 6034 // otherwise use the HAL / AudioStreamIn directly 6035 } else { 6036 ATRACE_BEGIN("read"); 6037 ssize_t bytesRead = mInput->stream->read(mInput->stream, 6038 (uint8_t*)mRsmpInBuffer + rear * mFrameSize, mBufferSize); 6039 ATRACE_END(); 6040 if (bytesRead < 0) { 6041 framesRead = bytesRead; 6042 } else { 6043 framesRead = bytesRead / mFrameSize; 6044 } 6045 } 6046 6047 // Update server timestamp with server stats 6048 // systemTime() is optional if the hardware supports timestamps. 6049 mTimestamp.mPosition[ExtendedTimestamp::LOCATION_SERVER] += framesRead; 6050 mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_SERVER] = systemTime(); 6051 6052 // Update server timestamp with kernel stats 6053 if (mInput->stream->get_capture_position != nullptr) { 6054 int64_t position, time; 6055 int ret = mInput->stream->get_capture_position(mInput->stream, &position, &time); 6056 if (ret == NO_ERROR) { 6057 mTimestamp.mPosition[ExtendedTimestamp::LOCATION_KERNEL] = position; 6058 mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL] = time; 6059 // Note: In general record buffers should tend to be empty in 6060 // a properly running pipeline. 6061 // 6062 // Also, it is not advantageous to call get_presentation_position during the read 6063 // as the read obtains a lock, preventing the timestamp call from executing. 6064 } 6065 } 6066 // Use this to track timestamp information 6067 // ALOGD("%s", mTimestamp.toString().c_str()); 6068 6069 if (framesRead < 0 || (framesRead == 0 && mPipeSource == 0)) { 6070 ALOGE("read failed: framesRead=%zd", framesRead); 6071 // Force input into standby so that it tries to recover at next read attempt 6072 inputStandBy(); 6073 sleepUs = kRecordThreadSleepUs; 6074 } 6075 if (framesRead <= 0) { 6076 goto unlock; 6077 } 6078 ALOG_ASSERT(framesRead > 0); 6079 6080 if (mTeeSink != 0) { 6081 (void) mTeeSink->write((uint8_t*)mRsmpInBuffer + rear * mFrameSize, framesRead); 6082 } 6083 // If destination is non-contiguous, we now correct for reading past end of buffer. 6084 { 6085 size_t part1 = mRsmpInFramesP2 - rear; 6086 if ((size_t) framesRead > part1) { 6087 memcpy(mRsmpInBuffer, (uint8_t*)mRsmpInBuffer + mRsmpInFramesP2 * mFrameSize, 6088 (framesRead - part1) * mFrameSize); 6089 } 6090 } 6091 rear = mRsmpInRear += framesRead; 6092 6093 size = activeTracks.size(); 6094 // loop over each active track 6095 for (size_t i = 0; i < size; i++) { 6096 activeTrack = activeTracks[i]; 6097 6098 // skip fast tracks, as those are handled directly by FastCapture 6099 if (activeTrack->isFastTrack()) { 6100 continue; 6101 } 6102 6103 // TODO: This code probably should be moved to RecordTrack. 6104 // TODO: Update the activeTrack buffer converter in case of reconfigure. 6105 6106 enum { 6107 OVERRUN_UNKNOWN, 6108 OVERRUN_TRUE, 6109 OVERRUN_FALSE 6110 } overrun = OVERRUN_UNKNOWN; 6111 6112 // loop over getNextBuffer to handle circular sink 6113 for (;;) { 6114 6115 activeTrack->mSink.frameCount = ~0; 6116 status_t status = activeTrack->getNextBuffer(&activeTrack->mSink); 6117 size_t framesOut = activeTrack->mSink.frameCount; 6118 LOG_ALWAYS_FATAL_IF((status == OK) != (framesOut > 0)); 6119 6120 // check available frames and handle overrun conditions 6121 // if the record track isn't draining fast enough. 6122 bool hasOverrun; 6123 size_t framesIn; 6124 activeTrack->mResamplerBufferProvider->sync(&framesIn, &hasOverrun); 6125 if (hasOverrun) { 6126 overrun = OVERRUN_TRUE; 6127 } 6128 if (framesOut == 0 || framesIn == 0) { 6129 break; 6130 } 6131 6132 // Don't allow framesOut to be larger than what is possible with resampling 6133 // from framesIn. 6134 // This isn't strictly necessary but helps limit buffer resizing in 6135 // RecordBufferConverter. TODO: remove when no longer needed. 6136 framesOut = min(framesOut, 6137 destinationFramesPossible( 6138 framesIn, mSampleRate, activeTrack->mSampleRate)); 6139 // process frames from the RecordThread buffer provider to the RecordTrack buffer 6140 framesOut = activeTrack->mRecordBufferConverter->convert( 6141 activeTrack->mSink.raw, activeTrack->mResamplerBufferProvider, framesOut); 6142 6143 if (framesOut > 0 && (overrun == OVERRUN_UNKNOWN)) { 6144 overrun = OVERRUN_FALSE; 6145 } 6146 6147 if (activeTrack->mFramesToDrop == 0) { 6148 if (framesOut > 0) { 6149 activeTrack->mSink.frameCount = framesOut; 6150 activeTrack->releaseBuffer(&activeTrack->mSink); 6151 } 6152 } else { 6153 // FIXME could do a partial drop of framesOut 6154 if (activeTrack->mFramesToDrop > 0) { 6155 activeTrack->mFramesToDrop -= framesOut; 6156 if (activeTrack->mFramesToDrop <= 0) { 6157 activeTrack->clearSyncStartEvent(); 6158 } 6159 } else { 6160 activeTrack->mFramesToDrop += framesOut; 6161 if (activeTrack->mFramesToDrop >= 0 || activeTrack->mSyncStartEvent == 0 || 6162 activeTrack->mSyncStartEvent->isCancelled()) { 6163 ALOGW("Synced record %s, session %d, trigger session %d", 6164 (activeTrack->mFramesToDrop >= 0) ? "timed out" : "cancelled", 6165 activeTrack->sessionId(), 6166 (activeTrack->mSyncStartEvent != 0) ? 6167 activeTrack->mSyncStartEvent->triggerSession() : 6168 AUDIO_SESSION_NONE); 6169 activeTrack->clearSyncStartEvent(); 6170 } 6171 } 6172 } 6173 6174 if (framesOut == 0) { 6175 break; 6176 } 6177 } 6178 6179 switch (overrun) { 6180 case OVERRUN_TRUE: 6181 // client isn't retrieving buffers fast enough 6182 if (!activeTrack->setOverflow()) { 6183 nsecs_t now = systemTime(); 6184 // FIXME should lastWarning per track? 