Threads.cpp revision e964d4e421e2d1ca937227a580c0c837091a11e3
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 const 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// checkEffectCompatibility_l() must be called with ThreadBase::mLock held 1260status_t AudioFlinger::RecordThread::checkEffectCompatibility_l( 1261 const effect_descriptor_t *desc, audio_session_t sessionId) 1262{ 1263 // No global effect sessions on record threads 1264 if (sessionId == AUDIO_SESSION_OUTPUT_MIX || sessionId == AUDIO_SESSION_OUTPUT_STAGE) { 1265 ALOGW("checkEffectCompatibility_l(): global effect %s on record thread %s", 1266 desc->name, mThreadName); 1267 return BAD_VALUE; 1268 } 1269 // only pre processing effects on record thread 1270 if ((desc->flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_PRE_PROC) { 1271 ALOGW("checkEffectCompatibility_l(): non pre processing effect %s on record thread %s", 1272 desc->name, mThreadName); 1273 return BAD_VALUE; 1274 } 1275 audio_input_flags_t flags = mInput->flags; 1276 if (hasFastCapture() || (flags & AUDIO_INPUT_FLAG_FAST)) { 1277 if (flags & AUDIO_INPUT_FLAG_RAW) { 1278 ALOGW("checkEffectCompatibility_l(): effect %s on record thread %s in raw mode", 1279 desc->name, mThreadName); 1280 return BAD_VALUE; 1281 } 1282 if ((desc->flags & EFFECT_FLAG_HW_ACC_TUNNEL) == 0) { 1283 ALOGW("checkEffectCompatibility_l(): non HW effect %s on record thread %s in fast mode", 1284 desc->name, mThreadName); 1285 return BAD_VALUE; 1286 } 1287 } 1288 return NO_ERROR; 1289} 1290 1291// checkEffectCompatibility_l() must be called with ThreadBase::mLock held 1292status_t AudioFlinger::PlaybackThread::checkEffectCompatibility_l( 1293 const effect_descriptor_t *desc, audio_session_t sessionId) 1294{ 1295 // no preprocessing on playback threads 1296 if ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC) { 1297 ALOGW("checkEffectCompatibility_l(): pre processing effect %s created on playback" 1298 " thread %s", desc->name, mThreadName); 1299 return BAD_VALUE; 1300 } 1301 1302 switch (mType) { 1303 case MIXER: { 1304 // Reject any effect on mixer multichannel sinks. 1305 // TODO: fix both format and multichannel issues with effects. 1306 if (mChannelCount != FCC_2) { 1307 ALOGW("checkEffectCompatibility_l(): effect %s for multichannel(%d) on MIXER" 1308 " thread %s", desc->name, mChannelCount, mThreadName); 1309 return BAD_VALUE; 1310 } 1311 audio_output_flags_t flags = mOutput->flags; 1312 if (hasFastMixer() || (flags & AUDIO_OUTPUT_FLAG_FAST)) { 1313 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 1314 // global effects are applied only to non fast tracks if they are SW 1315 if ((desc->flags & EFFECT_FLAG_HW_ACC_TUNNEL) == 0) { 1316 break; 1317 } 1318 } else if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) { 1319 // only post processing on output stage session 1320 if ((desc->flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_POST_PROC) { 1321 ALOGW("checkEffectCompatibility_l(): non post processing effect %s not allowed" 1322 " on output stage session", desc->name); 1323 return BAD_VALUE; 1324 } 1325 } else { 1326 // no restriction on effects applied on non fast tracks 1327 if ((hasAudioSession_l(sessionId) & ThreadBase::FAST_SESSION) == 0) { 1328 break; 1329 } 1330 } 1331 if (flags & AUDIO_OUTPUT_FLAG_RAW) { 1332 ALOGW("checkEffectCompatibility_l(): effect %s on playback thread in raw mode", 1333 desc->name); 1334 return BAD_VALUE; 1335 } 1336 if ((desc->flags & EFFECT_FLAG_HW_ACC_TUNNEL) == 0) { 1337 ALOGW("checkEffectCompatibility_l(): non HW effect %s on playback thread" 1338 " in fast mode", desc->name); 1339 return BAD_VALUE; 1340 } 1341 } 1342 } break; 1343 case OFFLOAD: 1344 // nothing actionable on offload threads, if the effect: 1345 // - is offloadable: the effect can be created 1346 // - is NOT offloadable: the effect should still be created, but EffectHandle::enable() 1347 // will take care of invalidating the tracks of the thread 1348 break; 1349 case DIRECT: 1350 // Reject any effect on Direct output threads for now, since the format of 1351 // mSinkBuffer is not guaranteed to be compatible with effect processing (PCM 16 stereo). 1352 ALOGW("checkEffectCompatibility_l(): effect %s on DIRECT output thread %s", 1353 desc->name, mThreadName); 1354 return BAD_VALUE; 1355 case DUPLICATING: 1356 // Reject any effect on mixer multichannel sinks. 1357 // TODO: fix both format and multichannel issues with effects. 1358 if (mChannelCount != FCC_2) { 1359 ALOGW("checkEffectCompatibility_l(): effect %s for multichannel(%d)" 1360 " on DUPLICATING thread %s", desc->name, mChannelCount, mThreadName); 1361 return BAD_VALUE; 1362 } 1363 if ((sessionId == AUDIO_SESSION_OUTPUT_STAGE) || (sessionId == AUDIO_SESSION_OUTPUT_MIX)) { 1364 ALOGW("checkEffectCompatibility_l(): global effect %s on DUPLICATING" 1365 " thread %s", desc->name, mThreadName); 1366 return BAD_VALUE; 1367 } 1368 if ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC) { 1369 ALOGW("checkEffectCompatibility_l(): post processing effect %s on" 1370 " DUPLICATING thread %s", desc->name, mThreadName); 1371 return BAD_VALUE; 1372 } 1373 if ((desc->flags & EFFECT_FLAG_HW_ACC_TUNNEL) != 0) { 1374 ALOGW("checkEffectCompatibility_l(): HW tunneled effect %s on" 1375 " DUPLICATING thread %s", desc->name, mThreadName); 1376 return BAD_VALUE; 1377 } 1378 break; 1379 default: 1380 LOG_ALWAYS_FATAL("checkEffectCompatibility_l(): wrong thread type %d", mType); 1381 } 1382 1383 return NO_ERROR; 1384} 1385 1386// ThreadBase::createEffect_l() must be called with AudioFlinger::mLock held 1387sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 1388 const sp<AudioFlinger::Client>& client, 1389 const sp<IEffectClient>& effectClient, 1390 int32_t priority, 1391 audio_session_t sessionId, 1392 effect_descriptor_t *desc, 1393 int *enabled, 1394 status_t *status) 1395{ 1396 sp<EffectModule> effect; 1397 sp<EffectHandle> handle; 1398 status_t lStatus; 1399 sp<EffectChain> chain; 1400 bool chainCreated = false; 1401 bool effectCreated = false; 1402 bool effectRegistered = false; 1403 1404 lStatus = initCheck(); 1405 if (lStatus != NO_ERROR) { 1406 ALOGW("createEffect_l() Audio driver not initialized."); 1407 goto Exit; 1408 } 1409 1410 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 1411 1412 { // scope for mLock 1413 Mutex::Autolock _l(mLock); 1414 1415 lStatus = checkEffectCompatibility_l(desc, sessionId); 1416 if (lStatus != NO_ERROR) { 1417 goto Exit; 1418 } 1419 1420 // check for existing effect chain with the requested audio session 1421 chain = getEffectChain_l(sessionId); 1422 if (chain == 0) { 1423 // create a new chain for this session 1424 ALOGV("createEffect_l() new effect chain for session %d", sessionId); 1425 chain = new EffectChain(this, sessionId); 1426 addEffectChain_l(chain); 1427 chain->setStrategy(getStrategyForSession_l(sessionId)); 1428 chainCreated = true; 1429 } else { 1430 effect = chain->getEffectFromDesc_l(desc); 1431 } 1432 1433 ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get()); 1434 1435 if (effect == 0) { 1436 audio_unique_id_t id = mAudioFlinger->nextUniqueId(AUDIO_UNIQUE_ID_USE_EFFECT); 1437 // Check CPU and memory usage 1438 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 1439 if (lStatus != NO_ERROR) { 1440 goto Exit; 1441 } 1442 effectRegistered = true; 1443 // create a new effect module if none present in the chain 1444 effect = new EffectModule(this, chain, desc, id, sessionId); 1445 lStatus = effect->status(); 1446 if (lStatus != NO_ERROR) { 1447 goto Exit; 1448 } 1449 effect->setOffloaded(mType == OFFLOAD, mId); 1450 1451 lStatus = chain->addEffect_l(effect); 1452 if (lStatus != NO_ERROR) { 1453 goto Exit; 1454 } 1455 effectCreated = true; 1456 1457 effect->setDevice(mOutDevice); 1458 effect->setDevice(mInDevice); 1459 effect->setMode(mAudioFlinger->getMode()); 1460 effect->setAudioSource(mAudioSource); 1461 } 1462 // create effect handle and connect it to effect module 1463 handle = new EffectHandle(effect, client, effectClient, priority); 1464 lStatus = handle->initCheck(); 1465 if (lStatus == OK) { 1466 lStatus = effect->addHandle(handle.get()); 1467 } 1468 if (enabled != NULL) { 1469 *enabled = (int)effect->isEnabled(); 1470 } 1471 } 1472 1473Exit: 1474 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 1475 Mutex::Autolock _l(mLock); 1476 if (effectCreated) { 1477 chain->removeEffect_l(effect); 1478 } 1479 if (effectRegistered) { 1480 AudioSystem::unregisterEffect(effect->id()); 1481 } 1482 if (chainCreated) { 1483 removeEffectChain_l(chain); 1484 } 1485 handle.clear(); 1486 } 1487 1488 *status = lStatus; 1489 return handle; 1490} 1491 1492sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect(audio_session_t sessionId, 1493 int effectId) 1494{ 1495 Mutex::Autolock _l(mLock); 1496 return getEffect_l(sessionId, effectId); 1497} 1498 1499sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(audio_session_t sessionId, 1500 int effectId) 1501{ 1502 sp<EffectChain> chain = getEffectChain_l(sessionId); 1503 return chain != 0 ? chain->getEffectFromId_l(effectId) : 0; 1504} 1505 1506// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 1507// PlaybackThread::mLock held 1508status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 1509{ 1510 // check for existing effect chain with the requested audio session 1511 audio_session_t sessionId = effect->sessionId(); 1512 sp<EffectChain> chain = getEffectChain_l(sessionId); 1513 bool chainCreated = false; 1514 1515 ALOGD_IF((mType == OFFLOAD) && !effect->isOffloadable(), 1516 "addEffect_l() on offloaded thread %p: effect %s does not support offload flags %x", 1517 this, effect->desc().name, effect->desc().flags); 1518 1519 if (chain == 0) { 1520 // create a new chain for this session 1521 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 1522 chain = new EffectChain(this, sessionId); 1523 addEffectChain_l(chain); 1524 chain->setStrategy(getStrategyForSession_l(sessionId)); 1525 chainCreated = true; 1526 } 1527 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 1528 1529 if (chain->getEffectFromId_l(effect->id()) != 0) { 1530 ALOGW("addEffect_l() %p effect %s already present in chain %p", 1531 this, effect->desc().name, chain.get()); 1532 return BAD_VALUE; 1533 } 1534 1535 effect->setOffloaded(mType == OFFLOAD, mId); 1536 1537 status_t status = chain->addEffect_l(effect); 1538 if (status != NO_ERROR) { 1539 if (chainCreated) { 1540 removeEffectChain_l(chain); 1541 } 1542 return status; 1543 } 1544 1545 effect->setDevice(mOutDevice); 1546 effect->setDevice(mInDevice); 1547 effect->setMode(mAudioFlinger->getMode()); 1548 effect->setAudioSource(mAudioSource); 1549 return NO_ERROR; 1550} 1551 1552void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 1553 1554 ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 1555 effect_descriptor_t desc = effect->desc(); 1556 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 1557 detachAuxEffect_l(effect->id()); 1558 } 1559 1560 sp<EffectChain> chain = effect->chain().promote(); 1561 if (chain != 0) { 1562 // remove effect chain if removing last effect 1563 if (chain->removeEffect_l(effect) == 0) { 1564 removeEffectChain_l(chain); 1565 } 1566 } else { 1567 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 1568 } 1569} 1570 1571void AudioFlinger::ThreadBase::lockEffectChains_l( 1572 Vector< sp<AudioFlinger::EffectChain> >& effectChains) 1573{ 1574 effectChains = mEffectChains; 1575 for (size_t i = 0; i < mEffectChains.size(); i++) { 1576 mEffectChains[i]->lock(); 1577 } 1578} 1579 1580void AudioFlinger::ThreadBase::unlockEffectChains( 1581 const Vector< sp<AudioFlinger::EffectChain> >& effectChains) 1582{ 1583 for (size_t i = 0; i < effectChains.size(); i++) { 1584 effectChains[i]->unlock(); 1585 } 1586} 1587 1588sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(audio_session_t sessionId) 1589{ 1590 Mutex::Autolock _l(mLock); 1591 return getEffectChain_l(sessionId); 1592} 1593 1594sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(audio_session_t sessionId) 1595 const 1596{ 1597 size_t size = mEffectChains.size(); 1598 for (size_t i = 0; i < size; i++) { 1599 if (mEffectChains[i]->sessionId() == sessionId) { 1600 return mEffectChains[i]; 1601 } 1602 } 1603 return 0; 1604} 1605 1606void AudioFlinger::ThreadBase::setMode(audio_mode_t mode) 1607{ 1608 Mutex::Autolock _l(mLock); 1609 size_t size = mEffectChains.size(); 1610 for (size_t i = 0; i < size; i++) { 1611 mEffectChains[i]->setMode_l(mode); 1612 } 1613} 1614 1615void AudioFlinger::ThreadBase::getAudioPortConfig(struct audio_port_config *config) 1616{ 1617 config->type = AUDIO_PORT_TYPE_MIX; 1618 config->ext.mix.handle = mId; 1619 config->sample_rate = mSampleRate; 1620 config->format = mFormat; 1621 config->channel_mask = mChannelMask; 1622 config->config_mask = AUDIO_PORT_CONFIG_SAMPLE_RATE|AUDIO_PORT_CONFIG_CHANNEL_MASK| 1623 AUDIO_PORT_CONFIG_FORMAT; 1624} 1625 1626void AudioFlinger::ThreadBase::systemReady() 1627{ 1628 Mutex::Autolock _l(mLock); 1629 if (mSystemReady) { 1630 return; 1631 } 1632 mSystemReady = true; 1633 1634 for (size_t i = 0; i < mPendingConfigEvents.size(); i++) { 1635 sendConfigEvent_l(mPendingConfigEvents.editItemAt(i)); 1636 } 1637 mPendingConfigEvents.clear(); 1638} 1639 1640 1641// ---------------------------------------------------------------------------- 1642// Playback 1643// ---------------------------------------------------------------------------- 1644 1645AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, 1646 AudioStreamOut* output, 1647 audio_io_handle_t id, 1648 audio_devices_t device, 1649 type_t type, 1650 bool systemReady) 1651 : ThreadBase(audioFlinger, id, device, AUDIO_DEVICE_NONE, type, systemReady), 1652 mNormalFrameCount(0), mSinkBuffer(NULL), 1653 mMixerBufferEnabled(AudioFlinger::kEnableExtendedPrecision), 1654 mMixerBuffer(NULL), 1655 mMixerBufferSize(0), 1656 mMixerBufferFormat(AUDIO_FORMAT_INVALID), 1657 mMixerBufferValid(false), 1658 mEffectBufferEnabled(AudioFlinger::kEnableExtendedPrecision), 1659 mEffectBuffer(NULL), 1660 mEffectBufferSize(0), 1661 mEffectBufferFormat(AUDIO_FORMAT_INVALID), 1662 mEffectBufferValid(false), 1663 mSuspended(0), mBytesWritten(0), 1664 mFramesWritten(0), 1665 mSuspendedFrames(0), 1666 mActiveTracksGeneration(0), 1667 // mStreamTypes[] initialized in constructor body 1668 mOutput(output), 1669 mLastWriteTime(-1), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false), 1670 mMixerStatus(MIXER_IDLE), 1671 mMixerStatusIgnoringFastTracks(MIXER_IDLE), 1672 mStandbyDelayNs(AudioFlinger::mStandbyTimeInNsecs), 1673 mBytesRemaining(0), 1674 mCurrentWriteLength(0), 1675 mUseAsyncWrite(false), 1676 mWriteAckSequence(0), 1677 mDrainSequence(0), 1678 mSignalPending(false), 1679 mScreenState(AudioFlinger::mScreenState), 1680 // index 0 is reserved for normal mixer's submix 1681 mFastTrackAvailMask(((1 << FastMixerState::sMaxFastTracks) - 1) & ~1), 1682 mHwSupportsPause(false), mHwPaused(false), mFlushPending(false) 1683{ 1684 snprintf(mThreadName, kThreadNameLength, "AudioOut_%X", id); 1685 mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mThreadName); 1686 1687 // Assumes constructor is called by AudioFlinger with it's mLock held, but 1688 // it would be safer to explicitly pass initial masterVolume/masterMute as 1689 // parameter. 1690 // 1691 // If the HAL we are using has support for master volume or master mute, 1692 // then do not attenuate or mute during mixing (just leave the volume at 1.0 1693 // and the mute set to false). 1694 mMasterVolume = audioFlinger->masterVolume_l(); 1695 mMasterMute = audioFlinger->masterMute_l(); 1696 if (mOutput && mOutput->audioHwDev) { 1697 if (mOutput->audioHwDev->canSetMasterVolume()) { 1698 mMasterVolume = 1.0; 1699 } 1700 1701 if (mOutput->audioHwDev->canSetMasterMute()) { 1702 mMasterMute = false; 1703 } 1704 } 1705 1706 readOutputParameters_l(); 1707 1708 // ++ operator does not compile 1709 for (audio_stream_type_t stream = AUDIO_STREAM_MIN; stream < AUDIO_STREAM_CNT; 1710 stream = (audio_stream_type_t) (stream + 1)) { 1711 mStreamTypes[stream].volume = mAudioFlinger->streamVolume_l(stream); 1712 mStreamTypes[stream].mute = mAudioFlinger->streamMute_l(stream); 1713 } 1714} 1715 1716AudioFlinger::PlaybackThread::~PlaybackThread() 1717{ 1718 mAudioFlinger->unregisterWriter(mNBLogWriter); 1719 free(mSinkBuffer); 1720 free(mMixerBuffer); 1721 free(mEffectBuffer); 1722} 1723 1724void AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) 1725{ 1726 dumpInternals(fd, args); 1727 dumpTracks(fd, args); 1728 dumpEffectChains(fd, args); 1729} 1730 1731void AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args __unused) 1732{ 1733 const size_t SIZE = 256; 1734 char buffer[SIZE]; 1735 String8 result; 1736 1737 result.appendFormat(" Stream volumes in dB: "); 1738 for (int i = 0; i < AUDIO_STREAM_CNT; ++i) { 1739 const stream_type_t *st = &mStreamTypes[i]; 1740 if (i > 0) { 1741 result.appendFormat(", "); 1742 } 1743 result.appendFormat("%d:%.2g", i, 20.0 * log10(st->volume)); 1744 if (st->mute) { 1745 result.append("M"); 1746 } 1747 } 1748 result.append("\n"); 1749 write(fd, result.string(), result.length()); 1750 result.clear(); 1751 1752 // These values are "raw"; they will wrap around. See prepareTracks_l() for a better way. 1753 FastTrackUnderruns underruns = getFastTrackUnderruns(0); 1754 dprintf(fd, " Normal mixer raw underrun counters: partial=%u empty=%u\n", 1755 underruns.mBitFields.mPartial, underruns.mBitFields.mEmpty); 1756 1757 size_t numtracks = mTracks.size(); 1758 size_t numactive = mActiveTracks.size(); 1759 dprintf(fd, " %zu Tracks", numtracks); 1760 size_t numactiveseen = 0; 1761 if (numtracks) { 1762 dprintf(fd, " of which %zu are active\n", numactive); 1763 Track::appendDumpHeader(result); 1764 for (size_t i = 0; i < numtracks; ++i) { 1765 sp<Track> track = mTracks[i]; 1766 if (track != 0) { 1767 bool active = mActiveTracks.indexOf(track) >= 0; 1768 if (active) { 1769 numactiveseen++; 1770 } 1771 track->dump(buffer, SIZE, active); 1772 result.append(buffer); 1773 } 1774 } 1775 } else { 1776 result.append("\n"); 1777 } 1778 if (numactiveseen != numactive) { 1779 // some tracks in the active list were not in the tracks list 1780 snprintf(buffer, SIZE, " The following tracks are in the active list but" 1781 " not in the track list\n"); 1782 result.append(buffer); 1783 Track::appendDumpHeader(result); 1784 for (size_t i = 0; i < numactive; ++i) { 1785 sp<Track> track = mActiveTracks[i].promote(); 1786 if (track != 0 && mTracks.indexOf(track) < 0) { 1787 track->dump(buffer, SIZE, true); 1788 result.append(buffer); 1789 } 1790 } 1791 } 1792 1793 write(fd, result.string(), result.size()); 1794} 1795 1796void AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 1797{ 1798 dprintf(fd, "\nOutput thread %p type %d (%s):\n", this, type(), threadTypeToString(type())); 1799 1800 dumpBase(fd, args); 1801 1802 dprintf(fd, " Normal frame count: %zu\n", mNormalFrameCount); 1803 dprintf(fd, " Last write occurred (msecs): %llu\n", 1804 (unsigned long long) ns2ms(systemTime() - mLastWriteTime)); 1805 dprintf(fd, " Total writes: %d\n", mNumWrites); 1806 dprintf(fd, " Delayed writes: %d\n", mNumDelayedWrites); 1807 dprintf(fd, " Blocked in write: %s\n", mInWrite ? "yes" : "no"); 1808 dprintf(fd, " Suspend count: %d\n", mSuspended); 1809 dprintf(fd, " Sink buffer : %p\n", mSinkBuffer); 1810 dprintf(fd, " Mixer buffer: %p\n", mMixerBuffer); 1811 dprintf(fd, " Effect buffer: %p\n", mEffectBuffer); 1812 dprintf(fd, " Fast track availMask=%#x\n", mFastTrackAvailMask); 1813 dprintf(fd, " Standby delay ns=%lld\n", (long long)mStandbyDelayNs); 1814 AudioStreamOut *output = mOutput; 1815 audio_output_flags_t flags = output != NULL ? output->flags : AUDIO_OUTPUT_FLAG_NONE; 1816 String8 flagsAsString = outputFlagsToString(flags); 1817 dprintf(fd, " AudioStreamOut: %p flags %#x (%s)\n", output, flags, flagsAsString.string()); 1818} 1819 1820// Thread virtuals 1821 1822void AudioFlinger::PlaybackThread::onFirstRef() 1823{ 1824 run(mThreadName, ANDROID_PRIORITY_URGENT_AUDIO); 1825} 1826 1827// ThreadBase virtuals 1828void AudioFlinger::PlaybackThread::preExit() 1829{ 1830 ALOGV(" preExit()"); 1831 // FIXME this is using hard-coded strings but in the future, this functionality will be 1832 // converted to use audio HAL extensions required to support tunneling 1833 mOutput->stream->common.set_parameters(&mOutput->stream->common, "exiting=1"); 1834} 1835 1836// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 1837sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 1838 const sp<AudioFlinger::Client>& client, 1839 audio_stream_type_t streamType, 1840 uint32_t sampleRate, 1841 audio_format_t format, 1842 audio_channel_mask_t channelMask, 1843 size_t *pFrameCount, 1844 const sp<IMemory>& sharedBuffer, 1845 audio_session_t sessionId, 1846 audio_output_flags_t *flags, 1847 pid_t tid, 1848 int uid, 1849 status_t *status) 1850{ 1851 size_t frameCount = *pFrameCount; 1852 sp<Track> track; 1853 status_t lStatus; 1854 audio_output_flags_t outputFlags = mOutput->flags; 1855 1856 // special case for FAST flag considered OK if fast mixer is present 1857 if (hasFastMixer()) { 1858 outputFlags = (audio_output_flags_t)(outputFlags | AUDIO_OUTPUT_FLAG_FAST); 1859 } 1860 1861 // Check if requested flags are compatible with output stream flags 1862 if ((*flags & outputFlags) != *flags) { 1863 ALOGW("createTrack_l(): mismatch between requested flags (%08x) and output flags (%08x)", 1864 *flags, outputFlags); 1865 *flags = (audio_output_flags_t)(*flags & outputFlags); 1866 } 1867 1868 // client expresses a preference for FAST, but we get the final say 1869 if (*flags & AUDIO_OUTPUT_FLAG_FAST) { 1870 if ( 1871 // PCM data 1872 audio_is_linear_pcm(format) && 1873 // TODO: extract as a data library function that checks that a computationally 1874 // expensive downmixer is not required: isFastOutputChannelConversion() 1875 (channelMask == mChannelMask || 1876 mChannelMask != AUDIO_CHANNEL_OUT_STEREO || 1877 (channelMask == AUDIO_CHANNEL_OUT_MONO 1878 /* && mChannelMask == AUDIO_CHANNEL_OUT_STEREO */)) && 1879 // hardware sample rate 1880 (sampleRate == mSampleRate) && 1881 // normal mixer has an associated fast mixer 1882 hasFastMixer() && 1883 // there are sufficient fast track slots available 1884 (mFastTrackAvailMask != 0) 1885 // FIXME test that MixerThread for this fast track has a capable output HAL 1886 // FIXME add a permission test also? 1887 ) { 1888 // static tracks can have any nonzero framecount, streaming tracks check against minimum. 