Threads.cpp revision 57fc788e8c924823c9026f1239282d39433da821
1/* 2** 3** Copyright 2012, The Android Open Source Project 4** 5** Licensed under the Apache License, Version 2.0 (the "License"); 6** you may not use this file except in compliance with the License. 7** You may obtain a copy of the License at 8** 9** http://www.apache.org/licenses/LICENSE-2.0 10** 11** Unless required by applicable law or agreed to in writing, software 12** distributed under the License is distributed on an "AS IS" BASIS, 13** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. 14** See the License for the specific language governing permissions and 15** limitations under the License. 16*/ 17 18 19#define LOG_TAG "AudioFlinger" 20//#define LOG_NDEBUG 0 21#define ATRACE_TAG ATRACE_TAG_AUDIO 22 23#include "Configuration.h" 24#include <math.h> 25#include <fcntl.h> 26#include <linux/futex.h> 27#include <sys/stat.h> 28#include <sys/syscall.h> 29#include <cutils/properties.h> 30#include <media/AudioParameter.h> 31#include <media/AudioResamplerPublic.h> 32#include <utils/Log.h> 33#include <utils/Trace.h> 34 35#include <private/media/AudioTrackShared.h> 36#include <hardware/audio.h> 37#include <audio_effects/effect_ns.h> 38#include <audio_effects/effect_aec.h> 39#include <audio_utils/conversion.h> 40#include <audio_utils/primitives.h> 41#include <audio_utils/format.h> 42#include <audio_utils/minifloat.h> 43 44// NBAIO implementations 45#include <media/nbaio/AudioStreamInSource.h> 46#include <media/nbaio/AudioStreamOutSink.h> 47#include <media/nbaio/MonoPipe.h> 48#include <media/nbaio/MonoPipeReader.h> 49#include <media/nbaio/Pipe.h> 50#include <media/nbaio/PipeReader.h> 51#include <media/nbaio/SourceAudioBufferProvider.h> 52#include <mediautils/BatteryNotifier.h> 53 54#include <powermanager/PowerManager.h> 55 56#include "AudioFlinger.h" 57#include "AudioMixer.h" 58#include "BufferProviders.h" 59#include "FastMixer.h" 60#include "FastCapture.h" 61#include "ServiceUtilities.h" 62#include "mediautils/SchedulingPolicyService.h" 63 64#ifdef ADD_BATTERY_DATA 65#include <media/IMediaPlayerService.h> 66#include <media/IMediaDeathNotifier.h> 67#endif 68 69#ifdef DEBUG_CPU_USAGE 70#include <cpustats/CentralTendencyStatistics.h> 71#include <cpustats/ThreadCpuUsage.h> 72#endif 73 74#include "AutoPark.h" 75 76// ---------------------------------------------------------------------------- 77 78// Note: the following macro is used for extremely verbose logging message. In 79// order to run with ALOG_ASSERT turned on, we need to have LOG_NDEBUG set to 80// 0; but one side effect of this is to turn all LOGV's as well. Some messages 81// are so verbose that we want to suppress them even when we have ALOG_ASSERT 82// turned on. Do not uncomment the #def below unless you really know what you 83// are doing and want to see all of the extremely verbose messages. 84//#define VERY_VERY_VERBOSE_LOGGING 85#ifdef VERY_VERY_VERBOSE_LOGGING 86#define ALOGVV ALOGV 87#else 88#define ALOGVV(a...) do { } while(0) 89#endif 90 91// TODO: Move these macro/inlines to a header file. 92#define max(a, b) ((a) > (b) ? (a) : (b)) 93template <typename T> 94static inline T min(const T& a, const T& b) 95{ 96 return a < b ? a : b; 97} 98 99#ifndef ARRAY_SIZE 100#define ARRAY_SIZE(a) (sizeof(a) / sizeof(a[0])) 101#endif 102 103namespace android { 104 105// retry counts for buffer fill timeout 106// 50 * ~20msecs = 1 second 107static const int8_t kMaxTrackRetries = 50; 108static const int8_t kMaxTrackStartupRetries = 50; 109// allow less retry attempts on direct output thread. 110// direct outputs can be a scarce resource in audio hardware and should 111// be released as quickly as possible. 112static const int8_t kMaxTrackRetriesDirect = 2; 113 114 115 116// don't warn about blocked writes or record buffer overflows more often than this 117static const nsecs_t kWarningThrottleNs = seconds(5); 118 119// RecordThread loop sleep time upon application overrun or audio HAL read error 120static const int kRecordThreadSleepUs = 5000; 121 122// maximum time to wait in sendConfigEvent_l() for a status to be received 123static const nsecs_t kConfigEventTimeoutNs = seconds(2); 124 125// minimum sleep time for the mixer thread loop when tracks are active but in underrun 126static const uint32_t kMinThreadSleepTimeUs = 5000; 127// maximum divider applied to the active sleep time in the mixer thread loop 128static const uint32_t kMaxThreadSleepTimeShift = 2; 129 130// minimum normal sink buffer size, expressed in milliseconds rather than frames 131// FIXME This should be based on experimentally observed scheduling jitter 132static const uint32_t kMinNormalSinkBufferSizeMs = 20; 133// maximum normal sink buffer size 134static const uint32_t kMaxNormalSinkBufferSizeMs = 24; 135 136// minimum capture buffer size in milliseconds to _not_ need a fast capture thread 137// FIXME This should be based on experimentally observed scheduling jitter 138static const uint32_t kMinNormalCaptureBufferSizeMs = 12; 139 140// Offloaded output thread standby delay: allows track transition without going to standby 141static const nsecs_t kOffloadStandbyDelayNs = seconds(1); 142 143// Direct output thread minimum sleep time in idle or active(underrun) state 144static const nsecs_t kDirectMinSleepTimeUs = 10000; 145 146 147// Whether to use fast mixer 148static const enum { 149 FastMixer_Never, // never initialize or use: for debugging only 150 FastMixer_Always, // always initialize and use, even if not needed: for debugging only 151 // normal mixer multiplier is 1 152 FastMixer_Static, // initialize if needed, then use all the time if initialized, 153 // multiplier is calculated based on min & max normal mixer buffer size 154 FastMixer_Dynamic, // initialize if needed, then use dynamically depending on track load, 155 // multiplier is calculated based on min & max normal mixer buffer size 156 // FIXME for FastMixer_Dynamic: 157 // Supporting this option will require fixing HALs that can't handle large writes. 158 // For example, one HAL implementation returns an error from a large write, 159 // and another HAL implementation corrupts memory, possibly in the sample rate converter. 160 // We could either fix the HAL implementations, or provide a wrapper that breaks 161 // up large writes into smaller ones, and the wrapper would need to deal with scheduler. 162} kUseFastMixer = FastMixer_Static; 163 164// Whether to use fast capture 165static const enum { 166 FastCapture_Never, // never initialize or use: for debugging only 167 FastCapture_Always, // always initialize and use, even if not needed: for debugging only 168 FastCapture_Static, // initialize if needed, then use all the time if initialized 169} kUseFastCapture = FastCapture_Static; 170 171// Priorities for requestPriority 172static const int kPriorityAudioApp = 2; 173static const int kPriorityFastMixer = 3; 174static const int kPriorityFastCapture = 3; 175 176// IAudioFlinger::createTrack() has an in/out parameter 'pFrameCount' for the total size of the 177// track buffer in shared memory. Zero on input means to use a default value. For fast tracks, 178// AudioFlinger derives the default from HAL buffer size and 'fast track multiplier'. 179 180// This is the default value, if not specified by property. 181static const int kFastTrackMultiplier = 2; 182 183// The minimum and maximum allowed values 184static const int kFastTrackMultiplierMin = 1; 185static const int kFastTrackMultiplierMax = 2; 186 187// The actual value to use, which can be specified per-device via property af.fast_track_multiplier. 188static int sFastTrackMultiplier = kFastTrackMultiplier; 189 190// See Thread::readOnlyHeap(). 191// Initially this heap is used to allocate client buffers for "fast" AudioRecord. 192// Eventually it will be the single buffer that FastCapture writes into via HAL read(), 193// and that all "fast" AudioRecord clients read from. In either case, the size can be small. 194static const size_t kRecordThreadReadOnlyHeapSize = 0x2000; 195 196// ---------------------------------------------------------------------------- 197 198static pthread_once_t sFastTrackMultiplierOnce = PTHREAD_ONCE_INIT; 199 200static void sFastTrackMultiplierInit() 201{ 202 char value[PROPERTY_VALUE_MAX]; 203 if (property_get("af.fast_track_multiplier", value, NULL) > 0) { 204 char *endptr; 205 unsigned long ul = strtoul(value, &endptr, 0); 206 if (*endptr == '\0' && kFastTrackMultiplierMin <= ul && ul <= kFastTrackMultiplierMax) { 207 sFastTrackMultiplier = (int) ul; 208 } 209 } 210} 211 212// ---------------------------------------------------------------------------- 213 214#ifdef ADD_BATTERY_DATA 215// To collect the amplifier usage 216static void addBatteryData(uint32_t params) { 217 sp<IMediaPlayerService> service = IMediaDeathNotifier::getMediaPlayerService(); 218 if (service == NULL) { 219 // it already logged 220 return; 221 } 222 223 service->addBatteryData(params); 224} 225#endif 226 227// Track the CLOCK_BOOTTIME versus CLOCK_MONOTONIC timebase offset 228struct { 229 // call when you acquire a partial wakelock 230 void acquire(const sp<IBinder> &wakeLockToken) { 231 pthread_mutex_lock(&mLock); 232 if (wakeLockToken.get() == nullptr) { 233 adjustTimebaseOffset(&mBoottimeOffset, ExtendedTimestamp::TIMEBASE_BOOTTIME); 234 } else { 235 if (mCount == 0) { 236 adjustTimebaseOffset(&mBoottimeOffset, ExtendedTimestamp::TIMEBASE_BOOTTIME); 237 } 238 ++mCount; 239 } 240 pthread_mutex_unlock(&mLock); 241 } 242 243 // call when you release a partial wakelock. 244 void release(const sp<IBinder> &wakeLockToken) { 245 if (wakeLockToken.get() == nullptr) { 246 return; 247 } 248 pthread_mutex_lock(&mLock); 249 if (--mCount < 0) { 250 ALOGE("negative wakelock count"); 251 mCount = 0; 252 } 253 pthread_mutex_unlock(&mLock); 254 } 255 256 // retrieves the boottime timebase offset from monotonic. 257 int64_t getBoottimeOffset() { 258 pthread_mutex_lock(&mLock); 259 int64_t boottimeOffset = mBoottimeOffset; 260 pthread_mutex_unlock(&mLock); 261 return boottimeOffset; 262 } 263 264 // Adjusts the timebase offset between TIMEBASE_MONOTONIC 265 // and the selected timebase. 266 // Currently only TIMEBASE_BOOTTIME is allowed. 267 // 268 // This only needs to be called upon acquiring the first partial wakelock 269 // after all other partial wakelocks are released. 270 // 271 // We do an empirical measurement of the offset rather than parsing 272 // /proc/timer_list since the latter is not a formal kernel ABI. 273 static void adjustTimebaseOffset(int64_t *offset, ExtendedTimestamp::Timebase timebase) { 274 int clockbase; 275 switch (timebase) { 276 case ExtendedTimestamp::TIMEBASE_BOOTTIME: 277 clockbase = SYSTEM_TIME_BOOTTIME; 278 break; 279 default: 280 LOG_ALWAYS_FATAL("invalid timebase %d", timebase); 281 break; 282 } 283 // try three times to get the clock offset, choose the one 284 // with the minimum gap in measurements. 285 const int tries = 3; 286 nsecs_t bestGap, measured; 287 for (int i = 0; i < tries; ++i) { 288 const nsecs_t tmono = systemTime(SYSTEM_TIME_MONOTONIC); 289 const nsecs_t tbase = systemTime(clockbase); 290 const nsecs_t tmono2 = systemTime(SYSTEM_TIME_MONOTONIC); 291 const nsecs_t gap = tmono2 - tmono; 292 if (i == 0 || gap < bestGap) { 293 bestGap = gap; 294 measured = tbase - ((tmono + tmono2) >> 1); 295 } 296 } 297 298 // to avoid micro-adjusting, we don't change the timebase 299 // unless it is significantly different. 300 // 301 // Assumption: It probably takes more than toleranceNs to 302 // suspend and resume the device. 303 static int64_t toleranceNs = 10000; // 10 us 304 if (llabs(*offset - measured) > toleranceNs) { 305 ALOGV("Adjusting timebase offset old: %lld new: %lld", 306 (long long)*offset, (long long)measured); 307 *offset = measured; 308 } 309 } 310 311 pthread_mutex_t mLock; 312 int32_t mCount; 313 int64_t mBoottimeOffset; 314} gBoottime = { PTHREAD_MUTEX_INITIALIZER, 0, 0 }; // static, so use POD initialization 315 316// ---------------------------------------------------------------------------- 317// CPU Stats 318// ---------------------------------------------------------------------------- 319 320class CpuStats { 321public: 322 CpuStats(); 323 void sample(const String8 &title); 324#ifdef DEBUG_CPU_USAGE 325private: 326 ThreadCpuUsage mCpuUsage; // instantaneous thread CPU usage in wall clock ns 327 CentralTendencyStatistics mWcStats; // statistics on thread CPU usage in wall clock ns 328 329 CentralTendencyStatistics mHzStats; // statistics on thread CPU usage in cycles 330 331 int mCpuNum; // thread's current CPU number 332 int mCpukHz; // frequency of thread's current CPU in kHz 333#endif 334}; 335 336CpuStats::CpuStats() 337#ifdef DEBUG_CPU_USAGE 338 : mCpuNum(-1), mCpukHz(-1) 339#endif 340{ 341} 342 343void CpuStats::sample(const String8 &title 344#ifndef DEBUG_CPU_USAGE 345 __unused 346#endif 347 ) { 348#ifdef DEBUG_CPU_USAGE 349 // get current thread's delta CPU time in wall clock ns 350 double wcNs; 351 bool valid = mCpuUsage.sampleAndEnable(wcNs); 352 353 // record sample for wall clock statistics 354 if (valid) { 355 mWcStats.sample(wcNs); 356 } 357 358 // get the current CPU number 359 int cpuNum = sched_getcpu(); 360 361 // get the current CPU frequency in kHz 362 int cpukHz = mCpuUsage.getCpukHz(cpuNum); 363 364 // check if either CPU number or frequency changed 365 if (cpuNum != mCpuNum || cpukHz != mCpukHz) { 366 mCpuNum = cpuNum; 367 mCpukHz = cpukHz; 368 // ignore sample for purposes of cycles 369 valid = false; 370 } 371 372 // if no change in CPU number or frequency, then record sample for cycle statistics 373 if (valid && mCpukHz > 0) { 374 double cycles = wcNs * cpukHz * 0.000001; 375 mHzStats.sample(cycles); 376 } 377 378 unsigned n = mWcStats.n(); 379 // mCpuUsage.elapsed() is expensive, so don't call it every loop 380 if ((n & 127) == 1) { 381 long long elapsed = mCpuUsage.elapsed(); 382 if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) { 383 double perLoop = elapsed / (double) n; 384 double perLoop100 = perLoop * 0.01; 385 double perLoop1k = perLoop * 0.001; 386 double mean = mWcStats.mean(); 387 double stddev = mWcStats.stddev(); 388 double minimum = mWcStats.minimum(); 389 double maximum = mWcStats.maximum(); 390 double meanCycles = mHzStats.mean(); 391 double stddevCycles = mHzStats.stddev(); 392 double minCycles = mHzStats.minimum(); 393 double maxCycles = mHzStats.maximum(); 394 mCpuUsage.resetElapsed(); 395 mWcStats.reset(); 396 mHzStats.reset(); 397 ALOGD("CPU usage for %s over past %.1f secs\n" 398 " (%u mixer loops at %.1f mean ms per loop):\n" 399 " us per mix loop: mean=%.0f stddev=%.0f min=%.0f max=%.0f\n" 400 " %% of wall: mean=%.1f stddev=%.1f min=%.1f max=%.1f\n" 401 " MHz: mean=%.1f, stddev=%.1f, min=%.1f max=%.1f", 402 title.string(), 403 elapsed * .000000001, n, perLoop * .000001, 404 mean * .001, 405 stddev * .001, 406 minimum * .001, 407 maximum * .001, 408 mean / perLoop100, 409 stddev / perLoop100, 410 minimum / perLoop100, 411 maximum / perLoop100, 412 meanCycles / perLoop1k, 413 stddevCycles / perLoop1k, 414 minCycles / perLoop1k, 415 maxCycles / perLoop1k); 416 417 } 418 } 419#endif 420}; 421 422// ---------------------------------------------------------------------------- 423// ThreadBase 424// ---------------------------------------------------------------------------- 425 426// static 427const char *AudioFlinger::ThreadBase::threadTypeToString(AudioFlinger::ThreadBase::type_t type) 428{ 429 switch (type) { 430 case MIXER: 431 return "MIXER"; 432 case DIRECT: 433 return "DIRECT"; 434 case DUPLICATING: 435 return "DUPLICATING"; 436 case RECORD: 437 return "RECORD"; 438 case OFFLOAD: 439 return "OFFLOAD"; 440 default: 441 return "unknown"; 442 } 443} 444 445String8 devicesToString(audio_devices_t devices) 446{ 447 static const struct mapping { 448 audio_devices_t mDevices; 449 const char * mString; 450 } mappingsOut[] = { 451 {AUDIO_DEVICE_OUT_EARPIECE, "EARPIECE"}, 452 {AUDIO_DEVICE_OUT_SPEAKER, "SPEAKER"}, 453 {AUDIO_DEVICE_OUT_WIRED_HEADSET, "WIRED_HEADSET"}, 454 {AUDIO_DEVICE_OUT_WIRED_HEADPHONE, "WIRED_HEADPHONE"}, 455 {AUDIO_DEVICE_OUT_BLUETOOTH_SCO, "BLUETOOTH_SCO"}, 456 {AUDIO_DEVICE_OUT_BLUETOOTH_SCO_HEADSET, "BLUETOOTH_SCO_HEADSET"}, 457 {AUDIO_DEVICE_OUT_BLUETOOTH_SCO_CARKIT, "BLUETOOTH_SCO_CARKIT"}, 458 {AUDIO_DEVICE_OUT_BLUETOOTH_A2DP, "BLUETOOTH_A2DP"}, 459 {AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES,"BLUETOOTH_A2DP_HEADPHONES"}, 460 {AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER, "BLUETOOTH_A2DP_SPEAKER"}, 461 {AUDIO_DEVICE_OUT_AUX_DIGITAL, "AUX_DIGITAL"}, 462 {AUDIO_DEVICE_OUT_HDMI, "HDMI"}, 463 {AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET,"ANLG_DOCK_HEADSET"}, 464 {AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET,"DGTL_DOCK_HEADSET"}, 465 {AUDIO_DEVICE_OUT_USB_ACCESSORY, "USB_ACCESSORY"}, 466 {AUDIO_DEVICE_OUT_USB_DEVICE, "USB_DEVICE"}, 467 {AUDIO_DEVICE_OUT_TELEPHONY_TX, "TELEPHONY_TX"}, 468 {AUDIO_DEVICE_OUT_LINE, "LINE"}, 469 {AUDIO_DEVICE_OUT_HDMI_ARC, "HDMI_ARC"}, 470 {AUDIO_DEVICE_OUT_SPDIF, "SPDIF"}, 471 {AUDIO_DEVICE_OUT_FM, "FM"}, 472 {AUDIO_DEVICE_OUT_AUX_LINE, "AUX_LINE"}, 473 {AUDIO_DEVICE_OUT_SPEAKER_SAFE, "SPEAKER_SAFE"}, 474 {AUDIO_DEVICE_OUT_IP, "IP"}, 475 {AUDIO_DEVICE_OUT_BUS, "BUS"}, 476 {AUDIO_DEVICE_NONE, "NONE"}, // must be last 477 }, mappingsIn[] = { 478 {AUDIO_DEVICE_IN_COMMUNICATION, "COMMUNICATION"}, 479 {AUDIO_DEVICE_IN_AMBIENT, "AMBIENT"}, 480 {AUDIO_DEVICE_IN_BUILTIN_MIC, "BUILTIN_MIC"}, 481 {AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET, "BLUETOOTH_SCO_HEADSET"}, 482 {AUDIO_DEVICE_IN_WIRED_HEADSET, "WIRED_HEADSET"}, 483 {AUDIO_DEVICE_IN_AUX_DIGITAL, "AUX_DIGITAL"}, 484 {AUDIO_DEVICE_IN_VOICE_CALL, "VOICE_CALL"}, 485 {AUDIO_DEVICE_IN_TELEPHONY_RX, "TELEPHONY_RX"}, 486 {AUDIO_DEVICE_IN_BACK_MIC, "BACK_MIC"}, 487 {AUDIO_DEVICE_IN_REMOTE_SUBMIX, "REMOTE_SUBMIX"}, 488 {AUDIO_DEVICE_IN_ANLG_DOCK_HEADSET, "ANLG_DOCK_HEADSET"}, 489 {AUDIO_DEVICE_IN_DGTL_DOCK_HEADSET, "DGTL_DOCK_HEADSET"}, 490 {AUDIO_DEVICE_IN_USB_ACCESSORY, "USB_ACCESSORY"}, 491 {AUDIO_DEVICE_IN_USB_DEVICE, "USB_DEVICE"}, 492 {AUDIO_DEVICE_IN_FM_TUNER, "FM_TUNER"}, 493 {AUDIO_DEVICE_IN_TV_TUNER, "TV_TUNER"}, 494 {AUDIO_DEVICE_IN_LINE, "LINE"}, 495 {AUDIO_DEVICE_IN_SPDIF, "SPDIF"}, 496 {AUDIO_DEVICE_IN_BLUETOOTH_A2DP, "BLUETOOTH_A2DP"}, 497 {AUDIO_DEVICE_IN_LOOPBACK, "LOOPBACK"}, 498 {AUDIO_DEVICE_IN_IP, "IP"}, 499 {AUDIO_DEVICE_IN_BUS, "BUS"}, 500 {AUDIO_DEVICE_NONE, "NONE"}, // must be last 501 }; 502 String8 result; 503 audio_devices_t allDevices = AUDIO_DEVICE_NONE; 504 const mapping *entry; 505 if (devices & AUDIO_DEVICE_BIT_IN) { 506 devices &= ~AUDIO_DEVICE_BIT_IN; 507 entry = mappingsIn; 508 } else { 509 entry = mappingsOut; 510 } 511 for ( ; entry->mDevices != AUDIO_DEVICE_NONE; entry++) { 512 allDevices = (audio_devices_t) (allDevices | entry->mDevices); 513 if (devices & entry->mDevices) { 514 if (!result.isEmpty()) { 515 result.append("|"); 516 } 517 result.append(entry->mString); 518 } 519 } 520 if (devices & ~allDevices) { 521 if (!result.isEmpty()) { 522 result.append("|"); 523 } 524 result.appendFormat("0x%X", devices & ~allDevices); 525 } 526 if (result.isEmpty()) { 527 result.append(entry->mString); 528 } 529 return result; 530} 531 532String8 inputFlagsToString(audio_input_flags_t flags) 533{ 534 static const struct mapping { 535 audio_input_flags_t mFlag; 536 const char * mString; 537 } mappings[] = { 538 {AUDIO_INPUT_FLAG_FAST, "FAST"}, 539 {AUDIO_INPUT_FLAG_HW_HOTWORD, "HW_HOTWORD"}, 540 {AUDIO_INPUT_FLAG_RAW, "RAW"}, 541 {AUDIO_INPUT_FLAG_SYNC, "SYNC"}, 542 {AUDIO_INPUT_FLAG_NONE, "NONE"}, // must be last 543 }; 544 String8 result; 545 audio_input_flags_t allFlags = AUDIO_INPUT_FLAG_NONE; 546 const mapping *entry; 547 for (entry = mappings; entry->mFlag != AUDIO_INPUT_FLAG_NONE; entry++) { 548 allFlags = (audio_input_flags_t) (allFlags | entry->mFlag); 549 if (flags & entry->mFlag) { 550 if (!result.isEmpty()) { 551 result.append("|"); 552 } 553 result.append(entry->mString); 554 } 555 } 556 if (flags & ~allFlags) { 557 if (!result.isEmpty()) { 558 result.append("|"); 559 } 560 result.appendFormat("0x%X", flags & ~allFlags); 561 } 562 if (result.isEmpty()) { 563 result.append(entry->mString); 564 } 565 return result; 566} 567 568String8 outputFlagsToString(audio_output_flags_t flags) 569{ 570 static const struct mapping { 571 audio_output_flags_t mFlag; 572 const char * mString; 573 } mappings[] = { 574 {AUDIO_OUTPUT_FLAG_DIRECT, "DIRECT"}, 575 {AUDIO_OUTPUT_FLAG_PRIMARY, "PRIMARY"}, 576 {AUDIO_OUTPUT_FLAG_FAST, "FAST"}, 577 {AUDIO_OUTPUT_FLAG_DEEP_BUFFER, "DEEP_BUFFER"}, 578 {AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD,"COMPRESS_OFFLOAD"}, 579 {AUDIO_OUTPUT_FLAG_NON_BLOCKING, "NON_BLOCKING"}, 580 {AUDIO_OUTPUT_FLAG_HW_AV_SYNC, "HW_AV_SYNC"}, 581 {AUDIO_OUTPUT_FLAG_RAW, "RAW"}, 582 {AUDIO_OUTPUT_FLAG_SYNC, "SYNC"}, 583 {AUDIO_OUTPUT_FLAG_IEC958_NONAUDIO, "IEC958_NONAUDIO"}, 584 {AUDIO_OUTPUT_FLAG_NONE, "NONE"}, // must be last 585 }; 586 String8 result; 587 audio_output_flags_t allFlags = AUDIO_OUTPUT_FLAG_NONE; 588 const mapping *entry; 589 for (entry = mappings; entry->mFlag != AUDIO_OUTPUT_FLAG_NONE; entry++) { 590 allFlags = (audio_output_flags_t) (allFlags | entry->mFlag); 591 if (flags & entry->mFlag) { 592 if (!result.isEmpty()) { 593 result.append("|"); 594 } 595 result.append(entry->mString); 596 } 597 } 598 if (flags & ~allFlags) { 599 if (!result.isEmpty()) { 600 result.append("|"); 601 } 602 result.appendFormat("0x%X", flags & ~allFlags); 603 } 604 if (result.isEmpty()) { 605 result.append(entry->mString); 606 } 607 return result; 608} 609 610const char *sourceToString(audio_source_t source) 611{ 612 switch (source) { 613 case AUDIO_SOURCE_DEFAULT: return "default"; 614 case AUDIO_SOURCE_MIC: return "mic"; 615 case AUDIO_SOURCE_VOICE_UPLINK: return "voice uplink"; 616 case AUDIO_SOURCE_VOICE_DOWNLINK: return "voice downlink"; 617 case AUDIO_SOURCE_VOICE_CALL: return "voice call"; 618 case AUDIO_SOURCE_CAMCORDER: return "camcorder"; 619 case AUDIO_SOURCE_VOICE_RECOGNITION: return "voice recognition"; 620 case AUDIO_SOURCE_VOICE_COMMUNICATION: return "voice communication"; 621 case AUDIO_SOURCE_REMOTE_SUBMIX: return "remote submix"; 622 case AUDIO_SOURCE_UNPROCESSED: return "unprocessed"; 623 case AUDIO_SOURCE_FM_TUNER: return "FM tuner"; 624 case AUDIO_SOURCE_HOTWORD: return "hotword"; 625 default: return "unknown"; 626 } 627} 628 629AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, 630 audio_devices_t outDevice, audio_devices_t inDevice, type_t type, bool systemReady) 631 : Thread(false /*canCallJava*/), 632 mType(type), 633 mAudioFlinger(audioFlinger), 634 // mSampleRate, mFrameCount, mChannelMask, mChannelCount, mFrameSize, mFormat, mBufferSize 635 // are set by PlaybackThread::readOutputParameters_l() or 636 // RecordThread::readInputParameters_l() 637 //FIXME: mStandby should be true here. Is this some kind of hack? 638 mStandby(false), mOutDevice(outDevice), mInDevice(inDevice), 639 mPrevOutDevice(AUDIO_DEVICE_NONE), mPrevInDevice(AUDIO_DEVICE_NONE), 640 mAudioSource(AUDIO_SOURCE_DEFAULT), mId(id), 641 // mName will be set by concrete (non-virtual) subclass 642 mDeathRecipient(new PMDeathRecipient(this)), 643 mSystemReady(systemReady), 644 mNotifiedBatteryStart(false) 645{ 646 memset(&mPatch, 0, sizeof(struct audio_patch)); 647} 648 649AudioFlinger::ThreadBase::~ThreadBase() 650{ 651 // mConfigEvents should be empty, but just in case it isn't, free the memory it owns 652 mConfigEvents.clear(); 653 654 // do not lock the mutex in destructor 655 releaseWakeLock_l(); 656 if (mPowerManager != 0) { 657 sp<IBinder> binder = IInterface::asBinder(mPowerManager); 658 binder->unlinkToDeath(mDeathRecipient); 659 } 660} 661 662status_t AudioFlinger::ThreadBase::readyToRun() 663{ 664 status_t status = initCheck(); 665 if (status == NO_ERROR) { 666 ALOGI("AudioFlinger's thread %p ready to run", this); 667 } else { 668 ALOGE("No working audio driver found."); 669 } 670 return status; 671} 672 673void AudioFlinger::ThreadBase::exit() 674{ 675 ALOGV("ThreadBase::exit"); 676 // do any cleanup required for exit to succeed 677 preExit(); 678 { 679 // This lock prevents the following race in thread (uniprocessor for illustration): 680 // if (!exitPending()) { 681 // // context switch from here to exit() 682 // // exit() calls requestExit(), what exitPending() observes 683 // // exit() calls signal(), which is dropped since no waiters 684 // // context switch back from exit() to here 685 // mWaitWorkCV.wait(...); 686 // // now thread is hung 687 // } 688 AutoMutex lock(mLock); 689 requestExit(); 690 mWaitWorkCV.broadcast(); 691 } 692 // When Thread::requestExitAndWait is made virtual and this method is renamed to 693 // "virtual status_t requestExitAndWait()", replace by "return Thread::requestExitAndWait();" 694 requestExitAndWait(); 695} 696 697status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs) 698{ 699 ALOGV("ThreadBase::setParameters() %s", keyValuePairs.string()); 700 Mutex::Autolock _l(mLock); 701 702 return sendSetParameterConfigEvent_l(keyValuePairs); 703} 704 705// sendConfigEvent_l() must be called with ThreadBase::mLock held 706// Can temporarily release the lock if waiting for a reply from processConfigEvents_l(). 707status_t AudioFlinger::ThreadBase::sendConfigEvent_l(sp<ConfigEvent>& event) 708{ 709 status_t status = NO_ERROR; 710 711 if (event->mRequiresSystemReady && !mSystemReady) { 712 event->mWaitStatus = false; 713 mPendingConfigEvents.add(event); 714 return status; 715 } 716 mConfigEvents.add(event); 717 ALOGV("sendConfigEvent_l() num events %zu event %d", mConfigEvents.size(), event->mType); 718 mWaitWorkCV.signal(); 719 mLock.unlock(); 720 { 721 Mutex::Autolock _l(event->mLock); 722 while (event->mWaitStatus) { 723 if (event->mCond.waitRelative(event->mLock, kConfigEventTimeoutNs) != NO_ERROR) { 724 event->mStatus = TIMED_OUT; 725 event->mWaitStatus = false; 726 } 727 } 728 status = event->mStatus; 729 } 730 mLock.lock(); 731 return status; 732} 733 734void AudioFlinger::ThreadBase::sendIoConfigEvent(audio_io_config_event event, pid_t pid) 735{ 736 Mutex::Autolock _l(mLock); 737 sendIoConfigEvent_l(event, pid); 738} 739 740// sendIoConfigEvent_l() must be called with ThreadBase::mLock held 741void AudioFlinger::ThreadBase::sendIoConfigEvent_l(audio_io_config_event event, pid_t pid) 742{ 743 sp<ConfigEvent> configEvent = (ConfigEvent *)new IoConfigEvent(event, pid); 744 sendConfigEvent_l(configEvent); 745} 746 747void AudioFlinger::ThreadBase::sendPrioConfigEvent(pid_t pid, pid_t tid, int32_t prio) 748{ 749 Mutex::Autolock _l(mLock); 750 sendPrioConfigEvent_l(pid, tid, prio); 751} 752 753// sendPrioConfigEvent_l() must be called with ThreadBase::mLock held 754void AudioFlinger::ThreadBase::sendPrioConfigEvent_l(pid_t pid, pid_t tid, int32_t prio) 755{ 756 sp<ConfigEvent> configEvent = (ConfigEvent *)new PrioConfigEvent(pid, tid, prio); 757 sendConfigEvent_l(configEvent); 758} 759 760// sendSetParameterConfigEvent_l() must be called with ThreadBase::mLock held 761status_t AudioFlinger::ThreadBase::sendSetParameterConfigEvent_l(const String8& keyValuePair) 762{ 763 sp<ConfigEvent> configEvent; 764 AudioParameter param(keyValuePair); 765 int value; 766 if (param.getInt(String8(AUDIO_PARAMETER_MONO_OUTPUT), value) == NO_ERROR) { 767 setMasterMono_l(value != 0); 768 if (param.size() == 1) { 769 return NO_ERROR; // should be a solo parameter - we don't pass down 770 } 771 param.remove(String8(AUDIO_PARAMETER_MONO_OUTPUT)); 772 configEvent = new SetParameterConfigEvent(param.toString()); 773 } else { 774 configEvent = new SetParameterConfigEvent(keyValuePair); 775 } 776 return sendConfigEvent_l(configEvent); 777} 778 779status_t AudioFlinger::ThreadBase::sendCreateAudioPatchConfigEvent( 780 const struct audio_patch *patch, 781 audio_patch_handle_t *handle) 782{ 783 Mutex::Autolock _l(mLock); 784 sp<ConfigEvent> configEvent = (ConfigEvent *)new CreateAudioPatchConfigEvent(*patch, *handle); 785 status_t status = sendConfigEvent_l(configEvent); 786 if (status == NO_ERROR) { 787 CreateAudioPatchConfigEventData *data = 788 (CreateAudioPatchConfigEventData *)configEvent->mData.get(); 789 *handle = data->mHandle; 790 } 791 return status; 792} 793 794status_t AudioFlinger::ThreadBase::sendReleaseAudioPatchConfigEvent( 795 const audio_patch_handle_t handle) 796{ 797 Mutex::Autolock _l(mLock); 798 sp<ConfigEvent> configEvent = (ConfigEvent *)new ReleaseAudioPatchConfigEvent(handle); 799 return sendConfigEvent_l(configEvent); 800} 801 802 803// post condition: mConfigEvents.isEmpty() 804void AudioFlinger::ThreadBase::processConfigEvents_l() 805{ 806 bool configChanged = false; 807 808 while (!mConfigEvents.isEmpty()) { 809 ALOGV("processConfigEvents_l() remaining events %zu", mConfigEvents.size()); 810 sp<ConfigEvent> event = mConfigEvents[0]; 811 mConfigEvents.removeAt(0); 812 switch (event->mType) { 813 case CFG_EVENT_PRIO: { 814 PrioConfigEventData *data = (PrioConfigEventData *)event->mData.get(); 815 // FIXME Need to understand why this has to be done asynchronously 816 int err = requestPriority(data->mPid, data->mTid, data->mPrio, 817 true /*asynchronous*/); 818 if (err != 0) { 819 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 820 data->mPrio, data->mPid, data->mTid, err); 821 } 822 } break; 823 case CFG_EVENT_IO: { 824 IoConfigEventData *data = (IoConfigEventData *)event->mData.get(); 825 ioConfigChanged(data->mEvent, data->mPid); 826 } break; 827 case CFG_EVENT_SET_PARAMETER: { 828 SetParameterConfigEventData *data = (SetParameterConfigEventData *)event->mData.get(); 829 if (checkForNewParameter_l(data->mKeyValuePairs, event->mStatus)) { 830 configChanged = true; 831 } 832 } break; 833 case CFG_EVENT_CREATE_AUDIO_PATCH: { 834 CreateAudioPatchConfigEventData *data = 835 (CreateAudioPatchConfigEventData *)event->mData.get(); 836 event->mStatus = createAudioPatch_l(&data->mPatch, &data->mHandle); 837 } break; 838 case CFG_EVENT_RELEASE_AUDIO_PATCH: { 839 ReleaseAudioPatchConfigEventData *data = 840 (ReleaseAudioPatchConfigEventData *)event->mData.get(); 841 event->mStatus = releaseAudioPatch_l(data->mHandle); 842 } break; 843 default: 844 ALOG_ASSERT(false, "processConfigEvents_l() unknown event type %d", event->mType); 845 break; 846 } 847 { 848 Mutex::Autolock _l(event->mLock); 849 if (event->mWaitStatus) { 850 event->mWaitStatus = false; 851 event->mCond.signal(); 852 } 853 } 854 ALOGV_IF(mConfigEvents.isEmpty(), "processConfigEvents_l() DONE thread %p", this); 855 } 856 857 if (configChanged) { 858 cacheParameters_l(); 859 } 860} 861 862String8 channelMaskToString(audio_channel_mask_t mask, bool output) { 863 String8 s; 864 const audio_channel_representation_t representation = 865 audio_channel_mask_get_representation(mask); 866 867 switch (representation) { 868 case AUDIO_CHANNEL_REPRESENTATION_POSITION: { 869 if (output) { 870 if (mask & AUDIO_CHANNEL_OUT_FRONT_LEFT) s.append("front-left, "); 871 if (mask & AUDIO_CHANNEL_OUT_FRONT_RIGHT) s.append("front-right, "); 872 if (mask & AUDIO_CHANNEL_OUT_FRONT_CENTER) s.append("front-center, "); 873 if (mask & AUDIO_CHANNEL_OUT_LOW_FREQUENCY) s.append("low freq, "); 874 if (mask & AUDIO_CHANNEL_OUT_BACK_LEFT) s.append("back-left, "); 875 if (mask & AUDIO_CHANNEL_OUT_BACK_RIGHT) s.append("back-right, "); 876 if (mask & AUDIO_CHANNEL_OUT_FRONT_LEFT_OF_CENTER) s.append("front-left-of-center, "); 877 if (mask & AUDIO_CHANNEL_OUT_FRONT_RIGHT_OF_CENTER) s.append("front-right-of-center, "); 878 if (mask & AUDIO_CHANNEL_OUT_BACK_CENTER) s.append("back-center, "); 879 if (mask & AUDIO_CHANNEL_OUT_SIDE_LEFT) s.append("side-left, "); 880 if (mask & AUDIO_CHANNEL_OUT_SIDE_RIGHT) s.append("side-right, "); 881 if (mask & AUDIO_CHANNEL_OUT_TOP_CENTER) s.append("top-center ,"); 882 if (mask & AUDIO_CHANNEL_OUT_TOP_FRONT_LEFT) s.append("top-front-left, "); 883 if (mask & AUDIO_CHANNEL_OUT_TOP_FRONT_CENTER) s.append("top-front-center, "); 884 if (mask & AUDIO_CHANNEL_OUT_TOP_FRONT_RIGHT) s.append("top-front-right, "); 885 if (mask & AUDIO_CHANNEL_OUT_TOP_BACK_LEFT) s.append("top-back-left, "); 886 if (mask & AUDIO_CHANNEL_OUT_TOP_BACK_CENTER) s.append("top-back-center, " ); 887 if (mask & AUDIO_CHANNEL_OUT_TOP_BACK_RIGHT) s.append("top-back-right, " ); 888 if (mask & ~AUDIO_CHANNEL_OUT_ALL) s.append("unknown, "); 889 } else { 890 if (mask & AUDIO_CHANNEL_IN_LEFT) s.append("left, "); 891 if (mask & AUDIO_CHANNEL_IN_RIGHT) s.append("right, "); 892 if (mask & AUDIO_CHANNEL_IN_FRONT) s.append("front, "); 893 if (mask & AUDIO_CHANNEL_IN_BACK) s.append("back, "); 894 if (mask & AUDIO_CHANNEL_IN_LEFT_PROCESSED) s.append("left-processed, "); 895 if (mask & AUDIO_CHANNEL_IN_RIGHT_PROCESSED) s.append("right-processed, "); 896 if (mask & AUDIO_CHANNEL_IN_FRONT_PROCESSED) s.append("front-processed, "); 897 if (mask & AUDIO_CHANNEL_IN_BACK_PROCESSED) s.append("back-processed, "); 898 if (mask & AUDIO_CHANNEL_IN_PRESSURE) s.append("pressure, "); 899 if (mask & AUDIO_CHANNEL_IN_X_AXIS) s.append("X, "); 900 if (mask & AUDIO_CHANNEL_IN_Y_AXIS) s.append("Y, "); 901 if (mask & AUDIO_CHANNEL_IN_Z_AXIS) s.append("Z, "); 902 if (mask & AUDIO_CHANNEL_IN_VOICE_UPLINK) s.append("voice-uplink, "); 903 if (mask & AUDIO_CHANNEL_IN_VOICE_DNLINK) s.append("voice-dnlink, "); 904 if (mask & ~AUDIO_CHANNEL_IN_ALL) s.append("unknown, "); 905 } 906 const int len = s.length(); 907 if (len > 2) { 908 (void) s.lockBuffer(len); // needed? 