6185 if ((now - lastWarning) > kWarningThrottleNs) { 6186 ALOGW("RecordThread: buffer overflow"); 6187 lastWarning = now; 6188 } 6189 } 6190 break; 6191 case OVERRUN_FALSE: 6192 activeTrack->clearOverflow(); 6193 break; 6194 case OVERRUN_UNKNOWN: 6195 break; 6196 } 6197 6198 // update frame information and push timestamp out 6199 activeTrack->updateTrackFrameInfo( 6200 activeTrack->mServerProxy->framesReleased(), 6201 mTimestamp.mPosition[ExtendedTimestamp::LOCATION_SERVER], 6202 mSampleRate, mTimestamp); 6203 } 6204 6205unlock: 6206 // enable changes in effect chain 6207 unlockEffectChains(effectChains); 6208 // effectChains doesn't need to be cleared, since it is cleared by destructor at scope end 6209 } 6210 6211 standbyIfNotAlreadyInStandby(); 6212 6213 { 6214 Mutex::Autolock _l(mLock); 6215 for (size_t i = 0; i < mTracks.size(); i++) { 6216 sp<RecordTrack> track = mTracks[i]; 6217 track->invalidate(); 6218 } 6219 mActiveTracks.clear(); 6220 mActiveTracksGen++; 6221 mStartStopCond.broadcast(); 6222 } 6223 6224 releaseWakeLock(); 6225 6226 ALOGV("RecordThread %p exiting", this); 6227 return false; 6228} 6229 6230void AudioFlinger::RecordThread::standbyIfNotAlreadyInStandby() 6231{ 6232 if (!mStandby) { 6233 inputStandBy(); 6234 mStandby = true; 6235 } 6236} 6237 6238void AudioFlinger::RecordThread::inputStandBy() 6239{ 6240 // Idle the fast capture if it's currently running 6241 if (mFastCapture != 0) { 6242 FastCaptureStateQueue *sq = mFastCapture->sq(); 6243 FastCaptureState *state = sq->begin(); 6244 if (!(state->mCommand & FastCaptureState::IDLE)) { 6245 state->mCommand = FastCaptureState::COLD_IDLE; 6246 state->mColdFutexAddr = &mFastCaptureFutex; 6247 state->mColdGen++; 6248 mFastCaptureFutex = 0; 6249 sq->end(); 6250 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 6251 sq->push(FastCaptureStateQueue::BLOCK_UNTIL_ACKED); 6252#if 0 6253 if (kUseFastCapture == FastCapture_Dynamic) { 6254 // FIXME 6255 } 6256#endif 6257#ifdef AUDIO_WATCHDOG 6258 // FIXME 6259#endif 6260 } else { 6261 sq->end(false /*didModify*/); 6262 } 6263 } 6264 mInput->stream->common.standby(&mInput->stream->common); 6265} 6266 6267// RecordThread::createRecordTrack_l() must be called with AudioFlinger::mLock held 6268sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 6269 const sp<AudioFlinger::Client>& client, 6270 uint32_t sampleRate, 6271 audio_format_t format, 6272 audio_channel_mask_t channelMask, 6273 size_t *pFrameCount, 6274 audio_session_t sessionId, 6275 size_t *notificationFrames, 6276 int uid, 6277 IAudioFlinger::track_flags_t *flags, 6278 pid_t tid, 6279 status_t *status) 6280{ 6281 size_t frameCount = *pFrameCount; 6282 sp<RecordTrack> track; 6283 status_t lStatus; 6284 6285 // client expresses a preference for FAST, but we get the final say 6286 if (*flags & IAudioFlinger::TRACK_FAST) { 6287 if ( 6288 // we formerly checked for a callback handler (non-0 tid), 6289 // but that is no longer required for TRANSFER_OBTAIN mode 6290 // 6291 // frame count is not specified, or is exactly the pipe depth 6292 ((frameCount == 0) || (frameCount == mPipeFramesP2)) && 6293 // PCM data 6294 audio_is_linear_pcm(format) && 6295 // hardware format 6296 (format == mFormat) && 6297 // hardware channel mask 6298 (channelMask == mChannelMask) && 6299 // hardware sample rate 6300 (sampleRate == mSampleRate) && 6301 // record thread has an associated fast capture 6302 hasFastCapture() && 6303 // there are sufficient fast track slots available 6304 mFastTrackAvail 6305 ) { 6306 ALOGV("AUDIO_INPUT_FLAG_FAST accepted: frameCount=%zu mFrameCount=%zu", 6307 frameCount, mFrameCount); 6308 } else { 6309 ALOGV("AUDIO_INPUT_FLAG_FAST denied: frameCount=%zu mFrameCount=%zu mPipeFramesP2=%zu " 6310 "format=%#x isLinear=%d channelMask=%#x sampleRate=%u mSampleRate=%u " 6311 "hasFastCapture=%d tid=%d mFastTrackAvail=%d", 6312 frameCount, mFrameCount, mPipeFramesP2, 6313 format, audio_is_linear_pcm(format), channelMask, sampleRate, mSampleRate, 6314 hasFastCapture(), tid, mFastTrackAvail); 6315 *flags &= ~IAudioFlinger::TRACK_FAST; 6316 } 6317 } 6318 6319 // compute track buffer size in frames, and suggest the notification frame count 6320 if (*flags & IAudioFlinger::TRACK_FAST) { 6321 // fast track: frame count is exactly the pipe depth 6322 frameCount = mPipeFramesP2; 6323 // ignore requested notificationFrames, and always notify exactly once every HAL buffer 6324 *notificationFrames = mFrameCount; 6325 } else { 6326 // not fast track: max notification period is resampled equivalent of one HAL buffer time 6327 // or 20 ms if there is a fast capture 6328 // TODO This could be a roundupRatio inline, and const 6329 size_t maxNotificationFrames = ((int64_t) (hasFastCapture() ? mSampleRate/50 : mFrameCount) 6330 * sampleRate + mSampleRate - 1) / mSampleRate; 6331 // minimum number of notification periods is at least kMinNotifications, 6332 // and at least kMinMs rounded up to a whole notification period (minNotificationsByMs) 6333 static const size_t kMinNotifications = 3; 6334 static const uint32_t kMinMs = 30; 6335 // TODO This could be a roundupRatio inline 6336 const size_t minFramesByMs = (sampleRate * kMinMs + 1000 - 1) / 1000; 6337 // TODO This could be a roundupRatio inline 6338 const size_t minNotificationsByMs = (minFramesByMs + maxNotificationFrames - 1) / 6339 maxNotificationFrames; 6340 const size_t minFrameCount = maxNotificationFrames * 6341 max(kMinNotifications, minNotificationsByMs); 6342 frameCount = max(frameCount, minFrameCount); 6343 if (*notificationFrames == 0 || *notificationFrames > maxNotificationFrames) { 6344 *notificationFrames = maxNotificationFrames; 6345 } 6346 } 6347 *pFrameCount = frameCount; 6348 6349 lStatus = initCheck(); 6350 if (lStatus != NO_ERROR) { 6351 ALOGE("createRecordTrack_l() audio driver not initialized"); 6352 goto Exit; 6353 } 6354 6355 { // scope for mLock 6356 Mutex::Autolock _l(mLock); 6357 6358 track = new RecordTrack(this, client, sampleRate, 6359 format, channelMask, frameCount, NULL, sessionId, uid, 6360 *flags, TrackBase::TYPE_DEFAULT); 6361 6362 lStatus = track->initCheck(); 6363 if (lStatus != NO_ERROR) { 6364 ALOGE("createRecordTrack_l() initCheck failed %d; no control block?", lStatus); 6365 // track must be cleared from the caller as the caller has the AF lock 6366 goto Exit; 6367 } 6368 mTracks.add(track); 6369 6370 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 6371 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 6372 mAudioFlinger->btNrecIsOff(); 6373 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 6374 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 6375 6376 