Threads.cpp revision 57c4e6f7464d458eb52d209c2a63524913d6406d
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// ---------------------------------------------------------------------------- 75 76// Note: the following macro is used for extremely verbose logging message. In 77// order to run with ALOG_ASSERT turned on, we need to have LOG_NDEBUG set to 78// 0; but one side effect of this is to turn all LOGV's as well. Some messages 79// are so verbose that we want to suppress them even when we have ALOG_ASSERT 80// turned on. Do not uncomment the #def below unless you really know what you 81// are doing and want to see all of the extremely verbose messages. 82//#define VERY_VERY_VERBOSE_LOGGING 83#ifdef VERY_VERY_VERBOSE_LOGGING 84#define ALOGVV ALOGV 85#else 86#define ALOGVV(a...) do { } while(0) 87#endif 88 89// TODO: Move these macro/inlines to a header file. 90#define max(a, b) ((a) > (b) ? (a) : (b)) 91template <typename T> 92static inline T min(const T& a, const T& b) 93{ 94 return a < b ? a : b; 95} 96 97#ifndef ARRAY_SIZE 98#define ARRAY_SIZE(a) (sizeof(a) / sizeof(a[0])) 99#endif 100 101namespace android { 102 103// retry counts for buffer fill timeout 104// 50 * ~20msecs = 1 second 105static const int8_t kMaxTrackRetries = 50; 106static const int8_t kMaxTrackStartupRetries = 50; 107// allow less retry attempts on direct output thread. 108// direct outputs can be a scarce resource in audio hardware and should 109// be released as quickly as possible. 110static const int8_t kMaxTrackRetriesDirect = 2; 111// retry count before removing active track in case of underrun on offloaded thread: 112// we need to make sure that AudioTrack client has enough time to send large buffers 113//FIXME may be more appropriate if expressed in time units. Need to revise how underrun is handled 114// for offloaded tracks 115static const int8_t kMaxTrackRetriesOffload = 10; 116static const int8_t kMaxTrackStartupRetriesOffload = 100; 117 118 119// don't warn about blocked writes or record buffer overflows more often than this 120static const nsecs_t kWarningThrottleNs = seconds(5); 121 122// RecordThread loop sleep time upon application overrun or audio HAL read error 123static const int kRecordThreadSleepUs = 5000; 124 125// maximum time to wait in sendConfigEvent_l() for a status to be received 126static const nsecs_t kConfigEventTimeoutNs = seconds(2); 127 128// minimum sleep time for the mixer thread loop when tracks are active but in underrun 129static const uint32_t kMinThreadSleepTimeUs = 5000; 130// maximum divider applied to the active sleep time in the mixer thread loop 131static const uint32_t kMaxThreadSleepTimeShift = 2; 132 133// minimum normal sink buffer size, expressed in milliseconds rather than frames 134// FIXME This should be based on experimentally observed scheduling jitter 135static const uint32_t kMinNormalSinkBufferSizeMs = 20; 136// maximum normal sink buffer size 137static const uint32_t kMaxNormalSinkBufferSizeMs = 24; 138 139// minimum capture buffer size in milliseconds to _not_ need a fast capture thread 140// FIXME This should be based on experimentally observed scheduling jitter 141static const uint32_t kMinNormalCaptureBufferSizeMs = 12; 142 143// Offloaded output thread standby delay: allows track transition without going to standby 144static const nsecs_t kOffloadStandbyDelayNs = seconds(1); 145 146// Direct output thread minimum sleep time in idle or active(underrun) state 147static const nsecs_t kDirectMinSleepTimeUs = 10000; 148 149// Offloaded output bit rate in bits per second when unknown. 150// Used for sleep time calculation, so use a high default bitrate to be conservative on sleep time. 151static const uint32_t kOffloadDefaultBitRateBps = 1500000; 152 153 154// Whether to use fast mixer 155static const enum { 156 FastMixer_Never, // never initialize or use: for debugging only 157 FastMixer_Always, // always initialize and use, even if not needed: for debugging only 158 // normal mixer multiplier is 1 159 FastMixer_Static, // initialize if needed, then use all the time if initialized, 160 // multiplier is calculated based on min & max normal mixer buffer size 161 FastMixer_Dynamic, // initialize if needed, then use dynamically depending on track load, 162 // multiplier is calculated based on min & max normal mixer buffer size 163 // FIXME for FastMixer_Dynamic: 164 // Supporting this option will require fixing HALs that can't handle large writes. 165 // For example, one HAL implementation returns an error from a large write, 166 // and another HAL implementation corrupts memory, possibly in the sample rate converter. 167 // We could either fix the HAL implementations, or provide a wrapper that breaks 168 // up large writes into smaller ones, and the wrapper would need to deal with scheduler. 169} kUseFastMixer = FastMixer_Static; 170 171// Whether to use fast capture 172static const enum { 173 FastCapture_Never, // never initialize or use: for debugging only 174 FastCapture_Always, // always initialize and use, even if not needed: for debugging only 175 FastCapture_Static, // initialize if needed, then use all the time if initialized 176} kUseFastCapture = FastCapture_Static; 177 178// Priorities for requestPriority 179static const int kPriorityAudioApp = 2; 180static const int kPriorityFastMixer = 3; 181static const int kPriorityFastCapture = 3; 182 183// IAudioFlinger::createTrack() reports back to client the total size of shared memory area 184// for the track. The client then sub-divides this into smaller buffers for its use. 185// Currently the client uses N-buffering by default, but doesn't tell us about the value of N. 186// So for now we just assume that client is double-buffered for fast tracks. 187// FIXME It would be better for client to tell AudioFlinger the value of N, 188// so AudioFlinger could allocate the right amount of memory. 189// See the client's minBufCount and mNotificationFramesAct calculations for details. 190 191// This is the default value, if not specified by property. 192static const int kFastTrackMultiplier = 2; 193 194// The minimum and maximum allowed values 195static const int kFastTrackMultiplierMin = 1; 196static const int kFastTrackMultiplierMax = 2; 197 198// The actual value to use, which can be specified per-device via property af.fast_track_multiplier. 199static int sFastTrackMultiplier = kFastTrackMultiplier; 200 201// See Thread::readOnlyHeap(). 202// Initially this heap is used to allocate client buffers for "fast" AudioRecord. 203// Eventually it will be the single buffer that FastCapture writes into via HAL read(), 204// and that all "fast" AudioRecord clients read from. In either case, the size can be small. 205static const size_t kRecordThreadReadOnlyHeapSize = 0x2000; 206 207// ---------------------------------------------------------------------------- 208 209static pthread_once_t sFastTrackMultiplierOnce = PTHREAD_ONCE_INIT; 210 211static void sFastTrackMultiplierInit() 212{ 213 char value[PROPERTY_VALUE_MAX]; 214 if (property_get("af.fast_track_multiplier", value, NULL) > 0) { 215 char *endptr; 216 unsigned long ul = strtoul(value, &endptr, 0); 217 if (*endptr == '\0' && kFastTrackMultiplierMin <= ul && ul <= kFastTrackMultiplierMax) { 218 sFastTrackMultiplier = (int) ul; 219 } 220 } 221} 222 223// ---------------------------------------------------------------------------- 224 225#ifdef ADD_BATTERY_DATA 226// To collect the amplifier usage 227static void addBatteryData(uint32_t params) { 228 sp<IMediaPlayerService> service = IMediaDeathNotifier::getMediaPlayerService(); 229 if (service == NULL) { 230 // it already logged 231 return; 232 } 233 234 service->addBatteryData(params); 235} 236#endif 237 238// Track the CLOCK_BOOTTIME versus CLOCK_MONOTONIC timebase offset 239struct { 240 // call when you acquire a partial wakelock 241 void acquire(const sp<IBinder> &wakeLockToken) { 242 pthread_mutex_lock(&mLock); 243 if (wakeLockToken.get() == nullptr) { 244 adjustTimebaseOffset(&mBoottimeOffset, ExtendedTimestamp::TIMEBASE_BOOTTIME); 245 } else { 246 if (mCount == 0) { 247 adjustTimebaseOffset(&mBoottimeOffset, ExtendedTimestamp::TIMEBASE_BOOTTIME); 248 } 249 ++mCount; 250 } 251 pthread_mutex_unlock(&mLock); 252 } 253 254 // call when you release a partial wakelock. 255 void release(const sp<IBinder> &wakeLockToken) { 256 if (wakeLockToken.get() == nullptr) { 257 return; 258 } 259 pthread_mutex_lock(&mLock); 260 if (--mCount < 0) { 261 ALOGE("negative wakelock count"); 262 mCount = 0; 263 } 264 pthread_mutex_unlock(&mLock); 265 } 266 267 // retrieves the boottime timebase offset from monotonic. 268 int64_t getBoottimeOffset() { 269 pthread_mutex_lock(&mLock); 270 int64_t boottimeOffset = mBoottimeOffset; 271 pthread_mutex_unlock(&mLock); 272 return boottimeOffset; 273 } 274 275 // Adjusts the timebase offset between TIMEBASE_MONOTONIC 276 // and the selected timebase. 277 // Currently only TIMEBASE_BOOTTIME is allowed. 278 // 279 // This only needs to be called upon acquiring the first partial wakelock 280 // after all other partial wakelocks are released. 281 // 282 // We do an empirical measurement of the offset rather than parsing 283 // /proc/timer_list since the latter is not a formal kernel ABI. 284 static void adjustTimebaseOffset(int64_t *offset, ExtendedTimestamp::Timebase timebase) { 285 int clockbase; 286 switch (timebase) { 287 case ExtendedTimestamp::TIMEBASE_BOOTTIME: 288 clockbase = SYSTEM_TIME_BOOTTIME; 289 break; 290 default: 291 LOG_ALWAYS_FATAL("invalid timebase %d", timebase); 292 break; 293 } 294 // try three times to get the clock offset, choose the one 295 // with the minimum gap in measurements. 296 const int tries = 3; 297 nsecs_t bestGap, measured; 298 for (int i = 0; i < tries; ++i) { 299 const nsecs_t tmono = systemTime(SYSTEM_TIME_MONOTONIC); 300 const nsecs_t tbase = systemTime(clockbase); 301 const nsecs_t tmono2 = systemTime(SYSTEM_TIME_MONOTONIC); 302 const nsecs_t gap = tmono2 - tmono; 303 if (i == 0 || gap < bestGap) { 304 bestGap = gap; 305 measured = tbase - ((tmono + tmono2) >> 1); 306 } 307 } 308 309 // to avoid micro-adjusting, we don't change the timebase 310 // unless it is significantly different. 311 // 312 // Assumption: It probably takes more than toleranceNs to 313 // suspend and resume the device. 314 static int64_t toleranceNs = 10000; // 10 us 315 if (llabs(*offset - measured) > toleranceNs) { 316 ALOGV("Adjusting timebase offset old: %lld new: %lld", 317 (long long)*offset, (long long)measured); 318 *offset = measured; 319 } 320 } 321 322 pthread_mutex_t mLock; 323 int32_t mCount; 324 int64_t mBoottimeOffset; 325} gBoottime = { PTHREAD_MUTEX_INITIALIZER, 0, 0 }; // static, so use POD initialization 326 327// ---------------------------------------------------------------------------- 328// CPU Stats 329// ---------------------------------------------------------------------------- 330 331class CpuStats { 332public: 333 CpuStats(); 334 void sample(const String8 &title); 335#ifdef DEBUG_CPU_USAGE 336private: 337 ThreadCpuUsage mCpuUsage; // instantaneous thread CPU usage in wall clock ns 338 CentralTendencyStatistics mWcStats; // statistics on thread CPU usage in wall clock ns 339 340 CentralTendencyStatistics mHzStats; // statistics on thread CPU usage in cycles 341 342 int mCpuNum; // thread's current CPU number 343 int mCpukHz; // frequency of thread's current CPU in kHz 344#endif 345}; 346 347CpuStats::CpuStats() 348#ifdef DEBUG_CPU_USAGE 349 : mCpuNum(-1), mCpukHz(-1) 350#endif 351{ 352} 353 354void CpuStats::sample(const String8 &title 355#ifndef DEBUG_CPU_USAGE 356 __unused 357#endif 358 ) { 359#ifdef DEBUG_CPU_USAGE 360 // get current thread's delta CPU time in wall clock ns 361 double wcNs; 362 bool valid = mCpuUsage.sampleAndEnable(wcNs); 363 364 // record sample for wall clock statistics 365 if (valid) { 366 mWcStats.sample(wcNs); 367 } 368 369 // get the current CPU number 370 int cpuNum = sched_getcpu(); 371 372 // get the current CPU frequency in kHz 373 int cpukHz = mCpuUsage.getCpukHz(cpuNum); 374 375 // check if either CPU number or frequency changed 376 if (cpuNum != mCpuNum || cpukHz != mCpukHz) { 377 mCpuNum = cpuNum; 378 mCpukHz = cpukHz; 379 // ignore sample for purposes of cycles 380 valid = false; 381 } 382 383 // if no change in CPU number or frequency, then record sample for cycle statistics 384 if (valid && mCpukHz > 0) { 385 double cycles = wcNs * cpukHz * 0.000001; 386 mHzStats.sample(cycles); 387 } 388 389 unsigned n = mWcStats.n(); 390 // mCpuUsage.elapsed() is expensive, so don't call it every loop 391 if ((n & 127) == 1) { 392 long long elapsed = mCpuUsage.elapsed(); 393 if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) { 394 double perLoop = elapsed / (double) n; 395 double perLoop100 = perLoop * 0.01; 396 double perLoop1k = perLoop * 0.001; 397 double mean = mWcStats.mean(); 398 double stddev = mWcStats.stddev(); 399 double minimum = mWcStats.minimum(); 400 double maximum = mWcStats.maximum(); 401 double meanCycles = mHzStats.mean(); 402 double stddevCycles = mHzStats.stddev(); 403 double minCycles = mHzStats.minimum(); 404 double maxCycles = mHzStats.maximum(); 405 mCpuUsage.resetElapsed(); 406 mWcStats.reset(); 407 mHzStats.reset(); 408 ALOGD("CPU usage for %s over past %.1f secs\n" 409 " (%u mixer loops at %.1f mean ms per loop):\n" 410 " us per mix loop: mean=%.0f stddev=%.0f min=%.0f max=%.0f\n" 411 " %% of wall: mean=%.1f stddev=%.1f min=%.1f max=%.1f\n" 412 " MHz: mean=%.1f, stddev=%.1f, min=%.1f max=%.1f", 413 title.string(), 414 elapsed * .000000001, n, perLoop * .000001, 415 mean * .001, 416 stddev * .001, 417 minimum * .001, 418 maximum * .001, 419 mean / perLoop100, 420 stddev / perLoop100, 421 minimum / perLoop100, 422 maximum / perLoop100, 423 meanCycles / perLoop1k, 424 stddevCycles / perLoop1k, 425 minCycles / perLoop1k, 426 maxCycles / perLoop1k); 427 428 } 429 } 430#endif 431}; 432 433// ---------------------------------------------------------------------------- 434// ThreadBase 435// ---------------------------------------------------------------------------- 436 437// static 438const char *AudioFlinger::ThreadBase::threadTypeToString(AudioFlinger::ThreadBase::type_t type) 439{ 440 switch (type) { 441 case MIXER: 442 return "MIXER"; 443 case DIRECT: 444 return "DIRECT"; 445 case DUPLICATING: 446 return "DUPLICATING"; 447 case RECORD: 448 return "RECORD"; 449 case OFFLOAD: 450 return "OFFLOAD"; 451 default: 452 return "unknown"; 453 } 454} 455 456String8 devicesToString(audio_devices_t devices) 457{ 458 static const struct mapping { 459 audio_devices_t mDevices; 460 const char * mString; 461 } mappingsOut[] = { 462 {AUDIO_DEVICE_OUT_EARPIECE, "EARPIECE"}, 463 {AUDIO_DEVICE_OUT_SPEAKER, "SPEAKER"}, 464 {AUDIO_DEVICE_OUT_WIRED_HEADSET, "WIRED_HEADSET"}, 465 {AUDIO_DEVICE_OUT_WIRED_HEADPHONE, "WIRED_HEADPHONE"}, 466 {AUDIO_DEVICE_OUT_BLUETOOTH_SCO, "BLUETOOTH_SCO"}, 467 {AUDIO_DEVICE_OUT_BLUETOOTH_SCO_HEADSET, "BLUETOOTH_SCO_HEADSET"}, 468 {AUDIO_DEVICE_OUT_BLUETOOTH_SCO_CARKIT, "BLUETOOTH_SCO_CARKIT"}, 469 {AUDIO_DEVICE_OUT_BLUETOOTH_A2DP, "BLUETOOTH_A2DP"}, 470 {AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES,"BLUETOOTH_A2DP_HEADPHONES"}, 471 {AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER, "BLUETOOTH_A2DP_SPEAKER"}, 472 {AUDIO_DEVICE_OUT_AUX_DIGITAL, "AUX_DIGITAL"}, 473 {AUDIO_DEVICE_OUT_HDMI, "HDMI"}, 474 {AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET,"ANLG_DOCK_HEADSET"}, 475 {AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET,"DGTL_DOCK_HEADSET"}, 476 {AUDIO_DEVICE_OUT_USB_ACCESSORY, "USB_ACCESSORY"}, 477 {AUDIO_DEVICE_OUT_USB_DEVICE, "USB_DEVICE"}, 478 {AUDIO_DEVICE_OUT_TELEPHONY_TX, "TELEPHONY_TX"}, 479 {AUDIO_DEVICE_OUT_LINE, "LINE"}, 480 {AUDIO_DEVICE_OUT_HDMI_ARC, "HDMI_ARC"}, 481 {AUDIO_DEVICE_OUT_SPDIF, "SPDIF"}, 482 {AUDIO_DEVICE_OUT_FM, "FM"}, 483 {AUDIO_DEVICE_OUT_AUX_LINE, "AUX_LINE"}, 484 {AUDIO_DEVICE_OUT_SPEAKER_SAFE, "SPEAKER_SAFE"}, 485 {AUDIO_DEVICE_OUT_IP, "IP"}, 486 {AUDIO_DEVICE_OUT_BUS, "BUS"}, 487 {AUDIO_DEVICE_NONE, "NONE"}, // must be last 488 }, mappingsIn[] = { 489 {AUDIO_DEVICE_IN_COMMUNICATION, "COMMUNICATION"}, 490 {AUDIO_DEVICE_IN_AMBIENT, "AMBIENT"}, 491 {AUDIO_DEVICE_IN_BUILTIN_MIC, "BUILTIN_MIC"}, 492 {AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET, "BLUETOOTH_SCO_HEADSET"}, 493 {AUDIO_DEVICE_IN_WIRED_HEADSET, "WIRED_HEADSET"}, 494 {AUDIO_DEVICE_IN_AUX_DIGITAL, "AUX_DIGITAL"}, 495 {AUDIO_DEVICE_IN_VOICE_CALL, "VOICE_CALL"}, 496 {AUDIO_DEVICE_IN_TELEPHONY_RX, "TELEPHONY_RX"}, 497 {AUDIO_DEVICE_IN_BACK_MIC, "BACK_MIC"}, 498 {AUDIO_DEVICE_IN_REMOTE_SUBMIX, "REMOTE_SUBMIX"}, 499 {AUDIO_DEVICE_IN_ANLG_DOCK_HEADSET, "ANLG_DOCK_HEADSET"}, 500 {AUDIO_DEVICE_IN_DGTL_DOCK_HEADSET, "DGTL_DOCK_HEADSET"}, 501 {AUDIO_DEVICE_IN_USB_ACCESSORY, "USB_ACCESSORY"}, 502 {AUDIO_DEVICE_IN_USB_DEVICE, "USB_DEVICE"}, 503 {AUDIO_DEVICE_IN_FM_TUNER, "FM_TUNER"}, 504 {AUDIO_DEVICE_IN_TV_TUNER, "TV_TUNER"}, 505 {AUDIO_DEVICE_IN_LINE, "LINE"}, 506 {AUDIO_DEVICE_IN_SPDIF, "SPDIF"}, 507 {AUDIO_DEVICE_IN_BLUETOOTH_A2DP, "BLUETOOTH_A2DP"}, 508 {AUDIO_DEVICE_IN_LOOPBACK, "LOOPBACK"}, 509 {AUDIO_DEVICE_IN_IP, "IP"}, 510 {AUDIO_DEVICE_IN_BUS, "BUS"}, 511 {AUDIO_DEVICE_NONE, "NONE"}, // must be last 512 }; 513 String8 result; 514 audio_devices_t allDevices = AUDIO_DEVICE_NONE; 515 const mapping *entry; 516 if (devices & AUDIO_DEVICE_BIT_IN) { 517 devices &= ~AUDIO_DEVICE_BIT_IN; 518 entry = mappingsIn; 519 } else { 520 entry = mappingsOut; 521 } 522 for ( ; entry->mDevices != AUDIO_DEVICE_NONE; entry++) { 523 allDevices = (audio_devices_t) (allDevices | entry->mDevices); 524 if (devices & entry->mDevices) { 525 if (!result.isEmpty()) { 526 result.append("|"); 527 } 528 result.append(entry->mString); 529 } 530 } 531 if (devices & ~allDevices) { 532 if (!result.isEmpty()) { 533 result.append("|"); 534 } 535 result.appendFormat("0x%X", devices & ~allDevices); 536 } 537 if (result.isEmpty()) { 538 result.append(entry->mString); 539 } 540 return result; 541} 542 543String8 inputFlagsToString(audio_input_flags_t flags) 544{ 545 static const struct mapping { 546 audio_input_flags_t mFlag; 547 const char * mString; 548 } mappings[] = { 549 {AUDIO_INPUT_FLAG_FAST, "FAST"}, 550 {AUDIO_INPUT_FLAG_HW_HOTWORD, "HW_HOTWORD"}, 551 {AUDIO_INPUT_FLAG_RAW, "RAW"}, 552 {AUDIO_INPUT_FLAG_SYNC, "SYNC"}, 553 {AUDIO_INPUT_FLAG_NONE, "NONE"}, // must be last 554 }; 555 String8 result; 556 audio_input_flags_t allFlags = AUDIO_INPUT_FLAG_NONE; 557 const mapping *entry; 558 for (entry = mappings; entry->mFlag != AUDIO_INPUT_FLAG_NONE; entry++) { 559 allFlags = (audio_input_flags_t) (allFlags | entry->mFlag); 560 if (flags & entry->mFlag) { 561 if (!result.isEmpty()) { 562 result.append("|"); 563 } 564 result.append(entry->mString); 565 } 566 } 567 if (flags & ~allFlags) { 568 if (!result.isEmpty()) { 569 result.append("|"); 570 } 571 result.appendFormat("0x%X", flags & ~allFlags); 572 } 573 if (result.isEmpty()) { 574 result.append(entry->mString); 575 } 576 return result; 577} 578 579String8 outputFlagsToString(audio_output_flags_t flags) 580{ 581 static const struct mapping { 582 audio_output_flags_t mFlag; 583 const char * mString; 584 } mappings[] = { 585 {AUDIO_OUTPUT_FLAG_DIRECT, "DIRECT"}, 586 {AUDIO_OUTPUT_FLAG_PRIMARY, "PRIMARY"}, 587 {AUDIO_OUTPUT_FLAG_FAST, "FAST"}, 588 {AUDIO_OUTPUT_FLAG_DEEP_BUFFER, "DEEP_BUFFER"}, 589 {AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD,"COMPRESS_OFFLOAD"}, 590 {AUDIO_OUTPUT_FLAG_NON_BLOCKING, "NON_BLOCKING"}, 591 {AUDIO_OUTPUT_FLAG_HW_AV_SYNC, "HW_AV_SYNC"}, 592 {AUDIO_OUTPUT_FLAG_RAW, "RAW"}, 593 {AUDIO_OUTPUT_FLAG_SYNC, "SYNC"}, 594 {AUDIO_OUTPUT_FLAG_IEC958_NONAUDIO, "IEC958_NONAUDIO"}, 595 {AUDIO_OUTPUT_FLAG_NONE, "NONE"}, // must be last 596 }; 597 String8 result; 598 audio_output_flags_t allFlags = AUDIO_OUTPUT_FLAG_NONE; 599 const mapping *entry; 600 for (entry = mappings; entry->mFlag != AUDIO_OUTPUT_FLAG_NONE; entry++) { 601 allFlags = (audio_output_flags_t) (allFlags | entry->mFlag); 602 if (flags & entry->mFlag) { 603 if (!result.isEmpty()) { 604 result.append("|"); 605 } 606 result.append(entry->mString); 607 } 608 } 609 if (flags & ~allFlags) { 610 if (!result.isEmpty()) { 611 result.append("|"); 612 } 613 result.appendFormat("0x%X", flags & ~allFlags); 614 } 615 if (result.isEmpty()) { 616 result.append(entry->mString); 617 } 618 return result; 619} 620 621const char *sourceToString(audio_source_t source) 622{ 623 switch (source) { 624 case AUDIO_SOURCE_DEFAULT: return "default"; 625 case AUDIO_SOURCE_MIC: return "mic"; 626 case AUDIO_SOURCE_VOICE_UPLINK: return "voice uplink"; 627 case AUDIO_SOURCE_VOICE_DOWNLINK: return "voice downlink"; 628 case AUDIO_SOURCE_VOICE_CALL: return "voice call"; 629 case AUDIO_SOURCE_CAMCORDER: return "camcorder"; 630 case AUDIO_SOURCE_VOICE_RECOGNITION: return "voice recognition"; 631 case AUDIO_SOURCE_VOICE_COMMUNICATION: return "voice communication"; 632 case AUDIO_SOURCE_REMOTE_SUBMIX: return "remote submix"; 633 case AUDIO_SOURCE_UNPROCESSED: return "unprocessed"; 634 case AUDIO_SOURCE_FM_TUNER: return "FM tuner"; 635 case AUDIO_SOURCE_HOTWORD: return "hotword"; 636 default: return "unknown"; 637 } 638} 639 640AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, 641 audio_devices_t outDevice, audio_devices_t inDevice, type_t type, bool systemReady) 642 : Thread(false /*canCallJava*/), 643 mType(type), 644 mAudioFlinger(audioFlinger), 645 // mSampleRate, mFrameCount, mChannelMask, mChannelCount, mFrameSize, mFormat, mBufferSize 646 // are set by PlaybackThread::readOutputParameters_l() or 647 // RecordThread::readInputParameters_l() 648 //FIXME: mStandby should be true here. Is this some kind of hack? 649 mStandby(false), mOutDevice(outDevice), mInDevice(inDevice), 650 mPrevOutDevice(AUDIO_DEVICE_NONE), mPrevInDevice(AUDIO_DEVICE_NONE), 651 mAudioSource(AUDIO_SOURCE_DEFAULT), mId(id), 652 // mName will be set by concrete (non-virtual) subclass 653 mDeathRecipient(new PMDeathRecipient(this)), 654 mSystemReady(systemReady), 655 mNotifiedBatteryStart(false) 656{ 657 memset(&mPatch, 0, sizeof(struct audio_patch)); 658} 659 660AudioFlinger::ThreadBase::~ThreadBase() 661{ 662 // mConfigEvents should be empty, but just in case it isn't, free the memory it owns 663 mConfigEvents.clear(); 664 665 // do not lock the mutex in destructor 666 releaseWakeLock_l(); 667 if (mPowerManager != 0) { 668 sp<IBinder> binder = IInterface::asBinder(mPowerManager); 669 binder->unlinkToDeath(mDeathRecipient); 670 } 671} 672 673status_t AudioFlinger::ThreadBase::readyToRun() 674{ 675 status_t status = initCheck(); 676 if (status == NO_ERROR) { 677 ALOGI("AudioFlinger's thread %p ready to run", this); 678 } else { 679 ALOGE("No working audio driver found."); 680 } 681 return status; 682} 683 684void AudioFlinger::ThreadBase::exit() 685{ 686 ALOGV("ThreadBase::exit"); 687 // do any cleanup required for exit to succeed 688 preExit(); 689 { 690 // This lock prevents the following race in thread (uniprocessor for illustration): 691 // if (!exitPending()) { 692 // // context switch from here to exit() 693 // // exit() calls requestExit(), what exitPending() observes 694 // // exit() calls signal(), which is dropped since no waiters 695 // // context switch back from exit() to here 696 // mWaitWorkCV.wait(...); 697 // // now thread is hung 698 // } 699 AutoMutex lock(mLock); 700 requestExit(); 701 mWaitWorkCV.broadcast(); 702 } 703 // When Thread::requestExitAndWait is made virtual and this method is renamed to 704 // "virtual status_t requestExitAndWait()", replace by "return Thread::requestExitAndWait();" 705 requestExitAndWait(); 706} 707 708status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs) 709{ 710 ALOGV("ThreadBase::setParameters() %s", keyValuePairs.string()); 711 Mutex::Autolock _l(mLock); 712 713 return sendSetParameterConfigEvent_l(keyValuePairs); 714} 715 716// sendConfigEvent_l() must be called with ThreadBase::mLock held 717// Can temporarily release the lock if waiting for a reply from processConfigEvents_l(). 718status_t AudioFlinger::ThreadBase::sendConfigEvent_l(sp<ConfigEvent>& event) 719{ 720 status_t status = NO_ERROR; 721 722 if (event->mRequiresSystemReady && !mSystemReady) { 723 event->mWaitStatus = false; 724 mPendingConfigEvents.add(event); 725 return status; 726 } 727 mConfigEvents.add(event); 728 ALOGV("sendConfigEvent_l() num events %d event %d", mConfigEvents.size(), event->mType); 729 mWaitWorkCV.signal(); 730 mLock.unlock(); 731 { 732 Mutex::Autolock _l(event->mLock); 733 while (event->mWaitStatus) { 734 if (event->mCond.waitRelative(event->mLock, kConfigEventTimeoutNs) != NO_ERROR) { 735 event->mStatus = TIMED_OUT; 736 event->mWaitStatus = false; 737 } 738 } 739 status = event->mStatus; 740 } 741 mLock.lock(); 742 return status; 743} 744 745void AudioFlinger::ThreadBase::sendIoConfigEvent(audio_io_config_event event, pid_t pid) 746{ 747 Mutex::Autolock _l(mLock); 748 sendIoConfigEvent_l(event, pid); 749} 750 751// sendIoConfigEvent_l() must be called with ThreadBase::mLock held 752void AudioFlinger::ThreadBase::sendIoConfigEvent_l(audio_io_config_event event, pid_t pid) 753{ 754 sp<ConfigEvent> configEvent = (ConfigEvent *)new IoConfigEvent(event, pid); 755 sendConfigEvent_l(configEvent); 756} 757 758void AudioFlinger::ThreadBase::sendPrioConfigEvent(pid_t pid, pid_t tid, int32_t prio) 759{ 760 Mutex::Autolock _l(mLock); 761 sendPrioConfigEvent_l(pid, tid, prio); 762} 763 764// sendPrioConfigEvent_l() must be called with ThreadBase::mLock held 765void AudioFlinger::ThreadBase::sendPrioConfigEvent_l(pid_t pid, pid_t tid, int32_t prio) 766{ 767 sp<ConfigEvent> configEvent = (ConfigEvent *)new PrioConfigEvent(pid, tid, prio); 768 sendConfigEvent_l(configEvent); 769} 770 771// sendSetParameterConfigEvent_l() must be called with ThreadBase::mLock held 772status_t AudioFlinger::ThreadBase::sendSetParameterConfigEvent_l(const String8& keyValuePair) 773{ 774 sp<ConfigEvent> configEvent; 775 AudioParameter param(keyValuePair); 776 int value; 777 if (param.getInt(String8(AUDIO_PARAMETER_MONO_OUTPUT), value) == NO_ERROR) { 778 setMasterMono_l(value != 0); 779 if (param.size() == 1) { 780 return NO_ERROR; // should be a solo parameter - we don't pass down 781 } 782 param.remove(String8(AUDIO_PARAMETER_MONO_OUTPUT)); 783 configEvent = new SetParameterConfigEvent(param.toString()); 784 } else { 785 configEvent = new SetParameterConfigEvent(keyValuePair); 786 } 787 return sendConfigEvent_l(configEvent); 788} 789 790status_t AudioFlinger::ThreadBase::sendCreateAudioPatchConfigEvent( 791 const struct audio_patch *patch, 792 audio_patch_handle_t *handle) 793{ 794 Mutex::Autolock _l(mLock); 795 sp<ConfigEvent> configEvent = (ConfigEvent *)new CreateAudioPatchConfigEvent(*patch, *handle); 796 status_t status = sendConfigEvent_l(configEvent); 797 if (status == NO_ERROR) { 798 CreateAudioPatchConfigEventData *data = 799 (CreateAudioPatchConfigEventData *)configEvent->mData.get(); 800 *handle = data->mHandle; 801 } 802 return status; 803} 804 805status_t AudioFlinger::ThreadBase::sendReleaseAudioPatchConfigEvent( 806 const audio_patch_handle_t handle) 807{ 808 Mutex::Autolock _l(mLock); 809 sp<ConfigEvent> configEvent = (ConfigEvent *)new ReleaseAudioPatchConfigEvent(handle); 810 return sendConfigEvent_l(configEvent); 811} 812 813 814// post condition: mConfigEvents.isEmpty() 815void AudioFlinger::ThreadBase::processConfigEvents_l() 816{ 817 bool configChanged = false; 818 819 while (!mConfigEvents.isEmpty()) { 820 ALOGV("processConfigEvents_l() remaining events %d", mConfigEvents.size()); 821 sp<ConfigEvent> event = mConfigEvents[0]; 822 mConfigEvents.removeAt(0); 823 switch (event->mType) { 824 case CFG_EVENT_PRIO: { 825 PrioConfigEventData *data = (PrioConfigEventData *)event->mData.get(); 826 // FIXME Need to understand why this has to be done asynchronously 827 int err = requestPriority(data->mPid, data->mTid, data->mPrio, 828 true /*asynchronous*/); 829 if (err != 0) { 830 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 831 data->mPrio, data->mPid, data->mTid, err); 832 } 833 } break; 834 case CFG_EVENT_IO: { 835 IoConfigEventData *data = (IoConfigEventData *)event->mData.get(); 836 ioConfigChanged(data->mEvent, data->mPid); 837 } break; 838 case CFG_EVENT_SET_PARAMETER: { 839 SetParameterConfigEventData *data = (SetParameterConfigEventData *)event->mData.get(); 840 if (checkForNewParameter_l(data->mKeyValuePairs, event->mStatus)) { 841 configChanged = true; 842 } 843 } break; 844 case CFG_EVENT_CREATE_AUDIO_PATCH: { 845 CreateAudioPatchConfigEventData *data = 846 (CreateAudioPatchConfigEventData *)event->mData.get(); 847 event->mStatus = createAudioPatch_l(&data->mPatch, &data->mHandle); 848 } break; 849 case CFG_EVENT_RELEASE_AUDIO_PATCH: { 850 ReleaseAudioPatchConfigEventData *data = 851 (ReleaseAudioPatchConfigEventData *)event->mData.get(); 852 event->mStatus = releaseAudioPatch_l(data->mHandle); 853 } break; 854 default: 855 ALOG_ASSERT(false, "processConfigEvents_l() unknown event type %d", event->mType); 856 break; 857 } 858 { 859 Mutex::Autolock _l(event->mLock); 860 if (event->mWaitStatus) { 861 event->mWaitStatus = false; 862 event->mCond.signal(); 863 } 864 } 865 ALOGV_IF(mConfigEvents.isEmpty(), "processConfigEvents_l() DONE thread %p", this); 866 } 867 868 if (configChanged) { 869 cacheParameters_l(); 870 } 871} 872 873String8 channelMaskToString(audio_channel_mask_t mask, bool output) { 874 String8 s; 875 const audio_channel_representation_t representation = 876 audio_channel_mask_get_representation(mask); 877 878 switch (representation) { 879 case AUDIO_CHANNEL_REPRESENTATION_POSITION: { 880 if (output) { 881 if (mask & AUDIO_CHANNEL_OUT_FRONT_LEFT) s.append("front-left, "); 882 if (mask & AUDIO_CHANNEL_OUT_FRONT_RIGHT) s.append("front-right, "); 883 if (mask & AUDIO_CHANNEL_OUT_FRONT_CENTER) s.append("front-center, "); 884 if (mask & AUDIO_CHANNEL_OUT_LOW_FREQUENCY) s.append("low freq, "); 885 if (mask & AUDIO_CHANNEL_OUT_BACK_LEFT) s.append("back-left, "); 886 if (mask & AUDIO_CHANNEL_OUT_BACK_RIGHT) s.append("back-right, "); 887 if (mask & AUDIO_CHANNEL_OUT_FRONT_LEFT_OF_CENTER) s.append("front-left-of-center, "); 888 if (mask & AUDIO_CHANNEL_OUT_FRONT_RIGHT_OF_CENTER) s.append("front-right-of-center, "); 889 if (mask & AUDIO_CHANNEL_OUT_BACK_CENTER) s.append("back-center, "); 890 if (mask & AUDIO_CHANNEL_OUT_SIDE_LEFT) s.append("side-left, "); 891 if (mask & AUDIO_CHANNEL_OUT_SIDE_RIGHT) s.append("side-right, "); 892 if (mask & AUDIO_CHANNEL_OUT_TOP_CENTER) s.append("top-center ,"); 893 if (mask & AUDIO_CHANNEL_OUT_TOP_FRONT_LEFT) s.append("top-front-left, "); 894 if (mask & AUDIO_CHANNEL_OUT_TOP_FRONT_CENTER) s.append("top-front-center, "); 895 if (mask & AUDIO_CHANNEL_OUT_TOP_FRONT_RIGHT) s.append("top-front-right, "); 896 if (mask & AUDIO_CHANNEL_OUT_TOP_BACK_LEFT) s.append("top-back-left, "); 897 if (mask & AUDIO_CHANNEL_OUT_TOP_BACK_CENTER) s.append("top-back-center, " ); 898 if (mask & AUDIO_CHANNEL_OUT_TOP_BACK_RIGHT) s.append("top-back-right, " ); 899 if (mask & ~AUDIO_CHANNEL_OUT_ALL) s.append("unknown, "); 900 } else { 901 if (mask & AUDIO_CHANNEL_IN_LEFT) s.append("left, "); 902 if (mask & AUDIO_CHANNEL_IN_RIGHT) s.append("right, "); 903 if (mask & AUDIO_CHANNEL_IN_FRONT) s.append("front, "); 904 if (mask & AUDIO_CHANNEL_IN_BACK) s.append("back, "); 905 if (mask & AUDIO_CHANNEL_IN_LEFT_PROCESSED) s.append("left-processed, "); 906 if (mask & AUDIO_CHANNEL_IN_RIGHT_PROCESSED) s.append("right-processed, "); 907 if (mask & AUDIO_CHANNEL_IN_FRONT_PROCESSED) s.append("front-processed, "); 908 if (mask & AUDIO_CHANNEL_IN_BACK_PROCESSED) s.append("back-processed, "); 909 if (mask & AUDIO_CHANNEL_IN_PRESSURE) s.append("pressure, "); 910 if (mask & AUDIO_CHANNEL_IN_X_AXIS) s.append("X, "); 911 if (mask & AUDIO_CHANNEL_IN_Y_AXIS) s.append("Y, "); 912 if (mask & AUDIO_CHANNEL_IN_Z_AXIS) s.append("Z, "); 913 if (mask & AUDIO_CHANNEL_IN_VOICE_UPLINK) s.append("voice-uplink, "); 914 if (mask & AUDIO_CHANNEL_IN_VOICE_DNLINK) s.append("voice-dnlink, "); 915 if (mask & ~AUDIO_CHANNEL_IN_ALL) s.append("unknown, "); 916 } 917 const int len = s.length(); 918 if (len > 2) { 919 (void) s.lockBuffer(len); // needed? 