909 s.unlockBuffer(len - 2); // remove trailing ", " 910 } 911 return s; 912 } 913 case AUDIO_CHANNEL_REPRESENTATION_INDEX: 914 s.appendFormat("index mask, bits:%#x", audio_channel_mask_get_bits(mask)); 915 return s; 916 default: 917 s.appendFormat("unknown mask, representation:%d bits:%#x", 918 representation, audio_channel_mask_get_bits(mask)); 919 return s; 920 } 921} 922 923void AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args __unused) 924{ 925 const size_t SIZE = 256; 926 char buffer[SIZE]; 927 String8 result; 928 929 bool locked = AudioFlinger::dumpTryLock(mLock); 930 if (!locked) { 931 dprintf(fd, "thread %p may be deadlocked\n", this); 932 } 933 934 dprintf(fd, " Thread name: %s\n", mThreadName); 935 dprintf(fd, " I/O handle: %d\n", mId); 936 dprintf(fd, " TID: %d\n", getTid()); 937 dprintf(fd, " Standby: %s\n", mStandby ? "yes" : "no"); 938 dprintf(fd, " Sample rate: %u Hz\n", mSampleRate); 939 dprintf(fd, " HAL frame count: %zu\n", mFrameCount); 940 dprintf(fd, " HAL format: 0x%x (%s)\n", mHALFormat, formatToString(mHALFormat)); 941 dprintf(fd, " HAL buffer size: %zu bytes\n", mBufferSize); 942 dprintf(fd, " Channel count: %u\n", mChannelCount); 943 dprintf(fd, " Channel mask: 0x%08x (%s)\n", mChannelMask, 944 channelMaskToString(mChannelMask, mType != RECORD).string()); 945 dprintf(fd, " Processing format: 0x%x (%s)\n", mFormat, formatToString(mFormat)); 946 dprintf(fd, " Processing frame size: %zu bytes\n", mFrameSize); 947 dprintf(fd, " Pending config events:"); 948 size_t numConfig = mConfigEvents.size(); 949 if (numConfig) { 950 for (size_t i = 0; i < numConfig; i++) { 951 mConfigEvents[i]->dump(buffer, SIZE); 952 dprintf(fd, "\n %s", buffer); 953 } 954 dprintf(fd, "\n"); 955 } else { 956 dprintf(fd, " none\n"); 957 } 958 dprintf(fd, " Output device: %#x (%s)\n", mOutDevice, devicesToString(mOutDevice).string()); 959 dprintf(fd, " Input device: %#x (%s)\n", mInDevice, devicesToString(mInDevice).string()); 960 dprintf(fd, " Audio source: %d (%s)\n", mAudioSource, sourceToString(mAudioSource)); 961 962 if (locked) { 963 mLock.unlock(); 964 } 965} 966 967void AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args) 968{ 969 const size_t SIZE = 256; 970 char buffer[SIZE]; 971 String8 result; 972 973 size_t numEffectChains = mEffectChains.size(); 974 snprintf(buffer, SIZE, " %zu Effect Chains\n", numEffectChains); 975 write(fd, buffer, strlen(buffer)); 976 977 for (size_t i = 0; i < numEffectChains; ++i) { 978 sp<EffectChain> chain = mEffectChains[i]; 979 if (chain != 0) { 980 chain->dump(fd, args); 981 } 982 } 983} 984 985void AudioFlinger::ThreadBase::acquireWakeLock(int uid) 986{ 987 Mutex::Autolock _l(mLock); 988 acquireWakeLock_l(uid); 989} 990 991String16 AudioFlinger::ThreadBase::getWakeLockTag() 992{ 993 switch (mType) { 994 case MIXER: 995 return String16("AudioMix"); 996 case DIRECT: 997 return String16("AudioDirectOut"); 998 case DUPLICATING: 999 return String16("AudioDup"); 1000 case RECORD: 1001 return String16("AudioIn"); 1002 case OFFLOAD: 1003 return String16("AudioOffload"); 1004 default: 1005 ALOG_ASSERT(false); 1006 return String16("AudioUnknown"); 1007 } 1008} 1009 1010void AudioFlinger::ThreadBase::acquireWakeLock_l(int uid) 1011{ 1012 getPowerManager_l(); 1013 if (mPowerManager != 0) { 1014 sp<IBinder> binder = new BBinder(); 1015 status_t status; 1016 if (uid >= 0) { 1017 status = mPowerManager->acquireWakeLockWithUid(POWERMANAGER_PARTIAL_WAKE_LOCK, 1018 binder, 1019 getWakeLockTag(), 1020 String16("audioserver"), 1021 uid, 1022 true /* FIXME force oneway contrary to .aidl */); 1023 } else { 1024 status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK, 1025 binder, 1026 getWakeLockTag(), 1027 String16("audioserver"), 1028 true /* FIXME force oneway contrary to .aidl */); 1029 } 1030 if (status == NO_ERROR) { 1031 mWakeLockToken = binder; 1032 } 1033 ALOGV("acquireWakeLock_l() %s status %d", mThreadName, status); 1034 } 1035 1036 if (!mNotifiedBatteryStart) { 1037 BatteryNotifier::getInstance().noteStartAudio(); 1038 mNotifiedBatteryStart = true; 1039 } 1040 gBoottime.acquire(mWakeLockToken); 1041 mTimestamp.mTimebaseOffset[ExtendedTimestamp::TIMEBASE_BOOTTIME] = 1042 gBoottime.getBoottimeOffset(); 1043} 1044 1045void AudioFlinger::ThreadBase::releaseWakeLock() 1046{ 1047 Mutex::Autolock _l(mLock); 1048 releaseWakeLock_l(); 1049} 1050 1051void AudioFlinger::ThreadBase::releaseWakeLock_l() 1052{ 1053 gBoottime.release(mWakeLockToken); 1054 if (mWakeLockToken != 0) { 1055 ALOGV("releaseWakeLock_l() %s", mThreadName); 1056 if (mPowerManager != 0) { 1057 mPowerManager->releaseWakeLock(mWakeLockToken, 0, 1058 true /* FIXME force oneway contrary to .aidl */); 1059 } 1060 mWakeLockToken.clear(); 1061 } 1062 1063 if (mNotifiedBatteryStart) { 1064 BatteryNotifier::getInstance().noteStopAudio(); 1065 mNotifiedBatteryStart = false; 1066 } 1067} 1068 1069void AudioFlinger::ThreadBase::updateWakeLockUids(const SortedVector<int> &uids) { 1070 Mutex::Autolock _l(mLock); 1071 updateWakeLockUids_l(uids); 1072} 1073 1074void AudioFlinger::ThreadBase::getPowerManager_l() { 1075 if (mSystemReady && mPowerManager == 0) { 1076 // use checkService() to avoid blocking if power service is not up yet 1077 sp<IBinder> binder = 1078 defaultServiceManager()->checkService(String16("power")); 1079 if (binder == 0) { 1080 ALOGW("Thread %s cannot connect to the power manager service", mThreadName); 1081 } else { 1082 mPowerManager = interface_cast<IPowerManager>(binder); 1083 binder->linkToDeath(mDeathRecipient); 1084 } 1085 } 1086} 1087 1088void AudioFlinger::ThreadBase::updateWakeLockUids_l(const SortedVector<int> &uids) { 1089 getPowerManager_l(); 1090 if (mWakeLockToken == NULL) { // token may be NULL if AudioFlinger::systemReady() not called. 1091 if (mSystemReady) { 1092 ALOGE("no wake lock to update, but system ready!"); 1093 } else { 1094 ALOGW("no wake lock to update, system not ready yet"); 1095 } 1096 return; 1097 } 1098 if (mPowerManager != 0) { 1099 sp<IBinder> binder = new BBinder(); 1100 status_t status; 1101 status = mPowerManager->updateWakeLockUids(mWakeLockToken, uids.size(), uids.array(), 1102 true /* FIXME force oneway contrary to .aidl */); 1103 ALOGV("updateWakeLockUids_l() %s status %d", mThreadName, status); 1104 } 1105} 1106 1107void AudioFlinger::ThreadBase::clearPowerManager() 1108{ 1109 Mutex::Autolock _l(mLock); 1110 releaseWakeLock_l(); 1111 mPowerManager.clear(); 1112} 1113 1114void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who __unused) 1115{ 1116 sp<ThreadBase> thread = mThread.promote(); 1117 if (thread != 0) { 1118 thread->clearPowerManager(); 1119 } 1120 ALOGW("power manager service died !!!"); 1121} 1122 1123void AudioFlinger::ThreadBase::setEffectSuspended( 1124 const effect_uuid_t *type, bool suspend, audio_session_t sessionId) 1125{ 1126 Mutex::Autolock _l(mLock); 1127 setEffectSuspended_l(type, suspend, sessionId); 1128} 1129 1130void AudioFlinger::ThreadBase::setEffectSuspended_l( 1131 const effect_uuid_t *type, bool suspend, audio_session_t sessionId) 1132{ 1133 sp<EffectChain> chain = getEffectChain_l(sessionId); 1134 if (chain != 0) { 1135 if (type != NULL) { 1136 chain->setEffectSuspended_l(type, suspend); 1137 } else { 1138 chain->setEffectSuspendedAll_l(suspend); 1139 } 1140 } 1141 1142 updateSuspendedSessions_l(type, suspend, sessionId); 1143} 1144 1145void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain) 1146{ 1147 ssize_t index = mSuspendedSessions.indexOfKey(chain->sessionId()); 1148 if (index < 0) { 1149 return; 1150 } 1151 1152 const KeyedVector <int, sp<SuspendedSessionDesc> >& sessionEffects = 1153 mSuspendedSessions.valueAt(index); 1154 1155 for (size_t i = 0; i < sessionEffects.size(); i++) { 1156 sp<SuspendedSessionDesc> desc = sessionEffects.valueAt(i); 1157 for (int j = 0; j < desc->mRefCount; j++) { 1158 if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) { 1159 chain->setEffectSuspendedAll_l(true); 1160 } else { 1161 ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x", 1162 desc->mType.timeLow); 1163 chain->setEffectSuspended_l(&desc->mType, true); 1164 } 1165 } 1166 } 1167} 1168 1169void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type, 1170 bool suspend, 1171 audio_session_t sessionId) 1172{ 1173 ssize_t index = mSuspendedSessions.indexOfKey(sessionId); 1174 1175 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects; 1176 1177 if (suspend) { 1178 if (index >= 0) { 1179 sessionEffects = mSuspendedSessions.valueAt(index); 1180 } else { 1181 mSuspendedSessions.add(sessionId, sessionEffects); 1182 } 1183 } else { 1184 if (index < 0) { 1185 return; 1186 } 1187 sessionEffects = mSuspendedSessions.valueAt(index); 1188 } 1189 1190 1191 int key = EffectChain::kKeyForSuspendAll; 1192 if (type != NULL) { 1193 key = type->timeLow; 1194 } 1195 index = sessionEffects.indexOfKey(key); 1196 1197 sp<SuspendedSessionDesc> desc; 1198 if (suspend) { 1199 if (index >= 0) { 1200 desc = sessionEffects.valueAt(index); 1201 } else { 1202 desc = new SuspendedSessionDesc(); 1203 if (type != NULL) { 1204 desc->mType = *type; 1205 } 1206 sessionEffects.add(key, desc); 1207 ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key); 1208 } 1209 desc->mRefCount++; 1210 } else { 1211 if (index < 0) { 1212 return; 1213 } 1214 desc = sessionEffects.valueAt(index); 1215 if (--desc->mRefCount == 0) { 1216 ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key); 1217 sessionEffects.removeItemsAt(index); 1218 if (sessionEffects.isEmpty()) { 1219 ALOGV("updateSuspendedSessions_l() restore removing session %d", 1220 sessionId); 1221 mSuspendedSessions.removeItem(sessionId); 1222 } 1223 } 1224 } 1225 if (!sessionEffects.isEmpty()) { 1226 mSuspendedSessions.replaceValueFor(sessionId, sessionEffects); 1227 } 1228} 1229 1230void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 1231 bool enabled, 1232 audio_session_t sessionId) 1233{ 1234 Mutex::Autolock _l(mLock); 1235 checkSuspendOnEffectEnabled_l(effect, enabled, sessionId); 1236} 1237 1238void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 1239 bool enabled, 1240 audio_session_t sessionId) 1241{ 1242 if (mType != RECORD) { 1243 // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on 1244 // another session. This gives the priority to well behaved effect control panels 1245 // and applications not using global effects. 1246 // Enabling post processing in AUDIO_SESSION_OUTPUT_STAGE session does not affect 1247 // global effects 1248 if ((sessionId != AUDIO_SESSION_OUTPUT_MIX) && (sessionId != AUDIO_SESSION_OUTPUT_STAGE)) { 1249 setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX); 1250 } 1251 } 1252 1253 sp<EffectChain> chain = getEffectChain_l(sessionId); 1254 if (chain != 0) { 1255 chain->checkSuspendOnEffectEnabled(effect, enabled); 1256 } 1257} 1258 1259// ThreadBase::createEffect_l() must be called with AudioFlinger::mLock held 1260sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 1261 const sp<AudioFlinger::Client>& client, 1262 const sp<IEffectClient>& effectClient, 1263 int32_t priority, 1264 audio_session_t sessionId, 1265 effect_descriptor_t *desc, 1266 int *enabled, 1267 status_t *status) 1268{ 1269 sp<EffectModule> effect; 1270 sp<EffectHandle> handle; 1271 status_t lStatus; 1272 sp<EffectChain> chain; 1273 bool chainCreated = false; 1274 bool effectCreated = false; 1275 bool effectRegistered = false; 1276 1277 lStatus = initCheck(); 1278 if (lStatus != NO_ERROR) { 1279 ALOGW("createEffect_l() Audio driver not initialized."); 1280 goto Exit; 1281 } 1282 1283 // Reject any effect on Direct output threads for now, since the format of 1284 // mSinkBuffer is not guaranteed to be compatible with effect processing (PCM 16 stereo). 1285 if (mType == DIRECT) { 1286 ALOGW("createEffect_l() Cannot add effect %s on Direct output type thread %s", 1287 desc->name, mThreadName); 1288 lStatus = BAD_VALUE; 1289 goto Exit; 1290 } 1291 1292 // Reject any effect on mixer or duplicating multichannel sinks. 1293 // TODO: fix both format and multichannel issues with effects. 1294 if ((mType == MIXER || mType == DUPLICATING) && mChannelCount != FCC_2) { 1295 ALOGW("createEffect_l() Cannot add effect %s for multichannel(%d) %s threads", 1296 desc->name, mChannelCount, mType == MIXER ? "MIXER" : "DUPLICATING"); 1297 lStatus = BAD_VALUE; 1298 goto Exit; 1299 } 1300 1301 // Allow global effects only on offloaded and mixer threads 1302 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 1303 switch (mType) { 1304 case MIXER: 1305 case OFFLOAD: 1306 break; 1307 case DIRECT: 1308 case DUPLICATING: 1309 case RECORD: 1310 default: 1311 ALOGW("createEffect_l() Cannot add global effect %s on thread %s", 1312 desc->name, mThreadName); 1313 lStatus = BAD_VALUE; 1314 goto Exit; 1315 } 1316 } 1317 1318 // Only Pre processor effects are allowed on input threads and only on input threads 1319 if ((mType == RECORD) != ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { 1320 ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d", 1321 desc->name, desc->flags, mType); 1322 lStatus = BAD_VALUE; 1323 goto Exit; 1324 } 1325 1326 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 1327 1328 { // scope for mLock 1329 Mutex::Autolock _l(mLock); 1330 1331 // check for existing effect chain with the requested audio session 1332 chain = getEffectChain_l(sessionId); 1333 if (chain == 0) { 1334 // create a new chain for this session 1335 ALOGV("createEffect_l() new effect chain for session %d", sessionId); 1336 chain = new EffectChain(this, sessionId); 1337 addEffectChain_l(chain); 1338 chain->setStrategy(getStrategyForSession_l(sessionId)); 1339 chainCreated = true; 1340 } else { 1341 effect = chain->getEffectFromDesc_l(desc); 1342 } 1343 1344 ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get()); 1345 1346 if (effect == 0) { 1347 audio_unique_id_t id = mAudioFlinger->nextUniqueId(AUDIO_UNIQUE_ID_USE_EFFECT); 1348 // Check CPU and memory usage 1349 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 1350 if (lStatus != NO_ERROR) { 1351 goto Exit; 1352 } 1353 effectRegistered = true; 1354 // create a new effect module if none present in the chain 1355 effect = new EffectModule(this, chain, desc, id, sessionId); 1356 lStatus = effect->status(); 1357 if (lStatus != NO_ERROR) { 1358 goto Exit; 1359 } 1360 effect->setOffloaded(mType == OFFLOAD, mId); 1361 1362 lStatus = chain->addEffect_l(effect); 1363 if (lStatus != NO_ERROR) { 1364 goto Exit; 1365 } 1366 effectCreated = true; 1367 1368 effect->setDevice(mOutDevice); 1369 effect->setDevice(mInDevice); 1370 effect->setMode(mAudioFlinger->getMode()); 1371 effect->setAudioSource(mAudioSource); 1372 } 1373 // create effect handle and connect it to effect module 1374 handle = new EffectHandle(effect, client, effectClient, priority); 1375 lStatus = handle->initCheck(); 1376 if (lStatus == OK) { 1377 lStatus = effect->addHandle(handle.get()); 1378 } 1379 if (enabled != NULL) { 1380 *enabled = (int)effect->isEnabled(); 1381 } 1382 } 1383 1384Exit: 1385 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 1386 Mutex::Autolock _l(mLock); 1387 if (effectCreated) { 1388 chain->removeEffect_l(effect); 1389 } 1390 if (effectRegistered) { 1391 AudioSystem::unregisterEffect(effect->id()); 1392 } 1393 if (chainCreated) { 1394 removeEffectChain_l(chain); 1395 } 1396 handle.clear(); 1397 } 1398 1399 *status = lStatus; 1400 return handle; 1401} 1402 1403sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect(audio_session_t sessionId, 1404 int effectId) 1405{ 1406 Mutex::Autolock _l(mLock); 1407 return getEffect_l(sessionId, effectId); 1408} 1409 1410sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(audio_session_t sessionId, 1411 int effectId) 1412{ 1413 sp<EffectChain> chain = getEffectChain_l(sessionId); 1414 return chain != 0 ? chain->getEffectFromId_l(effectId) : 0; 1415} 1416 1417// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 1418// PlaybackThread::mLock held 1419status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 1420{ 1421 // check for existing effect chain with the requested audio session 1422 audio_session_t sessionId = effect->sessionId(); 1423 sp<EffectChain> chain = getEffectChain_l(sessionId); 1424 bool chainCreated = false; 1425 1426 ALOGD_IF((mType == OFFLOAD) && !effect->isOffloadable(), 1427 "addEffect_l() on offloaded thread %p: effect %s does not support offload flags %x", 1428 this, effect->desc().name, effect->desc().flags); 1429 1430 if (chain == 0) { 1431 // create a new chain for this session 1432 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 1433 chain = new EffectChain(this, sessionId); 1434 addEffectChain_l(chain); 1435 chain->setStrategy(getStrategyForSession_l(sessionId)); 1436 chainCreated = true; 1437 } 1438 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 1439 1440 if (chain->getEffectFromId_l(effect->id()) != 0) { 1441 ALOGW("addEffect_l() %p effect %s already present in chain %p", 1442 this, effect->desc().name, chain.get()); 1443 return BAD_VALUE; 1444 } 1445 1446 effect->setOffloaded(mType == OFFLOAD, mId); 1447 1448 status_t status = chain->addEffect_l(effect); 1449 if (status != NO_ERROR) { 1450 if (chainCreated) { 1451 removeEffectChain_l(chain); 1452 } 1453 return status; 1454 } 1455 1456 effect->setDevice(mOutDevice); 1457 effect->setDevice(mInDevice); 1458 effect->setMode(mAudioFlinger->getMode()); 1459 effect->setAudioSource(mAudioSource); 1460 return NO_ERROR; 1461} 1462 1463void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 1464 1465 ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 1466 effect_descriptor_t desc = effect->desc(); 1467 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 1468 detachAuxEffect_l(effect->id()); 1469 } 1470 1471 sp<EffectChain> chain = effect->chain().promote(); 1472 if (chain != 0) { 1473 // remove effect chain if removing last effect 1474 if (chain->removeEffect_l(effect) == 0) { 1475 removeEffectChain_l(chain); 1476 } 1477 } else { 1478 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 1479 } 1480} 1481 1482void AudioFlinger::ThreadBase::lockEffectChains_l( 1483 Vector< sp<AudioFlinger::EffectChain> >& effectChains) 1484{ 1485 effectChains = mEffectChains; 1486 for (size_t i = 0; i < mEffectChains.size(); i++) { 1487 mEffectChains[i]->lock(); 1488 } 1489} 1490 1491void AudioFlinger::ThreadBase::unlockEffectChains( 1492 const Vector< sp<AudioFlinger::EffectChain> >& effectChains) 1493{ 1494 for (size_t i = 0; i < effectChains.size(); i++) { 1495 effectChains[i]->unlock(); 1496 } 1497} 1498 1499sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(audio_session_t sessionId) 1500{ 1501 Mutex::Autolock _l(mLock); 1502 return getEffectChain_l(sessionId); 1503} 1504 1505sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(audio_session_t sessionId) 1506 const 1507{ 1508 size_t size = mEffectChains.size(); 1509 for (size_t i = 0; i < size; i++) { 1510 if (mEffectChains[i]->sessionId() == sessionId) { 1511 return mEffectChains[i]; 1512 } 1513 } 1514 return 0; 1515} 1516 1517void AudioFlinger::ThreadBase::setMode(audio_mode_t mode) 1518{ 1519 Mutex::Autolock _l(mLock); 1520 size_t size = mEffectChains.size(); 1521 for (size_t i = 0; i < size; i++) { 1522 mEffectChains[i]->setMode_l(mode); 1523 } 1524} 1525 1526void AudioFlinger::ThreadBase::getAudioPortConfig(struct audio_port_config *config) 1527{ 1528 config->type = AUDIO_PORT_TYPE_MIX; 1529 config->ext.mix.handle = mId; 1530 config->sample_rate = mSampleRate; 1531 config->format = mFormat; 1532 config->channel_mask = mChannelMask; 1533 config->config_mask = AUDIO_PORT_CONFIG_SAMPLE_RATE|AUDIO_PORT_CONFIG_CHANNEL_MASK| 1534 AUDIO_PORT_CONFIG_FORMAT; 1535} 1536 1537void AudioFlinger::ThreadBase::systemReady() 1538{ 1539 Mutex::Autolock _l(mLock); 1540 if (mSystemReady) { 1541 return; 1542 } 1543 mSystemReady = true; 1544 1545 for (size_t i = 0; i < mPendingConfigEvents.size(); i++) { 1546 sendConfigEvent_l(mPendingConfigEvents.editItemAt(i)); 1547 } 1548 mPendingConfigEvents.clear(); 1549} 1550 1551 1552// ---------------------------------------------------------------------------- 1553// Playback 1554// ---------------------------------------------------------------------------- 1555 1556AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, 1557 AudioStreamOut* output, 1558 audio_io_handle_t id, 1559 audio_devices_t device, 1560 type_t type, 1561 bool systemReady) 1562 : ThreadBase(audioFlinger, id, device, AUDIO_DEVICE_NONE, type, systemReady), 1563 mNormalFrameCount(0), mSinkBuffer(NULL), 1564 mMixerBufferEnabled(AudioFlinger::kEnableExtendedPrecision), 1565 mMixerBuffer(NULL), 1566 mMixerBufferSize(0), 1567 mMixerBufferFormat(AUDIO_FORMAT_INVALID), 1568 mMixerBufferValid(false), 1569 mEffectBufferEnabled(AudioFlinger::kEnableExtendedPrecision), 1570 mEffectBuffer(NULL), 1571 mEffectBufferSize(0), 1572 mEffectBufferFormat(AUDIO_FORMAT_INVALID), 1573 mEffectBufferValid(false), 1574 mSuspended(0), mBytesWritten(0), 1575 mFramesWritten(0), 1576 mActiveTracksGeneration(0), 1577 // mStreamTypes[] initialized in constructor body 1578 mOutput(output), 1579 mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false), 1580 mMixerStatus(MIXER_IDLE), 1581 mMixerStatusIgnoringFastTracks(MIXER_IDLE), 1582 mStandbyDelayNs(AudioFlinger::mStandbyTimeInNsecs), 1583 mBytesRemaining(0), 1584 mCurrentWriteLength(0), 1585 mUseAsyncWrite(false), 1586 mWriteAckSequence(0), 1587 mDrainSequence(0), 1588 mSignalPending(false), 1589 mScreenState(AudioFlinger::mScreenState), 1590 // index 0 is reserved for normal mixer's submix 1591 mFastTrackAvailMask(((1 << FastMixerState::sMaxFastTracks) - 1) & ~1), 1592 mHwSupportsPause(false), mHwPaused(false), mFlushPending(false) 1593{ 1594 snprintf(mThreadName, kThreadNameLength, "AudioOut_%X", id); 1595 mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mThreadName); 1596 1597 // Assumes constructor is called by AudioFlinger with it's mLock held, but 1598 // it would be safer to explicitly pass initial masterVolume/masterMute as 1599 // parameter. 1600 // 1601 // If the HAL we are using has support for master volume or master mute, 1602 // then do not attenuate or mute during mixing (just leave the volume at 1.0 1603 // and the mute set to false). 1604 mMasterVolume = audioFlinger->masterVolume_l(); 1605 mMasterMute = audioFlinger->masterMute_l(); 1606 if (mOutput && mOutput->audioHwDev) { 1607 if (mOutput->audioHwDev->canSetMasterVolume()) { 1608 mMasterVolume = 1.0; 1609 } 1610 1611 if (mOutput->audioHwDev->canSetMasterMute()) { 1612 mMasterMute = false; 1613 } 1614 } 1615 1616 readOutputParameters_l(); 1617 1618 // ++ operator does not compile 1619 for (audio_stream_type_t stream = AUDIO_STREAM_MIN; stream < AUDIO_STREAM_CNT; 1620 stream = (audio_stream_type_t) (stream + 1)) { 1621 mStreamTypes[stream].volume = mAudioFlinger->streamVolume_l(stream); 1622 mStreamTypes[stream].mute = mAudioFlinger->streamMute_l(stream); 1623 } 1624} 1625 1626AudioFlinger::PlaybackThread::~PlaybackThread() 1627{ 1628 mAudioFlinger->unregisterWriter(mNBLogWriter); 1629 free(mSinkBuffer); 1630 free(mMixerBuffer); 1631 free(mEffectBuffer); 1632} 1633 1634void AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) 1635{ 1636 dumpInternals(fd, args); 1637 dumpTracks(fd, args); 1638 dumpEffectChains(fd, args); 1639} 1640 1641void AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args __unused) 1642{ 1643 const size_t SIZE = 256; 1644 char buffer[SIZE]; 1645 String8 result; 1646 1647 result.appendFormat(" Stream volumes in dB: "); 1648 for (int i = 0; i < AUDIO_STREAM_CNT; ++i) { 1649 const stream_type_t *st = &mStreamTypes[i]; 1650 if (i > 0) { 1651 result.appendFormat(", "); 1652 } 1653 result.appendFormat("%d:%.2g", i, 20.0 * log10(st->volume)); 1654 if (st->mute) { 1655 result.append("M"); 1656 } 1657 } 1658 result.append("\n"); 1659 write(fd, result.string(), result.length()); 1660 result.clear(); 1661 1662 // These values are "raw"; they will wrap around. See prepareTracks_l() for a better way. 1663 FastTrackUnderruns underruns = getFastTrackUnderruns(0); 1664 dprintf(fd, " Normal mixer raw underrun counters: partial=%u empty=%u\n", 1665 underruns.mBitFields.mPartial, underruns.mBitFields.mEmpty); 1666 1667 size_t numtracks = mTracks.size(); 1668 size_t numactive = mActiveTracks.size(); 1669 dprintf(fd, " %zu Tracks", numtracks); 1670 size_t numactiveseen = 0; 1671 if (numtracks) { 1672 dprintf(fd, " of which %zu are active\n", numactive); 1673 Track::appendDumpHeader(result); 1674 for (size_t i = 0; i < numtracks; ++i) { 1675 sp<Track> track = mTracks[i]; 1676 if (track != 0) { 1677 bool active = mActiveTracks.indexOf(track) >= 0; 1678 if (active) { 1679 numactiveseen++; 1680 } 1681 track->dump(buffer, SIZE, active); 1682 result.append(buffer); 1683 } 1684 } 1685 } else { 1686 result.append("\n"); 1687 } 1688 if (numactiveseen != numactive) { 1689 // some tracks in the active list were not in the tracks list 1690 snprintf(buffer, SIZE, " The following tracks are in the active list but" 1691 " not in the track list\n"); 1692 result.append(buffer); 1693 Track::appendDumpHeader(result); 1694 for (size_t i = 0; i < numactive; ++i) { 1695 sp<Track> track = mActiveTracks[i].promote(); 1696 if (track != 0 && mTracks.indexOf(track) < 0) { 1697 track->dump(buffer, SIZE, true); 1698 result.append(buffer); 1699 } 1700 } 1701 } 1702 1703 write(fd, result.string(), result.size()); 1704} 1705 1706void AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 1707{ 1708 dprintf(fd, "\nOutput thread %p type %d (%s):\n", this, type(), threadTypeToString(type())); 1709 1710 dumpBase(fd, args); 1711 1712 dprintf(fd, " Normal frame count: %zu\n", mNormalFrameCount); 1713 dprintf(fd, " Last write occurred (msecs): %llu\n", 1714 (unsigned long long) ns2ms(systemTime() - mLastWriteTime)); 1715 dprintf(fd, " Total writes: %d\n", mNumWrites); 1716 dprintf(fd, " Delayed writes: %d\n", mNumDelayedWrites); 1717 dprintf(fd, " Blocked in write: %s\n", mInWrite ? "yes" : "no"); 1718 dprintf(fd, " Suspend count: %d\n", mSuspended); 1719 dprintf(fd, " Sink buffer : %p\n", mSinkBuffer); 1720 dprintf(fd, " Mixer buffer: %p\n", mMixerBuffer); 1721 dprintf(fd, " Effect buffer: %p\n", mEffectBuffer); 1722 dprintf(fd, " Fast track availMask=%#x\n", mFastTrackAvailMask); 1723 dprintf(fd, " Standby delay ns=%lld\n", (long long)mStandbyDelayNs); 1724 AudioStreamOut *output = mOutput; 1725 audio_output_flags_t flags = output != NULL ? output->flags : AUDIO_OUTPUT_FLAG_NONE; 1726 String8 flagsAsString = outputFlagsToString(flags); 1727 dprintf(fd, " AudioStreamOut: %p flags %#x (%s)\n", output, flags, flagsAsString.string()); 1728} 1729 1730// Thread virtuals 1731 1732void AudioFlinger::PlaybackThread::onFirstRef() 1733{ 1734 run(mThreadName, ANDROID_PRIORITY_URGENT_AUDIO); 1735} 1736 1737// ThreadBase virtuals 1738void AudioFlinger::PlaybackThread::preExit() 1739{ 1740 ALOGV(" preExit()"); 1741 // FIXME this is using hard-coded strings but in the future, this functionality will be 1742 // converted to use audio HAL extensions required to support tunneling 1743 mOutput->stream->common.set_parameters(&mOutput->stream->common, "exiting=1"); 1744} 1745 1746// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 1747sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 1748 const sp<AudioFlinger::Client>& client, 1749 audio_stream_type_t streamType, 1750 uint32_t sampleRate, 1751 audio_format_t format, 1752 audio_channel_mask_t channelMask, 1753 size_t *pFrameCount, 1754 const sp<IMemory>& sharedBuffer, 1755 audio_session_t sessionId, 1756 IAudioFlinger::track_flags_t *flags, 1757 pid_t tid, 1758 int uid, 1759 status_t *status) 1760{ 1761 size_t frameCount = *pFrameCount; 1762 sp<Track> track; 1763 status_t lStatus; 1764 1765 // client expresses a preference for FAST, but we get the final say 1766 if (*flags & IAudioFlinger::TRACK_FAST) { 1767 if ( 1768 // PCM data 1769 audio_is_linear_pcm(format) && 1770 // TODO: extract as a data library function that checks that a computationally 1771 // expensive downmixer is not required: isFastOutputChannelConversion() 1772 (channelMask == mChannelMask || 1773 mChannelMask != AUDIO_CHANNEL_OUT_STEREO || 1774 (channelMask == AUDIO_CHANNEL_OUT_MONO 1775 /* && mChannelMask == AUDIO_CHANNEL_OUT_STEREO */)) && 1776 // hardware sample rate 1777 (sampleRate == mSampleRate) && 1778 // normal mixer has an associated fast mixer 1779 hasFastMixer() && 1780 // there are sufficient fast track slots available 1781 (mFastTrackAvailMask != 0) 1782 // FIXME test that MixerThread for this fast track has a capable output HAL 1783 // FIXME add a permission test also? 1784 ) { 1785 // static tracks can have any nonzero framecount, streaming tracks check against minimum. 1786 if (sharedBuffer == 0) { 1787 // read the fast track multiplier property the first time it is needed 1788 int ok = pthread_once(&sFastTrackMultiplierOnce, sFastTrackMultiplierInit); 1789 if (ok != 0) { 1790 ALOGE("%s pthread_once failed: %d", __func__, ok); 1791 } 1792 frameCount = max(frameCount, mFrameCount * sFastTrackMultiplier); // incl framecount 0 1793 } 1794 ALOGV("AUDIO_OUTPUT_FLAG_FAST accepted: frameCount=%zu mFrameCount=%zu", 1795 frameCount, mFrameCount); 1796 } else { 1797 ALOGV("AUDIO_OUTPUT_FLAG_FAST denied: sharedBuffer=%p frameCount=%zu " 1798 "mFrameCount=%zu format=%#x mFormat=%#x isLinear=%d channelMask=%#x " 1799 "sampleRate=%u mSampleRate=%u " 1800 "hasFastMixer=%d tid=%d fastTrackAvailMask=%#x", 1801 sharedBuffer.get(), frameCount, mFrameCount, format, mFormat, 1802 audio_is_linear_pcm(format), 1803 channelMask, sampleRate, mSampleRate, hasFastMixer(), tid, mFastTrackAvailMask); 1804 *flags &= ~IAudioFlinger::TRACK_FAST; 1805 } 1806 } 1807 // For normal PCM streaming tracks, update minimum frame count. 1808 // For compatibility with AudioTrack calculation, buffer depth is forced 1809 // to be at least 2 x the normal mixer frame count and cover audio hardware latency. 1810 // This is probably too conservative, but legacy application code may depend on it. 1811 // If you change this calculation, also review the start threshold which is related. 1812 if (!(*flags & IAudioFlinger::TRACK_FAST) 1813 && audio_has_proportional_frames(format) && sharedBuffer == 0) { 1814 // this must match AudioTrack.cpp calculateMinFrameCount(). 1815 // TODO: Move to a common library 1816 uint32_t latencyMs = mOutput->stream->get_latency(mOutput->stream); 1817 uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate); 1818 if (minBufCount < 2) { 1819 minBufCount = 2; 1820 } 1821 // For normal mixing tracks, if speed is > 1.0f (normal), AudioTrack 1822 // or the client should compute and pass in a larger buffer request. 