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 6377 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 6378 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 6379 // so ask activity manager to do this on our behalf 6380 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 6381 } 6382 } 6383 6384 lStatus = NO_ERROR; 6385 6386Exit: 6387 *status = lStatus; 6388 return track; 6389} 6390 6391status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, 6392 AudioSystem::sync_event_t event, 6393 audio_session_t triggerSession) 6394{ 6395 ALOGV("RecordThread::start event %d, triggerSession %d", event, triggerSession); 6396 sp<ThreadBase> strongMe = this; 6397 status_t status = NO_ERROR; 6398 6399 if (event == AudioSystem::SYNC_EVENT_NONE) { 6400 recordTrack->clearSyncStartEvent(); 6401 } else if (event != AudioSystem::SYNC_EVENT_SAME) { 6402 recordTrack->mSyncStartEvent = mAudioFlinger->createSyncEvent(event, 6403 triggerSession, 6404 recordTrack->sessionId(), 6405 syncStartEventCallback, 6406 recordTrack); 6407 // Sync event can be cancelled by the trigger session if the track is not in a 6408 // compatible state in which case we start record immediately 6409 if (recordTrack->mSyncStartEvent->isCancelled()) { 6410 recordTrack->clearSyncStartEvent(); 6411 } else { 6412 // do not wait for the event for more than AudioSystem::kSyncRecordStartTimeOutMs 6413 recordTrack->mFramesToDrop = - 6414 ((AudioSystem::kSyncRecordStartTimeOutMs * recordTrack->mSampleRate) / 1000); 6415 } 6416 } 6417 6418 { 6419 // This section is a rendezvous between binder thread executing start() and RecordThread 6420 AutoMutex lock(mLock); 6421 if (mActiveTracks.indexOf(recordTrack) >= 0) { 6422 if (recordTrack->mState == TrackBase::PAUSING) { 6423 ALOGV("active record track PAUSING -> ACTIVE"); 6424 recordTrack->mState = TrackBase::ACTIVE; 6425 } else { 6426 ALOGV("active record track state %d", recordTrack->mState); 6427 } 6428 return status; 6429 } 6430 6431 // TODO consider other ways of handling this, such as changing the state to :STARTING and 6432 // adding the track to mActiveTracks after returning from AudioSystem::startInput(), 6433 // or using a separate command thread 6434 recordTrack->mState = TrackBase::STARTING_1; 6435 mActiveTracks.add(recordTrack); 6436 mActiveTracksGen++; 6437 status_t status = NO_ERROR; 6438 if (recordTrack->isExternalTrack()) { 6439 mLock.unlock(); 6440 status = AudioSystem::startInput(mId, recordTrack->sessionId()); 6441 mLock.lock(); 6442 // FIXME should verify that recordTrack is still in mActiveTracks 6443 if (status != NO_ERROR) { 6444 mActiveTracks.remove(recordTrack); 6445 mActiveTracksGen++; 6446 recordTrack->clearSyncStartEvent(); 6447 ALOGV("RecordThread::start error %d", status); 6448 return status; 6449 } 6450 } 6451 // Catch up with current buffer indices if thread is already running. 6452 // This is what makes a new client discard all buffered data. If the track's mRsmpInFront 6453 // was initialized to some value closer to the thread's mRsmpInFront, then the track could 6454 // see previously buffered data before it called start(), but with greater risk of overrun. 6455 6456 recordTrack->mResamplerBufferProvider->reset(); 6457 // clear any converter state as new data will be discontinuous 6458 recordTrack->mRecordBufferConverter->reset(); 6459 recordTrack->mState = TrackBase::STARTING_2; 6460 // signal thread to start 6461 mWaitWorkCV.broadcast(); 6462 if (mActiveTracks.indexOf(recordTrack) < 0) { 6463 ALOGV("Record failed to start"); 6464 status = BAD_VALUE; 6465 goto startError; 6466 } 6467 return status; 6468 } 6469 6470startError: 6471 if (recordTrack->isExternalTrack()) { 6472 AudioSystem::stopInput(mId, recordTrack->sessionId()); 6473 } 6474 recordTrack->clearSyncStartEvent(); 6475 // FIXME I wonder why we do not reset the state here? 6476 return status; 6477} 6478 6479void AudioFlinger::RecordThread::syncStartEventCallback(const wp<SyncEvent>& event) 6480{ 6481 sp<SyncEvent> strongEvent = event.promote(); 6482 6483 if (strongEvent != 0) { 6484 sp<RefBase> ptr = strongEvent->cookie().promote(); 6485 if (ptr != 0) { 6486 RecordTrack *recordTrack = (RecordTrack *)ptr.get(); 6487 recordTrack->handleSyncStartEvent(strongEvent); 6488 } 6489 } 6490} 6491 6492bool AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 6493 ALOGV("RecordThread::stop"); 6494 AutoMutex _l(mLock); 6495 if (mActiveTracks.indexOf(recordTrack) != 0 || recordTrack->mState == TrackBase::PAUSING) { 6496 return false; 6497 } 6498 // note that threadLoop may still be processing the track at this point [without lock] 6499 recordTrack->mState = TrackBase::PAUSING; 6500 // do not wait for mStartStopCond if exiting 6501 if (exitPending()) { 6502 return true; 6503 } 6504 // FIXME incorrect usage of wait: no explicit predicate or loop 6505 mStartStopCond.wait(mLock); 6506 // if we have been restarted, recordTrack is in mActiveTracks here 6507 if (exitPending() || mActiveTracks.indexOf(recordTrack) != 0) { 6508 ALOGV("Record stopped OK"); 6509 return true; 6510 } 6511 return false; 6512} 6513 6514bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event __unused) const 6515{ 6516 return false; 6517} 6518 6519status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event __unused) 6520{ 6521#if 0 // This branch is currently dead code, but is preserved in case it will be needed in future 6522 if (!isValidSyncEvent(event)) { 6523 return BAD_VALUE; 6524 } 6525 6526 audio_session_t eventSession = event->triggerSession(); 6527 status_t ret = NAME_NOT_FOUND; 6528 6529 Mutex::Autolock _l(mLock); 6530 6531 for (size_t i = 0; i < mTracks.size(); i++) { 6532 sp<RecordTrack> track = mTracks[i]; 6533 if (eventSession == track->sessionId()) { 6534 (void) track->setSyncEvent(event); 6535 ret = NO_ERROR; 6536 } 6537 } 6538 return ret; 6539#else 6540 return BAD_VALUE; 6541#endif 6542} 6543 6544// destroyTrack_l() must be called with ThreadBase::mLock held 6545void AudioFlinger::RecordThread::destroyTrack_l(const sp<RecordTrack>& track) 6546{ 6547 track->terminate(); 6548 track->mState = TrackBase::STOPPED; 6549 // active tracks are removed by threadLoop() 6550 if (mActiveTracks.indexOf(track) < 0) { 6551 removeTrack_l(track); 6552 } 6553} 6554 6555void AudioFlinger::RecordThread::removeTrack_l(const sp<RecordTrack>& track) 6556{ 6557 mTracks.remove(track); 6558 // need anything related to effects here? 6559 if (track->isFastTrack()) { 6560 ALOG_ASSERT(!mFastTrackAvail); 6561 mFastTrackAvail = true; 6562 } 6563} 6564 6565void AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 6566{ 6567 dumpInternals(fd, args); 6568 dumpTracks(fd, args); 6569 dumpEffectChains(fd, args); 6570} 6571 6572void AudioFlinger::RecordThread::dumpInternals(int fd, const Vector<String16>& args) 6573{ 6574 dprintf(fd, "\nInput thread %p:\n", this); 6575 6576 dumpBase(fd, args); 6577 6578 if (mActiveTracks.size() == 0) { 6579 dprintf(fd, " No active record clients\n"); 6580 } 6581 dprintf(fd, " Fast capture thread: %s\n", hasFastCapture() ? "yes" : "no"); 6582 dprintf(fd, " Fast track available: %s\n", mFastTrackAvail ? "yes" : "no"); 6583 6584 // Make a non-atomic copy of fast capture dump state so it won't change underneath us 6585 // while we are dumping it. It may be inconsistent, but it won't mutate! 6586 // This is a large object so we place it on the heap. 6587 // FIXME 25972958: Need an intelligent copy constructor that does not touch unused pages. 6588 const FastCaptureDumpState *copy = new FastCaptureDumpState(mFastCaptureDumpState); 6589 copy->dump(fd); 6590 delete copy; 6591} 6592 6593void AudioFlinger::RecordThread::dumpTracks(int fd, const Vector<String16>& args __unused) 6594{ 6595 const size_t SIZE = 256; 6596 char buffer[SIZE]; 6597 String8 result; 6598 6599 size_t numtracks = mTracks.size(); 6600 size_t numactive = mActiveTracks.size(); 6601 size_t numactiveseen = 0; 6602 dprintf(fd, " %zu Tracks", numtracks); 6603 if (numtracks) { 6604 dprintf(fd, " of which %zu are active\n", numactive); 6605 RecordTrack::appendDumpHeader(result); 6606 for (size_t i = 0; i < numtracks ; ++i) { 6607 sp<RecordTrack> track = mTracks[i]; 6608 if (track != 0) { 6609 bool active = mActiveTracks.indexOf(track) >= 0; 6610 if (active) { 6611 numactiveseen++; 6612 } 6613 track->dump(buffer, SIZE, active); 6614 result.append(buffer); 6615 } 6616 } 6617 } else { 6618 dprintf(fd, "\n"); 6619 } 6620 6621 if (numactiveseen != numactive) { 6622 snprintf(buffer, SIZE, " The following tracks are in the active list but" 6623 " not in the track list\n"); 6624 result.append(buffer); 6625 RecordTrack::appendDumpHeader(result); 6626 for (size_t i = 0; i < numactive; ++i) { 6627 sp<RecordTrack> track = mActiveTracks[i]; 6628 if (mTracks.indexOf(track) < 0) { 6629 track->dump(buffer, SIZE, true); 6630 result.append(buffer); 6631 } 6632 } 6633 6634 } 6635 write(fd, result.string(), result.size()); 6636} 6637 6638 6639void AudioFlinger::RecordThread::ResamplerBufferProvider::reset() 6640{ 6641 sp<ThreadBase> threadBase = mRecordTrack->mThread.promote(); 6642 RecordThread *recordThread = (RecordThread *) threadBase.get(); 6643 mRsmpInFront = recordThread->mRsmpInRear; 6644 mRsmpInUnrel = 0; 6645} 6646 6647void AudioFlinger::RecordThread::ResamplerBufferProvider::sync( 6648 size_t *framesAvailable, bool *hasOverrun) 6649{ 6650 sp<ThreadBase> threadBase = mRecordTrack->mThread.promote(); 6651 RecordThread *recordThread = (RecordThread *) threadBase.get(); 6652 const int32_t rear = recordThread->mRsmpInRear; 6653 const int32_t front = mRsmpInFront; 6654 const ssize_t filled = rear - front; 6655 6656 size_t framesIn; 6657 bool overrun = false; 6658 if (filled < 0) { 6659 // should not happen, but treat like a massive overrun and re-sync 6660 framesIn = 0; 6661 mRsmpInFront = rear; 6662 overrun = true; 6663 } else if ((size_t) filled <= recordThread->mRsmpInFrames) { 6664 framesIn = (size_t) filled; 6665 } else { 6666 // client is not keeping up with server, but give it latest data 6667 framesIn = recordThread->mRsmpInFrames; 6668 mRsmpInFront = /* front = */ rear - framesIn; 6669 overrun = true; 6670 } 6671 if (framesAvailable != NULL) { 6672 *framesAvailable = framesIn; 6673 } 6674 if (hasOverrun != NULL) { 6675 *hasOverrun = overrun; 6676 } 6677} 6678 6679// AudioBufferProvider interface 6680status_t AudioFlinger::RecordThread::ResamplerBufferProvider::getNextBuffer( 6681 AudioBufferProvider::Buffer* buffer) 6682{ 6683 sp<ThreadBase> threadBase = mRecordTrack->mThread.promote(); 6684 if (threadBase == 0) { 6685 buffer->frameCount = 0; 6686 buffer->raw = NULL; 6687 return NOT_ENOUGH_DATA; 6688 } 6689 RecordThread *recordThread = (RecordThread *) threadBase.get(); 6690 int32_t rear = recordThread->mRsmpInRear; 6691 int32_t front = mRsmpInFront; 6692 ssize_t filled = rear - front; 6693 // FIXME should not be P2 (don't want to increase latency) 6694 // FIXME if client not keeping up, discard 6695 LOG_ALWAYS_FATAL_IF(!(0 <= filled && (size_t) filled <= recordThread->mRsmpInFrames)); 6696 // 'filled' may be non-contiguous, so return only the first contiguous chunk 6697 front &= recordThread->mRsmpInFramesP2 - 1; 6698 size_t part1 = recordThread->mRsmpInFramesP2 - front; 6699 if (part1 > (size_t) filled) { 6700 part1 = filled; 6701 } 6702 size_t ask = buffer->frameCount; 6703 ALOG_ASSERT(ask > 0); 6704 if (part1 > ask) { 6705 part1 = ask; 6706 } 6707 if (part1 == 0) { 6708 // out of data is fine since the resampler will return a short-count. 