1889 if (sharedBuffer == 0) { 1890 // read the fast track multiplier property the first time it is needed 1891 int ok = pthread_once(&sFastTrackMultiplierOnce, sFastTrackMultiplierInit); 1892 if (ok != 0) { 1893 ALOGE("%s pthread_once failed: %d", __func__, ok); 1894 } 1895 frameCount = max(frameCount, mFrameCount * sFastTrackMultiplier); // incl framecount 0 1896 } 1897 1898 // check compatibility with audio effects. 1899 { // scope for mLock 1900 Mutex::Autolock _l(mLock); 1901 // do not accept RAW flag if post processing are present. Note that post processing on 1902 // a fast mixer are necessarily hardware 1903 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_STAGE); 1904 if (chain != 0) { 1905 ALOGV_IF((*flags & AUDIO_OUTPUT_FLAG_RAW) != 0, 1906 "AUDIO_OUTPUT_FLAG_RAW denied: post processing effect present"); 1907 *flags = (audio_output_flags_t)(*flags & ~AUDIO_OUTPUT_FLAG_RAW); 1908 } 1909 // Do not accept FAST flag if software global effects are present 1910 chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 1911 if (chain != 0) { 1912 ALOGV_IF((*flags & AUDIO_OUTPUT_FLAG_RAW) != 0, 1913 "AUDIO_OUTPUT_FLAG_RAW denied: global effect present"); 1914 *flags = (audio_output_flags_t)(*flags & ~AUDIO_OUTPUT_FLAG_RAW); 1915 if (chain->hasSoftwareEffect()) { 1916 ALOGV("AUDIO_OUTPUT_FLAG_FAST denied: software global effect present"); 1917 *flags = (audio_output_flags_t)(*flags & ~AUDIO_OUTPUT_FLAG_FAST); 1918 } 1919 } 1920 // Do not accept FAST flag if the session has software effects 1921 chain = getEffectChain_l(sessionId); 1922 if (chain != 0) { 1923 ALOGV_IF((*flags & AUDIO_OUTPUT_FLAG_RAW) != 0, 1924 "AUDIO_OUTPUT_FLAG_RAW denied: effect present on session"); 1925 *flags = (audio_output_flags_t)(*flags & ~AUDIO_OUTPUT_FLAG_RAW); 1926 if (chain->hasSoftwareEffect()) { 1927 ALOGV("AUDIO_OUTPUT_FLAG_FAST denied: software effect present on session"); 1928 *flags = (audio_output_flags_t)(*flags & ~AUDIO_OUTPUT_FLAG_FAST); 1929 } 1930 } 1931 } 1932 ALOGV_IF((*flags & AUDIO_OUTPUT_FLAG_FAST) != 0, 1933 "AUDIO_OUTPUT_FLAG_FAST accepted: frameCount=%zu mFrameCount=%zu", 1934 frameCount, mFrameCount); 1935 } else { 1936 ALOGV("AUDIO_OUTPUT_FLAG_FAST denied: sharedBuffer=%p frameCount=%zu " 1937 "mFrameCount=%zu format=%#x mFormat=%#x isLinear=%d channelMask=%#x " 1938 "sampleRate=%u mSampleRate=%u " 1939 "hasFastMixer=%d tid=%d fastTrackAvailMask=%#x", 1940 sharedBuffer.get(), frameCount, mFrameCount, format, mFormat, 1941 audio_is_linear_pcm(format), 1942 channelMask, sampleRate, mSampleRate, hasFastMixer(), tid, mFastTrackAvailMask); 1943 *flags = (audio_output_flags_t)(*flags & ~AUDIO_OUTPUT_FLAG_FAST); 1944 } 1945 } 1946 // For normal PCM streaming tracks, update minimum frame count. 1947 // For compatibility with AudioTrack calculation, buffer depth is forced 1948 // to be at least 2 x the normal mixer frame count and cover audio hardware latency. 1949 // This is probably too conservative, but legacy application code may depend on it. 1950 // If you change this calculation, also review the start threshold which is related. 1951 if (!(*flags & AUDIO_OUTPUT_FLAG_FAST) 1952 && audio_has_proportional_frames(format) && sharedBuffer == 0) { 1953 // this must match AudioTrack.cpp calculateMinFrameCount(). 1954 // TODO: Move to a common library 1955 uint32_t latencyMs = mOutput->stream->get_latency(mOutput->stream); 1956 uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate); 1957 if (minBufCount < 2) { 1958 minBufCount = 2; 1959 } 1960 // For normal mixing tracks, if speed is > 1.0f (normal), AudioTrack 1961 // or the client should compute and pass in a larger buffer request. 1962 size_t minFrameCount = 1963 minBufCount * sourceFramesNeededWithTimestretch( 1964 sampleRate, mNormalFrameCount, 1965 mSampleRate, AUDIO_TIMESTRETCH_SPEED_NORMAL /*speed*/); 1966 if (frameCount < minFrameCount) { // including frameCount == 0 1967 frameCount = minFrameCount; 1968 } 1969 } 1970 *pFrameCount = frameCount; 1971 1972 switch (mType) { 1973 1974 case DIRECT: 1975 if (audio_is_linear_pcm(format)) { // TODO maybe use audio_has_proportional_frames()? 1976 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1977 ALOGE("createTrack_l() Bad parameter: sampleRate %u format %#x, channelMask 0x%08x " 1978 "for output %p with format %#x", 1979 sampleRate, format, channelMask, mOutput, mFormat); 1980 lStatus = BAD_VALUE; 1981 goto Exit; 1982 } 1983 } 1984 break; 1985 1986 case OFFLOAD: 1987 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1988 ALOGE("createTrack_l() Bad parameter: sampleRate %d format %#x, channelMask 0x%08x \"" 1989 "for output %p with format %#x", 1990 sampleRate, format, channelMask, mOutput, mFormat); 1991 lStatus = BAD_VALUE; 1992 goto Exit; 1993 } 1994 break; 1995 1996 default: 1997 if (!audio_is_linear_pcm(format)) { 1998 ALOGE("createTrack_l() Bad parameter: format %#x \"" 1999 "for output %p with format %#x", 2000 format, mOutput, mFormat); 2001 lStatus = BAD_VALUE; 2002 goto Exit; 2003 } 2004 if (sampleRate > mSampleRate * AUDIO_RESAMPLER_DOWN_RATIO_MAX) { 2005 ALOGE("Sample rate out of range: %u mSampleRate %u", sampleRate, mSampleRate); 2006 lStatus = BAD_VALUE; 2007 goto Exit; 2008 } 2009 break; 2010 2011 } 2012 2013 lStatus = initCheck(); 2014 if (lStatus != NO_ERROR) { 2015 ALOGE("createTrack_l() audio driver not initialized"); 2016 goto Exit; 2017 } 2018 2019 { // scope for mLock 2020 Mutex::Autolock _l(mLock); 2021 2022 // all tracks in same audio session must share the same routing strategy otherwise 2023 // conflicts will happen when tracks are moved from one output to another by audio policy 2024 // manager 2025 uint32_t strategy = AudioSystem::getStrategyForStream(streamType); 2026 for (size_t i = 0; i < mTracks.size(); ++i) { 2027 sp<Track> t = mTracks[i]; 2028 if (t != 0 && t->isExternalTrack()) { 2029 uint32_t actual = AudioSystem::getStrategyForStream(t->streamType()); 2030 if (sessionId == t->sessionId() && strategy != actual) { 2031 ALOGE("createTrack_l() mismatched strategy; expected %u but found %u", 2032 strategy, actual); 2033 lStatus = BAD_VALUE; 2034 goto Exit; 2035 } 2036 } 2037 } 2038 2039 track = new Track(this, client, streamType, sampleRate, format, 2040 channelMask, frameCount, NULL, sharedBuffer, 2041 sessionId, uid, *flags, TrackBase::TYPE_DEFAULT); 2042 2043 lStatus = track != 0 ? track->initCheck() : (status_t) NO_MEMORY; 2044 if (lStatus != NO_ERROR) { 2045 ALOGE("createTrack_l() initCheck failed %d; no control block?", lStatus); 2046 // track must be cleared from the caller as the caller has the AF lock 2047 goto Exit; 2048 } 2049 mTracks.add(track); 2050 2051 sp<EffectChain> chain = getEffectChain_l(sessionId); 2052 if (chain != 0) { 2053 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 2054 track->setMainBuffer(chain->inBuffer()); 2055 chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType())); 2056 chain->incTrackCnt(); 2057 } 2058 2059 if ((*flags & AUDIO_OUTPUT_FLAG_FAST) && (tid != -1)) { 2060 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 2061 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 2062 // so ask activity manager to do this on our behalf 2063 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 2064 } 2065 } 2066 2067 lStatus = NO_ERROR; 2068 2069Exit: 2070 *status = lStatus; 2071 return track; 2072} 2073 2074uint32_t AudioFlinger::PlaybackThread::correctLatency_l(uint32_t latency) const 2075{ 2076 return latency; 2077} 2078 2079uint32_t AudioFlinger::PlaybackThread::latency() const 2080{ 2081 Mutex::Autolock _l(mLock); 2082 return latency_l(); 2083} 2084uint32_t AudioFlinger::PlaybackThread::latency_l() const 2085{ 2086 if (initCheck() == NO_ERROR) { 2087 return correctLatency_l(mOutput->stream->get_latency(mOutput->stream)); 2088 } else { 2089 return 0; 2090 } 2091} 2092 2093void AudioFlinger::PlaybackThread::setMasterVolume(float value) 2094{ 2095 Mutex::Autolock _l(mLock); 2096 // Don't apply master volume in SW if our HAL can do it for us. 2097 if (mOutput && mOutput->audioHwDev && 2098 mOutput->audioHwDev->canSetMasterVolume()) { 2099 mMasterVolume = 1.0; 2100 } else { 2101 mMasterVolume = value; 2102 } 2103} 2104 2105void AudioFlinger::PlaybackThread::setMasterMute(bool muted) 2106{ 2107 Mutex::Autolock _l(mLock); 2108 // Don't apply master mute in SW if our HAL can do it for us. 2109 if (mOutput && mOutput->audioHwDev && 2110 mOutput->audioHwDev->canSetMasterMute()) { 2111 mMasterMute = false; 2112 } else { 2113 mMasterMute = muted; 2114 } 2115} 2116 2117void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) 2118{ 2119 Mutex::Autolock _l(mLock); 2120 mStreamTypes[stream].volume = value; 2121 broadcast_l(); 2122} 2123 2124void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) 2125{ 2126 Mutex::Autolock _l(mLock); 2127 mStreamTypes[stream].mute = muted; 2128 broadcast_l(); 2129} 2130 2131float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const 2132{ 2133 Mutex::Autolock _l(mLock); 2134 return mStreamTypes[stream].volume; 2135} 2136 2137// addTrack_l() must be called with ThreadBase::mLock held 2138status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 2139{ 2140 status_t status = ALREADY_EXISTS; 2141 2142 if (mActiveTracks.indexOf(track) < 0) { 2143 // the track is newly added, make sure it fills up all its 2144 // buffers before playing. This is to ensure the client will 2145 // effectively get the latency it requested. 2146 if (track->isExternalTrack()) { 2147 TrackBase::track_state state = track->mState; 2148 mLock.unlock(); 2149 status = AudioSystem::startOutput(mId, track->streamType(), 2150 track->sessionId()); 2151 mLock.lock(); 2152 // abort track was stopped/paused while we released the lock 2153 if (state != track->mState) { 2154 if (status == NO_ERROR) { 2155 mLock.unlock(); 2156 AudioSystem::stopOutput(mId, track->streamType(), 2157 track->sessionId()); 2158 mLock.lock(); 2159 } 2160 return INVALID_OPERATION; 2161 } 2162 // abort if start is rejected by audio policy manager 2163 if (status != NO_ERROR) { 2164 return PERMISSION_DENIED; 2165 } 2166#ifdef ADD_BATTERY_DATA 2167 // to track the speaker usage 2168 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 2169#endif 2170 } 2171 2172 // set retry count for buffer fill 2173 if (track->isOffloaded()) { 2174 if (track->isStopping_1()) { 2175 track->mRetryCount = kMaxTrackStopRetriesOffload; 2176 } else { 2177 track->mRetryCount = kMaxTrackStartupRetriesOffload; 2178 } 2179 track->mFillingUpStatus = mStandby ? Track::FS_FILLING : Track::FS_FILLED; 2180 } else { 2181 track->mRetryCount = kMaxTrackStartupRetries; 2182 track->mFillingUpStatus = 2183 track->sharedBuffer() != 0 ? Track::FS_FILLED : Track::FS_FILLING; 2184 } 2185 2186 track->mResetDone = false; 2187 track->mPresentationCompleteFrames = 0; 2188 mActiveTracks.add(track); 2189 mWakeLockUids.add(track->uid()); 2190 mActiveTracksGeneration++; 2191 mLatestActiveTrack = track; 2192 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 2193 if (chain != 0) { 2194 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), 2195 track->sessionId()); 2196 chain->incActiveTrackCnt(); 2197 } 2198 2199 status = NO_ERROR; 2200 } 2201 2202 onAddNewTrack_l(); 2203 return status; 2204} 2205 2206bool AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 2207{ 2208 track->terminate(); 2209 // active tracks are removed by threadLoop() 2210 bool trackActive = (mActiveTracks.indexOf(track) >= 0); 2211 track->mState = TrackBase::STOPPED; 2212 if (!trackActive) { 2213 removeTrack_l(track); 2214 } else if (track->isFastTrack() || track->isOffloaded() || track->isDirect()) { 2215 track->mState = TrackBase::STOPPING_1; 2216 } 2217 2218 return trackActive; 2219} 2220 2221void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 2222{ 2223 track->triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 2224 mTracks.remove(track); 2225 deleteTrackName_l(track->name()); 2226 // redundant as track is about to be destroyed, for dumpsys only 2227 track->mName = -1; 2228 if (track->isFastTrack()) { 2229 int index = track->mFastIndex; 2230 ALOG_ASSERT(0 < index && index < (int)FastMixerState::sMaxFastTracks); 2231 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << index))); 2232 mFastTrackAvailMask |= 1 << index; 2233 // redundant as track is about to be destroyed, for dumpsys only 2234 track->mFastIndex = -1; 2235 } 2236 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 2237 if (chain != 0) { 2238 chain->decTrackCnt(); 2239 } 2240} 2241 2242void AudioFlinger::PlaybackThread::broadcast_l() 2243{ 2244 // Thread could be blocked waiting for async 2245 // so signal it to handle state changes immediately 2246 // If threadLoop is currently unlocked a signal of mWaitWorkCV will 2247 // be lost so we also flag to prevent it blocking on mWaitWorkCV 2248 mSignalPending = true; 2249 mWaitWorkCV.broadcast(); 2250} 2251 2252String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 2253{ 2254 Mutex::Autolock _l(mLock); 2255 if (initCheck() != NO_ERROR) { 2256 return String8(); 2257 } 2258 2259 char *s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 2260 const String8 out_s8(s); 2261 free(s); 2262 return out_s8; 2263} 2264 2265void AudioFlinger::PlaybackThread::ioConfigChanged(audio_io_config_event event, pid_t pid) { 2266 sp<AudioIoDescriptor> desc = new AudioIoDescriptor(); 2267 ALOGV("PlaybackThread::ioConfigChanged, thread %p, event %d", this, event); 2268 2269 desc->mIoHandle = mId; 2270 2271 switch (event) { 2272 case AUDIO_OUTPUT_OPENED: 2273 case AUDIO_OUTPUT_CONFIG_CHANGED: 2274 desc->mPatch = mPatch; 2275 desc->mChannelMask = mChannelMask; 2276 desc->mSamplingRate = mSampleRate; 2277 desc->mFormat = mFormat; 2278 desc->mFrameCount = mNormalFrameCount; // FIXME see 2279 // AudioFlinger::frameCount(audio_io_handle_t) 2280 desc->mFrameCountHAL = mFrameCount; 2281 desc->mLatency = latency_l(); 2282 break; 2283 2284 case AUDIO_OUTPUT_CLOSED: 2285 default: 2286 break; 2287 } 2288 mAudioFlinger->ioConfigChanged(event, desc, pid); 2289} 2290 2291void AudioFlinger::PlaybackThread::writeCallback() 2292{ 2293 ALOG_ASSERT(mCallbackThread != 0); 2294 mCallbackThread->resetWriteBlocked(); 2295} 2296 2297void AudioFlinger::PlaybackThread::drainCallback() 2298{ 2299 ALOG_ASSERT(mCallbackThread != 0); 2300 mCallbackThread->resetDraining(); 2301} 2302 2303void AudioFlinger::PlaybackThread::errorCallback() 2304{ 2305 ALOG_ASSERT(mCallbackThread != 0); 2306 mCallbackThread->setAsyncError(); 2307} 2308 2309void AudioFlinger::PlaybackThread::resetWriteBlocked(uint32_t sequence) 2310{ 2311 Mutex::Autolock _l(mLock); 2312 // reject out of sequence requests 2313 if ((mWriteAckSequence & 1) && (sequence == mWriteAckSequence)) { 2314 mWriteAckSequence &= ~1; 2315 mWaitWorkCV.signal(); 2316 } 2317} 2318 2319void AudioFlinger::PlaybackThread::resetDraining(uint32_t sequence) 2320{ 2321 Mutex::Autolock _l(mLock); 2322 // reject out of sequence requests 2323 if ((mDrainSequence & 1) && (sequence == mDrainSequence)) { 2324 mDrainSequence &= ~1; 2325 mWaitWorkCV.signal(); 2326 } 2327} 2328 2329// static 2330int AudioFlinger::PlaybackThread::asyncCallback(stream_callback_event_t event, 2331 void *param __unused, 2332 void *cookie) 2333{ 2334 AudioFlinger::PlaybackThread *me = (AudioFlinger::PlaybackThread *)cookie; 2335 ALOGV("asyncCallback() event %d", event); 2336 switch (event) { 2337 case STREAM_CBK_EVENT_WRITE_READY: 2338 me->writeCallback(); 2339 break; 2340 case STREAM_CBK_EVENT_DRAIN_READY: 2341 me->drainCallback(); 2342 break; 2343 case STREAM_CBK_EVENT_ERROR: 2344 me->errorCallback(); 2345 break; 2346 default: 2347 ALOGW("asyncCallback() unknown event %d", event); 2348 break; 2349 } 2350 return 0; 2351} 2352 2353void AudioFlinger::PlaybackThread::readOutputParameters_l() 2354{ 2355 // unfortunately we have no way of recovering from errors here, hence the LOG_ALWAYS_FATAL 2356 mSampleRate = mOutput->getSampleRate(); 2357 mChannelMask = mOutput->getChannelMask(); 2358 if (!audio_is_output_channel(mChannelMask)) { 2359 LOG_ALWAYS_FATAL("HAL channel mask %#x not valid for output", mChannelMask); 2360 } 2361 if ((mType == MIXER || mType == DUPLICATING) 2362 && !isValidPcmSinkChannelMask(mChannelMask)) { 2363 LOG_ALWAYS_FATAL("HAL channel mask %#x not supported for mixed output", 2364 mChannelMask); 2365 } 2366 mChannelCount = audio_channel_count_from_out_mask(mChannelMask); 2367 2368 // Get actual HAL format. 2369 mHALFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 2370 // Get format from the shim, which will be different than the HAL format 2371 // if playing compressed audio over HDMI passthrough. 2372 mFormat = mOutput->getFormat(); 2373 if (!audio_is_valid_format(mFormat)) { 2374 LOG_ALWAYS_FATAL("HAL format %#x not valid for output", mFormat); 2375 } 2376 if ((mType == MIXER || mType == DUPLICATING) 2377 && !isValidPcmSinkFormat(mFormat)) { 2378 LOG_FATAL("HAL format %#x not supported for mixed output", 2379 mFormat); 2380 } 2381 mFrameSize = mOutput->getFrameSize(); 2382 mBufferSize = mOutput->stream->common.get_buffer_size(&mOutput->stream->common); 2383 mFrameCount = mBufferSize / mFrameSize; 2384 if (mFrameCount & 15) { 2385 ALOGW("HAL output buffer size is %zu frames but AudioMixer requires multiples of 16 frames", 2386 mFrameCount); 2387 } 2388 2389 if ((mOutput->flags & AUDIO_OUTPUT_FLAG_NON_BLOCKING) && 2390 (mOutput->stream->set_callback != NULL)) { 2391 if (mOutput->stream->set_callback(mOutput->stream, 2392 AudioFlinger::PlaybackThread::asyncCallback, this) == 0) { 2393 mUseAsyncWrite = true; 2394 mCallbackThread = new AudioFlinger::AsyncCallbackThread(this); 2395 } 2396 } 2397 2398 mHwSupportsPause = false; 2399 if (mOutput->flags & AUDIO_OUTPUT_FLAG_DIRECT) { 2400 if (mOutput->stream->pause != NULL) { 2401 if (mOutput->stream->resume != NULL) { 2402 mHwSupportsPause = true; 2403 } else { 2404 ALOGW("direct output implements pause but not resume"); 2405 } 2406 } else if (mOutput->stream->resume != NULL) { 2407 ALOGW("direct output implements resume but not pause"); 2408 } 2409 } 2410 if (!mHwSupportsPause && mOutput->flags & AUDIO_OUTPUT_FLAG_HW_AV_SYNC) { 2411 LOG_ALWAYS_FATAL("HW_AV_SYNC requested but HAL does not implement pause and resume"); 2412 } 2413 2414 if (mType == DUPLICATING && mMixerBufferEnabled && mEffectBufferEnabled) { 2415 // For best precision, we use float instead of the associated output 2416 // device format (typically PCM 16 bit). 