1823 size_t minFrameCount = 1824 minBufCount * sourceFramesNeededWithTimestretch( 1825 sampleRate, mNormalFrameCount, 1826 mSampleRate, AUDIO_TIMESTRETCH_SPEED_NORMAL /*speed*/); 1827 if (frameCount < minFrameCount) { // including frameCount == 0 1828 frameCount = minFrameCount; 1829 } 1830 } 1831 *pFrameCount = frameCount; 1832 1833 switch (mType) { 1834 1835 case DIRECT: 1836 if (audio_is_linear_pcm(format)) { // TODO maybe use audio_has_proportional_frames()? 1837 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1838 ALOGE("createTrack_l() Bad parameter: sampleRate %u format %#x, channelMask 0x%08x " 1839 "for output %p with format %#x", 1840 sampleRate, format, channelMask, mOutput, mFormat); 1841 lStatus = BAD_VALUE; 1842 goto Exit; 1843 } 1844 } 1845 break; 1846 1847 case OFFLOAD: 1848 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1849 ALOGE("createTrack_l() Bad parameter: sampleRate %d format %#x, channelMask 0x%08x \"" 1850 "for output %p with format %#x", 1851 sampleRate, format, channelMask, mOutput, mFormat); 1852 lStatus = BAD_VALUE; 1853 goto Exit; 1854 } 1855 break; 1856 1857 default: 1858 if (!audio_is_linear_pcm(format)) { 1859 ALOGE("createTrack_l() Bad parameter: format %#x \"" 1860 "for output %p with format %#x", 1861 format, mOutput, mFormat); 1862 lStatus = BAD_VALUE; 1863 goto Exit; 1864 } 1865 if (sampleRate > mSampleRate * AUDIO_RESAMPLER_DOWN_RATIO_MAX) { 1866 ALOGE("Sample rate out of range: %u mSampleRate %u", sampleRate, mSampleRate); 1867 lStatus = BAD_VALUE; 1868 goto Exit; 1869 } 1870 break; 1871 1872 } 1873 1874 lStatus = initCheck(); 1875 if (lStatus != NO_ERROR) { 1876 ALOGE("createTrack_l() audio driver not initialized"); 1877 goto Exit; 1878 } 1879 1880 { // scope for mLock 1881 Mutex::Autolock _l(mLock); 1882 1883 // all tracks in same audio session must share the same routing strategy otherwise 1884 // conflicts will happen when tracks are moved from one output to another by audio policy 1885 // manager 1886 uint32_t strategy = AudioSystem::getStrategyForStream(streamType); 1887 for (size_t i = 0; i < mTracks.size(); ++i) { 1888 sp<Track> t = mTracks[i]; 1889 if (t != 0 && t->isExternalTrack()) { 1890 uint32_t actual = AudioSystem::getStrategyForStream(t->streamType()); 1891 if (sessionId == t->sessionId() && strategy != actual) { 1892 ALOGE("createTrack_l() mismatched strategy; expected %u but found %u", 1893 strategy, actual); 1894 lStatus = BAD_VALUE; 1895 goto Exit; 1896 } 1897 } 1898 } 1899 1900 track = new Track(this, client, streamType, sampleRate, format, 1901 channelMask, frameCount, NULL, sharedBuffer, 1902 sessionId, uid, *flags, TrackBase::TYPE_DEFAULT); 1903 1904 lStatus = track != 0 ? track->initCheck() : (status_t) NO_MEMORY; 1905 if (lStatus != NO_ERROR) { 1906 ALOGE("createTrack_l() initCheck failed %d; no control block?", lStatus); 1907 // track must be cleared from the caller as the caller has the AF lock 1908 goto Exit; 1909 } 1910 mTracks.add(track); 1911 1912 sp<EffectChain> chain = getEffectChain_l(sessionId); 1913 if (chain != 0) { 1914 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1915 track->setMainBuffer(chain->inBuffer()); 1916 chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType())); 1917 chain->incTrackCnt(); 1918 } 1919 1920 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 1921 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 1922 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 1923 // so ask activity manager to do this on our behalf 1924 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 1925 } 1926 } 1927 1928 lStatus = NO_ERROR; 1929 1930Exit: 1931 *status = lStatus; 1932 return track; 1933} 1934 1935uint32_t AudioFlinger::PlaybackThread::correctLatency_l(uint32_t latency) const 1936{ 1937 return latency; 1938} 1939 1940uint32_t AudioFlinger::PlaybackThread::latency() const 1941{ 1942 Mutex::Autolock _l(mLock); 1943 return latency_l(); 1944} 1945uint32_t AudioFlinger::PlaybackThread::latency_l() const 1946{ 1947 if (initCheck() == NO_ERROR) { 1948 return correctLatency_l(mOutput->stream->get_latency(mOutput->stream)); 1949 } else { 1950 return 0; 1951 } 1952} 1953 1954void AudioFlinger::PlaybackThread::setMasterVolume(float value) 1955{ 1956 Mutex::Autolock _l(mLock); 1957 // Don't apply master volume in SW if our HAL can do it for us. 1958 if (mOutput && mOutput->audioHwDev && 1959 mOutput->audioHwDev->canSetMasterVolume()) { 1960 mMasterVolume = 1.0; 1961 } else { 1962 mMasterVolume = value; 1963 } 1964} 1965 1966void AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1967{ 1968 Mutex::Autolock _l(mLock); 1969 // Don't apply master mute in SW if our HAL can do it for us. 1970 if (mOutput && mOutput->audioHwDev && 1971 mOutput->audioHwDev->canSetMasterMute()) { 1972 mMasterMute = false; 1973 } else { 1974 mMasterMute = muted; 1975 } 1976} 1977 1978void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) 1979{ 1980 Mutex::Autolock _l(mLock); 1981 mStreamTypes[stream].volume = value; 1982 broadcast_l(); 1983} 1984 1985void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) 1986{ 1987 Mutex::Autolock _l(mLock); 1988 mStreamTypes[stream].mute = muted; 1989 broadcast_l(); 1990} 1991 1992float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const 1993{ 1994 Mutex::Autolock _l(mLock); 1995 return mStreamTypes[stream].volume; 1996} 1997 1998// addTrack_l() must be called with ThreadBase::mLock held 1999status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 2000{ 2001 status_t status = ALREADY_EXISTS; 2002 2003 if (mActiveTracks.indexOf(track) < 0) { 2004 // the track is newly added, make sure it fills up all its 2005 // buffers before playing. This is to ensure the client will 2006 // effectively get the latency it requested. 2007 if (track->isExternalTrack()) { 2008 TrackBase::track_state state = track->mState; 2009 mLock.unlock(); 2010 status = AudioSystem::startOutput(mId, track->streamType(), 2011 track->sessionId()); 2012 mLock.lock(); 2013 // abort track was stopped/paused while we released the lock 2014 if (state != track->mState) { 2015 if (status == NO_ERROR) { 2016 mLock.unlock(); 2017 AudioSystem::stopOutput(mId, track->streamType(), 2018 track->sessionId()); 2019 mLock.lock(); 2020 } 2021 return INVALID_OPERATION; 2022 } 2023 // abort if start is rejected by audio policy manager 2024 if (status != NO_ERROR) { 2025 return PERMISSION_DENIED; 2026 } 2027#ifdef ADD_BATTERY_DATA 2028 // to track the speaker usage 2029 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 2030#endif 2031 } 2032 2033 // set retry count for buffer fill 2034 if (track->isOffloaded()) { 2035 if (track->isStopping_1()) { 2036 track->mRetryCount = kMaxTrackStopRetriesOffload; 2037 } else { 2038 track->mRetryCount = kMaxTrackStartupRetriesOffload; 2039 } 2040 track->mFillingUpStatus = mStandby ? Track::FS_FILLING : Track::FS_FILLED; 2041 } else { 2042 track->mRetryCount = kMaxTrackStartupRetries; 2043 track->mFillingUpStatus = 2044 track->sharedBuffer() != 0 ? Track::FS_FILLED : Track::FS_FILLING; 2045 } 2046 2047 track->mResetDone = false; 2048 track->mPresentationCompleteFrames = 0; 2049 mActiveTracks.add(track); 2050 mWakeLockUids.add(track->uid()); 2051 mActiveTracksGeneration++; 2052 mLatestActiveTrack = track; 2053 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 2054 if (chain != 0) { 2055 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), 2056 track->sessionId()); 2057 chain->incActiveTrackCnt(); 2058 } 2059 2060 status = NO_ERROR; 2061 } 2062 2063 onAddNewTrack_l(); 2064 return status; 2065} 2066 2067bool AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 2068{ 2069 track->terminate(); 2070 // active tracks are removed by threadLoop() 2071 bool trackActive = (mActiveTracks.indexOf(track) >= 0); 2072 track->mState = TrackBase::STOPPED; 2073 if (!trackActive) { 2074 removeTrack_l(track); 2075 } else if (track->isFastTrack() || track->isOffloaded() || track->isDirect()) { 2076 track->mState = TrackBase::STOPPING_1; 2077 } 2078 2079 return trackActive; 2080} 2081 2082void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 2083{ 2084 track->triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 2085 mTracks.remove(track); 2086 deleteTrackName_l(track->name()); 2087 // redundant as track is about to be destroyed, for dumpsys only 2088 track->mName = -1; 2089 if (track->isFastTrack()) { 2090 int index = track->mFastIndex; 2091 ALOG_ASSERT(0 < index && index < (int)FastMixerState::sMaxFastTracks); 2092 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << index))); 2093 mFastTrackAvailMask |= 1 << index; 2094 // redundant as track is about to be destroyed, for dumpsys only 2095 track->mFastIndex = -1; 2096 } 2097 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 2098 if (chain != 0) { 2099 chain->decTrackCnt(); 2100 } 2101} 2102 2103void AudioFlinger::PlaybackThread::broadcast_l() 2104{ 2105 // Thread could be blocked waiting for async 2106 // so signal it to handle state changes immediately 2107 // If threadLoop is currently unlocked a signal of mWaitWorkCV will 2108 // be lost so we also flag to prevent it blocking on mWaitWorkCV 2109 mSignalPending = true; 2110 mWaitWorkCV.broadcast(); 2111} 2112 2113String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 2114{ 2115 Mutex::Autolock _l(mLock); 2116 if (initCheck() != NO_ERROR) { 2117 return String8(); 2118 } 2119 2120 char *s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 2121 const String8 out_s8(s); 2122 free(s); 2123 return out_s8; 2124} 2125 2126void AudioFlinger::PlaybackThread::ioConfigChanged(audio_io_config_event event, pid_t pid) { 2127 sp<AudioIoDescriptor> desc = new AudioIoDescriptor(); 2128 ALOGV("PlaybackThread::ioConfigChanged, thread %p, event %d", this, event); 2129 2130 desc->mIoHandle = mId; 2131 2132 switch (event) { 2133 case AUDIO_OUTPUT_OPENED: 2134 case AUDIO_OUTPUT_CONFIG_CHANGED: 2135 desc->mPatch = mPatch; 2136 desc->mChannelMask = mChannelMask; 2137 desc->mSamplingRate = mSampleRate; 2138 desc->mFormat = mFormat; 2139 desc->mFrameCount = mNormalFrameCount; // FIXME see 2140 // AudioFlinger::frameCount(audio_io_handle_t) 2141 desc->mFrameCountHAL = mFrameCount; 2142 desc->mLatency = latency_l(); 2143 break; 2144 2145 case AUDIO_OUTPUT_CLOSED: 2146 default: 2147 break; 2148 } 2149 mAudioFlinger->ioConfigChanged(event, desc, pid); 2150} 2151 2152void AudioFlinger::PlaybackThread::writeCallback() 2153{ 2154 ALOG_ASSERT(mCallbackThread != 0); 2155 mCallbackThread->resetWriteBlocked(); 2156} 2157 2158void AudioFlinger::PlaybackThread::drainCallback() 2159{ 2160 ALOG_ASSERT(mCallbackThread != 0); 2161 mCallbackThread->resetDraining(); 2162} 2163 2164void AudioFlinger::PlaybackThread::resetWriteBlocked(uint32_t sequence) 2165{ 2166 Mutex::Autolock _l(mLock); 2167 // reject out of sequence requests 2168 if ((mWriteAckSequence & 1) && (sequence == mWriteAckSequence)) { 2169 mWriteAckSequence &= ~1; 2170 mWaitWorkCV.signal(); 2171 } 2172} 2173 2174void AudioFlinger::PlaybackThread::resetDraining(uint32_t sequence) 2175{ 2176 Mutex::Autolock _l(mLock); 2177 // reject out of sequence requests 2178 if ((mDrainSequence & 1) && (sequence == mDrainSequence)) { 2179 mDrainSequence &= ~1; 2180 mWaitWorkCV.signal(); 2181 } 2182} 2183 2184// static 2185int AudioFlinger::PlaybackThread::asyncCallback(stream_callback_event_t event, 2186 void *param __unused, 2187 void *cookie) 2188{ 2189 AudioFlinger::PlaybackThread *me = (AudioFlinger::PlaybackThread *)cookie; 2190 ALOGV("asyncCallback() event %d", event); 2191 switch (event) { 2192 case STREAM_CBK_EVENT_WRITE_READY: 2193 me->writeCallback(); 2194 break; 2195 case STREAM_CBK_EVENT_DRAIN_READY: 2196 me->drainCallback(); 2197 break; 2198 default: 2199 ALOGW("asyncCallback() unknown event %d", event); 2200 break; 2201 } 2202 return 0; 2203} 2204 2205void AudioFlinger::PlaybackThread::readOutputParameters_l() 2206{ 2207 // unfortunately we have no way of recovering from errors here, hence the LOG_ALWAYS_FATAL 2208 mSampleRate = mOutput->getSampleRate(); 2209 mChannelMask = mOutput->getChannelMask(); 2210 if (!audio_is_output_channel(mChannelMask)) { 2211 LOG_ALWAYS_FATAL("HAL channel mask %#x not valid for output", mChannelMask); 2212 } 2213 if ((mType == MIXER || mType == DUPLICATING) 2214 && !isValidPcmSinkChannelMask(mChannelMask)) { 2215 LOG_ALWAYS_FATAL("HAL channel mask %#x not supported for mixed output", 2216 mChannelMask); 2217 } 2218 mChannelCount = audio_channel_count_from_out_mask(mChannelMask); 2219 2220 // Get actual HAL format. 2221 mHALFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 2222 // Get format from the shim, which will be different than the HAL format 2223 // if playing compressed audio over HDMI passthrough. 2224 mFormat = mOutput->getFormat(); 2225 if (!audio_is_valid_format(mFormat)) { 2226 LOG_ALWAYS_FATAL("HAL format %#x not valid for output", mFormat); 2227 } 2228 if ((mType == MIXER || mType == DUPLICATING) 2229 && !isValidPcmSinkFormat(mFormat)) { 2230 LOG_FATAL("HAL format %#x not supported for mixed output", 2231 mFormat); 2232 } 2233 mFrameSize = mOutput->getFrameSize(); 2234 mBufferSize = mOutput->stream->common.get_buffer_size(&mOutput->stream->common); 2235 mFrameCount = mBufferSize / mFrameSize; 2236 if (mFrameCount & 15) { 2237 ALOGW("HAL output buffer size is %zu frames but AudioMixer requires multiples of 16 frames", 2238 mFrameCount); 2239 } 2240 2241 if ((mOutput->flags & AUDIO_OUTPUT_FLAG_NON_BLOCKING) && 2242 (mOutput->stream->set_callback != NULL)) { 2243 if (mOutput->stream->set_callback(mOutput->stream, 2244 AudioFlinger::PlaybackThread::asyncCallback, this) == 0) { 2245 mUseAsyncWrite = true; 2246 mCallbackThread = new AudioFlinger::AsyncCallbackThread(this); 2247 } 2248 } 2249 2250 mHwSupportsPause = false; 2251 if (mOutput->flags & AUDIO_OUTPUT_FLAG_DIRECT) { 2252 if (mOutput->stream->pause != NULL) { 2253 if (mOutput->stream->resume != NULL) { 2254 mHwSupportsPause = true; 2255 } else { 2256 ALOGW("direct output implements pause but not resume"); 2257 } 2258 } else if (mOutput->stream->resume != NULL) { 2259 ALOGW("direct output implements resume but not pause"); 2260 } 2261 } 2262 if (!mHwSupportsPause && mOutput->flags & AUDIO_OUTPUT_FLAG_HW_AV_SYNC) { 2263 LOG_ALWAYS_FATAL("HW_AV_SYNC requested but HAL does not implement pause and resume"); 2264 } 2265 2266 if (mType == DUPLICATING && mMixerBufferEnabled && mEffectBufferEnabled) { 2267 // For best precision, we use float instead of the associated output 2268 // device format (typically PCM 16 bit). 2269 2270 mFormat = AUDIO_FORMAT_PCM_FLOAT; 2271 mFrameSize = mChannelCount * audio_bytes_per_sample(mFormat); 2272 mBufferSize = mFrameSize * mFrameCount; 2273 2274 // TODO: We currently use the associated output device channel mask and sample rate. 2275 // (1) Perhaps use the ORed channel mask of all downstream MixerThreads 2276 // (if a valid mask) to avoid premature downmix. 2277 // (2) Perhaps use the maximum sample rate of all downstream MixerThreads 2278 // instead of the output device sample rate to avoid loss of high frequency information. 2279 // This may need to be updated as MixerThread/OutputTracks are added and not here. 2280 } 2281 2282 // Calculate size of normal sink buffer relative to the HAL output buffer size 2283 double multiplier = 1.0; 2284 if (mType == MIXER && (kUseFastMixer == FastMixer_Static || 2285 kUseFastMixer == FastMixer_Dynamic)) { 2286 size_t minNormalFrameCount = (kMinNormalSinkBufferSizeMs * mSampleRate) / 1000; 2287 size_t maxNormalFrameCount = (kMaxNormalSinkBufferSizeMs * mSampleRate) / 1000; 2288 2289 // round up minimum and round down maximum to nearest 16 frames to satisfy AudioMixer 2290 minNormalFrameCount = (minNormalFrameCount + 15) & ~15; 2291 maxNormalFrameCount = maxNormalFrameCount & ~15; 2292 if (maxNormalFrameCount < minNormalFrameCount) { 2293 maxNormalFrameCount = minNormalFrameCount; 2294 } 2295 multiplier = (double) minNormalFrameCount / (double) mFrameCount; 2296 if (multiplier <= 1.0) { 2297 multiplier = 1.0; 2298 } else if (multiplier <= 2.0) { 2299 if (2 * mFrameCount <= maxNormalFrameCount) { 2300 multiplier = 2.0; 2301 } else { 2302 multiplier = (double) maxNormalFrameCount / (double) mFrameCount; 2303 } 2304 } else { 2305 multiplier = floor(multiplier); 2306 } 2307 } 2308 mNormalFrameCount = multiplier * mFrameCount; 2309 // round up to nearest 16 frames to satisfy AudioMixer 2310 if (mType == MIXER || mType == DUPLICATING) { 2311 mNormalFrameCount = (mNormalFrameCount + 15) & ~15; 2312 } 2313 ALOGI("HAL output buffer size %zu frames, normal sink buffer size %zu frames", mFrameCount, 2314 mNormalFrameCount); 2315 2316 // Check if we want to throttle the processing to no more than 2x normal rate 2317 mThreadThrottle = property_get_bool("af.thread.throttle", true /* default_value */); 2318 mThreadThrottleTimeMs = 0; 2319 mThreadThrottleEndMs = 0; 2320 mHalfBufferMs = mNormalFrameCount * 1000 / (2 * mSampleRate); 2321 2322 // mSinkBuffer is the sink buffer. Size is always multiple-of-16 frames. 2323 // Originally this was int16_t[] array, need to remove legacy implications. 2324 free(mSinkBuffer); 2325 mSinkBuffer = NULL; 2326 // For sink buffer size, we use the frame size from the downstream sink to avoid problems 2327 // with non PCM formats for compressed music, e.g. AAC, and Offload threads. 2328 const size_t sinkBufferSize = mNormalFrameCount * mFrameSize; 2329 (void)posix_memalign(&mSinkBuffer, 32, sinkBufferSize); 2330 2331 // We resize the mMixerBuffer according to the requirements of the sink buffer which 2332 // drives the output. 2333 free(mMixerBuffer); 2334 mMixerBuffer = NULL; 2335 if (mMixerBufferEnabled) { 2336 mMixerBufferFormat = AUDIO_FORMAT_PCM_FLOAT; // also valid: AUDIO_FORMAT_PCM_16_BIT. 2337 mMixerBufferSize = mNormalFrameCount * mChannelCount 2338 * audio_bytes_per_sample(mMixerBufferFormat); 2339 (void)posix_memalign(&mMixerBuffer, 32, mMixerBufferSize); 2340 } 2341 free(mEffectBuffer); 2342 mEffectBuffer = NULL; 2343 if (mEffectBufferEnabled) { 2344 mEffectBufferFormat = AUDIO_FORMAT_PCM_16_BIT; // Note: Effects support 16b only 2345 mEffectBufferSize = mNormalFrameCount * mChannelCount 2346 * audio_bytes_per_sample(mEffectBufferFormat); 2347 (void)posix_memalign(&mEffectBuffer, 32, mEffectBufferSize); 2348 } 2349 2350 // force reconfiguration of effect chains and engines to take new buffer size and audio 2351 // parameters into account 2352 // Note that mLock is not held when readOutputParameters_l() is called from the constructor 2353 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 2354 // matter. 2355 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 2356 Vector< sp<EffectChain> > effectChains = mEffectChains; 2357 for (size_t i = 0; i < effectChains.size(); i ++) { 2358 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 2359 } 2360} 2361 2362 2363status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames) 2364{ 2365 if (halFrames == NULL || dspFrames == NULL) { 2366 return BAD_VALUE; 2367 } 2368 Mutex::Autolock _l(mLock); 2369 if (initCheck() != NO_ERROR) { 2370 return INVALID_OPERATION; 2371 } 2372 int64_t framesWritten = mBytesWritten / mFrameSize; 2373 *halFrames = framesWritten; 2374 2375 if (isSuspended()) { 2376 // return an estimation of rendered frames when the output is suspended 2377 size_t latencyFrames = (latency_l() * mSampleRate) / 1000; 2378 *dspFrames = (uint32_t) 2379 (framesWritten >= (int64_t)latencyFrames ? framesWritten - latencyFrames : 0); 2380 return NO_ERROR; 2381 } else { 2382 status_t status; 2383 uint32_t frames; 2384 status = mOutput->getRenderPosition(&frames); 2385 *dspFrames = (size_t)frames; 2386 return status; 2387 } 2388} 2389 2390uint32_t AudioFlinger::PlaybackThread::hasAudioSession(audio_session_t sessionId) const 2391{ 2392 Mutex::Autolock _l(mLock); 2393 uint32_t result = 0; 2394 if (getEffectChain_l(sessionId) != 0) { 2395 result = EFFECT_SESSION; 2396 } 2397 2398 for (size_t i = 0; i < mTracks.size(); ++i) { 2399 sp<Track> track = mTracks[i]; 2400 if (sessionId == track->sessionId() && !track->isInvalid()) { 2401 result |= TRACK_SESSION; 2402 break; 2403 } 2404 } 2405 2406 return result; 2407} 2408 2409uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(audio_session_t sessionId) 2410{ 2411 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 2412 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 2413 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 2414 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 2415 } 2416 for (size_t i = 0; i < mTracks.size(); i++) { 2417 sp<Track> track = mTracks[i]; 2418 if (sessionId == track->sessionId() && !track->isInvalid()) { 2419 return AudioSystem::getStrategyForStream(track->streamType()); 2420 } 2421 } 2422 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 2423} 2424 2425 2426AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 2427{ 2428 Mutex::Autolock _l(mLock); 2429 return mOutput; 2430} 2431 2432AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 2433{ 2434 Mutex::Autolock _l(mLock); 2435 AudioStreamOut *output = mOutput; 2436 mOutput = NULL; 2437 // FIXME FastMixer might also have a raw ptr to mOutputSink; 2438 // must push a NULL and wait for ack 2439 mOutputSink.clear(); 2440 mPipeSink.clear(); 2441 mNormalSink.clear(); 2442 return output; 2443} 2444 2445// this method must always be called either with ThreadBase mLock held or inside the thread loop 2446audio_stream_t* AudioFlinger::PlaybackThread::stream() const 2447{ 2448 if (mOutput == NULL) { 2449 return NULL; 2450 } 2451 return &mOutput->stream->common; 2452} 2453 2454uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() const 2455{ 2456 return (uint32_t)((uint32_t)((mNormalFrameCount * 1000) / mSampleRate) * 1000); 2457} 2458 2459status_t AudioFlinger::PlaybackThread::setSyncEvent(const sp<SyncEvent>& event) 2460{ 2461 if (!isValidSyncEvent(event)) { 2462 return BAD_VALUE; 2463 } 2464 2465 Mutex::Autolock _l(mLock); 2466 2467 for (size_t i = 0; i < mTracks.size(); ++i) { 2468 sp<Track> track = mTracks[i]; 2469 if (event->triggerSession() == track->sessionId()) { 2470 (void) track->setSyncEvent(event); 2471 return NO_ERROR; 2472 } 2473 } 2474 2475 return NAME_NOT_FOUND; 2476} 2477 2478bool AudioFlinger::PlaybackThread::isValidSyncEvent(const sp<SyncEvent>& event) const 2479{ 2480 return event->type() == AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE; 2481} 2482 2483void AudioFlinger::PlaybackThread::threadLoop_removeTracks( 2484 const Vector< sp<Track> >& tracksToRemove) 2485{ 2486 size_t count = tracksToRemove.size(); 2487 if (count > 0) { 2488 for (size_t i = 0 ; i < count ; i++) { 2489 const sp<Track>& track = tracksToRemove.itemAt(i); 2490 if (track->isExternalTrack()) { 2491 AudioSystem::stopOutput(mId, track->streamType(), 2492 track->sessionId()); 2493#ifdef ADD_BATTERY_DATA 2494 // to track the speaker usage 2495 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 2496#endif 2497 if (track->isTerminated()) { 2498 AudioSystem::releaseOutput(mId, track->streamType(), 2499 track->sessionId()); 2500 } 2501 } 2502 } 2503 } 2504} 2505 2506void AudioFlinger::PlaybackThread::checkSilentMode_l() 2507{ 2508 if (!mMasterMute) { 2509 char value[PROPERTY_VALUE_MAX]; 2510 if (mOutDevice == AUDIO_DEVICE_OUT_REMOTE_SUBMIX) { 2511 ALOGD("ro.audio.silent will be ignored for threads on AUDIO_DEVICE_OUT_REMOTE_SUBMIX"); 2512 return; 2513 } 2514 if (property_get("ro.audio.silent", value, "0") > 0) { 2515 char *endptr; 2516 unsigned long ul = strtoul(value, &endptr, 0); 2517 if (*endptr == '\0' && ul != 0) { 2518 ALOGD("Silence is golden"); 2519 // The setprop command will not allow a property to be changed after 2520 // the first time it is set, so we don't have to worry about un-muting. 2521 setMasterMute_l(true); 2522 } 2523 } 2524 } 2525} 2526 2527// shared by MIXER and DIRECT, overridden by DUPLICATING 2528ssize_t AudioFlinger::PlaybackThread::threadLoop_write() 2529{ 2530 // FIXME rewrite to reduce number of system calls 2531 mLastWriteTime = systemTime(); 2532 mInWrite = true; 2533 ssize_t bytesWritten; 2534 const size_t offset = mCurrentWriteLength - mBytesRemaining; 2535 2536 // If an NBAIO sink is present, use it to write the normal mixer's submix 2537 if (mNormalSink != 0) { 2538 2539 const size_t count = mBytesRemaining / mFrameSize; 2540 2541 ATRACE_BEGIN("write"); 2542 // update the setpoint when AudioFlinger::mScreenState changes 2543 uint32_t screenState = AudioFlinger::mScreenState; 2544 if (screenState != mScreenState) { 2545 mScreenState = screenState; 2546 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 2547 if (pipe != NULL) { 2548 pipe->setAvgFrames((mScreenState & 1) ? 2549 (pipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2550 } 2551 } 2552 ssize_t framesWritten = mNormalSink->write((char *)mSinkBuffer + offset, count); 2553 ATRACE_END(); 2554 if (framesWritten > 0) { 2555 bytesWritten = framesWritten * mFrameSize; 2556 } else { 2557 bytesWritten = framesWritten; 2558 } 2559 // otherwise use the HAL / AudioStreamOut directly 2560 } else { 2561 // Direct output and offload threads 2562 2563 if (mUseAsyncWrite) { 2564 ALOGW_IF(mWriteAckSequence & 1, "threadLoop_write(): out of sequence write request"); 2565 mWriteAckSequence += 2; 2566 mWriteAckSequence |= 1; 2567 ALOG_ASSERT(mCallbackThread != 0); 2568 mCallbackThread->setWriteBlocked(mWriteAckSequence); 2569 } 2570 // FIXME We should have an implementation of timestamps for direct output threads. 2571 // They are used e.g for multichannel PCM playback over HDMI. 2572 bytesWritten = mOutput->write((char *)mSinkBuffer + offset, mBytesRemaining); 2573 2574 if (mUseAsyncWrite && 2575 ((bytesWritten < 0) || (bytesWritten == (ssize_t)mBytesRemaining))) { 2576 // do not wait for async callback in case of error of full write 2577 mWriteAckSequence &= ~1; 2578 ALOG_ASSERT(mCallbackThread != 0); 2579 mCallbackThread->setWriteBlocked(mWriteAckSequence); 2580 } 2581 } 2582 2583 mNumWrites++; 2584 mInWrite = false; 2585 mStandby = false; 2586 return bytesWritten; 2587} 2588 2589void AudioFlinger::PlaybackThread::threadLoop_drain() 2590{ 2591 if (mOutput->stream->drain) { 2592 ALOGV("draining %s", (mMixerStatus == MIXER_DRAIN_TRACK) ? "early" : "full"); 2593 if (mUseAsyncWrite) { 2594 ALOGW_IF(mDrainSequence & 1, "threadLoop_drain(): out of sequence drain request"); 2595 mDrainSequence |= 1; 2596 ALOG_ASSERT(mCallbackThread != 0); 2597 mCallbackThread->setDraining(mDrainSequence); 2598 } 2599 mOutput->stream->drain(mOutput->stream, 2600 (mMixerStatus == MIXER_DRAIN_TRACK) ? AUDIO_DRAIN_EARLY_NOTIFY 2601 : AUDIO_DRAIN_ALL); 2602 } 2603} 2604 2605void AudioFlinger::PlaybackThread::threadLoop_exit() 2606{ 2607 { 2608 Mutex::Autolock _l(mLock); 2609 for (size_t i = 0; i < mTracks.size(); i++) { 2610 sp<Track> track = mTracks[i]; 2611 track->invalidate(); 2612 } 2613 } 2614} 2615 2616/* 2617The derived values that are cached: 2618 - mSinkBufferSize from frame count * frame size 2619 - mActiveSleepTimeUs from activeSleepTimeUs() 2620 - mIdleSleepTimeUs from idleSleepTimeUs() 2621 - mStandbyDelayNs from mActiveSleepTimeUs (DIRECT only) or forced to at least 2622 kDefaultStandbyTimeInNsecs when connected to an A2DP device. 2623 - maxPeriod from frame count and sample rate (MIXER only) 2624 2625The parameters that affect these derived values are: 2626 - frame count 2627 - frame size 2628 - sample rate 2629 - device type: A2DP or not 2630 - device latency 2631 - format: PCM or not 2632 - active sleep time 2633 - idle sleep time 2634*/ 2635 2636void AudioFlinger::PlaybackThread::cacheParameters_l() 2637{ 2638 mSinkBufferSize = mNormalFrameCount * mFrameSize; 2639 mActiveSleepTimeUs = activeSleepTimeUs(); 2640 mIdleSleepTimeUs = idleSleepTimeUs(); 2641 2642 // make sure standby delay is not too short when connected to an A2DP sink to avoid 2643 // truncating audio when going to standby. 2644 mStandbyDelayNs = AudioFlinger::mStandbyTimeInNsecs; 2645 if ((mOutDevice & AUDIO_DEVICE_OUT_ALL_A2DP) != 0) { 2646 if (mStandbyDelayNs < kDefaultStandbyTimeInNsecs) { 2647 mStandbyDelayNs = kDefaultStandbyTimeInNsecs; 2648 } 2649 } 2650} 2651 2652bool AudioFlinger::PlaybackThread::invalidateTracks_l(audio_stream_type_t streamType) 2653{ 2654 ALOGV("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %zu", 2655 this, streamType, mTracks.size()); 2656 bool trackMatch = false; 2657 size_t size = mTracks.size(); 2658 for (size_t i = 0; i < size; i++) { 2659 sp<Track> t = mTracks[i]; 2660 if (t->streamType() == streamType && t->isExternalTrack()) { 2661 t->invalidate(); 2662 trackMatch = true; 2663 } 2664 } 2665 return trackMatch; 2666} 2667 2668void AudioFlinger::PlaybackThread::invalidateTracks(audio_stream_type_t streamType) 2669{ 2670 Mutex::Autolock _l(mLock); 2671 invalidateTracks_l(streamType); 2672} 2673 2674status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 2675{ 2676 audio_session_t session = chain->sessionId(); 2677 int16_t* buffer = reinterpret_cast<int16_t*>(mEffectBufferEnabled 2678 ? mEffectBuffer : mSinkBuffer); 2679 bool ownsBuffer = false; 2680 2681 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 2682 if (session > AUDIO_SESSION_OUTPUT_MIX) { 2683 // Only one effect chain can be present in direct output thread and it uses 2684 // the sink buffer as input 2685 if (mType != DIRECT) { 2686 size_t numSamples = mNormalFrameCount * mChannelCount; 2687 buffer = new int16_t[numSamples]; 2688 memset(buffer, 0, numSamples * sizeof(int16_t)); 2689 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 2690 ownsBuffer = true; 2691 } 2692 2693 // Attach all tracks with same session ID to this chain. 2694 for (size_t i = 0; i < mTracks.size(); ++i) { 2695 sp<Track> track = mTracks[i]; 2696 if (session == track->sessionId()) { 2697 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), 2698 buffer); 2699 track->setMainBuffer(buffer); 2700 chain->incTrackCnt(); 2701 } 2702 } 2703 2704 // indicate all active tracks in the chain 2705 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 2706 sp<Track> track = mActiveTracks[i].promote(); 2707 if (track == 0) { 2708 continue; 2709 } 2710 if (session == track->sessionId()) { 2711 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 2712 chain->incActiveTrackCnt(); 2713 } 2714 } 2715 } 2716 chain->setThread(this); 2717 chain->setInBuffer(buffer, ownsBuffer); 2718 chain->setOutBuffer(reinterpret_cast<int16_t*>(mEffectBufferEnabled 2719 ? mEffectBuffer : mSinkBuffer)); 2720 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 2721 // chains list in order to be processed last as it contains output stage effects. 2722 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 2723 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 2724 // after track specific effects and before output stage. 2725 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 2726 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX. 2727 // Effect chain for other sessions are inserted at beginning of effect 2728 // chains list to be processed before output mix effects. Relative order between other 2729 // sessions is not important. 