920 s.unlockBuffer(len - 2); // remove trailing ", " 921 } 922 return s; 923 } 924 case AUDIO_CHANNEL_REPRESENTATION_INDEX: 925 s.appendFormat("index mask, bits:%#x", audio_channel_mask_get_bits(mask)); 926 return s; 927 default: 928 s.appendFormat("unknown mask, representation:%d bits:%#x", 929 representation, audio_channel_mask_get_bits(mask)); 930 return s; 931 } 932} 933 934void AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args __unused) 935{ 936 const size_t SIZE = 256; 937 char buffer[SIZE]; 938 String8 result; 939 940 bool locked = AudioFlinger::dumpTryLock(mLock); 941 if (!locked) { 942 dprintf(fd, "thread %p may be deadlocked\n", this); 943 } 944 945 dprintf(fd, " Thread name: %s\n", mThreadName); 946 dprintf(fd, " I/O handle: %d\n", mId); 947 dprintf(fd, " TID: %d\n", getTid()); 948 dprintf(fd, " Standby: %s\n", mStandby ? "yes" : "no"); 949 dprintf(fd, " Sample rate: %u Hz\n", mSampleRate); 950 dprintf(fd, " HAL frame count: %zu\n", mFrameCount); 951 dprintf(fd, " HAL format: 0x%x (%s)\n", mHALFormat, formatToString(mHALFormat)); 952 dprintf(fd, " HAL buffer size: %u bytes\n", mBufferSize); 953 dprintf(fd, " Channel count: %u\n", mChannelCount); 954 dprintf(fd, " Channel mask: 0x%08x (%s)\n", mChannelMask, 955 channelMaskToString(mChannelMask, mType != RECORD).string()); 956 dprintf(fd, " Processing format: 0x%x (%s)\n", mFormat, formatToString(mFormat)); 957 dprintf(fd, " Processing frame size: %zu bytes\n", mFrameSize); 958 dprintf(fd, " Pending config events:"); 959 size_t numConfig = mConfigEvents.size(); 960 if (numConfig) { 961 for (size_t i = 0; i < numConfig; i++) { 962 mConfigEvents[i]->dump(buffer, SIZE); 963 dprintf(fd, "\n %s", buffer); 964 } 965 dprintf(fd, "\n"); 966 } else { 967 dprintf(fd, " none\n"); 968 } 969 dprintf(fd, " Output device: %#x (%s)\n", mOutDevice, devicesToString(mOutDevice).string()); 970 dprintf(fd, " Input device: %#x (%s)\n", mInDevice, devicesToString(mInDevice).string()); 971 dprintf(fd, " Audio source: %d (%s)\n", mAudioSource, sourceToString(mAudioSource)); 972 973 if (locked) { 974 mLock.unlock(); 975 } 976} 977 978void AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args) 979{ 980 const size_t SIZE = 256; 981 char buffer[SIZE]; 982 String8 result; 983 984 size_t numEffectChains = mEffectChains.size(); 985 snprintf(buffer, SIZE, " %zu Effect Chains\n", numEffectChains); 986 write(fd, buffer, strlen(buffer)); 987 988 for (size_t i = 0; i < numEffectChains; ++i) { 989 sp<EffectChain> chain = mEffectChains[i]; 990 if (chain != 0) { 991 chain->dump(fd, args); 992 } 993 } 994} 995 996void AudioFlinger::ThreadBase::acquireWakeLock(int uid) 997{ 998 Mutex::Autolock _l(mLock); 999 acquireWakeLock_l(uid); 1000} 1001 1002String16 AudioFlinger::ThreadBase::getWakeLockTag() 1003{ 1004 switch (mType) { 1005 case MIXER: 1006 return String16("AudioMix"); 1007 case DIRECT: 1008 return String16("AudioDirectOut"); 1009 case DUPLICATING: 1010 return String16("AudioDup"); 1011 case RECORD: 1012 return String16("AudioIn"); 1013 case OFFLOAD: 1014 return String16("AudioOffload"); 1015 default: 1016 ALOG_ASSERT(false); 1017 return String16("AudioUnknown"); 1018 } 1019} 1020 1021void AudioFlinger::ThreadBase::acquireWakeLock_l(int uid) 1022{ 1023 getPowerManager_l(); 1024 if (mPowerManager != 0) { 1025 sp<IBinder> binder = new BBinder(); 1026 status_t status; 1027 if (uid >= 0) { 1028 status = mPowerManager->acquireWakeLockWithUid(POWERMANAGER_PARTIAL_WAKE_LOCK, 1029 binder, 1030 getWakeLockTag(), 1031 String16("audioserver"), 1032 uid, 1033 true /* FIXME force oneway contrary to .aidl */); 1034 } else { 1035 status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK, 1036 binder, 1037 getWakeLockTag(), 1038 String16("audioserver"), 1039 true /* FIXME force oneway contrary to .aidl */); 1040 } 1041 if (status == NO_ERROR) { 1042 mWakeLockToken = binder; 1043 } 1044 ALOGV("acquireWakeLock_l() %s status %d", mThreadName, status); 1045 } 1046 1047 if (!mNotifiedBatteryStart) { 1048 BatteryNotifier::getInstance().noteStartAudio(); 1049 mNotifiedBatteryStart = true; 1050 } 1051 gBoottime.acquire(mWakeLockToken); 1052 mTimestamp.mTimebaseOffset[ExtendedTimestamp::TIMEBASE_BOOTTIME] = 1053 gBoottime.getBoottimeOffset(); 1054} 1055 1056void AudioFlinger::ThreadBase::releaseWakeLock() 1057{ 1058 Mutex::Autolock _l(mLock); 1059 releaseWakeLock_l(); 1060} 1061 1062void AudioFlinger::ThreadBase::releaseWakeLock_l() 1063{ 1064 gBoottime.release(mWakeLockToken); 1065 if (mWakeLockToken != 0) { 1066 ALOGV("releaseWakeLock_l() %s", mThreadName); 1067 if (mPowerManager != 0) { 1068 mPowerManager->releaseWakeLock(mWakeLockToken, 0, 1069 true /* FIXME force oneway contrary to .aidl */); 1070 } 1071 mWakeLockToken.clear(); 1072 } 1073 1074 if (mNotifiedBatteryStart) { 1075 BatteryNotifier::getInstance().noteStopAudio(); 1076 mNotifiedBatteryStart = false; 1077 } 1078} 1079 1080void AudioFlinger::ThreadBase::updateWakeLockUids(const SortedVector<int> &uids) { 1081 Mutex::Autolock _l(mLock); 1082 updateWakeLockUids_l(uids); 1083} 1084 1085void AudioFlinger::ThreadBase::getPowerManager_l() { 1086 if (mSystemReady && mPowerManager == 0) { 1087 // use checkService() to avoid blocking if power service is not up yet 1088 sp<IBinder> binder = 1089 defaultServiceManager()->checkService(String16("power")); 1090 if (binder == 0) { 1091 ALOGW("Thread %s cannot connect to the power manager service", mThreadName); 1092 } else { 1093 mPowerManager = interface_cast<IPowerManager>(binder); 1094 binder->linkToDeath(mDeathRecipient); 1095 } 1096 } 1097} 1098 1099void AudioFlinger::ThreadBase::updateWakeLockUids_l(const SortedVector<int> &uids) { 1100 getPowerManager_l(); 1101 if (mWakeLockToken == NULL) { // token may be NULL if AudioFlinger::systemReady() not called. 1102 if (mSystemReady) { 1103 ALOGE("no wake lock to update, but system ready!"); 1104 } else { 1105 ALOGW("no wake lock to update, system not ready yet"); 1106 } 1107 return; 1108 } 1109 if (mPowerManager != 0) { 1110 sp<IBinder> binder = new BBinder(); 1111 status_t status; 1112 status = mPowerManager->updateWakeLockUids(mWakeLockToken, uids.size(), uids.array(), 1113 true /* FIXME force oneway contrary to .aidl */); 1114 ALOGV("updateWakeLockUids_l() %s status %d", mThreadName, status); 1115 } 1116} 1117 1118void AudioFlinger::ThreadBase::clearPowerManager() 1119{ 1120 Mutex::Autolock _l(mLock); 1121 releaseWakeLock_l(); 1122 mPowerManager.clear(); 1123} 1124 1125void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who __unused) 1126{ 1127 sp<ThreadBase> thread = mThread.promote(); 1128 if (thread != 0) { 1129 thread->clearPowerManager(); 1130 } 1131 ALOGW("power manager service died !!!"); 1132} 1133 1134void AudioFlinger::ThreadBase::setEffectSuspended( 1135 const effect_uuid_t *type, bool suspend, audio_session_t sessionId) 1136{ 1137 Mutex::Autolock _l(mLock); 1138 setEffectSuspended_l(type, suspend, sessionId); 1139} 1140 1141void AudioFlinger::ThreadBase::setEffectSuspended_l( 1142 const effect_uuid_t *type, bool suspend, audio_session_t sessionId) 1143{ 1144 sp<EffectChain> chain = getEffectChain_l(sessionId); 1145 if (chain != 0) { 1146 if (type != NULL) { 1147 chain->setEffectSuspended_l(type, suspend); 1148 } else { 1149 chain->setEffectSuspendedAll_l(suspend); 1150 } 1151 } 1152 1153 updateSuspendedSessions_l(type, suspend, sessionId); 1154} 1155 1156void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain) 1157{ 1158 ssize_t index = mSuspendedSessions.indexOfKey(chain->sessionId()); 1159 if (index < 0) { 1160 return; 1161 } 1162 1163 const KeyedVector <int, sp<SuspendedSessionDesc> >& sessionEffects = 1164 mSuspendedSessions.valueAt(index); 1165 1166 for (size_t i = 0; i < sessionEffects.size(); i++) { 1167 sp<SuspendedSessionDesc> desc = sessionEffects.valueAt(i); 1168 for (int j = 0; j < desc->mRefCount; j++) { 1169 if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) { 1170 chain->setEffectSuspendedAll_l(true); 1171 } else { 1172 ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x", 1173 desc->mType.timeLow); 1174 chain->setEffectSuspended_l(&desc->mType, true); 1175 } 1176 } 1177 } 1178} 1179 1180void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type, 1181 bool suspend, 1182 audio_session_t sessionId) 1183{ 1184 ssize_t index = mSuspendedSessions.indexOfKey(sessionId); 1185 1186 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects; 1187 1188 if (suspend) { 1189 if (index >= 0) { 1190 sessionEffects = mSuspendedSessions.valueAt(index); 1191 } else { 1192 mSuspendedSessions.add(sessionId, sessionEffects); 1193 } 1194 } else { 1195 if (index < 0) { 1196 return; 1197 } 1198 sessionEffects = mSuspendedSessions.valueAt(index); 1199 } 1200 1201 1202 int key = EffectChain::kKeyForSuspendAll; 1203 if (type != NULL) { 1204 key = type->timeLow; 1205 } 1206 index = sessionEffects.indexOfKey(key); 1207 1208 sp<SuspendedSessionDesc> desc; 1209 if (suspend) { 1210 if (index >= 0) { 1211 desc = sessionEffects.valueAt(index); 1212 } else { 1213 desc = new SuspendedSessionDesc(); 1214 if (type != NULL) { 1215 desc->mType = *type; 1216 } 1217 sessionEffects.add(key, desc); 1218 ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key); 1219 } 1220 desc->mRefCount++; 1221 } else { 1222 if (index < 0) { 1223 return; 1224 } 1225 desc = sessionEffects.valueAt(index); 1226 if (--desc->mRefCount == 0) { 1227 ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key); 1228 sessionEffects.removeItemsAt(index); 1229 if (sessionEffects.isEmpty()) { 1230 ALOGV("updateSuspendedSessions_l() restore removing session %d", 1231 sessionId); 1232 mSuspendedSessions.removeItem(sessionId); 1233 } 1234 } 1235 } 1236 if (!sessionEffects.isEmpty()) { 1237 mSuspendedSessions.replaceValueFor(sessionId, sessionEffects); 1238 } 1239} 1240 1241void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 1242 bool enabled, 1243 audio_session_t sessionId) 1244{ 1245 Mutex::Autolock _l(mLock); 1246 checkSuspendOnEffectEnabled_l(effect, enabled, sessionId); 1247} 1248 1249void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 1250 bool enabled, 1251 audio_session_t sessionId) 1252{ 1253 if (mType != RECORD) { 1254 // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on 1255 // another session. This gives the priority to well behaved effect control panels 1256 // and applications not using global effects. 1257 // Enabling post processing in AUDIO_SESSION_OUTPUT_STAGE session does not affect 1258 // global effects 1259 if ((sessionId != AUDIO_SESSION_OUTPUT_MIX) && (sessionId != AUDIO_SESSION_OUTPUT_STAGE)) { 1260 setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX); 1261 } 1262 } 1263 1264 sp<EffectChain> chain = getEffectChain_l(sessionId); 1265 if (chain != 0) { 1266 chain->checkSuspendOnEffectEnabled(effect, enabled); 1267 } 1268} 1269 1270// ThreadBase::createEffect_l() must be called with AudioFlinger::mLock held 1271sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 1272 const sp<AudioFlinger::Client>& client, 1273 const sp<IEffectClient>& effectClient, 1274 int32_t priority, 1275 audio_session_t sessionId, 1276 effect_descriptor_t *desc, 1277 int *enabled, 1278 status_t *status) 1279{ 1280 sp<EffectModule> effect; 1281 sp<EffectHandle> handle; 1282 status_t lStatus; 1283 sp<EffectChain> chain; 1284 bool chainCreated = false; 1285 bool effectCreated = false; 1286 bool effectRegistered = false; 1287 1288 lStatus = initCheck(); 1289 if (lStatus != NO_ERROR) { 1290 ALOGW("createEffect_l() Audio driver not initialized."); 1291 goto Exit; 1292 } 1293 1294 // Reject any effect on Direct output threads for now, since the format of 1295 // mSinkBuffer is not guaranteed to be compatible with effect processing (PCM 16 stereo). 1296 if (mType == DIRECT) { 1297 ALOGW("createEffect_l() Cannot add effect %s on Direct output type thread %s", 1298 desc->name, mThreadName); 1299 lStatus = BAD_VALUE; 1300 goto Exit; 1301 } 1302 1303 // Reject any effect on mixer or duplicating multichannel sinks. 1304 // TODO: fix both format and multichannel issues with effects. 1305 if ((mType == MIXER || mType == DUPLICATING) && mChannelCount != FCC_2) { 1306 ALOGW("createEffect_l() Cannot add effect %s for multichannel(%d) %s threads", 1307 desc->name, mChannelCount, mType == MIXER ? "MIXER" : "DUPLICATING"); 1308 lStatus = BAD_VALUE; 1309 goto Exit; 1310 } 1311 1312 // Allow global effects only on offloaded and mixer threads 1313 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 1314 switch (mType) { 1315 case MIXER: 1316 case OFFLOAD: 1317 break; 1318 case DIRECT: 1319 case DUPLICATING: 1320 case RECORD: 1321 default: 1322 ALOGW("createEffect_l() Cannot add global effect %s on thread %s", 1323 desc->name, mThreadName); 1324 lStatus = BAD_VALUE; 1325 goto Exit; 1326 } 1327 } 1328 1329 // Only Pre processor effects are allowed on input threads and only on input threads 1330 if ((mType == RECORD) != ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { 1331 ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d", 1332 desc->name, desc->flags, mType); 1333 lStatus = BAD_VALUE; 1334 goto Exit; 1335 } 1336 1337 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 1338 1339 { // scope for mLock 1340 Mutex::Autolock _l(mLock); 1341 1342 // check for existing effect chain with the requested audio session 1343 chain = getEffectChain_l(sessionId); 1344 if (chain == 0) { 1345 // create a new chain for this session 1346 ALOGV("createEffect_l() new effect chain for session %d", sessionId); 1347 chain = new EffectChain(this, sessionId); 1348 addEffectChain_l(chain); 1349 chain->setStrategy(getStrategyForSession_l(sessionId)); 1350 chainCreated = true; 1351 } else { 1352 effect = chain->getEffectFromDesc_l(desc); 1353 } 1354 1355 ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get()); 1356 1357 if (effect == 0) { 1358 audio_unique_id_t id = mAudioFlinger->nextUniqueId(AUDIO_UNIQUE_ID_USE_EFFECT); 1359 // Check CPU and memory usage 1360 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 1361 if (lStatus != NO_ERROR) { 1362 goto Exit; 1363 } 1364 effectRegistered = true; 1365 // create a new effect module if none present in the chain 1366 effect = new EffectModule(this, chain, desc, id, sessionId); 1367 lStatus = effect->status(); 1368 if (lStatus != NO_ERROR) { 1369 goto Exit; 1370 } 1371 effect->setOffloaded(mType == OFFLOAD, mId); 1372 1373 lStatus = chain->addEffect_l(effect); 1374 if (lStatus != NO_ERROR) { 1375 goto Exit; 1376 } 1377 effectCreated = true; 1378 1379 effect->setDevice(mOutDevice); 1380 effect->setDevice(mInDevice); 1381 effect->setMode(mAudioFlinger->getMode()); 1382 effect->setAudioSource(mAudioSource); 1383 } 1384 // create effect handle and connect it to effect module 1385 handle = new EffectHandle(effect, client, effectClient, priority); 1386 lStatus = handle->initCheck(); 1387 if (lStatus == OK) { 1388 lStatus = effect->addHandle(handle.get()); 1389 } 1390 if (enabled != NULL) { 1391 *enabled = (int)effect->isEnabled(); 1392 } 1393 } 1394 1395Exit: 1396 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 1397 Mutex::Autolock _l(mLock); 1398 if (effectCreated) { 1399 chain->removeEffect_l(effect); 1400 } 1401 if (effectRegistered) { 1402 AudioSystem::unregisterEffect(effect->id()); 1403 } 1404 if (chainCreated) { 1405 removeEffectChain_l(chain); 1406 } 1407 handle.clear(); 1408 } 1409 1410 *status = lStatus; 1411 return handle; 1412} 1413 1414sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect(audio_session_t sessionId, 1415 int effectId) 1416{ 1417 Mutex::Autolock _l(mLock); 1418 return getEffect_l(sessionId, effectId); 1419} 1420 1421sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(audio_session_t sessionId, 1422 int effectId) 1423{ 1424 sp<EffectChain> chain = getEffectChain_l(sessionId); 1425 return chain != 0 ? chain->getEffectFromId_l(effectId) : 0; 1426} 1427 1428// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 1429// PlaybackThread::mLock held 1430status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 1431{ 1432 // check for existing effect chain with the requested audio session 1433 audio_session_t sessionId = effect->sessionId(); 1434 sp<EffectChain> chain = getEffectChain_l(sessionId); 1435 bool chainCreated = false; 1436 1437 ALOGD_IF((mType == OFFLOAD) && !effect->isOffloadable(), 1438 "addEffect_l() on offloaded thread %p: effect %s does not support offload flags %x", 1439 this, effect->desc().name, effect->desc().flags); 1440 1441 if (chain == 0) { 1442 // create a new chain for this session 1443 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 1444 chain = new EffectChain(this, sessionId); 1445 addEffectChain_l(chain); 1446 chain->setStrategy(getStrategyForSession_l(sessionId)); 1447 chainCreated = true; 1448 } 1449 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 1450 1451 if (chain->getEffectFromId_l(effect->id()) != 0) { 1452 ALOGW("addEffect_l() %p effect %s already present in chain %p", 1453 this, effect->desc().name, chain.get()); 1454 return BAD_VALUE; 1455 } 1456 1457 effect->setOffloaded(mType == OFFLOAD, mId); 1458 1459 status_t status = chain->addEffect_l(effect); 1460 if (status != NO_ERROR) { 1461 if (chainCreated) { 1462 removeEffectChain_l(chain); 1463 } 1464 return status; 1465 } 1466 1467 effect->setDevice(mOutDevice); 1468 effect->setDevice(mInDevice); 1469 effect->setMode(mAudioFlinger->getMode()); 1470 effect->setAudioSource(mAudioSource); 1471 return NO_ERROR; 1472} 1473 1474void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 1475 1476 ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 1477 effect_descriptor_t desc = effect->desc(); 1478 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 1479 detachAuxEffect_l(effect->id()); 1480 } 1481 1482 sp<EffectChain> chain = effect->chain().promote(); 1483 if (chain != 0) { 1484 // remove effect chain if removing last effect 1485 if (chain->removeEffect_l(effect) == 0) { 1486 removeEffectChain_l(chain); 1487 } 1488 } else { 1489 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 1490 } 1491} 1492 1493void AudioFlinger::ThreadBase::lockEffectChains_l( 1494 Vector< sp<AudioFlinger::EffectChain> >& effectChains) 1495{ 1496 effectChains = mEffectChains; 1497 for (size_t i = 0; i < mEffectChains.size(); i++) { 1498 mEffectChains[i]->lock(); 1499 } 1500} 1501 1502void AudioFlinger::ThreadBase::unlockEffectChains( 1503 const Vector< sp<AudioFlinger::EffectChain> >& effectChains) 1504{ 1505 for (size_t i = 0; i < effectChains.size(); i++) { 1506 effectChains[i]->unlock(); 1507 } 1508} 1509 1510sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(audio_session_t sessionId) 1511{ 1512 Mutex::Autolock _l(mLock); 1513 return getEffectChain_l(sessionId); 1514} 1515 1516sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(audio_session_t sessionId) 1517 const 1518{ 1519 size_t size = mEffectChains.size(); 1520 for (size_t i = 0; i < size; i++) { 1521 if (mEffectChains[i]->sessionId() == sessionId) { 1522 return mEffectChains[i]; 1523 } 1524 } 1525 return 0; 1526} 1527 1528void AudioFlinger::ThreadBase::setMode(audio_mode_t mode) 1529{ 1530 Mutex::Autolock _l(mLock); 1531 size_t size = mEffectChains.size(); 1532 for (size_t i = 0; i < size; i++) { 1533 mEffectChains[i]->setMode_l(mode); 1534 } 1535} 1536 1537void AudioFlinger::ThreadBase::getAudioPortConfig(struct audio_port_config *config) 1538{ 1539 config->type = AUDIO_PORT_TYPE_MIX; 1540 config->ext.mix.handle = mId; 1541 config->sample_rate = mSampleRate; 1542 config->format = mFormat; 1543 config->channel_mask = mChannelMask; 1544 config->config_mask = AUDIO_PORT_CONFIG_SAMPLE_RATE|AUDIO_PORT_CONFIG_CHANNEL_MASK| 1545 AUDIO_PORT_CONFIG_FORMAT; 1546} 1547 1548void AudioFlinger::ThreadBase::systemReady() 1549{ 1550 Mutex::Autolock _l(mLock); 1551 if (mSystemReady) { 1552 return; 1553 } 1554 mSystemReady = true; 1555 1556 for (size_t i = 0; i < mPendingConfigEvents.size(); i++) { 1557 sendConfigEvent_l(mPendingConfigEvents.editItemAt(i)); 1558 } 1559 mPendingConfigEvents.clear(); 1560} 1561 1562 1563// ---------------------------------------------------------------------------- 1564// Playback 1565// ---------------------------------------------------------------------------- 1566 1567AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, 1568 AudioStreamOut* output, 1569 audio_io_handle_t id, 1570 audio_devices_t device, 1571 type_t type, 1572 bool systemReady, 1573 uint32_t bitRate) 1574 : ThreadBase(audioFlinger, id, device, AUDIO_DEVICE_NONE, type, systemReady), 1575 mNormalFrameCount(0), mSinkBuffer(NULL), 1576 mMixerBufferEnabled(AudioFlinger::kEnableExtendedPrecision), 1577 mMixerBuffer(NULL), 1578 mMixerBufferSize(0), 1579 mMixerBufferFormat(AUDIO_FORMAT_INVALID), 1580 mMixerBufferValid(false), 1581 mEffectBufferEnabled(AudioFlinger::kEnableExtendedPrecision), 1582 mEffectBuffer(NULL), 1583 mEffectBufferSize(0), 1584 mEffectBufferFormat(AUDIO_FORMAT_INVALID), 1585 mEffectBufferValid(false), 1586 mSuspended(0), mBytesWritten(0), 1587 mFramesWritten(0), 1588 mActiveTracksGeneration(0), 1589 // mStreamTypes[] initialized in constructor body 1590 mOutput(output), 1591 mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false), 1592 mMixerStatus(MIXER_IDLE), 1593 mMixerStatusIgnoringFastTracks(MIXER_IDLE), 1594 mStandbyDelayNs(AudioFlinger::mStandbyTimeInNsecs), 1595 mBytesRemaining(0), 1596 mCurrentWriteLength(0), 1597 mUseAsyncWrite(false), 1598 mWriteAckSequence(0), 1599 mDrainSequence(0), 1600 mSignalPending(false), 1601 mScreenState(AudioFlinger::mScreenState), 1602 // index 0 is reserved for normal mixer's submix 1603 mFastTrackAvailMask(((1 << FastMixerState::kMaxFastTracks) - 1) & ~1), 1604 mHwSupportsPause(false), mHwPaused(false), mFlushPending(false) 1605{ 1606 snprintf(mThreadName, kThreadNameLength, "AudioOut_%X", id); 1607 mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mThreadName); 1608 1609 // Assumes constructor is called by AudioFlinger with it's mLock held, but 1610 // it would be safer to explicitly pass initial masterVolume/masterMute as 1611 // parameter. 1612 // 1613 // If the HAL we are using has support for master volume or master mute, 1614 // then do not attenuate or mute during mixing (just leave the volume at 1.0 1615 // and the mute set to false). 1616 mMasterVolume = audioFlinger->masterVolume_l(); 1617 mMasterMute = audioFlinger->masterMute_l(); 1618 if (mOutput && mOutput->audioHwDev) { 1619 if (mOutput->audioHwDev->canSetMasterVolume()) { 1620 mMasterVolume = 1.0; 1621 } 1622 1623 if (mOutput->audioHwDev->canSetMasterMute()) { 1624 mMasterMute = false; 1625 } 1626 } 1627 1628 readOutputParameters_l(); 1629 1630 // ++ operator does not compile 1631 for (audio_stream_type_t stream = AUDIO_STREAM_MIN; stream < AUDIO_STREAM_CNT; 1632 stream = (audio_stream_type_t) (stream + 1)) { 1633 mStreamTypes[stream].volume = mAudioFlinger->streamVolume_l(stream); 1634 mStreamTypes[stream].mute = mAudioFlinger->streamMute_l(stream); 1635 } 1636 1637 if (audio_has_proportional_frames(mFormat)) { 1638 mBufferDurationUs = (uint32_t)((mNormalFrameCount * 1000000LL) / mSampleRate); 1639 } else { 1640 bitRate = bitRate != 0 ? bitRate : kOffloadDefaultBitRateBps; 1641 mBufferDurationUs = (uint32_t)((mBufferSize * 8 * 1000000LL) / bitRate); 1642 } 1643} 1644 1645AudioFlinger::PlaybackThread::~PlaybackThread() 1646{ 1647 mAudioFlinger->unregisterWriter(mNBLogWriter); 1648 free(mSinkBuffer); 1649 free(mMixerBuffer); 1650 free(mEffectBuffer); 1651} 1652 1653void AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) 1654{ 1655 dumpInternals(fd, args); 1656 dumpTracks(fd, args); 1657 dumpEffectChains(fd, args); 1658} 1659 1660void AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args __unused) 1661{ 1662 const size_t SIZE = 256; 1663 char buffer[SIZE]; 1664 String8 result; 1665 1666 result.appendFormat(" Stream volumes in dB: "); 1667 for (int i = 0; i < AUDIO_STREAM_CNT; ++i) { 1668 const stream_type_t *st = &mStreamTypes[i]; 1669 if (i > 0) { 1670 result.appendFormat(", "); 1671 } 1672 result.appendFormat("%d:%.2g", i, 20.0 * log10(st->volume)); 1673 if (st->mute) { 1674 result.append("M"); 1675 } 1676 } 1677 result.append("\n"); 1678 write(fd, result.string(), result.length()); 1679 result.clear(); 1680 1681 // These values are "raw"; they will wrap around. See prepareTracks_l() for a better way. 1682 FastTrackUnderruns underruns = getFastTrackUnderruns(0); 1683 dprintf(fd, " Normal mixer raw underrun counters: partial=%u empty=%u\n", 1684 underruns.mBitFields.mPartial, underruns.mBitFields.mEmpty); 1685 1686 size_t numtracks = mTracks.size(); 1687 size_t numactive = mActiveTracks.size(); 1688 dprintf(fd, " %d Tracks", numtracks); 1689 size_t numactiveseen = 0; 1690 if (numtracks) { 1691 dprintf(fd, " of which %d are active\n", numactive); 1692 Track::appendDumpHeader(result); 1693 for (size_t i = 0; i < numtracks; ++i) { 1694 sp<Track> track = mTracks[i]; 1695 if (track != 0) { 1696 bool active = mActiveTracks.indexOf(track) >= 0; 1697 if (active) { 1698 numactiveseen++; 1699 } 1700 track->dump(buffer, SIZE, active); 1701 result.append(buffer); 1702 } 1703 } 1704 } else { 1705 result.append("\n"); 1706 } 1707 if (numactiveseen != numactive) { 1708 // some tracks in the active list were not in the tracks list 1709 snprintf(buffer, SIZE, " The following tracks are in the active list but" 1710 " not in the track list\n"); 1711 result.append(buffer); 1712 Track::appendDumpHeader(result); 1713 for (size_t i = 0; i < numactive; ++i) { 1714 sp<Track> track = mActiveTracks[i].promote(); 1715 if (track != 0 && mTracks.indexOf(track) < 0) { 1716 track->dump(buffer, SIZE, true); 1717 result.append(buffer); 1718 } 1719 } 1720 } 1721 1722 write(fd, result.string(), result.size()); 1723} 1724 1725void AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 1726{ 1727 dprintf(fd, "\nOutput thread %p type %d (%s):\n", this, type(), threadTypeToString(type())); 1728 1729 dumpBase(fd, args); 1730 1731 dprintf(fd, " Normal frame count: %zu\n", mNormalFrameCount); 1732 dprintf(fd, " Last write occurred (msecs): %llu\n", ns2ms(systemTime() - mLastWriteTime)); 1733 dprintf(fd, " Total writes: %d\n", mNumWrites); 1734 dprintf(fd, " Delayed writes: %d\n", mNumDelayedWrites); 1735 dprintf(fd, " Blocked in write: %s\n", mInWrite ? "yes" : "no"); 1736 dprintf(fd, " Suspend count: %d\n", mSuspended); 1737 dprintf(fd, " Sink buffer : %p\n", mSinkBuffer); 1738 dprintf(fd, " Mixer buffer: %p\n", mMixerBuffer); 1739 dprintf(fd, " Effect buffer: %p\n", mEffectBuffer); 1740 dprintf(fd, " Fast track availMask=%#x\n", mFastTrackAvailMask); 1741 dprintf(fd, " Standby delay ns=%lld\n", (long long)mStandbyDelayNs); 1742 AudioStreamOut *output = mOutput; 1743 audio_output_flags_t flags = output != NULL ? output->flags : AUDIO_OUTPUT_FLAG_NONE; 1744 String8 flagsAsString = outputFlagsToString(flags); 1745 dprintf(fd, " AudioStreamOut: %p flags %#x (%s)\n", output, flags, flagsAsString.string()); 1746} 1747 1748// Thread virtuals 1749 1750void AudioFlinger::PlaybackThread::onFirstRef() 1751{ 1752 run(mThreadName, ANDROID_PRIORITY_URGENT_AUDIO); 1753} 1754 1755// ThreadBase virtuals 1756void AudioFlinger::PlaybackThread::preExit() 1757{ 1758 ALOGV(" preExit()"); 1759 // FIXME this is using hard-coded strings but in the future, this functionality will be 1760 // converted to use audio HAL extensions required to support tunneling 1761 mOutput->stream->common.set_parameters(&mOutput->stream->common, "exiting=1"); 1762} 1763 1764// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 1765sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 1766 const sp<AudioFlinger::Client>& client, 1767 audio_stream_type_t streamType, 1768 uint32_t sampleRate, 1769 audio_format_t format, 1770 audio_channel_mask_t channelMask, 1771 size_t *pFrameCount, 1772 const sp<IMemory>& sharedBuffer, 1773 audio_session_t sessionId, 1774 IAudioFlinger::track_flags_t *flags, 1775 pid_t tid, 1776 int uid, 1777 status_t *status) 1778{ 1779 size_t frameCount = *pFrameCount; 1780 sp<Track> track; 1781 status_t lStatus; 1782 1783 // client expresses a preference for FAST, but we get the final say 1784 if (*flags & IAudioFlinger::TRACK_FAST) { 1785 if ( 1786 // either of these use cases: 1787 ( 1788 // use case 1: shared buffer with any frame count 1789 ( 1790 (sharedBuffer != 0) 1791 ) || 1792 // use case 2: frame count is default or at least as large as HAL 1793 ( 1794 // we formerly checked for a callback handler (non-0 tid), 1795 // but that is no longer required for TRANSFER_OBTAIN mode 1796 ((frameCount == 0) || 1797 (frameCount >= mFrameCount)) 1798 ) 1799 ) && 1800 // PCM data 1801 audio_is_linear_pcm(format) && 1802 // TODO: extract as a data library function that checks that a computationally 1803 // expensive downmixer is not required: isFastOutputChannelConversion() 1804 (channelMask == mChannelMask || 1805 mChannelMask != AUDIO_CHANNEL_OUT_STEREO || 1806 (channelMask == AUDIO_CHANNEL_OUT_MONO 1807 /* && mChannelMask == AUDIO_CHANNEL_OUT_STEREO */)) && 1808 // hardware sample rate 1809 (sampleRate == mSampleRate) && 1810 // normal mixer has an associated fast mixer 1811 hasFastMixer() && 1812 // there are sufficient fast track slots available 1813 (mFastTrackAvailMask != 0) 1814 // FIXME test that MixerThread for this fast track has a capable output HAL 1815 // FIXME add a permission test also? 1816 ) { 1817 // if frameCount not specified, then it defaults to fast mixer (HAL) frame count 1818 if (frameCount == 0) { 1819 // read the fast track multiplier property the first time it is needed 1820 int ok = pthread_once(&sFastTrackMultiplierOnce, sFastTrackMultiplierInit); 1821 if (ok != 0) { 1822 ALOGE("%s pthread_once failed: %d", __func__, ok); 1823 } 1824 frameCount = mFrameCount * sFastTrackMultiplier; 1825 } 1826 ALOGV("AUDIO_OUTPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d", 1827 frameCount, mFrameCount); 1828 } else { 1829 ALOGV("AUDIO_OUTPUT_FLAG_FAST denied: sharedBuffer=%p frameCount=%d " 1830 "mFrameCount=%d format=%#x mFormat=%#x isLinear=%d channelMask=%#x " 1831 "sampleRate=%u mSampleRate=%u " 1832 "hasFastMixer=%d tid=%d fastTrackAvailMask=%#x", 1833 sharedBuffer.get(), frameCount, mFrameCount, format, mFormat, 1834 audio_is_linear_pcm(format), 1835 channelMask, sampleRate, mSampleRate, hasFastMixer(), tid, mFastTrackAvailMask); 1836 *flags &= ~IAudioFlinger::TRACK_FAST; 1837 } 1838 } 1839 // For normal PCM streaming tracks, update minimum frame count. 1840 // For compatibility with AudioTrack calculation, buffer depth is forced 1841 // to be at least 2 x the normal mixer frame count and cover audio hardware latency. 1842 // This is probably too conservative, but legacy application code may depend on it. 1843 // If you change this calculation, also review the start threshold which is related. 1844 if (!(*flags & IAudioFlinger::TRACK_FAST) 1845 && audio_has_proportional_frames(format) && sharedBuffer == 0) { 1846 // this must match AudioTrack.cpp calculateMinFrameCount(). 1847 // TODO: Move to a common library 1848 uint32_t latencyMs = mOutput->stream->get_latency(mOutput->stream); 1849 uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate); 1850 if (minBufCount < 2) { 1851 minBufCount = 2; 1852 } 1853 // For normal mixing tracks, if speed is > 1.0f (normal), AudioTrack 1854 // or the client should compute and pass in a larger buffer request. 1855 size_t minFrameCount = 1856 minBufCount * sourceFramesNeededWithTimestretch( 1857 sampleRate, mNormalFrameCount, 1858 mSampleRate, AUDIO_TIMESTRETCH_SPEED_NORMAL /*speed*/); 1859 if (frameCount < minFrameCount) { // including frameCount == 0 1860 frameCount = minFrameCount; 1861 } 1862 } 1863 *pFrameCount = frameCount; 1864 1865 switch (mType) { 1866 1867 case DIRECT: 1868 if (audio_is_linear_pcm(format)) { // TODO maybe use audio_has_proportional_frames()? 