6709 buffer->raw = NULL; 6710 buffer->frameCount = 0; 6711 mRsmpInUnrel = 0; 6712 return NOT_ENOUGH_DATA; 6713 } 6714 6715 buffer->raw = (uint8_t*)recordThread->mRsmpInBuffer + front * recordThread->mFrameSize; 6716 buffer->frameCount = part1; 6717 mRsmpInUnrel = part1; 6718 return NO_ERROR; 6719} 6720 6721// AudioBufferProvider interface 6722void AudioFlinger::RecordThread::ResamplerBufferProvider::releaseBuffer( 6723 AudioBufferProvider::Buffer* buffer) 6724{ 6725 size_t stepCount = buffer->frameCount; 6726 if (stepCount == 0) { 6727 return; 6728 } 6729 ALOG_ASSERT(stepCount <= mRsmpInUnrel); 6730 mRsmpInUnrel -= stepCount; 6731 mRsmpInFront += stepCount; 6732 buffer->raw = NULL; 6733 buffer->frameCount = 0; 6734} 6735 6736AudioFlinger::RecordThread::RecordBufferConverter::RecordBufferConverter( 6737 audio_channel_mask_t srcChannelMask, audio_format_t srcFormat, 6738 uint32_t srcSampleRate, 6739 audio_channel_mask_t dstChannelMask, audio_format_t dstFormat, 6740 uint32_t dstSampleRate) : 6741 mSrcChannelMask(AUDIO_CHANNEL_INVALID), // updateParameters will set following vars 6742 // mSrcFormat 6743 // mSrcSampleRate 6744 // mDstChannelMask 6745 // mDstFormat 6746 // mDstSampleRate 6747 // mSrcChannelCount 6748 // mDstChannelCount 6749 // mDstFrameSize 6750 mBuf(NULL), mBufFrames(0), mBufFrameSize(0), 6751 mResampler(NULL), 6752 mIsLegacyDownmix(false), 6753 mIsLegacyUpmix(false), 6754 mRequiresFloat(false), 6755 mInputConverterProvider(NULL) 6756{ 6757 (void)updateParameters(srcChannelMask, srcFormat, srcSampleRate, 6758 dstChannelMask, dstFormat, dstSampleRate); 6759} 6760 6761AudioFlinger::RecordThread::RecordBufferConverter::~RecordBufferConverter() { 6762 free(mBuf); 6763 delete mResampler; 6764 delete mInputConverterProvider; 6765} 6766 6767size_t AudioFlinger::RecordThread::RecordBufferConverter::convert(void *dst, 6768 AudioBufferProvider *provider, size_t frames) 6769{ 6770 if (mInputConverterProvider != NULL) { 6771 mInputConverterProvider->setBufferProvider(provider); 6772 provider = mInputConverterProvider; 6773 } 6774 6775 if (mResampler == NULL) { 6776 ALOGVV("NO RESAMPLING sampleRate:%u mSrcFormat:%#x mDstFormat:%#x", 6777 mSrcSampleRate, mSrcFormat, mDstFormat); 6778 6779 AudioBufferProvider::Buffer buffer; 6780 for (size_t i = frames; i > 0; ) { 6781 buffer.frameCount = i; 6782 status_t status = provider->getNextBuffer(&buffer); 6783 if (status != OK || buffer.frameCount == 0) { 6784 frames -= i; // cannot fill request. 6785 break; 6786 } 6787 // format convert to destination buffer 6788 convertNoResampler(dst, buffer.raw, buffer.frameCount); 6789 6790 dst = (int8_t*)dst + buffer.frameCount * mDstFrameSize; 6791 i -= buffer.frameCount; 6792 provider->releaseBuffer(&buffer); 6793 } 6794 } else { 6795 ALOGVV("RESAMPLING mSrcSampleRate:%u mDstSampleRate:%u mSrcFormat:%#x mDstFormat:%#x", 6796 mSrcSampleRate, mDstSampleRate, mSrcFormat, mDstFormat); 6797 6798 // reallocate buffer if needed 6799 if (mBufFrameSize != 0 && mBufFrames < frames) { 6800 free(mBuf); 6801 mBufFrames = frames; 6802 (void)posix_memalign(&mBuf, 32, mBufFrames * mBufFrameSize); 6803 } 6804 // resampler accumulates, but we only have one source track 6805 memset(mBuf, 0, frames * mBufFrameSize); 6806 frames = mResampler->resample((int32_t*)mBuf, frames, provider); 6807 // format convert to destination buffer 6808 convertResampler(dst, mBuf, frames); 6809 } 6810 return frames; 6811} 6812 6813status_t AudioFlinger::RecordThread::RecordBufferConverter::updateParameters( 6814 audio_channel_mask_t srcChannelMask, audio_format_t srcFormat, 6815 uint32_t srcSampleRate, 6816 audio_channel_mask_t dstChannelMask, audio_format_t dstFormat, 6817 uint32_t dstSampleRate) 6818{ 6819 // quick evaluation if there is any change. 6820 if (mSrcFormat == srcFormat 6821 && mSrcChannelMask == srcChannelMask 6822 && mSrcSampleRate == srcSampleRate 6823 && mDstFormat == dstFormat 6824 && mDstChannelMask == dstChannelMask 6825 && mDstSampleRate == dstSampleRate) { 6826 return NO_ERROR; 6827 } 6828 6829 ALOGV("RecordBufferConverter updateParameters srcMask:%#x dstMask:%#x" 6830 " srcFormat:%#x dstFormat:%#x srcRate:%u dstRate:%u", 6831 srcChannelMask, dstChannelMask, srcFormat, dstFormat, srcSampleRate, dstSampleRate); 6832 const bool valid = 6833 audio_is_input_channel(srcChannelMask) 6834 && audio_is_input_channel(dstChannelMask) 6835 && audio_is_valid_format(srcFormat) && audio_is_linear_pcm(srcFormat) 6836 && audio_is_valid_format(dstFormat) && audio_is_linear_pcm(dstFormat) 6837 && (srcSampleRate <= dstSampleRate * AUDIO_RESAMPLER_DOWN_RATIO_MAX) 6838 ; // no upsampling checks for now 6839 if (!valid) { 6840 return BAD_VALUE; 6841 } 6842 6843 mSrcFormat = srcFormat; 6844 mSrcChannelMask = srcChannelMask; 6845 mSrcSampleRate = srcSampleRate; 6846 mDstFormat = dstFormat; 6847 mDstChannelMask = dstChannelMask; 6848 mDstSampleRate = dstSampleRate; 6849 6850 // compute derived parameters 6851 mSrcChannelCount = audio_channel_count_from_in_mask(srcChannelMask); 6852 mDstChannelCount = audio_channel_count_from_in_mask(dstChannelMask); 6853 mDstFrameSize = mDstChannelCount * audio_bytes_per_sample(mDstFormat); 6854 6855 // do we need to resample? 6856 delete mResampler; 6857 mResampler = NULL; 6858 if (mSrcSampleRate != mDstSampleRate) { 6859 mResampler = AudioResampler::create(AUDIO_FORMAT_PCM_FLOAT, 6860 mSrcChannelCount, mDstSampleRate); 6861 mResampler->setSampleRate(mSrcSampleRate); 6862 mResampler->setVolume(AudioMixer::UNITY_GAIN_FLOAT, AudioMixer::UNITY_GAIN_FLOAT); 6863 } 6864 6865 // are we running legacy channel conversion modes? 6866 mIsLegacyDownmix = (mSrcChannelMask == AUDIO_CHANNEL_IN_STEREO 6867 || mSrcChannelMask == AUDIO_CHANNEL_IN_FRONT_BACK) 6868 && mDstChannelMask == AUDIO_CHANNEL_IN_MONO; 6869 mIsLegacyUpmix = mSrcChannelMask == AUDIO_CHANNEL_IN_MONO 6870 && (mDstChannelMask == AUDIO_CHANNEL_IN_STEREO 6871 || mDstChannelMask == AUDIO_CHANNEL_IN_FRONT_BACK); 6872 6873 // do we need to process in float? 6874 mRequiresFloat = mResampler != NULL || mIsLegacyDownmix || mIsLegacyUpmix; 6875 6876 // do we need a staging buffer to convert for destination (we can still optimize this)? 6877 // we use mBufFrameSize > 0 to indicate both frame size as well as buffer necessity 6878 if (mResampler != NULL) { 6879 mBufFrameSize = max(mSrcChannelCount, FCC_2) 6880 * audio_bytes_per_sample(AUDIO_FORMAT_PCM_FLOAT); 6881 } else if (mIsLegacyUpmix || mIsLegacyDownmix) { // legacy modes always float 6882 mBufFrameSize = mDstChannelCount * audio_bytes_per_sample(AUDIO_FORMAT_PCM_FLOAT); 6883 } else if (mSrcChannelMask != mDstChannelMask && mDstFormat != mSrcFormat) { 6884 mBufFrameSize = mDstChannelCount * audio_bytes_per_sample(mSrcFormat); 6885 } else { 6886 mBufFrameSize = 0; 6887 } 6888 mBufFrames = 0; // force the buffer to be resized. 