2730 static_assert(AUDIO_SESSION_OUTPUT_MIX == 0 && 2731 AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX, 2732 "audio_session_t constants misdefined"); 2733 size_t size = mEffectChains.size(); 2734 size_t i = 0; 2735 for (i = 0; i < size; i++) { 2736 if (mEffectChains[i]->sessionId() < session) { 2737 break; 2738 } 2739 } 2740 mEffectChains.insertAt(chain, i); 2741 checkSuspendOnAddEffectChain_l(chain); 2742 2743 return NO_ERROR; 2744} 2745 2746size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 2747{ 2748 audio_session_t session = chain->sessionId(); 2749 2750 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 2751 2752 for (size_t i = 0; i < mEffectChains.size(); i++) { 2753 if (chain == mEffectChains[i]) { 2754 mEffectChains.removeAt(i); 2755 // detach all active tracks from the chain 2756 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 2757 sp<Track> track = mActiveTracks[i].promote(); 2758 if (track == 0) { 2759 continue; 2760 } 2761 if (session == track->sessionId()) { 2762 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 2763 chain.get(), session); 2764 chain->decActiveTrackCnt(); 2765 } 2766 } 2767 2768 // detach all tracks with same session ID from this chain 2769 for (size_t i = 0; i < mTracks.size(); ++i) { 2770 sp<Track> track = mTracks[i]; 2771 if (session == track->sessionId()) { 2772 track->setMainBuffer(reinterpret_cast<int16_t*>(mSinkBuffer)); 2773 chain->decTrackCnt(); 2774 } 2775 } 2776 break; 2777 } 2778 } 2779 return mEffectChains.size(); 2780} 2781 2782status_t AudioFlinger::PlaybackThread::attachAuxEffect( 2783 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2784{ 2785 Mutex::Autolock _l(mLock); 2786 return attachAuxEffect_l(track, EffectId); 2787} 2788 2789status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 2790 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2791{ 2792 status_t status = NO_ERROR; 2793 2794 if (EffectId == 0) { 2795 track->setAuxBuffer(0, NULL); 2796 } else { 2797 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 2798 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 2799 if (effect != 0) { 2800 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 2801 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 2802 } else { 2803 status = INVALID_OPERATION; 2804 } 2805 } else { 2806 status = BAD_VALUE; 2807 } 2808 } 2809 return status; 2810} 2811 2812void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 2813{ 2814 for (size_t i = 0; i < mTracks.size(); ++i) { 2815 sp<Track> track = mTracks[i]; 2816 if (track->auxEffectId() == effectId) { 2817 attachAuxEffect_l(track, 0); 2818 } 2819 } 2820} 2821 2822bool AudioFlinger::PlaybackThread::threadLoop() 2823{ 2824 Vector< sp<Track> > tracksToRemove; 2825 2826 mStandbyTimeNs = systemTime(); 2827 2828 // MIXER 2829 nsecs_t lastWarning = 0; 2830 nsecs_t mixStartNs = 0; 2831 2832 // DUPLICATING 2833 // FIXME could this be made local to while loop? 2834 writeFrames = 0; 2835 2836 int lastGeneration = 0; 2837 2838 cacheParameters_l(); 2839 mSleepTimeUs = mIdleSleepTimeUs; 2840 2841 if (mType == MIXER) { 2842 sleepTimeShift = 0; 2843 } 2844 2845 CpuStats cpuStats; 2846 const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid())); 2847 2848 acquireWakeLock(); 2849 2850 // mNBLogWriter->log can only be called while thread mutex mLock is held. 2851 // So if you need to log when mutex is unlocked, set logString to a non-NULL string, 2852 // and then that string will be logged at the next convenient opportunity. 2853 const char *logString = NULL; 2854 2855 checkSilentMode_l(); 2856 2857 while (!exitPending()) 2858 { 2859 cpuStats.sample(myName); 2860 2861 Vector< sp<EffectChain> > effectChains; 2862 2863 { // scope for mLock 2864 2865 Mutex::Autolock _l(mLock); 2866 2867 processConfigEvents_l(); 2868 2869 if (logString != NULL) { 2870 mNBLogWriter->logTimestamp(); 2871 mNBLogWriter->log(logString); 2872 logString = NULL; 2873 } 2874 2875 // Gather the framesReleased counters for all active tracks, 2876 // and associate with the sink frames written out. We need 2877 // this to convert the sink timestamp to the track timestamp. 2878 if (mNormalSink != 0) { 2879 // Note: The DuplicatingThread may not have a mNormalSink. 2880 // We always fetch the timestamp here because often the downstream 2881 // sink will block whie writing. 2882 ExtendedTimestamp timestamp; // use private copy to fetch 2883 (void) mNormalSink->getTimestamp(timestamp); 2884 2885 // We keep track of the last valid kernel position in case we are in underrun 2886 // and the normal mixer period is the same as the fast mixer period, or there 2887 // is some error from the HAL. 2888 if (mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL] >= 0) { 2889 mTimestamp.mPosition[ExtendedTimestamp::LOCATION_KERNEL_LASTKERNELOK] = 2890 mTimestamp.mPosition[ExtendedTimestamp::LOCATION_KERNEL]; 2891 mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL_LASTKERNELOK] = 2892 mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL]; 2893 2894 mTimestamp.mPosition[ExtendedTimestamp::LOCATION_SERVER_LASTKERNELOK] = 2895 mTimestamp.mPosition[ExtendedTimestamp::LOCATION_SERVER]; 2896 mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_SERVER_LASTKERNELOK] = 2897 mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_SERVER]; 2898 } else { 2899 ALOGV("getTimestamp error - no valid kernel position"); 2900 } 2901 2902 // copy over kernel info 2903 mTimestamp.mPosition[ExtendedTimestamp::LOCATION_KERNEL] = 2904 timestamp.mPosition[ExtendedTimestamp::LOCATION_KERNEL]; 2905 mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL] = 2906 timestamp.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL]; 2907 } 2908 // mFramesWritten for non-offloaded tracks are contiguous 2909 // even after standby() is called. This is useful for the track frame 2910 // to sink frame mapping. 2911 mTimestamp.mPosition[ExtendedTimestamp::LOCATION_SERVER] = mFramesWritten; 2912 mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_SERVER] = systemTime(); 2913 const size_t size = mActiveTracks.size(); 2914 for (size_t i = 0; i < size; ++i) { 2915 sp<Track> t = mActiveTracks[i].promote(); 2916 if (t != 0 && !t->isFastTrack()) { 2917 t->updateTrackFrameInfo( 2918 t->mAudioTrackServerProxy->framesReleased(), 2919 mFramesWritten, 2920 mTimestamp); 2921 } 2922 } 2923 2924 saveOutputTracks(); 2925 if (mSignalPending) { 2926 // A signal was raised while we were unlocked 2927 mSignalPending = false; 2928 } else if (waitingAsyncCallback_l()) { 2929 if (exitPending()) { 2930 break; 2931 } 2932 bool released = false; 2933 if (!keepWakeLock()) { 2934 releaseWakeLock_l(); 2935 released = true; 2936 } 2937 mWakeLockUids.clear(); 2938 mActiveTracksGeneration++; 2939 ALOGV("wait async completion"); 2940 mWaitWorkCV.wait(mLock); 2941 ALOGV("async completion/wake"); 2942 if (released) { 2943 acquireWakeLock_l(); 2944 } 2945 mStandbyTimeNs = systemTime() + mStandbyDelayNs; 2946 mSleepTimeUs = 0; 2947 2948 continue; 2949 } 2950 if ((!mActiveTracks.size() && systemTime() > mStandbyTimeNs) || 2951 isSuspended()) { 2952 // put audio hardware into standby after short delay 2953 if (shouldStandby_l()) { 2954 2955 threadLoop_standby(); 2956 2957 mStandby = true; 2958 } 2959 2960 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2961 // we're about to wait, flush the binder command buffer 2962 IPCThreadState::self()->flushCommands(); 2963 2964 clearOutputTracks(); 2965 2966 if (exitPending()) { 2967 break; 2968 } 2969 2970 releaseWakeLock_l(); 2971 mWakeLockUids.clear(); 2972 mActiveTracksGeneration++; 2973 // wait until we have something to do... 2974 ALOGV("%s going to sleep", myName.string()); 2975 mWaitWorkCV.wait(mLock); 2976 ALOGV("%s waking up", myName.string()); 2977 acquireWakeLock_l(); 2978 2979 mMixerStatus = MIXER_IDLE; 2980 mMixerStatusIgnoringFastTracks = MIXER_IDLE; 2981 mBytesWritten = 0; 2982 mBytesRemaining = 0; 2983 checkSilentMode_l(); 2984 2985 mStandbyTimeNs = systemTime() + mStandbyDelayNs; 2986 mSleepTimeUs = mIdleSleepTimeUs; 2987 if (mType == MIXER) { 2988 sleepTimeShift = 0; 2989 } 2990 2991 continue; 2992 } 2993 } 2994 // mMixerStatusIgnoringFastTracks is also updated internally 2995 mMixerStatus = prepareTracks_l(&tracksToRemove); 2996 2997 // compare with previously applied list 2998 if (lastGeneration != mActiveTracksGeneration) { 2999 // update wakelock 3000 updateWakeLockUids_l(mWakeLockUids); 3001 lastGeneration = mActiveTracksGeneration; 3002 } 3003 3004 // prevent any changes in effect chain list and in each effect chain 3005 // during mixing and effect process as the audio buffers could be deleted 3006 // or modified if an effect is created or deleted 3007 lockEffectChains_l(effectChains); 3008 } // mLock scope ends 3009 3010 if (mBytesRemaining == 0) { 3011 mCurrentWriteLength = 0; 3012 if (mMixerStatus == MIXER_TRACKS_READY) { 3013 mixStartNs = systemTime(); 3014 // threadLoop_mix() sets mCurrentWriteLength 3015 threadLoop_mix(); 3016 } else if ((mMixerStatus != MIXER_DRAIN_TRACK) 3017 && (mMixerStatus != MIXER_DRAIN_ALL)) { 3018 // threadLoop_sleepTime sets mSleepTimeUs to 0 if data 3019 // must be written to HAL 3020 threadLoop_sleepTime(); 3021 if (mSleepTimeUs == 0) { 3022 mCurrentWriteLength = mSinkBufferSize; 3023 } 3024 } 3025 // Either threadLoop_mix() or threadLoop_sleepTime() should have set 3026 // mMixerBuffer with data if mMixerBufferValid is true and mSleepTimeUs == 0. 3027 // Merge mMixerBuffer data into mEffectBuffer (if any effects are valid) 3028 // or mSinkBuffer (if there are no effects). 3029 // 3030 // This is done pre-effects computation; if effects change to 3031 // support higher precision, this needs to move. 3032 // 3033 // mMixerBufferValid is only set true by MixerThread::prepareTracks_l(). 3034 // TODO use mSleepTimeUs == 0 as an additional condition. 3035 if (mMixerBufferValid) { 3036 void *buffer = mEffectBufferValid ? mEffectBuffer : mSinkBuffer; 3037 audio_format_t format = mEffectBufferValid ? mEffectBufferFormat : mFormat; 3038 3039 // mono blend occurs for mixer threads only (not direct or offloaded) 3040 // and is handled here if we're going directly to the sink. 3041 if (requireMonoBlend() && !mEffectBufferValid) { 3042 mono_blend(mMixerBuffer, mMixerBufferFormat, mChannelCount, mNormalFrameCount, 3043 true /*limit*/); 3044 } 3045 3046 memcpy_by_audio_format(buffer, format, mMixerBuffer, mMixerBufferFormat, 3047 mNormalFrameCount * mChannelCount); 3048 } 3049 3050 mBytesRemaining = mCurrentWriteLength; 3051 if (isSuspended()) { 3052 mSleepTimeUs = suspendSleepTimeUs(); 3053 // simulate write to HAL when suspended 3054 mBytesWritten += mSinkBufferSize; 3055 mFramesWritten += mSinkBufferSize / mFrameSize; 3056 mBytesRemaining = 0; 3057 } 3058 3059 // only process effects if we're going to write 3060 if (mSleepTimeUs == 0 && mType != OFFLOAD) { 3061 for (size_t i = 0; i < effectChains.size(); i ++) { 3062 effectChains[i]->process_l(); 3063 } 3064 } 3065 } 3066 // Process effect chains for offloaded thread even if no audio 3067 // was read from audio track: process only updates effect state 3068 // and thus does have to be synchronized with audio writes but may have 3069 // to be called while waiting for async write callback 3070 if (mType == OFFLOAD) { 3071 for (size_t i = 0; i < effectChains.size(); i ++) { 3072 effectChains[i]->process_l(); 3073 } 3074 } 3075 3076 // Only if the Effects buffer is enabled and there is data in the 3077 // Effects buffer (buffer valid), we need to 3078 // copy into the sink buffer. 3079 // TODO use mSleepTimeUs == 0 as an additional condition. 3080 if (mEffectBufferValid) { 3081 //ALOGV("writing effect buffer to sink buffer format %#x", mFormat); 3082 3083 if (requireMonoBlend()) { 3084 mono_blend(mEffectBuffer, mEffectBufferFormat, mChannelCount, mNormalFrameCount, 3085 true /*limit*/); 3086 } 3087 3088 memcpy_by_audio_format(mSinkBuffer, mFormat, mEffectBuffer, mEffectBufferFormat, 3089 mNormalFrameCount * mChannelCount); 3090 } 3091 3092 // enable changes in effect chain 3093 unlockEffectChains(effectChains); 3094 3095 if (!waitingAsyncCallback()) { 3096 // mSleepTimeUs == 0 means we must write to audio hardware 3097 if (mSleepTimeUs == 0) { 3098 ssize_t ret = 0; 3099 if (mBytesRemaining) { 3100 ret = threadLoop_write(); 3101 if (ret < 0) { 3102 mBytesRemaining = 0; 3103 } else { 3104 mBytesWritten += ret; 3105 mBytesRemaining -= ret; 3106 mFramesWritten += ret / mFrameSize; 3107 } 3108 } else if ((mMixerStatus == MIXER_DRAIN_TRACK) || 3109 (mMixerStatus == MIXER_DRAIN_ALL)) { 3110 threadLoop_drain(); 3111 } 3112 if (mType == MIXER && !mStandby) { 3113 // write blocked detection 3114 nsecs_t now = systemTime(); 3115 nsecs_t delta = now - mLastWriteTime; 3116 if (delta > maxPeriod) { 3117 mNumDelayedWrites++; 3118 if ((now - lastWarning) > kWarningThrottleNs) { 3119 ATRACE_NAME("underrun"); 3120 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 3121 (unsigned long long) ns2ms(delta), mNumDelayedWrites, this); 3122 lastWarning = now; 3123 } 3124 } 3125 3126 if (mThreadThrottle 3127 && mMixerStatus == MIXER_TRACKS_READY // we are mixing (active tracks) 3128 && ret > 0) { // we wrote something 3129 // Limit MixerThread data processing to no more than twice the 3130 // expected processing rate. 3131 // 3132 // This helps prevent underruns with NuPlayer and other applications 3133 // which may set up buffers that are close to the minimum size, or use 3134 // deep buffers, and rely on a double-buffering sleep strategy to fill. 3135 // 3136 // The throttle smooths out sudden large data drains from the device, 3137 // e.g. when it comes out of standby, which often causes problems with 3138 // (1) mixer threads without a fast mixer (which has its own warm-up) 3139 // (2) minimum buffer sized tracks (even if the track is full, 3140 // the app won't fill fast enough to handle the sudden draw). 3141 // 3142 // Total time spent in last processing cycle equals time spent in 3143 // 1. threadLoop_write, as well as time spent in 3144 // 2. threadLoop_mix (significant for heavy mixing, especially 3145 // on low tier processors) 3146 3147 const int32_t deltaMs = (now - mixStartNs)/ 1000000; 3148 const int32_t throttleMs = mHalfBufferMs - deltaMs; 3149 if ((signed)mHalfBufferMs >= throttleMs && throttleMs > 0) { 3150 usleep(throttleMs * 1000); 3151 // notify of throttle start on verbose log 3152 ALOGV_IF(mThreadThrottleEndMs == mThreadThrottleTimeMs, 3153 "mixer(%p) throttle begin:" 3154 " ret(%zd) deltaMs(%d) requires sleep %d ms", 3155 this, ret, deltaMs, throttleMs); 3156 mThreadThrottleTimeMs += throttleMs; 3157 } else { 3158 uint32_t diff = mThreadThrottleTimeMs - mThreadThrottleEndMs; 3159 if (diff > 0) { 3160 // notify of throttle end on debug log 3161 // but prevent spamming for bluetooth 3162 ALOGD_IF(!audio_is_a2dp_out_device(outDevice()), 3163 "mixer(%p) throttle end: throttle time(%u)", this, diff); 3164 mThreadThrottleEndMs = mThreadThrottleTimeMs; 3165 } 3166 } 3167 } 3168 } 3169 3170 } else { 3171 ATRACE_BEGIN("sleep"); 3172 Mutex::Autolock _l(mLock); 3173 if (!mSignalPending && mConfigEvents.isEmpty() && !exitPending()) { 3174 mWaitWorkCV.waitRelative(mLock, microseconds((nsecs_t)mSleepTimeUs)); 3175 } 3176 ATRACE_END(); 3177 } 3178 } 3179 3180 // Finally let go of removed track(s), without the lock held 3181 // since we can't guarantee the destructors won't acquire that 3182 // same lock. This will also mutate and push a new fast mixer state. 3183 threadLoop_removeTracks(tracksToRemove); 3184 tracksToRemove.clear(); 3185 3186 // FIXME I don't understand the need for this here; 3187 // it was in the original code but maybe the 3188 // assignment in saveOutputTracks() makes this unnecessary? 3189 clearOutputTracks(); 3190 3191 // Effect chains will be actually deleted here if they were removed from 3192 // mEffectChains list during mixing or effects processing 3193 effectChains.clear(); 3194 3195 // FIXME Note that the above .clear() is no longer necessary since effectChains 3196 // is now local to this block, but will keep it for now (at least until merge done). 3197 } 3198 3199 threadLoop_exit(); 3200 3201 if (!mStandby) { 3202 threadLoop_standby(); 3203 mStandby = true; 3204 } 3205 3206 releaseWakeLock(); 3207 mWakeLockUids.clear(); 3208 mActiveTracksGeneration++; 3209 3210 ALOGV("Thread %p type %d exiting", this, mType); 3211 return false; 3212} 3213 3214// removeTracks_l() must be called with ThreadBase::mLock held 3215void AudioFlinger::PlaybackThread::removeTracks_l(const Vector< sp<Track> >& tracksToRemove) 3216{ 3217 size_t count = tracksToRemove.size(); 3218 if (count > 0) { 3219 for (size_t i=0 ; i<count ; i++) { 3220 const sp<Track>& track = tracksToRemove.itemAt(i); 3221 mActiveTracks.remove(track); 3222 mWakeLockUids.remove(track->uid()); 3223 mActiveTracksGeneration++; 3224 ALOGV("removeTracks_l removing track on session %d", track->sessionId()); 3225 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 3226 if (chain != 0) { 3227 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), 3228 track->sessionId()); 3229 chain->decActiveTrackCnt(); 3230 } 3231 if (track->isTerminated()) { 3232 removeTrack_l(track); 3233 } 3234 } 3235 } 3236 3237} 3238 3239status_t AudioFlinger::PlaybackThread::getTimestamp_l(AudioTimestamp& timestamp) 3240{ 3241 if (mNormalSink != 0) { 3242 ExtendedTimestamp ets; 3243 status_t status = mNormalSink->getTimestamp(ets); 3244 if (status == NO_ERROR) { 3245 status = ets.getBestTimestamp(×tamp); 3246 } 3247 return status; 3248 } 3249 if ((mType == OFFLOAD || mType == DIRECT) 3250 && mOutput != NULL && mOutput->stream->get_presentation_position) { 3251 uint64_t position64; 3252 int ret = mOutput->getPresentationPosition(&position64, ×tamp.mTime); 3253 if (ret == 0) { 3254 timestamp.mPosition = (uint32_t)position64; 3255 return NO_ERROR; 3256 } 3257 } 3258 return INVALID_OPERATION; 3259} 3260 3261status_t AudioFlinger::MixerThread::createAudioPatch_l(const struct audio_patch *patch, 3262 audio_patch_handle_t *handle) 3263{ 3264 AutoPark<FastMixer> park(mFastMixer); 3265 3266 status_t status = PlaybackThread::createAudioPatch_l(patch, handle); 3267 3268 return status; 3269} 3270 3271status_t AudioFlinger::PlaybackThread::createAudioPatch_l(const struct audio_patch *patch, 3272 audio_patch_handle_t *handle) 3273{ 3274 status_t status = NO_ERROR; 3275 3276 // store new device and send to effects 3277 audio_devices_t type = AUDIO_DEVICE_NONE; 3278 for (unsigned int i = 0; i < patch->num_sinks; i++) { 3279 type |= patch->sinks[i].ext.device.type; 3280 } 3281 3282#ifdef ADD_BATTERY_DATA 3283 // when changing the audio output device, call addBatteryData to notify 3284 // the change 3285 if (mOutDevice != type) { 3286 uint32_t params = 0; 3287 // check whether speaker is on 3288 if (type & AUDIO_DEVICE_OUT_SPEAKER) { 3289 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 3290 } 3291 3292 audio_devices_t deviceWithoutSpeaker 3293 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 3294 // check if any other device (except speaker) is on 3295 if (type & deviceWithoutSpeaker) { 3296 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 3297 } 3298 3299 if (params != 0) { 3300 addBatteryData(params); 3301 } 3302 } 3303#endif 3304 3305 for (size_t i = 0; i < mEffectChains.size(); i++) { 3306 mEffectChains[i]->setDevice_l(type); 3307 } 3308 3309 // mPrevOutDevice is the latest device set by createAudioPatch_l(). It is not set when 3310 // the thread is created so that the first patch creation triggers an ioConfigChanged callback 3311 bool configChanged = mPrevOutDevice != type; 3312 mOutDevice = type; 3313 mPatch = *patch; 3314 3315 if (mOutput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 3316 audio_hw_device_t *hwDevice = mOutput->audioHwDev->hwDevice(); 3317 status = hwDevice->create_audio_patch(hwDevice, 3318 patch->num_sources, 3319 patch->sources, 3320 patch->num_sinks, 3321 patch->sinks, 3322 handle); 3323 } else { 3324 char *address; 3325 if (strcmp(patch->sinks[0].ext.device.address, "") != 0) { 3326 //FIXME: we only support address on first sink with HAL version < 3.0 3327 address = audio_device_address_to_parameter( 3328 patch->sinks[0].ext.device.type, 3329 patch->sinks[0].ext.device.address); 3330 } else { 3331 address = (char *)calloc(1, 1); 3332 } 3333 AudioParameter param = AudioParameter(String8(address)); 3334 free(address); 3335 param.addInt(String8(AUDIO_PARAMETER_STREAM_ROUTING), (int)type); 3336 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3337 param.toString().string()); 3338 *handle = AUDIO_PATCH_HANDLE_NONE; 3339 } 3340 if (configChanged) { 3341 mPrevOutDevice = type; 3342 sendIoConfigEvent_l(AUDIO_OUTPUT_CONFIG_CHANGED); 3343 } 3344 return status; 3345} 3346 3347status_t AudioFlinger::MixerThread::releaseAudioPatch_l(const audio_patch_handle_t handle) 3348{ 3349 AutoPark<FastMixer> park(mFastMixer); 3350 3351 status_t status = PlaybackThread::releaseAudioPatch_l(handle); 3352 3353 return status; 3354} 3355 3356status_t AudioFlinger::PlaybackThread::releaseAudioPatch_l(const audio_patch_handle_t handle) 3357{ 3358 status_t status = NO_ERROR; 3359 3360 mOutDevice = AUDIO_DEVICE_NONE; 3361 3362 if (mOutput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 3363 audio_hw_device_t *hwDevice = mOutput->audioHwDev->hwDevice(); 3364 status = hwDevice->release_audio_patch(hwDevice, handle); 3365 } else { 3366 AudioParameter param; 3367 param.addInt(String8(AUDIO_PARAMETER_STREAM_ROUTING), 0); 3368 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3369 param.toString().string()); 3370 } 3371 return status; 3372} 3373 3374void AudioFlinger::PlaybackThread::addPatchTrack(const sp<PatchTrack>& track) 3375{ 3376 Mutex::Autolock _l(mLock); 3377 mTracks.add(track); 3378} 3379 3380void AudioFlinger::PlaybackThread::deletePatchTrack(const sp<PatchTrack>& track) 3381{ 3382 Mutex::Autolock _l(mLock); 3383 destroyTrack_l(track); 3384} 3385 3386void AudioFlinger::PlaybackThread::getAudioPortConfig(struct audio_port_config *config) 3387{ 3388 ThreadBase::getAudioPortConfig(config); 3389 config->role = AUDIO_PORT_ROLE_SOURCE; 3390 config->ext.mix.hw_module = mOutput->audioHwDev->handle(); 3391 config->ext.mix.usecase.stream = AUDIO_STREAM_DEFAULT; 3392} 3393 3394// ---------------------------------------------------------------------------- 3395 3396AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 3397 audio_io_handle_t id, audio_devices_t device, bool systemReady, type_t type) 3398 : PlaybackThread(audioFlinger, output, id, device, type, systemReady), 3399 // mAudioMixer below 3400 // mFastMixer below 3401 mFastMixerFutex(0), 3402 mMasterMono(false) 3403 // mOutputSink below 3404 // mPipeSink below 3405 // mNormalSink below 3406{ 3407 ALOGV("MixerThread() id=%d device=%#x type=%d", id, device, type); 3408 ALOGV("mSampleRate=%u, mChannelMask=%#x, mChannelCount=%u, mFormat=%d, mFrameSize=%zu, " 3409 "mFrameCount=%zu, mNormalFrameCount=%zu", 3410 mSampleRate, mChannelMask, mChannelCount, mFormat, mFrameSize, mFrameCount, 3411 mNormalFrameCount); 3412 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 3413 3414 if (type == DUPLICATING) { 3415 // The Duplicating thread uses the AudioMixer and delivers data to OutputTracks 3416 // (downstream MixerThreads) in DuplicatingThread::threadLoop_write(). 3417 // Do not create or use mFastMixer, mOutputSink, mPipeSink, or mNormalSink. 3418 return; 3419 } 3420 // create an NBAIO sink for the HAL output stream, and negotiate 3421 mOutputSink = new AudioStreamOutSink(output->stream); 3422 size_t numCounterOffers = 0; 3423 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount, mFormat)}; 3424#if !LOG_NDEBUG 3425 ssize_t index = 3426#else 3427 (void) 3428#endif 3429 mOutputSink->negotiate(offers, 1, NULL, numCounterOffers); 3430 ALOG_ASSERT(index == 0); 3431 3432 // initialize fast mixer depending on configuration 3433 bool initFastMixer; 3434 switch (kUseFastMixer) { 3435 case FastMixer_Never: 3436 initFastMixer = false; 3437 break; 3438 case FastMixer_Always: 3439 initFastMixer = true; 3440 break; 3441 case FastMixer_Static: 3442 case FastMixer_Dynamic: 3443 initFastMixer = mFrameCount < mNormalFrameCount; 3444 break; 3445 } 3446 if (initFastMixer) { 3447 audio_format_t fastMixerFormat; 3448 if (mMixerBufferEnabled && mEffectBufferEnabled) { 3449 fastMixerFormat = AUDIO_FORMAT_PCM_FLOAT; 3450 } else { 3451 fastMixerFormat = AUDIO_FORMAT_PCM_16_BIT; 3452 } 3453 if (mFormat != fastMixerFormat) { 3454 // change our Sink format to accept our intermediate precision 3455 mFormat = fastMixerFormat; 3456 free(mSinkBuffer); 3457 mFrameSize = mChannelCount * audio_bytes_per_sample(mFormat); 3458 const size_t sinkBufferSize = mNormalFrameCount * mFrameSize; 3459 (void)posix_memalign(&mSinkBuffer, 32, sinkBufferSize); 3460 } 3461 3462 // create a MonoPipe to connect our submix to FastMixer 3463 NBAIO_Format format = mOutputSink->format(); 3464#ifdef TEE_SINK 3465 NBAIO_Format origformat = format; 3466#endif 3467 // adjust format to match that of the Fast Mixer 3468 ALOGV("format changed from %d to %d", format.mFormat, fastMixerFormat); 3469 format.mFormat = fastMixerFormat; 3470 format.mFrameSize = audio_bytes_per_sample(format.mFormat) * format.mChannelCount; 3471 3472 // This pipe depth compensates for scheduling latency of the normal mixer thread. 3473 // When it wakes up after a maximum latency, it runs a few cycles quickly before 3474 // finally blocking. Note the pipe implementation rounds up the request to a power of 2. 3475 MonoPipe *monoPipe = new MonoPipe(mNormalFrameCount * 4, format, true /*writeCanBlock*/); 3476 const NBAIO_Format offers[1] = {format}; 3477 size_t numCounterOffers = 0; 3478#if !LOG_NDEBUG || defined(TEE_SINK) 3479 ssize_t index = 3480#else 3481 (void) 3482#endif 3483 monoPipe->negotiate(offers, 1, NULL, numCounterOffers); 3484 ALOG_ASSERT(index == 0); 3485 monoPipe->setAvgFrames((mScreenState & 1) ? 3486 (monoPipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 3487 mPipeSink = monoPipe; 3488 3489#ifdef TEE_SINK 3490 if (mTeeSinkOutputEnabled) { 3491 // create a Pipe to archive a copy of FastMixer's output for dumpsys 3492 Pipe *teeSink = new Pipe(mTeeSinkOutputFrames, origformat); 3493 const NBAIO_Format offers2[1] = {origformat}; 3494 numCounterOffers = 0; 3495 index = teeSink->negotiate(offers2, 1, NULL, numCounterOffers); 3496 ALOG_ASSERT(index == 0); 3497 mTeeSink = teeSink; 3498 PipeReader *teeSource = new PipeReader(*teeSink); 3499 numCounterOffers = 0; 3500 index = teeSource->negotiate(offers2, 1, NULL, numCounterOffers); 3501 ALOG_ASSERT(index == 0); 3502 mTeeSource = teeSource; 3503 } 3504#endif 3505 3506 // create fast mixer and configure it initially with just one fast track for our submix 3507 mFastMixer = new FastMixer(); 3508 FastMixerStateQueue *sq = mFastMixer->sq(); 3509#ifdef STATE_QUEUE_DUMP 3510 sq->setObserverDump(&mStateQueueObserverDump); 3511 sq->setMutatorDump(&mStateQueueMutatorDump); 3512#endif 3513 FastMixerState *state = sq->begin(); 3514 FastTrack *fastTrack = &state->mFastTracks[0]; 3515 // wrap the source side of the MonoPipe to make it an AudioBufferProvider 3516 fastTrack->mBufferProvider = new SourceAudioBufferProvider(new MonoPipeReader(monoPipe)); 3517 fastTrack->mVolumeProvider = NULL; 3518 fastTrack->mChannelMask = mChannelMask; // mPipeSink channel mask for audio to FastMixer 3519 fastTrack->mFormat = mFormat; // mPipeSink format for audio to FastMixer 3520 fastTrack->mGeneration++; 3521 state->mFastTracksGen++; 3522 state->mTrackMask = 1; 3523 // fast mixer will use the HAL output sink 3524 state->mOutputSink = mOutputSink.get(); 3525 state->mOutputSinkGen++; 3526 state->mFrameCount = mFrameCount; 3527 state->mCommand = FastMixerState::COLD_IDLE; 3528 // already done in constructor initialization list 3529 //mFastMixerFutex = 0; 3530 state->mColdFutexAddr = &mFastMixerFutex; 3531 state->mColdGen++; 3532 state->mDumpState = &mFastMixerDumpState; 3533#ifdef TEE_SINK 3534 state->mTeeSink = mTeeSink.get(); 3535#endif 3536 mFastMixerNBLogWriter = audioFlinger->newWriter_l(kFastMixerLogSize, "FastMixer"); 3537 state->mNBLogWriter = mFastMixerNBLogWriter.get(); 3538 sq->end(); 3539 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3540 3541 // start the fast mixer 3542 mFastMixer->run("FastMixer", PRIORITY_URGENT_AUDIO); 3543 pid_t tid = mFastMixer->getTid(); 3544 sendPrioConfigEvent(getpid_cached, tid, kPriorityFastMixer); 3545 3546#ifdef AUDIO_WATCHDOG 3547 // create and start the watchdog 3548 mAudioWatchdog = new AudioWatchdog(); 3549 mAudioWatchdog->setDump(&mAudioWatchdogDump); 3550 mAudioWatchdog->run("AudioWatchdog", PRIORITY_URGENT_AUDIO); 3551 tid = mAudioWatchdog->getTid(); 3552 sendPrioConfigEvent(getpid_cached, tid, kPriorityFastMixer); 3553#endif 3554 3555 } 3556 3557 switch (kUseFastMixer) { 3558 case FastMixer_Never: 3559 case FastMixer_Dynamic: 3560 mNormalSink = mOutputSink; 3561 break; 3562 case FastMixer_Always: 3563 mNormalSink = mPipeSink; 3564 break; 3565 case FastMixer_Static: 3566 mNormalSink = initFastMixer ? mPipeSink : mOutputSink; 3567 break; 3568 } 3569} 3570 3571AudioFlinger::MixerThread::~MixerThread() 3572{ 3573 if (mFastMixer != 0) { 3574 FastMixerStateQueue *sq = mFastMixer->sq(); 3575 FastMixerState *state = sq->begin(); 3576 if (state->mCommand == FastMixerState::COLD_IDLE) { 3577 int32_t old = android_atomic_inc(&mFastMixerFutex); 3578 if (old == -1) { 3579 (void) syscall(__NR_futex, &mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 3580 } 3581 } 3582 state->mCommand = FastMixerState::EXIT; 3583 sq->end(); 3584 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3585 mFastMixer->join(); 3586 // Though the fast mixer thread has exited, it's state queue is still valid. 3587 // We'll use that extract the final state which contains one remaining fast track 3588 // corresponding to our sub-mix. 3589 state = sq->begin(); 3590 ALOG_ASSERT(state->mTrackMask == 1); 3591 FastTrack *fastTrack = &state->mFastTracks[0]; 3592 ALOG_ASSERT(fastTrack->mBufferProvider != NULL); 3593 delete fastTrack->mBufferProvider; 3594 sq->end(false /*didModify*/); 3595 mFastMixer.clear(); 3596#ifdef AUDIO_WATCHDOG 3597 if (mAudioWatchdog != 0) { 3598 mAudioWatchdog->requestExit(); 3599 mAudioWatchdog->requestExitAndWait(); 3600 mAudioWatchdog.clear(); 3601 } 3602#endif 3603 } 3604 mAudioFlinger->unregisterWriter(mFastMixerNBLogWriter); 3605 delete mAudioMixer; 3606} 3607 3608 3609uint32_t AudioFlinger::MixerThread::correctLatency_l(uint32_t latency) const 3610{ 3611 if (mFastMixer != 0) { 3612 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 3613 latency += (pipe->getAvgFrames() * 1000) / mSampleRate; 3614 } 3615 return latency; 3616} 3617 3618 3619void AudioFlinger::MixerThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) 3620{ 3621 PlaybackThread::threadLoop_removeTracks(tracksToRemove); 3622} 3623 3624ssize_t AudioFlinger::MixerThread::threadLoop_write() 3625{ 3626 // FIXME we should only do one push per cycle; confirm this is true 3627 // Start the fast mixer if it's not already running 3628 if (mFastMixer != 0) { 3629 FastMixerStateQueue *sq = mFastMixer->sq(); 3630 FastMixerState *state = sq->begin(); 3631 if (state->mCommand != FastMixerState::MIX_WRITE && 3632 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)) { 3633 if (state->mCommand == FastMixerState::COLD_IDLE) { 3634 3635 // FIXME workaround for first HAL write being CPU bound on some devices 3636 ATRACE_BEGIN("write"); 3637 mOutput->write((char *)mSinkBuffer, 0); 3638 ATRACE_END(); 3639 3640 int32_t old = android_atomic_inc(&mFastMixerFutex); 3641 if (old == -1) { 3642 (void) syscall(__NR_futex, &mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 3643 } 3644#ifdef AUDIO_WATCHDOG 3645 if (mAudioWatchdog != 0) { 3646 mAudioWatchdog->resume(); 3647 } 3648#endif 3649 } 3650 state->mCommand = FastMixerState::MIX_WRITE; 3651#ifdef FAST_THREAD_STATISTICS 3652 mFastMixerDumpState.increaseSamplingN(mAudioFlinger->isLowRamDevice() ? 3653 FastThreadDumpState::kSamplingNforLowRamDevice : FastThreadDumpState::kSamplingN); 3654#endif 3655 sq->end(); 3656 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3657 if (kUseFastMixer == FastMixer_Dynamic) { 3658 mNormalSink = mPipeSink; 3659 } 3660 } else { 3661 sq->end(false /*didModify*/); 3662 } 3663 } 3664 return PlaybackThread::threadLoop_write(); 3665} 3666 3667void AudioFlinger::MixerThread::threadLoop_standby() 3668{ 3669 // Idle the fast mixer if it's currently running 3670 if (mFastMixer != 0) { 3671 FastMixerStateQueue *sq = mFastMixer->sq(); 3672 FastMixerState *state = sq->begin(); 3673 if (!