1869 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1870 ALOGE("createTrack_l() Bad parameter: sampleRate %u format %#x, channelMask 0x%08x " 1871 "for output %p with format %#x", 1872 sampleRate, format, channelMask, mOutput, mFormat); 1873 lStatus = BAD_VALUE; 1874 goto Exit; 1875 } 1876 } 1877 break; 1878 1879 case OFFLOAD: 1880 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1881 ALOGE("createTrack_l() Bad parameter: sampleRate %d format %#x, channelMask 0x%08x \"" 1882 "for output %p with format %#x", 1883 sampleRate, format, channelMask, mOutput, mFormat); 1884 lStatus = BAD_VALUE; 1885 goto Exit; 1886 } 1887 break; 1888 1889 default: 1890 if (!audio_is_linear_pcm(format)) { 1891 ALOGE("createTrack_l() Bad parameter: format %#x \"" 1892 "for output %p with format %#x", 1893 format, mOutput, mFormat); 1894 lStatus = BAD_VALUE; 1895 goto Exit; 1896 } 1897 if (sampleRate > mSampleRate * AUDIO_RESAMPLER_DOWN_RATIO_MAX) { 1898 ALOGE("Sample rate out of range: %u mSampleRate %u", sampleRate, mSampleRate); 1899 lStatus = BAD_VALUE; 1900 goto Exit; 1901 } 1902 break; 1903 1904 } 1905 1906 lStatus = initCheck(); 1907 if (lStatus != NO_ERROR) { 1908 ALOGE("createTrack_l() audio driver not initialized"); 1909 goto Exit; 1910 } 1911 1912 { // scope for mLock 1913 Mutex::Autolock _l(mLock); 1914 1915 // all tracks in same audio session must share the same routing strategy otherwise 1916 // conflicts will happen when tracks are moved from one output to another by audio policy 1917 // manager 1918 uint32_t strategy = AudioSystem::getStrategyForStream(streamType); 1919 for (size_t i = 0; i < mTracks.size(); ++i) { 1920 sp<Track> t = mTracks[i]; 1921 if (t != 0 && t->isExternalTrack()) { 1922 uint32_t actual = AudioSystem::getStrategyForStream(t->streamType()); 1923 if (sessionId == t->sessionId() && strategy != actual) { 1924 ALOGE("createTrack_l() mismatched strategy; expected %u but found %u", 1925 strategy, actual); 1926 lStatus = BAD_VALUE; 1927 goto Exit; 1928 } 1929 } 1930 } 1931 1932 track = new Track(this, client, streamType, sampleRate, format, 1933 channelMask, frameCount, NULL, sharedBuffer, 1934 sessionId, uid, *flags, TrackBase::TYPE_DEFAULT); 1935 1936 lStatus = track != 0 ? track->initCheck() : (status_t) NO_MEMORY; 1937 if (lStatus != NO_ERROR) { 1938 ALOGE("createTrack_l() initCheck failed %d; no control block?", lStatus); 1939 // track must be cleared from the caller as the caller has the AF lock 1940 goto Exit; 1941 } 1942 mTracks.add(track); 1943 1944 sp<EffectChain> chain = getEffectChain_l(sessionId); 1945 if (chain != 0) { 1946 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1947 track->setMainBuffer(chain->inBuffer()); 1948 chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType())); 1949 chain->incTrackCnt(); 1950 } 1951 1952 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 1953 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 1954 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 1955 // so ask activity manager to do this on our behalf 1956 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 1957 } 1958 } 1959 1960 lStatus = NO_ERROR; 1961 1962Exit: 1963 *status = lStatus; 1964 return track; 1965} 1966 1967uint32_t AudioFlinger::PlaybackThread::correctLatency_l(uint32_t latency) const 1968{ 1969 return latency; 1970} 1971 1972uint32_t AudioFlinger::PlaybackThread::latency() const 1973{ 1974 Mutex::Autolock _l(mLock); 1975 return latency_l(); 1976} 1977uint32_t AudioFlinger::PlaybackThread::latency_l() const 1978{ 1979 if (initCheck() == NO_ERROR) { 1980 return correctLatency_l(mOutput->stream->get_latency(mOutput->stream)); 1981 } else { 1982 return 0; 1983 } 1984} 1985 1986void AudioFlinger::PlaybackThread::setMasterVolume(float value) 1987{ 1988 Mutex::Autolock _l(mLock); 1989 // Don't apply master volume in SW if our HAL can do it for us. 1990 if (mOutput && mOutput->audioHwDev && 1991 mOutput->audioHwDev->canSetMasterVolume()) { 1992 mMasterVolume = 1.0; 1993 } else { 1994 mMasterVolume = value; 1995 } 1996} 1997 1998void AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1999{ 2000 Mutex::Autolock _l(mLock); 2001 // Don't apply master mute in SW if our HAL can do it for us. 2002 if (mOutput && mOutput->audioHwDev && 2003 mOutput->audioHwDev->canSetMasterMute()) { 2004 mMasterMute = false; 2005 } else { 2006 mMasterMute = muted; 2007 } 2008} 2009 2010void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) 2011{ 2012 Mutex::Autolock _l(mLock); 2013 mStreamTypes[stream].volume = value; 2014 broadcast_l(); 2015} 2016 2017void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) 2018{ 2019 Mutex::Autolock _l(mLock); 2020 mStreamTypes[stream].mute = muted; 2021 broadcast_l(); 2022} 2023 2024float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const 2025{ 2026 Mutex::Autolock _l(mLock); 2027 return mStreamTypes[stream].volume; 2028} 2029 2030// addTrack_l() must be called with ThreadBase::mLock held 2031status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 2032{ 2033 status_t status = ALREADY_EXISTS; 2034 2035 if (mActiveTracks.indexOf(track) < 0) { 2036 // the track is newly added, make sure it fills up all its 2037 // buffers before playing. This is to ensure the client will 2038 // effectively get the latency it requested. 2039 if (track->isExternalTrack()) { 2040 TrackBase::track_state state = track->mState; 2041 mLock.unlock(); 2042 status = AudioSystem::startOutput(mId, track->streamType(), 2043 track->sessionId()); 2044 mLock.lock(); 2045 // abort track was stopped/paused while we released the lock 2046 if (state != track->mState) { 2047 if (status == NO_ERROR) { 2048 mLock.unlock(); 2049 AudioSystem::stopOutput(mId, track->streamType(), 2050 track->sessionId()); 2051 mLock.lock(); 2052 } 2053 return INVALID_OPERATION; 2054 } 2055 // abort if start is rejected by audio policy manager 2056 if (status != NO_ERROR) { 2057 return PERMISSION_DENIED; 2058 } 2059#ifdef ADD_BATTERY_DATA 2060 // to track the speaker usage 2061 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 2062#endif 2063 } 2064 2065 // set retry count for buffer fill 2066 if (track->isOffloaded()) { 2067 track->mRetryCount = kMaxTrackStartupRetriesOffload; 2068 } else { 2069 track->mRetryCount = kMaxTrackStartupRetries; 2070 } 2071 2072 track->mFillingUpStatus = track->sharedBuffer() != 0 ? Track::FS_FILLED : Track::FS_FILLING; 2073 track->mResetDone = false; 2074 track->mPresentationCompleteFrames = 0; 2075 mActiveTracks.add(track); 2076 mWakeLockUids.add(track->uid()); 2077 mActiveTracksGeneration++; 2078 mLatestActiveTrack = track; 2079 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 2080 if (chain != 0) { 2081 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), 2082 track->sessionId()); 2083 chain->incActiveTrackCnt(); 2084 } 2085 2086 status = NO_ERROR; 2087 } 2088 2089 onAddNewTrack_l(); 2090 return status; 2091} 2092 2093bool AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 2094{ 2095 track->terminate(); 2096 // active tracks are removed by threadLoop() 2097 bool trackActive = (mActiveTracks.indexOf(track) >= 0); 2098 track->mState = TrackBase::STOPPED; 2099 if (!trackActive) { 2100 removeTrack_l(track); 2101 } else if (track->isFastTrack() || track->isOffloaded() || track->isDirect()) { 2102 track->mState = TrackBase::STOPPING_1; 2103 } 2104 2105 return trackActive; 2106} 2107 2108void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 2109{ 2110 track->triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 2111 mTracks.remove(track); 2112 deleteTrackName_l(track->name()); 2113 // redundant as track is about to be destroyed, for dumpsys only 2114 track->mName = -1; 2115 if (track->isFastTrack()) { 2116 int index = track->mFastIndex; 2117 ALOG_ASSERT(0 < index && index < (int)FastMixerState::kMaxFastTracks); 2118 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << index))); 2119 mFastTrackAvailMask |= 1 << index; 2120 // redundant as track is about to be destroyed, for dumpsys only 2121 track->mFastIndex = -1; 2122 } 2123 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 2124 if (chain != 0) { 2125 chain->decTrackCnt(); 2126 } 2127} 2128 2129void AudioFlinger::PlaybackThread::broadcast_l() 2130{ 2131 // Thread could be blocked waiting for async 2132 // so signal it to handle state changes immediately 2133 // If threadLoop is currently unlocked a signal of mWaitWorkCV will 2134 // be lost so we also flag to prevent it blocking on mWaitWorkCV 2135 mSignalPending = true; 2136 mWaitWorkCV.broadcast(); 2137} 2138 2139String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 2140{ 2141 Mutex::Autolock _l(mLock); 2142 if (initCheck() != NO_ERROR) { 2143 return String8(); 2144 } 2145 2146 char *s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 2147 const String8 out_s8(s); 2148 free(s); 2149 return out_s8; 2150} 2151 2152void AudioFlinger::PlaybackThread::ioConfigChanged(audio_io_config_event event, pid_t pid) { 2153 sp<AudioIoDescriptor> desc = new AudioIoDescriptor(); 2154 ALOGV("PlaybackThread::ioConfigChanged, thread %p, event %d", this, event); 2155 2156 desc->mIoHandle = mId; 2157 2158 switch (event) { 2159 case AUDIO_OUTPUT_OPENED: 2160 case AUDIO_OUTPUT_CONFIG_CHANGED: 2161 desc->mPatch = mPatch; 2162 desc->mChannelMask = mChannelMask; 2163 desc->mSamplingRate = mSampleRate; 2164 desc->mFormat = mFormat; 2165 desc->mFrameCount = mNormalFrameCount; // FIXME see 2166 // AudioFlinger::frameCount(audio_io_handle_t) 2167 desc->mLatency = latency_l(); 2168 break; 2169 2170 case AUDIO_OUTPUT_CLOSED: 2171 default: 2172 break; 2173 } 2174 mAudioFlinger->ioConfigChanged(event, desc, pid); 2175} 2176 2177void AudioFlinger::PlaybackThread::writeCallback() 2178{ 2179 ALOG_ASSERT(mCallbackThread != 0); 2180 mCallbackThread->resetWriteBlocked(); 2181} 2182 2183void AudioFlinger::PlaybackThread::drainCallback() 2184{ 2185 ALOG_ASSERT(mCallbackThread != 0); 2186 mCallbackThread->resetDraining(); 2187} 2188 2189void AudioFlinger::PlaybackThread::resetWriteBlocked(uint32_t sequence) 2190{ 2191 Mutex::Autolock _l(mLock); 2192 // reject out of sequence requests 2193 if ((mWriteAckSequence & 1) && (sequence == mWriteAckSequence)) { 2194 mWriteAckSequence &= ~1; 2195 mWaitWorkCV.signal(); 2196 } 2197} 2198 2199void AudioFlinger::PlaybackThread::resetDraining(uint32_t sequence) 2200{ 2201 Mutex::Autolock _l(mLock); 2202 // reject out of sequence requests 2203 if ((mDrainSequence & 1) && (sequence == mDrainSequence)) { 2204 mDrainSequence &= ~1; 2205 mWaitWorkCV.signal(); 2206 } 2207} 2208 2209// static 2210int AudioFlinger::PlaybackThread::asyncCallback(stream_callback_event_t event, 2211 void *param __unused, 2212 void *cookie) 2213{ 2214 AudioFlinger::PlaybackThread *me = (AudioFlinger::PlaybackThread *)cookie; 2215 ALOGV("asyncCallback() event %d", event); 2216 switch (event) { 2217 case STREAM_CBK_EVENT_WRITE_READY: 2218 me->writeCallback(); 2219 break; 2220 case STREAM_CBK_EVENT_DRAIN_READY: 2221 me->drainCallback(); 2222 break; 2223 default: 2224 ALOGW("asyncCallback() unknown event %d", event); 2225 break; 2226 } 2227 return 0; 2228} 2229 2230void AudioFlinger::PlaybackThread::readOutputParameters_l() 2231{ 2232 // unfortunately we have no way of recovering from errors here, hence the LOG_ALWAYS_FATAL 2233 mSampleRate = mOutput->getSampleRate(); 2234 mChannelMask = mOutput->getChannelMask(); 2235 if (!audio_is_output_channel(mChannelMask)) { 2236 LOG_ALWAYS_FATAL("HAL channel mask %#x not valid for output", mChannelMask); 2237 } 2238 if ((mType == MIXER || mType == DUPLICATING) 2239 && !isValidPcmSinkChannelMask(mChannelMask)) { 2240 LOG_ALWAYS_FATAL("HAL channel mask %#x not supported for mixed output", 2241 mChannelMask); 2242 } 2243 mChannelCount = audio_channel_count_from_out_mask(mChannelMask); 2244 2245 // Get actual HAL format. 2246 mHALFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 2247 // Get format from the shim, which will be different than the HAL format 2248 // if playing compressed audio over HDMI passthrough. 2249 mFormat = mOutput->getFormat(); 2250 if (!audio_is_valid_format(mFormat)) { 2251 LOG_ALWAYS_FATAL("HAL format %#x not valid for output", mFormat); 2252 } 2253 if ((mType == MIXER || mType == DUPLICATING) 2254 && !isValidPcmSinkFormat(mFormat)) { 2255 LOG_FATAL("HAL format %#x not supported for mixed output", 2256 mFormat); 2257 } 2258 mFrameSize = mOutput->getFrameSize(); 2259 mBufferSize = mOutput->stream->common.get_buffer_size(&mOutput->stream->common); 2260 mFrameCount = mBufferSize / mFrameSize; 2261 if (mFrameCount & 15) { 2262 ALOGW("HAL output buffer size is %u frames but AudioMixer requires multiples of 16 frames", 2263 mFrameCount); 2264 } 2265 2266 if ((mOutput->flags & AUDIO_OUTPUT_FLAG_NON_BLOCKING) && 2267 (mOutput->stream->set_callback != NULL)) { 2268 if (mOutput->stream->set_callback(mOutput->stream, 2269 AudioFlinger::PlaybackThread::asyncCallback, this) == 0) { 2270 mUseAsyncWrite = true; 2271 mCallbackThread = new AudioFlinger::AsyncCallbackThread(this); 2272 } 2273 } 2274 2275 mHwSupportsPause = false; 2276 if (mOutput->flags & AUDIO_OUTPUT_FLAG_DIRECT) { 2277 if (mOutput->stream->pause != NULL) { 2278 if (mOutput->stream->resume != NULL) { 2279 mHwSupportsPause = true; 2280 } else { 2281 ALOGW("direct output implements pause but not resume"); 2282 } 2283 } else if (mOutput->stream->resume != NULL) { 2284 ALOGW("direct output implements resume but not pause"); 2285 } 2286 } 2287 if (!mHwSupportsPause && mOutput->flags & AUDIO_OUTPUT_FLAG_HW_AV_SYNC) { 2288 LOG_ALWAYS_FATAL("HW_AV_SYNC requested but HAL does not implement pause and resume"); 2289 } 2290 2291 if (mType == DUPLICATING && mMixerBufferEnabled && mEffectBufferEnabled) { 2292 // For best precision, we use float instead of the associated output 2293 // device format (typically PCM 16 bit). 2294 2295 mFormat = AUDIO_FORMAT_PCM_FLOAT; 2296 mFrameSize = mChannelCount * audio_bytes_per_sample(mFormat); 2297 mBufferSize = mFrameSize * mFrameCount; 2298 2299 // TODO: We currently use the associated output device channel mask and sample rate. 2300 // (1) Perhaps use the ORed channel mask of all downstream MixerThreads 2301 // (if a valid mask) to avoid premature downmix. 2302 // (2) Perhaps use the maximum sample rate of all downstream MixerThreads 2303 // instead of the output device sample rate to avoid loss of high frequency information. 2304 // This may need to be updated as MixerThread/OutputTracks are added and not here. 2305 } 2306 2307 // Calculate size of normal sink buffer relative to the HAL output buffer size 2308 double multiplier = 1.0; 2309 if (mType == MIXER && (kUseFastMixer == FastMixer_Static || 2310 kUseFastMixer == FastMixer_Dynamic)) { 2311 size_t minNormalFrameCount = (kMinNormalSinkBufferSizeMs * mSampleRate) / 1000; 2312 size_t maxNormalFrameCount = (kMaxNormalSinkBufferSizeMs * mSampleRate) / 1000; 2313 // round up minimum and round down maximum to nearest 16 frames to satisfy AudioMixer 2314 minNormalFrameCount = (minNormalFrameCount + 15) & ~15; 2315 maxNormalFrameCount = maxNormalFrameCount & ~15; 2316 if (maxNormalFrameCount < minNormalFrameCount) { 2317 maxNormalFrameCount = minNormalFrameCount; 2318 } 2319 multiplier = (double) minNormalFrameCount / (double) mFrameCount; 2320 if (multiplier <= 1.0) { 2321 multiplier = 1.0; 2322 } else if (multiplier <= 2.0) { 2323 if (2 * mFrameCount <= maxNormalFrameCount) { 2324 multiplier = 2.0; 2325 } else { 2326 multiplier = (double) maxNormalFrameCount / (double) mFrameCount; 2327 } 2328 } else { 2329 // prefer an even multiplier, for compatibility with doubling of fast tracks due to HAL 2330 // SRC (it would be unusual for the normal sink buffer size to not be a multiple of fast 2331 // track, but we sometimes have to do this to satisfy the maximum frame count 2332 // constraint) 2333 // FIXME this rounding up should not be done if no HAL SRC 2334 uint32_t truncMult = (uint32_t) multiplier; 2335 if ((truncMult & 1)) { 2336 if ((truncMult + 1) * mFrameCount <= maxNormalFrameCount) { 2337 ++truncMult; 2338 } 2339 } 2340 multiplier = (double) truncMult; 2341 } 2342 } 2343 mNormalFrameCount = multiplier * mFrameCount; 2344 // round up to nearest 16 frames to satisfy AudioMixer 2345 if (mType == MIXER || mType == DUPLICATING) { 2346 mNormalFrameCount = (mNormalFrameCount + 15) & ~15; 2347 } 2348 ALOGI("HAL output buffer size %u frames, normal sink buffer size %u frames", mFrameCount, 2349 mNormalFrameCount); 2350 2351 // Check if we want to throttle the processing to no more than 2x normal rate 2352 mThreadThrottle = property_get_bool("af.thread.throttle", true /* default_value */); 2353 mThreadThrottleTimeMs = 0; 2354 mThreadThrottleEndMs = 0; 2355 mHalfBufferMs = mNormalFrameCount * 1000 / (2 * mSampleRate); 2356 2357 // mSinkBuffer is the sink buffer. Size is always multiple-of-16 frames. 2358 // Originally this was int16_t[] array, need to remove legacy implications. 2359 free(mSinkBuffer); 2360 mSinkBuffer = NULL; 2361 // For sink buffer size, we use the frame size from the downstream sink to avoid problems 2362 // with non PCM formats for compressed music, e.g. AAC, and Offload threads. 2363 const size_t sinkBufferSize = mNormalFrameCount * mFrameSize; 2364 (void)posix_memalign(&mSinkBuffer, 32, sinkBufferSize); 2365 2366 // We resize the mMixerBuffer according to the requirements of the sink buffer which 2367 // drives the output. 2368 free(mMixerBuffer); 2369 mMixerBuffer = NULL; 2370 if (mMixerBufferEnabled) { 2371 mMixerBufferFormat = AUDIO_FORMAT_PCM_FLOAT; // also valid: AUDIO_FORMAT_PCM_16_BIT. 2372 mMixerBufferSize = mNormalFrameCount * mChannelCount 2373 * audio_bytes_per_sample(mMixerBufferFormat); 2374 (void)posix_memalign(&mMixerBuffer, 32, mMixerBufferSize); 2375 } 2376 free(mEffectBuffer); 2377 mEffectBuffer = NULL; 2378 if (mEffectBufferEnabled) { 2379 mEffectBufferFormat = AUDIO_FORMAT_PCM_16_BIT; // Note: Effects support 16b only 2380 mEffectBufferSize = mNormalFrameCount * mChannelCount 2381 * audio_bytes_per_sample(mEffectBufferFormat); 2382 (void)posix_memalign(&mEffectBuffer, 32, mEffectBufferSize); 2383 } 2384 2385 // force reconfiguration of effect chains and engines to take new buffer size and audio 2386 // parameters into account 2387 // Note that mLock is not held when readOutputParameters_l() is called from the constructor 2388 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 2389 // matter. 2390 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 2391 Vector< sp<EffectChain> > effectChains = mEffectChains; 2392 for (size_t i = 0; i < effectChains.size(); i ++) { 2393 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 2394 } 2395} 2396 2397 2398status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames) 2399{ 2400 if (halFrames == NULL || dspFrames == NULL) { 2401 return BAD_VALUE; 2402 } 2403 Mutex::Autolock _l(mLock); 2404 if (initCheck() != NO_ERROR) { 2405 return INVALID_OPERATION; 2406 } 2407 int64_t framesWritten = mBytesWritten / mFrameSize; 2408 *halFrames = framesWritten; 2409 2410 if (isSuspended()) { 2411 // return an estimation of rendered frames when the output is suspended 2412 size_t latencyFrames = (latency_l() * mSampleRate) / 1000; 2413 *dspFrames = (uint32_t) 2414 (framesWritten >= (int64_t)latencyFrames ? framesWritten - latencyFrames : 0); 2415 return NO_ERROR; 2416 } else { 2417 status_t status; 2418 uint32_t frames; 2419 status = mOutput->getRenderPosition(&frames); 2420 *dspFrames = (size_t)frames; 2421 return status; 2422 } 2423} 2424 2425uint32_t AudioFlinger::PlaybackThread::hasAudioSession(audio_session_t sessionId) const 2426{ 2427 Mutex::Autolock _l(mLock); 2428 uint32_t result = 0; 2429 if (getEffectChain_l(sessionId) != 0) { 2430 result = EFFECT_SESSION; 2431 } 2432 2433 for (size_t i = 0; i < mTracks.size(); ++i) { 2434 sp<Track> track = mTracks[i]; 2435 if (sessionId == track->sessionId() && !track->isInvalid()) { 2436 result |= TRACK_SESSION; 2437 break; 2438 } 2439 } 2440 2441 return result; 2442} 2443 2444uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(audio_session_t sessionId) 2445{ 2446 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 2447 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 2448 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 2449 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 2450 } 2451 for (size_t i = 0; i < mTracks.size(); i++) { 2452 sp<Track> track = mTracks[i]; 2453 if (sessionId == track->sessionId() && !track->isInvalid()) { 2454 return AudioSystem::getStrategyForStream(track->streamType()); 2455 } 2456 } 2457 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 2458} 2459 2460 2461AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 2462{ 2463 Mutex::Autolock _l(mLock); 2464 return mOutput; 2465} 2466 2467AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 2468{ 2469 Mutex::Autolock _l(mLock); 2470 AudioStreamOut *output = mOutput; 2471 mOutput = NULL; 2472 // FIXME FastMixer might also have a raw ptr to mOutputSink; 2473 // must push a NULL and wait for ack 2474 mOutputSink.clear(); 2475 mPipeSink.clear(); 2476 mNormalSink.clear(); 2477 return output; 2478} 2479 2480// this method must always be called either with ThreadBase mLock held or inside the thread loop 2481audio_stream_t* AudioFlinger::PlaybackThread::stream() const 2482{ 2483 if (mOutput == NULL) { 2484 return NULL; 2485 } 2486 return &mOutput->stream->common; 2487} 2488 2489uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() const 2490{ 2491 return (uint32_t)((uint32_t)((mNormalFrameCount * 1000) / mSampleRate) * 1000); 2492} 2493 2494status_t AudioFlinger::PlaybackThread::setSyncEvent(const sp<SyncEvent>& event) 2495{ 2496 if (!isValidSyncEvent(event)) { 2497 return BAD_VALUE; 2498 } 2499 2500 Mutex::Autolock _l(mLock); 2501 2502 for (size_t i = 0; i < mTracks.size(); ++i) { 2503 sp<Track> track = mTracks[i]; 2504 if (event->triggerSession() == track->sessionId()) { 2505 (void) track->setSyncEvent(event); 2506 return NO_ERROR; 2507 } 2508 } 2509 2510 return NAME_NOT_FOUND; 2511} 2512 2513bool AudioFlinger::PlaybackThread::isValidSyncEvent(const sp<SyncEvent>& event) const 2514{ 2515 return event->type() == AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE; 2516} 2517 2518void AudioFlinger::PlaybackThread::threadLoop_removeTracks( 2519 const Vector< sp<Track> >& tracksToRemove) 2520{ 2521 size_t count = tracksToRemove.size(); 2522 if (count > 0) { 2523 for (size_t i = 0 ; i < count ; i++) { 2524 const sp<Track>& track = tracksToRemove.itemAt(i); 2525 if (track->isExternalTrack()) { 2526 AudioSystem::stopOutput(mId, track->streamType(), 2527 track->sessionId()); 2528#ifdef ADD_BATTERY_DATA 2529 // to track the speaker usage 2530 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 2531#endif 2532 if (track->isTerminated()) { 2533 AudioSystem::releaseOutput(mId, track->streamType(), 2534 track->sessionId()); 2535 } 2536 } 2537 } 2538 } 2539} 2540 2541void AudioFlinger::PlaybackThread::checkSilentMode_l() 2542{ 2543 if (!mMasterMute) { 2544 char value[PROPERTY_VALUE_MAX]; 2545 if (property_get("ro.audio.silent", value, "0") > 0) { 2546 char *endptr; 2547 unsigned long ul = strtoul(value, &endptr, 0); 2548 if (*endptr == '\0' && ul != 0) { 2549 ALOGD("Silence is golden"); 2550 // The setprop command will not allow a property to be changed after 2551 // the first time it is set, so we don't have to worry about un-muting. 2552 setMasterMute_l(true); 2553 } 2554 } 2555 } 2556} 2557 2558// shared by MIXER and DIRECT, overridden by DUPLICATING 2559ssize_t AudioFlinger::PlaybackThread::threadLoop_write() 2560{ 2561 // FIXME rewrite to reduce number of system calls 2562 mLastWriteTime = systemTime(); 2563 mInWrite = true; 2564 ssize_t bytesWritten; 2565 const size_t offset = mCurrentWriteLength - mBytesRemaining; 2566 2567 // If an NBAIO sink is present, use it to write the normal mixer's submix 2568 if (mNormalSink != 0) { 2569 2570 const size_t count = mBytesRemaining / mFrameSize; 2571 2572 ATRACE_BEGIN("write"); 2573 // update the setpoint when AudioFlinger::mScreenState changes 2574 uint32_t screenState = AudioFlinger::mScreenState; 2575 if (screenState != mScreenState) { 2576 mScreenState = screenState; 2577 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 2578 if (pipe != NULL) { 2579 pipe->setAvgFrames((mScreenState & 1) ? 2580 (pipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2581 } 2582 } 2583 ssize_t framesWritten = mNormalSink->write((char *)mSinkBuffer + offset, count); 2584 ATRACE_END(); 2585 if (framesWritten > 0) { 2586 bytesWritten = framesWritten * mFrameSize; 2587 } else { 2588 bytesWritten = framesWritten; 2589 } 2590 // otherwise use the HAL / AudioStreamOut directly 2591 } else { 2592 // Direct output and offload threads 2593 2594 if (mUseAsyncWrite) { 2595 ALOGW_IF(mWriteAckSequence & 1, "threadLoop_write(): out of sequence write request"); 2596 mWriteAckSequence += 2; 2597 mWriteAckSequence |= 1; 2598 ALOG_ASSERT(mCallbackThread != 0); 2599 mCallbackThread->setWriteBlocked(mWriteAckSequence); 2600 } 2601 // FIXME We should have an implementation of timestamps for direct output threads. 2602 // They are used e.g for multichannel PCM playback over HDMI. 2603 bytesWritten = mOutput->write((char *)mSinkBuffer + offset, mBytesRemaining); 2604 2605 if (mUseAsyncWrite && 2606 ((bytesWritten < 0) || (bytesWritten == (ssize_t)mBytesRemaining))) { 2607 // do not wait for async callback in case of error of full write 2608 mWriteAckSequence &= ~1; 2609 ALOG_ASSERT(mCallbackThread != 0); 2610 mCallbackThread->setWriteBlocked(mWriteAckSequence); 2611 } 2612 } 2613 2614 mNumWrites++; 2615 mInWrite = false; 2616 mStandby = false; 2617 return bytesWritten; 2618} 2619 2620void AudioFlinger::PlaybackThread::threadLoop_drain() 2621{ 2622 if (mOutput->stream->drain) { 2623 ALOGV("draining %s", (mMixerStatus == MIXER_DRAIN_TRACK) ? "early" : "full"); 2624 if (mUseAsyncWrite) { 2625 ALOGW_IF(mDrainSequence & 1, "threadLoop_drain(): out of sequence drain request"); 2626 mDrainSequence |= 1; 2627 ALOG_ASSERT(mCallbackThread != 0); 2628 mCallbackThread->setDraining(mDrainSequence); 2629 } 2630 mOutput->stream->drain(mOutput->stream, 2631 (mMixerStatus == MIXER_DRAIN_TRACK) ? AUDIO_DRAIN_EARLY_NOTIFY 2632 : AUDIO_DRAIN_ALL); 2633 } 2634} 2635 2636void AudioFlinger::PlaybackThread::threadLoop_exit() 2637{ 2638 { 2639 Mutex::Autolock _l(mLock); 2640 for (size_t i = 0; i < mTracks.size(); i++) { 2641 sp<Track> track = mTracks[i]; 2642 track->invalidate(); 2643 } 2644 } 2645} 2646 2647/* 2648The derived values that are cached: 2649 - mSinkBufferSize from frame count * frame size 2650 - mActiveSleepTimeUs from activeSleepTimeUs() 2651 - mIdleSleepTimeUs from idleSleepTimeUs() 2652 - mStandbyDelayNs from mActiveSleepTimeUs (DIRECT only) or forced to at least 2653 kDefaultStandbyTimeInNsecs when connected to an A2DP device. 2654 - maxPeriod from frame count and sample rate (MIXER only) 2655 2656The parameters that affect these derived values are: 2657 - frame count 2658 - frame size 2659 - sample rate 2660 - device type: A2DP or not 2661 - device latency 2662 - format: PCM or not 2663 - active sleep time 2664 - idle sleep time 2665*/ 2666 2667void AudioFlinger::PlaybackThread::cacheParameters_l() 2668{ 2669 mSinkBufferSize = mNormalFrameCount * mFrameSize; 2670 mActiveSleepTimeUs = activeSleepTimeUs(); 2671 mIdleSleepTimeUs = idleSleepTimeUs(); 2672 2673 // make sure standby delay is not too short when connected to an A2DP sink to avoid 2674 // truncating audio when going to standby. 2675 mStandbyDelayNs = AudioFlinger::mStandbyTimeInNsecs; 2676 if ((mOutDevice & AUDIO_DEVICE_OUT_ALL_A2DP) != 0) { 2677 if (mStandbyDelayNs < kDefaultStandbyTimeInNsecs) { 2678 mStandbyDelayNs = kDefaultStandbyTimeInNsecs; 2679 } 2680 } 2681} 2682 2683void AudioFlinger::PlaybackThread::invalidateTracks(audio_stream_type_t streamType) 2684{ 2685 ALOGV("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 2686 this, streamType, mTracks.size()); 2687 Mutex::Autolock _l(mLock); 2688 2689 size_t size = mTracks.size(); 2690 for (size_t i = 0; i < size; i++) { 2691 sp<Track> t = mTracks[i]; 2692 if (t->streamType() == streamType && t->isExternalTrack()) { 2693 t->invalidate(); 2694 } 2695 } 2696} 2697 2698status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 2699{ 2700 audio_session_t session = chain->sessionId(); 2701 int16_t* buffer = reinterpret_cast<int16_t*>(mEffectBufferEnabled 2702 ? mEffectBuffer : mSinkBuffer); 2703 bool ownsBuffer = false; 2704 2705 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 2706 if (session > AUDIO_SESSION_OUTPUT_MIX) { 2707 // Only one effect chain can be present in direct output thread and it uses 2708 // the sink buffer as input 2709 if (mType != DIRECT) { 2710 size_t numSamples = mNormalFrameCount * mChannelCount; 2711 buffer = new int16_t[numSamples]; 2712 memset(buffer, 0, numSamples * sizeof(int16_t)); 2713 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 2714 ownsBuffer = true; 2715 } 2716 2717 // Attach all tracks with same session ID to this chain. 2718 for (size_t i = 0; i < mTracks.size(); ++i) { 2719 sp<Track> track = mTracks[i]; 2720 if (session == track->sessionId()) { 2721 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), 2722 buffer); 2723 track->setMainBuffer(buffer); 2724 chain->incTrackCnt(); 2725 } 2726 } 2727 2728 // indicate all active tracks in the chain 2729 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 2730 sp<Track> track = mActiveTracks[i].promote(); 2731 if (track == 0) { 2732 continue; 2733 } 2734 if (session == track->sessionId()) { 2735 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 2736 chain->incActiveTrackCnt(); 2737 } 2738 } 2739 } 2740 chain->setThread(this); 2741 chain->setInBuffer(buffer, ownsBuffer); 2742 chain->setOutBuffer(reinterpret_cast<int16_t*>(mEffectBufferEnabled 2743 ? mEffectBuffer : mSinkBuffer)); 2744 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 2745 // chains list in order to be processed last as it contains output stage effects. 2746 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 2747 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 2748 // after track specific effects and before output stage. 2749 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 2750 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX. 2751 // Effect chain for other sessions are inserted at beginning of effect 2752 // chains list to be processed before output mix effects. Relative order between other 2753 // sessions is not important. 2754 static_assert(AUDIO_SESSION_OUTPUT_MIX == 0 && 2755 AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX, 2756 "audio_session_t constants misdefined"); 2757 size_t size = mEffectChains.size(); 2758 size_t i = 0; 2759 for (i = 0; i < size; i++) { 2760 if (mEffectChains[i]->sessionId() < session) { 2761 break; 2762 } 2763 } 2764 mEffectChains.insertAt(chain, i); 2765 checkSuspendOnAddEffectChain_l(chain); 2766 2767 return NO_ERROR; 2768} 2769 2770size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 2771{ 2772 audio_session_t session = chain->sessionId(); 2773 2774 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 2775 2776 for (size_t i = 0; i < mEffectChains.size(); i++) { 2777 if (chain == mEffectChains[i]) { 2778 mEffectChains.removeAt(i); 2779 // detach all active tracks from the chain 2780 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 2781 sp<Track> track = mActiveTracks[i].promote(); 2782 if (track == 0) { 2783 continue; 2784 } 2785 if (session == track->sessionId()) { 2786 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 2787 chain.get(), session); 2788 chain->decActiveTrackCnt(); 2789 } 2790 } 2791 2792 // detach all tracks with same session ID from this chain 2793 for (size_t i = 0; i < mTracks.size(); ++i) { 2794 sp<Track> track = mTracks[i]; 2795 if (session == track->sessionId()) { 2796 track->setMainBuffer(reinterpret_cast<int16_t*>(mSinkBuffer)); 2797 chain->decTrackCnt(); 2798 } 2799 } 2800 break; 2801 } 2802 } 2803 return mEffectChains.size(); 2804} 2805 2806status_t AudioFlinger::PlaybackThread::attachAuxEffect( 2807 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2808{ 2809 Mutex::Autolock _l(mLock); 2810 return attachAuxEffect_l(track, EffectId); 2811} 2812 2813status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 2814 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2815{ 2816 status_t status = NO_ERROR; 2817 2818 if (EffectId == 0) { 2819 track->setAuxBuffer(0, NULL); 2820 } else { 2821 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 2822 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 2823 if (effect != 0) { 2824 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 2825 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 2826 } else { 2827 status = INVALID_OPERATION; 2828 } 2829 } else { 2830 status = BAD_VALUE; 2831 } 2832 } 2833 return status; 2834} 2835 2836void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 2837{ 2838 for (size_t i = 0; i < mTracks.size(); ++i) { 2839 sp<Track> track = mTracks[i]; 2840 if (track->auxEffectId() == effectId) { 2841 attachAuxEffect_l(track, 0); 2842 } 2843 } 2844} 2845 2846bool AudioFlinger::PlaybackThread::threadLoop() 2847{ 2848 Vector< sp<Track> > tracksToRemove; 2849 2850 mStandbyTimeNs = systemTime(); 2851 2852 // MIXER 2853 nsecs_t lastWarning = 0; 2854 2855 // DUPLICATING 2856 // FIXME could this be made local to while loop? 2857 writeFrames = 0; 2858 2859 int lastGeneration = 0; 2860 2861 cacheParameters_l(); 2862 mSleepTimeUs = mIdleSleepTimeUs; 2863 2864 if (mType == MIXER) { 2865 sleepTimeShift = 0; 2866 } 2867 2868 CpuStats cpuStats; 2869 const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid())); 2870 2871 acquireWakeLock(); 2872 2873 // mNBLogWriter->log can only be called while thread mutex mLock is held. 2874 // So if you need to log when mutex is unlocked, set logString to a non-NULL string, 2875 // and then that string will be logged at the next convenient opportunity. 