6889 6890 // do we need an input converter buffer provider to give us float? 6891 delete mInputConverterProvider; 6892 mInputConverterProvider = NULL; 6893 if (mRequiresFloat && mSrcFormat != AUDIO_FORMAT_PCM_FLOAT) { 6894 mInputConverterProvider = new ReformatBufferProvider( 6895 audio_channel_count_from_in_mask(mSrcChannelMask), 6896 mSrcFormat, 6897 AUDIO_FORMAT_PCM_FLOAT, 6898 256 /* provider buffer frame count */); 6899 } 6900 6901 // do we need a remixer to do channel mask conversion 6902 if (!mIsLegacyDownmix && !mIsLegacyUpmix && mSrcChannelMask != mDstChannelMask) { 6903 (void) memcpy_by_index_array_initialization_from_channel_mask( 6904 mIdxAry, ARRAY_SIZE(mIdxAry), mDstChannelMask, mSrcChannelMask); 6905 } 6906 return NO_ERROR; 6907} 6908 6909void AudioFlinger::RecordThread::RecordBufferConverter::convertNoResampler( 6910 void *dst, const void *src, size_t frames) 6911{ 6912 // src is native type unless there is legacy upmix or downmix, whereupon it is float. 6913 if (mBufFrameSize != 0 && mBufFrames < frames) { 6914 free(mBuf); 6915 mBufFrames = frames; 6916 (void)posix_memalign(&mBuf, 32, mBufFrames * mBufFrameSize); 6917 } 6918 // do we need to do legacy upmix and downmix? 6919 if (mIsLegacyUpmix || mIsLegacyDownmix) { 6920 void *dstBuf = mBuf != NULL ? mBuf : dst; 6921 if (mIsLegacyUpmix) { 6922 upmix_to_stereo_float_from_mono_float((float *)dstBuf, 6923 (const float *)src, frames); 6924 } else /*mIsLegacyDownmix */ { 6925 downmix_to_mono_float_from_stereo_float((float *)dstBuf, 6926 (const float *)src, frames); 6927 } 6928 if (mBuf != NULL) { 6929 memcpy_by_audio_format(dst, mDstFormat, mBuf, AUDIO_FORMAT_PCM_FLOAT, 6930 frames * mDstChannelCount); 6931 } 6932 return; 6933 } 6934 // do we need to do channel mask conversion? 6935 if (mSrcChannelMask != mDstChannelMask) { 6936 void *dstBuf = mBuf != NULL ? mBuf : dst; 6937 memcpy_by_index_array(dstBuf, mDstChannelCount, 6938 src, mSrcChannelCount, mIdxAry, audio_bytes_per_sample(mSrcFormat), frames); 6939 if (dstBuf == dst) { 6940 return; // format is the same 6941 } 6942 } 6943 // convert to destination buffer 6944 const void *convertBuf = mBuf != NULL ? mBuf : src; 6945 memcpy_by_audio_format(dst, mDstFormat, convertBuf, mSrcFormat, 6946 frames * mDstChannelCount); 6947} 6948 6949void AudioFlinger::RecordThread::RecordBufferConverter::convertResampler( 6950 void *dst, /*not-a-const*/ void *src, size_t frames) 6951{ 6952 // src buffer format is ALWAYS float when entering this routine 6953 if (mIsLegacyUpmix) { 6954 ; // mono to stereo already handled by resampler 6955 } else if (mIsLegacyDownmix 6956 || (mSrcChannelMask == mDstChannelMask && mSrcChannelCount == 1)) { 6957 // the resampler outputs stereo for mono input channel (a feature?) 6958 // must convert to mono 6959 downmix_to_mono_float_from_stereo_float((float *)src, 6960 (const float *)src, frames); 6961 } else if (mSrcChannelMask != mDstChannelMask) { 6962 // convert to mono channel again for channel mask conversion (could be skipped 6963 // with further optimization). 6964 if (mSrcChannelCount == 1) { 6965 downmix_to_mono_float_from_stereo_float((float *)src, 6966 (const float *)src, frames); 6967 } 6968 // convert to destination format (in place, OK as float is larger than other types) 6969 if (mDstFormat != AUDIO_FORMAT_PCM_FLOAT) { 6970 memcpy_by_audio_format(src, mDstFormat, src, AUDIO_FORMAT_PCM_FLOAT, 6971 frames * mSrcChannelCount); 6972 } 6973 // channel convert and save to dst 6974 memcpy_by_index_array(dst, mDstChannelCount, 6975 src, mSrcChannelCount, mIdxAry, audio_bytes_per_sample(mDstFormat), frames); 6976 return; 6977 } 6978 // convert to destination format and save to dst 6979 memcpy_by_audio_format(dst, mDstFormat, src, AUDIO_FORMAT_PCM_FLOAT, 6980 frames * mDstChannelCount); 6981} 6982 6983bool AudioFlinger::RecordThread::checkForNewParameter_l(const String8& keyValuePair, 6984 status_t& status) 6985{ 6986 bool reconfig = false; 6987 6988 status = NO_ERROR; 6989 6990 audio_format_t reqFormat = mFormat; 6991 uint32_t samplingRate = mSampleRate; 6992 // TODO this may change if we want to support capture from HDMI PCM multi channel (e.g on TVs). 6993 audio_channel_mask_t channelMask = audio_channel_in_mask_from_count(mChannelCount); 6994 6995 AudioParameter param = AudioParameter(keyValuePair); 6996 int value; 6997 6998 // scope for AutoPark extends to end of method 6999 AutoPark<FastCapture> park(mFastCapture); 7000 7001 // TODO Investigate when this code runs. Check with audio policy when a sample rate and 7002 // channel count change can be requested. Do we mandate the first client defines the 7003 // HAL sampling rate and channel count or do we allow changes on the fly? 7004 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 7005 samplingRate = value; 7006 reconfig = true; 7007 } 7008 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 7009 if (!audio_is_linear_pcm((audio_format_t) value)) { 7010 status = BAD_VALUE; 7011 } else { 7012 reqFormat = (audio_format_t) value; 7013 reconfig = true; 7014 } 7015 } 7016 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 7017 audio_channel_mask_t mask = (audio_channel_mask_t) value; 7018 if (!audio_is_input_channel(mask) || 7019 audio_channel_count_from_in_mask(mask) > FCC_8) { 7020 status = BAD_VALUE; 7021 } else { 7022 channelMask = mask; 7023 reconfig = true; 7024 } 7025 } 7026 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 7027 // do not accept frame count changes if tracks are open as the track buffer 7028 // size depends on frame count and correct behavior would not be guaranteed 7029 // if frame count is changed after track creation 7030 if (mActiveTracks.size() > 0) { 7031 status = INVALID_OPERATION; 7032 } else { 7033 reconfig = true; 7034 } 7035 } 7036 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 7037 // forward device change to effects that have requested to be 7038 // aware of attached audio device. 