(state->mCommand & FastMixerState::IDLE)) { 3674 state->mCommand = FastMixerState::COLD_IDLE; 3675 state->mColdFutexAddr = &mFastMixerFutex; 3676 state->mColdGen++; 3677 mFastMixerFutex = 0; 3678 sq->end(); 3679 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 3680 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3681 if (kUseFastMixer == FastMixer_Dynamic) { 3682 mNormalSink = mOutputSink; 3683 } 3684#ifdef AUDIO_WATCHDOG 3685 if (mAudioWatchdog != 0) { 3686 mAudioWatchdog->pause(); 3687 } 3688#endif 3689 } else { 3690 sq->end(false /*didModify*/); 3691 } 3692 } 3693 PlaybackThread::threadLoop_standby(); 3694} 3695 3696bool AudioFlinger::PlaybackThread::waitingAsyncCallback_l() 3697{ 3698 return false; 3699} 3700 3701bool AudioFlinger::PlaybackThread::shouldStandby_l() 3702{ 3703 return !mStandby; 3704} 3705 3706bool AudioFlinger::PlaybackThread::waitingAsyncCallback() 3707{ 3708 Mutex::Autolock _l(mLock); 3709 return waitingAsyncCallback_l(); 3710} 3711 3712// shared by MIXER and DIRECT, overridden by DUPLICATING 3713void AudioFlinger::PlaybackThread::threadLoop_standby() 3714{ 3715 ALOGV("Audio hardware entering standby, mixer %p, suspend count %d", this, mSuspended); 3716 mOutput->standby(); 3717 if (mUseAsyncWrite != 0) { 3718 // discard any pending drain or write ack by incrementing sequence 3719 mWriteAckSequence = (mWriteAckSequence + 2) & ~1; 3720 mDrainSequence = (mDrainSequence + 2) & ~1; 3721 ALOG_ASSERT(mCallbackThread != 0); 3722 mCallbackThread->setWriteBlocked(mWriteAckSequence); 3723 mCallbackThread->setDraining(mDrainSequence); 3724 } 3725 mHwPaused = false; 3726} 3727 3728void AudioFlinger::PlaybackThread::onAddNewTrack_l() 3729{ 3730 ALOGV("signal playback thread"); 3731 broadcast_l(); 3732} 3733 3734void AudioFlinger::MixerThread::threadLoop_mix() 3735{ 3736 // mix buffers... 3737 mAudioMixer->process(); 3738 mCurrentWriteLength = mSinkBufferSize; 3739 // increase sleep time progressively when application underrun condition clears. 3740 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 3741 // that a steady state of alternating ready/not ready conditions keeps the sleep time 3742 // such that we would underrun the audio HAL. 3743 if ((mSleepTimeUs == 0) && (sleepTimeShift > 0)) { 3744 sleepTimeShift--; 3745 } 3746 mSleepTimeUs = 0; 3747 mStandbyTimeNs = systemTime() + mStandbyDelayNs; 3748 //TODO: delay standby when effects have a tail 3749 3750} 3751 3752void AudioFlinger::MixerThread::threadLoop_sleepTime() 3753{ 3754 // If no tracks are ready, sleep once for the duration of an output 3755 // buffer size, then write 0s to the output 3756 if (mSleepTimeUs == 0) { 3757 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3758 mSleepTimeUs = mActiveSleepTimeUs >> sleepTimeShift; 3759 if (mSleepTimeUs < kMinThreadSleepTimeUs) { 3760 mSleepTimeUs = kMinThreadSleepTimeUs; 3761 } 3762 // reduce sleep time in case of consecutive application underruns to avoid 3763 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 3764 // duration we would end up writing less data than needed by the audio HAL if 3765 // the condition persists. 3766 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 3767 sleepTimeShift++; 3768 } 3769 } else { 3770 mSleepTimeUs = mIdleSleepTimeUs; 3771 } 3772 } else if (mBytesWritten != 0 || (mMixerStatus == MIXER_TRACKS_ENABLED)) { 3773 // clear out mMixerBuffer or mSinkBuffer, to ensure buffers are cleared 3774 // before effects processing or output. 3775 if (mMixerBufferValid) { 3776 memset(mMixerBuffer, 0, mMixerBufferSize); 3777 } else { 3778 memset(mSinkBuffer, 0, mSinkBufferSize); 3779 } 3780 mSleepTimeUs = 0; 3781 ALOGV_IF(mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED), 3782 "anticipated start"); 3783 } 3784 // TODO add standby time extension fct of effect tail 3785} 3786 3787// prepareTracks_l() must be called with ThreadBase::mLock held 3788AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 3789 Vector< sp<Track> > *tracksToRemove) 3790{ 3791 3792 mixer_state mixerStatus = MIXER_IDLE; 3793 // find out which tracks need to be processed 3794 size_t count = mActiveTracks.size(); 3795 size_t mixedTracks = 0; 3796 size_t tracksWithEffect = 0; 3797 // counts only _active_ fast tracks 3798 size_t fastTracks = 0; 3799 uint32_t resetMask = 0; // bit mask of fast tracks that need to be reset 3800 3801 float masterVolume = mMasterVolume; 3802 bool masterMute = mMasterMute; 3803 3804 if (masterMute) { 3805 masterVolume = 0; 3806 } 3807 // Delegate master volume control to effect in output mix effect chain if needed 3808 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 3809 if (chain != 0) { 3810 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 3811 chain->setVolume_l(&v, &v); 3812 masterVolume = (float)((v + (1 << 23)) >> 24); 3813 chain.clear(); 3814 } 3815 3816 // prepare a new state to push 3817 FastMixerStateQueue *sq = NULL; 3818 FastMixerState *state = NULL; 3819 bool didModify = false; 3820 FastMixerStateQueue::block_t block = FastMixerStateQueue::BLOCK_UNTIL_PUSHED; 3821 if (mFastMixer != 0) { 3822 sq = mFastMixer->sq(); 3823 state = sq->begin(); 3824 } 3825 3826 mMixerBufferValid = false; // mMixerBuffer has no valid data until appropriate tracks found. 3827 mEffectBufferValid = false; // mEffectBuffer has no valid data until tracks found. 3828 3829 for (size_t i=0 ; i<count ; i++) { 3830 const sp<Track> t = mActiveTracks[i].promote(); 3831 if (t == 0) { 3832 continue; 3833 } 3834 3835 // this const just means the local variable doesn't change 3836 Track* const track = t.get(); 3837 3838 // process fast tracks 3839 if (track->isFastTrack()) { 3840 3841 // It's theoretically possible (though unlikely) for a fast track to be created 3842 // and then removed within the same normal mix cycle. This is not a problem, as 3843 // the track never becomes active so it's fast mixer slot is never touched. 3844 // The converse, of removing an (active) track and then creating a new track 3845 // at the identical fast mixer slot within the same normal mix cycle, 3846 // is impossible because the slot isn't marked available until the end of each cycle. 3847 int j = track->mFastIndex; 3848 ALOG_ASSERT(0 < j && j < (int)FastMixerState::sMaxFastTracks); 3849 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << j))); 3850 FastTrack *fastTrack = &state->mFastTracks[j]; 3851 3852 // Determine whether the track is currently in underrun condition, 3853 // and whether it had a recent underrun. 3854 FastTrackDump *ftDump = &mFastMixerDumpState.mTracks[j]; 3855 FastTrackUnderruns underruns = ftDump->mUnderruns; 3856 uint32_t recentFull = (underruns.mBitFields.mFull - 3857 track->mObservedUnderruns.mBitFields.mFull) & UNDERRUN_MASK; 3858 uint32_t recentPartial = (underruns.mBitFields.mPartial - 3859 track->mObservedUnderruns.mBitFields.mPartial) & UNDERRUN_MASK; 3860 uint32_t recentEmpty = (underruns.mBitFields.mEmpty - 3861 track->mObservedUnderruns.mBitFields.mEmpty) & UNDERRUN_MASK; 3862 uint32_t recentUnderruns = recentPartial + recentEmpty; 3863 track->mObservedUnderruns = underruns; 3864 // don't count underruns that occur while stopping or pausing 3865 // or stopped which can occur when flush() is called while active 3866 if (!(track->isStopping() || track->isPausing() || track->isStopped()) && 3867 recentUnderruns > 0) { 3868 // FIXME fast mixer will pull & mix partial buffers, but we count as a full underrun 3869 track->mAudioTrackServerProxy->tallyUnderrunFrames(recentUnderruns * mFrameCount); 3870 } else { 3871 track->mAudioTrackServerProxy->tallyUnderrunFrames(0); 3872 } 3873 3874 // This is similar to the state machine for normal tracks, 3875 // with a few modifications for fast tracks. 3876 bool isActive = true; 3877 switch (track->mState) { 3878 case TrackBase::STOPPING_1: 3879 // track stays active in STOPPING_1 state until first underrun 3880 if (recentUnderruns > 0 || track->isTerminated()) { 3881 track->mState = TrackBase::STOPPING_2; 3882 } 3883 break; 3884 case TrackBase::PAUSING: 3885 // ramp down is not yet implemented 3886 track->setPaused(); 3887 break; 3888 case TrackBase::RESUMING: 3889 // ramp up is not yet implemented 3890 track->mState = TrackBase::ACTIVE; 3891 break; 3892 case TrackBase::ACTIVE: 3893 if (recentFull > 0 || recentPartial > 0) { 3894 // track has provided at least some frames recently: reset retry count 3895 track->mRetryCount = kMaxTrackRetries; 3896 } 3897 if (recentUnderruns == 0) { 3898 // no recent underruns: stay active 3899 break; 3900 } 3901 // there has recently been an underrun of some kind 3902 if (track->sharedBuffer() == 0) { 3903 // were any of the recent underruns "empty" (no frames available)? 3904 if (recentEmpty == 0) { 3905 // no, then ignore the partial underruns as they are allowed indefinitely 3906 break; 3907 } 3908 // there has recently been an "empty" underrun: decrement the retry counter 3909 if (--(track->mRetryCount) > 0) { 3910 break; 3911 } 3912 // indicate to client process that the track was disabled because of underrun; 3913 // it will then automatically call start() when data is available 3914 track->disable(); 3915 // remove from active list, but state remains ACTIVE [confusing but true] 3916 isActive = false; 3917 break; 3918 } 3919 // fall through 3920 case TrackBase::STOPPING_2: 3921 case TrackBase::PAUSED: 3922 case TrackBase::STOPPED: 3923 case TrackBase::FLUSHED: // flush() while active 3924 // Check for presentation complete if track is inactive 3925 // We have consumed all the buffers of this track. 3926 // This would be incomplete if we auto-paused on underrun 3927 { 3928 size_t audioHALFrames = 3929 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 3930 int64_t framesWritten = mBytesWritten / mFrameSize; 3931 if (!(mStandby || track->presentationComplete(framesWritten, audioHALFrames))) { 3932 // track stays in active list until presentation is complete 3933 break; 3934 } 3935 } 3936 if (track->isStopping_2()) { 3937 track->mState = TrackBase::STOPPED; 3938 } 3939 if (track->isStopped()) { 3940 // Can't reset directly, as fast mixer is still polling this track 3941 // track->reset(); 3942 // So instead mark this track as needing to be reset after push with ack 3943 resetMask |= 1 << i; 3944 } 3945 isActive = false; 3946 break; 3947 case TrackBase::IDLE: 3948 default: 3949 LOG_ALWAYS_FATAL("unexpected track state %d", track->mState); 3950 } 3951 3952 if (isActive) { 3953 // was it previously inactive? 3954 if (!(state->mTrackMask & (1 << j))) { 3955 ExtendedAudioBufferProvider *eabp = track; 3956 VolumeProvider *vp = track; 3957 fastTrack->mBufferProvider = eabp; 3958 fastTrack->mVolumeProvider = vp; 3959 fastTrack->mChannelMask = track->mChannelMask; 3960 fastTrack->mFormat = track->mFormat; 3961 fastTrack->mGeneration++; 3962 state->mTrackMask |= 1 << j; 3963 didModify = true; 3964 // no acknowledgement required for newly active tracks 3965 } 3966 // cache the combined master volume and stream type volume for fast mixer; this 3967 // lacks any synchronization or barrier so VolumeProvider may read a stale value 3968 track->mCachedVolume = masterVolume * mStreamTypes[track->streamType()].volume; 3969 ++fastTracks; 3970 } else { 3971 // was it previously active? 3972 if (state->mTrackMask & (1 << j)) { 3973 fastTrack->mBufferProvider = NULL; 3974 fastTrack->mGeneration++; 3975 state->mTrackMask &= ~(1 << j); 3976 didModify = true; 3977 // If any fast tracks were removed, we must wait for acknowledgement 3978 // because we're about to decrement the last sp<> on those tracks. 3979 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3980 } else { 3981 LOG_ALWAYS_FATAL("fast track %d should have been active; " 3982 "mState=%d, mTrackMask=%#x, recentUnderruns=%u, isShared=%d", 3983 j, track->mState, state->mTrackMask, recentUnderruns, 3984 track->sharedBuffer() != 0); 3985 } 3986 tracksToRemove->add(track); 3987 // Avoids a misleading display in dumpsys 3988 track->mObservedUnderruns.mBitFields.mMostRecent = UNDERRUN_FULL; 3989 } 3990 continue; 3991 } 3992 3993 { // local variable scope to avoid goto warning 3994 3995 audio_track_cblk_t* cblk = track->cblk(); 3996 3997 // The first time a track is added we wait 3998 // for all its buffers to be filled before processing it 3999 int name = track->name(); 4000 // make sure that we have enough frames to mix one full buffer. 4001 // enforce this condition only once to enable draining the buffer in case the client 4002 // app does not call stop() and relies on underrun to stop: 4003 // hence the test on (mMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 4004 // during last round 4005 size_t desiredFrames; 4006 const uint32_t sampleRate = track->mAudioTrackServerProxy->getSampleRate(); 4007 AudioPlaybackRate playbackRate = track->mAudioTrackServerProxy->getPlaybackRate(); 4008 4009 desiredFrames = sourceFramesNeededWithTimestretch( 4010 sampleRate, mNormalFrameCount, mSampleRate, playbackRate.mSpeed); 4011 // TODO: ONLY USED FOR LEGACY RESAMPLERS, remove when they are removed. 4012 // add frames already consumed but not yet released by the resampler 4013 // because mAudioTrackServerProxy->framesReady() will include these frames 4014 desiredFrames += mAudioMixer->getUnreleasedFrames(track->name()); 4015 4016 uint32_t minFrames = 1; 4017 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() && 4018 (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) { 4019 minFrames = desiredFrames; 4020 } 4021 4022 size_t framesReady = track->framesReady(); 4023 if (ATRACE_ENABLED()) { 4024 // I wish we had formatted trace names 4025 char traceName[16]; 4026 strcpy(traceName, "nRdy"); 4027 int name = track->name(); 4028 if (AudioMixer::TRACK0 <= name && 4029 name < (int) (AudioMixer::TRACK0 + AudioMixer::MAX_NUM_TRACKS)) { 4030 name -= AudioMixer::TRACK0; 4031 traceName[4] = (name / 10) + '0'; 4032 traceName[5] = (name % 10) + '0'; 4033 } else { 4034 traceName[4] = '?'; 4035 traceName[5] = '?'; 4036 } 4037 traceName[6] = '\0'; 4038 ATRACE_INT(traceName, framesReady); 4039 } 4040 if ((framesReady >= minFrames) && track->isReady() && 4041 !track->isPaused() && !track->isTerminated()) 4042 { 4043 ALOGVV("track %d s=%08x [OK] on thread %p", name, cblk->mServer, this); 4044 4045 mixedTracks++; 4046 4047 // track->mainBuffer() != mSinkBuffer or mMixerBuffer means 4048 // there is an effect chain connected to the track 4049 chain.clear(); 4050 if (track->mainBuffer() != mSinkBuffer && 4051 track->mainBuffer() != mMixerBuffer) { 4052 if (mEffectBufferEnabled) { 4053 mEffectBufferValid = true; // Later can set directly. 4054 } 4055 chain = getEffectChain_l(track->sessionId()); 4056 // Delegate volume control to effect in track effect chain if needed 4057 if (chain != 0) { 4058 tracksWithEffect++; 4059 } else { 4060 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on " 4061 "session %d", 4062 name, track->sessionId()); 4063 } 4064 } 4065 4066 4067 int param = AudioMixer::VOLUME; 4068 if (track->mFillingUpStatus == Track::FS_FILLED) { 4069 // no ramp for the first volume setting 4070 track->mFillingUpStatus = Track::FS_ACTIVE; 4071 if (track->mState == TrackBase::RESUMING) { 4072 track->mState = TrackBase::ACTIVE; 4073 param = AudioMixer::RAMP_VOLUME; 4074 } 4075 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 4076 // FIXME should not make a decision based on mServer 4077 } else if (cblk->mServer != 0) { 4078 // If the track is stopped before the first frame was mixed, 4079 // do not apply ramp 4080 param = AudioMixer::RAMP_VOLUME; 4081 } 4082 4083 // compute volume for this track 4084 uint32_t vl, vr; // in U8.24 integer format 4085 float vlf, vrf, vaf; // in [0.0, 1.0] float format 4086 if (track->isPausing() || mStreamTypes[track->streamType()].mute) { 4087 vl = vr = 0; 4088 vlf = vrf = vaf = 0.; 4089 if (track->isPausing()) { 4090 track->setPaused(); 4091 } 4092 } else { 4093 4094 // read original volumes with volume control 4095 float typeVolume = mStreamTypes[track->streamType()].volume; 4096 float v = masterVolume * typeVolume; 4097 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 4098 gain_minifloat_packed_t vlr = proxy->getVolumeLR(); 4099 vlf = float_from_gain(gain_minifloat_unpack_left(vlr)); 4100 vrf = float_from_gain(gain_minifloat_unpack_right(vlr)); 4101 // track volumes come from shared memory, so can't be trusted and must be clamped 4102 if (vlf > GAIN_FLOAT_UNITY) { 4103 ALOGV("Track left volume out of range: %.3g", vlf); 4104 vlf = GAIN_FLOAT_UNITY; 4105 } 4106 if (vrf > GAIN_FLOAT_UNITY) { 4107 ALOGV("Track right volume out of range: %.3g", vrf); 4108 vrf = GAIN_FLOAT_UNITY; 4109 } 4110 // now apply the master volume and stream type volume 4111 vlf *= v; 4112 vrf *= v; 4113 // assuming master volume and stream type volume each go up to 1.0, 4114 // then derive vl and vr as U8.24 versions for the effect chain 4115 const float scaleto8_24 = MAX_GAIN_INT * MAX_GAIN_INT; 4116 vl = (uint32_t) (scaleto8_24 * vlf); 4117 vr = (uint32_t) (scaleto8_24 * vrf); 4118 // vl and vr are now in U8.24 format 4119 uint16_t sendLevel = proxy->getSendLevel_U4_12(); 4120 // send level comes from shared memory and so may be corrupt 4121 if (sendLevel > MAX_GAIN_INT) { 4122 ALOGV("Track send level out of range: %04X", sendLevel); 4123 sendLevel = MAX_GAIN_INT; 4124 } 4125 // vaf is represented as [0.0, 1.0] float by rescaling sendLevel 4126 vaf = v * sendLevel * (1. / MAX_GAIN_INT); 4127 } 4128 4129 // Delegate volume control to effect in track effect chain if needed 4130 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 4131 // Do not ramp volume if volume is controlled by effect 4132 param = AudioMixer::VOLUME; 4133 // Update remaining floating point volume levels 4134 vlf = (float)vl / (1 << 24); 4135 vrf = (float)vr / (1 << 24); 4136 track->mHasVolumeController = true; 4137 } else { 4138 // force no volume ramp when volume controller was just disabled or removed 4139 // from effect chain to avoid volume spike 4140 if (track->mHasVolumeController) { 4141 param = AudioMixer::VOLUME; 4142 } 4143 track->mHasVolumeController = false; 4144 } 4145 4146 // XXX: these things DON'T need to be done each time 4147 mAudioMixer->setBufferProvider(name, track); 4148 mAudioMixer->enable(name); 4149 4150 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, &vlf); 4151 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, &vrf); 4152 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, &vaf); 4153 mAudioMixer->setParameter( 4154 name, 4155 AudioMixer::TRACK, 4156 AudioMixer::FORMAT, (void *)track->format()); 4157 mAudioMixer->setParameter( 4158 name, 4159 AudioMixer::TRACK, 4160 AudioMixer::CHANNEL_MASK, (void *)(uintptr_t)track->channelMask()); 4161 mAudioMixer->setParameter( 4162 name, 4163 AudioMixer::TRACK, 4164 AudioMixer::MIXER_CHANNEL_MASK, (void *)(uintptr_t)mChannelMask); 4165 // limit track sample rate to 2 x output sample rate, which changes at re-configuration 4166 uint32_t maxSampleRate = mSampleRate * AUDIO_RESAMPLER_DOWN_RATIO_MAX; 4167 uint32_t reqSampleRate = track->mAudioTrackServerProxy->getSampleRate(); 4168 if (reqSampleRate == 0) { 4169 reqSampleRate = mSampleRate; 4170 } else if (reqSampleRate > maxSampleRate) { 4171 reqSampleRate = maxSampleRate; 4172 } 4173 mAudioMixer->setParameter( 4174 name, 4175 AudioMixer::RESAMPLE, 4176 AudioMixer::SAMPLE_RATE, 4177 (void *)(uintptr_t)reqSampleRate); 4178 4179 AudioPlaybackRate playbackRate = track->mAudioTrackServerProxy->getPlaybackRate(); 4180 mAudioMixer->setParameter( 4181 name, 4182 AudioMixer::TIMESTRETCH, 4183 AudioMixer::PLAYBACK_RATE, 4184 &playbackRate); 4185 4186 /* 4187 * Select the appropriate output buffer for the track. 4188 * 4189 * Tracks with effects go into their own effects chain buffer 4190 * and from there into either mEffectBuffer or mSinkBuffer. 4191 * 4192 * Other tracks can use mMixerBuffer for higher precision 4193 * channel accumulation. If this buffer is enabled 4194 * (mMixerBufferEnabled true), then selected tracks will accumulate 4195 * into it. 4196 * 4197 */ 4198 if (mMixerBufferEnabled 4199 && (track->mainBuffer() == mSinkBuffer 4200 || track->mainBuffer() == mMixerBuffer)) { 4201 mAudioMixer->setParameter( 4202 name, 4203 AudioMixer::TRACK, 4204 AudioMixer::MIXER_FORMAT, (void *)mMixerBufferFormat); 4205 mAudioMixer->setParameter( 4206 name, 4207 AudioMixer::TRACK, 4208 AudioMixer::MAIN_BUFFER, (void *)mMixerBuffer); 4209 // TODO: override track->mainBuffer()? 4210 mMixerBufferValid = true; 4211 } else { 4212 mAudioMixer->setParameter( 4213 name, 4214 AudioMixer::TRACK, 4215 AudioMixer::MIXER_FORMAT, (void *)AUDIO_FORMAT_PCM_16_BIT); 4216 mAudioMixer->setParameter( 4217 name, 4218 AudioMixer::TRACK, 4219 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 4220 } 4221 mAudioMixer->setParameter( 4222 name, 4223 AudioMixer::TRACK, 4224 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 4225 4226 // reset retry count 4227 track->mRetryCount = kMaxTrackRetries; 4228 4229 // If one track is ready, set the mixer ready if: 4230 // - the mixer was not ready during previous round OR 4231 // - no other track is not ready 4232 if (mMixerStatusIgnoringFastTracks != MIXER_TRACKS_READY || 4233 mixerStatus != MIXER_TRACKS_ENABLED) { 4234 mixerStatus = MIXER_TRACKS_READY; 4235 } 4236 } else { 4237 if (framesReady < desiredFrames && !track->isStopped() && !track->isPaused()) { 4238 ALOGV("track(%p) underrun, framesReady(%zu) < framesDesired(%zd)", 4239 track, framesReady, desiredFrames); 4240 track->mAudioTrackServerProxy->tallyUnderrunFrames(desiredFrames); 4241 } else { 4242 track->mAudioTrackServerProxy->tallyUnderrunFrames(0); 4243 } 4244 4245 // clear effect chain input buffer if an active track underruns to avoid sending 4246 // previous audio buffer again to effects 4247 chain = getEffectChain_l(track->sessionId()); 4248 if (chain != 0) { 4249 chain->clearInputBuffer(); 4250 } 4251 4252 ALOGVV("track %d s=%08x [NOT READY] on thread %p", name, cblk->mServer, this); 4253 if ((track->sharedBuffer() != 0) || track->isTerminated() || 4254 track->isStopped() || track->isPaused()) { 4255 // We have consumed all the buffers of this track. 4256 // Remove it from the list of active tracks. 4257 // TODO: use actual buffer filling status instead of latency when available from 4258 // audio HAL 4259 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 4260 int64_t framesWritten = mBytesWritten / mFrameSize; 4261 if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) { 4262 if (track->isStopped()) { 4263 track->reset(); 4264 } 4265 tracksToRemove->add(track); 4266 } 4267 } else { 4268 // No buffers for this track. Give it a few chances to 4269 // fill a buffer, then remove it from active list. 4270 if (--(track->mRetryCount) <= 0) { 4271 ALOGI("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 4272 tracksToRemove->add(track); 4273 // indicate to client process that the track was disabled because of underrun; 4274 // it will then automatically call start() when data is available 4275 track->disable(); 4276 // If one track is not ready, mark the mixer also not ready if: 4277 // - the mixer was ready during previous round OR 4278 // - no other track is ready 4279 } else if (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY || 4280 mixerStatus != MIXER_TRACKS_READY) { 4281 mixerStatus = MIXER_TRACKS_ENABLED; 4282 } 4283 } 4284 mAudioMixer->disable(name); 4285 } 4286 4287 } // local variable scope to avoid goto warning 4288 4289 } 4290 4291 // Push the new FastMixer state if necessary 4292 bool pauseAudioWatchdog = false; 4293 if (didModify) { 4294 state->mFastTracksGen++; 4295 // if the fast mixer was active, but now there are no fast tracks, then put it in cold idle 4296 if (kUseFastMixer == FastMixer_Dynamic && 4297 state->mCommand == FastMixerState::MIX_WRITE && state->mTrackMask <= 1) { 4298 state->mCommand = FastMixerState::COLD_IDLE; 4299 state->mColdFutexAddr = &mFastMixerFutex; 4300 state->mColdGen++; 4301 mFastMixerFutex = 0; 4302 if (kUseFastMixer == FastMixer_Dynamic) { 4303 mNormalSink = mOutputSink; 4304 } 4305 // If we go into cold idle, need to wait for acknowledgement 4306 // so that fast mixer stops doing I/O. 4307 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 4308 pauseAudioWatchdog = true; 4309 } 4310 } 4311 if (sq != NULL) { 4312 sq->end(didModify); 4313 sq->push(block); 4314 } 4315#ifdef AUDIO_WATCHDOG 4316 if (pauseAudioWatchdog && mAudioWatchdog != 0) { 4317 mAudioWatchdog->pause(); 4318 } 4319#endif 4320 4321 // Now perform the deferred reset on fast tracks that have stopped 4322 while (resetMask != 0) { 4323 size_t i = __builtin_ctz(resetMask); 4324 ALOG_ASSERT(i < count); 4325 resetMask &= ~(1 << i); 4326 sp<Track> t = mActiveTracks[i].promote(); 4327 if (t == 0) { 4328 continue; 4329 } 4330 Track* track = t.get(); 4331 ALOG_ASSERT(track->isFastTrack() && track->isStopped()); 4332 track->reset(); 4333 } 4334 4335 // remove all the tracks that need to be... 4336 removeTracks_l(*tracksToRemove); 4337 4338 if (getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX) != 0) { 4339 mEffectBufferValid = true; 4340 } 4341 4342 if (mEffectBufferValid) { 4343 // as long as there are effects we should clear the effects buffer, to avoid 4344 // passing a non-clean buffer to the effect chain 4345 memset(mEffectBuffer, 0, mEffectBufferSize); 4346 } 4347 // sink or mix buffer must be cleared if all tracks are connected to an 4348 // effect chain as in this case the mixer will not write to the sink or mix buffer 4349 // and track effects will accumulate into it 4350 if ((mBytesRemaining == 0) && ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 4351 (mixedTracks == 0 && fastTracks > 0))) { 4352 // FIXME as a performance optimization, should remember previous zero status 4353 if (mMixerBufferValid) { 4354 memset(mMixerBuffer, 0, mMixerBufferSize); 4355 // TODO: In testing, mSinkBuffer below need not be cleared because 4356 // the PlaybackThread::threadLoop() copies mMixerBuffer into mSinkBuffer 4357 // after mixing. 4358 // 4359 // To enforce this guarantee: 4360 // ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 4361 // (mixedTracks == 0 && fastTracks > 0)) 4362 // must imply MIXER_TRACKS_READY. 4363 // Later, we may clear buffers regardless, and skip much of this logic. 4364 } 4365 // FIXME as a performance optimization, should remember previous zero status 4366 memset(mSinkBuffer, 0, mNormalFrameCount * mFrameSize); 4367 } 4368 4369 // if any fast tracks, then status is ready 4370 mMixerStatusIgnoringFastTracks = mixerStatus; 4371 if (fastTracks > 0) { 4372 mixerStatus = MIXER_TRACKS_READY; 4373 } 4374 return mixerStatus; 4375} 4376 4377// getTrackName_l() must be called with ThreadBase::mLock held 4378int AudioFlinger::MixerThread::getTrackName_l(audio_channel_mask_t channelMask, 4379 audio_format_t format, audio_session_t sessionId) 4380{ 4381 return mAudioMixer->getTrackName(channelMask, format, sessionId); 4382} 4383 4384// deleteTrackName_l() must be called with ThreadBase::mLock held 4385void AudioFlinger::MixerThread::deleteTrackName_l(int name) 4386{ 4387 ALOGV("remove track (%d) and delete from mixer", name); 4388 mAudioMixer->deleteTrackName(name); 4389} 4390 4391// checkForNewParameter_l() must be called with ThreadBase::mLock held 4392bool AudioFlinger::MixerThread::checkForNewParameter_l(const String8& keyValuePair, 4393 status_t& status) 4394{ 4395 bool reconfig = false; 4396 bool a2dpDeviceChanged = false; 4397 4398 status = NO_ERROR; 4399 4400 AutoPark<FastMixer> park(mFastMixer); 4401 4402 AudioParameter param = AudioParameter(keyValuePair); 4403 int value; 4404 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 4405 reconfig = true; 4406 } 4407 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 4408 if (!isValidPcmSinkFormat((audio_format_t) value)) { 4409 status = BAD_VALUE; 4410 } else { 4411 // no need to save value, since it's constant 4412 reconfig = true; 4413 } 4414 } 4415 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 4416 if (!isValidPcmSinkChannelMask((audio_channel_mask_t) value)) { 4417 status = BAD_VALUE; 4418 } else { 4419 // no need to save value, since it's constant 4420 reconfig = true; 4421 } 4422 } 4423 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 4424 // do not accept frame count changes if tracks are open as the track buffer 4425 // size depends on frame count and correct behavior would not be guaranteed 4426 // if frame count is changed after track creation 4427 if (!mTracks.isEmpty()) { 4428 status = INVALID_OPERATION; 4429 } else { 4430 reconfig = true; 4431 } 4432 } 4433 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 4434#ifdef ADD_BATTERY_DATA 4435 // when changing the audio output device, call addBatteryData to notify 4436 // the change 4437 if (mOutDevice != value) { 4438 uint32_t params = 0; 4439 // check whether speaker is on 4440 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 4441 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 4442 } 4443 4444 audio_devices_t deviceWithoutSpeaker 4445 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 4446 // check if any other device (except speaker) is on 4447 if (value & deviceWithoutSpeaker) { 4448 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 4449 } 4450 4451 if (params != 0) { 4452 addBatteryData(params); 4453 } 4454 } 4455#endif 4456 4457 // forward device change to effects that have requested to be 4458 // aware of attached audio device. 4459 if (value != AUDIO_DEVICE_NONE) { 4460 a2dpDeviceChanged = 4461 (mOutDevice & AUDIO_DEVICE_OUT_ALL_A2DP) != (value & AUDIO_DEVICE_OUT_ALL_A2DP); 4462 mOutDevice = value; 4463 for (size_t i = 0; i < mEffectChains.size(); i++) { 4464 mEffectChains[i]->setDevice_l(mOutDevice); 4465 } 4466 } 4467 } 4468 4469 if (status == NO_ERROR) { 4470 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 4471 keyValuePair.string()); 4472 if (!mStandby && status == INVALID_OPERATION) { 4473 mOutput->standby(); 4474 mStandby = true; 4475 mBytesWritten = 0; 4476 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 4477 keyValuePair.string()); 4478 } 4479 if (status == NO_ERROR && reconfig) { 4480 readOutputParameters_l(); 4481 delete mAudioMixer; 4482 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 4483 for (size_t i = 0; i < mTracks.size() ; i++) { 4484 int name = getTrackName_l(mTracks[i]->mChannelMask, 4485 mTracks[i]->mFormat, mTracks[i]->mSessionId); 4486 if (name < 0) { 4487 break; 4488 } 4489 mTracks[i]->mName = name; 4490 } 4491 sendIoConfigEvent_l(AUDIO_OUTPUT_CONFIG_CHANGED); 4492 } 4493 } 4494 4495 return reconfig || a2dpDeviceChanged; 4496} 4497 4498 4499void AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 4500{ 4501 PlaybackThread::dumpInternals(fd, args); 4502 dprintf(fd, " Thread throttle time (msecs): %u\n", mThreadThrottleTimeMs); 4503 dprintf(fd, " AudioMixer tracks: 0x%08x\n", mAudioMixer->trackNames()); 4504 dprintf(fd, " Master mono: %s\n", mMasterMono ? "on" : "off"); 4505 4506 // Make a non-atomic copy of fast mixer dump state so it won't change underneath us 4507 // while we are dumping it. It may be inconsistent, but it won't mutate! 4508 // This is a large object so we place it on the heap. 4509 // FIXME 25972958: Need an intelligent copy constructor that does not touch unused pages. 4510 const FastMixerDumpState *copy = new FastMixerDumpState(mFastMixerDumpState); 4511 copy->dump(fd); 4512 delete copy; 4513 4514#ifdef STATE_QUEUE_DUMP 4515 // Similar for state queue 4516 StateQueueObserverDump observerCopy = mStateQueueObserverDump; 4517 observerCopy.dump(fd); 4518 StateQueueMutatorDump mutatorCopy = mStateQueueMutatorDump; 4519 mutatorCopy.dump(fd); 4520#endif 4521 4522#ifdef TEE_SINK 4523 // Write the tee output to a .wav file 4524 dumpTee(fd, mTeeSource, mId); 4525#endif 4526 4527#ifdef AUDIO_WATCHDOG 4528 if (mAudioWatchdog != 0) { 4529 // Make a non-atomic copy of audio watchdog dump so it won't change underneath us 4530 AudioWatchdogDump wdCopy = mAudioWatchdogDump; 4531 wdCopy.dump(fd); 4532 } 4533#endif 4534} 4535 4536uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const 4537{ 4538 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000) / 2; 4539} 4540 4541uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() const 4542{ 4543 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000); 4544} 4545 4546void AudioFlinger::MixerThread::cacheParameters_l() 4547{ 4548 PlaybackThread::cacheParameters_l(); 4549 4550 // FIXME: Relaxed timing because of a certain device that can't meet latency 4551 // Should be reduced to 2x after the vendor fixes the driver issue 4552 // increase threshold again due to low power audio mode. The way this warning 4553 // threshold is calculated and its usefulness should be reconsidered anyway. 