2417 2418 mFormat = AUDIO_FORMAT_PCM_FLOAT; 2419 mFrameSize = mChannelCount * audio_bytes_per_sample(mFormat); 2420 mBufferSize = mFrameSize * mFrameCount; 2421 2422 // TODO: We currently use the associated output device channel mask and sample rate. 2423 // (1) Perhaps use the ORed channel mask of all downstream MixerThreads 2424 // (if a valid mask) to avoid premature downmix. 2425 // (2) Perhaps use the maximum sample rate of all downstream MixerThreads 2426 // instead of the output device sample rate to avoid loss of high frequency information. 2427 // This may need to be updated as MixerThread/OutputTracks are added and not here. 2428 } 2429 2430 // Calculate size of normal sink buffer relative to the HAL output buffer size 2431 double multiplier = 1.0; 2432 if (mType == MIXER && (kUseFastMixer == FastMixer_Static || 2433 kUseFastMixer == FastMixer_Dynamic)) { 2434 size_t minNormalFrameCount = (kMinNormalSinkBufferSizeMs * mSampleRate) / 1000; 2435 size_t maxNormalFrameCount = (kMaxNormalSinkBufferSizeMs * mSampleRate) / 1000; 2436 2437 // round up minimum and round down maximum to nearest 16 frames to satisfy AudioMixer 2438 minNormalFrameCount = (minNormalFrameCount + 15) & ~15; 2439 maxNormalFrameCount = maxNormalFrameCount & ~15; 2440 if (maxNormalFrameCount < minNormalFrameCount) { 2441 maxNormalFrameCount = minNormalFrameCount; 2442 } 2443 multiplier = (double) minNormalFrameCount / (double) mFrameCount; 2444 if (multiplier <= 1.0) { 2445 multiplier = 1.0; 2446 } else if (multiplier <= 2.0) { 2447 if (2 * mFrameCount <= maxNormalFrameCount) { 2448 multiplier = 2.0; 2449 } else { 2450 multiplier = (double) maxNormalFrameCount / (double) mFrameCount; 2451 } 2452 } else { 2453 multiplier = floor(multiplier); 2454 } 2455 } 2456 mNormalFrameCount = multiplier * mFrameCount; 2457 // round up to nearest 16 frames to satisfy AudioMixer 2458 if (mType == MIXER || mType == DUPLICATING) { 2459 mNormalFrameCount = (mNormalFrameCount + 15) & ~15; 2460 } 2461 ALOGI("HAL output buffer size %zu frames, normal sink buffer size %zu frames", mFrameCount, 2462 mNormalFrameCount); 2463 2464 // Check if we want to throttle the processing to no more than 2x normal rate 2465 mThreadThrottle = property_get_bool("af.thread.throttle", true /* default_value */); 2466 mThreadThrottleTimeMs = 0; 2467 mThreadThrottleEndMs = 0; 2468 mHalfBufferMs = mNormalFrameCount * 1000 / (2 * mSampleRate); 2469 2470 // mSinkBuffer is the sink buffer. Size is always multiple-of-16 frames. 2471 // Originally this was int16_t[] array, need to remove legacy implications. 2472 free(mSinkBuffer); 2473 mSinkBuffer = NULL; 2474 // For sink buffer size, we use the frame size from the downstream sink to avoid problems 2475 // with non PCM formats for compressed music, e.g. AAC, and Offload threads. 2476 const size_t sinkBufferSize = mNormalFrameCount * mFrameSize; 2477 (void)posix_memalign(&mSinkBuffer, 32, sinkBufferSize); 2478 2479 // We resize the mMixerBuffer according to the requirements of the sink buffer which 2480 // drives the output. 2481 free(mMixerBuffer); 2482 mMixerBuffer = NULL; 2483 if (mMixerBufferEnabled) { 2484 mMixerBufferFormat = AUDIO_FORMAT_PCM_FLOAT; // also valid: AUDIO_FORMAT_PCM_16_BIT. 2485 mMixerBufferSize = mNormalFrameCount * mChannelCount 2486 * audio_bytes_per_sample(mMixerBufferFormat); 2487 (void)posix_memalign(&mMixerBuffer, 32, mMixerBufferSize); 2488 } 2489 free(mEffectBuffer); 2490 mEffectBuffer = NULL; 2491 if (mEffectBufferEnabled) { 2492 mEffectBufferFormat = AUDIO_FORMAT_PCM_16_BIT; // Note: Effects support 16b only 2493 mEffectBufferSize = mNormalFrameCount * mChannelCount 2494 * audio_bytes_per_sample(mEffectBufferFormat); 2495 (void)posix_memalign(&mEffectBuffer, 32, mEffectBufferSize); 2496 } 2497 2498 // force reconfiguration of effect chains and engines to take new buffer size and audio 2499 // parameters into account 2500 // Note that mLock is not held when readOutputParameters_l() is called from the constructor 2501 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 2502 // matter. 2503 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 2504 Vector< sp<EffectChain> > effectChains = mEffectChains; 2505 for (size_t i = 0; i < effectChains.size(); i ++) { 2506 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 2507 } 2508} 2509 2510 2511status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames) 2512{ 2513 if (halFrames == NULL || dspFrames == NULL) { 2514 return BAD_VALUE; 2515 } 2516 Mutex::Autolock _l(mLock); 2517 if (initCheck() != NO_ERROR) { 2518 return INVALID_OPERATION; 2519 } 2520 int64_t framesWritten = mBytesWritten / mFrameSize; 2521 *halFrames = framesWritten; 2522 2523 if (isSuspended()) { 2524 // return an estimation of rendered frames when the output is suspended 2525 size_t latencyFrames = (latency_l() * mSampleRate) / 1000; 2526 *dspFrames = (uint32_t) 2527 (framesWritten >= (int64_t)latencyFrames ? framesWritten - latencyFrames : 0); 2528 return NO_ERROR; 2529 } else { 2530 status_t status; 2531 uint32_t frames; 2532 status = mOutput->getRenderPosition(&frames); 2533 *dspFrames = (size_t)frames; 2534 return status; 2535 } 2536} 2537 2538// hasAudioSession_l() must be called with ThreadBase::mLock held 2539uint32_t AudioFlinger::PlaybackThread::hasAudioSession_l(audio_session_t sessionId) const 2540{ 2541 uint32_t result = 0; 2542 if (getEffectChain_l(sessionId) != 0) { 2543 result = EFFECT_SESSION; 2544 } 2545 2546 for (size_t i = 0; i < mTracks.size(); ++i) { 2547 sp<Track> track = mTracks[i]; 2548 if (sessionId == track->sessionId() && !track->isInvalid()) { 2549 result |= TRACK_SESSION; 2550 if (track->isFastTrack()) { 2551 result |= FAST_SESSION; 2552 } 2553 break; 2554 } 2555 } 2556 2557 return result; 2558} 2559 2560uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(audio_session_t sessionId) 2561{ 2562 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 2563 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 2564 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 2565 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 2566 } 2567 for (size_t i = 0; i < mTracks.size(); i++) { 2568 sp<Track> track = mTracks[i]; 2569 if (sessionId == track->sessionId() && !track->isInvalid()) { 2570 return AudioSystem::getStrategyForStream(track->streamType()); 2571 } 2572 } 2573 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 2574} 2575 2576 2577AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 2578{ 2579 Mutex::Autolock _l(mLock); 2580 return mOutput; 2581} 2582 2583AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 2584{ 2585 Mutex::Autolock _l(mLock); 2586 AudioStreamOut *output = mOutput; 2587 mOutput = NULL; 2588 // FIXME FastMixer might also have a raw ptr to mOutputSink; 2589 // must push a NULL and wait for ack 2590 mOutputSink.clear(); 2591 mPipeSink.clear(); 2592 mNormalSink.clear(); 2593 return output; 2594} 2595 2596// this method must always be called either with ThreadBase mLock held or inside the thread loop 2597audio_stream_t* AudioFlinger::PlaybackThread::stream() const 2598{ 2599 if (mOutput == NULL) { 2600 return NULL; 2601 } 2602 return &mOutput->stream->common; 2603} 2604 2605uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() const 2606{ 2607 return (uint32_t)((uint32_t)((mNormalFrameCount * 1000) / mSampleRate) * 1000); 2608} 2609 2610status_t AudioFlinger::PlaybackThread::setSyncEvent(const sp<SyncEvent>& event) 2611{ 2612 if (!isValidSyncEvent(event)) { 2613 return BAD_VALUE; 2614 } 2615 2616 Mutex::Autolock _l(mLock); 2617 2618 for (size_t i = 0; i < mTracks.size(); ++i) { 2619 sp<Track> track = mTracks[i]; 2620 if (event->triggerSession() == track->sessionId()) { 2621 (void) track->setSyncEvent(event); 2622 return NO_ERROR; 2623 } 2624 } 2625 2626 return NAME_NOT_FOUND; 2627} 2628 2629bool AudioFlinger::PlaybackThread::isValidSyncEvent(const sp<SyncEvent>& event) const 2630{ 2631 return event->type() == AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE; 2632} 2633 2634void AudioFlinger::PlaybackThread::threadLoop_removeTracks( 2635 const Vector< sp<Track> >& tracksToRemove) 2636{ 2637 size_t count = tracksToRemove.size(); 2638 if (count > 0) { 2639 for (size_t i = 0 ; i < count ; i++) { 2640 const sp<Track>& track = tracksToRemove.itemAt(i); 2641 if (track->isExternalTrack()) { 2642 AudioSystem::stopOutput(mId, track->streamType(), 2643 track->sessionId()); 2644#ifdef ADD_BATTERY_DATA 2645 // to track the speaker usage 2646 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 2647#endif 2648 if (track->isTerminated()) { 2649 AudioSystem::releaseOutput(mId, track->streamType(), 2650 track->sessionId()); 2651 } 2652 } 2653 } 2654 } 2655} 2656 2657void AudioFlinger::PlaybackThread::checkSilentMode_l() 2658{ 2659 if (!mMasterMute) { 2660 char value[PROPERTY_VALUE_MAX]; 2661 if (mOutDevice == AUDIO_DEVICE_OUT_REMOTE_SUBMIX) { 2662 ALOGD("ro.audio.silent will be ignored for threads on AUDIO_DEVICE_OUT_REMOTE_SUBMIX"); 2663 return; 2664 } 2665 if (property_get("ro.audio.silent", value, "0") > 0) { 2666 char *endptr; 2667 unsigned long ul = strtoul(value, &endptr, 0); 2668 if (*endptr == '\0' && ul != 0) { 2669 ALOGD("Silence is golden"); 2670 // The setprop command will not allow a property to be changed after 2671 // the first time it is set, so we don't have to worry about un-muting. 2672 setMasterMute_l(true); 2673 } 2674 } 2675 } 2676} 2677 2678// shared by MIXER and DIRECT, overridden by DUPLICATING 2679ssize_t AudioFlinger::PlaybackThread::threadLoop_write() 2680{ 2681 mInWrite = true; 2682 ssize_t bytesWritten; 2683 const size_t offset = mCurrentWriteLength - mBytesRemaining; 2684 2685 // If an NBAIO sink is present, use it to write the normal mixer's submix 2686 if (mNormalSink != 0) { 2687 2688 const size_t count = mBytesRemaining / mFrameSize; 2689 2690 ATRACE_BEGIN("write"); 2691 // update the setpoint when AudioFlinger::mScreenState changes 2692 uint32_t screenState = AudioFlinger::mScreenState; 2693 if (screenState != mScreenState) { 2694 mScreenState = screenState; 2695 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 2696 if (pipe != NULL) { 2697 pipe->setAvgFrames((mScreenState & 1) ? 2698 (pipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2699 } 2700 } 2701 ssize_t framesWritten = mNormalSink->write((char *)mSinkBuffer + offset, count); 2702 ATRACE_END(); 2703 if (framesWritten > 0) { 2704 bytesWritten = framesWritten * mFrameSize; 2705 } else { 2706 bytesWritten = framesWritten; 2707 } 2708 // otherwise use the HAL / AudioStreamOut directly 2709 } else { 2710 // Direct output and offload threads 2711 2712 if (mUseAsyncWrite) { 2713 ALOGW_IF(mWriteAckSequence & 1, "threadLoop_write(): out of sequence write request"); 2714 mWriteAckSequence += 2; 2715 mWriteAckSequence |= 1; 2716 ALOG_ASSERT(mCallbackThread != 0); 2717 mCallbackThread->setWriteBlocked(mWriteAckSequence); 2718 } 2719 // FIXME We should have an implementation of timestamps for direct output threads. 2720 // They are used e.g for multichannel PCM playback over HDMI. 2721 bytesWritten = mOutput->write((char *)mSinkBuffer + offset, mBytesRemaining); 2722 2723 if (mUseAsyncWrite && 2724 ((bytesWritten < 0) || (bytesWritten == (ssize_t)mBytesRemaining))) { 2725 // do not wait for async callback in case of error of full write 2726 mWriteAckSequence &= ~1; 2727 ALOG_ASSERT(mCallbackThread != 0); 2728 mCallbackThread->setWriteBlocked(mWriteAckSequence); 2729 } 2730 } 2731 2732 mNumWrites++; 2733 mInWrite = false; 2734 mStandby = false; 2735 return bytesWritten; 2736} 2737 2738void AudioFlinger::PlaybackThread::threadLoop_drain() 2739{ 2740 if (mOutput->stream->drain) { 2741 ALOGV("draining %s", (mMixerStatus == MIXER_DRAIN_TRACK) ? "early" : "full"); 2742 if (mUseAsyncWrite) { 2743 ALOGW_IF(mDrainSequence & 1, "threadLoop_drain(): out of sequence drain request"); 2744 mDrainSequence |= 1; 2745 ALOG_ASSERT(mCallbackThread != 0); 2746 mCallbackThread->setDraining(mDrainSequence); 2747 } 2748 mOutput->stream->drain(mOutput->stream, 2749 (mMixerStatus == MIXER_DRAIN_TRACK) ? AUDIO_DRAIN_EARLY_NOTIFY 2750 : AUDIO_DRAIN_ALL); 2751 } 2752} 2753 2754void AudioFlinger::PlaybackThread::threadLoop_exit() 2755{ 2756 { 2757 Mutex::Autolock _l(mLock); 2758 for (size_t i = 0; i < mTracks.size(); i++) { 2759 sp<Track> track = mTracks[i]; 2760 track->invalidate(); 2761 } 2762 } 2763} 2764 2765/* 2766The derived values that are cached: 2767 - mSinkBufferSize from frame count * frame size 2768 - mActiveSleepTimeUs from activeSleepTimeUs() 2769 - mIdleSleepTimeUs from idleSleepTimeUs() 2770 - mStandbyDelayNs from mActiveSleepTimeUs (DIRECT only) or forced to at least 2771 kDefaultStandbyTimeInNsecs when connected to an A2DP device. 2772 - maxPeriod from frame count and sample rate (MIXER only) 2773 2774The parameters that affect these derived values are: 2775 - frame count 2776 - frame size 2777 - sample rate 2778 - device type: A2DP or not 2779 - device latency 2780 - format: PCM or not 2781 - active sleep time 2782 - idle sleep time 2783*/ 2784 2785void AudioFlinger::PlaybackThread::cacheParameters_l() 2786{ 2787 mSinkBufferSize = mNormalFrameCount * mFrameSize; 2788 mActiveSleepTimeUs = activeSleepTimeUs(); 2789 mIdleSleepTimeUs = idleSleepTimeUs(); 2790 2791 // make sure standby delay is not too short when connected to an A2DP sink to avoid 2792 // truncating audio when going to standby. 2793 mStandbyDelayNs = AudioFlinger::mStandbyTimeInNsecs; 2794 if ((mOutDevice & AUDIO_DEVICE_OUT_ALL_A2DP) != 0) { 2795 if (mStandbyDelayNs < kDefaultStandbyTimeInNsecs) { 2796 mStandbyDelayNs = kDefaultStandbyTimeInNsecs; 2797 } 2798 } 2799} 2800 2801bool AudioFlinger::PlaybackThread::invalidateTracks_l(audio_stream_type_t streamType) 2802{ 2803 ALOGV("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %zu", 2804 this, streamType, mTracks.size()); 2805 bool trackMatch = false; 2806 size_t size = mTracks.size(); 2807 for (size_t i = 0; i < size; i++) { 2808 sp<Track> t = mTracks[i]; 2809 if (t->streamType() == streamType && t->isExternalTrack()) { 2810 t->invalidate(); 2811 trackMatch = true; 2812 } 2813 } 2814 return trackMatch; 2815} 2816 2817void AudioFlinger::PlaybackThread::invalidateTracks(audio_stream_type_t streamType) 2818{ 2819 Mutex::Autolock _l(mLock); 2820 invalidateTracks_l(streamType); 2821} 2822 2823status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 2824{ 2825 audio_session_t session = chain->sessionId(); 2826 int16_t* buffer = reinterpret_cast<int16_t*>(mEffectBufferEnabled 2827 ? mEffectBuffer : mSinkBuffer); 2828 bool ownsBuffer = false; 2829 2830 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 2831 if (session > AUDIO_SESSION_OUTPUT_MIX) { 2832 // Only one effect chain can be present in direct output thread and it uses 2833 // the sink buffer as input 2834 if (mType != DIRECT) { 2835 size_t numSamples = mNormalFrameCount * mChannelCount; 2836 buffer = new int16_t[numSamples]; 2837 memset(buffer, 0, numSamples * sizeof(int16_t)); 2838 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 2839 ownsBuffer = true; 2840 } 2841 2842 // Attach all tracks with same session ID to this chain. 2843 for (size_t i = 0; i < mTracks.size(); ++i) { 2844 sp<Track> track = mTracks[i]; 2845 if (session == track->sessionId()) { 2846 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), 2847 buffer); 2848 track->setMainBuffer(buffer); 2849 chain->incTrackCnt(); 2850 } 2851 } 2852 2853 // indicate all active tracks in the chain 2854 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 2855 sp<Track> track = mActiveTracks[i].promote(); 2856 if (track == 0) { 2857 continue; 2858 } 2859 if (session == track->sessionId()) { 2860 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 2861 chain->incActiveTrackCnt(); 2862 } 2863 } 2864 } 2865 chain->setThread(this); 2866 chain->setInBuffer(buffer, ownsBuffer); 2867 chain->setOutBuffer(reinterpret_cast<int16_t*>(mEffectBufferEnabled 2868 ? mEffectBuffer : mSinkBuffer)); 2869 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 2870 // chains list in order to be processed last as it contains output stage effects. 2871 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 2872 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 2873 // after track specific effects and before output stage. 2874 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 2875 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX. 2876 // Effect chain for other sessions are inserted at beginning of effect 2877 // chains list to be processed before output mix effects. Relative order between other 2878 // sessions is not important. 2879 static_assert(AUDIO_SESSION_OUTPUT_MIX == 0 && 2880 AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX, 2881 "audio_session_t constants misdefined"); 2882 size_t size = mEffectChains.size(); 2883 size_t i = 0; 2884 for (i = 0; i < size; i++) { 2885 if (mEffectChains[i]->sessionId() < session) { 2886 break; 2887 } 2888 } 2889 mEffectChains.insertAt(chain, i); 2890 checkSuspendOnAddEffectChain_l(chain); 2891 2892 return NO_ERROR; 2893} 2894 2895size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 2896{ 2897 audio_session_t session = chain->sessionId(); 2898 2899 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 2900 2901 for (size_t i = 0; i < mEffectChains.size(); i++) { 2902 if (chain == mEffectChains[i]) { 2903 mEffectChains.removeAt(i); 2904 // detach all active tracks from the chain 2905 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 2906 sp<Track> track = mActiveTracks[i].promote(); 2907 if (track == 0) { 2908 continue; 2909 } 2910 if (session == track->sessionId()) { 2911 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 2912 chain.get(), session); 2913 chain->decActiveTrackCnt(); 2914 } 2915 } 2916 2917 // detach all tracks with same session ID from this chain 2918 for (size_t i = 0; i < mTracks.size(); ++i) { 2919 sp<Track> track = mTracks[i]; 2920 if (session == track->sessionId()) { 2921 track->setMainBuffer(reinterpret_cast<int16_t*>(mSinkBuffer)); 2922 chain->decTrackCnt(); 2923 } 2924 } 2925 break; 2926 } 2927 } 2928 return mEffectChains.size(); 2929} 2930 2931status_t AudioFlinger::PlaybackThread::attachAuxEffect( 2932 const sp<AudioFlinger::PlaybackThread::Track>& track, int EffectId) 2933{ 2934 Mutex::Autolock _l(mLock); 2935 return attachAuxEffect_l(track, EffectId); 2936} 2937 2938status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 2939 const sp<AudioFlinger::PlaybackThread::Track>& track, int EffectId) 2940{ 2941 status_t status = NO_ERROR; 2942 2943 if (EffectId == 0) { 2944 track->setAuxBuffer(0, NULL); 2945 } else { 2946 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 2947 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 2948 if (effect != 0) { 2949 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 2950 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 2951 } else { 2952 status = INVALID_OPERATION; 2953 } 2954 } else { 2955 status = BAD_VALUE; 2956 } 2957 } 2958 return status; 2959} 2960 2961void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 2962{ 2963 for (size_t i = 0; i < mTracks.size(); ++i) { 2964 sp<Track> track = mTracks[i]; 2965 if (track->auxEffectId() == effectId) { 2966 attachAuxEffect_l(track, 0); 2967 } 2968 } 2969} 2970 2971bool AudioFlinger::PlaybackThread::threadLoop() 2972{ 2973 Vector< sp<Track> > tracksToRemove; 2974 2975 mStandbyTimeNs = systemTime(); 2976 nsecs_t lastWriteFinished = -1; // time last server write completed 2977 int64_t lastFramesWritten = -1; // track changes in timestamp server frames written 2978 2979 // MIXER 2980 nsecs_t lastWarning = 0; 2981 2982 // DUPLICATING 2983 // FIXME could this be made local to while loop? 2984 writeFrames = 0; 2985 2986 int lastGeneration = 0; 2987 2988 cacheParameters_l(); 2989 mSleepTimeUs = mIdleSleepTimeUs; 2990 2991 if (mType == MIXER) { 2992 sleepTimeShift = 0; 2993 } 2994 2995 CpuStats cpuStats; 2996 const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid())); 2997 2998 acquireWakeLock(); 2999 3000 // mNBLogWriter->log can only be called while thread mutex mLock is held. 3001 // So if you need to log when mutex is unlocked, set logString to a non-NULL string, 3002 // and then that string will be logged at the next convenient opportunity. 