7039 for (size_t i = 0; i < mEffectChains.size(); i++) { 7040 mEffectChains[i]->setDevice_l(value); 7041 } 7042 7043 // store input device and output device but do not forward output device to audio HAL. 7044 // Note that status is ignored by the caller for output device 7045 // (see AudioFlinger::setParameters() 7046 if (audio_is_output_devices(value)) { 7047 mOutDevice = value; 7048 status = BAD_VALUE; 7049 } else { 7050 mInDevice = value; 7051 if (value != AUDIO_DEVICE_NONE) { 7052 mPrevInDevice = value; 7053 } 7054 // disable AEC and NS if the device is a BT SCO headset supporting those 7055 // pre processings 7056 if (mTracks.size() > 0) { 7057 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 7058 mAudioFlinger->btNrecIsOff(); 7059 for (size_t i = 0; i < mTracks.size(); i++) { 7060 sp<RecordTrack> track = mTracks[i]; 7061 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId()); 7062 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId()); 7063 } 7064 } 7065 } 7066 } 7067 if (param.getInt(String8(AudioParameter::keyInputSource), value) == NO_ERROR && 7068 mAudioSource != (audio_source_t)value) { 7069 // forward device change to effects that have requested to be 7070 // aware of attached audio device. 7071 for (size_t i = 0; i < mEffectChains.size(); i++) { 7072 mEffectChains[i]->setAudioSource_l((audio_source_t)value); 7073 } 7074 mAudioSource = (audio_source_t)value; 7075 } 7076 7077 if (status == NO_ERROR) { 7078 status = mInput->stream->common.set_parameters(&mInput->stream->common, 7079 keyValuePair.string()); 7080 if (status == INVALID_OPERATION) { 7081 inputStandBy(); 7082 status = mInput->stream->common.set_parameters(&mInput->stream->common, 7083 keyValuePair.string()); 7084 } 7085 if (reconfig) { 7086 if (status == BAD_VALUE && 7087 audio_is_linear_pcm(mInput->stream->common.get_format(&mInput->stream->common)) && 7088 audio_is_linear_pcm(reqFormat) && 7089 (mInput->stream->common.get_sample_rate(&mInput->stream->common) 7090 <= (AUDIO_RESAMPLER_DOWN_RATIO_MAX * samplingRate)) && 7091 audio_channel_count_from_in_mask( 7092 mInput->stream->common.get_channels(&mInput->stream->common)) <= FCC_8) { 7093 status = NO_ERROR; 7094 } 7095 if (status == NO_ERROR) { 7096 readInputParameters_l(); 7097 sendIoConfigEvent_l(AUDIO_INPUT_CONFIG_CHANGED); 7098 } 7099 } 7100 } 7101 7102 return reconfig; 7103} 7104 7105String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 7106{ 7107 Mutex::Autolock _l(mLock); 7108 if (initCheck() != NO_ERROR) { 7109 return String8(); 7110 } 7111 7112 char *s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 7113 const String8 out_s8(s); 7114 free(s); 7115 return out_s8; 7116} 7117 7118void AudioFlinger::RecordThread::ioConfigChanged(audio_io_config_event event, pid_t pid) { 7119 sp<AudioIoDescriptor> desc = new AudioIoDescriptor(); 7120 7121 desc->mIoHandle = mId; 7122 7123 switch (event) { 7124 case AUDIO_INPUT_OPENED: 7125 case AUDIO_INPUT_CONFIG_CHANGED: 7126 desc->mPatch = mPatch; 7127 desc->mChannelMask = mChannelMask; 7128 desc->mSamplingRate = mSampleRate; 7129 desc->mFormat = mFormat; 7130 desc->mFrameCount = mFrameCount; 7131 desc->mFrameCountHAL = mFrameCount; 7132 desc->mLatency = 0; 7133 break; 7134 7135 case AUDIO_INPUT_CLOSED: 7136 default: 7137 break; 7138 } 7139 mAudioFlinger->ioConfigChanged(event, desc, pid); 7140} 7141 7142void AudioFlinger::RecordThread::readInputParameters_l() 7143{ 7144 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 7145 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 7146 mChannelCount = audio_channel_count_from_in_mask(mChannelMask); 7147 if (mChannelCount > FCC_8) { 7148 ALOGE("HAL channel count %d > %d", mChannelCount, FCC_8); 7149 } 7150 mHALFormat = mInput->stream->common.get_format(&mInput->stream->common); 7151 mFormat = mHALFormat; 7152 if (!audio_is_linear_pcm(mFormat)) { 7153 ALOGE("HAL format %#x is not linear pcm", mFormat); 7154 } 7155 mFrameSize = audio_stream_in_frame_size(mInput->stream); 7156 mBufferSize = mInput->stream->common.get_buffer_size(&mInput->stream->common); 7157 mFrameCount = mBufferSize / mFrameSize; 7158 // This is the formula for calculating the temporary buffer size. 7159 // With 7 HAL buffers, we can guarantee ability to down-sample the input by ratio of 6:1 to 7160 // 1 full output buffer, regardless of the alignment of the available input. 7161 // The value is somewhat arbitrary, and could probably be even larger. 7162 // A larger value should allow more old data to be read after a track calls start(), 7163 // without increasing latency. 7164 // 7165 // Note this is independent of the maximum downsampling ratio permitted for capture. 7166 mRsmpInFrames = mFrameCount * 7; 7167 mRsmpInFramesP2 = roundup(mRsmpInFrames); 7168 free(mRsmpInBuffer); 7169 mRsmpInBuffer = NULL; 7170 7171 // TODO optimize audio capture buffer sizes ... 7172 // Here we calculate the size of the sliding buffer used as a source 7173 // for resampling. mRsmpInFramesP2 is currently roundup(mFrameCount * 7). 7174 // For current HAL frame counts, this is usually 2048 = 40 ms. It would 7175 // be better to have it derived from the pipe depth in the long term. 7176 // The current value is higher than necessary. However it should not add to latency. 7177 7178 // Over-allocate beyond mRsmpInFramesP2 to permit a HAL read past end of buffer 7179 size_t bufferSize = (mRsmpInFramesP2 + mFrameCount - 1) * mFrameSize; 7180 (void)posix_memalign(&mRsmpInBuffer, 32, bufferSize); 7181 memset(mRsmpInBuffer, 0, bufferSize); // if posix_memalign fails, will segv here. 7182 7183 // AudioRecord mSampleRate and mChannelCount are constant due to AudioRecord API constraints. 7184 // But if thread's mSampleRate or mChannelCount changes, how will that affect active tracks? 