4554 maxPeriod = seconds(mNormalFrameCount) / mSampleRate * 15; 4555} 4556 4557// ---------------------------------------------------------------------------- 4558 4559AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 4560 AudioStreamOut* output, audio_io_handle_t id, audio_devices_t device, bool systemReady) 4561 : PlaybackThread(audioFlinger, output, id, device, DIRECT, systemReady) 4562 // mLeftVolFloat, mRightVolFloat 4563{ 4564} 4565 4566AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 4567 AudioStreamOut* output, audio_io_handle_t id, uint32_t device, 4568 ThreadBase::type_t type, bool systemReady) 4569 : PlaybackThread(audioFlinger, output, id, device, type, systemReady) 4570 // mLeftVolFloat, mRightVolFloat 4571{ 4572} 4573 4574AudioFlinger::DirectOutputThread::~DirectOutputThread() 4575{ 4576} 4577 4578void AudioFlinger::DirectOutputThread::processVolume_l(Track *track, bool lastTrack) 4579{ 4580 float left, right; 4581 4582 if (mMasterMute || mStreamTypes[track->streamType()].mute) { 4583 left = right = 0; 4584 } else { 4585 float typeVolume = mStreamTypes[track->streamType()].volume; 4586 float v = mMasterVolume * typeVolume; 4587 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 4588 gain_minifloat_packed_t vlr = proxy->getVolumeLR(); 4589 left = float_from_gain(gain_minifloat_unpack_left(vlr)); 4590 if (left > GAIN_FLOAT_UNITY) { 4591 left = GAIN_FLOAT_UNITY; 4592 } 4593 left *= v; 4594 right = float_from_gain(gain_minifloat_unpack_right(vlr)); 4595 if (right > GAIN_FLOAT_UNITY) { 4596 right = GAIN_FLOAT_UNITY; 4597 } 4598 right *= v; 4599 } 4600 4601 if (lastTrack) { 4602 if (left != mLeftVolFloat || right != mRightVolFloat) { 4603 mLeftVolFloat = left; 4604 mRightVolFloat = right; 4605 4606 // Convert volumes from float to 8.24 4607 uint32_t vl = (uint32_t)(left * (1 << 24)); 4608 uint32_t vr = (uint32_t)(right * (1 << 24)); 4609 4610 // Delegate volume control to effect in track effect chain if needed 4611 // only one effect chain can be present on DirectOutputThread, so if 4612 // there is one, the track is connected to it 4613 if (!mEffectChains.isEmpty()) { 4614 mEffectChains[0]->setVolume_l(&vl, &vr); 4615 left = (float)vl / (1 << 24); 4616 right = (float)vr / (1 << 24); 4617 } 4618 if (mOutput->stream->set_volume) { 4619 mOutput->stream->set_volume(mOutput->stream, left, right); 4620 } 4621 } 4622 } 4623} 4624 4625void AudioFlinger::DirectOutputThread::onAddNewTrack_l() 4626{ 4627 sp<Track> previousTrack = mPreviousTrack.promote(); 4628 sp<Track> latestTrack = mLatestActiveTrack.promote(); 4629 4630 if (previousTrack != 0 && latestTrack != 0) { 4631 if (mType == DIRECT) { 4632 if (previousTrack.get() != latestTrack.get()) { 4633 mFlushPending = true; 4634 } 4635 } else /* mType == OFFLOAD */ { 4636 if (previousTrack->sessionId() != latestTrack->sessionId()) { 4637 mFlushPending = true; 4638 } 4639 } 4640 } 4641 PlaybackThread::onAddNewTrack_l(); 4642} 4643 4644AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l( 4645 Vector< sp<Track> > *tracksToRemove 4646) 4647{ 4648 size_t count = mActiveTracks.size(); 4649 mixer_state mixerStatus = MIXER_IDLE; 4650 bool doHwPause = false; 4651 bool doHwResume = false; 4652 4653 // find out which tracks need to be processed 4654 for (size_t i = 0; i < count; i++) { 4655 sp<Track> t = mActiveTracks[i].promote(); 4656 // The track died recently 4657 if (t == 0) { 4658 continue; 4659 } 4660 4661 if (t->isInvalid()) { 4662 ALOGW("An invalidated track shouldn't be in active list"); 4663 tracksToRemove->add(t); 4664 continue; 4665 } 4666 4667 Track* const track = t.get(); 4668#ifdef VERY_VERY_VERBOSE_LOGGING 4669 audio_track_cblk_t* cblk = track->cblk(); 4670#endif 4671 // Only consider last track started for volume and mixer state control. 4672 // In theory an older track could underrun and restart after the new one starts 4673 // but as we only care about the transition phase between two tracks on a 4674 // direct output, it is not a problem to ignore the underrun case. 4675 sp<Track> l = mLatestActiveTrack.promote(); 4676 bool last = l.get() == track; 4677 4678 if (track->isPausing()) { 4679 track->setPaused(); 4680 if (mHwSupportsPause && last && !mHwPaused) { 4681 doHwPause = true; 4682 mHwPaused = true; 4683 } 4684 tracksToRemove->add(track); 4685 } else if (track->isFlushPending()) { 4686 track->flushAck(); 4687 if (last) { 4688 mFlushPending = true; 4689 } 4690 } else if (track->isResumePending()) { 4691 track->resumeAck(); 4692 if (last && mHwPaused) { 4693 doHwResume = true; 4694 mHwPaused = false; 4695 } 4696 } 4697 4698 // The first time a track is added we wait 4699 // for all its buffers to be filled before processing it. 4700 // Allow draining the buffer in case the client 4701 // app does not call stop() and relies on underrun to stop: 4702 // hence the test on (track->mRetryCount > 1). 4703 // If retryCount<=1 then track is about to underrun and be removed. 4704 // Do not use a high threshold for compressed audio. 4705 uint32_t minFrames; 4706 if ((track->sharedBuffer() == 0) && !track->isStopping_1() && !track->isPausing() 4707 && (track->mRetryCount > 1) && audio_has_proportional_frames(mFormat)) { 4708 minFrames = mNormalFrameCount; 4709 } else { 4710 minFrames = 1; 4711 } 4712 4713 if ((track->framesReady() >= minFrames) && track->isReady() && !track->isPaused() && 4714 !track->isStopping_2() && !track->isStopped()) 4715 { 4716 ALOGVV("track %d s=%08x [OK]", track->name(), cblk->mServer); 4717 4718 if (track->mFillingUpStatus == Track::FS_FILLED) { 4719 track->mFillingUpStatus = Track::FS_ACTIVE; 4720 // make sure processVolume_l() will apply new volume even if 0 4721 mLeftVolFloat = mRightVolFloat = -1.0; 4722 if (!mHwSupportsPause) { 4723 track->resumeAck(); 4724 } 4725 } 4726 4727 // compute volume for this track 4728 processVolume_l(track, last); 4729 if (last) { 4730 sp<Track> previousTrack = mPreviousTrack.promote(); 4731 if (previousTrack != 0) { 4732 if (track != previousTrack.get()) { 4733 // Flush any data still being written from last track 4734 mBytesRemaining = 0; 4735 // Invalidate previous track to force a seek when resuming. 4736 previousTrack->invalidate(); 4737 } 4738 } 4739 mPreviousTrack = track; 4740 4741 // reset retry count 4742 track->mRetryCount = kMaxTrackRetriesDirect; 4743 mActiveTrack = t; 4744 mixerStatus = MIXER_TRACKS_READY; 4745 if (mHwPaused) { 4746 doHwResume = true; 4747 mHwPaused = false; 4748 } 4749 } 4750 } else { 4751 // clear effect chain input buffer if the last active track started underruns 4752 // to avoid sending previous audio buffer again to effects 4753 if (!mEffectChains.isEmpty() && last) { 4754 mEffectChains[0]->clearInputBuffer(); 4755 } 4756 if (track->isStopping_1()) { 4757 track->mState = TrackBase::STOPPING_2; 4758 if (last && mHwPaused) { 4759 doHwResume = true; 4760 mHwPaused = false; 4761 } 4762 } 4763 if ((track->sharedBuffer() != 0) || track->isStopped() || 4764 track->isStopping_2() || track->isPaused()) { 4765 // We have consumed all the buffers of this track. 4766 // Remove it from the list of active tracks. 4767 size_t audioHALFrames; 4768 if (audio_has_proportional_frames(mFormat)) { 4769 audioHALFrames = (latency_l() * mSampleRate) / 1000; 4770 } else { 4771 audioHALFrames = 0; 4772 } 4773 4774 int64_t framesWritten = mBytesWritten / mFrameSize; 4775 if (mStandby || !last || 4776 track->presentationComplete(framesWritten, audioHALFrames)) { 4777 if (track->isStopping_2()) { 4778 track->mState = TrackBase::STOPPED; 4779 } 4780 if (track->isStopped()) { 4781 track->reset(); 4782 } 4783 tracksToRemove->add(track); 4784 } 4785 } else { 4786 // No buffers for this track. Give it a few chances to 4787 // fill a buffer, then remove it from active list. 4788 // Only consider last track started for mixer state control 4789 if (--(track->mRetryCount) <= 0) { 4790 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 4791 tracksToRemove->add(track); 4792 // indicate to client process that the track was disabled because of underrun; 4793 // it will then automatically call start() when data is available 4794 track->disable(); 4795 } else if (last) { 4796 ALOGW("pause because of UNDERRUN, framesReady = %zu," 4797 "minFrames = %u, mFormat = %#x", 4798 track->framesReady(), minFrames, mFormat); 4799 mixerStatus = MIXER_TRACKS_ENABLED; 4800 if (mHwSupportsPause && !mHwPaused && !mStandby) { 4801 doHwPause = true; 4802 mHwPaused = true; 4803 } 4804 } 4805 } 4806 } 4807 } 4808 4809 // if an active track did not command a flush, check for pending flush on stopped tracks 4810 if (!mFlushPending) { 4811 for (size_t i = 0; i < mTracks.size(); i++) { 4812 if (mTracks[i]->isFlushPending()) { 4813 mTracks[i]->flushAck(); 4814 mFlushPending = true; 4815 } 4816 } 4817 } 4818 4819 // make sure the pause/flush/resume sequence is executed in the right order. 4820 // If a flush is pending and a track is active but the HW is not paused, force a HW pause 4821 // before flush and then resume HW. This can happen in case of pause/flush/resume 4822 // if resume is received before pause is executed. 4823 if (mHwSupportsPause && !mStandby && 4824 (doHwPause || (mFlushPending && !mHwPaused && (count != 0)))) { 4825 mOutput->stream->pause(mOutput->stream); 4826 } 4827 if (mFlushPending) { 4828 flushHw_l(); 4829 } 4830 if (mHwSupportsPause && !mStandby && doHwResume) { 4831 mOutput->stream->resume(mOutput->stream); 4832 } 4833 // remove all the tracks that need to be... 4834 removeTracks_l(*tracksToRemove); 4835 4836 return mixerStatus; 4837} 4838 4839void AudioFlinger::DirectOutputThread::threadLoop_mix() 4840{ 4841 size_t frameCount = mFrameCount; 4842 int8_t *curBuf = (int8_t *)mSinkBuffer; 4843 // output audio to hardware 4844 while (frameCount) { 4845 AudioBufferProvider::Buffer buffer; 4846 buffer.frameCount = frameCount; 4847 status_t status = mActiveTrack->getNextBuffer(&buffer); 4848 if (status != NO_ERROR || buffer.raw == NULL) { 4849 // no need to pad with 0 for compressed audio 4850 if (audio_has_proportional_frames(mFormat)) { 4851 memset(curBuf, 0, frameCount * mFrameSize); 4852 } 4853 break; 4854 } 4855 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 4856 frameCount -= buffer.frameCount; 4857 curBuf += buffer.frameCount * mFrameSize; 4858 mActiveTrack->releaseBuffer(&buffer); 4859 } 4860 mCurrentWriteLength = curBuf - (int8_t *)mSinkBuffer; 4861 mSleepTimeUs = 0; 4862 mStandbyTimeNs = systemTime() + mStandbyDelayNs; 4863 mActiveTrack.clear(); 4864} 4865 4866void AudioFlinger::DirectOutputThread::threadLoop_sleepTime() 4867{ 4868 // do not write to HAL when paused 4869 if (mHwPaused || (usesHwAvSync() && mStandby)) { 4870 mSleepTimeUs = mIdleSleepTimeUs; 4871 return; 4872 } 4873 if (mSleepTimeUs == 0) { 4874 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4875 mSleepTimeUs = mActiveSleepTimeUs; 4876 } else { 4877 mSleepTimeUs = mIdleSleepTimeUs; 4878 } 4879 } else if (mBytesWritten != 0 && audio_has_proportional_frames(mFormat)) { 4880 memset(mSinkBuffer, 0, mFrameCount * mFrameSize); 4881 mSleepTimeUs = 0; 4882 } 4883} 4884 4885void AudioFlinger::DirectOutputThread::threadLoop_exit() 4886{ 4887 { 4888 Mutex::Autolock _l(mLock); 4889 for (size_t i = 0; i < mTracks.size(); i++) { 4890 if (mTracks[i]->isFlushPending()) { 4891 mTracks[i]->flushAck(); 4892 mFlushPending = true; 4893 } 4894 } 4895 if (mFlushPending) { 4896 flushHw_l(); 4897 } 4898 } 4899 PlaybackThread::threadLoop_exit(); 4900} 4901 4902// must be called with thread mutex locked 4903bool AudioFlinger::DirectOutputThread::shouldStandby_l() 4904{ 4905 bool trackPaused = false; 4906 bool trackStopped = false; 4907 4908 if ((mType == DIRECT) && audio_is_linear_pcm(mFormat) && !usesHwAvSync()) { 4909 return !mStandby; 4910 } 4911 4912 // do not put the HAL in standby when paused. AwesomePlayer clear the offloaded AudioTrack 4913 // after a timeout and we will enter standby then. 4914 if (mTracks.size() > 0) { 4915 trackPaused = mTracks[mTracks.size() - 1]->isPaused(); 4916 trackStopped = mTracks[mTracks.size() - 1]->isStopped() || 4917 mTracks[mTracks.size() - 1]->mState == TrackBase::IDLE; 4918 } 4919 4920 return !mStandby && !(trackPaused || (mHwPaused && !trackStopped)); 4921} 4922 4923// getTrackName_l() must be called with ThreadBase::mLock held 4924int AudioFlinger::DirectOutputThread::getTrackName_l(audio_channel_mask_t channelMask __unused, 4925 audio_format_t format __unused, audio_session_t sessionId __unused) 4926{ 4927 return 0; 4928} 4929 4930// deleteTrackName_l() must be called with ThreadBase::mLock held 4931void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name __unused) 4932{ 4933} 4934 4935// checkForNewParameter_l() must be called with ThreadBase::mLock held 4936bool AudioFlinger::DirectOutputThread::checkForNewParameter_l(const String8& keyValuePair, 4937 status_t& status) 4938{ 4939 bool reconfig = false; 4940 bool a2dpDeviceChanged = false; 4941 4942 status = NO_ERROR; 4943 4944 AudioParameter param = AudioParameter(keyValuePair); 4945 int value; 4946 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 4947 // forward device change to effects that have requested to be 4948 // aware of attached audio device. 4949 if (value != AUDIO_DEVICE_NONE) { 4950 a2dpDeviceChanged = 4951 (mOutDevice & AUDIO_DEVICE_OUT_ALL_A2DP) != (value & AUDIO_DEVICE_OUT_ALL_A2DP); 4952 mOutDevice = value; 4953 for (size_t i = 0; i < mEffectChains.size(); i++) { 4954 mEffectChains[i]->setDevice_l(mOutDevice); 4955 } 4956 } 4957 } 4958 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 4959 // do not accept frame count changes if tracks are open as the track buffer 4960 // size depends on frame count and correct behavior would not be garantied 4961 // if frame count is changed after track creation 4962 if (!mTracks.isEmpty()) { 4963 status = INVALID_OPERATION; 4964 } else { 4965 reconfig = true; 4966 } 4967 } 4968 if (status == NO_ERROR) { 4969 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 4970 keyValuePair.string()); 4971 if (!mStandby && status == INVALID_OPERATION) { 4972 mOutput->standby(); 4973 mStandby = true; 4974 mBytesWritten = 0; 4975 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 4976 keyValuePair.string()); 4977 } 4978 if (status == NO_ERROR && reconfig) { 4979 readOutputParameters_l(); 4980 sendIoConfigEvent_l(AUDIO_OUTPUT_CONFIG_CHANGED); 4981 } 4982 } 4983 4984 return reconfig || a2dpDeviceChanged; 4985} 4986 4987uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() const 4988{ 4989 uint32_t time; 4990 if (audio_has_proportional_frames(mFormat)) { 4991 time = PlaybackThread::activeSleepTimeUs(); 4992 } else { 4993 time = kDirectMinSleepTimeUs; 4994 } 4995 return time; 4996} 4997 4998uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() const 4999{ 5000 uint32_t time; 5001 if (audio_has_proportional_frames(mFormat)) { 5002 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 5003 } else { 5004 time = kDirectMinSleepTimeUs; 5005 } 5006 return time; 5007} 5008 5009uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() const 5010{ 5011 uint32_t time; 5012 if (audio_has_proportional_frames(mFormat)) { 5013 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 5014 } else { 5015 time = kDirectMinSleepTimeUs; 5016 } 5017 return time; 5018} 5019 5020void AudioFlinger::DirectOutputThread::cacheParameters_l() 5021{ 5022 PlaybackThread::cacheParameters_l(); 5023 5024 // use shorter standby delay as on normal output to release 5025 // hardware resources as soon as possible 5026 // no delay on outputs with HW A/V sync 5027 if (usesHwAvSync()) { 5028 mStandbyDelayNs = 0; 5029 } else if ((mType == OFFLOAD) && !audio_has_proportional_frames(mFormat)) { 5030 mStandbyDelayNs = kOffloadStandbyDelayNs; 5031 } else { 5032 mStandbyDelayNs = microseconds(mActiveSleepTimeUs*2); 5033 } 5034} 5035 5036void AudioFlinger::DirectOutputThread::flushHw_l() 5037{ 5038 mOutput->flush(); 5039 mHwPaused = false; 5040 mFlushPending = false; 5041} 5042 5043// ---------------------------------------------------------------------------- 5044 5045AudioFlinger::AsyncCallbackThread::AsyncCallbackThread( 5046 const wp<AudioFlinger::PlaybackThread>& playbackThread) 5047 : Thread(false /*canCallJava*/), 5048 mPlaybackThread(playbackThread), 5049 mWriteAckSequence(0), 5050 mDrainSequence(0) 5051{ 5052} 5053 5054AudioFlinger::AsyncCallbackThread::~AsyncCallbackThread() 5055{ 5056} 5057 5058void AudioFlinger::AsyncCallbackThread::onFirstRef() 5059{ 5060 run("Offload Cbk", ANDROID_PRIORITY_URGENT_AUDIO); 5061} 5062 5063bool AudioFlinger::AsyncCallbackThread::threadLoop() 5064{ 5065 while (!exitPending()) { 5066 uint32_t writeAckSequence; 5067 uint32_t drainSequence; 5068 5069 { 5070 Mutex::Autolock _l(mLock); 5071 while (!((mWriteAckSequence & 1) || 5072 (mDrainSequence & 1) || 5073 exitPending())) { 5074 mWaitWorkCV.wait(mLock); 5075 } 5076 5077 if (exitPending()) { 5078 break; 5079 } 5080 ALOGV("AsyncCallbackThread mWriteAckSequence %d mDrainSequence %d", 5081 mWriteAckSequence, mDrainSequence); 5082 writeAckSequence = mWriteAckSequence; 5083 mWriteAckSequence &= ~1; 5084 drainSequence = mDrainSequence; 5085 mDrainSequence &= ~1; 5086 } 5087 { 5088 sp<AudioFlinger::PlaybackThread> playbackThread = mPlaybackThread.promote(); 5089 if (playbackThread != 0) { 5090 if (writeAckSequence & 1) { 5091 playbackThread->resetWriteBlocked(writeAckSequence >> 1); 5092 } 5093 if (drainSequence & 1) { 5094 playbackThread->resetDraining(drainSequence >> 1); 5095 } 5096 } 5097 } 5098 } 5099 return false; 5100} 5101 5102void AudioFlinger::AsyncCallbackThread::exit() 5103{ 5104 ALOGV("AsyncCallbackThread::exit"); 5105 Mutex::Autolock _l(mLock); 5106 requestExit(); 5107 mWaitWorkCV.broadcast(); 5108} 5109 5110void AudioFlinger::AsyncCallbackThread::setWriteBlocked(uint32_t sequence) 5111{ 5112 Mutex::Autolock _l(mLock); 5113 // bit 0 is cleared 5114 mWriteAckSequence = sequence << 1; 5115} 5116 5117void AudioFlinger::AsyncCallbackThread::resetWriteBlocked() 5118{ 5119 Mutex::Autolock _l(mLock); 5120 // ignore unexpected callbacks 5121 if (mWriteAckSequence & 2) { 5122 mWriteAckSequence |= 1; 5123 mWaitWorkCV.signal(); 5124 } 5125} 5126 5127void AudioFlinger::AsyncCallbackThread::setDraining(uint32_t sequence) 5128{ 5129 Mutex::Autolock _l(mLock); 5130 // bit 0 is cleared 5131 mDrainSequence = sequence << 1; 5132} 5133 5134void AudioFlinger::AsyncCallbackThread::resetDraining() 5135{ 5136 Mutex::Autolock _l(mLock); 5137 // ignore unexpected callbacks 5138 if (mDrainSequence & 2) { 5139 mDrainSequence |= 1; 5140 mWaitWorkCV.signal(); 5141 } 5142} 5143 5144 5145// ---------------------------------------------------------------------------- 5146AudioFlinger::OffloadThread::OffloadThread(const sp<AudioFlinger>& audioFlinger, 5147 AudioStreamOut* output, audio_io_handle_t id, uint32_t device, bool systemReady) 5148 : DirectOutputThread(audioFlinger, output, id, device, OFFLOAD, systemReady), 5149 mPausedWriteLength(0), mPausedBytesRemaining(0), mKeepWakeLock(true) 5150{ 5151 //FIXME: mStandby should be set to true by ThreadBase constructor 5152 mStandby = true; 5153 mKeepWakeLock = property_get_bool("ro.audio.offload_wakelock", true /* default_value */); 5154} 5155 5156void AudioFlinger::OffloadThread::threadLoop_exit() 5157{ 5158 if (mFlushPending || mHwPaused) { 5159 // If a flush is pending or track was paused, just discard buffered data 5160 flushHw_l(); 5161 } else { 5162 mMixerStatus = MIXER_DRAIN_ALL; 5163 threadLoop_drain(); 5164 } 5165 if (mUseAsyncWrite) { 5166 ALOG_ASSERT(mCallbackThread != 0); 5167 mCallbackThread->exit(); 5168 } 5169 PlaybackThread::threadLoop_exit(); 5170} 5171 5172AudioFlinger::PlaybackThread::mixer_state AudioFlinger::OffloadThread::prepareTracks_l( 5173 Vector< sp<Track> > *tracksToRemove 5174) 5175{ 5176 size_t count = mActiveTracks.size(); 5177 5178 mixer_state mixerStatus = MIXER_IDLE; 5179 bool doHwPause = false; 5180 bool doHwResume = false; 5181 5182 ALOGV("OffloadThread::prepareTracks_l active tracks %zu", count); 5183 5184 // find out which tracks need to be processed 5185 for (size_t i = 0; i < count; i++) { 5186 sp<Track> t = mActiveTracks[i].promote(); 5187 // The track died recently 5188 if (t == 0) { 5189 continue; 5190 } 5191 Track* const track = t.get(); 5192#ifdef VERY_VERY_VERBOSE_LOGGING 5193 audio_track_cblk_t* cblk = track->cblk(); 5194#endif 5195 // Only consider last track started for volume and mixer state control. 5196 // In theory an older track could underrun and restart after the new one starts 5197 // but as we only care about the transition phase between two tracks on a 5198 // direct output, it is not a problem to ignore the underrun case. 5199 sp<Track> l = mLatestActiveTrack.promote(); 5200 bool last = l.get() == track; 5201 5202 if (track->isInvalid()) { 5203 ALOGW("An invalidated track shouldn't be in active list"); 5204 tracksToRemove->add(track); 5205 continue; 5206 } 5207 5208 if (track->mState == TrackBase::IDLE) { 5209 ALOGW("An idle track shouldn't be in active list"); 5210 continue; 5211 } 5212 5213 if (track->isPausing()) { 5214 track->setPaused(); 5215 if (last) { 5216 if (mHwSupportsPause && !mHwPaused) { 5217 doHwPause = true; 5218 mHwPaused = true; 5219 } 5220 // If we were part way through writing the mixbuffer to 5221 // the HAL we must save this until we resume 5222 // BUG - this will be wrong if a different track is made active, 5223 // in that case we want to discard the pending data in the 5224 // mixbuffer and tell the client to present it again when the 5225 // track is resumed 5226 mPausedWriteLength = mCurrentWriteLength; 5227 mPausedBytesRemaining = mBytesRemaining; 5228 mBytesRemaining = 0; // stop writing 5229 } 5230 tracksToRemove->add(track); 5231 } else if (track->isFlushPending()) { 5232 if (track->isStopping_1()) { 5233 track->mRetryCount = kMaxTrackStopRetriesOffload; 5234 } else { 5235 track->mRetryCount = kMaxTrackRetriesOffload; 5236 } 5237 track->flushAck(); 5238 if (last) { 5239 mFlushPending = true; 5240 } 5241 } else if (track->isResumePending()){ 5242 track->resumeAck(); 5243 if (last) { 5244 if (mPausedBytesRemaining) { 5245 // Need to continue write that was interrupted 5246 mCurrentWriteLength = mPausedWriteLength; 5247 mBytesRemaining = mPausedBytesRemaining; 5248 mPausedBytesRemaining = 0; 5249 } 5250 if (mHwPaused) { 5251 doHwResume = true; 5252 mHwPaused = false; 5253 // threadLoop_mix() will handle the case that we need to 5254 // resume an interrupted write 5255 } 5256 // enable write to audio HAL 5257 mSleepTimeUs = 0; 5258 5259 // Do not handle new data in this iteration even if track->framesReady() 5260 mixerStatus = MIXER_TRACKS_ENABLED; 5261 } 5262 } else if (track->framesReady() && track->isReady() && 5263 !track->isPaused() && !track->isTerminated() && !track->isStopping_2()) { 5264 ALOGVV("OffloadThread: track %d s=%08x [OK]", track->name(), cblk->mServer); 5265 if (track->mFillingUpStatus == Track::FS_FILLED) { 5266 track->mFillingUpStatus = Track::FS_ACTIVE; 5267 // make sure processVolume_l() will apply new volume even if 0 5268 mLeftVolFloat = mRightVolFloat = -1.0; 5269 } 5270 5271 if (last) { 5272 sp<Track> previousTrack = mPreviousTrack.promote(); 5273 if (previousTrack != 0) { 5274 if (track != previousTrack.get()) { 5275 // Flush any data still being written from last track 5276 mBytesRemaining = 0; 5277 if (mPausedBytesRemaining) { 5278 // Last track was paused so we also need to flush saved 5279 // mixbuffer state and invalidate track so that it will 5280 // re-submit that unwritten data when it is next resumed 5281 mPausedBytesRemaining = 0; 5282 // Invalidate is a bit drastic - would be more efficient 5283 // to have a flag to tell client that some of the 5284 // previously written data was lost 5285 previousTrack->invalidate(); 5286 } 5287 // flush data already sent to the DSP if changing audio session as audio 5288 // comes from a different source. Also invalidate previous track to force a 5289 // seek when resuming. 5290 if (previousTrack->sessionId() != track->sessionId()) { 5291 previousTrack->invalidate(); 5292 } 5293 } 5294 } 5295 mPreviousTrack = track; 5296 // reset retry count 5297 if (track->isStopping_1()) { 5298 track->mRetryCount = kMaxTrackStopRetriesOffload; 5299 } else { 5300 track->mRetryCount = kMaxTrackRetriesOffload; 5301 } 5302 mActiveTrack = t; 5303 mixerStatus = MIXER_TRACKS_READY; 5304 } 5305 } else { 5306 ALOGVV("OffloadThread: track %d s=%08x [NOT READY]", track->name(), cblk->mServer); 5307 if (track->isStopping_1()) { 5308 if (--(track->mRetryCount) <= 0) { 5309 // Hardware buffer can hold a large amount of audio so we must 5310 // wait for all current track's data to drain before we say 5311 // that the track is stopped. 5312 if (mBytesRemaining == 0) { 5313 // Only start draining when all data in mixbuffer 5314 // has been written 5315 ALOGV("OffloadThread: underrun and STOPPING_1 -> draining, STOPPING_2"); 5316 track->mState = TrackBase::STOPPING_2; // so presentation completes after 5317 // drain do not drain if no data was ever sent to HAL (mStandby == true) 5318 if (last && !mStandby) { 5319 // do not modify drain sequence if we are already draining. This happens 5320 // when resuming from pause after drain. 5321 if ((mDrainSequence & 1) == 0) { 5322 mSleepTimeUs = 0; 5323 mStandbyTimeNs = systemTime() + mStandbyDelayNs; 5324 mixerStatus = MIXER_DRAIN_TRACK; 5325 mDrainSequence += 2; 5326 } 5327 if (mHwPaused) { 5328 // It is possible to move from PAUSED to STOPPING_1 without 5329 // a resume so we must ensure hardware is running 5330 doHwResume = true; 5331 mHwPaused = false; 5332 } 5333 } 5334 } 5335 } else if (last) { 5336 ALOGV("stopping1 underrun retries left %d", track->mRetryCount); 5337 mixerStatus = MIXER_TRACKS_ENABLED; 5338 } 5339 } else if (track->isStopping_2()) { 5340 // Drain has completed or we are in standby, signal presentation complete 5341 if (!(mDrainSequence & 1) || !last || mStandby) { 5342 track->mState = TrackBase::STOPPED; 5343 size_t audioHALFrames = 5344 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 5345 int64_t framesWritten = 5346 mBytesWritten / mOutput->getFrameSize(); 5347 track->presentationComplete(framesWritten, audioHALFrames); 5348 track->reset(); 5349 tracksToRemove->add(track); 5350 } 5351 } else { 5352 // No buffers for this track. Give it a few chances to 5353 // fill a buffer, then remove it from active list. 5354 if (--(track->mRetryCount) <= 0) { 5355 ALOGV("OffloadThread: BUFFER TIMEOUT: remove(%d) from active list", 5356 track->name()); 5357 tracksToRemove->add(track); 5358 // indicate to client process that the track was disabled because of underrun; 5359 // it will then automatically call start() when data is available 5360 track->disable(); 5361 } else if (last){ 5362 mixerStatus = MIXER_TRACKS_ENABLED; 5363 } 5364 } 5365 } 5366 // compute volume for this track 5367 processVolume_l(track, last); 5368 } 5369 5370 // make sure the pause/flush/resume sequence is executed in the right order. 5371 // If a flush is pending and a track is active but the HW is not paused, force a HW pause 5372 // before flush and then resume HW. This can happen in case of pause/flush/resume 5373 // if resume is received before pause is executed. 5374 if (!mStandby && (doHwPause || (mFlushPending && !mHwPaused && (count != 0)))) { 5375 mOutput->stream->pause(mOutput->stream); 5376 } 5377 if (mFlushPending) { 5378 flushHw_l(); 5379 } 5380 if (!mStandby && doHwResume) { 5381 mOutput->stream->resume(mOutput->stream); 5382 } 5383 5384 // remove all the tracks that need to be... 5385 removeTracks_l(*tracksToRemove); 5386 5387 return mixerStatus; 5388} 5389 5390// must be called with thread mutex locked 5391bool AudioFlinger::OffloadThread::waitingAsyncCallback_l() 5392{ 5393 ALOGVV("waitingAsyncCallback_l mWriteAckSequence %d mDrainSequence %d", 5394 mWriteAckSequence, mDrainSequence); 5395 if (mUseAsyncWrite && ((mWriteAckSequence & 1) || (mDrainSequence & 1))) { 5396 return true; 5397 } 5398 return false; 5399} 5400 5401bool AudioFlinger::OffloadThread::waitingAsyncCallback() 5402{ 5403 Mutex::Autolock _l(mLock); 5404 return waitingAsyncCallback_l(); 5405} 5406 5407void AudioFlinger::OffloadThread::flushHw_l() 5408{ 5409 DirectOutputThread::flushHw_l(); 5410 // Flush anything still waiting in the mixbuffer 5411 mCurrentWriteLength = 0; 5412 mBytesRemaining = 0; 5413 mPausedWriteLength = 0; 5414 mPausedBytesRemaining = 0; 5415 // reset bytes written count to reflect that DSP buffers are empty after flush. 5416 mBytesWritten = 0; 5417 5418 if (mUseAsyncWrite) { 5419 // discard any pending drain or write ack by incrementing sequence 5420 mWriteAckSequence = (mWriteAckSequence + 2) & ~1; 5421 mDrainSequence = (mDrainSequence + 2) & ~1; 5422 ALOG_ASSERT(mCallbackThread != 0); 5423 mCallbackThread->setWriteBlocked(mWriteAckSequence); 5424 mCallbackThread->setDraining(mDrainSequence); 5425 } 5426} 5427 5428void AudioFlinger::OffloadThread::invalidateTracks(audio_stream_type_t streamType) 5429{ 5430 Mutex::Autolock _l(mLock); 5431 if (PlaybackThread::invalidateTracks_l(streamType)) { 5432 mFlushPending = true; 5433 } 5434} 5435 5436// ---------------------------------------------------------------------------- 5437 5438AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 5439 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id, bool systemReady) 5440 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->outDevice(), 5441 systemReady, DUPLICATING), 5442 mWaitTimeMs(UINT_MAX) 5443{ 5444 addOutputTrack(mainThread); 5445} 5446 5447AudioFlinger::DuplicatingThread::~DuplicatingThread() 5448{ 5449 for (size_t i = 0; i < mOutputTracks.size(); i++) { 5450 mOutputTracks[i]->destroy(); 5451 } 5452} 5453 5454void AudioFlinger::DuplicatingThread::threadLoop_mix() 5455{ 5456 // mix buffers... 5457 if (outputsReady(outputTracks)) { 5458 mAudioMixer->process(); 5459 } else { 5460 if (mMixerBufferValid) { 5461 memset(mMixerBuffer, 0, mMixerBufferSize); 5462 } else { 5463 memset(mSinkBuffer, 0, mSinkBufferSize); 5464 } 5465 } 5466 mSleepTimeUs = 0; 5467 writeFrames = mNormalFrameCount; 5468 mCurrentWriteLength = mSinkBufferSize; 5469 mStandbyTimeNs = systemTime() + mStandbyDelayNs; 5470} 5471 5472void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() 5473{ 5474 if (mSleepTimeUs == 0) { 5475 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 5476 mSleepTimeUs = mActiveSleepTimeUs; 5477 } else { 5478 mSleepTimeUs = mIdleSleepTimeUs; 5479 } 5480 } else if (mBytesWritten != 0) { 5481 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 5482 writeFrames = mNormalFrameCount; 5483 memset(mSinkBuffer, 0, mSinkBufferSize); 5484 } else { 5485 // flush remaining overflow buffers in output tracks 5486 writeFrames = 0; 5487 } 5488 mSleepTimeUs = 0; 5489 } 5490} 5491 5492ssize_t AudioFlinger::DuplicatingThread::threadLoop_write() 5493{ 5494 for (size_t i = 0; i < outputTracks.size(); i++) { 5495 outputTracks[i]->write(mSinkBuffer, writeFrames); 5496 } 5497 mStandby = false; 5498 return (ssize_t)mSinkBufferSize; 5499} 5500 5501void AudioFlinger::DuplicatingThread::threadLoop_standby() 5502{ 5503 // DuplicatingThread implements standby by stopping all tracks 5504 for (size_t i = 0; i < outputTracks.size(); i++) { 5505 outputTracks[i]->stop(); 5506 } 5507} 5508 5509void AudioFlinger::DuplicatingThread::saveOutputTracks() 5510{ 5511 outputTracks = mOutputTracks; 5512} 5513 5514void AudioFlinger::DuplicatingThread::clearOutputTracks() 5515{ 5516 outputTracks.clear(); 5517} 5518 5519void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 5520{ 5521 Mutex::Autolock _l(mLock); 5522 // The downstream MixerThread consumes thread->frameCount() amount of frames per mix pass. 5523 // Adjust for thread->sampleRate() to determine minimum buffer frame count. 