3003 const char *logString = NULL; 3004 3005 checkSilentMode_l(); 3006 3007 while (!exitPending()) 3008 { 3009 cpuStats.sample(myName); 3010 3011 Vector< sp<EffectChain> > effectChains; 3012 3013 { // scope for mLock 3014 3015 Mutex::Autolock _l(mLock); 3016 3017 processConfigEvents_l(); 3018 3019 if (logString != NULL) { 3020 mNBLogWriter->logTimestamp(); 3021 mNBLogWriter->log(logString); 3022 logString = NULL; 3023 } 3024 3025 // Gather the framesReleased counters for all active tracks, 3026 // and associate with the sink frames written out. We need 3027 // this to convert the sink timestamp to the track timestamp. 3028 bool kernelLocationUpdate = false; 3029 if (mNormalSink != 0) { 3030 // Note: The DuplicatingThread may not have a mNormalSink. 3031 // We always fetch the timestamp here because often the downstream 3032 // sink will block while writing. 3033 ExtendedTimestamp timestamp; // use private copy to fetch 3034 (void) mNormalSink->getTimestamp(timestamp); 3035 3036 // We keep track of the last valid kernel position in case we are in underrun 3037 // and the normal mixer period is the same as the fast mixer period, or there 3038 // is some error from the HAL. 3039 if (mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL] >= 0) { 3040 mTimestamp.mPosition[ExtendedTimestamp::LOCATION_KERNEL_LASTKERNELOK] = 3041 mTimestamp.mPosition[ExtendedTimestamp::LOCATION_KERNEL]; 3042 mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL_LASTKERNELOK] = 3043 mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL]; 3044 3045 mTimestamp.mPosition[ExtendedTimestamp::LOCATION_SERVER_LASTKERNELOK] = 3046 mTimestamp.mPosition[ExtendedTimestamp::LOCATION_SERVER]; 3047 mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_SERVER_LASTKERNELOK] = 3048 mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_SERVER]; 3049 } 3050 3051 if (timestamp.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL] >= 0) { 3052 kernelLocationUpdate = true; 3053 } else { 3054 ALOGVV("getTimestamp error - no valid kernel position"); 3055 } 3056 3057 // copy over kernel info 3058 mTimestamp.mPosition[ExtendedTimestamp::LOCATION_KERNEL] = 3059 timestamp.mPosition[ExtendedTimestamp::LOCATION_KERNEL] 3060 + mSuspendedFrames; // add frames discarded when suspended 3061 mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL] = 3062 timestamp.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL]; 3063 } 3064 // mFramesWritten for non-offloaded tracks are contiguous 3065 // even after standby() is called. This is useful for the track frame 3066 // to sink frame mapping. 3067 bool serverLocationUpdate = false; 3068 if (mFramesWritten != lastFramesWritten) { 3069 serverLocationUpdate = true; 3070 lastFramesWritten = mFramesWritten; 3071 } 3072 // Only update timestamps if there is a meaningful change. 3073 // Either the kernel timestamp must be valid or we have written something. 3074 if (kernelLocationUpdate || serverLocationUpdate) { 3075 if (serverLocationUpdate) { 3076 // use the time before we called the HAL write - it is a bit more accurate 3077 // to when the server last read data than the current time here. 3078 // 3079 // If we haven't written anything, mLastWriteTime will be -1 3080 // and we use systemTime(). 3081 mTimestamp.mPosition[ExtendedTimestamp::LOCATION_SERVER] = mFramesWritten; 3082 mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_SERVER] = mLastWriteTime == -1 3083 ? systemTime() : mLastWriteTime; 3084 } 3085 const size_t size = mActiveTracks.size(); 3086 for (size_t i = 0; i < size; ++i) { 3087 sp<Track> t = mActiveTracks[i].promote(); 3088 if (t != 0 && !t->isFastTrack()) { 3089 t->updateTrackFrameInfo( 3090 t->mAudioTrackServerProxy->framesReleased(), 3091 mFramesWritten, 3092 mTimestamp); 3093 } 3094 } 3095 } 3096 3097 saveOutputTracks(); 3098 if (mSignalPending) { 3099 // A signal was raised while we were unlocked 3100 mSignalPending = false; 3101 } else if (waitingAsyncCallback_l()) { 3102 if (exitPending()) { 3103 break; 3104 } 3105 bool released = false; 3106 if (!keepWakeLock()) { 3107 releaseWakeLock_l(); 3108 released = true; 3109 } 3110 mWakeLockUids.clear(); 3111 mActiveTracksGeneration++; 3112 ALOGV("wait async completion"); 3113 mWaitWorkCV.wait(mLock); 3114 ALOGV("async completion/wake"); 3115 if (released) { 3116 acquireWakeLock_l(); 3117 } 3118 mStandbyTimeNs = systemTime() + mStandbyDelayNs; 3119 mSleepTimeUs = 0; 3120 3121 continue; 3122 } 3123 if ((!mActiveTracks.size() && systemTime() > mStandbyTimeNs) || 3124 isSuspended()) { 3125 // put audio hardware into standby after short delay 3126 if (shouldStandby_l()) { 3127 3128 threadLoop_standby(); 3129 3130 mStandby = true; 3131 } 3132 3133 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 3134 // we're about to wait, flush the binder command buffer 3135 IPCThreadState::self()->flushCommands(); 3136 3137 clearOutputTracks(); 3138 3139 if (exitPending()) { 3140 break; 3141 } 3142 3143 releaseWakeLock_l(); 3144 mWakeLockUids.clear(); 3145 mActiveTracksGeneration++; 3146 // wait until we have something to do... 3147 ALOGV("%s going to sleep", myName.string()); 3148 mWaitWorkCV.wait(mLock); 3149 ALOGV("%s waking up", myName.string()); 3150 acquireWakeLock_l(); 3151 3152 mMixerStatus = MIXER_IDLE; 3153 mMixerStatusIgnoringFastTracks = MIXER_IDLE; 3154 mBytesWritten = 0; 3155 mBytesRemaining = 0; 3156 checkSilentMode_l(); 3157 3158 mStandbyTimeNs = systemTime() + mStandbyDelayNs; 3159 mSleepTimeUs = mIdleSleepTimeUs; 3160 if (mType == MIXER) { 3161 sleepTimeShift = 0; 3162 } 3163 3164 continue; 3165 } 3166 } 3167 // mMixerStatusIgnoringFastTracks is also updated internally 3168 mMixerStatus = prepareTracks_l(&tracksToRemove); 3169 3170 // compare with previously applied list 3171 if (lastGeneration != mActiveTracksGeneration) { 3172 // update wakelock 3173 updateWakeLockUids_l(mWakeLockUids); 3174 lastGeneration = mActiveTracksGeneration; 3175 } 3176 3177 // prevent any changes in effect chain list and in each effect chain 3178 // during mixing and effect process as the audio buffers could be deleted 3179 // or modified if an effect is created or deleted 3180 lockEffectChains_l(effectChains); 3181 } // mLock scope ends 3182 3183 if (mBytesRemaining == 0) { 3184 mCurrentWriteLength = 0; 3185 if (mMixerStatus == MIXER_TRACKS_READY) { 3186 // threadLoop_mix() sets mCurrentWriteLength 3187 threadLoop_mix(); 3188 } else if ((mMixerStatus != MIXER_DRAIN_TRACK) 3189 && (mMixerStatus != MIXER_DRAIN_ALL)) { 3190 // threadLoop_sleepTime sets mSleepTimeUs to 0 if data 3191 // must be written to HAL 3192 threadLoop_sleepTime(); 3193 if (mSleepTimeUs == 0) { 3194 mCurrentWriteLength = mSinkBufferSize; 3195 } 3196 } 3197 // Either threadLoop_mix() or threadLoop_sleepTime() should have set 3198 // mMixerBuffer with data if mMixerBufferValid is true and mSleepTimeUs == 0. 3199 // Merge mMixerBuffer data into mEffectBuffer (if any effects are valid) 3200 // or mSinkBuffer (if there are no effects). 3201 // 3202 // This is done pre-effects computation; if effects change to 3203 // support higher precision, this needs to move. 3204 // 3205 // mMixerBufferValid is only set true by MixerThread::prepareTracks_l(). 3206 // TODO use mSleepTimeUs == 0 as an additional condition. 3207 if (mMixerBufferValid) { 3208 void *buffer = mEffectBufferValid ? mEffectBuffer : mSinkBuffer; 3209 audio_format_t format = mEffectBufferValid ? mEffectBufferFormat : mFormat; 3210 3211 // mono blend occurs for mixer threads only (not direct or offloaded) 3212 // and is handled here if we're going directly to the sink. 3213 if (requireMonoBlend() && !mEffectBufferValid) { 3214 mono_blend(mMixerBuffer, mMixerBufferFormat, mChannelCount, mNormalFrameCount, 3215 true /*limit*/); 3216 } 3217 3218 memcpy_by_audio_format(buffer, format, mMixerBuffer, mMixerBufferFormat, 3219 mNormalFrameCount * mChannelCount); 3220 } 3221 3222 mBytesRemaining = mCurrentWriteLength; 3223 if (isSuspended()) { 3224 // Simulate write to HAL when suspended (e.g. BT SCO phone call). 3225 mSleepTimeUs = suspendSleepTimeUs(); // assumes full buffer. 3226 const size_t framesRemaining = mBytesRemaining / mFrameSize; 3227 mBytesWritten += mBytesRemaining; 3228 mFramesWritten += framesRemaining; 3229 mSuspendedFrames += framesRemaining; // to adjust kernel HAL position 3230 mBytesRemaining = 0; 3231 } 3232 3233 // only process effects if we're going to write 3234 if (mSleepTimeUs == 0 && mType != OFFLOAD) { 3235 for (size_t i = 0; i < effectChains.size(); i ++) { 3236 effectChains[i]->process_l(); 3237 } 3238 } 3239 } 3240 // Process effect chains for offloaded thread even if no audio 3241 // was read from audio track: process only updates effect state 3242 // and thus does have to be synchronized with audio writes but may have 3243 // to be called while waiting for async write callback 3244 if (mType == OFFLOAD) { 3245 for (size_t i = 0; i < effectChains.size(); i ++) { 3246 effectChains[i]->process_l(); 3247 } 3248 } 3249 3250 // Only if the Effects buffer is enabled and there is data in the 3251 // Effects buffer (buffer valid), we need to 3252 // copy into the sink buffer. 3253 // TODO use mSleepTimeUs == 0 as an additional condition. 3254 if (mEffectBufferValid) { 3255 //ALOGV("writing effect buffer to sink buffer format %#x", mFormat); 3256 3257 if (requireMonoBlend()) { 3258 mono_blend(mEffectBuffer, mEffectBufferFormat, mChannelCount, mNormalFrameCount, 3259 true /*limit*/); 3260 } 3261 3262 memcpy_by_audio_format(mSinkBuffer, mFormat, mEffectBuffer, mEffectBufferFormat, 3263 mNormalFrameCount * mChannelCount); 3264 } 3265 3266 // enable changes in effect chain 3267 unlockEffectChains(effectChains); 3268 3269 if (!waitingAsyncCallback()) { 3270 // mSleepTimeUs == 0 means we must write to audio hardware 3271 if (mSleepTimeUs == 0) { 3272 ssize_t ret = 0; 3273 // We save lastWriteFinished here, as previousLastWriteFinished, 3274 // for throttling. On thread start, previousLastWriteFinished will be 3275 // set to -1, which properly results in no throttling after the first write. 3276 nsecs_t previousLastWriteFinished = lastWriteFinished; 3277 nsecs_t delta = 0; 3278 if (mBytesRemaining) { 3279 // FIXME rewrite to reduce number of system calls 3280 mLastWriteTime = systemTime(); // also used for dumpsys 3281 ret = threadLoop_write(); 3282 lastWriteFinished = systemTime(); 3283 delta = lastWriteFinished - mLastWriteTime; 3284 if (ret < 0) { 3285 mBytesRemaining = 0; 3286 } else { 3287 mBytesWritten += ret; 3288 mBytesRemaining -= ret; 3289 mFramesWritten += ret / mFrameSize; 3290 } 3291 } else if ((mMixerStatus == MIXER_DRAIN_TRACK) || 3292 (mMixerStatus == MIXER_DRAIN_ALL)) { 3293 threadLoop_drain(); 3294 } 3295 if (mType == MIXER && !mStandby) { 3296 // write blocked detection 3297 if (delta > maxPeriod) { 3298 mNumDelayedWrites++; 3299 if ((lastWriteFinished - lastWarning) > kWarningThrottleNs) { 3300 ATRACE_NAME("underrun"); 3301 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 3302 (unsigned long long) ns2ms(delta), mNumDelayedWrites, this); 3303 lastWarning = lastWriteFinished; 3304 } 3305 } 3306 3307 if (mThreadThrottle 3308 && mMixerStatus == MIXER_TRACKS_READY // we are mixing (active tracks) 3309 && ret > 0) { // we wrote something 3310 // Limit MixerThread data processing to no more than twice the 3311 // expected processing rate. 3312 // 3313 // This helps prevent underruns with NuPlayer and other applications 3314 // which may set up buffers that are close to the minimum size, or use 3315 // deep buffers, and rely on a double-buffering sleep strategy to fill. 3316 // 3317 // The throttle smooths out sudden large data drains from the device, 3318 // e.g. when it comes out of standby, which often causes problems with 3319 // (1) mixer threads without a fast mixer (which has its own warm-up) 3320 // (2) minimum buffer sized tracks (even if the track is full, 3321 // the app won't fill fast enough to handle the sudden draw). 3322 // 3323 // Total time spent in last processing cycle equals time spent in 3324 // 1. threadLoop_write, as well as time spent in 3325 // 2. threadLoop_mix (significant for heavy mixing, especially 3326 // on low tier processors) 3327 3328 // it's OK if deltaMs is an overestimate. 3329 const int32_t deltaMs = 3330 (lastWriteFinished - previousLastWriteFinished) / 1000000; 3331 const int32_t throttleMs = mHalfBufferMs - deltaMs; 3332 if ((signed)mHalfBufferMs >= throttleMs && throttleMs > 0) { 3333 usleep(throttleMs * 1000); 3334 // notify of throttle start on verbose log 3335 ALOGV_IF(mThreadThrottleEndMs == mThreadThrottleTimeMs, 3336 "mixer(%p) throttle begin:" 3337 " ret(%zd) deltaMs(%d) requires sleep %d ms", 3338 this, ret, deltaMs, throttleMs); 3339 mThreadThrottleTimeMs += throttleMs; 3340 // Throttle must be attributed to the previous mixer loop's write time 3341 // to allow back-to-back throttling. 3342 lastWriteFinished += throttleMs * 1000000; 3343 } else { 3344 uint32_t diff = mThreadThrottleTimeMs - mThreadThrottleEndMs; 3345 if (diff > 0) { 3346 // notify of throttle end on debug log 3347 // but prevent spamming for bluetooth 3348 ALOGD_IF(!audio_is_a2dp_out_device(outDevice()), 3349 "mixer(%p) throttle end: throttle time(%u)", this, diff); 3350 mThreadThrottleEndMs = mThreadThrottleTimeMs; 3351 } 3352 } 3353 } 3354 } 3355 3356 } else { 3357 ATRACE_BEGIN("sleep"); 3358 Mutex::Autolock _l(mLock); 3359 if (!mSignalPending && mConfigEvents.isEmpty() && !exitPending()) { 3360 mWaitWorkCV.waitRelative(mLock, microseconds((nsecs_t)mSleepTimeUs)); 3361 } 3362 ATRACE_END(); 3363 } 3364 } 3365 3366 // Finally let go of removed track(s), without the lock held 3367 // since we can't guarantee the destructors won't acquire that 3368 // same lock. This will also mutate and push a new fast mixer state. 3369 threadLoop_removeTracks(tracksToRemove); 3370 tracksToRemove.clear(); 3371 3372 // FIXME I don't understand the need for this here; 3373 // it was in the original code but maybe the 3374 // assignment in saveOutputTracks() makes this unnecessary? 3375 clearOutputTracks(); 3376 3377 // Effect chains will be actually deleted here if they were removed from 3378 // mEffectChains list during mixing or effects processing 3379 effectChains.clear(); 3380 3381 // FIXME Note that the above .clear() is no longer necessary since effectChains 3382 // is now local to this block, but will keep it for now (at least until merge done). 3383 } 3384 3385 threadLoop_exit(); 3386 3387 if (!mStandby) { 3388 threadLoop_standby(); 3389 mStandby = true; 3390 } 3391 3392 releaseWakeLock(); 3393 mWakeLockUids.clear(); 3394 mActiveTracksGeneration++; 3395 3396 ALOGV("Thread %p type %d exiting", this, mType); 3397 return false; 3398} 3399 3400// removeTracks_l() must be called with ThreadBase::mLock held 3401void AudioFlinger::PlaybackThread::removeTracks_l(const Vector< sp<Track> >& tracksToRemove) 3402{ 3403 size_t count = tracksToRemove.size(); 3404 if (count > 0) { 3405 for (size_t i=0 ; i<count ; i++) { 3406 const sp<Track>& track = tracksToRemove.itemAt(i); 3407 mActiveTracks.remove(track); 3408 mWakeLockUids.remove(track->uid()); 3409 mActiveTracksGeneration++; 3410 ALOGV("removeTracks_l removing track on session %d", track->sessionId()); 3411 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 3412 if (chain != 0) { 3413 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), 3414 track->sessionId()); 3415 chain->decActiveTrackCnt(); 3416 } 3417 if (track->isTerminated()) { 3418 removeTrack_l(track); 3419 } 3420 } 3421 } 3422 3423} 3424 3425status_t AudioFlinger::PlaybackThread::getTimestamp_l(AudioTimestamp& timestamp) 3426{ 3427 if (mNormalSink != 0) { 3428 ExtendedTimestamp ets; 3429 status_t status = mNormalSink->getTimestamp(ets); 3430 if (status == NO_ERROR) { 3431 status = ets.getBestTimestamp(×tamp); 3432 } 3433 return status; 3434 } 3435 if ((mType == OFFLOAD || mType == DIRECT) 3436 && mOutput != NULL && mOutput->stream->get_presentation_position) { 3437 uint64_t position64; 3438 int ret = mOutput->getPresentationPosition(&position64, ×tamp.mTime); 3439 if (ret == 0) { 3440 timestamp.mPosition = (uint32_t)position64; 3441 return NO_ERROR; 3442 } 3443 } 3444 return INVALID_OPERATION; 3445} 3446 3447status_t AudioFlinger::MixerThread::createAudioPatch_l(const struct audio_patch *patch, 3448 audio_patch_handle_t *handle) 3449{ 3450 AutoPark<FastMixer> park(mFastMixer); 3451 3452 status_t status = PlaybackThread::createAudioPatch_l(patch, handle); 3453 3454 return status; 3455} 3456 3457status_t AudioFlinger::PlaybackThread::createAudioPatch_l(const struct audio_patch *patch, 3458 audio_patch_handle_t *handle) 3459{ 3460 status_t status = NO_ERROR; 3461 3462 // store new device and send to effects 3463 audio_devices_t type = AUDIO_DEVICE_NONE; 3464 for (unsigned int i = 0; i < patch->num_sinks; i++) { 3465 type |= patch->sinks[i].ext.device.type; 3466 } 3467 3468#ifdef ADD_BATTERY_DATA 3469 // when changing the audio output device, call addBatteryData to notify 3470 // the change 3471 if (mOutDevice != type) { 3472 uint32_t params = 0; 3473 // check whether speaker is on 3474 if (type & AUDIO_DEVICE_OUT_SPEAKER) { 3475 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 3476 } 3477 3478 audio_devices_t deviceWithoutSpeaker 3479 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 3480 // check if any other device (except speaker) is on 3481 if (type & deviceWithoutSpeaker) { 3482 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 3483 } 3484 3485 if (params != 0) { 3486 addBatteryData(params); 3487 } 3488 } 3489#endif 3490 3491 for (size_t i = 0; i < mEffectChains.size(); i++) { 3492 mEffectChains[i]->setDevice_l(type); 3493 } 3494 3495 // mPrevOutDevice is the latest device set by createAudioPatch_l(). It is not set when 3496 // the thread is created so that the first patch creation triggers an ioConfigChanged callback 3497 bool configChanged = mPrevOutDevice != type; 3498 mOutDevice = type; 3499 mPatch = *patch; 3500 3501 if (mOutput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 3502 audio_hw_device_t *hwDevice = mOutput->audioHwDev->hwDevice(); 3503 status = hwDevice->create_audio_patch(hwDevice, 3504 patch->num_sources, 3505 patch->sources, 3506 patch->num_sinks, 3507 patch->sinks, 3508 handle); 3509 } else { 3510 char *address; 3511 if (strcmp(patch->sinks[0].ext.device.address, "") != 0) { 3512 //FIXME: we only support address on first sink with HAL version < 3.0 3513 address = audio_device_address_to_parameter( 3514 patch->sinks[0].ext.device.type, 3515 patch->sinks[0].ext.device.address); 3516 } else { 3517 address = (char *)calloc(1, 1); 3518 } 3519 AudioParameter param = AudioParameter(String8(address)); 3520 free(address); 3521 param.addInt(String8(AUDIO_PARAMETER_STREAM_ROUTING), (int)type); 3522 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3523 param.toString().string()); 3524 *handle = AUDIO_PATCH_HANDLE_NONE; 3525 } 3526 if (configChanged) { 3527 mPrevOutDevice = type; 3528 sendIoConfigEvent_l(AUDIO_OUTPUT_CONFIG_CHANGED); 3529 } 3530 return status; 3531} 3532 3533status_t AudioFlinger::MixerThread::releaseAudioPatch_l(const audio_patch_handle_t handle) 3534{ 3535 AutoPark<FastMixer> park(mFastMixer); 3536 3537 status_t status = PlaybackThread::releaseAudioPatch_l(handle); 3538 3539 return status; 3540} 3541 3542status_t AudioFlinger::PlaybackThread::releaseAudioPatch_l(const audio_patch_handle_t handle) 3543{ 3544 status_t status = NO_ERROR; 3545 3546 mOutDevice = AUDIO_DEVICE_NONE; 3547 3548 if (mOutput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 3549 audio_hw_device_t *hwDevice = mOutput->audioHwDev->hwDevice(); 3550 status = hwDevice->release_audio_patch(hwDevice, handle); 3551 } else { 3552 AudioParameter param; 3553 param.addInt(String8(AUDIO_PARAMETER_STREAM_ROUTING), 0); 3554 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3555 param.toString().string()); 3556 } 3557 return status; 3558} 3559 3560void AudioFlinger::PlaybackThread::addPatchTrack(const sp<PatchTrack>& track) 3561{ 3562 Mutex::Autolock _l(mLock); 3563 mTracks.add(track); 3564} 3565 3566void AudioFlinger::PlaybackThread::deletePatchTrack(const sp<PatchTrack>& track) 3567{ 3568 Mutex::Autolock _l(mLock); 3569 destroyTrack_l(track); 3570} 3571 3572void AudioFlinger::PlaybackThread::getAudioPortConfig(struct audio_port_config *config) 3573{ 3574 ThreadBase::getAudioPortConfig(config); 3575 config->role = AUDIO_PORT_ROLE_SOURCE; 3576 config->ext.mix.hw_module = mOutput->audioHwDev->handle(); 3577 config->ext.mix.usecase.stream = AUDIO_STREAM_DEFAULT; 3578} 3579 3580// ---------------------------------------------------------------------------- 3581 3582AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 3583 audio_io_handle_t id, audio_devices_t device, bool systemReady, type_t type) 3584 : PlaybackThread(audioFlinger, output, id, device, type, systemReady), 3585 // mAudioMixer below 3586 // mFastMixer below 3587 mFastMixerFutex(0), 3588 mMasterMono(false) 3589 // mOutputSink below 3590 // mPipeSink below 3591 // mNormalSink below 3592{ 3593 ALOGV("MixerThread() id=%d device=%#x type=%d", id, device, type); 3594 ALOGV("mSampleRate=%u, mChannelMask=%#x, mChannelCount=%u, mFormat=%d, mFrameSize=%zu, " 3595 "mFrameCount=%zu, mNormalFrameCount=%zu", 3596 mSampleRate, mChannelMask, mChannelCount, mFormat, mFrameSize, mFrameCount, 3597 mNormalFrameCount); 3598 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 3599 3600 if (type == DUPLICATING) { 3601 // The Duplicating thread uses the AudioMixer and delivers data to OutputTracks 3602 // (downstream MixerThreads) in DuplicatingThread::threadLoop_write(). 