2876 const char *logString = NULL; 2877 2878 checkSilentMode_l(); 2879 2880 while (!exitPending()) 2881 { 2882 cpuStats.sample(myName); 2883 2884 Vector< sp<EffectChain> > effectChains; 2885 2886 { // scope for mLock 2887 2888 Mutex::Autolock _l(mLock); 2889 2890 processConfigEvents_l(); 2891 2892 if (logString != NULL) { 2893 mNBLogWriter->logTimestamp(); 2894 mNBLogWriter->log(logString); 2895 logString = NULL; 2896 } 2897 2898 // Gather the framesReleased counters for all active tracks, 2899 // and associate with the sink frames written out. We need 2900 // this to convert the sink timestamp to the track timestamp. 2901 if (mNormalSink != 0) { 2902 // Note: The DuplicatingThread may not have a mNormalSink. 2903 // We always fetch the timestamp here because often the downstream 2904 // sink will block whie writing. 2905 ExtendedTimestamp timestamp; // use private copy to fetch 2906 (void) mNormalSink->getTimestamp(timestamp); 2907 // copy over kernel info 2908 mTimestamp.mPosition[ExtendedTimestamp::LOCATION_KERNEL] = 2909 timestamp.mPosition[ExtendedTimestamp::LOCATION_KERNEL]; 2910 mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL] = 2911 timestamp.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL]; 2912 } 2913 // mFramesWritten for non-offloaded tracks are contiguous 2914 // even after standby() is called. This is useful for the track frame 2915 // to sink frame mapping. 2916 mTimestamp.mPosition[ExtendedTimestamp::LOCATION_SERVER] = mFramesWritten; 2917 mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_SERVER] = systemTime(); 2918 const size_t size = mActiveTracks.size(); 2919 for (size_t i = 0; i < size; ++i) { 2920 sp<Track> t = mActiveTracks[i].promote(); 2921 if (t != 0 && !t->isFastTrack()) { 2922 t->updateTrackFrameInfo( 2923 t->mAudioTrackServerProxy->framesReleased(), 2924 mFramesWritten, 2925 mTimestamp); 2926 } 2927 } 2928 2929 saveOutputTracks(); 2930 if (mSignalPending) { 2931 // A signal was raised while we were unlocked 2932 mSignalPending = false; 2933 } else if (waitingAsyncCallback_l()) { 2934 if (exitPending()) { 2935 break; 2936 } 2937 bool released = false; 2938 // The following works around a bug in the offload driver. Ideally we would release 2939 // the wake lock every time, but that causes the last offload buffer(s) to be 2940 // dropped while the device is on battery, so we need to hold a wake lock during 2941 // the drain phase. 2942 if (mBytesRemaining && !(mDrainSequence & 1)) { 2943 releaseWakeLock_l(); 2944 released = true; 2945 } 2946 mWakeLockUids.clear(); 2947 mActiveTracksGeneration++; 2948 ALOGV("wait async completion"); 2949 mWaitWorkCV.wait(mLock); 2950 ALOGV("async completion/wake"); 2951 if (released) { 2952 acquireWakeLock_l(); 2953 } 2954 mStandbyTimeNs = systemTime() + mStandbyDelayNs; 2955 mSleepTimeUs = 0; 2956 2957 continue; 2958 } 2959 if ((!mActiveTracks.size() && systemTime() > mStandbyTimeNs) || 2960 isSuspended()) { 2961 // put audio hardware into standby after short delay 2962 if (shouldStandby_l()) { 2963 2964 threadLoop_standby(); 2965 2966 mStandby = true; 2967 } 2968 2969 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2970 // we're about to wait, flush the binder command buffer 2971 IPCThreadState::self()->flushCommands(); 2972 2973 clearOutputTracks(); 2974 2975 if (exitPending()) { 2976 break; 2977 } 2978 2979 releaseWakeLock_l(); 2980 mWakeLockUids.clear(); 2981 mActiveTracksGeneration++; 2982 // wait until we have something to do... 2983 ALOGV("%s going to sleep", myName.string()); 2984 mWaitWorkCV.wait(mLock); 2985 ALOGV("%s waking up", myName.string()); 2986 acquireWakeLock_l(); 2987 2988 mMixerStatus = MIXER_IDLE; 2989 mMixerStatusIgnoringFastTracks = MIXER_IDLE; 2990 mBytesWritten = 0; 2991 mBytesRemaining = 0; 2992 checkSilentMode_l(); 2993 2994 mStandbyTimeNs = systemTime() + mStandbyDelayNs; 2995 mSleepTimeUs = mIdleSleepTimeUs; 2996 if (mType == MIXER) { 2997 sleepTimeShift = 0; 2998 } 2999 3000 continue; 3001 } 3002 } 3003 // mMixerStatusIgnoringFastTracks is also updated internally 3004 mMixerStatus = prepareTracks_l(&tracksToRemove); 3005 3006 // compare with previously applied list 3007 if (lastGeneration != mActiveTracksGeneration) { 3008 // update wakelock 3009 updateWakeLockUids_l(mWakeLockUids); 3010 lastGeneration = mActiveTracksGeneration; 3011 } 3012 3013 // prevent any changes in effect chain list and in each effect chain 3014 // during mixing and effect process as the audio buffers could be deleted 3015 // or modified if an effect is created or deleted 3016 lockEffectChains_l(effectChains); 3017 } // mLock scope ends 3018 3019 if (mBytesRemaining == 0) { 3020 mCurrentWriteLength = 0; 3021 if (mMixerStatus == MIXER_TRACKS_READY) { 3022 // threadLoop_mix() sets mCurrentWriteLength 3023 threadLoop_mix(); 3024 } else if ((mMixerStatus != MIXER_DRAIN_TRACK) 3025 && (mMixerStatus != MIXER_DRAIN_ALL)) { 3026 // threadLoop_sleepTime sets mSleepTimeUs to 0 if data 3027 // must be written to HAL 3028 threadLoop_sleepTime(); 3029 if (mSleepTimeUs == 0) { 3030 mCurrentWriteLength = mSinkBufferSize; 3031 } 3032 } 3033 // Either threadLoop_mix() or threadLoop_sleepTime() should have set 3034 // mMixerBuffer with data if mMixerBufferValid is true and mSleepTimeUs == 0. 3035 // Merge mMixerBuffer data into mEffectBuffer (if any effects are valid) 3036 // or mSinkBuffer (if there are no effects). 3037 // 3038 // This is done pre-effects computation; if effects change to 3039 // support higher precision, this needs to move. 3040 // 3041 // mMixerBufferValid is only set true by MixerThread::prepareTracks_l(). 3042 // TODO use mSleepTimeUs == 0 as an additional condition. 3043 if (mMixerBufferValid) { 3044 void *buffer = mEffectBufferValid ? mEffectBuffer : mSinkBuffer; 3045 audio_format_t format = mEffectBufferValid ? mEffectBufferFormat : mFormat; 3046 3047 // mono blend occurs for mixer threads only (not direct or offloaded) 3048 // and is handled here if we're going directly to the sink. 3049 if (requireMonoBlend() && !mEffectBufferValid) { 3050 mono_blend(mMixerBuffer, mMixerBufferFormat, mChannelCount, mNormalFrameCount, 3051 true /*limit*/); 3052 } 3053 3054 memcpy_by_audio_format(buffer, format, mMixerBuffer, mMixerBufferFormat, 3055 mNormalFrameCount * mChannelCount); 3056 } 3057 3058 mBytesRemaining = mCurrentWriteLength; 3059 if (isSuspended()) { 3060 mSleepTimeUs = suspendSleepTimeUs(); 3061 // simulate write to HAL when suspended 3062 mBytesWritten += mSinkBufferSize; 3063 mFramesWritten += mSinkBufferSize / mFrameSize; 3064 mBytesRemaining = 0; 3065 } 3066 3067 // only process effects if we're going to write 3068 if (mSleepTimeUs == 0 && mType != OFFLOAD) { 3069 for (size_t i = 0; i < effectChains.size(); i ++) { 3070 effectChains[i]->process_l(); 3071 } 3072 } 3073 } 3074 // Process effect chains for offloaded thread even if no audio 3075 // was read from audio track: process only updates effect state 3076 // and thus does have to be synchronized with audio writes but may have 3077 // to be called while waiting for async write callback 3078 if (mType == OFFLOAD) { 3079 for (size_t i = 0; i < effectChains.size(); i ++) { 3080 effectChains[i]->process_l(); 3081 } 3082 } 3083 3084 // Only if the Effects buffer is enabled and there is data in the 3085 // Effects buffer (buffer valid), we need to 3086 // copy into the sink buffer. 3087 // TODO use mSleepTimeUs == 0 as an additional condition. 3088 if (mEffectBufferValid) { 3089 //ALOGV("writing effect buffer to sink buffer format %#x", mFormat); 3090 3091 if (requireMonoBlend()) { 3092 mono_blend(mEffectBuffer, mEffectBufferFormat, mChannelCount, mNormalFrameCount, 3093 true /*limit*/); 3094 } 3095 3096 memcpy_by_audio_format(mSinkBuffer, mFormat, mEffectBuffer, mEffectBufferFormat, 3097 mNormalFrameCount * mChannelCount); 3098 } 3099 3100 // enable changes in effect chain 3101 unlockEffectChains(effectChains); 3102 3103 if (!waitingAsyncCallback()) { 3104 // mSleepTimeUs == 0 means we must write to audio hardware 3105 if (mSleepTimeUs == 0) { 3106 ssize_t ret = 0; 3107 if (mBytesRemaining) { 3108 ret = threadLoop_write(); 3109 if (ret < 0) { 3110 mBytesRemaining = 0; 3111 } else { 3112 mBytesWritten += ret; 3113 mBytesRemaining -= ret; 3114 mFramesWritten += ret / mFrameSize; 3115 } 3116 } else if ((mMixerStatus == MIXER_DRAIN_TRACK) || 3117 (mMixerStatus == MIXER_DRAIN_ALL)) { 3118 threadLoop_drain(); 3119 } 3120 if (mType == MIXER && !mStandby) { 3121 // write blocked detection 3122 nsecs_t now = systemTime(); 3123 nsecs_t delta = now - mLastWriteTime; 3124 if (delta > maxPeriod) { 3125 mNumDelayedWrites++; 3126 if ((now - lastWarning) > kWarningThrottleNs) { 3127 ATRACE_NAME("underrun"); 3128 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 3129 ns2ms(delta), mNumDelayedWrites, this); 3130 lastWarning = now; 3131 } 3132 } 3133 3134 if (mThreadThrottle 3135 && mMixerStatus == MIXER_TRACKS_READY // we are mixing (active tracks) 3136 && ret > 0) { // we wrote something 3137 // Limit MixerThread data processing to no more than twice the 3138 // expected processing rate. 3139 // 3140 // This helps prevent underruns with NuPlayer and other applications 3141 // which may set up buffers that are close to the minimum size, or use 3142 // deep buffers, and rely on a double-buffering sleep strategy to fill. 3143 // 3144 // The throttle smooths out sudden large data drains from the device, 3145 // e.g. when it comes out of standby, which often causes problems with 3146 // (1) mixer threads without a fast mixer (which has its own warm-up) 3147 // (2) minimum buffer sized tracks (even if the track is full, 3148 // the app won't fill fast enough to handle the sudden draw). 3149 3150 const int32_t deltaMs = delta / 1000000; 3151 const int32_t throttleMs = mHalfBufferMs - deltaMs; 3152 if ((signed)mHalfBufferMs >= throttleMs && throttleMs > 0) { 3153 usleep(throttleMs * 1000); 3154 // notify of throttle start on verbose log 3155 ALOGV_IF(mThreadThrottleEndMs == mThreadThrottleTimeMs, 3156 "mixer(%p) throttle begin:" 3157 " ret(%zd) deltaMs(%d) requires sleep %d ms", 3158 this, ret, deltaMs, throttleMs); 3159 mThreadThrottleTimeMs += throttleMs; 3160 } else { 3161 uint32_t diff = mThreadThrottleTimeMs - mThreadThrottleEndMs; 3162 if (diff > 0) { 3163 // notify of throttle end on debug log 3164 ALOGD("mixer(%p) throttle end: throttle time(%u)", this, diff); 3165 mThreadThrottleEndMs = mThreadThrottleTimeMs; 3166 } 3167 } 3168 } 3169 } 3170 3171 } else { 3172 ATRACE_BEGIN("sleep"); 3173 if ((mType == OFFLOAD) && !audio_has_proportional_frames(mFormat)) { 3174 Mutex::Autolock _l(mLock); 3175 if (!mSignalPending && !exitPending()) { 3176 // Do not sleep more than one buffer duration since last write and not 3177 // less than kDirectMinSleepTimeUs 3178 // Wake up if a command is received 3179 nsecs_t now = systemTime(); 3180 uint32_t deltaUs = (uint32_t)((now - mLastWriteTime) / 1000); 3181 uint32_t timeoutUs = mSleepTimeUs; 3182 if (timeoutUs + deltaUs > mBufferDurationUs) { 3183 if (mBufferDurationUs > deltaUs) { 3184 timeoutUs = mBufferDurationUs - deltaUs; 3185 if (timeoutUs < kDirectMinSleepTimeUs) { 3186 timeoutUs = kDirectMinSleepTimeUs; 3187 } 3188 } else { 3189 timeoutUs = kDirectMinSleepTimeUs; 3190 } 3191 } 3192 mWaitWorkCV.waitRelative(mLock, microseconds((nsecs_t)timeoutUs)); 3193 } 3194 } else { 3195 usleep(mSleepTimeUs); 3196 } 3197 ATRACE_END(); 3198 } 3199 } 3200 3201 // Finally let go of removed track(s), without the lock held 3202 // since we can't guarantee the destructors won't acquire that 3203 // same lock. This will also mutate and push a new fast mixer state. 3204 threadLoop_removeTracks(tracksToRemove); 3205 tracksToRemove.clear(); 3206 3207 // FIXME I don't understand the need for this here; 3208 // it was in the original code but maybe the 3209 // assignment in saveOutputTracks() makes this unnecessary? 3210 clearOutputTracks(); 3211 3212 // Effect chains will be actually deleted here if they were removed from 3213 // mEffectChains list during mixing or effects processing 3214 effectChains.clear(); 3215 3216 // FIXME Note that the above .clear() is no longer necessary since effectChains 3217 // is now local to this block, but will keep it for now (at least until merge done). 3218 } 3219 3220 threadLoop_exit(); 3221 3222 if (!mStandby) { 3223 threadLoop_standby(); 3224 mStandby = true; 3225 } 3226 3227 releaseWakeLock(); 3228 mWakeLockUids.clear(); 3229 mActiveTracksGeneration++; 3230 3231 ALOGV("Thread %p type %d exiting", this, mType); 3232 return false; 3233} 3234 3235// removeTracks_l() must be called with ThreadBase::mLock held 3236void AudioFlinger::PlaybackThread::removeTracks_l(const Vector< sp<Track> >& tracksToRemove) 3237{ 3238 size_t count = tracksToRemove.size(); 3239 if (count > 0) { 3240 for (size_t i=0 ; i<count ; i++) { 3241 const sp<Track>& track = tracksToRemove.itemAt(i); 3242 mActiveTracks.remove(track); 3243 mWakeLockUids.remove(track->uid()); 3244 mActiveTracksGeneration++; 3245 ALOGV("removeTracks_l removing track on session %d", track->sessionId()); 3246 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 3247 if (chain != 0) { 3248 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), 3249 track->sessionId()); 3250 chain->decActiveTrackCnt(); 3251 } 3252 if (track->isTerminated()) { 3253 removeTrack_l(track); 3254 } 3255 } 3256 } 3257 3258} 3259 3260status_t AudioFlinger::PlaybackThread::getTimestamp_l(AudioTimestamp& timestamp) 3261{ 3262 if (mNormalSink != 0) { 3263 ExtendedTimestamp ets; 3264 status_t status = mNormalSink->getTimestamp(ets); 3265 if (status == NO_ERROR) { 3266 status = ets.getBestTimestamp(×tamp); 3267 } 3268 return status; 3269 } 3270 if ((mType == OFFLOAD || mType == DIRECT) 3271 && mOutput != NULL && mOutput->stream->get_presentation_position) { 3272 uint64_t position64; 3273 int ret = mOutput->getPresentationPosition(&position64, ×tamp.mTime); 3274 if (ret == 0) { 3275 timestamp.mPosition = (uint32_t)position64; 3276 return NO_ERROR; 3277 } 3278 } 3279 return INVALID_OPERATION; 3280} 3281 3282status_t AudioFlinger::MixerThread::createAudioPatch_l(const struct audio_patch *patch, 3283 audio_patch_handle_t *handle) 3284{ 3285 // if !&IDLE, holds the FastMixer state to restore after new parameters processed 3286 FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE; 3287 if (mFastMixer != 0) { 3288 FastMixerStateQueue *sq = mFastMixer->sq(); 3289 FastMixerState *state = sq->begin(); 3290 if (!(state->mCommand & FastMixerState::IDLE)) { 3291 previousCommand = state->mCommand; 3292 state->mCommand = FastMixerState::HOT_IDLE; 3293 sq->end(); 3294 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3295 } else { 3296 sq->end(false /*didModify*/); 3297 } 3298 } 3299 status_t status = PlaybackThread::createAudioPatch_l(patch, handle); 3300 3301 if (!(previousCommand & FastMixerState::IDLE)) { 3302 ALOG_ASSERT(mFastMixer != 0); 3303 FastMixerStateQueue *sq = mFastMixer->sq(); 3304 FastMixerState *state = sq->begin(); 3305 ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE); 3306 state->mCommand = previousCommand; 3307 sq->end(); 3308 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3309 } 3310 3311 return status; 3312} 3313 3314status_t AudioFlinger::PlaybackThread::createAudioPatch_l(const struct audio_patch *patch, 3315 audio_patch_handle_t *handle) 3316{ 3317 status_t status = NO_ERROR; 3318 3319 // store new device and send to effects 3320 audio_devices_t type = AUDIO_DEVICE_NONE; 3321 for (unsigned int i = 0; i < patch->num_sinks; i++) { 3322 type |= patch->sinks[i].ext.device.type; 3323 } 3324 3325#ifdef ADD_BATTERY_DATA 3326 // when changing the audio output device, call addBatteryData to notify 3327 // the change 3328 if (mOutDevice != type) { 3329 uint32_t params = 0; 3330 // check whether speaker is on 3331 if (type & AUDIO_DEVICE_OUT_SPEAKER) { 3332 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 3333 } 3334 3335 audio_devices_t deviceWithoutSpeaker 3336 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 3337 // check if any other device (except speaker) is on 3338 if (type & deviceWithoutSpeaker) { 3339 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 3340 } 3341 3342 if (params != 0) { 3343 addBatteryData(params); 3344 } 3345 } 3346#endif 3347 3348 for (size_t i = 0; i < mEffectChains.size(); i++) { 3349 mEffectChains[i]->setDevice_l(type); 3350 } 3351 3352 // mPrevOutDevice is the latest device set by createAudioPatch_l(). It is not set when 3353 // the thread is created so that the first patch creation triggers an ioConfigChanged callback 3354 bool configChanged = mPrevOutDevice != type; 3355 mOutDevice = type; 3356 mPatch = *patch; 3357 3358 if (mOutput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 3359 audio_hw_device_t *hwDevice = mOutput->audioHwDev->hwDevice(); 3360 status = hwDevice->create_audio_patch(hwDevice, 3361 patch->num_sources, 3362 patch->sources, 3363 patch->num_sinks, 3364 patch->sinks, 3365 handle); 3366 } else { 3367 char *address; 3368 if (strcmp(patch->sinks[0].ext.device.address, "") != 0) { 3369 //FIXME: we only support address on first sink with HAL version < 3.0 3370 address = audio_device_address_to_parameter( 3371 patch->sinks[0].ext.device.type, 3372 patch->sinks[0].ext.device.address); 3373 } else { 3374 address = (char *)calloc(1, 1); 3375 } 3376 AudioParameter param = AudioParameter(String8(address)); 3377 free(address); 3378 param.addInt(String8(AUDIO_PARAMETER_STREAM_ROUTING), (int)type); 3379 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3380 param.toString().string()); 3381 *handle = AUDIO_PATCH_HANDLE_NONE; 3382 } 3383 if (configChanged) { 3384 mPrevOutDevice = type; 3385 sendIoConfigEvent_l(AUDIO_OUTPUT_CONFIG_CHANGED); 3386 } 3387 return status; 3388} 3389 3390status_t AudioFlinger::MixerThread::releaseAudioPatch_l(const audio_patch_handle_t handle) 3391{ 3392 // if !&IDLE, holds the FastMixer state to restore after new parameters processed 3393 FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE; 3394 if (mFastMixer != 0) { 3395 FastMixerStateQueue *sq = mFastMixer->sq(); 3396 FastMixerState *state = sq->begin(); 3397 if (!(state->mCommand & FastMixerState::IDLE)) { 3398 previousCommand = state->mCommand; 3399 state->mCommand = FastMixerState::HOT_IDLE; 3400 sq->end(); 3401 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3402 } else { 3403 sq->end(false /*didModify*/); 3404 } 3405 } 3406 3407 status_t status = PlaybackThread::releaseAudioPatch_l(handle); 3408 3409 if (!(previousCommand & FastMixerState::IDLE)) { 3410 ALOG_ASSERT(mFastMixer != 0); 3411 FastMixerStateQueue *sq = mFastMixer->sq(); 3412 FastMixerState *state = sq->begin(); 3413 ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE); 3414 state->mCommand = previousCommand; 3415 sq->end(); 3416 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3417 } 3418 3419 return status; 3420} 3421 3422status_t AudioFlinger::PlaybackThread::releaseAudioPatch_l(const audio_patch_handle_t handle) 3423{ 3424 status_t status = NO_ERROR; 3425 3426 mOutDevice = AUDIO_DEVICE_NONE; 3427 3428 if (mOutput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 3429 audio_hw_device_t *hwDevice = mOutput->audioHwDev->hwDevice(); 3430 status = hwDevice->release_audio_patch(hwDevice, handle); 3431 } else { 3432 AudioParameter param; 3433 param.addInt(String8(AUDIO_PARAMETER_STREAM_ROUTING), 0); 3434 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3435 param.toString().string()); 3436 } 3437 return status; 3438} 3439 3440void AudioFlinger::PlaybackThread::addPatchTrack(const sp<PatchTrack>& track) 3441{ 3442 Mutex::Autolock _l(mLock); 3443 mTracks.add(track); 3444} 3445 3446void AudioFlinger::PlaybackThread::deletePatchTrack(const sp<PatchTrack>& track) 3447{ 3448 Mutex::Autolock _l(mLock); 3449 destroyTrack_l(track); 3450} 3451 3452void AudioFlinger::PlaybackThread::getAudioPortConfig(struct audio_port_config *config) 3453{ 3454 ThreadBase::getAudioPortConfig(config); 3455 config->role = AUDIO_PORT_ROLE_SOURCE; 3456 config->ext.mix.hw_module = mOutput->audioHwDev->handle(); 3457 config->ext.mix.usecase.stream = AUDIO_STREAM_DEFAULT; 3458} 3459 3460// ---------------------------------------------------------------------------- 3461 3462AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 3463 audio_io_handle_t id, audio_devices_t device, bool systemReady, type_t type) 3464 : PlaybackThread(audioFlinger, output, id, device, type, systemReady), 3465 // mAudioMixer below 3466 // mFastMixer below 3467 mFastMixerFutex(0), 3468 mMasterMono(false) 3469 // mOutputSink below 3470 // mPipeSink below 3471 // mNormalSink below 3472{ 3473 ALOGV("MixerThread() id=%d device=%#x type=%d", id, device, type); 3474 ALOGV("mSampleRate=%u, mChannelMask=%#x, mChannelCount=%u, mFormat=%d, mFrameSize=%u, " 3475 "mFrameCount=%d, mNormalFrameCount=%d", 3476 mSampleRate, mChannelMask, mChannelCount, mFormat, mFrameSize, mFrameCount, 3477 mNormalFrameCount); 3478 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 3479 3480 if (type == DUPLICATING) { 3481 // The Duplicating thread uses the AudioMixer and delivers data to OutputTracks 3482 // (downstream MixerThreads) in DuplicatingThread::threadLoop_write(). 3483 // Do not create or use mFastMixer, mOutputSink, mPipeSink, or mNormalSink. 3484 return; 3485 } 3486 // create an NBAIO sink for the HAL output stream, and negotiate 3487 mOutputSink = new AudioStreamOutSink(output->stream); 3488 size_t numCounterOffers = 0; 3489 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount, mFormat)}; 3490#if !LOG_NDEBUG 3491 ssize_t index = 3492#else 3493 (void) 3494#endif 3495 mOutputSink->negotiate(offers, 1, NULL, numCounterOffers); 3496 ALOG_ASSERT(index == 0); 3497 3498 // initialize fast mixer depending on configuration 3499 bool initFastMixer; 3500 switch (kUseFastMixer) { 3501 case FastMixer_Never: 3502 initFastMixer = false; 3503 break; 3504 case FastMixer_Always: 3505 initFastMixer = true; 3506 break; 3507 case FastMixer_Static: 3508 case FastMixer_Dynamic: 3509 initFastMixer = mFrameCount < mNormalFrameCount; 3510 break; 3511 } 3512 if (initFastMixer) { 3513 audio_format_t fastMixerFormat; 3514 if (mMixerBufferEnabled && mEffectBufferEnabled) { 3515 fastMixerFormat = AUDIO_FORMAT_PCM_FLOAT; 3516 } else { 3517 fastMixerFormat = AUDIO_FORMAT_PCM_16_BIT; 3518 } 3519 if (mFormat != fastMixerFormat) { 3520 // change our Sink format to accept our intermediate precision 3521 mFormat = fastMixerFormat; 3522 free(mSinkBuffer); 3523 mFrameSize = mChannelCount * audio_bytes_per_sample(mFormat); 3524 const size_t sinkBufferSize = mNormalFrameCount * mFrameSize; 3525 (void)posix_memalign(&mSinkBuffer, 32, sinkBufferSize); 3526 } 3527 3528 // create a MonoPipe to connect our submix to FastMixer 3529 NBAIO_Format format = mOutputSink->format(); 3530#ifdef TEE_SINK 3531 NBAIO_Format origformat = format; 3532#endif 3533 // adjust format to match that of the Fast Mixer 3534 ALOGV("format changed from %d to %d", format.mFormat, fastMixerFormat); 3535 format.mFormat = fastMixerFormat; 3536 format.mFrameSize = audio_bytes_per_sample(format.mFormat) * format.mChannelCount; 3537 3538 // This pipe depth compensates for scheduling latency of the normal mixer thread. 3539 // When it wakes up after a maximum latency, it runs a few cycles quickly before 3540 // finally blocking. Note the pipe implementation rounds up the request to a power of 2. 3541 MonoPipe *monoPipe = new MonoPipe(mNormalFrameCount * 4, format, true /*writeCanBlock*/); 3542 const NBAIO_Format offers[1] = {format}; 3543 size_t numCounterOffers = 0; 3544#if !LOG_NDEBUG 3545 ssize_t index = 3546#else 3547 (void) 3548#endif 3549 monoPipe->negotiate(offers, 1, NULL, numCounterOffers); 3550 ALOG_ASSERT(index == 0); 3551 monoPipe->setAvgFrames((mScreenState & 1) ? 3552 (monoPipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 3553 mPipeSink = monoPipe; 3554 3555#ifdef TEE_SINK 3556 if (mTeeSinkOutputEnabled) { 3557 // create a Pipe to archive a copy of FastMixer's output for dumpsys 3558 Pipe *teeSink = new Pipe(mTeeSinkOutputFrames, origformat); 3559 const NBAIO_Format offers2[1] = {origformat}; 3560 numCounterOffers = 0; 3561 index = teeSink->negotiate(offers2, 1, NULL, numCounterOffers); 3562 ALOG_ASSERT(index == 0); 3563 mTeeSink = teeSink; 3564 PipeReader *teeSource = new PipeReader(*teeSink); 3565 numCounterOffers = 0; 3566 index = teeSource->negotiate(offers2, 1, NULL, numCounterOffers); 3567 ALOG_ASSERT(index == 0); 3568 mTeeSource = teeSource; 3569 } 3570#endif 3571 3572 // create fast mixer and configure it initially with just one fast track for our submix 3573 mFastMixer = new FastMixer(); 3574 FastMixerStateQueue *sq = mFastMixer->sq(); 3575#ifdef STATE_QUEUE_DUMP 3576 sq->setObserverDump(&mStateQueueObserverDump); 3577 sq->setMutatorDump(&mStateQueueMutatorDump); 3578#endif 3579 FastMixerState *state = sq->begin(); 3580 FastTrack *fastTrack = &state->mFastTracks[0]; 3581 // wrap the source side of the MonoPipe to make it an AudioBufferProvider 3582 fastTrack->mBufferProvider = new SourceAudioBufferProvider(new MonoPipeReader(monoPipe)); 3583 fastTrack->mVolumeProvider = NULL; 3584 fastTrack->mChannelMask = mChannelMask; // mPipeSink channel mask for audio to FastMixer 3585 fastTrack->mFormat = mFormat; // mPipeSink format for audio to FastMixer 3586 fastTrack->mGeneration++; 3587 state->mFastTracksGen++; 3588 state->mTrackMask = 1; 3589 // fast mixer will use the HAL output sink 3590 state->mOutputSink = mOutputSink.get(); 3591 state->mOutputSinkGen++; 3592 state->mFrameCount = mFrameCount; 3593 state->mCommand = FastMixerState::COLD_IDLE; 3594 // already done in constructor initialization list 3595 //mFastMixerFutex = 0; 3596 state->mColdFutexAddr = &mFastMixerFutex; 3597 state->mColdGen++; 3598 state->mDumpState = &mFastMixerDumpState; 3599#ifdef TEE_SINK 3600 state->mTeeSink = mTeeSink.get(); 3601#endif 3602 mFastMixerNBLogWriter = audioFlinger->newWriter_l(kFastMixerLogSize, "FastMixer"); 3603 state->mNBLogWriter = mFastMixerNBLogWriter.get(); 3604 sq->end(); 3605 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3606 3607 // start the fast mixer 3608 mFastMixer->run("FastMixer", PRIORITY_URGENT_AUDIO); 3609 pid_t tid = mFastMixer->getTid(); 3610 sendPrioConfigEvent(getpid_cached, tid, kPriorityFastMixer); 3611 3612#ifdef AUDIO_WATCHDOG 3613 // create and start the watchdog 3614 mAudioWatchdog = new AudioWatchdog(); 3615 mAudioWatchdog->setDump(&mAudioWatchdogDump); 3616 mAudioWatchdog->run("AudioWatchdog", PRIORITY_URGENT_AUDIO); 3617 tid = mAudioWatchdog->getTid(); 3618 sendPrioConfigEvent(getpid_cached, tid, kPriorityFastMixer); 3619#endif 3620 3621 } 3622 3623 switch (kUseFastMixer) { 3624 case FastMixer_Never: 3625 case FastMixer_Dynamic: 3626 mNormalSink = mOutputSink; 3627 break; 3628 case FastMixer_Always: 3629 mNormalSink = mPipeSink; 3630 break; 3631 case FastMixer_Static: 3632 mNormalSink = initFastMixer ? mPipeSink : mOutputSink; 3633 break; 3634 } 3635} 3636 3637AudioFlinger::MixerThread::~MixerThread() 3638{ 3639 if (mFastMixer != 0) { 3640 FastMixerStateQueue *sq = mFastMixer->sq(); 3641 FastMixerState *state = sq->begin(); 3642 if (state->mCommand == FastMixerState::COLD_IDLE) { 3643 int32_t old = android_atomic_inc(&mFastMixerFutex); 3644 if (old == -1) { 3645 (void) syscall(__NR_futex, &mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 3646 } 3647 } 3648 state->mCommand = FastMixerState::EXIT; 3649 sq->end(); 3650 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3651 mFastMixer->join(); 3652 // Though the fast mixer thread has exited, it's state queue is still valid. 3653 // We'll use that extract the final state which contains one remaining fast track 3654 // corresponding to our sub-mix. 3655 state = sq->begin(); 3656 ALOG_ASSERT(state->mTrackMask == 1); 3657 FastTrack *fastTrack = &state->mFastTracks[0]; 3658 ALOG_ASSERT(fastTrack->mBufferProvider != NULL); 3659 delete fastTrack->mBufferProvider; 3660 sq->end(false /*didModify*/); 3661 mFastMixer.clear(); 3662#ifdef AUDIO_WATCHDOG 3663 if (mAudioWatchdog != 0) { 3664 mAudioWatchdog->requestExit(); 3665 mAudioWatchdog->requestExitAndWait(); 3666 mAudioWatchdog.clear(); 3667 } 3668#endif 3669 } 3670 mAudioFlinger->unregisterWriter(mFastMixerNBLogWriter); 3671 delete mAudioMixer; 3672} 3673 3674 3675uint32_t AudioFlinger::MixerThread::correctLatency_l(uint32_t latency) const 3676{ 3677 if (mFastMixer != 0) { 3678 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 3679 latency += (pipe->getAvgFrames() * 1000) / mSampleRate; 3680 } 3681 return latency; 3682} 3683 3684 3685void AudioFlinger::MixerThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) 3686{ 3687 PlaybackThread::threadLoop_removeTracks(tracksToRemove); 3688} 3689 3690ssize_t AudioFlinger::MixerThread::threadLoop_write() 3691{ 3692 // FIXME we should only do one push per cycle; confirm this is true 3693 // Start the fast mixer if it's not already running 3694 if (mFastMixer != 0) { 3695 FastMixerStateQueue *sq = mFastMixer->sq(); 3696 FastMixerState *state = sq->begin(); 3697 if (state->mCommand != FastMixerState::MIX_WRITE && 3698 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)) { 3699 if (state->mCommand == FastMixerState::COLD_IDLE) { 3700 3701 // FIXME workaround for first HAL write being CPU bound on some devices 3702 ATRACE_BEGIN("write"); 3703 mOutput->write((char *)mSinkBuffer, 0); 3704 ATRACE_END(); 3705 3706 int32_t old = android_atomic_inc(&mFastMixerFutex); 3707 if (old == -1) { 3708 (void) syscall(__NR_futex, &mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 3709 } 3710#ifdef AUDIO_WATCHDOG 3711 if (mAudioWatchdog != 0) { 3712 mAudioWatchdog->resume(); 3713 } 3714#endif 3715 } 3716 state->mCommand = FastMixerState::MIX_WRITE; 3717#ifdef FAST_THREAD_STATISTICS 3718 mFastMixerDumpState.increaseSamplingN(mAudioFlinger->isLowRamDevice() ? 3719 FastThreadDumpState::kSamplingNforLowRamDevice : FastThreadDumpState::kSamplingN); 3720#endif 3721 sq->end(); 3722 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3723 if (kUseFastMixer == FastMixer_Dynamic) { 3724 mNormalSink = mPipeSink; 3725 } 3726 } else { 3727 sq->end(false /*didModify*/); 3728 } 3729 } 3730 return PlaybackThread::threadLoop_write(); 3731} 3732 3733void AudioFlinger::MixerThread::threadLoop_standby() 3734{ 3735 // Idle the fast mixer if it's currently running 3736 if (mFastMixer != 0) { 3737 FastMixerStateQueue *sq = mFastMixer->sq(); 3738 FastMixerState *state = sq->begin(); 3739 if (!