5524 // Then triple buffer because Threads do not run synchronously and may not be clock locked. 5525 const size_t frameCount = 5526 3 * sourceFramesNeeded(mSampleRate, thread->frameCount(), thread->sampleRate()); 5527 // TODO: Consider asynchronous sample rate conversion to handle clock disparity 5528 // from different OutputTracks and their associated MixerThreads (e.g. one may 5529 // nearly empty and the other may be dropping data). 5530 5531 sp<OutputTrack> outputTrack = new OutputTrack(thread, 5532 this, 5533 mSampleRate, 5534 mFormat, 5535 mChannelMask, 5536 frameCount, 5537 IPCThreadState::self()->getCallingUid()); 5538 if (outputTrack->cblk() != NULL) { 5539 thread->setStreamVolume(AUDIO_STREAM_PATCH, 1.0f); 5540 mOutputTracks.add(outputTrack); 5541 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack.get(), thread); 5542 updateWaitTime_l(); 5543 } 5544} 5545 5546void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 5547{ 5548 Mutex::Autolock _l(mLock); 5549 for (size_t i = 0; i < mOutputTracks.size(); i++) { 5550 if (mOutputTracks[i]->thread() == thread) { 5551 mOutputTracks[i]->destroy(); 5552 mOutputTracks.removeAt(i); 5553 updateWaitTime_l(); 5554 if (thread->getOutput() == mOutput) { 5555 mOutput = NULL; 5556 } 5557 return; 5558 } 5559 } 5560 ALOGV("removeOutputTrack(): unknown thread: %p", thread); 5561} 5562 5563// caller must hold mLock 5564void AudioFlinger::DuplicatingThread::updateWaitTime_l() 5565{ 5566 mWaitTimeMs = UINT_MAX; 5567 for (size_t i = 0; i < mOutputTracks.size(); i++) { 5568 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 5569 if (strong != 0) { 5570 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 5571 if (waitTimeMs < mWaitTimeMs) { 5572 mWaitTimeMs = waitTimeMs; 5573 } 5574 } 5575 } 5576} 5577 5578 5579bool AudioFlinger::DuplicatingThread::outputsReady( 5580 const SortedVector< sp<OutputTrack> > &outputTracks) 5581{ 5582 for (size_t i = 0; i < outputTracks.size(); i++) { 5583 sp<ThreadBase> thread = outputTracks[i]->thread().promote(); 5584 if (thread == 0) { 5585 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", 5586 outputTracks[i].get()); 5587 return false; 5588 } 5589 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 5590 // see note at standby() declaration 5591 if (playbackThread->standby() && !playbackThread->isSuspended()) { 5592 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), 5593 thread.get()); 5594 return false; 5595 } 5596 } 5597 return true; 5598} 5599 5600uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() const 5601{ 5602 return (mWaitTimeMs * 1000) / 2; 5603} 5604 5605void AudioFlinger::DuplicatingThread::cacheParameters_l() 5606{ 5607 // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first 5608 updateWaitTime_l(); 5609 5610 MixerThread::cacheParameters_l(); 5611} 5612 5613// ---------------------------------------------------------------------------- 5614// Record 5615// ---------------------------------------------------------------------------- 5616 5617AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 5618 AudioStreamIn *input, 5619 audio_io_handle_t id, 5620 audio_devices_t outDevice, 5621 audio_devices_t inDevice, 5622 bool systemReady 5623#ifdef TEE_SINK 5624 , const sp<NBAIO_Sink>& teeSink 5625#endif 5626 ) : 5627 ThreadBase(audioFlinger, id, outDevice, inDevice, RECORD, systemReady), 5628 mInput(input), mActiveTracksGen(0), mRsmpInBuffer(NULL), 5629 // mRsmpInFrames and mRsmpInFramesP2 are set by readInputParameters_l() 5630 mRsmpInRear(0) 5631#ifdef TEE_SINK 5632 , mTeeSink(teeSink) 5633#endif 5634 , mReadOnlyHeap(new MemoryDealer(kRecordThreadReadOnlyHeapSize, 5635 "RecordThreadRO", MemoryHeapBase::READ_ONLY)) 5636 // mFastCapture below 5637 , mFastCaptureFutex(0) 5638 // mInputSource 5639 // mPipeSink 5640 // mPipeSource 5641 , mPipeFramesP2(0) 5642 // mPipeMemory 5643 // mFastCaptureNBLogWriter 5644 , mFastTrackAvail(false) 5645{ 5646 snprintf(mThreadName, kThreadNameLength, "AudioIn_%X", id); 5647 mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mThreadName); 5648 5649 readInputParameters_l(); 5650 5651 // create an NBAIO source for the HAL input stream, and negotiate 5652 mInputSource = new AudioStreamInSource(input->stream); 5653 size_t numCounterOffers = 0; 5654 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount, mFormat)}; 5655#if !LOG_NDEBUG 5656 ssize_t index = 5657#else 5658 (void) 5659#endif 5660 mInputSource->negotiate(offers, 1, NULL, numCounterOffers); 5661 ALOG_ASSERT(index == 0); 5662 5663 // initialize fast capture depending on configuration 5664 bool initFastCapture; 5665 switch (kUseFastCapture) { 5666 case FastCapture_Never: 5667 initFastCapture = false; 5668 break; 5669 case FastCapture_Always: 5670 initFastCapture = true; 5671 break; 5672 case FastCapture_Static: 5673 initFastCapture = (mFrameCount * 1000) / mSampleRate < kMinNormalCaptureBufferSizeMs; 5674 break; 5675 // case FastCapture_Dynamic: 5676 } 5677 5678 if (initFastCapture) { 5679 // create a Pipe for FastCapture to write to, and for us and fast tracks to read from 5680 NBAIO_Format format = mInputSource->format(); 5681 size_t pipeFramesP2 = roundup(mSampleRate / 25); // double-buffering of 20 ms each 5682 size_t pipeSize = pipeFramesP2 * Format_frameSize(format); 5683 void *pipeBuffer; 5684 const sp<MemoryDealer> roHeap(readOnlyHeap()); 5685 sp<IMemory> pipeMemory; 5686 if ((roHeap == 0) || 5687 (pipeMemory = roHeap->allocate(pipeSize)) == 0 || 5688 (pipeBuffer = pipeMemory->pointer()) == NULL) { 5689 ALOGE("not enough memory for pipe buffer size=%zu", pipeSize); 5690 goto failed; 5691 } 5692 // pipe will be shared directly with fast clients, so clear to avoid leaking old information 5693 memset(pipeBuffer, 0, pipeSize); 5694 Pipe *pipe = new Pipe(pipeFramesP2, format, pipeBuffer); 5695 const NBAIO_Format offers[1] = {format}; 5696 size_t numCounterOffers = 0; 5697 ssize_t index = pipe->negotiate(offers, 1, NULL, numCounterOffers); 5698 ALOG_ASSERT(index == 0); 5699 mPipeSink = pipe; 5700 PipeReader *pipeReader = new PipeReader(*pipe); 5701 numCounterOffers = 0; 5702 index = pipeReader->negotiate(offers, 1, NULL, numCounterOffers); 5703 ALOG_ASSERT(index == 0); 5704 mPipeSource = pipeReader; 5705 mPipeFramesP2 = pipeFramesP2; 5706 mPipeMemory = pipeMemory; 5707 5708 // create fast capture 5709 mFastCapture = new FastCapture(); 5710 FastCaptureStateQueue *sq = mFastCapture->sq(); 5711#ifdef STATE_QUEUE_DUMP 5712 // FIXME 5713#endif 5714 FastCaptureState *state = sq->begin(); 5715 state->mCblk = NULL; 5716 state->mInputSource = mInputSource.get(); 5717 state->mInputSourceGen++; 5718 state->mPipeSink = pipe; 5719 state->mPipeSinkGen++; 5720 state->mFrameCount = mFrameCount; 5721 state->mCommand = FastCaptureState::COLD_IDLE; 5722 // already done in constructor initialization list 5723 //mFastCaptureFutex = 0; 5724 state->mColdFutexAddr = &mFastCaptureFutex; 5725 state->mColdGen++; 5726 state->mDumpState = &mFastCaptureDumpState; 5727#ifdef TEE_SINK 5728 // FIXME 5729#endif 5730 mFastCaptureNBLogWriter = audioFlinger->newWriter_l(kFastCaptureLogSize, "FastCapture"); 5731 state->mNBLogWriter = mFastCaptureNBLogWriter.get(); 5732 sq->end(); 5733 sq->push(FastCaptureStateQueue::BLOCK_UNTIL_PUSHED); 5734 5735 // start the fast capture 5736 mFastCapture->run("FastCapture", ANDROID_PRIORITY_URGENT_AUDIO); 5737 pid_t tid = mFastCapture->getTid(); 5738 sendPrioConfigEvent(getpid_cached, tid, kPriorityFastCapture); 5739#ifdef AUDIO_WATCHDOG 5740 // FIXME 5741#endif 5742 5743 mFastTrackAvail = true; 5744 } 5745failed: ; 5746 5747 // FIXME mNormalSource 5748} 5749 5750AudioFlinger::RecordThread::~RecordThread() 5751{ 5752 if (mFastCapture != 0) { 5753 FastCaptureStateQueue *sq = mFastCapture->sq(); 5754 FastCaptureState *state = sq->begin(); 5755 if (state->mCommand == FastCaptureState::COLD_IDLE) { 5756 int32_t old = android_atomic_inc(&mFastCaptureFutex); 5757 if (old == -1) { 5758 (void) syscall(__NR_futex, &mFastCaptureFutex, FUTEX_WAKE_PRIVATE, 1); 5759 } 5760 } 5761 state->mCommand = FastCaptureState::EXIT; 5762 sq->end(); 5763 sq->push(FastCaptureStateQueue::BLOCK_UNTIL_PUSHED); 5764 mFastCapture->join(); 5765 mFastCapture.clear(); 5766 } 5767 mAudioFlinger->unregisterWriter(mFastCaptureNBLogWriter); 5768 mAudioFlinger->unregisterWriter(mNBLogWriter); 5769 free(mRsmpInBuffer); 5770} 5771 5772void AudioFlinger::RecordThread::onFirstRef() 5773{ 5774 run(mThreadName, PRIORITY_URGENT_AUDIO); 5775} 5776 5777bool AudioFlinger::RecordThread::threadLoop() 5778{ 5779 nsecs_t lastWarning = 0; 5780 5781 inputStandBy(); 5782 5783reacquire_wakelock: 5784 sp<RecordTrack> activeTrack; 5785 int activeTracksGen; 5786 { 5787 Mutex::Autolock _l(mLock); 5788 size_t size = mActiveTracks.size(); 5789 activeTracksGen = mActiveTracksGen; 5790 if (size > 0) { 5791 // FIXME an arbitrary choice 5792 activeTrack = mActiveTracks[0]; 5793 acquireWakeLock_l(activeTrack->uid()); 5794 if (size > 1) { 5795 SortedVector<int> tmp; 5796 for (size_t i = 0; i < size; i++) { 5797 tmp.add(mActiveTracks[i]->uid()); 5798 } 5799 updateWakeLockUids_l(tmp); 5800 } 5801 } else { 5802 acquireWakeLock_l(-1); 5803 } 5804 } 5805 5806 // used to request a deferred sleep, to be executed later while mutex is unlocked 5807 uint32_t sleepUs = 0; 5808 5809 // loop while there is work to do 5810 for (;;) { 5811 Vector< sp<EffectChain> > effectChains; 5812 5813 // sleep with mutex unlocked 5814 if (sleepUs > 0) { 5815 ATRACE_BEGIN("sleep"); 5816 usleep(sleepUs); 5817 ATRACE_END(); 5818 sleepUs = 0; 5819 } 5820 5821 // activeTracks accumulates a copy of a subset of mActiveTracks 5822 Vector< sp<RecordTrack> > activeTracks; 5823 5824 // reference to the (first and only) active fast track 5825 sp<RecordTrack> fastTrack; 5826 5827 // reference to a fast track which is about to be removed 5828 sp<RecordTrack> fastTrackToRemove; 5829 5830 { // scope for mLock 5831 Mutex::Autolock _l(mLock); 5832 5833 processConfigEvents_l(); 5834 5835 // check exitPending here because checkForNewParameters_l() and 5836 // checkForNewParameters_l() can temporarily release mLock 5837 if (exitPending()) { 5838 break; 5839 } 5840 5841 // if no active track(s), then standby and release wakelock 5842 size_t size = mActiveTracks.size(); 5843 if (size == 0) { 5844 standbyIfNotAlreadyInStandby(); 5845 // exitPending() can't become true here 5846 releaseWakeLock_l(); 5847 ALOGV("RecordThread: loop stopping"); 5848 // go to sleep 5849 mWaitWorkCV.wait(mLock); 5850 ALOGV("RecordThread: loop starting"); 5851 goto reacquire_wakelock; 5852 } 5853 5854 if (mActiveTracksGen != activeTracksGen) { 5855 activeTracksGen = mActiveTracksGen; 5856 SortedVector<int> tmp; 5857 for (size_t i = 0; i < size; i++) { 5858 tmp.add(mActiveTracks[i]->uid()); 5859 } 5860 updateWakeLockUids_l(tmp); 5861 } 5862 5863 bool doBroadcast = false; 5864 for (size_t i = 0; i < size; ) { 5865 5866 activeTrack = mActiveTracks[i]; 5867 if (activeTrack->isTerminated()) { 5868 if (activeTrack->isFastTrack()) { 5869 ALOG_ASSERT(fastTrackToRemove == 0); 5870 fastTrackToRemove = activeTrack; 5871 } 5872 removeTrack_l(activeTrack); 5873 mActiveTracks.remove(activeTrack); 5874 mActiveTracksGen++; 5875 size--; 5876 continue; 5877 } 5878 5879 TrackBase::track_state activeTrackState = activeTrack->mState; 5880 switch (activeTrackState) { 5881 5882 case TrackBase::PAUSING: 5883 mActiveTracks.remove(activeTrack); 5884 mActiveTracksGen++; 5885 doBroadcast = true; 5886 size--; 5887 continue; 5888 5889 case TrackBase::STARTING_1: 5890 sleepUs = 10000; 5891 i++; 5892 continue; 5893 5894 case TrackBase::STARTING_2: 5895 doBroadcast = true; 5896 mStandby = false; 5897 activeTrack->mState = TrackBase::ACTIVE; 5898 break; 5899 5900 case TrackBase::ACTIVE: 5901 break; 5902 5903 case TrackBase::IDLE: 5904 i++; 5905 continue; 5906 5907 default: 5908 LOG_ALWAYS_FATAL("Unexpected activeTrackState %d", activeTrackState); 5909 } 5910 5911 activeTracks.add(activeTrack); 5912 i++; 5913 5914 if (activeTrack->isFastTrack()) { 5915 ALOG_ASSERT(!mFastTrackAvail); 5916 ALOG_ASSERT(fastTrack == 0); 5917 fastTrack = activeTrack; 5918 } 5919 } 5920 if (doBroadcast) { 5921 mStartStopCond.broadcast(); 5922 } 5923 5924 // sleep if there are no active tracks to process 5925 if (activeTracks.size() == 0) { 5926 if (sleepUs == 0) { 5927 sleepUs = kRecordThreadSleepUs; 5928 } 5929 continue; 5930 } 5931 sleepUs = 0; 5932 5933 lockEffectChains_l(effectChains); 5934 } 5935 5936 // thread mutex is now unlocked, mActiveTracks unknown, activeTracks.size() > 0 5937 5938 size_t size = effectChains.size(); 5939 for (size_t i = 0; i < size; i++) { 5940 // thread mutex is not locked, but effect chain is locked 5941 effectChains[i]->process_l(); 5942 } 5943 5944 // Push a new fast capture state if fast capture is not already running, or cblk change 5945 if (mFastCapture != 0) { 5946 FastCaptureStateQueue *sq = mFastCapture->sq(); 5947 FastCaptureState *state = sq->begin(); 5948 bool didModify = false; 5949 FastCaptureStateQueue::block_t block = FastCaptureStateQueue::BLOCK_UNTIL_PUSHED; 5950 if (state->mCommand != FastCaptureState::READ_WRITE /* FIXME && 5951 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)*/) { 5952 if (state->mCommand == FastCaptureState::COLD_IDLE) { 5953 int32_t old = android_atomic_inc(&mFastCaptureFutex); 5954 if (old == -1) { 5955 (void) syscall(__NR_futex, &mFastCaptureFutex, FUTEX_WAKE_PRIVATE, 1); 5956 } 5957 } 5958 state->mCommand = FastCaptureState::READ_WRITE; 5959#if 0 // FIXME 5960 mFastCaptureDumpState.increaseSamplingN(mAudioFlinger->isLowRamDevice() ? 5961 FastThreadDumpState::kSamplingNforLowRamDevice : 5962 FastThreadDumpState::kSamplingN); 5963#endif 5964 didModify = true; 5965 } 5966 audio_track_cblk_t *cblkOld = state->mCblk; 5967 audio_track_cblk_t *cblkNew = fastTrack != 0 ? fastTrack->cblk() : NULL; 5968 if (cblkNew != cblkOld) { 5969 state->mCblk = cblkNew; 5970 // block until acked if removing a fast track 5971 if (cblkOld != NULL) { 5972 block = FastCaptureStateQueue::BLOCK_UNTIL_ACKED; 5973 } 5974 didModify = true; 5975 } 5976 sq->end(didModify); 5977 if (didModify) { 5978 sq->push(block); 5979#if 0 5980 if (kUseFastCapture == FastCapture_Dynamic) { 5981 mNormalSource = mPipeSource; 5982 } 5983#endif 5984 } 5985 } 5986 5987 // now run the fast track destructor with thread mutex unlocked 5988 fastTrackToRemove.clear(); 5989 5990 // Read from HAL to keep up with fastest client if multiple active tracks, not slowest one. 5991 // Only the client(s) that are too slow will overrun. But if even the fastest client is too 5992 // slow, then this RecordThread will overrun by not calling HAL read often enough. 5993 // If destination is non-contiguous, first read past the nominal end of buffer, then 5994 // copy to the right place. Permitted because mRsmpInBuffer was over-allocated. 5995 5996 int32_t rear = mRsmpInRear & (mRsmpInFramesP2 - 1); 5997 ssize_t framesRead; 5998 5999 // If an NBAIO source is present, use it to read the normal capture's data 6000 if (mPipeSource != 0) { 6001 size_t framesToRead = mBufferSize / mFrameSize; 6002 framesRead = mPipeSource->read((uint8_t*)mRsmpInBuffer + rear * mFrameSize, 6003 framesToRead); 6004 if (framesRead == 0) { 6005 // since pipe is non-blocking, simulate blocking input 6006 sleepUs = (framesToRead * 1000000LL) / mSampleRate; 6007 } 6008 // otherwise use the HAL / AudioStreamIn directly 6009 } else { 6010 ATRACE_BEGIN("read"); 6011 ssize_t bytesRead = mInput->stream->read(mInput->stream, 6012 (uint8_t*)mRsmpInBuffer + rear * mFrameSize, mBufferSize); 6013 ATRACE_END(); 6014 if (bytesRead < 0) { 6015 framesRead = bytesRead; 6016 } else { 6017 framesRead = bytesRead / mFrameSize; 6018 } 6019 } 6020 6021 // Update server timestamp with server stats 6022 // systemTime() is optional if the hardware supports timestamps. 6023 mTimestamp.mPosition[ExtendedTimestamp::LOCATION_SERVER] += framesRead; 6024 mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_SERVER] = systemTime(); 6025 6026 // Update server timestamp with kernel stats 6027 if (mInput->stream->get_capture_position != nullptr) { 6028 int64_t position, time; 6029 int ret = mInput->stream->get_capture_position(mInput->stream, &position, &time); 6030 if (ret == NO_ERROR) { 6031 mTimestamp.mPosition[ExtendedTimestamp::LOCATION_KERNEL] = position; 6032 mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL] = time; 6033 // Note: In general record buffers should tend to be empty in 6034 // a properly running pipeline. 6035 // 6036 // Also, it is not advantageous to call get_presentation_position during the read 6037 // as the read obtains a lock, preventing the timestamp call from executing. 6038 } 6039 } 6040 // Use this to track timestamp information 6041 // ALOGD("%s", mTimestamp.toString().c_str()); 6042 6043 if (framesRead < 0 || (framesRead == 0 && mPipeSource == 0)) { 6044 ALOGE("read failed: framesRead=%zd", framesRead); 6045 // Force input into standby so that it tries to recover at next read attempt 6046 inputStandBy(); 6047 sleepUs = kRecordThreadSleepUs; 6048 } 6049 if (framesRead <= 0) { 6050 goto unlock; 6051 } 6052 ALOG_ASSERT(framesRead > 0); 6053 6054 if (mTeeSink != 0) { 6055 (void) mTeeSink->write((uint8_t*)mRsmpInBuffer + rear * mFrameSize, framesRead); 6056 } 6057 // If destination is non-contiguous, we now correct for reading past end of buffer. 6058 { 6059 size_t part1 = mRsmpInFramesP2 - rear; 6060 if ((size_t) framesRead > part1) { 6061 memcpy(mRsmpInBuffer, (uint8_t*)mRsmpInBuffer + mRsmpInFramesP2 * mFrameSize, 6062 (framesRead - part1) * mFrameSize); 6063 } 6064 } 6065 rear = mRsmpInRear += framesRead; 6066 6067 size = activeTracks.size(); 6068 // loop over each active track 6069 for (size_t i = 0; i < size; i++) { 6070 activeTrack = activeTracks[i]; 6071 6072 // skip fast tracks, as those are handled directly by FastCapture 6073 if (activeTrack->isFastTrack()) { 6074 continue; 6075 } 6076 6077 // TODO: This code probably should be moved to RecordTrack. 6078 // TODO: Update the activeTrack buffer converter in case of reconfigure. 6079 6080 enum { 6081 OVERRUN_UNKNOWN, 6082 OVERRUN_TRUE, 6083 OVERRUN_FALSE 6084 } overrun = OVERRUN_UNKNOWN; 6085 6086 // loop over getNextBuffer to handle circular sink 6087 for (;;) { 6088 6089 activeTrack->mSink.frameCount = ~0; 6090 status_t status = activeTrack->getNextBuffer(&activeTrack->mSink); 6091 size_t framesOut = activeTrack->mSink.frameCount; 6092 LOG_ALWAYS_FATAL_IF((status == OK) != (framesOut > 0)); 6093 6094 // check available frames and handle overrun conditions 6095 // if the record track isn't draining fast enough. 6096 bool hasOverrun; 6097 size_t framesIn; 6098 activeTrack->mResamplerBufferProvider->sync(&framesIn, &hasOverrun); 6099 if (hasOverrun) { 6100 overrun = OVERRUN_TRUE; 6101 } 6102 if (framesOut == 0 || framesIn == 0) { 6103 break; 6104 } 6105 6106 // Don't allow framesOut to be larger than what is possible with resampling 6107 // from framesIn. 6108 // This isn't strictly necessary but helps limit buffer resizing in 6109 // RecordBufferConverter. TODO: remove when no longer needed. 6110 framesOut = min(framesOut, 6111 destinationFramesPossible( 6112 framesIn, mSampleRate, activeTrack->mSampleRate)); 6113 // process frames from the RecordThread buffer provider to the RecordTrack buffer 6114 framesOut = activeTrack->mRecordBufferConverter->convert( 6115 activeTrack->mSink.raw, activeTrack->mResamplerBufferProvider, framesOut); 6116 6117 if (framesOut > 0 && (overrun == OVERRUN_UNKNOWN)) { 6118 overrun = OVERRUN_FALSE; 6119 } 6120 6121 if (activeTrack->mFramesToDrop == 0) { 6122 if (framesOut > 0) { 6123 activeTrack->mSink.frameCount = framesOut; 6124 activeTrack->releaseBuffer(&activeTrack->mSink); 6125 } 6126 } else { 6127 // FIXME could do a partial drop of framesOut 6128 if (activeTrack->mFramesToDrop > 0) { 6129 activeTrack->mFramesToDrop -= framesOut; 6130 if (activeTrack->mFramesToDrop <= 0) { 6131 activeTrack->clearSyncStartEvent(); 6132 } 6133 } else { 6134 activeTrack->mFramesToDrop += framesOut; 6135 if (activeTrack->mFramesToDrop >= 0 || activeTrack->mSyncStartEvent == 0 || 6136 activeTrack->mSyncStartEvent->isCancelled()) { 6137 ALOGW("Synced record %s, session %d, trigger session %d", 6138 (activeTrack->mFramesToDrop >= 0) ? "timed out" : "cancelled", 6139 activeTrack->sessionId(), 6140 (activeTrack->mSyncStartEvent != 0) ? 6141 activeTrack->mSyncStartEvent->triggerSession() : 6142 AUDIO_SESSION_NONE); 6143 activeTrack->clearSyncStartEvent(); 6144 } 6145 } 6146 } 6147 6148 if (framesOut == 0) { 6149 break; 6150 } 6151 } 6152 6153 switch (overrun) { 6154 case OVERRUN_TRUE: 6155 // client isn't retrieving buffers fast enough 6156 if (!activeTrack->setOverflow()) { 6157 nsecs_t now = systemTime(); 6158 // FIXME should lastWarning per track? 6159 if ((now - lastWarning) > kWarningThrottleNs) { 6160 ALOGW("RecordThread: buffer overflow"); 6161 lastWarning = now; 6162 } 6163 } 6164 break; 6165 case OVERRUN_FALSE: 6166 activeTrack->clearOverflow(); 6167 break; 6168 case OVERRUN_UNKNOWN: 6169 break; 6170 } 6171 6172 // update frame information and push timestamp out 6173 activeTrack->updateTrackFrameInfo( 6174 activeTrack->mServerProxy->framesReleased(), 6175 mTimestamp.mPosition[ExtendedTimestamp::LOCATION_SERVER], 6176 mSampleRate, mTimestamp); 6177 } 6178 6179unlock: 6180 // enable changes in effect chain 6181 unlockEffectChains(effectChains); 6182 // effectChains doesn't need to be cleared, since it is cleared by destructor at scope end 6183 } 6184 6185 standbyIfNotAlreadyInStandby(); 6186 6187 { 6188 Mutex::Autolock _l(mLock); 6189 for (size_t i = 0; i < mTracks.size(); i++) { 6190 sp<RecordTrack> track = mTracks[i]; 6191 track->invalidate(); 6192 } 6193 mActiveTracks.clear(); 6194 mActiveTracksGen++; 6195 mStartStopCond.broadcast(); 6196 } 6197 6198 releaseWakeLock(); 6199 6200 ALOGV("RecordThread %p exiting", this); 6201 return false; 6202} 6203 6204void AudioFlinger::RecordThread::standbyIfNotAlreadyInStandby() 6205{ 6206 if (!mStandby) { 6207 inputStandBy(); 6208 mStandby = true; 6209 } 6210} 6211 6212void AudioFlinger::RecordThread::inputStandBy() 6213{ 6214 // Idle the fast capture if it's currently running 6215 if (mFastCapture != 0) { 6216 FastCaptureStateQueue *sq = mFastCapture->sq(); 6217 FastCaptureState *state = sq->begin(); 6218 if (!(state->mCommand & FastCaptureState::IDLE)) { 6219 state->mCommand = FastCaptureState::COLD_IDLE; 6220 state->mColdFutexAddr = &mFastCaptureFutex; 6221 state->mColdGen++; 6222 mFastCaptureFutex = 0; 6223 sq->end(); 6224 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 6225 sq->push(FastCaptureStateQueue::BLOCK_UNTIL_ACKED); 6226#if 0 6227 if (kUseFastCapture == FastCapture_Dynamic) { 6228 // FIXME 6229 } 6230#endif 6231#ifdef AUDIO_WATCHDOG 6232 // FIXME 6233#endif 6234 } else { 6235 sq->end(false /*didModify*/); 6236 } 6237 } 6238 mInput->stream->common.standby(&mInput->stream->common); 6239} 6240 6241// RecordThread::createRecordTrack_l() must be called with AudioFlinger::mLock held 6242sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 6243 const sp<AudioFlinger::Client>& client, 6244 uint32_t sampleRate, 6245 audio_format_t format, 6246 audio_channel_mask_t channelMask, 6247 size_t *pFrameCount, 6248 audio_session_t sessionId, 6249 size_t *notificationFrames, 6250 int uid, 6251 IAudioFlinger::track_flags_t *flags, 6252 pid_t tid, 6253 status_t *status) 6254{ 6255 size_t frameCount = *pFrameCount; 6256 sp<RecordTrack> track; 6257 status_t lStatus; 6258 6259 // client expresses a preference for FAST, but we get the final say 6260 if (*flags & IAudioFlinger::TRACK_FAST) { 6261 if ( 6262 // we formerly checked for a callback handler (non-0 tid), 6263 // but that is no longer required for TRANSFER_OBTAIN mode 6264 // 6265 // frame count is not specified, or is exactly the pipe depth 6266 ((frameCount == 0) || (frameCount == mPipeFramesP2)) && 6267 // PCM data 6268 audio_is_linear_pcm(format) && 6269 // hardware format 6270 (format == mFormat) && 6271 // hardware channel mask 6272 (channelMask == mChannelMask) && 6273 // hardware sample rate 6274 (sampleRate == mSampleRate) && 6275 // record thread has an associated fast capture 6276 hasFastCapture() && 6277 // there are sufficient fast track slots available 6278 mFastTrackAvail 6279 ) { 6280 ALOGV("AUDIO_INPUT_FLAG_FAST accepted: frameCount=%zu mFrameCount=%zu", 6281 frameCount, mFrameCount); 6282 } else { 6283 ALOGV("AUDIO_INPUT_FLAG_FAST denied: frameCount=%zu mFrameCount=%zu mPipeFramesP2=%zu " 6284 "format=%#x isLinear=%d channelMask=%#x sampleRate=%u mSampleRate=%u " 6285 "hasFastCapture=%d tid=%d mFastTrackAvail=%d", 6286 frameCount, mFrameCount, mPipeFramesP2, 6287 format, audio_is_linear_pcm(format), channelMask, sampleRate, mSampleRate, 6288 hasFastCapture(), tid, mFastTrackAvail); 6289 *flags &= ~IAudioFlinger::TRACK_FAST; 6290 } 6291 } 6292 6293 // compute track buffer size in frames, and suggest the notification frame count 6294 if (*flags & IAudioFlinger::TRACK_FAST) { 6295 // fast track: frame count is exactly the pipe depth 6296 frameCount = mPipeFramesP2; 6297 // ignore requested notificationFrames, and always notify exactly once every HAL buffer 6298 *notificationFrames = mFrameCount; 6299 } else { 6300 // not fast track: max notification period is resampled equivalent of one HAL buffer time 6301 // or 20 ms if there is a fast capture 6302 // TODO This could be a roundupRatio inline, and const 6303 size_t maxNotificationFrames = ((int64_t) (hasFastCapture() ? mSampleRate/50 : mFrameCount) 6304 * sampleRate + mSampleRate - 1) / mSampleRate; 6305 // minimum number of notification periods is at least kMinNotifications, 6306 // and at least kMinMs rounded up to a whole notification period (minNotificationsByMs) 6307 static const size_t kMinNotifications = 3; 6308 static const uint32_t kMinMs = 30; 6309 // TODO This could be a roundupRatio inline 6310 const size_t minFramesByMs = (sampleRate * kMinMs + 1000 - 1) / 1000; 6311 // TODO This could be a roundupRatio inline 6312 const size_t minNotificationsByMs = (minFramesByMs + maxNotificationFrames - 1) / 6313 maxNotificationFrames; 6314 const size_t minFrameCount = maxNotificationFrames * 6315 max(kMinNotifications, minNotificationsByMs); 6316 frameCount = max(frameCount, minFrameCount); 6317 if (*notificationFrames == 0 || *notificationFrames > maxNotificationFrames) { 6318 *notificationFrames = maxNotificationFrames; 6319 } 6320 } 6321 *pFrameCount = frameCount; 6322 6323 lStatus = initCheck(); 6324 if (lStatus != NO_ERROR) { 6325 ALOGE("createRecordTrack_l() audio driver not initialized"); 6326 goto Exit; 6327 } 6328 6329 { // scope for mLock 6330 Mutex::Autolock _l(mLock); 6331 6332 track = new RecordTrack(this, client, sampleRate, 6333 format, channelMask, frameCount, NULL, sessionId, uid, 6334 *flags, TrackBase::TYPE_DEFAULT); 6335 6336 lStatus = track->initCheck(); 6337 if (lStatus != NO_ERROR) { 6338 ALOGE("createRecordTrack_l() initCheck failed %d; no control block?", lStatus); 6339 // track must be cleared from the caller as the caller has the AF lock 6340 goto Exit; 6341 } 6342 mTracks.add(track); 6343 6344 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 6345 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 6346 mAudioFlinger->btNrecIsOff(); 6347 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 6348 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 6349 6350 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 6351 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 6352 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 6353 // so ask activity manager to do this on our behalf 6354 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 6355 } 6356 } 6357 6358 lStatus = NO_ERROR; 6359 6360Exit: 6361 *status = lStatus; 6362 return track; 6363} 6364 6365status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, 6366 AudioSystem::sync_event_t event, 6367 audio_session_t triggerSession) 6368{ 6369 ALOGV("RecordThread::start event %d, triggerSession %d", event, triggerSession); 6370 sp<ThreadBase> strongMe = this; 6371 status_t status = NO_ERROR; 6372 6373 if (event == AudioSystem::SYNC_EVENT_NONE) { 6374 recordTrack->clearSyncStartEvent(); 6375 } else if (event != AudioSystem::SYNC_EVENT_SAME) { 6376 recordTrack->mSyncStartEvent = mAudioFlinger->createSyncEvent(event, 6377 triggerSession, 6378 recordTrack->sessionId(), 6379 syncStartEventCallback, 6380 recordTrack); 6381 // Sync event can be cancelled by the trigger session if the track is not in a 6382 // compatible state in which case we start record immediately 6383 if (recordTrack->mSyncStartEvent->isCancelled()) { 6384 recordTrack->clearSyncStartEvent(); 6385 } else { 6386 // do not wait for the event for more than AudioSystem::kSyncRecordStartTimeOutMs 6387 recordTrack->mFramesToDrop = - 6388 ((AudioSystem::kSyncRecordStartTimeOutMs * recordTrack->mSampleRate) / 1000); 6389 } 6390 } 6391 6392 { 6393 // This section is a rendezvous between binder thread executing start() and RecordThread 6394 AutoMutex lock(mLock); 6395 if (mActiveTracks.indexOf(recordTrack) >= 0) { 6396 if (recordTrack->mState == TrackBase::PAUSING) { 6397 ALOGV("active record track PAUSING -> ACTIVE"); 6398 recordTrack->mState = TrackBase::ACTIVE; 6399 } else { 6400 ALOGV("active record track state %d", recordTrack->mState); 6401 } 6402 return status; 6403 } 6404 6405 // TODO consider other ways of handling this, such as changing the state to :STARTING and 6406 // adding the track to mActiveTracks after returning from AudioSystem::startInput(), 6407 // or using a separate command thread 6408 recordTrack->mState = TrackBase::STARTING_1; 6409 mActiveTracks.add(recordTrack); 6410 mActiveTracksGen++; 6411 status_t status = NO_ERROR; 6412 if (recordTrack->isExternalTrack()) { 6413 mLock.unlock(); 6414 status = AudioSystem::startInput(mId, recordTrack->sessionId()); 6415 mLock.lock(); 6416 // FIXME should verify that recordTrack is still in mActiveTracks 6417 if (status != NO_ERROR) { 6418 mActiveTracks.remove(recordTrack); 6419 mActiveTracksGen++; 6420 recordTrack->clearSyncStartEvent(); 6421 ALOGV("RecordThread::start error %d", status); 6422 return status; 6423 } 6424 } 6425 // Catch up with current buffer indices if thread is already running. 6426 // This is what makes a new client discard all buffered data. If the track's mRsmpInFront 6427 // was initialized to some value closer to the thread's mRsmpInFront, then the track could 6428 // see previously buffered data before it called start(), but with greater risk of overrun. 