3603 // Do not create or use mFastMixer, mOutputSink, mPipeSink, or mNormalSink. 3604 return; 3605 } 3606 // create an NBAIO sink for the HAL output stream, and negotiate 3607 mOutputSink = new AudioStreamOutSink(output->stream); 3608 size_t numCounterOffers = 0; 3609 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount, mFormat)}; 3610#if !LOG_NDEBUG 3611 ssize_t index = 3612#else 3613 (void) 3614#endif 3615 mOutputSink->negotiate(offers, 1, NULL, numCounterOffers); 3616 ALOG_ASSERT(index == 0); 3617 3618 // initialize fast mixer depending on configuration 3619 bool initFastMixer; 3620 switch (kUseFastMixer) { 3621 case FastMixer_Never: 3622 initFastMixer = false; 3623 break; 3624 case FastMixer_Always: 3625 initFastMixer = true; 3626 break; 3627 case FastMixer_Static: 3628 case FastMixer_Dynamic: 3629 initFastMixer = mFrameCount < mNormalFrameCount; 3630 break; 3631 } 3632 if (initFastMixer) { 3633 audio_format_t fastMixerFormat; 3634 if (mMixerBufferEnabled && mEffectBufferEnabled) { 3635 fastMixerFormat = AUDIO_FORMAT_PCM_FLOAT; 3636 } else { 3637 fastMixerFormat = AUDIO_FORMAT_PCM_16_BIT; 3638 } 3639 if (mFormat != fastMixerFormat) { 3640 // change our Sink format to accept our intermediate precision 3641 mFormat = fastMixerFormat; 3642 free(mSinkBuffer); 3643 mFrameSize = mChannelCount * audio_bytes_per_sample(mFormat); 3644 const size_t sinkBufferSize = mNormalFrameCount * mFrameSize; 3645 (void)posix_memalign(&mSinkBuffer, 32, sinkBufferSize); 3646 } 3647 3648 // create a MonoPipe to connect our submix to FastMixer 3649 NBAIO_Format format = mOutputSink->format(); 3650#ifdef TEE_SINK 3651 NBAIO_Format origformat = format; 3652#endif 3653 // adjust format to match that of the Fast Mixer 3654 ALOGV("format changed from %d to %d", format.mFormat, fastMixerFormat); 3655 format.mFormat = fastMixerFormat; 3656 format.mFrameSize = audio_bytes_per_sample(format.mFormat) * format.mChannelCount; 3657 3658 // This pipe depth compensates for scheduling latency of the normal mixer thread. 3659 // When it wakes up after a maximum latency, it runs a few cycles quickly before 3660 // finally blocking. Note the pipe implementation rounds up the request to a power of 2. 3661 MonoPipe *monoPipe = new MonoPipe(mNormalFrameCount * 4, format, true /*writeCanBlock*/); 3662 const NBAIO_Format offers[1] = {format}; 3663 size_t numCounterOffers = 0; 3664#if !LOG_NDEBUG || defined(TEE_SINK) 3665 ssize_t index = 3666#else 3667 (void) 3668#endif 3669 monoPipe->negotiate(offers, 1, NULL, numCounterOffers); 3670 ALOG_ASSERT(index == 0); 3671 monoPipe->setAvgFrames((mScreenState & 1) ? 3672 (monoPipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 3673 mPipeSink = monoPipe; 3674 3675#ifdef TEE_SINK 3676 if (mTeeSinkOutputEnabled) { 3677 // create a Pipe to archive a copy of FastMixer's output for dumpsys 3678 Pipe *teeSink = new Pipe(mTeeSinkOutputFrames, origformat); 3679 const NBAIO_Format offers2[1] = {origformat}; 3680 numCounterOffers = 0; 3681 index = teeSink->negotiate(offers2, 1, NULL, numCounterOffers); 3682 ALOG_ASSERT(index == 0); 3683 mTeeSink = teeSink; 3684 PipeReader *teeSource = new PipeReader(*teeSink); 3685 numCounterOffers = 0; 3686 index = teeSource->negotiate(offers2, 1, NULL, numCounterOffers); 3687 ALOG_ASSERT(index == 0); 3688 mTeeSource = teeSource; 3689 } 3690#endif 3691 3692 // create fast mixer and configure it initially with just one fast track for our submix 3693 mFastMixer = new FastMixer(); 3694 FastMixerStateQueue *sq = mFastMixer->sq(); 3695#ifdef STATE_QUEUE_DUMP 3696 sq->setObserverDump(&mStateQueueObserverDump); 3697 sq->setMutatorDump(&mStateQueueMutatorDump); 3698#endif 3699 FastMixerState *state = sq->begin(); 3700 FastTrack *fastTrack = &state->mFastTracks[0]; 3701 // wrap the source side of the MonoPipe to make it an AudioBufferProvider 3702 fastTrack->mBufferProvider = new SourceAudioBufferProvider(new MonoPipeReader(monoPipe)); 3703 fastTrack->mVolumeProvider = NULL; 3704 fastTrack->mChannelMask = mChannelMask; // mPipeSink channel mask for audio to FastMixer 3705 fastTrack->mFormat = mFormat; // mPipeSink format for audio to FastMixer 3706 fastTrack->mGeneration++; 3707 state->mFastTracksGen++; 3708 state->mTrackMask = 1; 3709 // fast mixer will use the HAL output sink 3710 state->mOutputSink = mOutputSink.get(); 3711 state->mOutputSinkGen++; 3712 state->mFrameCount = mFrameCount; 3713 state->mCommand = FastMixerState::COLD_IDLE; 3714 // already done in constructor initialization list 3715 //mFastMixerFutex = 0; 3716 state->mColdFutexAddr = &mFastMixerFutex; 3717 state->mColdGen++; 3718 state->mDumpState = &mFastMixerDumpState; 3719#ifdef TEE_SINK 3720 state->mTeeSink = mTeeSink.get(); 3721#endif 3722 mFastMixerNBLogWriter = audioFlinger->newWriter_l(kFastMixerLogSize, "FastMixer"); 3723 state->mNBLogWriter = mFastMixerNBLogWriter.get(); 3724 sq->end(); 3725 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3726 3727 // start the fast mixer 3728 mFastMixer->run("FastMixer", PRIORITY_URGENT_AUDIO); 3729 pid_t tid = mFastMixer->getTid(); 3730 sendPrioConfigEvent(getpid_cached, tid, kPriorityFastMixer); 3731 3732#ifdef AUDIO_WATCHDOG 3733 // create and start the watchdog 3734 mAudioWatchdog = new AudioWatchdog(); 3735 mAudioWatchdog->setDump(&mAudioWatchdogDump); 3736 mAudioWatchdog->run("AudioWatchdog", PRIORITY_URGENT_AUDIO); 3737 tid = mAudioWatchdog->getTid(); 3738 sendPrioConfigEvent(getpid_cached, tid, kPriorityFastMixer); 3739#endif 3740 3741 } 3742 3743 switch (kUseFastMixer) { 3744 case FastMixer_Never: 3745 case FastMixer_Dynamic: 3746 mNormalSink = mOutputSink; 3747 break; 3748 case FastMixer_Always: 3749 mNormalSink = mPipeSink; 3750 break; 3751 case FastMixer_Static: 3752 mNormalSink = initFastMixer ? mPipeSink : mOutputSink; 3753 break; 3754 } 3755} 3756 3757AudioFlinger::MixerThread::~MixerThread() 3758{ 3759 if (mFastMixer != 0) { 3760 FastMixerStateQueue *sq = mFastMixer->sq(); 3761 FastMixerState *state = sq->begin(); 3762 if (state->mCommand == FastMixerState::COLD_IDLE) { 3763 int32_t old = android_atomic_inc(&mFastMixerFutex); 3764 if (old == -1) { 3765 (void) syscall(__NR_futex, &mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 3766 } 3767 } 3768 state->mCommand = FastMixerState::EXIT; 3769 sq->end(); 3770 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3771 mFastMixer->join(); 3772 // Though the fast mixer thread has exited, it's state queue is still valid. 3773 // We'll use that extract the final state which contains one remaining fast track 3774 // corresponding to our sub-mix. 3775 state = sq->begin(); 3776 ALOG_ASSERT(state->mTrackMask == 1); 3777 FastTrack *fastTrack = &state->mFastTracks[0]; 3778 ALOG_ASSERT(fastTrack->mBufferProvider != NULL); 3779 delete fastTrack->mBufferProvider; 3780 sq->end(false /*didModify*/); 3781 mFastMixer.clear(); 3782#ifdef AUDIO_WATCHDOG 3783 if (mAudioWatchdog != 0) { 3784 mAudioWatchdog->requestExit(); 3785 mAudioWatchdog->requestExitAndWait(); 3786 mAudioWatchdog.clear(); 3787 } 3788#endif 3789 } 3790 mAudioFlinger->unregisterWriter(mFastMixerNBLogWriter); 3791 delete mAudioMixer; 3792} 3793 3794 3795uint32_t AudioFlinger::MixerThread::correctLatency_l(uint32_t latency) const 3796{ 3797 if (mFastMixer != 0) { 3798 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 3799 latency += (pipe->getAvgFrames() * 1000) / mSampleRate; 3800 } 3801 return latency; 3802} 3803 3804 3805void AudioFlinger::MixerThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) 3806{ 3807 PlaybackThread::threadLoop_removeTracks(tracksToRemove); 3808} 3809 3810ssize_t AudioFlinger::MixerThread::threadLoop_write() 3811{ 3812 // FIXME we should only do one push per cycle; confirm this is true 3813 // Start the fast mixer if it's not already running 3814 if (mFastMixer != 0) { 3815 FastMixerStateQueue *sq = mFastMixer->sq(); 3816 FastMixerState *state = sq->begin(); 3817 if (state->mCommand != FastMixerState::MIX_WRITE && 3818 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)) { 3819 if (state->mCommand == FastMixerState::COLD_IDLE) { 3820 3821 // FIXME workaround for first HAL write being CPU bound on some devices 3822 ATRACE_BEGIN("write"); 3823 mOutput->write((char *)mSinkBuffer, 0); 3824 ATRACE_END(); 3825 3826 int32_t old = android_atomic_inc(&mFastMixerFutex); 3827 if (old == -1) { 3828 (void) syscall(__NR_futex, &mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 3829 } 3830#ifdef AUDIO_WATCHDOG 3831 if (mAudioWatchdog != 0) { 3832 mAudioWatchdog->resume(); 3833 } 3834#endif 3835 } 3836 state->mCommand = FastMixerState::MIX_WRITE; 3837#ifdef FAST_THREAD_STATISTICS 3838 mFastMixerDumpState.increaseSamplingN(mAudioFlinger->isLowRamDevice() ? 3839 FastThreadDumpState::kSamplingNforLowRamDevice : FastThreadDumpState::kSamplingN); 3840#endif 3841 sq->end(); 3842 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3843 if (kUseFastMixer == FastMixer_Dynamic) { 3844 mNormalSink = mPipeSink; 3845 } 3846 } else { 3847 sq->end(false /*didModify*/); 3848 } 3849 } 3850 return PlaybackThread::threadLoop_write(); 3851} 3852 3853void AudioFlinger::MixerThread::threadLoop_standby() 3854{ 3855 // Idle the fast mixer if it's currently running 3856 if (mFastMixer != 0) { 3857 FastMixerStateQueue *sq = mFastMixer->sq(); 3858 FastMixerState *state = sq->begin(); 3859 if (!(state->mCommand & FastMixerState::IDLE)) { 3860 state->mCommand = FastMixerState::COLD_IDLE; 3861 state->mColdFutexAddr = &mFastMixerFutex; 3862 state->mColdGen++; 3863 mFastMixerFutex = 0; 3864 sq->end(); 3865 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 3866 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3867 if (kUseFastMixer == FastMixer_Dynamic) { 3868 mNormalSink = mOutputSink; 3869 } 3870#ifdef AUDIO_WATCHDOG 3871 if (mAudioWatchdog != 0) { 3872 mAudioWatchdog->pause(); 3873 } 3874#endif 3875 } else { 3876 sq->end(false /*didModify*/); 3877 } 3878 } 3879 PlaybackThread::threadLoop_standby(); 3880} 3881 3882bool AudioFlinger::PlaybackThread::waitingAsyncCallback_l() 3883{ 3884 return false; 3885} 3886 3887bool AudioFlinger::PlaybackThread::shouldStandby_l() 3888{ 3889 return !mStandby; 3890} 3891 3892bool AudioFlinger::PlaybackThread::waitingAsyncCallback() 3893{ 3894 Mutex::Autolock _l(mLock); 3895 return waitingAsyncCallback_l(); 3896} 3897 3898// shared by MIXER and DIRECT, overridden by DUPLICATING 3899void AudioFlinger::PlaybackThread::threadLoop_standby() 3900{ 3901 ALOGV("Audio hardware entering standby, mixer %p, suspend count %d", this, mSuspended); 3902 mOutput->standby(); 3903 if (mUseAsyncWrite != 0) { 3904 // discard any pending drain or write ack by incrementing sequence 3905 mWriteAckSequence = (mWriteAckSequence + 2) & ~1; 3906 mDrainSequence = (mDrainSequence + 2) & ~1; 3907 ALOG_ASSERT(mCallbackThread != 0); 3908 mCallbackThread->setWriteBlocked(mWriteAckSequence); 3909 mCallbackThread->setDraining(mDrainSequence); 3910 } 3911 mHwPaused = false; 3912} 3913 3914void AudioFlinger::PlaybackThread::onAddNewTrack_l() 3915{ 3916 ALOGV("signal playback thread"); 3917 broadcast_l(); 3918} 3919 3920void AudioFlinger::PlaybackThread::onAsyncError() 3921{ 3922 for (int i = AUDIO_STREAM_SYSTEM; i < (int)AUDIO_STREAM_CNT; i++) { 3923 invalidateTracks((audio_stream_type_t)i); 3924 } 3925} 3926 3927void AudioFlinger::MixerThread::threadLoop_mix() 3928{ 3929 // mix buffers... 3930 mAudioMixer->process(); 3931 mCurrentWriteLength = mSinkBufferSize; 3932 // increase sleep time progressively when application underrun condition clears. 3933 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 3934 // that a steady state of alternating ready/not ready conditions keeps the sleep time 3935 // such that we would underrun the audio HAL. 3936 if ((mSleepTimeUs == 0) && (sleepTimeShift > 0)) { 3937 sleepTimeShift--; 3938 } 3939 mSleepTimeUs = 0; 3940 mStandbyTimeNs = systemTime() + mStandbyDelayNs; 3941 //TODO: delay standby when effects have a tail 3942 3943} 3944 3945void AudioFlinger::MixerThread::threadLoop_sleepTime() 3946{ 3947 // If no tracks are ready, sleep once for the duration of an output 3948 // buffer size, then write 0s to the output 3949 if (mSleepTimeUs == 0) { 3950 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3951 mSleepTimeUs = mActiveSleepTimeUs >> sleepTimeShift; 3952 if (mSleepTimeUs < kMinThreadSleepTimeUs) { 3953 mSleepTimeUs = kMinThreadSleepTimeUs; 3954 } 3955 // reduce sleep time in case of consecutive application underruns to avoid 3956 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 3957 // duration we would end up writing less data than needed by the audio HAL if 3958 // the condition persists. 3959 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 3960 sleepTimeShift++; 3961 } 3962 } else { 3963 mSleepTimeUs = mIdleSleepTimeUs; 3964 } 3965 } else if (mBytesWritten != 0 || (mMixerStatus == MIXER_TRACKS_ENABLED)) { 3966 // clear out mMixerBuffer or mSinkBuffer, to ensure buffers are cleared 3967 // before effects processing or output. 3968 if (mMixerBufferValid) { 3969 memset(mMixerBuffer, 0, mMixerBufferSize); 3970 } else { 3971 memset(mSinkBuffer, 0, mSinkBufferSize); 3972 } 3973 mSleepTimeUs = 0; 3974 ALOGV_IF(mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED), 3975 "anticipated start"); 3976 } 3977 // TODO add standby time extension fct of effect tail 3978} 3979 3980// prepareTracks_l() must be called with ThreadBase::mLock held 3981AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 3982 Vector< sp<Track> > *tracksToRemove) 3983{ 3984 3985 mixer_state mixerStatus = MIXER_IDLE; 3986 // find out which tracks need to be processed 3987 size_t count = mActiveTracks.size(); 3988 size_t mixedTracks = 0; 3989 size_t tracksWithEffect = 0; 3990 // counts only _active_ fast tracks 3991 size_t fastTracks = 0; 3992 uint32_t resetMask = 0; // bit mask of fast tracks that need to be reset 3993 3994 float masterVolume = mMasterVolume; 3995 bool masterMute = mMasterMute; 3996 3997 if (masterMute) { 3998 masterVolume = 0; 3999 } 4000 // Delegate master volume control to effect in output mix effect chain if needed 4001 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 4002 if (chain != 0) { 4003 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 4004 chain->setVolume_l(&v, &v); 4005 masterVolume = (float)((v + (1 << 23)) >> 24); 4006 chain.clear(); 4007 } 4008 4009 // prepare a new state to push 4010 FastMixerStateQueue *sq = NULL; 4011 FastMixerState *state = NULL; 4012 bool didModify = false; 4013 FastMixerStateQueue::block_t block = FastMixerStateQueue::BLOCK_UNTIL_PUSHED; 4014 if (mFastMixer != 0) { 4015 sq = mFastMixer->sq(); 4016 state = sq->begin(); 4017 } 4018 4019 mMixerBufferValid = false; // mMixerBuffer has no valid data until appropriate tracks found. 4020 mEffectBufferValid = false; // mEffectBuffer has no valid data until tracks found. 4021 4022 for (size_t i=0 ; i<count ; i++) { 4023 const sp<Track> t = mActiveTracks[i].promote(); 4024 if (t == 0) { 4025 continue; 4026 } 4027 4028 // this const just means the local variable doesn't change 4029 Track* const track = t.get(); 4030 4031 // process fast tracks 4032 if (track->isFastTrack()) { 4033 4034 // It's theoretically possible (though unlikely) for a fast track to be created 4035 // and then removed within the same normal mix cycle. This is not a problem, as 4036 // the track never becomes active so it's fast mixer slot is never touched. 4037 // The converse, of removing an (active) track and then creating a new track 4038 // at the identical fast mixer slot within the same normal mix cycle, 4039 // is impossible because the slot isn't marked available until the end of each cycle. 4040 int j = track->mFastIndex; 4041 ALOG_ASSERT(0 < j && j < (int)FastMixerState::sMaxFastTracks); 4042 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << j))); 4043 FastTrack *fastTrack = &state->mFastTracks[j]; 4044 4045 // Determine whether the track is currently in underrun condition, 4046 // and whether it had a recent underrun. 4047 FastTrackDump *ftDump = &mFastMixerDumpState.mTracks[j]; 4048 FastTrackUnderruns underruns = ftDump->mUnderruns; 4049 uint32_t recentFull = (underruns.mBitFields.mFull - 4050 track->mObservedUnderruns.mBitFields.mFull) & UNDERRUN_MASK; 4051 uint32_t recentPartial = (underruns.mBitFields.mPartial - 4052 track->mObservedUnderruns.mBitFields.mPartial) & UNDERRUN_MASK; 4053 uint32_t recentEmpty = (underruns.mBitFields.mEmpty - 4054 track->mObservedUnderruns.mBitFields.mEmpty) & UNDERRUN_MASK; 4055 uint32_t recentUnderruns = recentPartial + recentEmpty; 4056 track->mObservedUnderruns = underruns; 4057 // don't count underruns that occur while stopping or pausing 4058 // or stopped which can occur when flush() is called while active 4059 if (!(track->isStopping() || track->isPausing() || track->isStopped()) && 4060 recentUnderruns > 0) { 4061 // FIXME fast mixer will pull & mix partial buffers, but we count as a full underrun 4062 track->mAudioTrackServerProxy->tallyUnderrunFrames(recentUnderruns * mFrameCount); 4063 } else { 4064 track->mAudioTrackServerProxy->tallyUnderrunFrames(0); 4065 } 4066 4067 // This is similar to the state machine for normal tracks, 4068 // with a few modifications for fast tracks. 4069 bool isActive = true; 4070 switch (track->mState) { 4071 case TrackBase::STOPPING_1: 4072 // track stays active in STOPPING_1 state until first underrun 4073 if (recentUnderruns > 0 || track->isTerminated()) { 4074 track->mState = TrackBase::STOPPING_2; 4075 } 4076 break; 4077 case TrackBase::PAUSING: 4078 // ramp down is not yet implemented 4079 track->setPaused(); 4080 break; 4081 case TrackBase::RESUMING: 4082 // ramp up is not yet implemented 4083 track->mState = TrackBase::ACTIVE; 4084 break; 4085 case TrackBase::ACTIVE: 4086 if (recentFull > 0 || recentPartial > 0) { 4087 // track has provided at least some frames recently: reset retry count 4088 track->mRetryCount = kMaxTrackRetries; 4089 } 4090 if (recentUnderruns == 0) { 4091 // no recent underruns: stay active 4092 break; 4093 } 4094 // there has recently been an underrun of some kind 4095 if (track->sharedBuffer() == 0) { 4096 // were any of the recent underruns "empty" (no frames available)? 