(state->mCommand & FastMixerState::IDLE)) { 3740 state->mCommand = FastMixerState::COLD_IDLE; 3741 state->mColdFutexAddr = &mFastMixerFutex; 3742 state->mColdGen++; 3743 mFastMixerFutex = 0; 3744 sq->end(); 3745 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 3746 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3747 if (kUseFastMixer == FastMixer_Dynamic) { 3748 mNormalSink = mOutputSink; 3749 } 3750#ifdef AUDIO_WATCHDOG 3751 if (mAudioWatchdog != 0) { 3752 mAudioWatchdog->pause(); 3753 } 3754#endif 3755 } else { 3756 sq->end(false /*didModify*/); 3757 } 3758 } 3759 PlaybackThread::threadLoop_standby(); 3760} 3761 3762bool AudioFlinger::PlaybackThread::waitingAsyncCallback_l() 3763{ 3764 return false; 3765} 3766 3767bool AudioFlinger::PlaybackThread::shouldStandby_l() 3768{ 3769 return !mStandby; 3770} 3771 3772bool AudioFlinger::PlaybackThread::waitingAsyncCallback() 3773{ 3774 Mutex::Autolock _l(mLock); 3775 return waitingAsyncCallback_l(); 3776} 3777 3778// shared by MIXER and DIRECT, overridden by DUPLICATING 3779void AudioFlinger::PlaybackThread::threadLoop_standby() 3780{ 3781 ALOGV("Audio hardware entering standby, mixer %p, suspend count %d", this, mSuspended); 3782 mOutput->standby(); 3783 if (mUseAsyncWrite != 0) { 3784 // discard any pending drain or write ack by incrementing sequence 3785 mWriteAckSequence = (mWriteAckSequence + 2) & ~1; 3786 mDrainSequence = (mDrainSequence + 2) & ~1; 3787 ALOG_ASSERT(mCallbackThread != 0); 3788 mCallbackThread->setWriteBlocked(mWriteAckSequence); 3789 mCallbackThread->setDraining(mDrainSequence); 3790 } 3791 mHwPaused = false; 3792} 3793 3794void AudioFlinger::PlaybackThread::onAddNewTrack_l() 3795{ 3796 ALOGV("signal playback thread"); 3797 broadcast_l(); 3798} 3799 3800void AudioFlinger::MixerThread::threadLoop_mix() 3801{ 3802 // mix buffers... 3803 mAudioMixer->process(); 3804 mCurrentWriteLength = mSinkBufferSize; 3805 // increase sleep time progressively when application underrun condition clears. 3806 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 3807 // that a steady state of alternating ready/not ready conditions keeps the sleep time 3808 // such that we would underrun the audio HAL. 3809 if ((mSleepTimeUs == 0) && (sleepTimeShift > 0)) { 3810 sleepTimeShift--; 3811 } 3812 mSleepTimeUs = 0; 3813 mStandbyTimeNs = systemTime() + mStandbyDelayNs; 3814 //TODO: delay standby when effects have a tail 3815 3816} 3817 3818void AudioFlinger::MixerThread::threadLoop_sleepTime() 3819{ 3820 // If no tracks are ready, sleep once for the duration of an output 3821 // buffer size, then write 0s to the output 3822 if (mSleepTimeUs == 0) { 3823 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3824 mSleepTimeUs = mActiveSleepTimeUs >> sleepTimeShift; 3825 if (mSleepTimeUs < kMinThreadSleepTimeUs) { 3826 mSleepTimeUs = kMinThreadSleepTimeUs; 3827 } 3828 // reduce sleep time in case of consecutive application underruns to avoid 3829 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 3830 // duration we would end up writing less data than needed by the audio HAL if 3831 // the condition persists. 3832 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 3833 sleepTimeShift++; 3834 } 3835 } else { 3836 mSleepTimeUs = mIdleSleepTimeUs; 3837 } 3838 } else if (mBytesWritten != 0 || (mMixerStatus == MIXER_TRACKS_ENABLED)) { 3839 // clear out mMixerBuffer or mSinkBuffer, to ensure buffers are cleared 3840 // before effects processing or output. 3841 if (mMixerBufferValid) { 3842 memset(mMixerBuffer, 0, mMixerBufferSize); 3843 } else { 3844 memset(mSinkBuffer, 0, mSinkBufferSize); 3845 } 3846 mSleepTimeUs = 0; 3847 ALOGV_IF(mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED), 3848 "anticipated start"); 3849 } 3850 // TODO add standby time extension fct of effect tail 3851} 3852 3853// prepareTracks_l() must be called with ThreadBase::mLock held 3854AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 3855 Vector< sp<Track> > *tracksToRemove) 3856{ 3857 3858 mixer_state mixerStatus = MIXER_IDLE; 3859 // find out which tracks need to be processed 3860 size_t count = mActiveTracks.size(); 3861 size_t mixedTracks = 0; 3862 size_t tracksWithEffect = 0; 3863 // counts only _active_ fast tracks 3864 size_t fastTracks = 0; 3865 uint32_t resetMask = 0; // bit mask of fast tracks that need to be reset 3866 3867 float masterVolume = mMasterVolume; 3868 bool masterMute = mMasterMute; 3869 3870 if (masterMute) { 3871 masterVolume = 0; 3872 } 3873 // Delegate master volume control to effect in output mix effect chain if needed 3874 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 3875 if (chain != 0) { 3876 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 3877 chain->setVolume_l(&v, &v); 3878 masterVolume = (float)((v + (1 << 23)) >> 24); 3879 chain.clear(); 3880 } 3881 3882 // prepare a new state to push 3883 FastMixerStateQueue *sq = NULL; 3884 FastMixerState *state = NULL; 3885 bool didModify = false; 3886 FastMixerStateQueue::block_t block = FastMixerStateQueue::BLOCK_UNTIL_PUSHED; 3887 if (mFastMixer != 0) { 3888 sq = mFastMixer->sq(); 3889 state = sq->begin(); 3890 } 3891 3892 mMixerBufferValid = false; // mMixerBuffer has no valid data until appropriate tracks found. 3893 mEffectBufferValid = false; // mEffectBuffer has no valid data until tracks found. 3894 3895 for (size_t i=0 ; i<count ; i++) { 3896 const sp<Track> t = mActiveTracks[i].promote(); 3897 if (t == 0) { 3898 continue; 3899 } 3900 3901 // this const just means the local variable doesn't change 3902 Track* const track = t.get(); 3903 3904 // process fast tracks 3905 if (track->isFastTrack()) { 3906 3907 // It's theoretically possible (though unlikely) for a fast track to be created 3908 // and then removed within the same normal mix cycle. This is not a problem, as 3909 // the track never becomes active so it's fast mixer slot is never touched. 3910 // The converse, of removing an (active) track and then creating a new track 3911 // at the identical fast mixer slot within the same normal mix cycle, 3912 // is impossible because the slot isn't marked available until the end of each cycle. 3913 int j = track->mFastIndex; 3914 ALOG_ASSERT(0 < j && j < (int)FastMixerState::kMaxFastTracks); 3915 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << j))); 3916 FastTrack *fastTrack = &state->mFastTracks[j]; 3917 3918 // Determine whether the track is currently in underrun condition, 3919 // and whether it had a recent underrun. 3920 FastTrackDump *ftDump = &mFastMixerDumpState.mTracks[j]; 3921 FastTrackUnderruns underruns = ftDump->mUnderruns; 3922 uint32_t recentFull = (underruns.mBitFields.mFull - 3923 track->mObservedUnderruns.mBitFields.mFull) & UNDERRUN_MASK; 3924 uint32_t recentPartial = (underruns.mBitFields.mPartial - 3925 track->mObservedUnderruns.mBitFields.mPartial) & UNDERRUN_MASK; 3926 uint32_t recentEmpty = (underruns.mBitFields.mEmpty - 3927 track->mObservedUnderruns.mBitFields.mEmpty) & UNDERRUN_MASK; 3928 uint32_t recentUnderruns = recentPartial + recentEmpty; 3929 track->mObservedUnderruns = underruns; 3930 // don't count underruns that occur while stopping or pausing 3931 // or stopped which can occur when flush() is called while active 3932 if (!(track->isStopping() || track->isPausing() || track->isStopped()) && 3933 recentUnderruns > 0) { 3934 // FIXME fast mixer will pull & mix partial buffers, but we count as a full underrun 3935 track->mAudioTrackServerProxy->tallyUnderrunFrames(recentUnderruns * mFrameCount); 3936 } else { 3937 track->mAudioTrackServerProxy->tallyUnderrunFrames(0); 3938 } 3939 3940 // This is similar to the state machine for normal tracks, 3941 // with a few modifications for fast tracks. 3942 bool isActive = true; 3943 switch (track->mState) { 3944 case TrackBase::STOPPING_1: 3945 // track stays active in STOPPING_1 state until first underrun 3946 if (recentUnderruns > 0 || track->isTerminated()) { 3947 track->mState = TrackBase::STOPPING_2; 3948 } 3949 break; 3950 case TrackBase::PAUSING: 3951 // ramp down is not yet implemented 3952 track->setPaused(); 3953 break; 3954 case TrackBase::RESUMING: 3955 // ramp up is not yet implemented 3956 track->mState = TrackBase::ACTIVE; 3957 break; 3958 case TrackBase::ACTIVE: 3959 if (recentFull > 0 || recentPartial > 0) { 3960 // track has provided at least some frames recently: reset retry count 3961 track->mRetryCount = kMaxTrackRetries; 3962 } 3963 if (recentUnderruns == 0) { 3964 // no recent underruns: stay active 3965 break; 3966 } 3967 // there has recently been an underrun of some kind 3968 if (track->sharedBuffer() == 0) { 3969 // were any of the recent underruns "empty" (no frames available)? 3970 if (recentEmpty == 0) { 3971 // no, then ignore the partial underruns as they are allowed indefinitely 3972 break; 3973 } 3974 // there has recently been an "empty" underrun: decrement the retry counter 3975 if (--(track->mRetryCount) > 0) { 3976 break; 3977 } 3978 // indicate to client process that the track was disabled because of underrun; 3979 // it will then automatically call start() when data is available 3980 track->disable(); 3981 // remove from active list, but state remains ACTIVE [confusing but true] 3982 isActive = false; 3983 break; 3984 } 3985 // fall through 3986 case TrackBase::STOPPING_2: 3987 case TrackBase::PAUSED: 3988 case TrackBase::STOPPED: 3989 case TrackBase::FLUSHED: // flush() while active 3990 // Check for presentation complete if track is inactive 3991 // We have consumed all the buffers of this track. 3992 // This would be incomplete if we auto-paused on underrun 3993 { 3994 size_t audioHALFrames = 3995 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 3996 int64_t framesWritten = mBytesWritten / mFrameSize; 3997 if (!(mStandby || track->presentationComplete(framesWritten, audioHALFrames))) { 3998 // track stays in active list until presentation is complete 3999 break; 4000 } 4001 } 4002 if (track->isStopping_2()) { 4003 track->mState = TrackBase::STOPPED; 4004 } 4005 if (track->isStopped()) { 4006 // Can't reset directly, as fast mixer is still polling this track 4007 // track->reset(); 4008 // So instead mark this track as needing to be reset after push with ack 4009 resetMask |= 1 << i; 4010 } 4011 isActive = false; 4012 break; 4013 case TrackBase::IDLE: 4014 default: 4015 LOG_ALWAYS_FATAL("unexpected track state %d", track->mState); 4016 } 4017 4018 if (isActive) { 4019 // was it previously inactive? 4020 if (!(state->mTrackMask & (1 << j))) { 4021 ExtendedAudioBufferProvider *eabp = track; 4022 VolumeProvider *vp = track; 4023 fastTrack->mBufferProvider = eabp; 4024 fastTrack->mVolumeProvider = vp; 4025 fastTrack->mChannelMask = track->mChannelMask; 4026 fastTrack->mFormat = track->mFormat; 4027 fastTrack->mGeneration++; 4028 state->mTrackMask |= 1 << j; 4029 didModify = true; 4030 // no acknowledgement required for newly active tracks 4031 } 4032 // cache the combined master volume and stream type volume for fast mixer; this 4033 // lacks any synchronization or barrier so VolumeProvider may read a stale value 4034 track->mCachedVolume = masterVolume * mStreamTypes[track->streamType()].volume; 4035 ++fastTracks; 4036 } else { 4037 // was it previously active? 4038 if (state->mTrackMask & (1 << j)) { 4039 fastTrack->mBufferProvider = NULL; 4040 fastTrack->mGeneration++; 4041 state->mTrackMask &= ~(1 << j); 4042 didModify = true; 4043 // If any fast tracks were removed, we must wait for acknowledgement 4044 // because we're about to decrement the last sp<> on those tracks. 4045 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 4046 } else { 4047 LOG_ALWAYS_FATAL("fast track %d should have been active; " 4048 "mState=%d, mTrackMask=%#x, recentUnderruns=%u, isShared=%d", 4049 j, track->mState, state->mTrackMask, recentUnderruns, 4050 track->sharedBuffer() != 0); 4051 } 4052 tracksToRemove->add(track); 4053 // Avoids a misleading display in dumpsys 4054 track->mObservedUnderruns.mBitFields.mMostRecent = UNDERRUN_FULL; 4055 } 4056 continue; 4057 } 4058 4059 { // local variable scope to avoid goto warning 4060 4061 audio_track_cblk_t* cblk = track->cblk(); 4062 4063 // The first time a track is added we wait 4064 // for all its buffers to be filled before processing it 4065 int name = track->name(); 4066 // make sure that we have enough frames to mix one full buffer. 4067 // enforce this condition only once to enable draining the buffer in case the client 4068 // app does not call stop() and relies on underrun to stop: 4069 // hence the test on (mMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 4070 // during last round 4071 size_t desiredFrames; 4072 const uint32_t sampleRate = track->mAudioTrackServerProxy->getSampleRate(); 4073 AudioPlaybackRate playbackRate = track->mAudioTrackServerProxy->getPlaybackRate(); 4074 4075 desiredFrames = sourceFramesNeededWithTimestretch( 4076 sampleRate, mNormalFrameCount, mSampleRate, playbackRate.mSpeed); 4077 // TODO: ONLY USED FOR LEGACY RESAMPLERS, remove when they are removed. 4078 // add frames already consumed but not yet released by the resampler 4079 // because mAudioTrackServerProxy->framesReady() will include these frames 4080 desiredFrames += mAudioMixer->getUnreleasedFrames(track->name()); 4081 4082 uint32_t minFrames = 1; 4083 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() && 4084 (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) { 4085 minFrames = desiredFrames; 4086 } 4087 4088 size_t framesReady = track->framesReady(); 4089 if (ATRACE_ENABLED()) { 4090 // I wish we had formatted trace names 4091 char traceName[16]; 4092 strcpy(traceName, "nRdy"); 4093 int name = track->name(); 4094 if (AudioMixer::TRACK0 <= name && 4095 name < (int) (AudioMixer::TRACK0 + AudioMixer::MAX_NUM_TRACKS)) { 4096 name -= AudioMixer::TRACK0; 4097 traceName[4] = (name / 10) + '0'; 4098 traceName[5] = (name % 10) + '0'; 4099 } else { 4100 traceName[4] = '?'; 4101 traceName[5] = '?'; 4102 } 4103 traceName[6] = '\0'; 4104 ATRACE_INT(traceName, framesReady); 4105 } 4106 if ((framesReady >= minFrames) && track->isReady() && 4107 !track->isPaused() && !track->isTerminated()) 4108 { 4109 ALOGVV("track %d s=%08x [OK] on thread %p", name, cblk->mServer, this); 4110 4111 mixedTracks++; 4112 4113 // track->mainBuffer() != mSinkBuffer or mMixerBuffer means 4114 // there is an effect chain connected to the track 4115 chain.clear(); 4116 if (track->mainBuffer() != mSinkBuffer && 4117 track->mainBuffer() != mMixerBuffer) { 4118 if (mEffectBufferEnabled) { 4119 mEffectBufferValid = true; // Later can set directly. 4120 } 4121 chain = getEffectChain_l(track->sessionId()); 4122 // Delegate volume control to effect in track effect chain if needed 4123 if (chain != 0) { 4124 tracksWithEffect++; 4125 } else { 4126 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on " 4127 "session %d", 4128 name, track->sessionId()); 4129 } 4130 } 4131 4132 4133 int param = AudioMixer::VOLUME; 4134 if (track->mFillingUpStatus == Track::FS_FILLED) { 4135 // no ramp for the first volume setting 4136 track->mFillingUpStatus = Track::FS_ACTIVE; 4137 if (track->mState == TrackBase::RESUMING) { 4138 track->mState = TrackBase::ACTIVE; 4139 param = AudioMixer::RAMP_VOLUME; 4140 } 4141 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 4142 // FIXME should not make a decision based on mServer 4143 } else if (cblk->mServer != 0) { 4144 // If the track is stopped before the first frame was mixed, 4145 // do not apply ramp 4146 param = AudioMixer::RAMP_VOLUME; 4147 } 4148 4149 // compute volume for this track 4150 uint32_t vl, vr; // in U8.24 integer format 4151 float vlf, vrf, vaf; // in [0.0, 1.0] float format 4152 if (track->isPausing() || mStreamTypes[track->streamType()].mute) { 4153 vl = vr = 0; 4154 vlf = vrf = vaf = 0.; 4155 if (track->isPausing()) { 4156 track->setPaused(); 4157 } 4158 } else { 4159 4160 // read original volumes with volume control 4161 float typeVolume = mStreamTypes[track->streamType()].volume; 4162 float v = masterVolume * typeVolume; 4163 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 4164 gain_minifloat_packed_t vlr = proxy->getVolumeLR(); 4165 vlf = float_from_gain(gain_minifloat_unpack_left(vlr)); 4166 vrf = float_from_gain(gain_minifloat_unpack_right(vlr)); 4167 // track volumes come from shared memory, so can't be trusted and must be clamped 4168 if (vlf > GAIN_FLOAT_UNITY) { 4169 ALOGV("Track left volume out of range: %.3g", vlf); 4170 vlf = GAIN_FLOAT_UNITY; 4171 } 4172 if (vrf > GAIN_FLOAT_UNITY) { 4173 ALOGV("Track right volume out of range: %.3g", vrf); 4174 vrf = GAIN_FLOAT_UNITY; 4175 } 4176 // now apply the master volume and stream type volume 4177 vlf *= v; 4178 vrf *= v; 4179 // assuming master volume and stream type volume each go up to 1.0, 4180 // then derive vl and vr as U8.24 versions for the effect chain 4181 const float scaleto8_24 = MAX_GAIN_INT * MAX_GAIN_INT; 4182 vl = (uint32_t) (scaleto8_24 * vlf); 4183 vr = (uint32_t) (scaleto8_24 * vrf); 4184 // vl and vr are now in U8.24 format 4185 uint16_t sendLevel = proxy->getSendLevel_U4_12(); 4186 // send level comes from shared memory and so may be corrupt 4187 if (sendLevel > MAX_GAIN_INT) { 4188 ALOGV("Track send level out of range: %04X", sendLevel); 4189 sendLevel = MAX_GAIN_INT; 4190 } 4191 // vaf is represented as [0.0, 1.0] float by rescaling sendLevel 4192 vaf = v * sendLevel * (1. / MAX_GAIN_INT); 4193 } 4194 4195 // Delegate volume control to effect in track effect chain if needed 4196 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 4197 // Do not ramp volume if volume is controlled by effect 4198 param = AudioMixer::VOLUME; 4199 // Update remaining floating point volume levels 4200 vlf = (float)vl / (1 << 24); 4201 vrf = (float)vr / (1 << 24); 4202 track->mHasVolumeController = true; 4203 } else { 4204 // force no volume ramp when volume controller was just disabled or removed 4205 // from effect chain to avoid volume spike 4206 if (track->mHasVolumeController) { 4207 param = AudioMixer::VOLUME; 4208 } 4209 track->mHasVolumeController = false; 4210 } 4211 4212 // XXX: these things DON'T need to be done each time 4213 mAudioMixer->setBufferProvider(name, track); 4214 mAudioMixer->enable(name); 4215 4216 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, &vlf); 4217 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, &vrf); 4218 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, &vaf); 4219 mAudioMixer->setParameter( 4220 name, 4221 AudioMixer::TRACK, 4222 AudioMixer::FORMAT, (void *)track->format()); 4223 mAudioMixer->setParameter( 4224 name, 4225 AudioMixer::TRACK, 4226 AudioMixer::CHANNEL_MASK, (void *)(uintptr_t)track->channelMask()); 4227 mAudioMixer->setParameter( 4228 name, 4229 AudioMixer::TRACK, 4230 AudioMixer::MIXER_CHANNEL_MASK, (void *)(uintptr_t)mChannelMask); 4231 // limit track sample rate to 2 x output sample rate, which changes at re-configuration 4232 uint32_t maxSampleRate = mSampleRate * AUDIO_RESAMPLER_DOWN_RATIO_MAX; 4233 uint32_t reqSampleRate = track->mAudioTrackServerProxy->getSampleRate(); 4234 if (reqSampleRate == 0) { 4235 reqSampleRate = mSampleRate; 4236 } else if (reqSampleRate > maxSampleRate) { 4237 reqSampleRate = maxSampleRate; 4238 } 4239 mAudioMixer->setParameter( 4240 name, 4241 AudioMixer::RESAMPLE, 4242 AudioMixer::SAMPLE_RATE, 4243 (void *)(uintptr_t)reqSampleRate); 4244 4245 AudioPlaybackRate playbackRate = track->mAudioTrackServerProxy->getPlaybackRate(); 4246 mAudioMixer->setParameter( 4247 name, 4248 AudioMixer::TIMESTRETCH, 4249 AudioMixer::PLAYBACK_RATE, 4250 &playbackRate); 4251 4252 /* 4253 * Select the appropriate output buffer for the track. 4254 * 4255 * Tracks with effects go into their own effects chain buffer 4256 * and from there into either mEffectBuffer or mSinkBuffer. 4257 * 4258 * Other tracks can use mMixerBuffer for higher precision 4259 * channel accumulation. If this buffer is enabled 4260 * (mMixerBufferEnabled true), then selected tracks will accumulate 4261 * into it. 4262 * 4263 */ 4264 if (mMixerBufferEnabled 4265 && (track->mainBuffer() == mSinkBuffer 4266 || track->mainBuffer() == mMixerBuffer)) { 4267 mAudioMixer->setParameter( 4268 name, 4269 AudioMixer::TRACK, 4270 AudioMixer::MIXER_FORMAT, (void *)mMixerBufferFormat); 4271 mAudioMixer->setParameter( 4272 name, 4273 AudioMixer::TRACK, 4274 AudioMixer::MAIN_BUFFER, (void *)mMixerBuffer); 4275 // TODO: override track->mainBuffer()? 4276 mMixerBufferValid = true; 4277 } else { 4278 mAudioMixer->setParameter( 4279 name, 4280 AudioMixer::TRACK, 4281 AudioMixer::MIXER_FORMAT, (void *)AUDIO_FORMAT_PCM_16_BIT); 4282 mAudioMixer->setParameter( 4283 name, 4284 AudioMixer::TRACK, 4285 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 4286 } 4287 mAudioMixer->setParameter( 4288 name, 4289 AudioMixer::TRACK, 4290 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 4291 4292 // reset retry count 4293 track->mRetryCount = kMaxTrackRetries; 4294 4295 // If one track is ready, set the mixer ready if: 4296 // - the mixer was not ready during previous round OR 4297 // - no other track is not ready 4298 if (mMixerStatusIgnoringFastTracks != MIXER_TRACKS_READY || 4299 mixerStatus != MIXER_TRACKS_ENABLED) { 4300 mixerStatus = MIXER_TRACKS_READY; 4301 } 4302 } else { 4303 if (framesReady < desiredFrames && !track->isStopped() && !track->isPaused()) { 4304 ALOGV("track(%p) underrun, framesReady(%zu) < framesDesired(%zd)", 4305 track, framesReady, desiredFrames); 4306 track->mAudioTrackServerProxy->tallyUnderrunFrames(desiredFrames); 4307 } else { 4308 track->mAudioTrackServerProxy->tallyUnderrunFrames(0); 4309 } 4310 4311 // clear effect chain input buffer if an active track underruns to avoid sending 4312 // previous audio buffer again to effects 4313 chain = getEffectChain_l(track->sessionId()); 4314 if (chain != 0) { 4315 chain->clearInputBuffer(); 4316 } 4317 4318 ALOGVV("track %d s=%08x [NOT READY] on thread %p", name, cblk->mServer, this); 4319 if ((track->sharedBuffer() != 0) || track->isTerminated() || 4320 track->isStopped() || track->isPaused()) { 4321 // We have consumed all the buffers of this track. 4322 // Remove it from the list of active tracks. 4323 // TODO: use actual buffer filling status instead of latency when available from 4324 // audio HAL 4325 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 4326 int64_t framesWritten = mBytesWritten / mFrameSize; 4327 if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) { 4328 if (track->isStopped()) { 4329 track->reset(); 4330 } 4331 tracksToRemove->add(track); 4332 } 4333 } else { 4334 // No buffers for this track. Give it a few chances to 4335 // fill a buffer, then remove it from active list. 4336 if (--(track->mRetryCount) <= 0) { 4337 ALOGI("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 4338 tracksToRemove->add(track); 4339 // indicate to client process that the track was disabled because of underrun; 4340 // it will then automatically call start() when data is available 4341 track->disable(); 4342 // If one track is not ready, mark the mixer also not ready if: 4343 // - the mixer was ready during previous round OR 4344 // - no other track is ready 4345 } else if (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY || 4346 mixerStatus != MIXER_TRACKS_READY) { 4347 mixerStatus = MIXER_TRACKS_ENABLED; 4348 } 4349 } 4350 mAudioMixer->disable(name); 4351 } 4352 4353 } // local variable scope to avoid goto warning 4354 4355 } 4356 4357 // Push the new FastMixer state if necessary 4358 bool pauseAudioWatchdog = false; 4359 if (didModify) { 4360 state->mFastTracksGen++; 4361 // if the fast mixer was active, but now there are no fast tracks, then put it in cold idle 4362 if (kUseFastMixer == FastMixer_Dynamic && 4363 state->mCommand == FastMixerState::MIX_WRITE && state->mTrackMask <= 1) { 4364 state->mCommand = FastMixerState::COLD_IDLE; 4365 state->mColdFutexAddr = &mFastMixerFutex; 4366 state->mColdGen++; 4367 mFastMixerFutex = 0; 4368 if (kUseFastMixer == FastMixer_Dynamic) { 4369 mNormalSink = mOutputSink; 4370 } 4371 // If we go into cold idle, need to wait for acknowledgement 4372 // so that fast mixer stops doing I/O. 4373 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 4374 pauseAudioWatchdog = true; 4375 } 4376 } 4377 if (sq != NULL) { 4378 sq->end(didModify); 4379 sq->push(block); 4380 } 4381#ifdef AUDIO_WATCHDOG 4382 if (pauseAudioWatchdog && mAudioWatchdog != 0) { 4383 mAudioWatchdog->pause(); 4384 } 4385#endif 4386 4387 // Now perform the deferred reset on fast tracks that have stopped 4388 while (resetMask != 0) { 4389 size_t i = __builtin_ctz(resetMask); 4390 ALOG_ASSERT(i < count); 4391 resetMask &= ~(1 << i); 4392 sp<Track> t = mActiveTracks[i].promote(); 4393 if (t == 0) { 4394 continue; 4395 } 4396 Track* track = t.get(); 4397 ALOG_ASSERT(track->isFastTrack() && track->isStopped()); 4398 track->reset(); 4399 } 4400 4401 // remove all the tracks that need to be... 4402 removeTracks_l(*tracksToRemove); 4403 4404 if (getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX) != 0) { 4405 mEffectBufferValid = true; 4406 } 4407 4408 if (mEffectBufferValid) { 4409 // as long as there are effects we should clear the effects buffer, to avoid 4410 // passing a non-clean buffer to the effect chain 4411 memset(mEffectBuffer, 0, mEffectBufferSize); 4412 } 4413 // sink or mix buffer must be cleared if all tracks are connected to an 4414 // effect chain as in this case the mixer will not write to the sink or mix buffer 4415 // and track effects will accumulate into it 4416 if ((mBytesRemaining == 0) && ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 4417 (mixedTracks == 0 && fastTracks > 0))) { 4418 // FIXME as a performance optimization, should remember previous zero status 4419 if (mMixerBufferValid) { 4420 memset(mMixerBuffer, 0, mMixerBufferSize); 4421 // TODO: In testing, mSinkBuffer below need not be cleared because 4422 // the PlaybackThread::threadLoop() copies mMixerBuffer into mSinkBuffer 4423 // after mixing. 4424 // 4425 // To enforce this guarantee: 4426 // ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 4427 // (mixedTracks == 0 && fastTracks > 0)) 4428 // must imply MIXER_TRACKS_READY. 4429 // Later, we may clear buffers regardless, and skip much of this logic. 4430 } 4431 // FIXME as a performance optimization, should remember previous zero status 4432 memset(mSinkBuffer, 0, mNormalFrameCount * mFrameSize); 4433 } 4434 4435 // if any fast tracks, then status is ready 4436 mMixerStatusIgnoringFastTracks = mixerStatus; 4437 if (fastTracks > 0) { 4438 mixerStatus = MIXER_TRACKS_READY; 4439 } 4440 return mixerStatus; 4441} 4442 4443// getTrackName_l() must be called with ThreadBase::mLock held 4444int AudioFlinger::MixerThread::getTrackName_l(audio_channel_mask_t channelMask, 4445 audio_format_t format, audio_session_t sessionId) 4446{ 4447 return mAudioMixer->getTrackName(channelMask, format, sessionId); 4448} 4449 4450// deleteTrackName_l() must be called with ThreadBase::mLock held 4451void AudioFlinger::MixerThread::deleteTrackName_l(int name) 4452{ 4453 ALOGV("remove track (%d) and delete from mixer", name); 4454 mAudioMixer->deleteTrackName(name); 4455} 4456 4457// checkForNewParameter_l() must be called with ThreadBase::mLock held 4458bool AudioFlinger::MixerThread::checkForNewParameter_l(const String8& keyValuePair, 4459 status_t& status) 4460{ 4461 bool reconfig = false; 4462 bool a2dpDeviceChanged = false; 4463 4464 status = NO_ERROR; 4465 4466 // if !&IDLE, holds the FastMixer state to restore after new parameters processed 4467 FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE; 4468 if (mFastMixer != 0) { 4469 FastMixerStateQueue *sq = mFastMixer->sq(); 4470 FastMixerState *state = sq->begin(); 4471 if (!(state->mCommand & FastMixerState::IDLE)) { 4472 previousCommand = state->mCommand; 4473 state->mCommand = FastMixerState::HOT_IDLE; 4474 sq->end(); 4475 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 4476 } else { 4477 sq->end(false /*didModify*/); 4478 } 4479 } 4480 4481 AudioParameter param = AudioParameter(keyValuePair); 4482 int value; 4483 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 4484 reconfig = true; 4485 } 4486 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 4487 if (!isValidPcmSinkFormat((audio_format_t) value)) { 4488 status = BAD_VALUE; 4489 } else { 4490 // no need to save value, since it's constant 4491 reconfig = true; 4492 } 4493 } 4494 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 4495 if (!isValidPcmSinkChannelMask((audio_channel_mask_t) value)) { 4496 status = BAD_VALUE; 4497 } else { 4498 // no need to save value, since it's constant 4499 reconfig = true; 4500 } 4501 } 4502 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 4503 // do not accept frame count changes if tracks are open as the track buffer 4504 // size depends on frame count and correct behavior would not be guaranteed 4505 // if frame count is changed after track creation 4506 if (!mTracks.isEmpty()) { 4507 status = INVALID_OPERATION; 4508 } else { 4509 reconfig = true; 4510 } 4511 } 4512 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 4513#ifdef ADD_BATTERY_DATA 4514 // when changing the audio output device, call addBatteryData to notify 4515 // the change 4516 if (mOutDevice != value) { 4517 uint32_t params = 0; 4518 // check whether speaker is on 4519 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 4520 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 4521 } 4522 4523 audio_devices_t deviceWithoutSpeaker 4524 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 4525 // check if any other device (except speaker) is on 4526 if (value & deviceWithoutSpeaker) { 4527 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 4528 } 4529 4530 if (params != 0) { 4531 addBatteryData(params); 4532 } 4533 } 4534#endif 4535 4536 // forward device change to effects that have requested to be 4537 // aware of attached audio device. 4538 if (value != AUDIO_DEVICE_NONE) { 4539 a2dpDeviceChanged = 4540 (mOutDevice & AUDIO_DEVICE_OUT_ALL_A2DP) != (value & AUDIO_DEVICE_OUT_ALL_A2DP); 4541 mOutDevice = value; 4542 for (size_t i = 0; i < mEffectChains.size(); i++) { 4543 mEffectChains[i]->setDevice_l(mOutDevice); 4544 } 4545 } 4546 } 4547 4548 if (status == NO_ERROR) { 4549 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 4550 keyValuePair.string()); 4551 if (!mStandby && status == INVALID_OPERATION) { 4552 mOutput->standby(); 4553 mStandby = true; 4554 mBytesWritten = 0; 4555 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 4556 keyValuePair.string()); 4557 } 4558 if (status == NO_ERROR && reconfig) { 4559 readOutputParameters_l(); 4560 delete mAudioMixer; 4561 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 4562 for (size_t i = 0; i < mTracks.size() ; i++) { 4563 int name = getTrackName_l(mTracks[i]->mChannelMask, 4564 mTracks[i]->mFormat, mTracks[i]->mSessionId); 4565 if (name < 0) { 4566 break; 4567 } 4568 mTracks[i]->mName = name; 4569 } 4570 sendIoConfigEvent_l(AUDIO_OUTPUT_CONFIG_CHANGED); 4571 } 4572 } 4573 4574 if (!(previousCommand & FastMixerState::IDLE)) { 4575 ALOG_ASSERT(mFastMixer != 0); 4576 FastMixerStateQueue *sq = mFastMixer->sq(); 4577 FastMixerState *state = sq->begin(); 4578 ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE); 4579 state->mCommand = previousCommand; 4580 sq->end(); 4581 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 4582 } 4583 4584 return reconfig || a2dpDeviceChanged; 4585} 4586 4587 4588void AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 4589{ 4590 PlaybackThread::dumpInternals(fd, args); 4591 dprintf(fd, " Thread throttle time (msecs): %u\n", mThreadThrottleTimeMs); 4592 dprintf(fd, " AudioMixer tracks: 0x%08x\n", mAudioMixer->trackNames()); 4593 dprintf(fd, " Master mono: %s\n", mMasterMono ? "on" : "off"); 4594 4595 // Make a non-atomic copy of fast mixer dump state so it won't change underneath us 4596 // while we are dumping it. It may be inconsistent, but it won't mutate! 4597 // This is a large object so we place it on the heap. 4598 // FIXME 25972958: Need an intelligent copy constructor that does not touch unused pages. 