6429 6430 recordTrack->mResamplerBufferProvider->reset(); 6431 // clear any converter state as new data will be discontinuous 6432 recordTrack->mRecordBufferConverter->reset(); 6433 recordTrack->mState = TrackBase::STARTING_2; 6434 // signal thread to start 6435 mWaitWorkCV.broadcast(); 6436 if (mActiveTracks.indexOf(recordTrack) < 0) { 6437 ALOGV("Record failed to start"); 6438 status = BAD_VALUE; 6439 goto startError; 6440 } 6441 return status; 6442 } 6443 6444startError: 6445 if (recordTrack->isExternalTrack()) { 6446 AudioSystem::stopInput(mId, recordTrack->sessionId()); 6447 } 6448 recordTrack->clearSyncStartEvent(); 6449 // FIXME I wonder why we do not reset the state here? 6450 return status; 6451} 6452 6453void AudioFlinger::RecordThread::syncStartEventCallback(const wp<SyncEvent>& event) 6454{ 6455 sp<SyncEvent> strongEvent = event.promote(); 6456 6457 if (strongEvent != 0) { 6458 sp<RefBase> ptr = strongEvent->cookie().promote(); 6459 if (ptr != 0) { 6460 RecordTrack *recordTrack = (RecordTrack *)ptr.get(); 6461 recordTrack->handleSyncStartEvent(strongEvent); 6462 } 6463 } 6464} 6465 6466bool AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 6467 ALOGV("RecordThread::stop"); 6468 AutoMutex _l(mLock); 6469 if (mActiveTracks.indexOf(recordTrack) != 0 || recordTrack->mState == TrackBase::PAUSING) { 6470 return false; 6471 } 6472 // note that threadLoop may still be processing the track at this point [without lock] 6473 recordTrack->mState = TrackBase::PAUSING; 6474 // do not wait for mStartStopCond if exiting 6475 if (exitPending()) { 6476 return true; 6477 } 6478 // FIXME incorrect usage of wait: no explicit predicate or loop 6479 mStartStopCond.wait(mLock); 6480 // if we have been restarted, recordTrack is in mActiveTracks here 6481 if (exitPending() || mActiveTracks.indexOf(recordTrack) != 0) { 6482 ALOGV("Record stopped OK"); 6483 return true; 6484 } 6485 return false; 6486} 6487 6488bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event __unused) const 6489{ 6490 return false; 6491} 6492 6493status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event __unused) 6494{ 6495#if 0 // This branch is currently dead code, but is preserved in case it will be needed in future 6496 if (!isValidSyncEvent(event)) { 6497 return BAD_VALUE; 6498 } 6499 6500 audio_session_t eventSession = event->triggerSession(); 6501 status_t ret = NAME_NOT_FOUND; 6502 6503 Mutex::Autolock _l(mLock); 6504 6505 for (size_t i = 0; i < mTracks.size(); i++) { 6506 sp<RecordTrack> track = mTracks[i]; 6507 if (eventSession == track->sessionId()) { 6508 (void) track->setSyncEvent(event); 6509 ret = NO_ERROR; 6510 } 6511 } 6512 return ret; 6513#else 6514 return BAD_VALUE; 6515#endif 6516} 6517 6518// destroyTrack_l() must be called with ThreadBase::mLock held 6519void AudioFlinger::RecordThread::destroyTrack_l(const sp<RecordTrack>& track) 6520{ 6521 track->terminate(); 6522 track->mState = TrackBase::STOPPED; 6523 // active tracks are removed by threadLoop() 6524 if (mActiveTracks.indexOf(track) < 0) { 6525 removeTrack_l(track); 6526 } 6527} 6528 6529void AudioFlinger::RecordThread::removeTrack_l(const sp<RecordTrack>& track) 6530{ 6531 mTracks.remove(track); 6532 // need anything related to effects here? 6533 if (track->isFastTrack()) { 6534 ALOG_ASSERT(!mFastTrackAvail); 6535 mFastTrackAvail = true; 6536 } 6537} 6538 6539void AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 6540{ 6541 dumpInternals(fd, args); 6542 dumpTracks(fd, args); 6543 dumpEffectChains(fd, args); 6544} 6545 6546void AudioFlinger::RecordThread::dumpInternals(int fd, const Vector<String16>& args) 6547{ 6548 dprintf(fd, "\nInput thread %p:\n", this); 6549 6550 dumpBase(fd, args); 6551 6552 if (mActiveTracks.size() == 0) { 6553 dprintf(fd, " No active record clients\n"); 6554 } 6555 dprintf(fd, " Fast capture thread: %s\n", hasFastCapture() ? "yes" : "no"); 6556 dprintf(fd, " Fast track available: %s\n", mFastTrackAvail ? "yes" : "no"); 6557 6558 // Make a non-atomic copy of fast capture dump state so it won't change underneath us 6559 // while we are dumping it. It may be inconsistent, but it won't mutate! 6560 // This is a large object so we place it on the heap. 6561 // FIXME 25972958: Need an intelligent copy constructor that does not touch unused pages. 6562 const FastCaptureDumpState *copy = new FastCaptureDumpState(mFastCaptureDumpState); 6563 copy->dump(fd); 6564 delete copy; 6565} 6566 6567void AudioFlinger::RecordThread::dumpTracks(int fd, const Vector<String16>& args __unused) 6568{ 6569 const size_t SIZE = 256; 6570 char buffer[SIZE]; 6571 String8 result; 6572 6573 size_t numtracks = mTracks.size(); 6574 size_t numactive = mActiveTracks.size(); 6575 size_t numactiveseen = 0; 6576 dprintf(fd, " %zu Tracks", numtracks); 6577 if (numtracks) { 6578 dprintf(fd, " of which %zu are active\n", numactive); 6579 RecordTrack::appendDumpHeader(result); 6580 for (size_t i = 0; i < numtracks ; ++i) { 6581 sp<RecordTrack> track = mTracks[i]; 6582 if (track != 0) { 6583 bool active = mActiveTracks.indexOf(track) >= 0; 6584 if (active) { 6585 numactiveseen++; 6586 } 6587 track->dump(buffer, SIZE, active); 6588 result.append(buffer); 6589 } 6590 } 6591 } else { 6592 dprintf(fd, "\n"); 6593 } 6594 6595 if (numactiveseen != numactive) { 6596 snprintf(buffer, SIZE, " The following tracks are in the active list but" 6597 " not in the track list\n"); 6598 result.append(buffer); 6599 RecordTrack::appendDumpHeader(result); 6600 for (size_t i = 0; i < numactive; ++i) { 6601 sp<RecordTrack> track = mActiveTracks[i]; 6602 if (mTracks.indexOf(track) < 0) { 6603 track->dump(buffer, SIZE, true); 6604 result.append(buffer); 6605 } 6606 } 6607 6608 } 6609 write(fd, result.string(), result.size()); 6610} 6611 6612 6613void AudioFlinger::RecordThread::ResamplerBufferProvider::reset() 6614{ 6615 sp<ThreadBase> threadBase = mRecordTrack->mThread.promote(); 6616 RecordThread *recordThread = (RecordThread *) threadBase.get(); 6617 mRsmpInFront = recordThread->mRsmpInRear; 6618 mRsmpInUnrel = 0; 6619} 6620 6621void AudioFlinger::RecordThread::ResamplerBufferProvider::sync( 6622 size_t *framesAvailable, bool *hasOverrun) 6623{ 6624 sp<ThreadBase> threadBase = mRecordTrack->mThread.promote(); 6625 RecordThread *recordThread = (RecordThread *) threadBase.get(); 6626 const int32_t rear = recordThread->mRsmpInRear; 6627 const int32_t front = mRsmpInFront; 6628 const ssize_t filled = rear - front; 6629 6630 size_t framesIn; 6631 bool overrun = false; 6632 if (filled < 0) { 6633 // should not happen, but treat like a massive overrun and re-sync 6634 framesIn = 0; 6635 mRsmpInFront = rear; 6636 overrun = true; 6637 } else if ((size_t) filled <= recordThread->mRsmpInFrames) { 6638 framesIn = (size_t) filled; 6639 } else { 6640 // client is not keeping up with server, but give it latest data 6641 framesIn = recordThread->mRsmpInFrames; 6642 mRsmpInFront = /* front = */ rear - framesIn; 6643 overrun = true; 6644 } 6645 if (framesAvailable != NULL) { 6646 *framesAvailable = framesIn; 6647 } 6648 if (hasOverrun != NULL) { 6649 *hasOverrun = overrun; 6650 } 6651} 6652 6653// AudioBufferProvider interface 6654status_t AudioFlinger::RecordThread::ResamplerBufferProvider::getNextBuffer( 6655 AudioBufferProvider::Buffer* buffer) 6656{ 6657 sp<ThreadBase> threadBase = mRecordTrack->mThread.promote(); 6658 if (threadBase == 0) { 6659 buffer->frameCount = 0; 6660 buffer->raw = NULL; 6661 return NOT_ENOUGH_DATA; 6662 } 6663 RecordThread *recordThread = (RecordThread *) threadBase.get(); 6664 int32_t rear = recordThread->mRsmpInRear; 6665 int32_t front = mRsmpInFront; 6666 ssize_t filled = rear - front; 6667 // FIXME should not be P2 (don't want to increase latency) 6668 // FIXME if client not keeping up, discard 6669 LOG_ALWAYS_FATAL_IF(!(0 <= filled && (size_t) filled <= recordThread->mRsmpInFrames)); 6670 // 'filled' may be non-contiguous, so return only the first contiguous chunk 6671 front &= recordThread->mRsmpInFramesP2 - 1; 6672 size_t part1 = recordThread->mRsmpInFramesP2 - front; 6673 if (part1 > (size_t) filled) { 6674 part1 = filled; 6675 } 6676 size_t ask = buffer->frameCount; 6677 ALOG_ASSERT(ask > 0); 6678 if (part1 > ask) { 6679 part1 = ask; 6680 } 6681 if (part1 == 0) { 6682 // out of data is fine since the resampler will return a short-count. 6683 buffer->raw = NULL; 6684 buffer->frameCount = 0; 6685 mRsmpInUnrel = 0; 6686 return NOT_ENOUGH_DATA; 6687 } 6688 6689 buffer->raw = (uint8_t*)recordThread->mRsmpInBuffer + front * recordThread->mFrameSize; 6690 buffer->frameCount = part1; 6691 mRsmpInUnrel = part1; 6692 return NO_ERROR; 6693} 6694 6695// AudioBufferProvider interface 6696void AudioFlinger::RecordThread::ResamplerBufferProvider::releaseBuffer( 6697 AudioBufferProvider::Buffer* buffer) 6698{ 6699 size_t stepCount = buffer->frameCount; 6700 if (stepCount == 0) { 6701 return; 6702 } 6703 ALOG_ASSERT(stepCount <= mRsmpInUnrel); 6704 mRsmpInUnrel -= stepCount; 6705 mRsmpInFront += stepCount; 6706 buffer->raw = NULL; 6707 buffer->frameCount = 0; 6708} 6709 6710AudioFlinger::RecordThread::RecordBufferConverter::RecordBufferConverter( 6711 audio_channel_mask_t srcChannelMask, audio_format_t srcFormat, 6712 uint32_t srcSampleRate, 6713 audio_channel_mask_t dstChannelMask, audio_format_t dstFormat, 6714 uint32_t dstSampleRate) : 6715 mSrcChannelMask(AUDIO_CHANNEL_INVALID), // updateParameters will set following vars 6716 // mSrcFormat 6717 // mSrcSampleRate 6718 // mDstChannelMask 6719 // mDstFormat 6720 // mDstSampleRate 6721 // mSrcChannelCount 6722 // mDstChannelCount 6723 // mDstFrameSize 6724 mBuf(NULL), mBufFrames(0), mBufFrameSize(0), 6725 mResampler(NULL), 6726 mIsLegacyDownmix(false), 6727 mIsLegacyUpmix(false), 6728 mRequiresFloat(false), 6729 mInputConverterProvider(NULL) 6730{ 6731 (void)updateParameters(srcChannelMask, srcFormat, srcSampleRate, 6732 dstChannelMask, dstFormat, dstSampleRate); 6733} 6734 6735AudioFlinger::RecordThread::RecordBufferConverter::~RecordBufferConverter() { 6736 free(mBuf); 6737 delete mResampler; 6738 delete mInputConverterProvider; 6739} 6740 6741size_t AudioFlinger::RecordThread::RecordBufferConverter::convert(void *dst, 6742 AudioBufferProvider *provider, size_t frames) 6743{ 6744 if (mInputConverterProvider != NULL) { 6745 mInputConverterProvider->setBufferProvider(provider); 6746 provider = mInputConverterProvider; 6747 } 6748 6749 if (mResampler == NULL) { 6750 ALOGVV("NO RESAMPLING sampleRate:%u mSrcFormat:%#x mDstFormat:%#x", 6751 mSrcSampleRate, mSrcFormat, mDstFormat); 6752 6753 AudioBufferProvider::Buffer buffer; 6754 for (size_t i = frames; i > 0; ) { 6755 buffer.frameCount = i; 6756 status_t status = provider->getNextBuffer(&buffer); 6757 if (status != OK || buffer.frameCount == 0) { 6758 frames -= i; // cannot fill request. 6759 break; 6760 } 6761 // format convert to destination buffer 6762 convertNoResampler(dst, buffer.raw, buffer.frameCount); 6763 6764 dst = (int8_t*)dst + buffer.frameCount * mDstFrameSize; 6765 i -= buffer.frameCount; 6766 provider->releaseBuffer(&buffer); 6767 } 6768 } else { 6769 ALOGVV("RESAMPLING mSrcSampleRate:%u mDstSampleRate:%u mSrcFormat:%#x mDstFormat:%#x", 6770 mSrcSampleRate, mDstSampleRate, mSrcFormat, mDstFormat); 6771 6772 // reallocate buffer if needed 6773 if (mBufFrameSize != 0 && mBufFrames < frames) { 6774 free(mBuf); 6775 mBufFrames = frames; 6776 (void)posix_memalign(&mBuf, 32, mBufFrames * mBufFrameSize); 6777 } 6778 // resampler accumulates, but we only have one source track 6779 memset(mBuf, 0, frames * mBufFrameSize); 6780 frames = mResampler->resample((int32_t*)mBuf, frames, provider); 6781 // format convert to destination buffer 6782 convertResampler(dst, mBuf, frames); 6783 } 6784 return frames; 6785} 6786 6787status_t AudioFlinger::RecordThread::RecordBufferConverter::updateParameters( 6788 audio_channel_mask_t srcChannelMask, audio_format_t srcFormat, 6789 uint32_t srcSampleRate, 6790 audio_channel_mask_t dstChannelMask, audio_format_t dstFormat, 6791 uint32_t dstSampleRate) 6792{ 6793 // quick evaluation if there is any change. 6794 if (mSrcFormat == srcFormat 6795 && mSrcChannelMask == srcChannelMask 6796 && mSrcSampleRate == srcSampleRate 6797 && mDstFormat == dstFormat 6798 && mDstChannelMask == dstChannelMask 6799 && mDstSampleRate == dstSampleRate) { 6800 return NO_ERROR; 6801 } 6802 6803 ALOGV("RecordBufferConverter updateParameters srcMask:%#x dstMask:%#x" 6804 " srcFormat:%#x dstFormat:%#x srcRate:%u dstRate:%u", 6805 srcChannelMask, dstChannelMask, srcFormat, dstFormat, srcSampleRate, dstSampleRate); 6806 const bool valid = 6807 audio_is_input_channel(srcChannelMask) 6808 && audio_is_input_channel(dstChannelMask) 6809 && audio_is_valid_format(srcFormat) && audio_is_linear_pcm(srcFormat) 6810 && audio_is_valid_format(dstFormat) && audio_is_linear_pcm(dstFormat) 6811 && (srcSampleRate <= dstSampleRate * AUDIO_RESAMPLER_DOWN_RATIO_MAX) 6812 ; // no upsampling checks for now 6813 if (!valid) { 6814 return BAD_VALUE; 6815 } 6816 6817 mSrcFormat = srcFormat; 6818 mSrcChannelMask = srcChannelMask; 6819 mSrcSampleRate = srcSampleRate; 6820 mDstFormat = dstFormat; 6821 mDstChannelMask = dstChannelMask; 6822 mDstSampleRate = dstSampleRate; 6823 6824 // compute derived parameters 6825 mSrcChannelCount = audio_channel_count_from_in_mask(srcChannelMask); 6826 mDstChannelCount = audio_channel_count_from_in_mask(dstChannelMask); 6827 mDstFrameSize = mDstChannelCount * audio_bytes_per_sample(mDstFormat); 6828 6829 // do we need to resample? 6830 delete mResampler; 6831 mResampler = NULL; 6832 if (mSrcSampleRate != mDstSampleRate) { 6833 mResampler = AudioResampler::create(AUDIO_FORMAT_PCM_FLOAT, 6834 mSrcChannelCount, mDstSampleRate); 6835 mResampler->setSampleRate(mSrcSampleRate); 6836 mResampler->setVolume(AudioMixer::UNITY_GAIN_FLOAT, AudioMixer::UNITY_GAIN_FLOAT); 6837 } 6838 6839 // are we running legacy channel conversion modes? 6840 mIsLegacyDownmix = (mSrcChannelMask == AUDIO_CHANNEL_IN_STEREO 6841 || mSrcChannelMask == AUDIO_CHANNEL_IN_FRONT_BACK) 6842 && mDstChannelMask == AUDIO_CHANNEL_IN_MONO; 6843 mIsLegacyUpmix = mSrcChannelMask == AUDIO_CHANNEL_IN_MONO 6844 && (mDstChannelMask == AUDIO_CHANNEL_IN_STEREO 6845 || mDstChannelMask == AUDIO_CHANNEL_IN_FRONT_BACK); 6846 6847 // do we need to process in float? 6848 mRequiresFloat = mResampler != NULL || mIsLegacyDownmix || mIsLegacyUpmix; 6849 6850 // do we need a staging buffer to convert for destination (we can still optimize this)? 6851 // we use mBufFrameSize > 0 to indicate both frame size as well as buffer necessity 6852 if (mResampler != NULL) { 6853 mBufFrameSize = max(mSrcChannelCount, FCC_2) 6854 * audio_bytes_per_sample(AUDIO_FORMAT_PCM_FLOAT); 6855 } else if (mIsLegacyUpmix || mIsLegacyDownmix) { // legacy modes always float 6856 mBufFrameSize = mDstChannelCount * audio_bytes_per_sample(AUDIO_FORMAT_PCM_FLOAT); 6857 } else if (mSrcChannelMask != mDstChannelMask && mDstFormat != mSrcFormat) { 6858 mBufFrameSize = mDstChannelCount * audio_bytes_per_sample(mSrcFormat); 6859 } else { 6860 mBufFrameSize = 0; 6861 } 6862 mBufFrames = 0; // force the buffer to be resized. 6863 6864 // do we need an input converter buffer provider to give us float? 6865 delete mInputConverterProvider; 6866 mInputConverterProvider = NULL; 6867 if (mRequiresFloat && mSrcFormat != AUDIO_FORMAT_PCM_FLOAT) { 6868 mInputConverterProvider = new ReformatBufferProvider( 6869 audio_channel_count_from_in_mask(mSrcChannelMask), 6870 mSrcFormat, 6871 AUDIO_FORMAT_PCM_FLOAT, 6872 256 /* provider buffer frame count */); 6873 } 6874 6875 // do we need a remixer to do channel mask conversion 6876 if (!mIsLegacyDownmix && !mIsLegacyUpmix && mSrcChannelMask != mDstChannelMask) { 6877 (void) memcpy_by_index_array_initialization_from_channel_mask( 6878 mIdxAry, ARRAY_SIZE(mIdxAry), mDstChannelMask, mSrcChannelMask); 6879 } 6880 return NO_ERROR; 6881} 6882 6883void AudioFlinger::RecordThread::RecordBufferConverter::convertNoResampler( 6884 void *dst, const void *src, size_t frames) 6885{ 6886 // src is native type unless there is legacy upmix or downmix, whereupon it is float. 6887 if (mBufFrameSize != 0 && mBufFrames < frames) { 6888 free(mBuf); 6889 mBufFrames = frames; 6890 (void)posix_memalign(&mBuf, 32, mBufFrames * mBufFrameSize); 6891 } 6892 // do we need to do legacy upmix and downmix? 6893 if (mIsLegacyUpmix || mIsLegacyDownmix) { 6894 void *dstBuf = mBuf != NULL ? mBuf : dst; 6895 if (mIsLegacyUpmix) { 6896 upmix_to_stereo_float_from_mono_float((float *)dstBuf, 6897 (const float *)src, frames); 6898 } else /*mIsLegacyDownmix */ { 6899 downmix_to_mono_float_from_stereo_float((float *)dstBuf, 6900 (const float *)src, frames); 6901 } 6902 if (mBuf != NULL) { 6903 memcpy_by_audio_format(dst, mDstFormat, mBuf, AUDIO_FORMAT_PCM_FLOAT, 6904 frames * mDstChannelCount); 6905 } 6906 return; 6907 } 6908 // do we need to do channel mask conversion? 6909 if (mSrcChannelMask != mDstChannelMask) { 6910 void *dstBuf = mBuf != NULL ? mBuf : dst; 6911 memcpy_by_index_array(dstBuf, mDstChannelCount, 6912 src, mSrcChannelCount, mIdxAry, audio_bytes_per_sample(mSrcFormat), frames); 6913 if (dstBuf == dst) { 6914 return; // format is the same 6915 } 6916 } 6917 // convert to destination buffer 6918 const void *convertBuf = mBuf != NULL ? mBuf : src; 6919 memcpy_by_audio_format(dst, mDstFormat, convertBuf, mSrcFormat, 6920 frames * mDstChannelCount); 6921} 6922 6923void AudioFlinger::RecordThread::RecordBufferConverter::convertResampler( 6924 void *dst, /*not-a-const*/ void *src, size_t frames) 6925{ 6926 // src buffer format is ALWAYS float when entering this routine 6927 if (mIsLegacyUpmix) { 6928 ; // mono to stereo already handled by resampler 6929 } else if (mIsLegacyDownmix 6930 || (mSrcChannelMask == mDstChannelMask && mSrcChannelCount == 1)) { 6931 // the resampler outputs stereo for mono input channel (a feature?) 6932 // must convert to mono 6933 downmix_to_mono_float_from_stereo_float((float *)src, 6934 (const float *)src, frames); 6935 } else if (mSrcChannelMask != mDstChannelMask) { 6936 // convert to mono channel again for channel mask conversion (could be skipped 6937 // with further optimization). 6938 if (mSrcChannelCount == 1) { 6939 downmix_to_mono_float_from_stereo_float((float *)src, 6940 (const float *)src, frames); 6941 } 6942 // convert to destination format (in place, OK as float is larger than other types) 6943 if (mDstFormat != AUDIO_FORMAT_PCM_FLOAT) { 6944 memcpy_by_audio_format(src, mDstFormat, src, AUDIO_FORMAT_PCM_FLOAT, 6945 frames * mSrcChannelCount); 6946 } 6947 // channel convert and save to dst 6948 memcpy_by_index_array(dst, mDstChannelCount, 6949 src, mSrcChannelCount, mIdxAry, audio_bytes_per_sample(mDstFormat), frames); 6950 return; 6951 } 6952 // convert to destination format and save to dst 6953 memcpy_by_audio_format(dst, mDstFormat, src, AUDIO_FORMAT_PCM_FLOAT, 6954 frames * mDstChannelCount); 6955} 6956 6957bool AudioFlinger::RecordThread::checkForNewParameter_l(const String8& keyValuePair, 6958 status_t& status) 6959{ 6960 bool reconfig = false; 6961 6962 status = NO_ERROR; 6963 6964 audio_format_t reqFormat = mFormat; 6965 uint32_t samplingRate = mSampleRate; 6966 // TODO this may change if we want to support capture from HDMI PCM multi channel (e.g on TVs). 6967 audio_channel_mask_t channelMask = audio_channel_in_mask_from_count(mChannelCount); 6968 6969 AudioParameter param = AudioParameter(keyValuePair); 6970 int value; 6971 6972 // scope for AutoPark extends to end of method 6973 AutoPark<FastCapture> park(mFastCapture); 6974 6975 // TODO Investigate when this code runs. Check with audio policy when a sample rate and 6976 // channel count change can be requested. Do we mandate the first client defines the 6977 // HAL sampling rate and channel count or do we allow changes on the fly? 6978 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 6979 samplingRate = value; 6980 reconfig = true; 6981 } 6982 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 6983 if (!audio_is_linear_pcm((audio_format_t) value)) { 6984 status = BAD_VALUE; 6985 } else { 6986 reqFormat = (audio_format_t) value; 6987 reconfig = true; 6988 } 6989 } 6990 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 6991 audio_channel_mask_t mask = (audio_channel_mask_t) value; 6992 if (!audio_is_input_channel(mask) || 6993 audio_channel_count_from_in_mask(mask) > FCC_8) { 6994 status = BAD_VALUE; 6995 } else { 6996 channelMask = mask; 6997 reconfig = true; 6998 } 6999 } 7000 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 7001 // do not accept frame count changes if tracks are open as the track buffer 7002 // size depends on frame count and correct behavior would not be guaranteed 7003 // if frame count is changed after track creation 7004 if (mActiveTracks.size() > 0) { 7005 status = INVALID_OPERATION; 7006 } else { 7007 reconfig = true; 7008 } 7009 } 7010 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 7011 // forward device change to effects that have requested to be 7012 // aware of attached audio device. 7013 for (size_t i = 0; i < mEffectChains.size(); i++) { 7014 mEffectChains[i]->setDevice_l(value); 7015 } 7016 7017 // store input device and output device but do not forward output device to audio HAL. 7018 // Note that status is ignored by the caller for output device 7019 // (see AudioFlinger::setParameters() 7020 if (audio_is_output_devices(value)) { 7021 mOutDevice = value; 7022 status = BAD_VALUE; 7023 } else { 7024 mInDevice = value; 7025 if (value != AUDIO_DEVICE_NONE) { 7026 mPrevInDevice = value; 7027 } 7028 // disable AEC and NS if the device is a BT SCO headset supporting those 7029 // pre processings 7030 if (mTracks.size() > 0) { 7031 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 7032 mAudioFlinger->btNrecIsOff(); 7033 for (size_t i = 0; i < mTracks.size(); i++) { 7034 sp<RecordTrack> track = mTracks[i]; 7035 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId()); 7036 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId()); 7037 } 7038 } 7039 } 7040 } 7041 if (param.getInt(String8(AudioParameter::keyInputSource), value) == NO_ERROR && 7042 mAudioSource != (audio_source_t)value) { 7043 // forward device change to effects that have requested to be 7044 // aware of attached audio device. 7045 for (size_t i = 0; i < mEffectChains.size(); i++) { 7046 mEffectChains[i]->setAudioSource_l((audio_source_t)value); 7047 } 7048 mAudioSource = (audio_source_t)value; 7049 } 7050 7051 if (status == NO_ERROR) { 7052 status = mInput->stream->common.set_parameters(&mInput->stream->common, 7053 keyValuePair.string()); 7054 if (status == INVALID_OPERATION) { 7055 inputStandBy(); 7056 status = mInput->stream->common.set_parameters(&mInput->stream->common, 7057 keyValuePair.string()); 7058 } 7059 if (reconfig) { 7060 if (status == BAD_VALUE && 7061 audio_is_linear_pcm(mInput->stream->common.get_format(&mInput->stream->common)) && 7062 audio_is_linear_pcm(reqFormat) && 7063 (mInput->stream->common.get_sample_rate(&mInput->stream->common) 7064 <= (AUDIO_RESAMPLER_DOWN_RATIO_MAX * samplingRate)) && 7065 audio_channel_count_from_in_mask( 7066 mInput->stream->common.get_channels(&mInput->stream->common)) <= FCC_8) { 7067 status = NO_ERROR; 7068 } 7069 if (status == NO_ERROR) { 7070 readInputParameters_l(); 7071 sendIoConfigEvent_l(AUDIO_INPUT_CONFIG_CHANGED); 7072 } 7073 } 7074 } 7075 7076 return reconfig; 7077} 7078 7079String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 7080{ 7081 Mutex::Autolock _l(mLock); 7082 if (initCheck() != NO_ERROR) { 7083 return String8(); 7084 } 7085 7086 char *s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 7087 const String8 out_s8(s); 7088 free(s); 7089 return out_s8; 7090} 7091 7092void AudioFlinger::RecordThread::ioConfigChanged(audio_io_config_event event, pid_t pid) { 7093 sp<AudioIoDescriptor> desc = new AudioIoDescriptor(); 7094 7095 desc->mIoHandle = mId; 7096 7097 switch (event) { 7098 case AUDIO_INPUT_OPENED: 7099 case AUDIO_INPUT_CONFIG_CHANGED: 7100 desc->mPatch = mPatch; 7101 desc->mChannelMask = mChannelMask; 7102 desc->mSamplingRate = mSampleRate; 7103 desc->mFormat = mFormat; 7104 desc->mFrameCount = mFrameCount; 7105 desc->mFrameCountHAL = mFrameCount; 7106 desc->mLatency = 0; 7107 break; 7108 7109 case AUDIO_INPUT_CLOSED: 7110 default: 7111 break; 7112 } 7113 mAudioFlinger->ioConfigChanged(event, desc, pid); 7114} 7115 7116void AudioFlinger::RecordThread::readInputParameters_l() 7117{ 7118 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 7119 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 7120 mChannelCount = audio_channel_count_from_in_mask(mChannelMask); 7121 if (mChannelCount > FCC_8) { 7122 ALOGE("HAL channel count %d > %d", mChannelCount, FCC_8); 7123 } 7124 mHALFormat = mInput->stream->common.get_format(&mInput->stream->common); 7125 mFormat = mHALFormat; 7126 if (!audio_is_linear_pcm(mFormat)) { 7127 ALOGE("HAL format %#x is not linear pcm", mFormat); 7128 } 7129 mFrameSize = audio_stream_in_frame_size(mInput->stream); 7130 mBufferSize = mInput->stream->common.get_buffer_size(&mInput->stream->common); 7131 mFrameCount = mBufferSize / mFrameSize; 7132 // This is the formula for calculating the temporary buffer size. 7133 // With 7 HAL buffers, we can guarantee ability to down-sample the input by ratio of 6:1 to 7134 // 1 full output buffer, regardless of the alignment of the available input. 7135 // The value is somewhat arbitrary, and could probably be even larger. 7136 // A larger value should allow more old data to be read after a track calls start(), 7137 // without increasing latency. 7138 // 7139 // Note this is independent of the maximum downsampling ratio permitted for capture. 7140 mRsmpInFrames = mFrameCount * 7; 7141 mRsmpInFramesP2 = roundup(mRsmpInFrames); 7142 free(mRsmpInBuffer); 7143 mRsmpInBuffer = NULL; 7144 7145 // TODO optimize audio capture buffer sizes ... 7146 // Here we calculate the size of the sliding buffer used as a source 7147 // for resampling. mRsmpInFramesP2 is currently roundup(mFrameCount * 7). 7148 // For current HAL frame counts, this is usually 2048 = 40 ms. It would 7149 // be better to have it derived from the pipe depth in the long term. 7150 // The current value is higher than necessary. However it should not add to latency. 7151 7152 // Over-allocate beyond mRsmpInFramesP2 to permit a HAL read past end of buffer 7153 size_t bufferSize = (mRsmpInFramesP2 + mFrameCount - 1) * mFrameSize; 7154 (void)posix_memalign(&mRsmpInBuffer, 32, bufferSize); 7155 memset(mRsmpInBuffer, 0, bufferSize); // if posix_memalign fails, will segv here. 7156 7157 // AudioRecord mSampleRate and mChannelCount are constant due to AudioRecord API constraints. 7158 // But if thread's mSampleRate or mChannelCount changes, how will that affect active tracks? 7159} 7160 7161uint32_t AudioFlinger::RecordThread::getInputFramesLost() 7162{ 7163 Mutex::Autolock _l(mLock); 7164 if (initCheck() != NO_ERROR) { 7165 return 0; 7166 } 7167 7168 return mInput->stream->get_input_frames_lost(mInput->stream); 7169} 7170 7171uint32_t AudioFlinger::RecordThread::hasAudioSession(audio_session_t sessionId) const 7172{ 7173 Mutex::Autolock _l(mLock); 7174 uint32_t result = 0; 7175 if (getEffectChain_l(sessionId) != 0) { 7176 result = EFFECT_SESSION; 7177 } 7178 7179 for (size_t i = 0; i < mTracks.size(); ++i) { 7180 if (sessionId == mTracks[i]->sessionId()) { 7181 result |= TRACK_SESSION; 7182 break; 7183 } 7184 } 7185 7186 return result; 7187} 7188 7189KeyedVector<audio_session_t, bool> AudioFlinger::RecordThread::sessionIds() const 7190{ 7191 KeyedVector<audio_session_t, bool> ids; 7192 Mutex::Autolock _l(mLock); 7193 for (size_t j = 0; j < mTracks.size(); ++j) { 7194 sp<RecordThread::RecordTrack> track = mTracks[j]; 7195 audio_session_t sessionId = track->sessionId(); 7196 if (ids.indexOfKey(sessionId) < 0) { 7197 ids.add(sessionId, true); 7198 } 7199 } 7200 return ids; 7201} 7202 7203AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 7204{ 7205 Mutex::Autolock _l(mLock); 7206 AudioStreamIn *input = mInput; 7207 mInput = NULL; 7208 return input; 7209} 7210 7211// this method must always be called either with ThreadBase mLock held or inside the thread loop 7212audio_stream_t* AudioFlinger::RecordThread::stream() const 7213{ 7214 if (mInput == NULL) { 7215 return NULL; 7216 } 7217 return &mInput->stream->common; 7218} 7219 7220status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 7221{ 7222 // only one chain per input thread 7223 if (mEffectChains.size() != 0) { 7224 ALOGW("addEffectChain_l() already one chain %p on thread %p", chain.get(), this); 7225 return INVALID_OPERATION; 7226 } 7227 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 7228 chain->setThread(this); 7229 chain->setInBuffer(NULL); 7230 chain->setOutBuffer(NULL); 7231 7232 checkSuspendOnAddEffectChain_l(chain); 7233 7234 // make sure enabled pre processing effects state is communicated to the HAL as we 7235 // just moved them to a new input stream. 7236 chain->syncHalEffectsState(); 7237 7238 mEffectChains.add(chain); 7239 7240 return NO_ERROR; 7241} 7242 7243size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 7244{ 7245 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 7246 ALOGW_IF(mEffectChains.size() != 1, 7247 "removeEffectChain_l() %p invalid chain size %zu on thread %p", 7248 chain.get(), mEffectChains.size(), this); 7249 if (mEffectChains.size() == 1) { 7250 mEffectChains.removeAt(0); 7251 } 7252 return 0; 7253} 7254 7255status_t AudioFlinger::RecordThread::createAudioPatch_l(const struct audio_patch *patch, 7256 audio_patch_handle_t *handle) 7257{ 7258 status_t status = NO_ERROR; 7259 7260 // store new device and send to effects 7261 mInDevice = patch->sources[0].ext.device.type; 7262 mPatch = *patch; 7263 for (size_t i = 0; i < mEffectChains.size(); i++) { 7264 mEffectChains[i]->setDevice_l(mInDevice); 7265 } 7266 7267 // disable AEC and NS if the device is a BT SCO headset supporting those 7268 // pre processings 7269 if (mTracks.size() > 0) { 7270 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 7271 mAudioFlinger->btNrecIsOff(); 7272 for (size_t i = 0; i < mTracks.size(); i++) { 7273 sp<RecordTrack> track = mTracks[i]; 7274 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId()); 7275 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId()); 7276 } 7277 } 7278 7279 // store new source and send to effects 7280 if (mAudioSource != patch->sinks[0].ext.mix.usecase.source) { 7281 mAudioSource = patch->sinks[0].ext.mix.usecase.source; 7282 for (size_t i = 0; i < mEffectChains.size(); i++) { 7283 mEffectChains[i]->setAudioSource_l(mAudioSource); 7284 } 7285 } 7286 7287 if (mInput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 7288 audio_hw_device_t *hwDevice = mInput->audioHwDev->hwDevice(); 7289 status = hwDevice->create_audio_patch(hwDevice, 7290 patch->num_sources, 7291 patch->sources, 7292 patch->num_sinks, 7293 patch->sinks, 7294 handle); 7295 } else { 7296 char *address; 7297 if (strcmp(patch->sources[0].ext.device.address, "") != 0) { 7298 address = audio_device_address_to_parameter( 7299 patch->sources[0].ext.device.type, 7300 patch->sources[0].ext.device.address); 7301 } else { 7302 address = (char *)calloc(1, 1); 7303 } 7304 AudioParameter param = AudioParameter(String8(address)); 7305 free(address); 7306 param.addInt(String8(AUDIO_PARAMETER_STREAM_ROUTING), 7307 (int)patch->sources[0].ext.device.type); 7308 param.addInt(String8(AUDIO_PARAMETER_STREAM_INPUT_SOURCE), 7309 (int)patch->sinks[0].ext.mix.usecase.source); 7310 status = mInput->stream->common.set_parameters(&mInput->stream->common, 7311 param.toString().string()); 7312 *handle = AUDIO_PATCH_HANDLE_NONE; 7313 } 7314 7315 if (mInDevice != mPrevInDevice) { 7316 sendIoConfigEvent_l(AUDIO_INPUT_CONFIG_CHANGED); 7317 mPrevInDevice = mInDevice; 7318 } 7319 7320 return status; 7321} 7322 7323status_t AudioFlinger::RecordThread::releaseAudioPatch_l(const audio_patch_handle_t handle) 7324{ 7325 status_t status = NO_ERROR; 7326 7327 mInDevice = AUDIO_DEVICE_NONE; 7328 7329 if (mInput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 7330 audio_hw_device_t *hwDevice = mInput->audioHwDev->hwDevice(); 7331 status = hwDevice->release_audio_patch(hwDevice, handle); 7332 } else { 7333 AudioParameter param; 7334 param.addInt(String8(AUDIO_PARAMETER_STREAM_ROUTING), 0); 7335 status = mInput->stream->common.set_parameters(&mInput->stream->common, 7336 param.toString().string()); 7337 } 7338 return status; 7339} 7340 7341void AudioFlinger::RecordThread::addPatchRecord(const sp<PatchRecord>& record) 7342{ 7343 Mutex::Autolock _l(mLock); 7344 mTracks.add(record); 7345} 7346 7347void AudioFlinger::RecordThread::deletePatchRecord(const sp<PatchRecord>& record) 7348{ 7349 Mutex::Autolock _l(mLock); 7350 destroyTrack_l(record); 7351} 7352 7353void AudioFlinger::RecordThread::getAudioPortConfig(struct audio_port_config *config) 7354{ 7355 ThreadBase::getAudioPortConfig(config); 7356 config->role = AUDIO_PORT_ROLE_SINK; 7357 config->ext.mix.hw_module = mInput->audioHwDev->handle(); 7358 config->ext.mix.usecase.source = mAudioSource; 7359} 7360 7361} // namespace android 7362