4097 if (recentEmpty == 0) { 4098 // no, then ignore the partial underruns as they are allowed indefinitely 4099 break; 4100 } 4101 // there has recently been an "empty" underrun: decrement the retry counter 4102 if (--(track->mRetryCount) > 0) { 4103 break; 4104 } 4105 // indicate to client process that the track was disabled because of underrun; 4106 // it will then automatically call start() when data is available 4107 track->disable(); 4108 // remove from active list, but state remains ACTIVE [confusing but true] 4109 isActive = false; 4110 break; 4111 } 4112 // fall through 4113 case TrackBase::STOPPING_2: 4114 case TrackBase::PAUSED: 4115 case TrackBase::STOPPED: 4116 case TrackBase::FLUSHED: // flush() while active 4117 // Check for presentation complete if track is inactive 4118 // We have consumed all the buffers of this track. 4119 // This would be incomplete if we auto-paused on underrun 4120 { 4121 size_t audioHALFrames = 4122 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 4123 int64_t framesWritten = mBytesWritten / mFrameSize; 4124 if (!(mStandby || track->presentationComplete(framesWritten, audioHALFrames))) { 4125 // track stays in active list until presentation is complete 4126 break; 4127 } 4128 } 4129 if (track->isStopping_2()) { 4130 track->mState = TrackBase::STOPPED; 4131 } 4132 if (track->isStopped()) { 4133 // Can't reset directly, as fast mixer is still polling this track 4134 // track->reset(); 4135 // So instead mark this track as needing to be reset after push with ack 4136 resetMask |= 1 << i; 4137 } 4138 isActive = false; 4139 break; 4140 case TrackBase::IDLE: 4141 default: 4142 LOG_ALWAYS_FATAL("unexpected track state %d", track->mState); 4143 } 4144 4145 if (isActive) { 4146 // was it previously inactive? 4147 if (!(state->mTrackMask & (1 << j))) { 4148 ExtendedAudioBufferProvider *eabp = track; 4149 VolumeProvider *vp = track; 4150 fastTrack->mBufferProvider = eabp; 4151 fastTrack->mVolumeProvider = vp; 4152 fastTrack->mChannelMask = track->mChannelMask; 4153 fastTrack->mFormat = track->mFormat; 4154 fastTrack->mGeneration++; 4155 state->mTrackMask |= 1 << j; 4156 didModify = true; 4157 // no acknowledgement required for newly active tracks 4158 } 4159 // cache the combined master volume and stream type volume for fast mixer; this 4160 // lacks any synchronization or barrier so VolumeProvider may read a stale value 4161 track->mCachedVolume = masterVolume * mStreamTypes[track->streamType()].volume; 4162 ++fastTracks; 4163 } else { 4164 // was it previously active? 4165 if (state->mTrackMask & (1 << j)) { 4166 fastTrack->mBufferProvider = NULL; 4167 fastTrack->mGeneration++; 4168 state->mTrackMask &= ~(1 << j); 4169 didModify = true; 4170 // If any fast tracks were removed, we must wait for acknowledgement 4171 // because we're about to decrement the last sp<> on those tracks. 4172 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 4173 } else { 4174 LOG_ALWAYS_FATAL("fast track %d should have been active; " 4175 "mState=%d, mTrackMask=%#x, recentUnderruns=%u, isShared=%d", 4176 j, track->mState, state->mTrackMask, recentUnderruns, 4177 track->sharedBuffer() != 0); 4178 } 4179 tracksToRemove->add(track); 4180 // Avoids a misleading display in dumpsys 4181 track->mObservedUnderruns.mBitFields.mMostRecent = UNDERRUN_FULL; 4182 } 4183 continue; 4184 } 4185 4186 { // local variable scope to avoid goto warning 4187 4188 audio_track_cblk_t* cblk = track->cblk(); 4189 4190 // The first time a track is added we wait 4191 // for all its buffers to be filled before processing it 4192 int name = track->name(); 4193 // make sure that we have enough frames to mix one full buffer. 4194 // enforce this condition only once to enable draining the buffer in case the client 4195 // app does not call stop() and relies on underrun to stop: 4196 // hence the test on (mMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 4197 // during last round 4198 size_t desiredFrames; 4199 const uint32_t sampleRate = track->mAudioTrackServerProxy->getSampleRate(); 4200 AudioPlaybackRate playbackRate = track->mAudioTrackServerProxy->getPlaybackRate(); 4201 4202 desiredFrames = sourceFramesNeededWithTimestretch( 4203 sampleRate, mNormalFrameCount, mSampleRate, playbackRate.mSpeed); 4204 // TODO: ONLY USED FOR LEGACY RESAMPLERS, remove when they are removed. 4205 // add frames already consumed but not yet released by the resampler 4206 // because mAudioTrackServerProxy->framesReady() will include these frames 4207 desiredFrames += mAudioMixer->getUnreleasedFrames(track->name()); 4208 4209 uint32_t minFrames = 1; 4210 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() && 4211 (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) { 4212 minFrames = desiredFrames; 4213 } 4214 4215 size_t framesReady = track->framesReady(); 4216 if (ATRACE_ENABLED()) { 4217 // I wish we had formatted trace names 4218 char traceName[16]; 4219 strcpy(traceName, "nRdy"); 4220 int name = track->name(); 4221 if (AudioMixer::TRACK0 <= name && 4222 name < (int) (AudioMixer::TRACK0 + AudioMixer::MAX_NUM_TRACKS)) { 4223 name -= AudioMixer::TRACK0; 4224 traceName[4] = (name / 10) + '0'; 4225 traceName[5] = (name % 10) + '0'; 4226 } else { 4227 traceName[4] = '?'; 4228 traceName[5] = '?'; 4229 } 4230 traceName[6] = '\0'; 4231 ATRACE_INT(traceName, framesReady); 4232 } 4233 if ((framesReady >= minFrames) && track->isReady() && 4234 !track->isPaused() && !track->isTerminated()) 4235 { 4236 ALOGVV("track %d s=%08x [OK] on thread %p", name, cblk->mServer, this); 4237 4238 mixedTracks++; 4239 4240 // track->mainBuffer() != mSinkBuffer or mMixerBuffer means 4241 // there is an effect chain connected to the track 4242 chain.clear(); 4243 if (track->mainBuffer() != mSinkBuffer && 4244 track->mainBuffer() != mMixerBuffer) { 4245 if (mEffectBufferEnabled) { 4246 mEffectBufferValid = true; // Later can set directly. 4247 } 4248 chain = getEffectChain_l(track->sessionId()); 4249 // Delegate volume control to effect in track effect chain if needed 4250 if (chain != 0) { 4251 tracksWithEffect++; 4252 } else { 4253 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on " 4254 "session %d", 4255 name, track->sessionId()); 4256 } 4257 } 4258 4259 4260 int param = AudioMixer::VOLUME; 4261 if (track->mFillingUpStatus == Track::FS_FILLED) { 4262 // no ramp for the first volume setting 4263 track->mFillingUpStatus = Track::FS_ACTIVE; 4264 if (track->mState == TrackBase::RESUMING) { 4265 track->mState = TrackBase::ACTIVE; 4266 param = AudioMixer::RAMP_VOLUME; 4267 } 4268 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 4269 // FIXME should not make a decision based on mServer 4270 } else if (cblk->mServer != 0) { 4271 // If the track is stopped before the first frame was mixed, 4272 // do not apply ramp 4273 param = AudioMixer::RAMP_VOLUME; 4274 } 4275 4276 // compute volume for this track 4277 uint32_t vl, vr; // in U8.24 integer format 4278 float vlf, vrf, vaf; // in [0.0, 1.0] float format 4279 if (track->isPausing() || mStreamTypes[track->streamType()].mute) { 4280 vl = vr = 0; 4281 vlf = vrf = vaf = 0.; 4282 if (track->isPausing()) { 4283 track->setPaused(); 4284 } 4285 } else { 4286 4287 // read original volumes with volume control 4288 float typeVolume = mStreamTypes[track->streamType()].volume; 4289 float v = masterVolume * typeVolume; 4290 sp<AudioTrackServerProxy> proxy = track->mAudioTrackServerProxy; 4291 gain_minifloat_packed_t vlr = proxy->getVolumeLR(); 4292 vlf = float_from_gain(gain_minifloat_unpack_left(vlr)); 4293 vrf = float_from_gain(gain_minifloat_unpack_right(vlr)); 4294 // track volumes come from shared memory, so can't be trusted and must be clamped 4295 if (vlf > GAIN_FLOAT_UNITY) { 4296 ALOGV("Track left volume out of range: %.3g", vlf); 4297 vlf = GAIN_FLOAT_UNITY; 4298 } 4299 if (vrf > GAIN_FLOAT_UNITY) { 4300 ALOGV("Track right volume out of range: %.3g", vrf); 4301 vrf = GAIN_FLOAT_UNITY; 4302 } 4303 // now apply the master volume and stream type volume 4304 vlf *= v; 4305 vrf *= v; 4306 // assuming master volume and stream type volume each go up to 1.0, 4307 // then derive vl and vr as U8.24 versions for the effect chain 4308 const float scaleto8_24 = MAX_GAIN_INT * MAX_GAIN_INT; 4309 vl = (uint32_t) (scaleto8_24 * vlf); 4310 vr = (uint32_t) (scaleto8_24 * vrf); 4311 // vl and vr are now in U8.24 format 4312 uint16_t sendLevel = proxy->getSendLevel_U4_12(); 4313 // send level comes from shared memory and so may be corrupt 4314 if (sendLevel > MAX_GAIN_INT) { 4315 ALOGV("Track send level out of range: %04X", sendLevel); 4316 sendLevel = MAX_GAIN_INT; 4317 } 4318 // vaf is represented as [0.0, 1.0] float by rescaling sendLevel 4319 vaf = v * sendLevel * (1. / MAX_GAIN_INT); 4320 } 4321 4322 // Delegate volume control to effect in track effect chain if needed 4323 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 4324 // Do not ramp volume if volume is controlled by effect 4325 param = AudioMixer::VOLUME; 4326 // Update remaining floating point volume levels 4327 vlf = (float)vl / (1 << 24); 4328 vrf = (float)vr / (1 << 24); 4329 track->mHasVolumeController = true; 4330 } else { 4331 // force no volume ramp when volume controller was just disabled or removed 4332 // from effect chain to avoid volume spike 4333 if (track->mHasVolumeController) { 4334 param = AudioMixer::VOLUME; 4335 } 4336 track->mHasVolumeController = false; 4337 } 4338 4339 // XXX: these things DON'T need to be done each time 4340 mAudioMixer->setBufferProvider(name, track); 4341 mAudioMixer->enable(name); 4342 4343 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, &vlf); 4344 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, &vrf); 4345 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, &vaf); 4346 mAudioMixer->setParameter( 4347 name, 4348 AudioMixer::TRACK, 4349 AudioMixer::FORMAT, (void *)track->format()); 4350 mAudioMixer->setParameter( 4351 name, 4352 AudioMixer::TRACK, 4353 AudioMixer::CHANNEL_MASK, (void *)(uintptr_t)track->channelMask()); 4354 mAudioMixer->setParameter( 4355 name, 4356 AudioMixer::TRACK, 4357 AudioMixer::MIXER_CHANNEL_MASK, (void *)(uintptr_t)mChannelMask); 4358 // limit track sample rate to 2 x output sample rate, which changes at re-configuration 4359 uint32_t maxSampleRate = mSampleRate * AUDIO_RESAMPLER_DOWN_RATIO_MAX; 4360 uint32_t reqSampleRate = track->mAudioTrackServerProxy->getSampleRate(); 4361 if (reqSampleRate == 0) { 4362 reqSampleRate = mSampleRate; 4363 } else if (reqSampleRate > maxSampleRate) { 4364 reqSampleRate = maxSampleRate; 4365 } 4366 mAudioMixer->setParameter( 4367 name, 4368 AudioMixer::RESAMPLE, 4369 AudioMixer::SAMPLE_RATE, 4370 (void *)(uintptr_t)reqSampleRate); 4371 4372 AudioPlaybackRate playbackRate = track->mAudioTrackServerProxy->getPlaybackRate(); 4373 mAudioMixer->setParameter( 4374 name, 4375 AudioMixer::TIMESTRETCH, 4376 AudioMixer::PLAYBACK_RATE, 4377 &playbackRate); 4378 4379 /* 4380 * Select the appropriate output buffer for the track. 4381 * 4382 * Tracks with effects go into their own effects chain buffer 4383 * and from there into either mEffectBuffer or mSinkBuffer. 4384 * 4385 * Other tracks can use mMixerBuffer for higher precision 4386 * channel accumulation. If this buffer is enabled 4387 * (mMixerBufferEnabled true), then selected tracks will accumulate 4388 * into it. 4389 * 4390 */ 4391 if (mMixerBufferEnabled 4392 && (track->mainBuffer() == mSinkBuffer 4393 || track->mainBuffer() == mMixerBuffer)) { 4394 mAudioMixer->setParameter( 4395 name, 4396 AudioMixer::TRACK, 4397 AudioMixer::MIXER_FORMAT, (void *)mMixerBufferFormat); 4398 mAudioMixer->setParameter( 4399 name, 4400 AudioMixer::TRACK, 4401 AudioMixer::MAIN_BUFFER, (void *)mMixerBuffer); 4402 // TODO: override track->mainBuffer()? 4403 mMixerBufferValid = true; 4404 } else { 4405 mAudioMixer->setParameter( 4406 name, 4407 AudioMixer::TRACK, 4408 AudioMixer::MIXER_FORMAT, (void *)AUDIO_FORMAT_PCM_16_BIT); 4409 mAudioMixer->setParameter( 4410 name, 4411 AudioMixer::TRACK, 4412 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 4413 } 4414 mAudioMixer->setParameter( 4415 name, 4416 AudioMixer::TRACK, 4417 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 4418 4419 // reset retry count 4420 track->mRetryCount = kMaxTrackRetries; 4421 4422 // If one track is ready, set the mixer ready if: 4423 // - the mixer was not ready during previous round OR 4424 // - no other track is not ready 4425 if (mMixerStatusIgnoringFastTracks != MIXER_TRACKS_READY || 4426 mixerStatus != MIXER_TRACKS_ENABLED) { 4427 mixerStatus = MIXER_TRACKS_READY; 4428 } 4429 } else { 4430 if (framesReady < desiredFrames && !track->isStopped() && !track->isPaused()) { 4431 ALOGV("track(%p) underrun, framesReady(%zu) < framesDesired(%zd)", 4432 track, framesReady, desiredFrames); 4433 track->mAudioTrackServerProxy->tallyUnderrunFrames(desiredFrames); 4434 } else { 4435 track->mAudioTrackServerProxy->tallyUnderrunFrames(0); 4436 } 4437 4438 // clear effect chain input buffer if an active track underruns to avoid sending 4439 // previous audio buffer again to effects 4440 chain = getEffectChain_l(track->sessionId()); 4441 if (chain != 0) { 4442 chain->clearInputBuffer(); 4443 } 4444 4445 ALOGVV("track %d s=%08x [NOT READY] on thread %p", name, cblk->mServer, this); 4446 if ((track->sharedBuffer() != 0) || track->isTerminated() || 4447 track->isStopped() || track->isPaused()) { 4448 // We have consumed all the buffers of this track. 4449 // Remove it from the list of active tracks. 4450 // TODO: use actual buffer filling status instead of latency when available from 4451 // audio HAL 4452 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 4453 int64_t framesWritten = mBytesWritten / mFrameSize; 4454 if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) { 4455 if (track->isStopped()) { 4456 track->reset(); 4457 } 4458 tracksToRemove->add(track); 4459 } 4460 } else { 4461 // No buffers for this track. Give it a few chances to 4462 // fill a buffer, then remove it from active list. 4463 if (--(track->mRetryCount) <= 0) { 4464 ALOGI("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 4465 tracksToRemove->add(track); 4466 // indicate to client process that the track was disabled because of underrun; 4467 // it will then automatically call start() when data is available 4468 track->disable(); 4469 // If one track is not ready, mark the mixer also not ready if: 4470 // - the mixer was ready during previous round OR 4471 // - no other track is ready 4472 } else if (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY || 4473 mixerStatus != MIXER_TRACKS_READY) { 4474 mixerStatus = MIXER_TRACKS_ENABLED; 4475 } 4476 } 4477 mAudioMixer->disable(name); 4478 } 4479 4480 } // local variable scope to avoid goto warning 4481 4482 } 4483 4484 // Push the new FastMixer state if necessary 4485 bool pauseAudioWatchdog = false; 4486 if (didModify) { 4487 state->mFastTracksGen++; 4488 // if the fast mixer was active, but now there are no fast tracks, then put it in cold idle 4489 if (kUseFastMixer == FastMixer_Dynamic && 4490 state->mCommand == FastMixerState::MIX_WRITE && state->mTrackMask <= 1) { 4491 state->mCommand = FastMixerState::COLD_IDLE; 4492 state->mColdFutexAddr = &mFastMixerFutex; 4493 state->mColdGen++; 4494 mFastMixerFutex = 0; 4495 if (kUseFastMixer == FastMixer_Dynamic) { 4496 mNormalSink = mOutputSink; 4497 } 4498 // If we go into cold idle, need to wait for acknowledgement 4499 // so that fast mixer stops doing I/O. 4500 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 4501 pauseAudioWatchdog = true; 4502 } 4503 } 4504 if (sq != NULL) { 4505 sq->end(didModify); 4506 sq->push(block); 4507 } 4508#ifdef AUDIO_WATCHDOG 4509 if (pauseAudioWatchdog && mAudioWatchdog != 0) { 4510 mAudioWatchdog->pause(); 4511 } 4512#endif 4513 4514 // Now perform the deferred reset on fast tracks that have stopped 4515 while (resetMask != 0) { 4516 size_t i = __builtin_ctz(resetMask); 4517 ALOG_ASSERT(i < count); 4518 resetMask &= ~(1 << i); 4519 sp<Track> t = mActiveTracks[i].promote(); 4520 if (t == 0) { 4521 continue; 4522 } 4523 Track* track = t.get(); 4524 ALOG_ASSERT(track->isFastTrack() && track->isStopped()); 4525 track->reset(); 4526 } 4527 4528 // remove all the tracks that need to be... 4529 removeTracks_l(*tracksToRemove); 4530 4531 if (getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX) != 0) { 4532 mEffectBufferValid = true; 4533 } 4534 4535 if (mEffectBufferValid) { 4536 // as long as there are effects we should clear the effects buffer, to avoid 4537 // passing a non-clean buffer to the effect chain 4538 memset(mEffectBuffer, 0, mEffectBufferSize); 4539 } 4540 // sink or mix buffer must be cleared if all tracks are connected to an 4541 // effect chain as in this case the mixer will not write to the sink or mix buffer 4542 // and track effects will accumulate into it 4543 if ((mBytesRemaining == 0) && ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 4544 (mixedTracks == 0 && fastTracks > 0))) { 4545 // FIXME as a performance optimization, should remember previous zero status 4546 if (mMixerBufferValid) { 4547 memset(mMixerBuffer, 0, mMixerBufferSize); 4548 // TODO: In testing, mSinkBuffer below need not be cleared because 4549 // the PlaybackThread::threadLoop() copies mMixerBuffer into mSinkBuffer 4550 // after mixing. 4551 // 4552 // To enforce this guarantee: 4553 // ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 4554 // (mixedTracks == 0 && fastTracks > 0)) 4555 // must imply MIXER_TRACKS_READY. 4556 // Later, we may clear buffers regardless, and skip much of this logic. 4557 } 4558 // FIXME as a performance optimization, should remember previous zero status 4559 memset(mSinkBuffer, 0, mNormalFrameCount * mFrameSize); 4560 } 4561 4562 // if any fast tracks, then status is ready 4563 mMixerStatusIgnoringFastTracks = mixerStatus; 4564 if (fastTracks > 0) { 4565 mixerStatus = MIXER_TRACKS_READY; 4566 } 4567 return mixerStatus; 4568} 4569 4570// getTrackName_l() must be called with ThreadBase::mLock held 4571int AudioFlinger::MixerThread::getTrackName_l(audio_channel_mask_t channelMask, 4572 audio_format_t format, audio_session_t sessionId) 4573{ 4574 return mAudioMixer->getTrackName(channelMask, format, sessionId); 4575} 4576 4577// deleteTrackName_l() must be called with ThreadBase::mLock held 4578void AudioFlinger::MixerThread::deleteTrackName_l(int name) 4579{ 4580 ALOGV("remove track (%d) and delete from mixer", name); 4581 mAudioMixer->deleteTrackName(name); 4582} 4583 4584// checkForNewParameter_l() must be called with ThreadBase::mLock held 4585bool AudioFlinger::MixerThread::checkForNewParameter_l(const String8& keyValuePair, 4586 status_t& status) 4587{ 4588 bool reconfig = false; 4589 bool a2dpDeviceChanged = false; 4590 4591 status = NO_ERROR; 4592 4593 AutoPark<FastMixer> park(mFastMixer); 4594 4595 AudioParameter param = AudioParameter(keyValuePair); 4596 int value; 4597 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 4598 reconfig = true; 4599 } 4600 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 4601 if (!isValidPcmSinkFormat((audio_format_t) value)) { 4602 status = BAD_VALUE; 4603 } else { 4604 // no need to save value, since it's constant 4605 reconfig = true; 4606 } 4607 } 4608 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 4609 if (!isValidPcmSinkChannelMask((audio_channel_mask_t) value)) { 4610 status = BAD_VALUE; 4611 } else { 4612 // no need to save value, since it's constant 4613 reconfig = true; 4614 } 4615 } 4616 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 4617 // do not accept frame count changes if tracks are open as the track buffer 4618 // size depends on frame count and correct behavior would not be guaranteed 4619 // if frame count is changed after track creation 4620 if (!mTracks.isEmpty()) { 4621 status = INVALID_OPERATION; 4622 } else { 4623 reconfig = true; 4624 } 4625 } 4626 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 4627#ifdef ADD_BATTERY_DATA 4628 // when changing the audio output device, call addBatteryData to notify 4629 // the change 4630 if (mOutDevice != value) { 4631 uint32_t params = 0; 4632 // check whether speaker is on 4633 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 4634 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 4635 } 4636 4637 audio_devices_t deviceWithoutSpeaker 4638 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 4639 // check if any other device (except speaker) is on 4640 if (value & deviceWithoutSpeaker) { 4641 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 4642 } 4643 4644 if (params != 0) { 4645 addBatteryData(params); 4646 } 4647 } 4648#endif 4649 4650 // forward device change to effects that have requested to be 4651 // aware of attached audio device. 