4599 const FastMixerDumpState *copy = new FastMixerDumpState(mFastMixerDumpState); 4600 copy->dump(fd); 4601 delete copy; 4602 4603#ifdef STATE_QUEUE_DUMP 4604 // Similar for state queue 4605 StateQueueObserverDump observerCopy = mStateQueueObserverDump; 4606 observerCopy.dump(fd); 4607 StateQueueMutatorDump mutatorCopy = mStateQueueMutatorDump; 4608 mutatorCopy.dump(fd); 4609#endif 4610 4611#ifdef TEE_SINK 4612 // Write the tee output to a .wav file 4613 dumpTee(fd, mTeeSource, mId); 4614#endif 4615 4616#ifdef AUDIO_WATCHDOG 4617 if (mAudioWatchdog != 0) { 4618 // Make a non-atomic copy of audio watchdog dump so it won't change underneath us 4619 AudioWatchdogDump wdCopy = mAudioWatchdogDump; 4620 wdCopy.dump(fd); 4621 } 4622#endif 4623} 4624 4625uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const 4626{ 4627 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000) / 2; 4628} 4629 4630uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() const 4631{ 4632 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000); 4633} 4634 4635void AudioFlinger::MixerThread::cacheParameters_l() 4636{ 4637 PlaybackThread::cacheParameters_l(); 4638 4639 // FIXME: Relaxed timing because of a certain device that can't meet latency 4640 // Should be reduced to 2x after the vendor fixes the driver issue 4641 // increase threshold again due to low power audio mode. The way this warning 4642 // threshold is calculated and its usefulness should be reconsidered anyway. 4643 maxPeriod = seconds(mNormalFrameCount) / mSampleRate * 15; 4644} 4645 4646// ---------------------------------------------------------------------------- 4647 4648AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 4649 AudioStreamOut* output, audio_io_handle_t id, audio_devices_t device, bool systemReady, 4650 uint32_t bitRate) 4651 : PlaybackThread(audioFlinger, output, id, device, DIRECT, systemReady, bitRate) 4652 // mLeftVolFloat, mRightVolFloat 4653{ 4654} 4655 4656AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 4657 AudioStreamOut* output, audio_io_handle_t id, uint32_t device, 4658 ThreadBase::type_t type, bool systemReady, uint32_t bitRate) 4659 : PlaybackThread(audioFlinger, output, id, device, type, systemReady, bitRate) 4660 // mLeftVolFloat, mRightVolFloat 4661{ 4662} 4663 4664AudioFlinger::DirectOutputThread::~DirectOutputThread() 4665{ 4666} 4667 4668void AudioFlinger::DirectOutputThread::processVolume_l(Track *track, bool lastTrack) 4669{ 4670 float left, right; 4671 4672 if (mMasterMute || mStreamTypes[track->streamType()].mute) { 4673 left = right = 0; 4674 } else { 4675 float typeVolume = mStreamTypes[track->streamType()].volume; 4676 float v = mMasterVolume * typeVolume; 4677 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 4678 gain_minifloat_packed_t vlr = proxy->getVolumeLR(); 4679 left = float_from_gain(gain_minifloat_unpack_left(vlr)); 4680 if (left > GAIN_FLOAT_UNITY) { 4681 left = GAIN_FLOAT_UNITY; 4682 } 4683 left *= v; 4684 right = float_from_gain(gain_minifloat_unpack_right(vlr)); 4685 if (right > GAIN_FLOAT_UNITY) { 4686 right = GAIN_FLOAT_UNITY; 4687 } 4688 right *= v; 4689 } 4690 4691 if (lastTrack) { 4692 if (left != mLeftVolFloat || right != mRightVolFloat) { 4693 mLeftVolFloat = left; 4694 mRightVolFloat = right; 4695 4696 // Convert volumes from float to 8.24 4697 uint32_t vl = (uint32_t)(left * (1 << 24)); 4698 uint32_t vr = (uint32_t)(right * (1 << 24)); 4699 4700 // Delegate volume control to effect in track effect chain if needed 4701 // only one effect chain can be present on DirectOutputThread, so if 4702 // there is one, the track is connected to it 4703 if (!mEffectChains.isEmpty()) { 4704 mEffectChains[0]->setVolume_l(&vl, &vr); 4705 left = (float)vl / (1 << 24); 4706 right = (float)vr / (1 << 24); 4707 } 4708 if (mOutput->stream->set_volume) { 4709 mOutput->stream->set_volume(mOutput->stream, left, right); 4710 } 4711 } 4712 } 4713} 4714 4715void AudioFlinger::DirectOutputThread::onAddNewTrack_l() 4716{ 4717 sp<Track> previousTrack = mPreviousTrack.promote(); 4718 sp<Track> latestTrack = mLatestActiveTrack.promote(); 4719 4720 if (previousTrack != 0 && latestTrack != 0) { 4721 if (mType == DIRECT) { 4722 if (previousTrack.get() != latestTrack.get()) { 4723 mFlushPending = true; 4724 } 4725 } else /* mType == OFFLOAD */ { 4726 if (previousTrack->sessionId() != latestTrack->sessionId()) { 4727 mFlushPending = true; 4728 } 4729 } 4730 } 4731 PlaybackThread::onAddNewTrack_l(); 4732} 4733 4734AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l( 4735 Vector< sp<Track> > *tracksToRemove 4736) 4737{ 4738 size_t count = mActiveTracks.size(); 4739 mixer_state mixerStatus = MIXER_IDLE; 4740 bool doHwPause = false; 4741 bool doHwResume = false; 4742 4743 // find out which tracks need to be processed 4744 for (size_t i = 0; i < count; i++) { 4745 sp<Track> t = mActiveTracks[i].promote(); 4746 // The track died recently 4747 if (t == 0) { 4748 continue; 4749 } 4750 4751 if (t->isInvalid()) { 4752 ALOGW("An invalidated track shouldn't be in active list"); 4753 tracksToRemove->add(t); 4754 continue; 4755 } 4756 4757 Track* const track = t.get(); 4758#ifdef VERY_VERY_VERBOSE_LOGGING 4759 audio_track_cblk_t* cblk = track->cblk(); 4760#endif 4761 // Only consider last track started for volume and mixer state control. 4762 // In theory an older track could underrun and restart after the new one starts 4763 // but as we only care about the transition phase between two tracks on a 4764 // direct output, it is not a problem to ignore the underrun case. 4765 sp<Track> l = mLatestActiveTrack.promote(); 4766 bool last = l.get() == track; 4767 4768 if (track->isPausing()) { 4769 track->setPaused(); 4770 if (mHwSupportsPause && last && !mHwPaused) { 4771 doHwPause = true; 4772 mHwPaused = true; 4773 } 4774 tracksToRemove->add(track); 4775 } else if (track->isFlushPending()) { 4776 track->flushAck(); 4777 if (last) { 4778 mFlushPending = true; 4779 } 4780 } else if (track->isResumePending()) { 4781 track->resumeAck(); 4782 if (last && mHwPaused) { 4783 doHwResume = true; 4784 mHwPaused = false; 4785 } 4786 } 4787 4788 // The first time a track is added we wait 4789 // for all its buffers to be filled before processing it. 4790 // Allow draining the buffer in case the client 4791 // app does not call stop() and relies on underrun to stop: 4792 // hence the test on (track->mRetryCount > 1). 4793 // If retryCount<=1 then track is about to underrun and be removed. 4794 // Do not use a high threshold for compressed audio. 4795 uint32_t minFrames; 4796 if ((track->sharedBuffer() == 0) && !track->isStopping_1() && !track->isPausing() 4797 && (track->mRetryCount > 1) && audio_has_proportional_frames(mFormat)) { 4798 minFrames = mNormalFrameCount; 4799 } else { 4800 minFrames = 1; 4801 } 4802 4803 if ((track->framesReady() >= minFrames) && track->isReady() && !track->isPaused() && 4804 !track->isStopping_2() && !track->isStopped()) 4805 { 4806 ALOGVV("track %d s=%08x [OK]", track->name(), cblk->mServer); 4807 4808 if (track->mFillingUpStatus == Track::FS_FILLED) { 4809 track->mFillingUpStatus = Track::FS_ACTIVE; 4810 // make sure processVolume_l() will apply new volume even if 0 4811 mLeftVolFloat = mRightVolFloat = -1.0; 4812 if (!mHwSupportsPause) { 4813 track->resumeAck(); 4814 } 4815 } 4816 4817 // compute volume for this track 4818 processVolume_l(track, last); 4819 if (last) { 4820 sp<Track> previousTrack = mPreviousTrack.promote(); 4821 if (previousTrack != 0) { 4822 if (track != previousTrack.get()) { 4823 // Flush any data still being written from last track 4824 mBytesRemaining = 0; 4825 // Invalidate previous track to force a seek when resuming. 4826 previousTrack->invalidate(); 4827 } 4828 } 4829 mPreviousTrack = track; 4830 4831 // reset retry count 4832 track->mRetryCount = kMaxTrackRetriesDirect; 4833 mActiveTrack = t; 4834 mixerStatus = MIXER_TRACKS_READY; 4835 if (mHwPaused) { 4836 doHwResume = true; 4837 mHwPaused = false; 4838 } 4839 } 4840 } else { 4841 // clear effect chain input buffer if the last active track started underruns 4842 // to avoid sending previous audio buffer again to effects 4843 if (!mEffectChains.isEmpty() && last) { 4844 mEffectChains[0]->clearInputBuffer(); 4845 } 4846 if (track->isStopping_1()) { 4847 track->mState = TrackBase::STOPPING_2; 4848 if (last && mHwPaused) { 4849 doHwResume = true; 4850 mHwPaused = false; 4851 } 4852 } 4853 if ((track->sharedBuffer() != 0) || track->isStopped() || 4854 track->isStopping_2() || track->isPaused()) { 4855 // We have consumed all the buffers of this track. 4856 // Remove it from the list of active tracks. 4857 size_t audioHALFrames; 4858 if (audio_has_proportional_frames(mFormat)) { 4859 audioHALFrames = (latency_l() * mSampleRate) / 1000; 4860 } else { 4861 audioHALFrames = 0; 4862 } 4863 4864 int64_t framesWritten = mBytesWritten / mFrameSize; 4865 if (mStandby || !last || 4866 track->presentationComplete(framesWritten, audioHALFrames)) { 4867 if (track->isStopping_2()) { 4868 track->mState = TrackBase::STOPPED; 4869 } 4870 if (track->isStopped()) { 4871 track->reset(); 4872 } 4873 tracksToRemove->add(track); 4874 } 4875 } else { 4876 // No buffers for this track. Give it a few chances to 4877 // fill a buffer, then remove it from active list. 4878 // Only consider last track started for mixer state control 4879 if (--(track->mRetryCount) <= 0) { 4880 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 4881 tracksToRemove->add(track); 4882 // indicate to client process that the track was disabled because of underrun; 4883 // it will then automatically call start() when data is available 4884 track->disable(); 4885 } else if (last) { 4886 ALOGW("pause because of UNDERRUN, framesReady = %zu," 4887 "minFrames = %u, mFormat = %#x", 4888 track->framesReady(), minFrames, mFormat); 4889 mixerStatus = MIXER_TRACKS_ENABLED; 4890 if (mHwSupportsPause && !mHwPaused && !mStandby) { 4891 doHwPause = true; 4892 mHwPaused = true; 4893 } 4894 } 4895 } 4896 } 4897 } 4898 4899 // if an active track did not command a flush, check for pending flush on stopped tracks 4900 if (!mFlushPending) { 4901 for (size_t i = 0; i < mTracks.size(); i++) { 4902 if (mTracks[i]->isFlushPending()) { 4903 mTracks[i]->flushAck(); 4904 mFlushPending = true; 4905 } 4906 } 4907 } 4908 4909 // make sure the pause/flush/resume sequence is executed in the right order. 4910 // If a flush is pending and a track is active but the HW is not paused, force a HW pause 4911 // before flush and then resume HW. This can happen in case of pause/flush/resume 4912 // if resume is received before pause is executed. 4913 if (mHwSupportsPause && !mStandby && 4914 (doHwPause || (mFlushPending && !mHwPaused && (count != 0)))) { 4915 mOutput->stream->pause(mOutput->stream); 4916 } 4917 if (mFlushPending) { 4918 flushHw_l(); 4919 } 4920 if (mHwSupportsPause && !mStandby && doHwResume) { 4921 mOutput->stream->resume(mOutput->stream); 4922 } 4923 // remove all the tracks that need to be... 4924 removeTracks_l(*tracksToRemove); 4925 4926 return mixerStatus; 4927} 4928 4929void AudioFlinger::DirectOutputThread::threadLoop_mix() 4930{ 4931 size_t frameCount = mFrameCount; 4932 int8_t *curBuf = (int8_t *)mSinkBuffer; 4933 // output audio to hardware 4934 while (frameCount) { 4935 AudioBufferProvider::Buffer buffer; 4936 buffer.frameCount = frameCount; 4937 status_t status = mActiveTrack->getNextBuffer(&buffer); 4938 if (status != NO_ERROR || buffer.raw == NULL) { 4939 // no need to pad with 0 for compressed audio 4940 if (audio_has_proportional_frames(mFormat)) { 4941 memset(curBuf, 0, frameCount * mFrameSize); 4942 } 4943 break; 4944 } 4945 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 4946 frameCount -= buffer.frameCount; 4947 curBuf += buffer.frameCount * mFrameSize; 4948 mActiveTrack->releaseBuffer(&buffer); 4949 } 4950 mCurrentWriteLength = curBuf - (int8_t *)mSinkBuffer; 4951 mSleepTimeUs = 0; 4952 mStandbyTimeNs = systemTime() + mStandbyDelayNs; 4953 mActiveTrack.clear(); 4954} 4955 4956void AudioFlinger::DirectOutputThread::threadLoop_sleepTime() 4957{ 4958 // do not write to HAL when paused 4959 if (mHwPaused || (usesHwAvSync() && mStandby)) { 4960 mSleepTimeUs = mIdleSleepTimeUs; 4961 return; 4962 } 4963 if (mSleepTimeUs == 0) { 4964 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4965 // For compressed offload, use faster sleep time when underruning until more than an 4966 // entire buffer was written to the audio HAL 4967 if (!audio_has_proportional_frames(mFormat) && 4968 (mType == OFFLOAD) && (mBytesWritten < mBufferSize)) { 4969 mSleepTimeUs = kDirectMinSleepTimeUs; 4970 } else { 4971 mSleepTimeUs = mActiveSleepTimeUs; 4972 } 4973 } else { 4974 mSleepTimeUs = mIdleSleepTimeUs; 4975 } 4976 } else if (mBytesWritten != 0 && audio_has_proportional_frames(mFormat)) { 4977 memset(mSinkBuffer, 0, mFrameCount * mFrameSize); 4978 mSleepTimeUs = 0; 4979 } 4980} 4981 4982void AudioFlinger::DirectOutputThread::threadLoop_exit() 4983{ 4984 { 4985 Mutex::Autolock _l(mLock); 4986 for (size_t i = 0; i < mTracks.size(); i++) { 4987 if (mTracks[i]->isFlushPending()) { 4988 mTracks[i]->flushAck(); 4989 mFlushPending = true; 4990 } 4991 } 4992 if (mFlushPending) { 4993 flushHw_l(); 4994 } 4995 } 4996 PlaybackThread::threadLoop_exit(); 4997} 4998 4999// must be called with thread mutex locked 5000bool AudioFlinger::DirectOutputThread::shouldStandby_l() 5001{ 5002 bool trackPaused = false; 5003 bool trackStopped = false; 5004 5005 // do not put the HAL in standby when paused. AwesomePlayer clear the offloaded AudioTrack 5006 // after a timeout and we will enter standby then. 5007 if (mTracks.size() > 0) { 5008 trackPaused = mTracks[mTracks.size() - 1]->isPaused(); 5009 trackStopped = mTracks[mTracks.size() - 1]->isStopped() || 5010 mTracks[mTracks.size() - 1]->mState == TrackBase::IDLE; 5011 } 5012 5013 return !mStandby && !(trackPaused || (mHwPaused && !trackStopped)); 5014} 5015 5016// getTrackName_l() must be called with ThreadBase::mLock held 5017int AudioFlinger::DirectOutputThread::getTrackName_l(audio_channel_mask_t channelMask __unused, 5018 audio_format_t format __unused, audio_session_t sessionId __unused) 5019{ 5020 return 0; 5021} 5022 5023// deleteTrackName_l() must be called with ThreadBase::mLock held 5024void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name __unused) 5025{ 5026} 5027 5028// checkForNewParameter_l() must be called with ThreadBase::mLock held 5029bool AudioFlinger::DirectOutputThread::checkForNewParameter_l(const String8& keyValuePair, 5030 status_t& status) 5031{ 5032 bool reconfig = false; 5033 bool a2dpDeviceChanged = false; 5034 5035 status = NO_ERROR; 5036 5037 AudioParameter param = AudioParameter(keyValuePair); 5038 int value; 5039 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 5040 // forward device change to effects that have requested to be 5041 // aware of attached audio device. 5042 if (value != AUDIO_DEVICE_NONE) { 5043 a2dpDeviceChanged = 5044 (mOutDevice & AUDIO_DEVICE_OUT_ALL_A2DP) != (value & AUDIO_DEVICE_OUT_ALL_A2DP); 5045 mOutDevice = value; 5046 for (size_t i = 0; i < mEffectChains.size(); i++) { 5047 mEffectChains[i]->setDevice_l(mOutDevice); 5048 } 5049 } 5050 } 5051 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 5052 // do not accept frame count changes if tracks are open as the track buffer 5053 // size depends on frame count and correct behavior would not be garantied 5054 // if frame count is changed after track creation 5055 if (!mTracks.isEmpty()) { 5056 status = INVALID_OPERATION; 5057 } else { 5058 reconfig = true; 5059 } 5060 } 5061 if (status == NO_ERROR) { 5062 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 5063 keyValuePair.string()); 5064 if (!mStandby && status == INVALID_OPERATION) { 5065 mOutput->standby(); 5066 mStandby = true; 5067 mBytesWritten = 0; 5068 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 5069 keyValuePair.string()); 5070 } 5071 if (status == NO_ERROR && reconfig) { 5072 readOutputParameters_l(); 5073 sendIoConfigEvent_l(AUDIO_OUTPUT_CONFIG_CHANGED); 5074 } 5075 } 5076 5077 return reconfig || a2dpDeviceChanged; 5078} 5079 5080uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() const 5081{ 5082 uint32_t time; 5083 if (audio_has_proportional_frames(mFormat)) { 5084 time = PlaybackThread::activeSleepTimeUs(); 5085 } else { 5086 time = kDirectMinSleepTimeUs; 5087 } 5088 return time; 5089} 5090 5091uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() const 5092{ 5093 uint32_t time; 5094 if (audio_has_proportional_frames(mFormat)) { 5095 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 5096 } else { 5097 time = kDirectMinSleepTimeUs; 5098 } 5099 return time; 5100} 5101 5102uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() const 5103{ 5104 uint32_t time; 5105 if (audio_has_proportional_frames(mFormat)) { 5106 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 5107 } else { 5108 time = kDirectMinSleepTimeUs; 5109 } 5110 return time; 5111} 5112 5113void AudioFlinger::DirectOutputThread::cacheParameters_l() 5114{ 5115 PlaybackThread::cacheParameters_l(); 5116 5117 // use shorter standby delay as on normal output to release 5118 // hardware resources as soon as possible 5119 // no delay on outputs with HW A/V sync 5120 if (usesHwAvSync()) { 5121 mStandbyDelayNs = 0; 5122 } else if ((mType == OFFLOAD) && !audio_has_proportional_frames(mFormat)) { 5123 mStandbyDelayNs = kOffloadStandbyDelayNs; 5124 } else { 5125 mStandbyDelayNs = microseconds(mActiveSleepTimeUs*2); 5126 } 5127} 5128 5129void AudioFlinger::DirectOutputThread::flushHw_l() 5130{ 5131 mOutput->flush(); 5132 mHwPaused = false; 5133 mFlushPending = false; 5134} 5135 5136// ---------------------------------------------------------------------------- 5137 5138AudioFlinger::AsyncCallbackThread::AsyncCallbackThread( 5139 const wp<AudioFlinger::PlaybackThread>& playbackThread) 5140 : Thread(false /*canCallJava*/), 5141 mPlaybackThread(playbackThread), 5142 mWriteAckSequence(0), 5143 mDrainSequence(0) 5144{ 5145} 5146 5147AudioFlinger::AsyncCallbackThread::~AsyncCallbackThread() 5148{ 5149} 5150 5151void AudioFlinger::AsyncCallbackThread::onFirstRef() 5152{ 5153 run("Offload Cbk", ANDROID_PRIORITY_URGENT_AUDIO); 5154} 5155 5156bool AudioFlinger::AsyncCallbackThread::threadLoop() 5157{ 5158 while (!exitPending()) { 5159 uint32_t writeAckSequence; 5160 uint32_t drainSequence; 5161 5162 { 5163 Mutex::Autolock _l(mLock); 5164 while (!((mWriteAckSequence & 1) || 5165 (mDrainSequence & 1) || 5166 exitPending())) { 5167 mWaitWorkCV.wait(mLock); 5168 } 5169 5170 if (exitPending()) { 5171 break; 5172 } 5173 ALOGV("AsyncCallbackThread mWriteAckSequence %d mDrainSequence %d", 5174 mWriteAckSequence, mDrainSequence); 5175 writeAckSequence = mWriteAckSequence; 5176 mWriteAckSequence &= ~1; 5177 drainSequence = mDrainSequence; 5178 mDrainSequence &= ~1; 5179 } 5180 { 5181 sp<AudioFlinger::PlaybackThread> playbackThread = mPlaybackThread.promote(); 5182 if (playbackThread != 0) { 5183 if (writeAckSequence & 1) { 5184 playbackThread->resetWriteBlocked(writeAckSequence >> 1); 5185 } 5186 if (drainSequence & 1) { 5187 playbackThread->resetDraining(drainSequence >> 1); 5188 } 5189 } 5190 } 5191 } 5192 return false; 5193} 5194 5195void AudioFlinger::AsyncCallbackThread::exit() 5196{ 5197 ALOGV("AsyncCallbackThread::exit"); 5198 Mutex::Autolock _l(mLock); 5199 requestExit(); 5200 mWaitWorkCV.broadcast(); 5201} 5202 5203void AudioFlinger::AsyncCallbackThread::setWriteBlocked(uint32_t sequence) 5204{ 5205 Mutex::Autolock _l(mLock); 5206 // bit 0 is cleared 5207 mWriteAckSequence = sequence << 1; 5208} 5209 5210void AudioFlinger::AsyncCallbackThread::resetWriteBlocked() 5211{ 5212 Mutex::Autolock _l(mLock); 5213 // ignore unexpected callbacks 5214 if (mWriteAckSequence & 2) { 5215 mWriteAckSequence |= 1; 5216 mWaitWorkCV.signal(); 5217 } 5218} 5219 5220void AudioFlinger::AsyncCallbackThread::setDraining(uint32_t sequence) 5221{ 5222 Mutex::Autolock _l(mLock); 5223 // bit 0 is cleared 5224 mDrainSequence = sequence << 1; 5225} 5226 5227void AudioFlinger::AsyncCallbackThread::resetDraining() 5228{ 5229 Mutex::Autolock _l(mLock); 5230 // ignore unexpected callbacks 5231 if (mDrainSequence & 2) { 5232 mDrainSequence |= 1; 5233 mWaitWorkCV.signal(); 5234 } 5235} 5236 5237 5238// ---------------------------------------------------------------------------- 5239AudioFlinger::OffloadThread::OffloadThread(const sp<AudioFlinger>& audioFlinger, 5240 AudioStreamOut* output, audio_io_handle_t id, uint32_t device, bool systemReady, 5241 uint32_t bitRate) 5242 : DirectOutputThread(audioFlinger, output, id, device, OFFLOAD, systemReady, bitRate), 5243 mPausedBytesRemaining(0) 5244{ 5245 //FIXME: mStandby should be set to true by ThreadBase constructor 5246 mStandby = true; 5247} 5248 5249void AudioFlinger::OffloadThread::threadLoop_exit() 5250{ 5251 if (mFlushPending || mHwPaused) { 5252 // If a flush is pending or track was paused, just discard buffered data 5253 flushHw_l(); 5254 } else { 5255 mMixerStatus = MIXER_DRAIN_ALL; 5256 threadLoop_drain(); 5257 } 5258 if (mUseAsyncWrite) { 5259 ALOG_ASSERT(mCallbackThread != 0); 5260 mCallbackThread->exit(); 5261 } 5262 PlaybackThread::threadLoop_exit(); 5263} 5264 5265AudioFlinger::PlaybackThread::mixer_state AudioFlinger::OffloadThread::prepareTracks_l( 5266 Vector< sp<Track> > *tracksToRemove 5267) 5268{ 5269 size_t count = mActiveTracks.size(); 5270 5271 mixer_state mixerStatus = MIXER_IDLE; 5272 bool doHwPause = false; 5273 bool doHwResume = false; 5274 5275 ALOGV("OffloadThread::prepareTracks_l active tracks %d", count); 5276 5277 // find out which tracks need to be processed 5278 for (size_t i = 0; i < count; i++) { 5279 sp<Track> t = mActiveTracks[i].promote(); 5280 // The track died recently 5281 if (t == 0) { 5282 continue; 5283 } 5284 Track* const track = t.get(); 5285#ifdef VERY_VERY_VERBOSE_LOGGING 5286 audio_track_cblk_t* cblk = track->cblk(); 5287#endif 5288 // Only consider last track started for volume and mixer state control. 5289 // In theory an older track could underrun and restart after the new one starts 5290 // but as we only care about the transition phase between two tracks on a 5291 // direct output, it is not a problem to ignore the underrun case. 5292 sp<Track> l = mLatestActiveTrack.promote(); 5293 bool last = l.get() == track; 5294 5295 if (track->isInvalid()) { 5296 ALOGW("An invalidated track shouldn't be in active list"); 5297 tracksToRemove->add(track); 5298 continue; 5299 } 5300 5301 if (track->mState == TrackBase::IDLE) { 5302 ALOGW("An idle track shouldn't be in active list"); 5303 continue; 5304 } 5305 5306 if (track->isPausing()) { 5307 track->setPaused(); 5308 if (last) { 5309 if (mHwSupportsPause && !mHwPaused) { 5310 doHwPause = true; 5311 mHwPaused = true; 5312 } 5313 // If we were part way through writing the mixbuffer to 5314 // the HAL we must save this until we resume 5315 // BUG - this will be wrong if a different track is made active, 5316 // in that case we want to discard the pending data in the 5317 // mixbuffer and tell the client to present it again when the 5318 // track is resumed 5319 mPausedWriteLength = mCurrentWriteLength; 5320 mPausedBytesRemaining = mBytesRemaining; 5321 mBytesRemaining = 0; // stop writing 5322 } 5323 tracksToRemove->add(track); 5324 } else if (track->isFlushPending()) { 5325 track->mRetryCount = kMaxTrackRetriesOffload; 5326 track->flushAck(); 5327 if (last) { 5328 mFlushPending = true; 5329 } 5330 } else if (track->isResumePending()){ 5331 track->resumeAck(); 5332 if (last) { 5333 if (mPausedBytesRemaining) { 5334 // Need to continue write that was interrupted 5335 mCurrentWriteLength = mPausedWriteLength; 5336 mBytesRemaining = mPausedBytesRemaining; 5337 mPausedBytesRemaining = 0; 5338 } 5339 if (mHwPaused) { 5340 doHwResume = true; 5341 mHwPaused = false; 5342 // threadLoop_mix() will handle the case that we need to 5343 // resume an interrupted write 5344 } 5345 // enable write to audio HAL 5346 mSleepTimeUs = 0; 5347 5348 // Do not handle new data in this iteration even if track->framesReady() 5349 mixerStatus = MIXER_TRACKS_ENABLED; 5350 } 5351 } else if (track->framesReady() && track->isReady() && 5352 !track->isPaused() && !track->isTerminated() && !track->isStopping_2()) { 5353 ALOGVV("OffloadThread: track %d s=%08x [OK]", track->name(), cblk->mServer); 5354 if (track->mFillingUpStatus == Track::FS_FILLED) { 5355 track->mFillingUpStatus = Track::FS_ACTIVE; 5356 // make sure processVolume_l() will apply new volume even if 0 5357 mLeftVolFloat = mRightVolFloat = -1.0; 5358 } 5359 5360 if (last) { 5361 sp<Track> previousTrack = mPreviousTrack.promote(); 5362 if (previousTrack != 0) { 5363 if (track != previousTrack.get()) { 5364 // Flush any data still being written from last track 5365 mBytesRemaining = 0; 5366 if (mPausedBytesRemaining) { 5367 // Last track was paused so we also need to flush saved 5368 // mixbuffer state and invalidate track so that it will 5369 // re-submit that unwritten data when it is next resumed 5370 mPausedBytesRemaining = 0; 5371 // Invalidate is a bit drastic - would be more efficient 5372 // to have a flag to tell client that some of the 5373 // previously written data was lost 5374 previousTrack->invalidate(); 5375 } 5376 // flush data already sent to the DSP if changing audio session as audio 5377 // comes from a different source. Also invalidate previous track to force a 5378 // seek when resuming. 5379 if (previousTrack->sessionId() != track->sessionId()) { 5380 previousTrack->invalidate(); 5381 } 5382 } 5383 } 5384 mPreviousTrack = track; 5385 // reset retry count 5386 track->mRetryCount = kMaxTrackRetriesOffload; 5387 mActiveTrack = t; 5388 mixerStatus = MIXER_TRACKS_READY; 5389 } 5390 } else { 5391 ALOGVV("OffloadThread: track %d s=%08x [NOT READY]", track->name(), cblk->mServer); 5392 if (track->isStopping_1()) { 5393 // Hardware buffer can hold a large amount of audio so we must 5394 // wait for all current track's data to drain before we say 5395 // that the track is stopped. 5396 if (mBytesRemaining == 0) { 5397 // Only start draining when all data in mixbuffer 5398 // has been written 5399 ALOGV("OffloadThread: underrun and STOPPING_1 -> draining, STOPPING_2"); 5400 track->mState = TrackBase::STOPPING_2; // so presentation completes after drain 5401 // do not drain if no data was ever sent to HAL (mStandby == true) 5402 if (last && !mStandby) { 5403 // do not modify drain sequence if we are already draining. This happens 5404 // when resuming from pause after drain. 5405 if ((mDrainSequence & 1) == 0) { 5406 mSleepTimeUs = 0; 5407 mStandbyTimeNs = systemTime() + mStandbyDelayNs; 5408 mixerStatus = MIXER_DRAIN_TRACK; 5409 mDrainSequence += 2; 5410 } 5411 if (mHwPaused) { 5412 // It is possible to move from PAUSED to STOPPING_1 without 5413 // a resume so we must ensure hardware is running 5414 doHwResume = true; 5415 mHwPaused = false; 5416 } 5417 } 5418 } 5419 } else if (track->isStopping_2()) { 5420 // Drain has completed or we are in standby, signal presentation complete 5421 if (!(mDrainSequence & 1) || !last || mStandby) { 5422 track->mState = TrackBase::STOPPED; 5423 size_t audioHALFrames = 5424 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 5425 int64_t framesWritten = 5426 mBytesWritten / mOutput->getFrameSize(); 5427 track->presentationComplete(framesWritten, audioHALFrames); 5428 track->reset(); 5429 tracksToRemove->add(track); 5430 } 5431 } else { 5432 // No buffers for this track. Give it a few chances to 5433 // fill a buffer, then remove it from active list. 5434 if (--(track->mRetryCount) <= 0) { 5435 ALOGV("OffloadThread: BUFFER TIMEOUT: remove(%d) from active list", 5436 track->name()); 5437 tracksToRemove->add(track); 5438 // indicate to client process that the track was disabled because of underrun; 5439 // it will then automatically call start() when data is available 5440 track->disable(); 5441 } else if (last){ 5442 mixerStatus = MIXER_TRACKS_ENABLED; 5443 } 5444 } 5445 } 5446 // compute volume for this track 5447 processVolume_l(track, last); 5448 } 5449 5450 // make sure the pause/flush/resume sequence is executed in the right order. 5451 // If a flush is pending and a track is active but the HW is not paused, force a HW pause 5452 // before flush and then resume HW. This can happen in case of pause/flush/resume 5453 // if resume is received before pause is executed. 5454 if (!mStandby && (doHwPause || (mFlushPending && !mHwPaused && (count != 0)))) { 5455 mOutput->stream->pause(mOutput->stream); 5456 } 5457 if (mFlushPending) { 5458 flushHw_l(); 5459 } 5460 if (!mStandby && doHwResume) { 5461 mOutput->stream->resume(mOutput->stream); 5462 } 5463 5464 // remove all the tracks that need to be... 5465 removeTracks_l(*tracksToRemove); 5466 5467 return mixerStatus; 5468} 5469 5470// must be called with thread mutex locked 5471bool AudioFlinger::OffloadThread::waitingAsyncCallback_l() 5472{ 5473 ALOGVV("waitingAsyncCallback_l mWriteAckSequence %d mDrainSequence %d", 5474 mWriteAckSequence, mDrainSequence); 5475 if (mUseAsyncWrite && ((mWriteAckSequence & 1) || (mDrainSequence & 1))) { 5476 return true; 5477 } 5478 return false; 5479} 5480 5481bool AudioFlinger::OffloadThread::waitingAsyncCallback() 5482{ 5483 Mutex::Autolock _l(mLock); 5484 return waitingAsyncCallback_l(); 5485} 5486 5487void AudioFlinger::OffloadThread::flushHw_l() 5488{ 5489 DirectOutputThread::flushHw_l(); 5490 // Flush anything still waiting in the mixbuffer 5491 mCurrentWriteLength = 0; 5492 mBytesRemaining = 0; 5493 mPausedWriteLength = 0; 5494 mPausedBytesRemaining = 0; 5495 5496 if (mUseAsyncWrite) { 5497 // discard any pending drain or write ack by incrementing sequence 5498 mWriteAckSequence = (mWriteAckSequence + 2) & ~1; 5499 mDrainSequence = (mDrainSequence + 2) & ~1; 5500 ALOG_ASSERT(mCallbackThread != 0); 5501 mCallbackThread->setWriteBlocked(mWriteAckSequence); 5502 mCallbackThread->setDraining(mDrainSequence); 5503 } 5504} 5505 5506uint32_t AudioFlinger::OffloadThread::activeSleepTimeUs() const 5507{ 5508 uint32_t time; 5509 if (audio_has_proportional_frames(mFormat)) { 5510 time = PlaybackThread::activeSleepTimeUs(); 5511 } else { 5512 // sleep time is half the duration of an audio HAL buffer. 5513 // Note: This can be problematic in case of underrun with variable bit rate and 5514 // current rate is much less than initial rate. 5515 time = (uint32_t)max(kDirectMinSleepTimeUs, mBufferDurationUs / 2); 5516 } 5517 return time; 5518} 5519 5520// ---------------------------------------------------------------------------- 5521 5522AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 5523 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id, bool systemReady) 5524 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->outDevice(), 5525 systemReady, DUPLICATING), 5526 mWaitTimeMs(UINT_MAX) 5527{ 5528 addOutputTrack(mainThread); 5529} 5530 5531AudioFlinger::DuplicatingThread::~DuplicatingThread() 5532{ 5533 for (size_t i = 0; i < mOutputTracks.size(); i++) { 5534 mOutputTracks[i]->destroy(); 5535 } 5536} 5537 5538void AudioFlinger::DuplicatingThread::threadLoop_mix() 5539{ 5540 // mix buffers... 5541 if (outputsReady(outputTracks)) { 5542 mAudioMixer->process(); 5543 } else { 5544 if (mMixerBufferValid) { 5545 memset(mMixerBuffer, 0, mMixerBufferSize); 5546 } else { 5547 memset(mSinkBuffer, 0, mSinkBufferSize); 5548 } 5549 } 5550 mSleepTimeUs = 0; 5551 writeFrames = mNormalFrameCount; 5552 mCurrentWriteLength = mSinkBufferSize; 5553 mStandbyTimeNs = systemTime() + mStandbyDelayNs; 5554} 5555 5556void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() 5557{ 5558 if (mSleepTimeUs == 0) { 5559 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 5560 mSleepTimeUs = mActiveSleepTimeUs; 5561 } else { 5562 mSleepTimeUs = mIdleSleepTimeUs; 5563 } 5564 } else if (mBytesWritten != 0) { 5565 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 5566 writeFrames = mNormalFrameCount; 5567 memset(mSinkBuffer, 0, mSinkBufferSize); 5568 } else { 5569 // flush remaining overflow buffers in output tracks 5570 writeFrames = 0; 5571 } 5572 mSleepTimeUs = 0; 5573 } 5574} 5575 5576ssize_t AudioFlinger::DuplicatingThread::threadLoop_write() 5577{ 5578 for (size_t i = 0; i < outputTracks.size(); i++) { 5579 outputTracks[i]->write(mSinkBuffer, writeFrames); 5580 } 5581 mStandby = false; 5582 return (ssize_t)mSinkBufferSize; 5583} 5584 5585void AudioFlinger::DuplicatingThread::threadLoop_standby() 5586{ 5587 // DuplicatingThread implements standby by stopping all tracks 5588 for (size_t i = 0; i < outputTracks.size(); i++) { 5589 outputTracks[i]->stop(); 5590 } 5591} 5592 5593void AudioFlinger::DuplicatingThread::saveOutputTracks() 5594{ 5595 outputTracks = mOutputTracks; 5596} 5597 5598void AudioFlinger::DuplicatingThread::clearOutputTracks() 5599{ 5600 outputTracks.clear(); 5601} 5602 5603void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 5604{ 5605 Mutex::Autolock _l(mLock); 5606 // The downstream MixerThread consumes thread->frameCount() amount of frames per mix pass. 5607 // Adjust for thread->sampleRate() to determine minimum buffer frame count. 