7185} 7186 7187uint32_t AudioFlinger::RecordThread::getInputFramesLost() 7188{ 7189 Mutex::Autolock _l(mLock); 7190 if (initCheck() != NO_ERROR) { 7191 return 0; 7192 } 7193 7194 return mInput->stream->get_input_frames_lost(mInput->stream); 7195} 7196 7197uint32_t AudioFlinger::RecordThread::hasAudioSession(audio_session_t sessionId) const 7198{ 7199 Mutex::Autolock _l(mLock); 7200 uint32_t result = 0; 7201 if (getEffectChain_l(sessionId) != 0) { 7202 result = EFFECT_SESSION; 7203 } 7204 7205 for (size_t i = 0; i < mTracks.size(); ++i) { 7206 if (sessionId == mTracks[i]->sessionId()) { 7207 result |= TRACK_SESSION; 7208 break; 7209 } 7210 } 7211 7212 return result; 7213} 7214 7215KeyedVector<audio_session_t, bool> AudioFlinger::RecordThread::sessionIds() const 7216{ 7217 KeyedVector<audio_session_t, bool> ids; 7218 Mutex::Autolock _l(mLock); 7219 for (size_t j = 0; j < mTracks.size(); ++j) { 7220 sp<RecordThread::RecordTrack> track = mTracks[j]; 7221 audio_session_t sessionId = track->sessionId(); 7222 if (ids.indexOfKey(sessionId) < 0) { 7223 ids.add(sessionId, true); 7224 } 7225 } 7226 return ids; 7227} 7228 7229AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 7230{ 7231 Mutex::Autolock _l(mLock); 7232 AudioStreamIn *input = mInput; 7233 mInput = NULL; 7234 return input; 7235} 7236 7237// this method must always be called either with ThreadBase mLock held or inside the thread loop 7238audio_stream_t* AudioFlinger::RecordThread::stream() const 7239{ 7240 if (mInput == NULL) { 7241 return NULL; 7242 } 7243 return &mInput->stream->common; 7244} 7245 7246status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 7247{ 7248 // only one chain per input thread 7249 if (mEffectChains.size() != 0) { 7250 ALOGW("addEffectChain_l() already one chain %p on thread %p", chain.get(), this); 7251 return INVALID_OPERATION; 7252 } 7253 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 7254 chain->setThread(this); 7255 chain->setInBuffer(NULL); 7256 chain->setOutBuffer(NULL); 7257 7258 checkSuspendOnAddEffectChain_l(chain); 7259 7260 // make sure enabled pre processing effects state is communicated to the HAL as we 7261 // just moved them to a new input stream. 7262 chain->syncHalEffectsState(); 7263 7264 mEffectChains.add(chain); 7265 7266 return NO_ERROR; 7267} 7268 7269size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 7270{ 7271 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 7272 ALOGW_IF(mEffectChains.size() != 1, 7273 "removeEffectChain_l() %p invalid chain size %zu on thread %p", 7274 chain.get(), mEffectChains.size(), this); 7275 if (mEffectChains.size() == 1) { 7276 mEffectChains.removeAt(0); 7277 } 7278 return 0; 7279} 7280 7281status_t AudioFlinger::RecordThread::createAudioPatch_l(const struct audio_patch *patch, 7282 audio_patch_handle_t *handle) 7283{ 7284 status_t status = NO_ERROR; 7285 7286 // store new device and send to effects 7287 mInDevice = patch->sources[0].ext.device.type; 7288 mPatch = *patch; 7289 for (size_t i = 0; i < mEffectChains.size(); i++) { 7290 mEffectChains[i]->setDevice_l(mInDevice); 7291 } 7292 7293 // disable AEC and NS if the device is a BT SCO headset supporting those 7294 // pre processings 7295 if (mTracks.size() > 0) { 7296 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 7297 mAudioFlinger->btNrecIsOff(); 7298 for (size_t i = 0; i < mTracks.size(); i++) { 7299 sp<RecordTrack> track = mTracks[i]; 7300 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId()); 7301 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId()); 7302 } 7303 } 7304 7305 // store new source and send to effects 7306 if (mAudioSource != patch->sinks[0].ext.mix.usecase.source) { 7307 mAudioSource = patch->sinks[0].ext.mix.usecase.source; 7308 for (size_t i = 0; i < mEffectChains.size(); i++) { 7309 mEffectChains[i]->setAudioSource_l(mAudioSource); 7310 } 7311 } 7312 7313 if (mInput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 7314 audio_hw_device_t *hwDevice = mInput->audioHwDev->hwDevice(); 7315 status = hwDevice->create_audio_patch(hwDevice, 7316 patch->num_sources, 7317 patch->sources, 7318 patch->num_sinks, 7319 patch->sinks, 7320 handle); 7321 } else { 7322 char *address; 7323 if (strcmp(patch->sources[0].ext.device.address, "") != 0) { 7324 address = audio_device_address_to_parameter( 7325 patch->sources[0].ext.device.type, 7326 patch->sources[0].ext.device.address); 7327 } else { 7328 address = (char *)calloc(1, 1); 7329 } 7330 AudioParameter param = AudioParameter(String8(address)); 7331 free(address); 7332 param.addInt(String8(AUDIO_PARAMETER_STREAM_ROUTING), 7333 (int)patch->sources[0].ext.device.type); 7334 param.addInt(String8(AUDIO_PARAMETER_STREAM_INPUT_SOURCE), 7335 (int)patch->sinks[0].ext.mix.usecase.source); 7336 status = mInput->stream->common.set_parameters(&mInput->stream->common, 7337 param.toString().string()); 7338 *handle = AUDIO_PATCH_HANDLE_NONE; 7339 } 7340 7341 if (mInDevice != mPrevInDevice) { 7342 sendIoConfigEvent_l(AUDIO_INPUT_CONFIG_CHANGED); 7343 mPrevInDevice = mInDevice; 7344 } 7345 7346 return status; 7347} 7348 7349status_t AudioFlinger::RecordThread::releaseAudioPatch_l(const audio_patch_handle_t handle) 7350{ 7351 status_t status = NO_ERROR; 7352 7353 mInDevice = AUDIO_DEVICE_NONE; 7354 7355 if (mInput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 7356 audio_hw_device_t *hwDevice = mInput->audioHwDev->hwDevice(); 7357 status = hwDevice->release_audio_patch(hwDevice, handle); 7358 } else { 7359 AudioParameter param; 7360 param.addInt(String8(AUDIO_PARAMETER_STREAM_ROUTING), 0); 7361 status = mInput->stream->common.set_parameters(&mInput->stream->common, 7362 param.toString().string()); 7363 } 7364 return status; 7365} 7366 7367void AudioFlinger::RecordThread::addPatchRecord(const sp<PatchRecord>& record) 7368{ 7369 Mutex::Autolock _l(mLock); 7370 mTracks.add(record); 7371} 7372 7373void AudioFlinger::RecordThread::deletePatchRecord(const sp<PatchRecord>& record) 7374{ 7375 Mutex::Autolock _l(mLock); 7376 destroyTrack_l(record); 7377} 7378 7379void AudioFlinger::RecordThread::getAudioPortConfig(struct audio_port_config *config) 7380{ 7381 ThreadBase::getAudioPortConfig(config); 7382 config->role = AUDIO_PORT_ROLE_SINK; 7383 config->ext.mix.hw_module = mInput->audioHwDev->handle(); 7384 config->ext.mix.usecase.source = mAudioSource; 7385} 7386 7387} // namespace android 7388