4652 if (value != AUDIO_DEVICE_NONE) { 4653 a2dpDeviceChanged = 4654 (mOutDevice & AUDIO_DEVICE_OUT_ALL_A2DP) != (value & AUDIO_DEVICE_OUT_ALL_A2DP); 4655 mOutDevice = value; 4656 for (size_t i = 0; i < mEffectChains.size(); i++) { 4657 mEffectChains[i]->setDevice_l(mOutDevice); 4658 } 4659 } 4660 } 4661 4662 if (status == NO_ERROR) { 4663 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 4664 keyValuePair.string()); 4665 if (!mStandby && status == INVALID_OPERATION) { 4666 mOutput->standby(); 4667 mStandby = true; 4668 mBytesWritten = 0; 4669 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 4670 keyValuePair.string()); 4671 } 4672 if (status == NO_ERROR && reconfig) { 4673 readOutputParameters_l(); 4674 delete mAudioMixer; 4675 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 4676 for (size_t i = 0; i < mTracks.size() ; i++) { 4677 int name = getTrackName_l(mTracks[i]->mChannelMask, 4678 mTracks[i]->mFormat, mTracks[i]->mSessionId); 4679 if (name < 0) { 4680 break; 4681 } 4682 mTracks[i]->mName = name; 4683 } 4684 sendIoConfigEvent_l(AUDIO_OUTPUT_CONFIG_CHANGED); 4685 } 4686 } 4687 4688 return reconfig || a2dpDeviceChanged; 4689} 4690 4691 4692void AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 4693{ 4694 PlaybackThread::dumpInternals(fd, args); 4695 dprintf(fd, " Thread throttle time (msecs): %u\n", mThreadThrottleTimeMs); 4696 dprintf(fd, " AudioMixer tracks: 0x%08x\n", mAudioMixer->trackNames()); 4697 dprintf(fd, " Master mono: %s\n", mMasterMono ? "on" : "off"); 4698 4699 // Make a non-atomic copy of fast mixer dump state so it won't change underneath us 4700 // while we are dumping it. It may be inconsistent, but it won't mutate! 4701 // This is a large object so we place it on the heap. 4702 // FIXME 25972958: Need an intelligent copy constructor that does not touch unused pages. 4703 const FastMixerDumpState *copy = new FastMixerDumpState(mFastMixerDumpState); 4704 copy->dump(fd); 4705 delete copy; 4706 4707#ifdef STATE_QUEUE_DUMP 4708 // Similar for state queue 4709 StateQueueObserverDump observerCopy = mStateQueueObserverDump; 4710 observerCopy.dump(fd); 4711 StateQueueMutatorDump mutatorCopy = mStateQueueMutatorDump; 4712 mutatorCopy.dump(fd); 4713#endif 4714 4715#ifdef TEE_SINK 4716 // Write the tee output to a .wav file 4717 dumpTee(fd, mTeeSource, mId); 4718#endif 4719 4720#ifdef AUDIO_WATCHDOG 4721 if (mAudioWatchdog != 0) { 4722 // Make a non-atomic copy of audio watchdog dump so it won't change underneath us 4723 AudioWatchdogDump wdCopy = mAudioWatchdogDump; 4724 wdCopy.dump(fd); 4725 } 4726#endif 4727} 4728 4729uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const 4730{ 4731 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000) / 2; 4732} 4733 4734uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() const 4735{ 4736 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000); 4737} 4738 4739void AudioFlinger::MixerThread::cacheParameters_l() 4740{ 4741 PlaybackThread::cacheParameters_l(); 4742 4743 // FIXME: Relaxed timing because of a certain device that can't meet latency 4744 // Should be reduced to 2x after the vendor fixes the driver issue 4745 // increase threshold again due to low power audio mode. The way this warning 4746 // threshold is calculated and its usefulness should be reconsidered anyway. 4747 maxPeriod = seconds(mNormalFrameCount) / mSampleRate * 15; 4748} 4749 4750// ---------------------------------------------------------------------------- 4751 4752AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 4753 AudioStreamOut* output, audio_io_handle_t id, audio_devices_t device, bool systemReady) 4754 : PlaybackThread(audioFlinger, output, id, device, DIRECT, systemReady) 4755 // mLeftVolFloat, mRightVolFloat 4756{ 4757} 4758 4759AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 4760 AudioStreamOut* output, audio_io_handle_t id, uint32_t device, 4761 ThreadBase::type_t type, bool systemReady) 4762 : PlaybackThread(audioFlinger, output, id, device, type, systemReady) 4763 // mLeftVolFloat, mRightVolFloat 4764{ 4765} 4766 4767AudioFlinger::DirectOutputThread::~DirectOutputThread() 4768{ 4769} 4770 4771void AudioFlinger::DirectOutputThread::processVolume_l(Track *track, bool lastTrack) 4772{ 4773 float left, right; 4774 4775 if (mMasterMute || mStreamTypes[track->streamType()].mute) { 4776 left = right = 0; 4777 } else { 4778 float typeVolume = mStreamTypes[track->streamType()].volume; 4779 float v = mMasterVolume * typeVolume; 4780 sp<AudioTrackServerProxy> proxy = track->mAudioTrackServerProxy; 4781 gain_minifloat_packed_t vlr = proxy->getVolumeLR(); 4782 left = float_from_gain(gain_minifloat_unpack_left(vlr)); 4783 if (left > GAIN_FLOAT_UNITY) { 4784 left = GAIN_FLOAT_UNITY; 4785 } 4786 left *= v; 4787 right = float_from_gain(gain_minifloat_unpack_right(vlr)); 4788 if (right > GAIN_FLOAT_UNITY) { 4789 right = GAIN_FLOAT_UNITY; 4790 } 4791 right *= v; 4792 } 4793 4794 if (lastTrack) { 4795 if (left != mLeftVolFloat || right != mRightVolFloat) { 4796 mLeftVolFloat = left; 4797 mRightVolFloat = right; 4798 4799 // Convert volumes from float to 8.24 4800 uint32_t vl = (uint32_t)(left * (1 << 24)); 4801 uint32_t vr = (uint32_t)(right * (1 << 24)); 4802 4803 // Delegate volume control to effect in track effect chain if needed 4804 // only one effect chain can be present on DirectOutputThread, so if 4805 // there is one, the track is connected to it 4806 if (!mEffectChains.isEmpty()) { 4807 mEffectChains[0]->setVolume_l(&vl, &vr); 4808 left = (float)vl / (1 << 24); 4809 right = (float)vr / (1 << 24); 4810 } 4811 if (mOutput->stream->set_volume) { 4812 mOutput->stream->set_volume(mOutput->stream, left, right); 4813 } 4814 } 4815 } 4816} 4817 4818void AudioFlinger::DirectOutputThread::onAddNewTrack_l() 4819{ 4820 sp<Track> previousTrack = mPreviousTrack.promote(); 4821 sp<Track> latestTrack = mLatestActiveTrack.promote(); 4822 4823 if (previousTrack != 0 && latestTrack != 0) { 4824 if (mType == DIRECT) { 4825 if (previousTrack.get() != latestTrack.get()) { 4826 mFlushPending = true; 4827 } 4828 } else /* mType == OFFLOAD */ { 4829 if (previousTrack->sessionId() != latestTrack->sessionId()) { 4830 mFlushPending = true; 4831 } 4832 } 4833 } 4834 PlaybackThread::onAddNewTrack_l(); 4835} 4836 4837AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l( 4838 Vector< sp<Track> > *tracksToRemove 4839) 4840{ 4841 size_t count = mActiveTracks.size(); 4842 mixer_state mixerStatus = MIXER_IDLE; 4843 bool doHwPause = false; 4844 bool doHwResume = false; 4845 4846 // find out which tracks need to be processed 4847 for (size_t i = 0; i < count; i++) { 4848 sp<Track> t = mActiveTracks[i].promote(); 4849 // The track died recently 4850 if (t == 0) { 4851 continue; 4852 } 4853 4854 if (t->isInvalid()) { 4855 ALOGW("An invalidated track shouldn't be in active list"); 4856 tracksToRemove->add(t); 4857 continue; 4858 } 4859 4860 Track* const track = t.get(); 4861#ifdef VERY_VERY_VERBOSE_LOGGING 4862 audio_track_cblk_t* cblk = track->cblk(); 4863#endif 4864 // Only consider last track started for volume and mixer state control. 4865 // In theory an older track could underrun and restart after the new one starts 4866 // but as we only care about the transition phase between two tracks on a 4867 // direct output, it is not a problem to ignore the underrun case. 4868 sp<Track> l = mLatestActiveTrack.promote(); 4869 bool last = l.get() == track; 4870 4871 if (track->isPausing()) { 4872 track->setPaused(); 4873 if (mHwSupportsPause && last && !mHwPaused) { 4874 doHwPause = true; 4875 mHwPaused = true; 4876 } 4877 tracksToRemove->add(track); 4878 } else if (track->isFlushPending()) { 4879 track->flushAck(); 4880 if (last) { 4881 mFlushPending = true; 4882 } 4883 } else if (track->isResumePending()) { 4884 track->resumeAck(); 4885 if (last) { 4886 mLeftVolFloat = mRightVolFloat = -1.0; 4887 if (mHwPaused) { 4888 doHwResume = true; 4889 mHwPaused = false; 4890 } 4891 } 4892 } 4893 4894 // The first time a track is added we wait 4895 // for all its buffers to be filled before processing it. 4896 // Allow draining the buffer in case the client 4897 // app does not call stop() and relies on underrun to stop: 4898 // hence the test on (track->mRetryCount > 1). 4899 // If retryCount<=1 then track is about to underrun and be removed. 4900 // Do not use a high threshold for compressed audio. 4901 uint32_t minFrames; 4902 if ((track->sharedBuffer() == 0) && !track->isStopping_1() && !track->isPausing() 4903 && (track->mRetryCount > 1) && audio_has_proportional_frames(mFormat)) { 4904 minFrames = mNormalFrameCount; 4905 } else { 4906 minFrames = 1; 4907 } 4908 4909 if ((track->framesReady() >= minFrames) && track->isReady() && !track->isPaused() && 4910 !track->isStopping_2() && !track->isStopped()) 4911 { 4912 ALOGVV("track %d s=%08x [OK]", track->name(), cblk->mServer); 4913 4914 if (track->mFillingUpStatus == Track::FS_FILLED) { 4915 track->mFillingUpStatus = Track::FS_ACTIVE; 4916 if (last) { 4917 // make sure processVolume_l() will apply new volume even if 0 4918 mLeftVolFloat = mRightVolFloat = -1.0; 4919 } 4920 if (!mHwSupportsPause) { 4921 track->resumeAck(); 4922 } 4923 } 4924 4925 // compute volume for this track 4926 processVolume_l(track, last); 4927 if (last) { 4928 sp<Track> previousTrack = mPreviousTrack.promote(); 4929 if (previousTrack != 0) { 4930 if (track != previousTrack.get()) { 4931 // Flush any data still being written from last track 4932 mBytesRemaining = 0; 4933 // Invalidate previous track to force a seek when resuming. 4934 previousTrack->invalidate(); 4935 } 4936 } 4937 mPreviousTrack = track; 4938 4939 // reset retry count 4940 track->mRetryCount = kMaxTrackRetriesDirect; 4941 mActiveTrack = t; 4942 mixerStatus = MIXER_TRACKS_READY; 4943 if (mHwPaused) { 4944 doHwResume = true; 4945 mHwPaused = false; 4946 } 4947 } 4948 } else { 4949 // clear effect chain input buffer if the last active track started underruns 4950 // to avoid sending previous audio buffer again to effects 4951 if (!mEffectChains.isEmpty() && last) { 4952 mEffectChains[0]->clearInputBuffer(); 4953 } 4954 if (track->isStopping_1()) { 4955 track->mState = TrackBase::STOPPING_2; 4956 if (last && mHwPaused) { 4957 doHwResume = true; 4958 mHwPaused = false; 4959 } 4960 } 4961 if ((track->sharedBuffer() != 0) || track->isStopped() || 4962 track->isStopping_2() || track->isPaused()) { 4963 // We have consumed all the buffers of this track. 4964 // Remove it from the list of active tracks. 4965 size_t audioHALFrames; 4966 if (audio_has_proportional_frames(mFormat)) { 4967 audioHALFrames = (latency_l() * mSampleRate) / 1000; 4968 } else { 4969 audioHALFrames = 0; 4970 } 4971 4972 int64_t framesWritten = mBytesWritten / mFrameSize; 4973 if (mStandby || !last || 4974 track->presentationComplete(framesWritten, audioHALFrames)) { 4975 if (track->isStopping_2()) { 4976 track->mState = TrackBase::STOPPED; 4977 } 4978 if (track->isStopped()) { 4979 track->reset(); 4980 } 4981 tracksToRemove->add(track); 4982 } 4983 } else { 4984 // No buffers for this track. Give it a few chances to 4985 // fill a buffer, then remove it from active list. 4986 // Only consider last track started for mixer state control 4987 if (--(track->mRetryCount) <= 0) { 4988 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 4989 tracksToRemove->add(track); 4990 // indicate to client process that the track was disabled because of underrun; 4991 // it will then automatically call start() when data is available 4992 track->disable(); 4993 } else if (last) { 4994 ALOGW("pause because of UNDERRUN, framesReady = %zu," 4995 "minFrames = %u, mFormat = %#x", 4996 track->framesReady(), minFrames, mFormat); 4997 mixerStatus = MIXER_TRACKS_ENABLED; 4998 if (mHwSupportsPause && !mHwPaused && !mStandby) { 4999 doHwPause = true; 5000 mHwPaused = true; 5001 } 5002 } 5003 } 5004 } 5005 } 5006 5007 // if an active track did not command a flush, check for pending flush on stopped tracks 5008 if (!mFlushPending) { 5009 for (size_t i = 0; i < mTracks.size(); i++) { 5010 if (mTracks[i]->isFlushPending()) { 5011 mTracks[i]->flushAck(); 5012 mFlushPending = true; 5013 } 5014 } 5015 } 5016 5017 // make sure the pause/flush/resume sequence is executed in the right order. 5018 // If a flush is pending and a track is active but the HW is not paused, force a HW pause 5019 // before flush and then resume HW. This can happen in case of pause/flush/resume 5020 // if resume is received before pause is executed. 5021 if (mHwSupportsPause && !mStandby && 5022 (doHwPause || (mFlushPending && !mHwPaused && (count != 0)))) { 5023 mOutput->stream->pause(mOutput->stream); 5024 } 5025 if (mFlushPending) { 5026 flushHw_l(); 5027 } 5028 if (mHwSupportsPause && !mStandby && doHwResume) { 5029 mOutput->stream->resume(mOutput->stream); 5030 } 5031 // remove all the tracks that need to be... 5032 removeTracks_l(*tracksToRemove); 5033 5034 return mixerStatus; 5035} 5036 5037void AudioFlinger::DirectOutputThread::threadLoop_mix() 5038{ 5039 size_t frameCount = mFrameCount; 5040 int8_t *curBuf = (int8_t *)mSinkBuffer; 5041 // output audio to hardware 5042 while (frameCount) { 5043 AudioBufferProvider::Buffer buffer; 5044 buffer.frameCount = frameCount; 5045 status_t status = mActiveTrack->getNextBuffer(&buffer); 5046 if (status != NO_ERROR || buffer.raw == NULL) { 5047 // no need to pad with 0 for compressed audio 5048 if (audio_has_proportional_frames(mFormat)) { 5049 memset(curBuf, 0, frameCount * mFrameSize); 5050 } 5051 break; 5052 } 5053 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 5054 frameCount -= buffer.frameCount; 5055 curBuf += buffer.frameCount * mFrameSize; 5056 mActiveTrack->releaseBuffer(&buffer); 5057 } 5058 mCurrentWriteLength = curBuf - (int8_t *)mSinkBuffer; 5059 mSleepTimeUs = 0; 5060 mStandbyTimeNs = systemTime() + mStandbyDelayNs; 5061 mActiveTrack.clear(); 5062} 5063 5064void AudioFlinger::DirectOutputThread::threadLoop_sleepTime() 5065{ 5066 // do not write to HAL when paused 5067 if (mHwPaused || (usesHwAvSync() && mStandby)) { 5068 mSleepTimeUs = mIdleSleepTimeUs; 5069 return; 5070 } 5071 if (mSleepTimeUs == 0) { 5072 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 5073 mSleepTimeUs = mActiveSleepTimeUs; 5074 } else { 5075 mSleepTimeUs = mIdleSleepTimeUs; 5076 } 5077 } else if (mBytesWritten != 0 && audio_has_proportional_frames(mFormat)) { 5078 memset(mSinkBuffer, 0, mFrameCount * mFrameSize); 5079 mSleepTimeUs = 0; 5080 } 5081} 5082 5083void AudioFlinger::DirectOutputThread::threadLoop_exit() 5084{ 5085 { 5086 Mutex::Autolock _l(mLock); 5087 for (size_t i = 0; i < mTracks.size(); i++) { 5088 if (mTracks[i]->isFlushPending()) { 5089 mTracks[i]->flushAck(); 5090 mFlushPending = true; 5091 } 5092 } 5093 if (mFlushPending) { 5094 flushHw_l(); 5095 } 5096 } 5097 PlaybackThread::threadLoop_exit(); 5098} 5099 5100// must be called with thread mutex locked 5101bool AudioFlinger::DirectOutputThread::shouldStandby_l() 5102{ 5103 bool trackPaused = false; 5104 bool trackStopped = false; 5105 5106 if ((mType == DIRECT) && audio_is_linear_pcm(mFormat) && !usesHwAvSync()) { 5107 return !mStandby; 5108 } 5109 5110 // do not put the HAL in standby when paused. AwesomePlayer clear the offloaded AudioTrack 5111 // after a timeout and we will enter standby then. 5112 if (mTracks.size() > 0) { 5113 trackPaused = mTracks[mTracks.size() - 1]->isPaused(); 5114 trackStopped = mTracks[mTracks.size() - 1]->isStopped() || 5115 mTracks[mTracks.size() - 1]->mState == TrackBase::IDLE; 5116 } 5117 5118 return !mStandby && !(trackPaused || (mHwPaused && !trackStopped)); 5119} 5120 5121// getTrackName_l() must be called with ThreadBase::mLock held 5122int AudioFlinger::DirectOutputThread::getTrackName_l(audio_channel_mask_t channelMask __unused, 5123 audio_format_t format __unused, audio_session_t sessionId __unused) 5124{ 5125 return 0; 5126} 5127 5128// deleteTrackName_l() must be called with ThreadBase::mLock held 5129void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name __unused) 5130{ 5131} 5132 5133// checkForNewParameter_l() must be called with ThreadBase::mLock held 5134bool AudioFlinger::DirectOutputThread::checkForNewParameter_l(const String8& keyValuePair, 5135 status_t& status) 5136{ 5137 bool reconfig = false; 5138 bool a2dpDeviceChanged = false; 5139 5140 status = NO_ERROR; 5141 5142 AudioParameter param = AudioParameter(keyValuePair); 5143 int value; 5144 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 5145 // forward device change to effects that have requested to be 5146 // aware of attached audio device. 5147 if (value != AUDIO_DEVICE_NONE) { 5148 a2dpDeviceChanged = 5149 (mOutDevice & AUDIO_DEVICE_OUT_ALL_A2DP) != (value & AUDIO_DEVICE_OUT_ALL_A2DP); 5150 mOutDevice = value; 5151 for (size_t i = 0; i < mEffectChains.size(); i++) { 5152 mEffectChains[i]->setDevice_l(mOutDevice); 5153 } 5154 } 5155 } 5156 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 5157 // do not accept frame count changes if tracks are open as the track buffer 5158 // size depends on frame count and correct behavior would not be garantied 5159 // if frame count is changed after track creation 5160 if (!mTracks.isEmpty()) { 5161 status = INVALID_OPERATION; 5162 } else { 5163 reconfig = true; 5164 } 5165 } 5166 if (status == NO_ERROR) { 5167 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 5168 keyValuePair.string()); 5169 if (!mStandby && status == INVALID_OPERATION) { 5170 mOutput->standby(); 5171 mStandby = true; 5172 mBytesWritten = 0; 5173 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 5174 keyValuePair.string()); 5175 } 5176 if (status == NO_ERROR && reconfig) { 5177 readOutputParameters_l(); 5178 sendIoConfigEvent_l(AUDIO_OUTPUT_CONFIG_CHANGED); 5179 } 5180 } 5181 5182 return reconfig || a2dpDeviceChanged; 5183} 5184 5185uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() const 5186{ 5187 uint32_t time; 5188 if (audio_has_proportional_frames(mFormat)) { 5189 time = PlaybackThread::activeSleepTimeUs(); 5190 } else { 5191 time = kDirectMinSleepTimeUs; 5192 } 5193 return time; 5194} 5195 5196uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() const 5197{ 5198 uint32_t time; 5199 if (audio_has_proportional_frames(mFormat)) { 5200 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 5201 } else { 5202 time = kDirectMinSleepTimeUs; 5203 } 5204 return time; 5205} 5206 5207uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() const 5208{ 5209 uint32_t time; 5210 if (audio_has_proportional_frames(mFormat)) { 5211 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 5212 } else { 5213 time = kDirectMinSleepTimeUs; 5214 } 5215 return time; 5216} 5217 5218void AudioFlinger::DirectOutputThread::cacheParameters_l() 5219{ 5220 PlaybackThread::cacheParameters_l(); 5221 5222 // use shorter standby delay as on normal output to release 5223 // hardware resources as soon as possible 5224 // no delay on outputs with HW A/V sync 5225 if (usesHwAvSync()) { 5226 mStandbyDelayNs = 0; 5227 } else if ((mType == OFFLOAD) && !audio_has_proportional_frames(mFormat)) { 5228 mStandbyDelayNs = kOffloadStandbyDelayNs; 5229 } else { 5230 mStandbyDelayNs = microseconds(mActiveSleepTimeUs*2); 5231 } 5232} 5233 5234void AudioFlinger::DirectOutputThread::flushHw_l() 5235{ 5236 mOutput->flush(); 5237 mHwPaused = false; 5238 mFlushPending = false; 5239} 5240 5241// ---------------------------------------------------------------------------- 5242 5243AudioFlinger::AsyncCallbackThread::AsyncCallbackThread( 5244 const wp<AudioFlinger::PlaybackThread>& playbackThread) 5245 : Thread(false /*canCallJava*/), 5246 mPlaybackThread(playbackThread), 5247 mWriteAckSequence(0), 5248 mDrainSequence(0), 5249 mAsyncError(false) 5250{ 5251} 5252 5253AudioFlinger::AsyncCallbackThread::~AsyncCallbackThread() 5254{ 5255} 5256 5257void AudioFlinger::AsyncCallbackThread::onFirstRef() 5258{ 5259 run("Offload Cbk", ANDROID_PRIORITY_URGENT_AUDIO); 5260} 5261 5262bool AudioFlinger::AsyncCallbackThread::threadLoop() 5263{ 5264 while (!exitPending()) { 5265 uint32_t writeAckSequence; 5266 uint32_t drainSequence; 5267 bool asyncError; 5268 5269 { 5270 Mutex::Autolock _l(mLock); 5271 while (!