5608 // Then triple buffer because Threads do not run synchronously and may not be clock locked. 5609 const size_t frameCount = 5610 3 * sourceFramesNeeded(mSampleRate, thread->frameCount(), thread->sampleRate()); 5611 // TODO: Consider asynchronous sample rate conversion to handle clock disparity 5612 // from different OutputTracks and their associated MixerThreads (e.g. one may 5613 // nearly empty and the other may be dropping data). 5614 5615 sp<OutputTrack> outputTrack = new OutputTrack(thread, 5616 this, 5617 mSampleRate, 5618 mFormat, 5619 mChannelMask, 5620 frameCount, 5621 IPCThreadState::self()->getCallingUid()); 5622 if (outputTrack->cblk() != NULL) { 5623 thread->setStreamVolume(AUDIO_STREAM_PATCH, 1.0f); 5624 mOutputTracks.add(outputTrack); 5625 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack.get(), thread); 5626 updateWaitTime_l(); 5627 } 5628} 5629 5630void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 5631{ 5632 Mutex::Autolock _l(mLock); 5633 for (size_t i = 0; i < mOutputTracks.size(); i++) { 5634 if (mOutputTracks[i]->thread() == thread) { 5635 mOutputTracks[i]->destroy(); 5636 mOutputTracks.removeAt(i); 5637 updateWaitTime_l(); 5638 if (thread->getOutput() == mOutput) { 5639 mOutput = NULL; 5640 } 5641 return; 5642 } 5643 } 5644 ALOGV("removeOutputTrack(): unknown thread: %p", thread); 5645} 5646 5647// caller must hold mLock 5648void AudioFlinger::DuplicatingThread::updateWaitTime_l() 5649{ 5650 mWaitTimeMs = UINT_MAX; 5651 for (size_t i = 0; i < mOutputTracks.size(); i++) { 5652 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 5653 if (strong != 0) { 5654 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 5655 if (waitTimeMs < mWaitTimeMs) { 5656 mWaitTimeMs = waitTimeMs; 5657 } 5658 } 5659 } 5660} 5661 5662 5663bool AudioFlinger::DuplicatingThread::outputsReady( 5664 const SortedVector< sp<OutputTrack> > &outputTracks) 5665{ 5666 for (size_t i = 0; i < outputTracks.size(); i++) { 5667 sp<ThreadBase> thread = outputTracks[i]->thread().promote(); 5668 if (thread == 0) { 5669 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", 5670 outputTracks[i].get()); 5671 return false; 5672 } 5673 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 5674 // see note at standby() declaration 5675 if (playbackThread->standby() && !playbackThread->isSuspended()) { 5676 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), 5677 thread.get()); 5678 return false; 5679 } 5680 } 5681 return true; 5682} 5683 5684uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() const 5685{ 5686 return (mWaitTimeMs * 1000) / 2; 5687} 5688 5689void AudioFlinger::DuplicatingThread::cacheParameters_l() 5690{ 5691 // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first 5692 updateWaitTime_l(); 5693 5694 MixerThread::cacheParameters_l(); 5695} 5696 5697// ---------------------------------------------------------------------------- 5698// Record 5699// ---------------------------------------------------------------------------- 5700 5701AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 5702 AudioStreamIn *input, 5703 audio_io_handle_t id, 5704 audio_devices_t outDevice, 5705 audio_devices_t inDevice, 5706 bool systemReady 5707#ifdef TEE_SINK 5708 , const sp<NBAIO_Sink>& teeSink 5709#endif 5710 ) : 5711 ThreadBase(audioFlinger, id, outDevice, inDevice, RECORD, systemReady), 5712 mInput(input), mActiveTracksGen(0), mRsmpInBuffer(NULL), 5713 // mRsmpInFrames and mRsmpInFramesP2 are set by readInputParameters_l() 5714 mRsmpInRear(0) 5715#ifdef TEE_SINK 5716 , mTeeSink(teeSink) 5717#endif 5718 , mReadOnlyHeap(new MemoryDealer(kRecordThreadReadOnlyHeapSize, 5719 "RecordThreadRO", MemoryHeapBase::READ_ONLY)) 5720 // mFastCapture below 5721 , mFastCaptureFutex(0) 5722 // mInputSource 5723 // mPipeSink 5724 // mPipeSource 5725 , mPipeFramesP2(0) 5726 // mPipeMemory 5727 // mFastCaptureNBLogWriter 5728 , mFastTrackAvail(false) 5729{ 5730 snprintf(mThreadName, kThreadNameLength, "AudioIn_%X", id); 5731 mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mThreadName); 5732 5733 readInputParameters_l(); 5734 5735 // create an NBAIO source for the HAL input stream, and negotiate 5736 mInputSource = new AudioStreamInSource(input->stream); 5737 size_t numCounterOffers = 0; 5738 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount, mFormat)}; 5739#if !LOG_NDEBUG 5740 ssize_t index = 5741#else 5742 (void) 5743#endif 5744 mInputSource->negotiate(offers, 1, NULL, numCounterOffers); 5745 ALOG_ASSERT(index == 0); 5746 5747 // initialize fast capture depending on configuration 5748 bool initFastCapture; 5749 switch (kUseFastCapture) { 5750 case FastCapture_Never: 5751 initFastCapture = false; 5752 break; 5753 case FastCapture_Always: 5754 initFastCapture = true; 5755 break; 5756 case FastCapture_Static: 5757 initFastCapture = (mFrameCount * 1000) / mSampleRate < kMinNormalCaptureBufferSizeMs; 5758 break; 5759 // case FastCapture_Dynamic: 5760 } 5761 5762 if (initFastCapture) { 5763 // create a Pipe for FastCapture to write to, and for us and fast tracks to read from 5764 NBAIO_Format format = mInputSource->format(); 5765 size_t pipeFramesP2 = roundup(mSampleRate / 25); // double-buffering of 20 ms each 5766 size_t pipeSize = pipeFramesP2 * Format_frameSize(format); 5767 void *pipeBuffer; 5768 const sp<MemoryDealer> roHeap(readOnlyHeap()); 5769 sp<IMemory> pipeMemory; 5770 if ((roHeap == 0) || 5771 (pipeMemory = roHeap->allocate(pipeSize)) == 0 || 5772 (pipeBuffer = pipeMemory->pointer()) == NULL) { 5773 ALOGE("not enough memory for pipe buffer size=%zu", pipeSize); 5774 goto failed; 5775 } 5776 // pipe will be shared directly with fast clients, so clear to avoid leaking old information 5777 memset(pipeBuffer, 0, pipeSize); 5778 Pipe *pipe = new Pipe(pipeFramesP2, format, pipeBuffer); 5779 const NBAIO_Format offers[1] = {format}; 5780 size_t numCounterOffers = 0; 5781 ssize_t index = pipe->negotiate(offers, 1, NULL, numCounterOffers); 5782 ALOG_ASSERT(index == 0); 5783 mPipeSink = pipe; 5784 PipeReader *pipeReader = new PipeReader(*pipe); 5785 numCounterOffers = 0; 5786 index = pipeReader->negotiate(offers, 1, NULL, numCounterOffers); 5787 ALOG_ASSERT(index == 0); 5788 mPipeSource = pipeReader; 5789 mPipeFramesP2 = pipeFramesP2; 5790 mPipeMemory = pipeMemory; 5791 5792 // create fast capture 5793 mFastCapture = new FastCapture(); 5794 FastCaptureStateQueue *sq = mFastCapture->sq(); 5795#ifdef STATE_QUEUE_DUMP 5796 // FIXME 5797#endif 5798 FastCaptureState *state = sq->begin(); 5799 state->mCblk = NULL; 5800 state->mInputSource = mInputSource.get(); 5801 state->mInputSourceGen++; 5802 state->mPipeSink = pipe; 5803 state->mPipeSinkGen++; 5804 state->mFrameCount = mFrameCount; 5805 state->mCommand = FastCaptureState::COLD_IDLE; 5806 // already done in constructor initialization list 5807 //mFastCaptureFutex = 0; 5808 state->mColdFutexAddr = &mFastCaptureFutex; 5809 state->mColdGen++; 5810 state->mDumpState = &mFastCaptureDumpState; 5811#ifdef TEE_SINK 5812 // FIXME 5813#endif 5814 mFastCaptureNBLogWriter = audioFlinger->newWriter_l(kFastCaptureLogSize, "FastCapture"); 5815 state->mNBLogWriter = mFastCaptureNBLogWriter.get(); 5816 sq->end(); 5817 sq->push(FastCaptureStateQueue::BLOCK_UNTIL_PUSHED); 5818 5819 // start the fast capture 5820 mFastCapture->run("FastCapture", ANDROID_PRIORITY_URGENT_AUDIO); 5821 pid_t tid = mFastCapture->getTid(); 5822 sendPrioConfigEvent(getpid_cached, tid, kPriorityFastMixer); 5823#ifdef AUDIO_WATCHDOG 5824 // FIXME 5825#endif 5826 5827 mFastTrackAvail = true; 5828 } 5829failed: ; 5830 5831 // FIXME mNormalSource 5832} 5833 5834AudioFlinger::RecordThread::~RecordThread() 5835{ 5836 if (mFastCapture != 0) { 5837 FastCaptureStateQueue *sq = mFastCapture->sq(); 5838 FastCaptureState *state = sq->begin(); 5839 if (state->mCommand == FastCaptureState::COLD_IDLE) { 5840 int32_t old = android_atomic_inc(&mFastCaptureFutex); 5841 if (old == -1) { 5842 (void) syscall(__NR_futex, &mFastCaptureFutex, FUTEX_WAKE_PRIVATE, 1); 5843 } 5844 } 5845 state->mCommand = FastCaptureState::EXIT; 5846 sq->end(); 5847 sq->push(FastCaptureStateQueue::BLOCK_UNTIL_PUSHED); 5848 mFastCapture->join(); 5849 mFastCapture.clear(); 5850 } 5851 mAudioFlinger->unregisterWriter(mFastCaptureNBLogWriter); 5852 mAudioFlinger->unregisterWriter(mNBLogWriter); 5853 free(mRsmpInBuffer); 5854} 5855 5856void AudioFlinger::RecordThread::onFirstRef() 5857{ 5858 run(mThreadName, PRIORITY_URGENT_AUDIO); 5859} 5860 5861bool AudioFlinger::RecordThread::threadLoop() 5862{ 5863 nsecs_t lastWarning = 0; 5864 5865 inputStandBy(); 5866 5867reacquire_wakelock: 5868 sp<RecordTrack> activeTrack; 5869 int activeTracksGen; 5870 { 5871 Mutex::Autolock _l(mLock); 5872 size_t size = mActiveTracks.size(); 5873 activeTracksGen = mActiveTracksGen; 5874 if (size > 0) { 5875 // FIXME an arbitrary choice 5876 activeTrack = mActiveTracks[0]; 5877 acquireWakeLock_l(activeTrack->uid()); 5878 if (size > 1) { 5879 SortedVector<int> tmp; 5880 for (size_t i = 0; i < size; i++) { 5881 tmp.add(mActiveTracks[i]->uid()); 5882 } 5883 updateWakeLockUids_l(tmp); 5884 } 5885 } else { 5886 acquireWakeLock_l(-1); 5887 } 5888 } 5889 5890 // used to request a deferred sleep, to be executed later while mutex is unlocked 5891 uint32_t sleepUs = 0; 5892 5893 // loop while there is work to do 5894 for (;;) { 5895 Vector< sp<EffectChain> > effectChains; 5896 5897 // sleep with mutex unlocked 5898 if (sleepUs > 0) { 5899 ATRACE_BEGIN("sleep"); 5900 usleep(sleepUs); 5901 ATRACE_END(); 5902 sleepUs = 0; 5903 } 5904 5905 // activeTracks accumulates a copy of a subset of mActiveTracks 5906 Vector< sp<RecordTrack> > activeTracks; 5907 5908 // reference to the (first and only) active fast track 5909 sp<RecordTrack> fastTrack; 5910 5911 // reference to a fast track which is about to be removed 5912 sp<RecordTrack> fastTrackToRemove; 5913 5914 { // scope for mLock 5915 Mutex::Autolock _l(mLock); 5916 5917 processConfigEvents_l(); 5918 5919 // check exitPending here because checkForNewParameters_l() and 5920 // checkForNewParameters_l() can temporarily release mLock 5921 if (exitPending()) { 5922 break; 5923 } 5924 5925 // if no active track(s), then standby and release wakelock 5926 size_t size = mActiveTracks.size(); 5927 if (size == 0) { 5928 standbyIfNotAlreadyInStandby(); 5929 // exitPending() can't become true here 5930 releaseWakeLock_l(); 5931 ALOGV("RecordThread: loop stopping"); 5932 // go to sleep 5933 mWaitWorkCV.wait(mLock); 5934 ALOGV("RecordThread: loop starting"); 5935 goto reacquire_wakelock; 5936 } 5937 5938 if (mActiveTracksGen != activeTracksGen) { 5939 activeTracksGen = mActiveTracksGen; 5940 SortedVector<int> tmp; 5941 for (size_t i = 0; i < size; i++) { 5942 tmp.add(mActiveTracks[i]->uid()); 5943 } 5944 updateWakeLockUids_l(tmp); 5945 } 5946 5947 bool doBroadcast = false; 5948 for (size_t i = 0; i < size; ) { 5949 5950 activeTrack = mActiveTracks[i]; 5951 if (activeTrack->isTerminated()) { 5952 if (activeTrack->isFastTrack()) { 5953 ALOG_ASSERT(fastTrackToRemove == 0); 5954 fastTrackToRemove = activeTrack; 5955 } 5956 removeTrack_l(activeTrack); 5957 mActiveTracks.remove(activeTrack); 5958 mActiveTracksGen++; 5959 size--; 5960 continue; 5961 } 5962 5963 TrackBase::track_state activeTrackState = activeTrack->mState; 5964 switch (activeTrackState) { 5965 5966 case TrackBase::PAUSING: 5967 mActiveTracks.remove(activeTrack); 5968 mActiveTracksGen++; 5969 doBroadcast = true; 5970 size--; 5971 continue; 5972 5973 case TrackBase::STARTING_1: 5974 sleepUs = 10000; 5975 i++; 5976 continue; 5977 5978 case TrackBase::STARTING_2: 5979 doBroadcast = true; 5980 mStandby = false; 5981 activeTrack->mState = TrackBase::ACTIVE; 5982 break; 5983 5984 case TrackBase::ACTIVE: 5985 break; 5986 5987 case TrackBase::IDLE: 5988 i++; 5989 continue; 5990 5991 default: 5992 LOG_ALWAYS_FATAL("Unexpected activeTrackState %d", activeTrackState); 5993 } 5994 5995 activeTracks.add(activeTrack); 5996 i++; 5997 5998 if (activeTrack->isFastTrack()) { 5999 ALOG_ASSERT(!mFastTrackAvail); 6000 ALOG_ASSERT(fastTrack == 0); 6001 fastTrack = activeTrack; 6002 } 6003 } 6004 if (doBroadcast) { 6005 mStartStopCond.broadcast(); 6006 } 6007 6008 // sleep if there are no active tracks to process 6009 if (activeTracks.size() == 0) { 6010 if (sleepUs == 0) { 6011 sleepUs = kRecordThreadSleepUs; 6012 } 6013 continue; 6014 } 6015 sleepUs = 0; 6016 6017 lockEffectChains_l(effectChains); 6018 } 6019 6020 // thread mutex is now unlocked, mActiveTracks unknown, activeTracks.size() > 0 6021 6022 size_t size = effectChains.size(); 6023 for (size_t i = 0; i < size; i++) { 6024 // thread mutex is not locked, but effect chain is locked 6025 effectChains[i]->process_l(); 6026 } 6027 6028 // Push a new fast capture state if fast capture is not already running, or cblk change 6029 if (mFastCapture != 0) { 6030 FastCaptureStateQueue *sq = mFastCapture->sq(); 6031 FastCaptureState *state = sq->begin(); 6032 bool didModify = false; 6033 FastCaptureStateQueue::block_t block = FastCaptureStateQueue::BLOCK_UNTIL_PUSHED; 6034 if (state->mCommand != FastCaptureState::READ_WRITE /* FIXME && 6035 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)*/) { 6036 if (state->mCommand == FastCaptureState::COLD_IDLE) { 6037 int32_t old = android_atomic_inc(&mFastCaptureFutex); 6038 if (old == -1) { 6039 (void) syscall(__NR_futex, &mFastCaptureFutex, FUTEX_WAKE_PRIVATE, 1); 6040 } 6041 } 6042 state->mCommand = FastCaptureState::READ_WRITE; 6043#if 0 // FIXME 6044 mFastCaptureDumpState.increaseSamplingN(mAudioFlinger->isLowRamDevice() ? 6045 FastThreadDumpState::kSamplingNforLowRamDevice : 6046 FastThreadDumpState::kSamplingN); 6047#endif 6048 didModify = true; 6049 } 6050 audio_track_cblk_t *cblkOld = state->mCblk; 6051 audio_track_cblk_t *cblkNew = fastTrack != 0 ? fastTrack->cblk() : NULL; 6052 if (cblkNew != cblkOld) { 6053 state->mCblk = cblkNew; 6054 // block until acked if removing a fast track 6055 if (cblkOld != NULL) { 6056 block = FastCaptureStateQueue::BLOCK_UNTIL_ACKED; 6057 } 6058 didModify = true; 6059 } 6060 sq->end(didModify); 6061 if (didModify) { 6062 sq->push(block); 6063#if 0 6064 if (kUseFastCapture == FastCapture_Dynamic) { 6065 mNormalSource = mPipeSource; 6066 } 6067#endif 6068 } 6069 } 6070 6071 // now run the fast track destructor with thread mutex unlocked 6072 fastTrackToRemove.clear(); 6073 6074 // Read from HAL to keep up with fastest client if multiple active tracks, not slowest one. 6075 // Only the client(s) that are too slow will overrun. But if even the fastest client is too 6076 // slow, then this RecordThread will overrun by not calling HAL read often enough. 6077 // If destination is non-contiguous, first read past the nominal end of buffer, then 6078 // copy to the right place. Permitted because mRsmpInBuffer was over-allocated. 6079 6080 int32_t rear = mRsmpInRear & (mRsmpInFramesP2 - 1); 6081 ssize_t framesRead; 6082 6083 // If an NBAIO source is present, use it to read the normal capture's data 6084 if (mPipeSource != 0) { 6085 size_t framesToRead = mBufferSize / mFrameSize; 6086 framesRead = mPipeSource->read((uint8_t*)mRsmpInBuffer + rear * mFrameSize, 6087 framesToRead); 6088 if (framesRead == 0) { 6089 // since pipe is non-blocking, simulate blocking input 6090 sleepUs = (framesToRead * 1000000LL) / mSampleRate; 6091 } 6092 // otherwise use the HAL / AudioStreamIn directly 6093 } else { 6094 ssize_t bytesRead = mInput->stream->read(mInput->stream, 6095 (uint8_t*)mRsmpInBuffer + rear * mFrameSize, mBufferSize); 6096 if (bytesRead < 0) { 6097 framesRead = bytesRead; 6098 } else { 6099 framesRead = bytesRead / mFrameSize; 6100 } 6101 } 6102 6103 // Update server timestamp with server stats 6104 // systemTime() is optional if the hardware supports timestamps. 6105 mTimestamp.mPosition[ExtendedTimestamp::LOCATION_SERVER] += framesRead; 6106 mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_SERVER] = systemTime(); 6107 6108 // Update server timestamp with kernel stats 6109 if (mInput->stream->get_capture_position != nullptr) { 6110 int64_t position, time; 6111 int ret = mInput->stream->get_capture_position(mInput->stream, &position, &time); 6112 if (ret == NO_ERROR) { 6113 mTimestamp.mPosition[ExtendedTimestamp::LOCATION_KERNEL] = position; 6114 mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL] = time; 6115 // Note: In general record buffers should tend to be empty in 6116 // a properly running pipeline. 6117 // 6118 // Also, it is not advantageous to call get_presentation_position during the read 6119 // as the read obtains a lock, preventing the timestamp call from executing. 6120 } 6121 } 6122 // Use this to track timestamp information 6123 // ALOGD("%s", mTimestamp.toString().c_str()); 6124 6125 if (framesRead < 0 || (framesRead == 0 && mPipeSource == 0)) { 6126 ALOGE("read failed: framesRead=%d", framesRead); 6127 // Force input into standby so that it tries to recover at next read attempt 6128 inputStandBy(); 6129 sleepUs = kRecordThreadSleepUs; 6130 } 6131 if (framesRead <= 0) { 6132 goto unlock; 6133 } 6134 ALOG_ASSERT(framesRead > 0); 6135 6136 if (mTeeSink != 0) { 6137 (void) mTeeSink->write((uint8_t*)mRsmpInBuffer + rear * mFrameSize, framesRead); 6138 } 6139 // If destination is non-contiguous, we now correct for reading past end of buffer. 6140 { 6141 size_t part1 = mRsmpInFramesP2 - rear; 6142 if ((size_t) framesRead > part1) { 6143 memcpy(mRsmpInBuffer, (uint8_t*)mRsmpInBuffer + mRsmpInFramesP2 * mFrameSize, 6144 (framesRead - part1) * mFrameSize); 6145 } 6146 } 6147 rear = mRsmpInRear += framesRead; 6148 6149 size = activeTracks.size(); 6150 // loop over each active track 6151 for (size_t i = 0; i < size; i++) { 6152 activeTrack = activeTracks[i]; 6153 6154 // skip fast tracks, as those are handled directly by FastCapture 6155 if (activeTrack->isFastTrack()) { 6156 continue; 6157 } 6158 6159 // TODO: This code probably should be moved to RecordTrack. 6160 // TODO: Update the activeTrack buffer converter in case of reconfigure. 6161 6162 enum { 6163 OVERRUN_UNKNOWN, 6164 OVERRUN_TRUE, 6165 OVERRUN_FALSE 6166 } overrun = OVERRUN_UNKNOWN; 6167 6168 // loop over getNextBuffer to handle circular sink 6169 for (;;) { 6170 6171 activeTrack->mSink.frameCount = ~0; 6172 status_t status = activeTrack->getNextBuffer(&activeTrack->mSink); 6173 size_t framesOut = activeTrack->mSink.frameCount; 6174 LOG_ALWAYS_FATAL_IF((status == OK) != (framesOut > 0)); 6175 6176 // check available frames and handle overrun conditions 6177 // if the record track isn't draining fast enough. 6178 bool hasOverrun; 6179 size_t framesIn; 6180 activeTrack->mResamplerBufferProvider->sync(&framesIn, &hasOverrun); 6181 if (hasOverrun) { 6182 overrun = OVERRUN_TRUE; 6183 } 6184 if (framesOut == 0 || framesIn == 0) { 6185 break; 6186 } 6187 6188 // Don't allow framesOut to be larger than what is possible with resampling 6189 // from framesIn. 6190 // This isn't strictly necessary but helps limit buffer resizing in 6191 // RecordBufferConverter. TODO: remove when no longer needed. 6192 framesOut = min(framesOut, 6193 destinationFramesPossible( 6194 framesIn, mSampleRate, activeTrack->mSampleRate)); 6195 // process frames from the RecordThread buffer provider to the RecordTrack buffer 6196 framesOut = activeTrack->mRecordBufferConverter->convert( 6197 activeTrack->mSink.raw, activeTrack->mResamplerBufferProvider, framesOut); 6198 6199 if (framesOut > 0 && (overrun == OVERRUN_UNKNOWN)) { 6200 overrun = OVERRUN_FALSE; 6201 } 6202 6203 if (activeTrack->mFramesToDrop == 0) { 6204 if (framesOut > 0) { 6205 activeTrack->mSink.frameCount = framesOut; 6206 activeTrack->releaseBuffer(&activeTrack->mSink); 6207 } 6208 } else { 6209 // FIXME could do a partial drop of framesOut 6210 if (activeTrack->mFramesToDrop > 0) { 6211 activeTrack->mFramesToDrop -= framesOut; 6212 if (activeTrack->mFramesToDrop <= 0) { 6213 activeTrack->clearSyncStartEvent(); 6214 } 6215 } else { 6216 activeTrack->mFramesToDrop += framesOut; 6217 if (activeTrack->mFramesToDrop >= 0 || activeTrack->mSyncStartEvent == 0 || 6218 activeTrack->mSyncStartEvent->isCancelled()) { 6219 ALOGW("Synced record %s, session %d, trigger session %d", 6220 (activeTrack->mFramesToDrop >= 0) ? "timed out" : "cancelled", 6221 activeTrack->sessionId(), 6222 (activeTrack->mSyncStartEvent != 0) ? 6223 activeTrack->mSyncStartEvent->triggerSession() : 6224 AUDIO_SESSION_NONE); 6225 activeTrack->clearSyncStartEvent(); 6226 } 6227 } 6228 } 6229 6230 if (framesOut == 0) { 6231 break; 6232 } 6233 } 6234 6235 switch (overrun) { 6236 case OVERRUN_TRUE: 6237 // client isn't retrieving buffers fast enough 6238 if (!activeTrack->setOverflow()) { 6239 nsecs_t now = systemTime(); 6240 // FIXME should lastWarning per track? 6241 if ((now - lastWarning) > kWarningThrottleNs) { 6242 ALOGW("RecordThread: buffer overflow"); 6243 lastWarning = now; 6244 } 6245 } 6246 break; 6247 case OVERRUN_FALSE: 6248 activeTrack->clearOverflow(); 6249 break; 6250 case OVERRUN_UNKNOWN: 6251 break; 6252 } 6253 6254 // update frame information and push timestamp out 6255 activeTrack->updateTrackFrameInfo( 6256 activeTrack->mServerProxy->framesReleased(), 6257 mTimestamp.mPosition[ExtendedTimestamp::LOCATION_SERVER], 6258 mSampleRate, mTimestamp); 6259 } 6260 6261unlock: 6262 // enable changes in effect chain 6263 unlockEffectChains(effectChains); 6264 // effectChains doesn't need to be cleared, since it is cleared by destructor at scope end 6265 } 6266 6267 standbyIfNotAlreadyInStandby(); 6268 6269 { 6270 Mutex::Autolock _l(mLock); 6271 for (size_t i = 0; i < mTracks.size(); i++) { 6272 sp<RecordTrack> track = mTracks[i]; 6273 track->invalidate(); 6274 } 6275 mActiveTracks.clear(); 6276 mActiveTracksGen++; 6277 mStartStopCond.broadcast(); 6278 } 6279 6280 releaseWakeLock(); 6281 6282 ALOGV("RecordThread %p exiting", this); 6283 return false; 6284} 6285 6286void AudioFlinger::RecordThread::standbyIfNotAlreadyInStandby() 6287{ 6288 if (!mStandby) { 6289 inputStandBy(); 6290 mStandby = true; 6291 } 6292} 6293 6294void AudioFlinger::RecordThread::inputStandBy() 6295{ 6296 // Idle the fast capture if it's currently running 6297 if (mFastCapture != 0) { 6298 FastCaptureStateQueue *sq = mFastCapture->sq(); 6299 FastCaptureState *state = sq->begin(); 6300 if (!(state->mCommand & FastCaptureState::IDLE)) { 6301 state->mCommand = FastCaptureState::COLD_IDLE; 6302 state->mColdFutexAddr = &mFastCaptureFutex; 6303 state->mColdGen++; 6304 mFastCaptureFutex = 0; 6305 sq->end(); 6306 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 6307 sq->push(FastCaptureStateQueue::BLOCK_UNTIL_ACKED); 6308#if 0 6309 if (kUseFastCapture == FastCapture_Dynamic) { 6310 // FIXME 6311 } 6312#endif 6313#ifdef AUDIO_WATCHDOG 6314 // FIXME 6315#endif 6316 } else { 6317 sq->end(false /*didModify*/); 6318 } 6319 } 6320 mInput->stream->common.standby(&mInput->stream->common); 6321} 6322 6323// RecordThread::createRecordTrack_l() must be called with AudioFlinger::mLock held 6324sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 6325 const sp<AudioFlinger::Client>& client, 6326 uint32_t sampleRate, 6327 audio_format_t format, 6328 audio_channel_mask_t channelMask, 6329 size_t *pFrameCount, 6330 audio_session_t sessionId, 6331 size_t *notificationFrames, 6332 int uid, 6333 IAudioFlinger::track_flags_t *flags, 6334 pid_t tid, 6335 status_t *status) 6336{ 6337 size_t frameCount = *pFrameCount; 6338 sp<RecordTrack> track; 6339 status_t lStatus; 6340 6341 // client expresses a preference for FAST, but we get the final say 6342 if (*flags & IAudioFlinger::TRACK_FAST) { 6343 if ( 6344 // we formerly checked for a callback handler (non-0 tid), 6345 // but that is no longer required for TRANSFER_OBTAIN mode 6346 // 6347 // frame count is not specified, or is exactly the pipe depth 6348 ((frameCount == 0) || (frameCount == mPipeFramesP2)) && 6349 // PCM data 6350 audio_is_linear_pcm(format) && 6351 // hardware format 6352 (format == mFormat) && 6353 // hardware channel mask 6354 (channelMask == mChannelMask) && 6355 // hardware sample rate 6356 (sampleRate == mSampleRate) && 6357 // record thread has an associated fast capture 6358 hasFastCapture() && 6359 // there are sufficient fast track slots available 6360 mFastTrackAvail 6361 ) { 6362 ALOGV("AUDIO_INPUT_FLAG_FAST accepted: frameCount=%u mFrameCount=%u", 6363 frameCount, mFrameCount); 6364 } else { 6365 ALOGV("AUDIO_INPUT_FLAG_FAST denied: frameCount=%u mFrameCount=%u mPipeFramesP2=%u " 6366 "format=%#x isLinear=%d channelMask=%#x sampleRate=%u mSampleRate=%u " 6367 "hasFastCapture=%d tid=%d mFastTrackAvail=%d", 6368 frameCount, mFrameCount, mPipeFramesP2, 6369 format, audio_is_linear_pcm(format), channelMask, sampleRate, mSampleRate, 6370 hasFastCapture(), tid, mFastTrackAvail); 6371 *flags &= ~IAudioFlinger::TRACK_FAST; 6372 } 6373 } 6374 6375 // compute track buffer size in frames, and suggest the notification frame count 6376 if (*flags & IAudioFlinger::TRACK_FAST) { 6377 // fast track: frame count is exactly the pipe depth 6378 frameCount = mPipeFramesP2; 6379 // ignore requested notificationFrames, and always notify exactly once every HAL buffer 6380 *notificationFrames = mFrameCount; 6381 } else { 6382 // not fast track: max notification period is resampled equivalent of one HAL buffer time 6383 // or 20 ms if there is a fast capture 6384 // TODO This could be a roundupRatio inline, and const 6385 size_t maxNotificationFrames = ((int64_t) (hasFastCapture() ? mSampleRate/50 : mFrameCount) 6386 * sampleRate + mSampleRate - 1) / mSampleRate; 6387 // minimum number of notification periods is at least kMinNotifications, 6388 // and at least kMinMs rounded up to a whole notification period (minNotificationsByMs) 6389 static const size_t kMinNotifications = 3; 6390 static const uint32_t kMinMs = 30; 6391 // TODO This could be a roundupRatio inline 6392 const size_t minFramesByMs = (sampleRate * kMinMs + 1000 - 1) / 1000; 6393 // TODO This could be a roundupRatio inline 6394 const size_t minNotificationsByMs = (minFramesByMs + maxNotificationFrames - 1) / 6395 maxNotificationFrames; 6396 const size_t minFrameCount = maxNotificationFrames * 6397 max(kMinNotifications, minNotificationsByMs); 6398 frameCount = max(frameCount, minFrameCount); 6399 if (*notificationFrames == 0 || *notificationFrames > maxNotificationFrames) { 6400 *notificationFrames = maxNotificationFrames; 6401 } 6402 } 6403 *pFrameCount = frameCount; 6404 6405 lStatus = initCheck(); 6406 if (lStatus != NO_ERROR) { 6407 ALOGE("createRecordTrack_l() audio driver not initialized"); 6408 goto Exit; 6409 } 6410 6411 { // scope for mLock 6412 Mutex::Autolock _l(mLock); 6413 6414 track = new RecordTrack(this, client, sampleRate, 6415 format, channelMask, frameCount, NULL, sessionId, uid, 6416 *flags, TrackBase::TYPE_DEFAULT); 6417 6418 lStatus = track->initCheck(); 6419 if (lStatus != NO_ERROR) { 6420 ALOGE("createRecordTrack_l() initCheck failed %d; no control block?", lStatus); 6421 // track must be cleared from the caller as the caller has the AF lock 6422 goto Exit; 6423 } 6424 mTracks.add(track); 6425 6426 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 6427 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 6428 mAudioFlinger->btNrecIsOff(); 6429 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 6430 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 6431 6432 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 6433 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 6434 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 6435 // so ask activity manager to do this on our behalf 6436 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 6437 } 6438 } 6439 6440 lStatus = NO_ERROR; 6441 6442Exit: 6443 *status = lStatus; 6444 return track; 6445} 6446 6447status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, 6448 AudioSystem::sync_event_t event, 6449 audio_session_t triggerSession) 6450{ 6451 ALOGV("RecordThread::start event %d, triggerSession %d", event, triggerSession); 6452 sp<ThreadBase> strongMe = this; 6453 status_t status = NO_ERROR; 6454 6455 if (event == AudioSystem::SYNC_EVENT_NONE) { 6456 recordTrack->clearSyncStartEvent(); 6457 } else if (event != AudioSystem::SYNC_EVENT_SAME) { 6458 recordTrack->mSyncStartEvent = mAudioFlinger->createSyncEvent(event, 6459 triggerSession, 6460 recordTrack->sessionId(), 6461 syncStartEventCallback, 6462 recordTrack); 6463 // Sync event can be cancelled by the trigger session if the track is not in a 6464 // compatible state in which case we start record immediately 6465 if (recordTrack->mSyncStartEvent->isCancelled()) { 6466 recordTrack->clearSyncStartEvent(); 6467 } else { 6468 // do not wait for the event for more than AudioSystem::kSyncRecordStartTimeOutMs 6469 recordTrack->mFramesToDrop = - 6470 ((AudioSystem::kSyncRecordStartTimeOutMs * recordTrack->mSampleRate) / 1000); 6471 } 6472 } 6473 6474 { 6475 // This section is a rendezvous between binder thread executing start() and RecordThread 6476 AutoMutex lock(mLock); 6477 if (mActiveTracks.indexOf(recordTrack) >= 0) { 6478 if (recordTrack->mState == TrackBase::PAUSING) { 6479 ALOGV("active record track PAUSING -> ACTIVE"); 6480 recordTrack->mState = TrackBase::ACTIVE; 6481 } else { 6482 ALOGV("active record track state %d", recordTrack->mState); 6483 } 6484 return status; 6485 } 6486 6487 // TODO consider other ways of handling this, such as changing the state to :STARTING and 6488 // adding the track to mActiveTracks after returning from AudioSystem::startInput(), 6489 // or using a separate command thread 6490 recordTrack->mState = TrackBase::STARTING_1; 6491 mActiveTracks.add(recordTrack); 6492 mActiveTracksGen++; 6493 status_t status = NO_ERROR; 6494 if (recordTrack->isExternalTrack()) { 6495 mLock.unlock(); 6496 status = AudioSystem::startInput(mId, recordTrack->sessionId()); 6497 mLock.lock(); 6498 // FIXME should verify that recordTrack is still in mActiveTracks 6499 if (status != NO_ERROR) { 6500 mActiveTracks.remove(recordTrack); 6501 mActiveTracksGen++; 6502 recordTrack->clearSyncStartEvent(); 6503 ALOGV("RecordThread::start error %d", status); 6504 return status; 6505 } 6506 } 6507 // Catch up with current buffer indices if thread is already running. 6508 // This is what makes a new client discard all buffered data. If the track's mRsmpInFront 6509 // was initialized to some value closer to the thread's mRsmpInFront, then the track could 6510 // see previously buffered data before it called start(), but with greater risk of overrun. 6511 6512 recordTrack->mResamplerBufferProvider->reset(); 6513 // clear any converter state as new data will be discontinuous 6514 recordTrack->mRecordBufferConverter->reset(); 6515 recordTrack->mState = TrackBase::STARTING_2; 6516 // signal thread to start 6517 mWaitWorkCV.broadcast(); 6518 if (mActiveTracks.indexOf(recordTrack) < 0) { 6519 ALOGV("Record failed to start"); 6520 status = BAD_VALUE; 6521 goto startError; 6522 } 6523 return status; 6524 } 6525 6526startError: 6527 if (recordTrack->isExternalTrack()) { 6528 AudioSystem::stopInput(mId, recordTrack->sessionId()); 6529 } 6530 recordTrack->clearSyncStartEvent(); 6531 // FIXME I wonder why we do not reset the state here? 6532 return status; 6533} 6534 6535void AudioFlinger::RecordThread::syncStartEventCallback(const wp<SyncEvent>& event) 6536{ 6537 sp<SyncEvent> strongEvent = event.promote(); 6538 6539 if (strongEvent != 0) { 6540 sp<RefBase> ptr = strongEvent->cookie().promote(); 6541 if (ptr != 0) { 6542 RecordTrack *recordTrack = (RecordTrack *)ptr.get(); 6543 recordTrack->handleSyncStartEvent(strongEvent); 6544 } 6545 } 6546} 6547 6548bool AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 6549 ALOGV("RecordThread::stop"); 6550 AutoMutex _l(mLock); 6551 if (mActiveTracks.indexOf(recordTrack) != 0 || recordTrack->mState == TrackBase::PAUSING) { 6552 return false; 6553 } 6554 // note that threadLoop may still be processing the track at this point [without lock] 6555 recordTrack->mState = TrackBase::PAUSING; 6556 // do not wait for mStartStopCond if exiting 6557 if (exitPending()) { 6558 return true; 6559 } 6560 // FIXME incorrect usage of wait: no explicit predicate or loop 6561 mStartStopCond.wait(mLock); 6562 // if we have been restarted, recordTrack is in mActiveTracks here 6563 if (exitPending() || mActiveTracks.indexOf(recordTrack) != 0) { 6564 ALOGV("Record stopped OK"); 6565 return true; 6566 } 6567 return false; 6568} 6569 6570bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event __unused) const 6571{ 6572 return false; 6573} 6574 6575status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event __unused) 6576{ 6577#if 0 // This branch is currently dead code, but is preserved in case it will be needed in future 6578 if (!isValidSyncEvent(event)) { 6579 return BAD_VALUE; 6580 } 6581 6582 audio_session_t eventSession = event->triggerSession(); 6583 status_t ret = NAME_NOT_FOUND; 6584 6585 Mutex::Autolock _l(mLock); 6586 6587 for (size_t i = 0; i < mTracks.size(); i++) { 6588 sp<RecordTrack> track = mTracks[i]; 6589 if (eventSession == track->sessionId()) { 6590 (void) track->setSyncEvent(event); 6591 ret = NO_ERROR; 6592 } 6593 } 6594 return ret; 6595#else 6596 return BAD_VALUE; 6597#endif 6598} 6599 6600// destroyTrack_l() must be called with ThreadBase::mLock held 6601void AudioFlinger::RecordThread::destroyTrack_l(const sp<RecordTrack>& track) 6602{ 6603 track->terminate(); 6604 track->mState = TrackBase::STOPPED; 6605 // active tracks are removed by threadLoop() 6606 if (mActiveTracks.indexOf(track) < 0) { 6607 removeTrack_l(track); 6608 } 6609} 6610 6611void AudioFlinger::RecordThread::removeTrack_l(const sp<RecordTrack>& track) 6612{ 6613 mTracks.remove(track); 6614 // need anything related to effects here? 6615 if (track->isFastTrack()) { 6616 ALOG_ASSERT(!mFastTrackAvail); 6617 mFastTrackAvail = true; 6618 } 6619} 6620 6621void AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 6622{ 6623 dumpInternals(fd, args); 6624 dumpTracks(fd, args); 6625 dumpEffectChains(fd, args); 6626} 6627 6628void AudioFlinger::RecordThread::dumpInternals(int fd, const Vector<String16>& args) 6629{ 6630 dprintf(fd, "\nInput thread %p:\n", this); 6631 6632 dumpBase(fd, args); 6633 6634 if (mActiveTracks.size() == 0) { 6635 dprintf(fd, " No active record clients\n"); 6636 } 6637 dprintf(fd, " Fast capture thread: %s\n", hasFastCapture() ? "yes" : "no"); 6638 dprintf(fd, " Fast track available: %s\n", mFastTrackAvail ? "yes" : "no"); 6639 6640 // Make a non-atomic copy of fast capture dump state so it won't change underneath us 6641 // while we are dumping it. It may be inconsistent, but it won't mutate! 6642 // This is a large object so we place it on the heap. 6643 // FIXME 25972958: Need an intelligent copy constructor that does not touch unused pages. 6644 const FastCaptureDumpState *copy = new FastCaptureDumpState(mFastCaptureDumpState); 6645 copy->dump(fd); 6646 delete copy; 6647} 6648 6649void AudioFlinger::RecordThread::dumpTracks(int fd, const Vector<String16>& args __unused) 6650{ 6651 const size_t SIZE = 256; 6652 char buffer[SIZE]; 6653 String8 result; 6654 6655 size_t numtracks = mTracks.size(); 6656 size_t numactive = mActiveTracks.size(); 6657 size_t numactiveseen = 0; 6658 dprintf(fd, " %d Tracks", numtracks); 6659 if (numtracks) { 6660 dprintf(fd, " of which %d are active\n", numactive); 6661 RecordTrack::appendDumpHeader(result); 6662 for (size_t i = 0; i < numtracks ; ++i) { 6663 sp<RecordTrack> track = mTracks[i]; 6664 if (track != 0) { 6665 bool active = mActiveTracks.indexOf(track) >= 0; 6666 if (active) { 6667 numactiveseen++; 6668 } 6669 track->dump(buffer, SIZE, active); 6670 result.append(buffer); 6671 } 6672 } 6673 } else { 6674 dprintf(fd, "\n"); 6675 } 6676 6677 if (numactiveseen != numactive) { 6678 snprintf(buffer, SIZE, " The following tracks are in the active list but" 6679 " not in the track list\n"); 6680 result.append(buffer); 6681 RecordTrack::appendDumpHeader(result); 6682 for (size_t i = 0; i < numactive; ++i) { 6683 sp<RecordTrack> track = mActiveTracks[i]; 6684 if (mTracks.indexOf(track) < 0) { 6685 track->dump(buffer, SIZE, true); 6686 result.append(buffer); 6687 } 6688 } 6689 6690 } 6691 write(fd, result.string(), result.size()); 6692} 6693 6694 6695void AudioFlinger::RecordThread::ResamplerBufferProvider::reset() 6696{ 6697 sp<ThreadBase> threadBase = mRecordTrack->mThread.promote(); 6698 RecordThread *recordThread = (RecordThread *) threadBase.get(); 6699 mRsmpInFront = recordThread->mRsmpInRear; 6700 mRsmpInUnrel = 0; 6701} 6702 6703void AudioFlinger::RecordThread::ResamplerBufferProvider::sync( 6704 size_t *framesAvailable, bool *hasOverrun) 6705{ 6706 sp<ThreadBase> threadBase = mRecordTrack->mThread.promote(); 6707 RecordThread *recordThread = (RecordThread *) threadBase.get(); 6708 const int32_t rear = recordThread->mRsmpInRear; 6709 const int32_t front = mRsmpInFront; 6710 const ssize_t filled = rear - front; 6711 6712 size_t framesIn; 6713 bool overrun = false; 6714 if (filled < 0) { 6715 // should not happen, but treat like a massive overrun and re-sync 6716 framesIn = 0; 6717 mRsmpInFront = rear; 6718 overrun = true; 6719 } else if ((size_t) filled <= recordThread->mRsmpInFrames) { 6720 framesIn = (size_t) filled; 6721 } else { 6722 // client is not keeping up with server, but give it latest data 6723 framesIn = recordThread->mRsmpInFrames; 6724 mRsmpInFront = /* front = */ rear - framesIn; 6725 overrun = true; 6726 } 6727 if (framesAvailable != NULL) { 6728 *framesAvailable = framesIn; 6729 } 6730 if (hasOverrun != NULL) { 6731 *hasOverrun = overrun; 6732 } 6733} 6734 6735// AudioBufferProvider interface 6736status_t AudioFlinger::RecordThread::ResamplerBufferProvider::getNextBuffer( 6737 AudioBufferProvider::Buffer* buffer) 6738{ 6739 sp<ThreadBase> threadBase = mRecordTrack->mThread.promote(); 6740 if (threadBase == 0) { 6741 buffer->frameCount = 0; 6742 buffer->raw = NULL; 6743 return NOT_ENOUGH_DATA; 6744 } 6745 RecordThread *recordThread = (RecordThread *) threadBase.get(); 6746 int32_t rear = recordThread->mRsmpInRear; 6747 int32_t front = mRsmpInFront; 6748 ssize_t filled = rear - front; 6749 // FIXME should not be P2 (don't want to increase latency) 6750 // FIXME if client not keeping up, discard 6751 LOG_ALWAYS_FATAL_IF(!(0 <= filled && (size_t) filled <= recordThread->mRsmpInFrames)); 6752 // 'filled' may be non-contiguous, so return only the first contiguous chunk 6753 front &= recordThread->mRsmpInFramesP2 - 1; 6754 size_t part1 = recordThread->mRsmpInFramesP2 - front; 6755 if (part1 > (size_t) filled) { 6756 part1 = filled; 6757 } 6758 size_t ask = buffer->frameCount; 6759 ALOG_ASSERT(ask > 0); 6760 if (part1 > ask) { 6761 part1 = ask; 6762 } 6763 if (part1 == 0) { 6764 // out of data is fine since the resampler will return a short-count. 6765 buffer->raw = NULL; 6766 buffer->frameCount = 0; 6767 mRsmpInUnrel = 0; 6768 return NOT_ENOUGH_DATA; 6769 } 6770 6771 buffer->raw = (uint8_t*)recordThread->mRsmpInBuffer + front * recordThread->mFrameSize; 6772 buffer->frameCount = part1; 6773 mRsmpInUnrel = part1; 6774 return NO_ERROR; 6775} 6776 6777// AudioBufferProvider interface 6778void AudioFlinger::RecordThread::ResamplerBufferProvider::releaseBuffer( 6779 AudioBufferProvider::Buffer* buffer) 6780{ 6781 size_t stepCount = buffer->frameCount; 6782 if (stepCount == 0) { 6783 return; 6784 } 6785 ALOG_ASSERT(stepCount <= mRsmpInUnrel); 6786 mRsmpInUnrel -= stepCount; 6787 mRsmpInFront += stepCount; 6788 buffer->raw = NULL; 6789 buffer->frameCount = 0; 6790} 6791 6792AudioFlinger::RecordThread::RecordBufferConverter::RecordBufferConverter( 6793 audio_channel_mask_t srcChannelMask, audio_format_t srcFormat, 6794 uint32_t srcSampleRate, 6795 audio_channel_mask_t dstChannelMask, audio_format_t dstFormat, 6796 uint32_t dstSampleRate) : 6797 mSrcChannelMask(AUDIO_CHANNEL_INVALID), // updateParameters will set following vars 6798 // mSrcFormat 6799 // mSrcSampleRate 6800 // mDstChannelMask 6801 // mDstFormat 6802 // mDstSampleRate 6803 // mSrcChannelCount 6804 // mDstChannelCount 6805 // mDstFrameSize 6806 mBuf(NULL), mBufFrames(0), mBufFrameSize(0), 6807 mResampler(NULL), 6808 mIsLegacyDownmix(false), 6809 mIsLegacyUpmix(false), 6810 mRequiresFloat(false), 6811 mInputConverterProvider(NULL) 6812{ 6813 (void)updateParameters(srcChannelMask, srcFormat, srcSampleRate, 6814 dstChannelMask, dstFormat, dstSampleRate); 6815} 6816 6817AudioFlinger::RecordThread::RecordBufferConverter::~RecordBufferConverter() { 6818 free(mBuf); 6819 delete mResampler; 6820 delete mInputConverterProvider; 6821} 6822 6823size_t AudioFlinger::RecordThread::RecordBufferConverter::convert(void *dst, 6824 AudioBufferProvider *provider, size_t frames) 6825{ 6826 if (mInputConverterProvider != NULL) { 6827 mInputConverterProvider->setBufferProvider(provider); 6828 provider = mInputConverterProvider; 6829 } 6830 6831 if (mResampler == NULL) { 6832 ALOGVV("NO RESAMPLING sampleRate:%u mSrcFormat:%#x mDstFormat:%#x", 6833 mSrcSampleRate, mSrcFormat, mDstFormat); 6834 6835 AudioBufferProvider::Buffer buffer; 6836 for (size_t i = frames; i > 0; ) { 6837 buffer.frameCount = i; 6838 status_t status = provider->getNextBuffer(&buffer); 6839 if (status != OK || buffer.frameCount == 0) { 6840 frames -= i; // cannot fill request. 6841 break; 6842 } 6843 // format convert to destination buffer 6844 convertNoResampler(dst, buffer.raw, buffer.frameCount); 6845 6846 dst = (int8_t*)dst + buffer.frameCount * mDstFrameSize; 6847 i -= buffer.frameCount; 6848 provider->releaseBuffer(&buffer); 6849 } 6850 } else { 6851 ALOGVV("RESAMPLING mSrcSampleRate:%u mDstSampleRate:%u mSrcFormat:%#x mDstFormat:%#x", 6852 mSrcSampleRate, mDstSampleRate, mSrcFormat, mDstFormat); 6853 6854 // reallocate buffer if needed 6855 if (mBufFrameSize != 0 && mBufFrames < frames) { 6856 free(mBuf); 6857 mBufFrames = frames; 6858 (void)posix_memalign(&mBuf, 32, mBufFrames * mBufFrameSize); 6859 } 6860 // resampler accumulates, but we only have one source track 6861 memset(mBuf, 0, frames * mBufFrameSize); 6862 frames = mResampler->resample((int32_t*)mBuf, frames, provider); 6863 // format convert to destination buffer 6864 convertResampler(dst, mBuf, frames); 6865 } 6866 return frames; 6867} 6868 6869status_t AudioFlinger::RecordThread::RecordBufferConverter::updateParameters( 6870 audio_channel_mask_t srcChannelMask, audio_format_t srcFormat, 6871 uint32_t srcSampleRate, 6872 audio_channel_mask_t dstChannelMask, audio_format_t dstFormat, 6873 uint32_t dstSampleRate) 6874{ 6875 // quick evaluation if there is any change. 6876 if (mSrcFormat == srcFormat 6877 && mSrcChannelMask == srcChannelMask 6878 && mSrcSampleRate == srcSampleRate 6879 && mDstFormat == dstFormat 6880 && mDstChannelMask == dstChannelMask 6881 && mDstSampleRate == dstSampleRate) { 6882 return NO_ERROR; 6883 } 6884 6885 ALOGV("RecordBufferConverter updateParameters srcMask:%#x dstMask:%#x" 6886 " srcFormat:%#x dstFormat:%#x srcRate:%u dstRate:%u", 6887 srcChannelMask, dstChannelMask, srcFormat, dstFormat, srcSampleRate, dstSampleRate); 6888 const bool valid = 6889 audio_is_input_channel(srcChannelMask) 6890 && audio_is_input_channel(dstChannelMask) 6891 && audio_is_valid_format(srcFormat) && audio_is_linear_pcm(srcFormat) 6892 && audio_is_valid_format(dstFormat) && audio_is_linear_pcm(dstFormat) 6893 && (srcSampleRate <= dstSampleRate * AUDIO_RESAMPLER_DOWN_RATIO_MAX) 6894 ; // no upsampling checks for now 6895 if (!valid) { 6896 return BAD_VALUE; 6897 } 6898 6899 mSrcFormat = srcFormat; 6900 mSrcChannelMask = srcChannelMask; 6901 mSrcSampleRate = srcSampleRate; 6902 mDstFormat = dstFormat; 6903 mDstChannelMask = dstChannelMask; 6904 mDstSampleRate = dstSampleRate; 6905 6906 // compute derived parameters 6907 mSrcChannelCount = audio_channel_count_from_in_mask(srcChannelMask); 6908 mDstChannelCount = audio_channel_count_from_in_mask(dstChannelMask); 6909 mDstFrameSize = mDstChannelCount * audio_bytes_per_sample(mDstFormat); 6910 6911 // do we need to resample? 6912 delete mResampler; 6913 mResampler = NULL; 6914 if (mSrcSampleRate != mDstSampleRate) { 6915 mResampler = AudioResampler::create(AUDIO_FORMAT_PCM_FLOAT, 6916 mSrcChannelCount, mDstSampleRate); 6917 mResampler->setSampleRate(mSrcSampleRate); 6918 mResampler->setVolume(AudioMixer::UNITY_GAIN_FLOAT, AudioMixer::UNITY_GAIN_FLOAT); 6919 } 6920 6921 // are we running legacy channel conversion modes? 6922 mIsLegacyDownmix = (mSrcChannelMask == AUDIO_CHANNEL_IN_STEREO 6923 || mSrcChannelMask == AUDIO_CHANNEL_IN_FRONT_BACK) 6924 && mDstChannelMask == AUDIO_CHANNEL_IN_MONO; 6925 mIsLegacyUpmix = mSrcChannelMask == AUDIO_CHANNEL_IN_MONO 6926 && (mDstChannelMask == AUDIO_CHANNEL_IN_STEREO 6927 || mDstChannelMask == AUDIO_CHANNEL_IN_FRONT_BACK); 6928 6929 // do we need to process in float? 6930 mRequiresFloat = mResampler != NULL || mIsLegacyDownmix || mIsLegacyUpmix; 6931 6932 // do we need a staging buffer to convert for destination (we can still optimize this)? 6933 // we use mBufFrameSize > 0 to indicate both frame size as well as buffer necessity 6934 if (mResampler != NULL) { 6935 mBufFrameSize = max(mSrcChannelCount, FCC_2) 6936 * audio_bytes_per_sample(AUDIO_FORMAT_PCM_FLOAT); 6937 } else if (mIsLegacyUpmix || mIsLegacyDownmix) { // legacy modes always float 6938 mBufFrameSize = mDstChannelCount * audio_bytes_per_sample(AUDIO_FORMAT_PCM_FLOAT); 6939 } else if (mSrcChannelMask != mDstChannelMask && mDstFormat != mSrcFormat) { 6940 mBufFrameSize = mDstChannelCount * audio_bytes_per_sample(mSrcFormat); 6941 } else { 6942 mBufFrameSize = 0; 6943 } 6944 mBufFrames = 0; // force the buffer to be resized. 6945 6946 // do we need an input converter buffer provider to give us float? 6947 delete mInputConverterProvider; 6948 mInputConverterProvider = NULL; 6949 if (mRequiresFloat && mSrcFormat != AUDIO_FORMAT_PCM_FLOAT) { 6950 mInputConverterProvider = new ReformatBufferProvider( 6951 audio_channel_count_from_in_mask(mSrcChannelMask), 6952 mSrcFormat, 6953 AUDIO_FORMAT_PCM_FLOAT, 6954 256 /* provider buffer frame count */); 6955 } 6956 6957 // do we need a remixer to do channel mask conversion 6958 if (!mIsLegacyDownmix && !mIsLegacyUpmix && mSrcChannelMask != mDstChannelMask) { 6959 (void) memcpy_by_index_array_initialization_from_channel_mask( 6960 mIdxAry, ARRAY_SIZE(mIdxAry), mDstChannelMask, mSrcChannelMask); 6961 } 6962 return NO_ERROR; 6963} 6964 6965void AudioFlinger::RecordThread::RecordBufferConverter::convertNoResampler( 6966 void *dst, const void *src, size_t frames) 6967{ 6968 // src is native type unless there is legacy upmix or downmix, whereupon it is float. 6969 if (mBufFrameSize != 0 && mBufFrames < frames) { 6970 free(mBuf); 6971 mBufFrames = frames; 6972 (void)posix_memalign(&mBuf, 32, mBufFrames * mBufFrameSize); 6973 } 6974 // do we need to do legacy upmix and downmix? 6975 if (mIsLegacyUpmix || mIsLegacyDownmix) { 6976 void *dstBuf = mBuf != NULL ? mBuf : dst; 6977 if (mIsLegacyUpmix) { 6978 upmix_to_stereo_float_from_mono_float((float *)dstBuf, 6979 (const float *)src, frames); 6980 } else /*mIsLegacyDownmix */ { 6981 downmix_to_mono_float_from_stereo_float((float *)dstBuf, 6982 (const float *)src, frames); 6983 } 6984 if (mBuf != NULL) { 6985 memcpy_by_audio_format(dst, mDstFormat, mBuf, AUDIO_FORMAT_PCM_FLOAT, 6986 frames * mDstChannelCount); 6987 } 6988 return; 6989 } 6990 // do we need to do channel mask conversion? 6991 if (mSrcChannelMask != mDstChannelMask) { 6992 void *dstBuf = mBuf != NULL ? mBuf : dst; 6993 memcpy_by_index_array(dstBuf, mDstChannelCount, 6994 src, mSrcChannelCount, mIdxAry, audio_bytes_per_sample(mSrcFormat), frames); 6995 if (dstBuf == dst) { 6996 return; // format is the same 6997 } 6998 } 6999 // convert to destination buffer 7000 const void *convertBuf = mBuf != NULL ? mBuf : src; 7001 memcpy_by_audio_format(dst, mDstFormat, convertBuf, mSrcFormat, 7002 frames * mDstChannelCount); 7003} 7004 7005void AudioFlinger::RecordThread::RecordBufferConverter::convertResampler( 7006 void *dst, /*not-a-const*/ void *src, size_t frames) 7007{ 7008 // src buffer format is ALWAYS float when entering this routine 7009 if (mIsLegacyUpmix) { 7010 ; // mono to stereo already handled by resampler 7011 } else if (mIsLegacyDownmix 7012 || (mSrcChannelMask == mDstChannelMask && mSrcChannelCount == 1)) { 7013 // the resampler outputs stereo for mono input channel (a feature?) 7014 // must convert to mono 7015 downmix_to_mono_float_from_stereo_float((float *)src, 7016 (const float *)src, frames); 7017 } else if (mSrcChannelMask != mDstChannelMask) { 7018 // convert to mono channel again for channel mask conversion (could be skipped 7019 // with further optimization). 7020 if (mSrcChannelCount == 1) { 7021 downmix_to_mono_float_from_stereo_float((float *)src, 7022 (const float *)src, frames); 7023 } 7024 // convert to destination format (in place, OK as float is larger than other types) 7025 if (mDstFormat != AUDIO_FORMAT_PCM_FLOAT) { 7026 memcpy_by_audio_format(src, mDstFormat, src, AUDIO_FORMAT_PCM_FLOAT, 7027 frames * mSrcChannelCount); 7028 } 7029 // channel convert and save to dst 7030 memcpy_by_index_array(dst, mDstChannelCount, 7031 src, mSrcChannelCount, mIdxAry, audio_bytes_per_sample(mDstFormat), frames); 7032 return; 7033 } 7034 // convert to destination format and save to dst 7035 memcpy_by_audio_format(dst, mDstFormat, src, AUDIO_FORMAT_PCM_FLOAT, 7036 frames * mDstChannelCount); 7037} 7038 7039bool AudioFlinger::RecordThread::checkForNewParameter_l(const String8& keyValuePair, 7040 status_t& status) 7041{ 7042 bool reconfig = false; 7043 7044 status = NO_ERROR; 7045 7046 audio_format_t reqFormat = mFormat; 7047 uint32_t samplingRate = mSampleRate; 7048 // TODO this may change if we want to support capture from HDMI PCM multi channel (e.g on TVs). 7049 audio_channel_mask_t channelMask = audio_channel_in_mask_from_count(mChannelCount); 7050 7051 AudioParameter param = AudioParameter(keyValuePair); 7052 int value; 7053 // TODO Investigate when this code runs. Check with audio policy when a sample rate and 7054 // channel count change can be requested. Do we mandate the first client defines the 7055 // HAL sampling rate and channel count or do we allow changes on the fly? 7056 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 7057 samplingRate = value; 7058 reconfig = true; 7059 } 7060 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 7061 if (!audio_is_linear_pcm((audio_format_t) value)) { 7062 status = BAD_VALUE; 7063 } else { 7064 reqFormat = (audio_format_t) value; 7065 reconfig = true; 7066 } 7067 } 7068 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 7069 audio_channel_mask_t mask = (audio_channel_mask_t) value; 7070 if (!audio_is_input_channel(mask) || 7071 audio_channel_count_from_in_mask(mask) > FCC_8) { 7072 status = BAD_VALUE; 7073 } else { 7074 channelMask = mask; 7075 reconfig = true; 7076 } 7077 } 7078 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 7079 // do not accept frame count changes if tracks are open as the track buffer 7080 // size depends on frame count and correct behavior would not be guaranteed 7081 // if frame count is changed after track creation 7082 if (mActiveTracks.size() > 0) { 7083 status = INVALID_OPERATION; 7084 } else { 7085 reconfig = true; 7086 } 7087 } 7088 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 7089 // forward device change to effects that have requested to be 7090 // aware of attached audio device. 7091 for (size_t i = 0; i < mEffectChains.size(); i++) { 7092 mEffectChains[i]->setDevice_l(value); 7093 } 7094 7095 // store input device and output device but do not forward output device to audio HAL. 7096 // Note that status is ignored by the caller for output device 7097 // (see AudioFlinger::setParameters() 7098 if (audio_is_output_devices(value)) { 7099 mOutDevice = value; 7100 status = BAD_VALUE; 7101 } else { 7102 mInDevice = value; 7103 if (value != AUDIO_DEVICE_NONE) { 7104 mPrevInDevice = value; 7105 } 7106 // disable AEC and NS if the device is a BT SCO headset supporting those 7107 // pre processings 7108 if (mTracks.size() > 0) { 7109 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 7110 mAudioFlinger->btNrecIsOff(); 7111 for (size_t i = 0; i < mTracks.size(); i++) { 7112 sp<RecordTrack> track = mTracks[i]; 7113 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId()); 7114 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId()); 7115 } 7116 } 7117 } 7118 } 7119 if (param.getInt(String8(AudioParameter::keyInputSource), value) == NO_ERROR && 7120 mAudioSource != (audio_source_t)value) { 7121 // forward device change to effects that have requested to be 7122 // aware of attached audio device. 7123 for (size_t i = 0; i < mEffectChains.size(); i++) { 7124 mEffectChains[i]->setAudioSource_l((audio_source_t)value); 7125 } 7126 mAudioSource = (audio_source_t)value; 7127 } 7128 7129 if (status == NO_ERROR) { 7130 status = mInput->stream->common.set_parameters(&mInput->stream->common, 7131 keyValuePair.string()); 7132 if (status == INVALID_OPERATION) { 7133 inputStandBy(); 7134 status = mInput->stream->common.set_parameters(&mInput->stream->common, 7135 keyValuePair.string()); 7136 } 7137 if (reconfig) { 7138 if (status == BAD_VALUE && 7139 audio_is_linear_pcm(mInput->stream->common.get_format(&mInput->stream->common)) && 7140 audio_is_linear_pcm(reqFormat) && 7141 (mInput->stream->common.get_sample_rate(&mInput->stream->common) 7142 <= (AUDIO_RESAMPLER_DOWN_RATIO_MAX * samplingRate)) && 7143 audio_channel_count_from_in_mask( 7144 mInput->stream->common.get_channels(&mInput->stream->common)) <= FCC_8) { 7145 status = NO_ERROR; 7146 } 7147 if (status == NO_ERROR) { 7148 readInputParameters_l(); 7149 sendIoConfigEvent_l(AUDIO_INPUT_CONFIG_CHANGED); 7150 } 7151 } 7152 } 7153 7154 return reconfig; 7155} 7156 7157String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 7158{ 7159 Mutex::Autolock _l(mLock); 7160 if (initCheck() != NO_ERROR) { 7161 return String8(); 7162 } 7163 7164 char *s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 7165 const String8 out_s8(s); 7166 free(s); 7167 return out_s8; 7168} 7169 7170void AudioFlinger::RecordThread::ioConfigChanged(audio_io_config_event event, pid_t pid) { 7171 sp<AudioIoDescriptor> desc = new AudioIoDescriptor(); 7172 7173 desc->mIoHandle = mId; 7174 7175 switch (event) { 7176 case AUDIO_INPUT_OPENED: 7177 case AUDIO_INPUT_CONFIG_CHANGED: 7178 desc->mPatch = mPatch; 7179 desc->mChannelMask = mChannelMask; 7180 desc->mSamplingRate = mSampleRate; 7181 desc->mFormat = mFormat; 7182 desc->mFrameCount = mFrameCount; 7183 desc->mLatency = 0; 7184 break; 7185 7186 case AUDIO_INPUT_CLOSED: 7187 default: 7188 break; 7189 } 7190 mAudioFlinger->ioConfigChanged(event, desc, pid); 7191} 7192 7193void AudioFlinger::RecordThread::readInputParameters_l() 7194{ 7195 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 7196 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 7197 mChannelCount = audio_channel_count_from_in_mask(mChannelMask); 7198 if (mChannelCount > FCC_8) { 7199 ALOGE("HAL channel count %d > %d", mChannelCount, FCC_8); 7200 } 7201 mHALFormat = mInput->stream->common.get_format(&mInput->stream->common); 7202 mFormat = mHALFormat; 7203 if (!audio_is_linear_pcm(mFormat)) { 7204 ALOGE("HAL format %#x is not linear pcm", mFormat); 7205 } 7206 mFrameSize = audio_stream_in_frame_size(mInput->stream); 7207 mBufferSize = mInput->stream->common.get_buffer_size(&mInput->stream->common); 7208 mFrameCount = mBufferSize / mFrameSize; 7209 // This is the formula for calculating the temporary buffer size. 7210 // With 7 HAL buffers, we can guarantee ability to down-sample the input by ratio of 6:1 to 7211 // 1 full output buffer, regardless of the alignment of the available input. 7212 // The value is somewhat arbitrary, and could probably be even larger. 7213 // A larger value should allow more old data to be read after a track calls start(), 7214 // without increasing latency. 7215 // 7216 // Note this is independent of the maximum downsampling ratio permitted for capture. 7217 mRsmpInFrames = mFrameCount * 7; 7218 mRsmpInFramesP2 = roundup(mRsmpInFrames); 7219 free(mRsmpInBuffer); 7220 mRsmpInBuffer = NULL; 7221 7222 // TODO optimize audio capture buffer sizes ... 7223 // Here we calculate the size of the sliding buffer used as a source 7224 // for resampling. mRsmpInFramesP2 is currently roundup(mFrameCount * 7). 7225 // For current HAL frame counts, this is usually 2048 = 40 ms. It would 7226 // be better to have it derived from the pipe depth in the long term. 7227 // The current value is higher than necessary. However it should not add to latency. 7228 7229 // Over-allocate beyond mRsmpInFramesP2 to permit a HAL read past end of buffer 7230 size_t bufferSize = (mRsmpInFramesP2 + mFrameCount - 1) * mFrameSize; 7231 (void)posix_memalign(&mRsmpInBuffer, 32, bufferSize); 7232 memset(mRsmpInBuffer, 0, bufferSize); // if posix_memalign fails, will segv here. 7233 7234 // AudioRecord mSampleRate and mChannelCount are constant due to AudioRecord API constraints. 7235 // But if thread's mSampleRate or mChannelCount changes, how will that affect active tracks? 7236} 7237 7238uint32_t AudioFlinger::RecordThread::getInputFramesLost() 7239{ 7240 Mutex::Autolock _l(mLock); 7241 if (initCheck() != NO_ERROR) { 7242 return 0; 7243 } 7244 7245 return mInput->stream->get_input_frames_lost(mInput->stream); 7246} 7247 7248uint32_t AudioFlinger::RecordThread::hasAudioSession(audio_session_t sessionId) const 7249{ 7250 Mutex::Autolock _l(mLock); 7251 uint32_t result = 0; 7252 if (getEffectChain_l(sessionId) != 0) { 7253 result = EFFECT_SESSION; 7254 } 7255 7256 for (size_t i = 0; i < mTracks.size(); ++i) { 7257 if (sessionId == mTracks[i]->sessionId()) { 7258 result |= TRACK_SESSION; 7259 break; 7260 } 7261 } 7262 7263 return result; 7264} 7265 7266KeyedVector<audio_session_t, bool> AudioFlinger::RecordThread::sessionIds() const 7267{ 7268 KeyedVector<audio_session_t, bool> ids; 7269 Mutex::Autolock _l(mLock); 7270 for (size_t j = 0; j < mTracks.size(); ++j) { 7271 sp<RecordThread::RecordTrack> track = mTracks[j]; 7272 audio_session_t sessionId = track->sessionId(); 7273 if (ids.indexOfKey(sessionId) < 0) { 7274 ids.add(sessionId, true); 7275 } 7276 } 7277 return ids; 7278} 7279 7280AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 7281{ 7282 Mutex::Autolock _l(mLock); 7283 AudioStreamIn *input = mInput; 7284 mInput = NULL; 7285 return input; 7286} 7287 7288// this method must always be called either with ThreadBase mLock held or inside the thread loop 7289audio_stream_t* AudioFlinger::RecordThread::stream() const 7290{ 7291 if (mInput == NULL) { 7292 return NULL; 7293 } 7294 return &mInput->stream->common; 7295} 7296 7297status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 7298{ 7299 // only one chain per input thread 7300 if (mEffectChains.size() != 0) { 7301 ALOGW("addEffectChain_l() already one chain %p on thread %p", chain.get(), this); 7302 return INVALID_OPERATION; 7303 } 7304 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 7305 chain->setThread(this); 7306 chain->setInBuffer(NULL); 7307 chain->setOutBuffer(NULL); 7308 7309 checkSuspendOnAddEffectChain_l(chain); 7310 7311 // make sure enabled pre processing effects state is communicated to the HAL as we 7312 // just moved them to a new input stream. 7313 chain->syncHalEffectsState(); 7314 7315 mEffectChains.add(chain); 7316 7317 return NO_ERROR; 7318} 7319 7320size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 7321{ 7322 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 7323 ALOGW_IF(mEffectChains.size() != 1, 7324 "removeEffectChain_l() %p invalid chain size %d on thread %p", 7325 chain.get(), mEffectChains.size(), this); 7326 if (mEffectChains.size() == 1) { 7327 mEffectChains.removeAt(0); 7328 } 7329 return 0; 7330} 7331 7332status_t AudioFlinger::RecordThread::createAudioPatch_l(const struct audio_patch *patch, 7333 audio_patch_handle_t *handle) 7334{ 7335 status_t status = NO_ERROR; 7336 7337 // store new device and send to effects 7338 mInDevice = patch->sources[0].ext.device.type; 7339 mPatch = *patch; 7340 for (size_t i = 0; i < mEffectChains.size(); i++) { 7341 mEffectChains[i]->setDevice_l(mInDevice); 7342 } 7343 7344 // disable AEC and NS if the device is a BT SCO headset supporting those 7345 // pre processings 7346 if (mTracks.size() > 0) { 7347 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 7348 mAudioFlinger->btNrecIsOff(); 7349 for (size_t i = 0; i < mTracks.size(); i++) { 7350 sp<RecordTrack> track = mTracks[i]; 7351 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId()); 7352 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId()); 7353 } 7354 } 7355 7356 // store new source and send to effects 7357 if (mAudioSource != patch->sinks[0].ext.mix.usecase.source) { 7358 mAudioSource = patch->sinks[0].ext.mix.usecase.source; 7359 for (size_t i = 0; i < mEffectChains.size(); i++) { 7360 mEffectChains[i]->setAudioSource_l(mAudioSource); 7361 } 7362 } 7363 7364 if (mInput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 7365 audio_hw_device_t *hwDevice = mInput->audioHwDev->hwDevice(); 7366 status = hwDevice->create_audio_patch(hwDevice, 7367 patch->num_sources, 7368 patch->sources, 7369 patch->num_sinks, 7370 patch->sinks, 7371 handle); 7372 } else { 7373 char *address; 7374 if (strcmp(patch->sources[0].ext.device.address, "") != 0) { 7375 address = audio_device_address_to_parameter( 7376 patch->sources[0].ext.device.type, 7377 patch->sources[0].ext.device.address); 7378 } else { 7379 address = (char *)calloc(1, 1); 7380 } 7381 AudioParameter param = AudioParameter(String8(address)); 7382 free(address); 7383 param.addInt(String8(AUDIO_PARAMETER_STREAM_ROUTING), 7384 (int)patch->sources[0].ext.device.type); 7385 param.addInt(String8(AUDIO_PARAMETER_STREAM_INPUT_SOURCE), 7386 (int)patch->sinks[0].ext.mix.usecase.source); 7387 status = mInput->stream->common.set_parameters(&mInput->stream->common, 7388 param.toString().string()); 7389 *handle = AUDIO_PATCH_HANDLE_NONE; 7390 } 7391 7392 if (mInDevice != mPrevInDevice) { 7393 sendIoConfigEvent_l(AUDIO_INPUT_CONFIG_CHANGED); 7394 mPrevInDevice = mInDevice; 7395 } 7396 7397 return status; 7398} 7399 7400status_t AudioFlinger::RecordThread::releaseAudioPatch_l(const audio_patch_handle_t handle) 7401{ 7402 status_t status = NO_ERROR; 7403 7404 mInDevice = AUDIO_DEVICE_NONE; 7405 7406 if (mInput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 7407 audio_hw_device_t *hwDevice = mInput->audioHwDev->hwDevice(); 7408 status = hwDevice->release_audio_patch(hwDevice, handle); 7409 } else { 7410 AudioParameter param; 7411 param.addInt(String8(AUDIO_PARAMETER_STREAM_ROUTING), 0); 7412 status = mInput->stream->common.set_parameters(&mInput->stream->common, 7413 param.toString().string()); 7414 } 7415 return status; 7416} 7417 7418void AudioFlinger::RecordThread::addPatchRecord(const sp<PatchRecord>& record) 7419{ 7420 Mutex::Autolock _l(mLock); 7421 mTracks.add(record); 7422} 7423 7424void AudioFlinger::RecordThread::deletePatchRecord(const sp<PatchRecord>& record) 7425{ 7426 Mutex::Autolock _l(mLock); 7427 destroyTrack_l(record); 7428} 7429 7430void AudioFlinger::RecordThread::getAudioPortConfig(struct audio_port_config *config) 7431{ 7432 ThreadBase::getAudioPortConfig(config); 7433 config->role = AUDIO_PORT_ROLE_SINK; 7434 config->ext.mix.hw_module = mInput->audioHwDev->handle(); 7435 config->ext.mix.usecase.source = mAudioSource; 7436} 7437 7438} // namespace android 7439