((mWriteAckSequence & 1) || 5272 (mDrainSequence & 1) || 5273 mAsyncError || 5274 exitPending())) { 5275 mWaitWorkCV.wait(mLock); 5276 } 5277 5278 if (exitPending()) { 5279 break; 5280 } 5281 ALOGV("AsyncCallbackThread mWriteAckSequence %d mDrainSequence %d", 5282 mWriteAckSequence, mDrainSequence); 5283 writeAckSequence = mWriteAckSequence; 5284 mWriteAckSequence &= ~1; 5285 drainSequence = mDrainSequence; 5286 mDrainSequence &= ~1; 5287 asyncError = mAsyncError; 5288 mAsyncError = false; 5289 } 5290 { 5291 sp<AudioFlinger::PlaybackThread> playbackThread = mPlaybackThread.promote(); 5292 if (playbackThread != 0) { 5293 if (writeAckSequence & 1) { 5294 playbackThread->resetWriteBlocked(writeAckSequence >> 1); 5295 } 5296 if (drainSequence & 1) { 5297 playbackThread->resetDraining(drainSequence >> 1); 5298 } 5299 if (asyncError) { 5300 playbackThread->onAsyncError(); 5301 } 5302 } 5303 } 5304 } 5305 return false; 5306} 5307 5308void AudioFlinger::AsyncCallbackThread::exit() 5309{ 5310 ALOGV("AsyncCallbackThread::exit"); 5311 Mutex::Autolock _l(mLock); 5312 requestExit(); 5313 mWaitWorkCV.broadcast(); 5314} 5315 5316void AudioFlinger::AsyncCallbackThread::setWriteBlocked(uint32_t sequence) 5317{ 5318 Mutex::Autolock _l(mLock); 5319 // bit 0 is cleared 5320 mWriteAckSequence = sequence << 1; 5321} 5322 5323void AudioFlinger::AsyncCallbackThread::resetWriteBlocked() 5324{ 5325 Mutex::Autolock _l(mLock); 5326 // ignore unexpected callbacks 5327 if (mWriteAckSequence & 2) { 5328 mWriteAckSequence |= 1; 5329 mWaitWorkCV.signal(); 5330 } 5331} 5332 5333void AudioFlinger::AsyncCallbackThread::setDraining(uint32_t sequence) 5334{ 5335 Mutex::Autolock _l(mLock); 5336 // bit 0 is cleared 5337 mDrainSequence = sequence << 1; 5338} 5339 5340void AudioFlinger::AsyncCallbackThread::resetDraining() 5341{ 5342 Mutex::Autolock _l(mLock); 5343 // ignore unexpected callbacks 5344 if (mDrainSequence & 2) { 5345 mDrainSequence |= 1; 5346 mWaitWorkCV.signal(); 5347 } 5348} 5349 5350void AudioFlinger::AsyncCallbackThread::setAsyncError() 5351{ 5352 Mutex::Autolock _l(mLock); 5353 mAsyncError = true; 5354 mWaitWorkCV.signal(); 5355} 5356 5357 5358// ---------------------------------------------------------------------------- 5359AudioFlinger::OffloadThread::OffloadThread(const sp<AudioFlinger>& audioFlinger, 5360 AudioStreamOut* output, audio_io_handle_t id, uint32_t device, bool systemReady) 5361 : DirectOutputThread(audioFlinger, output, id, device, OFFLOAD, systemReady), 5362 mPausedWriteLength(0), mPausedBytesRemaining(0), mKeepWakeLock(true), 5363 mOffloadUnderrunPosition(~0LL) 5364{ 5365 //FIXME: mStandby should be set to true by ThreadBase constructor 5366 mStandby = true; 5367 mKeepWakeLock = property_get_bool("ro.audio.offload_wakelock", true /* default_value */); 5368} 5369 5370void AudioFlinger::OffloadThread::threadLoop_exit() 5371{ 5372 if (mFlushPending || mHwPaused) { 5373 // If a flush is pending or track was paused, just discard buffered data 5374 flushHw_l(); 5375 } else { 5376 mMixerStatus = MIXER_DRAIN_ALL; 5377 threadLoop_drain(); 5378 } 5379 if (mUseAsyncWrite) { 5380 ALOG_ASSERT(mCallbackThread != 0); 5381 mCallbackThread->exit(); 5382 } 5383 PlaybackThread::threadLoop_exit(); 5384} 5385 5386AudioFlinger::PlaybackThread::mixer_state AudioFlinger::OffloadThread::prepareTracks_l( 5387 Vector< sp<Track> > *tracksToRemove 5388) 5389{ 5390 size_t count = mActiveTracks.size(); 5391 5392 mixer_state mixerStatus = MIXER_IDLE; 5393 bool doHwPause = false; 5394 bool doHwResume = false; 5395 5396 ALOGV("OffloadThread::prepareTracks_l active tracks %zu", count); 5397 5398 // find out which tracks need to be processed 5399 for (size_t i = 0; i < count; i++) { 5400 sp<Track> t = mActiveTracks[i].promote(); 5401 // The track died recently 5402 if (t == 0) { 5403 continue; 5404 } 5405 Track* const track = t.get(); 5406#ifdef VERY_VERY_VERBOSE_LOGGING 5407 audio_track_cblk_t* cblk = track->cblk(); 5408#endif 5409 // Only consider last track started for volume and mixer state control. 5410 // In theory an older track could underrun and restart after the new one starts 5411 // but as we only care about the transition phase between two tracks on a 5412 // direct output, it is not a problem to ignore the underrun case. 5413 sp<Track> l = mLatestActiveTrack.promote(); 5414 bool last = l.get() == track; 5415 5416 if (track->isInvalid()) { 5417 ALOGW("An invalidated track shouldn't be in active list"); 5418 tracksToRemove->add(track); 5419 continue; 5420 } 5421 5422 if (track->mState == TrackBase::IDLE) { 5423 ALOGW("An idle track shouldn't be in active list"); 5424 continue; 5425 } 5426 5427 if (track->isPausing()) { 5428 track->setPaused(); 5429 if (last) { 5430 if (mHwSupportsPause && !mHwPaused) { 5431 doHwPause = true; 5432 mHwPaused = true; 5433 } 5434 // If we were part way through writing the mixbuffer to 5435 // the HAL we must save this until we resume 5436 // BUG - this will be wrong if a different track is made active, 5437 // in that case we want to discard the pending data in the 5438 // mixbuffer and tell the client to present it again when the 5439 // track is resumed 5440 mPausedWriteLength = mCurrentWriteLength; 5441 mPausedBytesRemaining = mBytesRemaining; 5442 mBytesRemaining = 0; // stop writing 5443 } 5444 tracksToRemove->add(track); 5445 } else if (track->isFlushPending()) { 5446 if (track->isStopping_1()) { 5447 track->mRetryCount = kMaxTrackStopRetriesOffload; 5448 } else { 5449 track->mRetryCount = kMaxTrackRetriesOffload; 5450 } 5451 track->flushAck(); 5452 if (last) { 5453 mFlushPending = true; 5454 } 5455 } else if (track->isResumePending()){ 5456 track->resumeAck(); 5457 if (last) { 5458 if (mPausedBytesRemaining) { 5459 // Need to continue write that was interrupted 5460 mCurrentWriteLength = mPausedWriteLength; 5461 mBytesRemaining = mPausedBytesRemaining; 5462 mPausedBytesRemaining = 0; 5463 } 5464 if (mHwPaused) { 5465 doHwResume = true; 5466 mHwPaused = false; 5467 // threadLoop_mix() will handle the case that we need to 5468 // resume an interrupted write 5469 } 5470 // enable write to audio HAL 5471 mSleepTimeUs = 0; 5472 5473 mLeftVolFloat = mRightVolFloat = -1.0; 5474 5475 // Do not handle new data in this iteration even if track->framesReady() 5476 mixerStatus = MIXER_TRACKS_ENABLED; 5477 } 5478 } else if (track->framesReady() && track->isReady() && 5479 !track->isPaused() && !track->isTerminated() && !track->isStopping_2()) { 5480 ALOGVV("OffloadThread: track %d s=%08x [OK]", track->name(), cblk->mServer); 5481 if (track->mFillingUpStatus == Track::FS_FILLED) { 5482 track->mFillingUpStatus = Track::FS_ACTIVE; 5483 if (last) { 5484 // make sure processVolume_l() will apply new volume even if 0 5485 mLeftVolFloat = mRightVolFloat = -1.0; 5486 } 5487 } 5488 5489 if (last) { 5490 sp<Track> previousTrack = mPreviousTrack.promote(); 5491 if (previousTrack != 0) { 5492 if (track != previousTrack.get()) { 5493 // Flush any data still being written from last track 5494 mBytesRemaining = 0; 5495 if (mPausedBytesRemaining) { 5496 // Last track was paused so we also need to flush saved 5497 // mixbuffer state and invalidate track so that it will 5498 // re-submit that unwritten data when it is next resumed 5499 mPausedBytesRemaining = 0; 5500 // Invalidate is a bit drastic - would be more efficient 5501 // to have a flag to tell client that some of the 5502 // previously written data was lost 5503 previousTrack->invalidate(); 5504 } 5505 // flush data already sent to the DSP if changing audio session as audio 5506 // comes from a different source. Also invalidate previous track to force a 5507 // seek when resuming. 5508 if (previousTrack->sessionId() != track->sessionId()) { 5509 previousTrack->invalidate(); 5510 } 5511 } 5512 } 5513 mPreviousTrack = track; 5514 // reset retry count 5515 if (track->isStopping_1()) { 5516 track->mRetryCount = kMaxTrackStopRetriesOffload; 5517 } else { 5518 track->mRetryCount = kMaxTrackRetriesOffload; 5519 } 5520 mActiveTrack = t; 5521 mixerStatus = MIXER_TRACKS_READY; 5522 } 5523 } else { 5524 ALOGVV("OffloadThread: track %d s=%08x [NOT READY]", track->name(), cblk->mServer); 5525 if (track->isStopping_1()) { 5526 if (--(track->mRetryCount) <= 0) { 5527 // Hardware buffer can hold a large amount of audio so we must 5528 // wait for all current track's data to drain before we say 5529 // that the track is stopped. 5530 if (mBytesRemaining == 0) { 5531 // Only start draining when all data in mixbuffer 5532 // has been written 5533 ALOGV("OffloadThread: underrun and STOPPING_1 -> draining, STOPPING_2"); 5534 track->mState = TrackBase::STOPPING_2; // so presentation completes after 5535 // drain do not drain if no data was ever sent to HAL (mStandby == true) 5536 if (last && !mStandby) { 5537 // do not modify drain sequence if we are already draining. This happens 5538 // when resuming from pause after drain. 5539 if ((mDrainSequence & 1) == 0) { 5540 mSleepTimeUs = 0; 5541 mStandbyTimeNs = systemTime() + mStandbyDelayNs; 5542 mixerStatus = MIXER_DRAIN_TRACK; 5543 mDrainSequence += 2; 5544 } 5545 if (mHwPaused) { 5546 // It is possible to move from PAUSED to STOPPING_1 without 5547 // a resume so we must ensure hardware is running 5548 doHwResume = true; 5549 mHwPaused = false; 5550 } 5551 } 5552 } 5553 } else if (last) { 5554 ALOGV("stopping1 underrun retries left %d", track->mRetryCount); 5555 mixerStatus = MIXER_TRACKS_ENABLED; 5556 } 5557 } else if (track->isStopping_2()) { 5558 // Drain has completed or we are in standby, signal presentation complete 5559 if (!(mDrainSequence & 1) || !last || mStandby) { 5560 track->mState = TrackBase::STOPPED; 5561 size_t audioHALFrames = 5562 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 5563 int64_t framesWritten = 5564 mBytesWritten / mOutput->getFrameSize(); 5565 track->presentationComplete(framesWritten, audioHALFrames); 5566 track->reset(); 5567 tracksToRemove->add(track); 5568 } 5569 } else { 5570 // No buffers for this track. Give it a few chances to 5571 // fill a buffer, then remove it from active list. 5572 if (--(track->mRetryCount) <= 0) { 5573 bool running = false; 5574 if (mOutput->stream->get_presentation_position != nullptr) { 5575 uint64_t position = 0; 5576 struct timespec unused; 5577 // The running check restarts the retry counter at least once. 5578 int ret = mOutput->stream->get_presentation_position( 5579 mOutput->stream, &position, &unused); 5580 if (ret == NO_ERROR && position != mOffloadUnderrunPosition) { 5581 running = true; 5582 mOffloadUnderrunPosition = position; 5583 } 5584 ALOGVV("underrun counter, running(%d): %lld vs %lld", running, 5585 (long long)position, (long long)mOffloadUnderrunPosition); 5586 } 5587 if (running) { // still running, give us more time. 5588 track->mRetryCount = kMaxTrackRetriesOffload; 5589 } else { 5590 ALOGV("OffloadThread: BUFFER TIMEOUT: remove(%d) from active list", 5591 track->name()); 5592 tracksToRemove->add(track); 5593 // indicate to client process that the track was disabled because of underrun; 5594 // it will then automatically call start() when data is available 5595 track->disable(); 5596 } 5597 } else if (last){ 5598 mixerStatus = MIXER_TRACKS_ENABLED; 5599 } 5600 } 5601 } 5602 // compute volume for this track 5603 processVolume_l(track, last); 5604 } 5605 5606 // make sure the pause/flush/resume sequence is executed in the right order. 5607 // If a flush is pending and a track is active but the HW is not paused, force a HW pause 5608 // before flush and then resume HW. This can happen in case of pause/flush/resume 5609 // if resume is received before pause is executed. 5610 if (!mStandby && (doHwPause || (mFlushPending && !mHwPaused && (count != 0)))) { 5611 mOutput->stream->pause(mOutput->stream); 5612 } 5613 if (mFlushPending) { 5614 flushHw_l(); 5615 } 5616 if (!mStandby && doHwResume) { 5617 mOutput->stream->resume(mOutput->stream); 5618 } 5619 5620 // remove all the tracks that need to be... 5621 removeTracks_l(*tracksToRemove); 5622 5623 return mixerStatus; 5624} 5625 5626// must be called with thread mutex locked 5627bool AudioFlinger::OffloadThread::waitingAsyncCallback_l() 5628{ 5629 ALOGVV("waitingAsyncCallback_l mWriteAckSequence %d mDrainSequence %d", 5630 mWriteAckSequence, mDrainSequence); 5631 if (mUseAsyncWrite && ((mWriteAckSequence & 1) || (mDrainSequence & 1))) { 5632 return true; 5633 } 5634 return false; 5635} 5636 5637bool AudioFlinger::OffloadThread::waitingAsyncCallback() 5638{ 5639 Mutex::Autolock _l(mLock); 5640 return waitingAsyncCallback_l(); 5641} 5642 5643void AudioFlinger::OffloadThread::flushHw_l() 5644{ 5645 DirectOutputThread::flushHw_l(); 5646 // Flush anything still waiting in the mixbuffer 5647 mCurrentWriteLength = 0; 5648 mBytesRemaining = 0; 5649 mPausedWriteLength = 0; 5650 mPausedBytesRemaining = 0; 5651 // reset bytes written count to reflect that DSP buffers are empty after flush. 5652 mBytesWritten = 0; 5653 mOffloadUnderrunPosition = ~0LL; 5654 5655 if (mUseAsyncWrite) { 5656 // discard any pending drain or write ack by incrementing sequence 5657 mWriteAckSequence = (mWriteAckSequence + 2) & ~1; 5658 mDrainSequence = (mDrainSequence + 2) & ~1; 5659 ALOG_ASSERT(mCallbackThread != 0); 5660 mCallbackThread->setWriteBlocked(mWriteAckSequence); 5661 mCallbackThread->setDraining(mDrainSequence); 5662 } 5663} 5664 5665void AudioFlinger::OffloadThread::invalidateTracks(audio_stream_type_t streamType) 5666{ 5667 Mutex::Autolock _l(mLock); 5668 if (PlaybackThread::invalidateTracks_l(streamType)) { 5669 mFlushPending = true; 5670 } 5671} 5672 5673// ---------------------------------------------------------------------------- 5674 5675AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 5676 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id, bool systemReady) 5677 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->outDevice(), 5678 systemReady, DUPLICATING), 5679 mWaitTimeMs(UINT_MAX) 5680{ 5681 addOutputTrack(mainThread); 5682} 5683 5684AudioFlinger::DuplicatingThread::~DuplicatingThread() 5685{ 5686 for (size_t i = 0; i < mOutputTracks.size(); i++) { 5687 mOutputTracks[i]->destroy(); 5688 } 5689} 5690 5691void AudioFlinger::DuplicatingThread::threadLoop_mix() 5692{ 5693 // mix buffers... 5694 if (outputsReady(outputTracks)) { 5695 mAudioMixer->process(); 5696 } else { 5697 if (mMixerBufferValid) { 5698 memset(mMixerBuffer, 0, mMixerBufferSize); 5699 } else { 5700 memset(mSinkBuffer, 0, mSinkBufferSize); 5701 } 5702 } 5703 mSleepTimeUs = 0; 5704 writeFrames = mNormalFrameCount; 5705 mCurrentWriteLength = mSinkBufferSize; 5706 mStandbyTimeNs = systemTime() + mStandbyDelayNs; 5707} 5708 5709void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() 5710{ 5711 if (mSleepTimeUs == 0) { 5712 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 5713 mSleepTimeUs = mActiveSleepTimeUs; 5714 } else { 5715 mSleepTimeUs = mIdleSleepTimeUs; 5716 } 5717 } else if (mBytesWritten != 0) { 5718 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 5719 writeFrames = mNormalFrameCount; 5720 memset(mSinkBuffer, 0, mSinkBufferSize); 5721 } else { 5722 // flush remaining overflow buffers in output tracks 5723 writeFrames = 0; 5724 } 5725 mSleepTimeUs = 0; 5726 } 5727} 5728 5729ssize_t AudioFlinger::DuplicatingThread::threadLoop_write() 5730{ 5731 for (size_t i = 0; i < outputTracks.size(); i++) { 5732 outputTracks[i]->write(mSinkBuffer, writeFrames); 5733 } 5734 mStandby = false; 5735 return (ssize_t)mSinkBufferSize; 5736} 5737 5738void AudioFlinger::DuplicatingThread::threadLoop_standby() 5739{ 5740 // DuplicatingThread implements standby by stopping all tracks 5741 for (size_t i = 0; i < outputTracks.size(); i++) { 5742 outputTracks[i]->stop(); 5743 } 5744} 5745 5746void AudioFlinger::DuplicatingThread::saveOutputTracks() 5747{ 5748 outputTracks = mOutputTracks; 5749} 5750 5751void AudioFlinger::DuplicatingThread::clearOutputTracks() 5752{ 5753 outputTracks.clear(); 5754} 5755 5756void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 5757{ 5758 Mutex::Autolock _l(mLock); 5759 // The downstream MixerThread consumes thread->frameCount() amount of frames per mix pass. 5760 // Adjust for thread->sampleRate() to determine minimum buffer frame count. 5761 // Then triple buffer because Threads do not run synchronously and may not be clock locked. 5762 const size_t frameCount = 5763 3 * sourceFramesNeeded(mSampleRate, thread->frameCount(), thread->sampleRate()); 5764 // TODO: Consider asynchronous sample rate conversion to handle clock disparity 5765 // from different OutputTracks and their associated MixerThreads (e.g. one may 5766 // nearly empty and the other may be dropping data). 5767 5768 sp<OutputTrack> outputTrack = new OutputTrack(thread, 5769 this, 5770 mSampleRate, 5771 mFormat, 5772 mChannelMask, 5773 frameCount, 5774 IPCThreadState::self()->getCallingUid()); 5775 status_t status = outputTrack != 0 ? outputTrack->initCheck() : (status_t) NO_MEMORY; 5776 if (status != NO_ERROR) { 5777 ALOGE("addOutputTrack() initCheck failed %d", status); 5778 return; 5779 } 5780 thread->setStreamVolume(AUDIO_STREAM_PATCH, 1.0f); 5781 mOutputTracks.add(outputTrack); 5782 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack.get(), thread); 5783 updateWaitTime_l(); 5784} 5785 5786void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 5787{ 5788 Mutex::Autolock _l(mLock); 5789 for (size_t i = 0; i < mOutputTracks.size(); i++) { 5790 if (mOutputTracks[i]->thread() == thread) { 5791 mOutputTracks[i]->destroy(); 5792 mOutputTracks.removeAt(i); 5793 updateWaitTime_l(); 5794 if (thread->getOutput() == mOutput) { 5795 mOutput = NULL; 5796 } 5797 return; 5798 } 5799 } 5800 ALOGV("removeOutputTrack(): unknown thread: %p", thread); 5801} 5802 5803// caller must hold mLock 5804void AudioFlinger::DuplicatingThread::updateWaitTime_l() 5805{ 5806 mWaitTimeMs = UINT_MAX; 5807 for (size_t i = 0; i < mOutputTracks.size(); i++) { 5808 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 5809 if (strong != 0) { 5810 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 5811 if (waitTimeMs < mWaitTimeMs) { 5812 mWaitTimeMs = waitTimeMs; 5813 } 5814 } 5815 } 5816} 5817 5818 5819bool AudioFlinger::DuplicatingThread::outputsReady( 5820 const SortedVector< sp<OutputTrack> > &outputTracks) 5821{ 5822 for (size_t i = 0; i < outputTracks.size(); i++) { 5823 sp<ThreadBase> thread = outputTracks[i]->thread().promote(); 5824 if (thread == 0) { 5825 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", 5826 outputTracks[i].get()); 5827 return false; 5828 } 5829 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 5830 // see note at standby() declaration 5831 if (playbackThread->standby() && !playbackThread->isSuspended()) { 5832 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), 5833 thread.get()); 5834 return false; 5835 } 5836 } 5837 return true; 5838} 5839 5840uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() const 5841{ 5842 return (mWaitTimeMs * 1000) / 2; 5843} 5844 5845void AudioFlinger::DuplicatingThread::cacheParameters_l() 5846{ 5847 // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first 5848 updateWaitTime_l(); 5849 5850 MixerThread::cacheParameters_l(); 5851} 5852 5853// ---------------------------------------------------------------------------- 5854// Record 5855// ---------------------------------------------------------------------------- 5856 5857AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 5858 AudioStreamIn *input, 5859 audio_io_handle_t id, 5860 audio_devices_t outDevice, 5861 audio_devices_t inDevice, 5862 bool systemReady 5863#ifdef TEE_SINK 5864 , const sp<NBAIO_Sink>& teeSink 5865#endif 5866 ) : 5867 ThreadBase(audioFlinger, id, outDevice, inDevice, RECORD, systemReady), 5868 mInput(input), mActiveTracksGen(0), mRsmpInBuffer(NULL), 5869 // mRsmpInFrames and mRsmpInFramesP2 are set by readInputParameters_l() 5870 mRsmpInRear(0) 5871#ifdef TEE_SINK 5872 , mTeeSink(teeSink) 5873#endif 5874 , mReadOnlyHeap(new MemoryDealer(kRecordThreadReadOnlyHeapSize, 5875 "RecordThreadRO", MemoryHeapBase::READ_ONLY)) 5876 // mFastCapture below 5877 , mFastCaptureFutex(0) 5878 // mInputSource 5879 // mPipeSink 5880 // mPipeSource 5881 , mPipeFramesP2(0) 5882 // mPipeMemory 5883 // mFastCaptureNBLogWriter 5884 , mFastTrackAvail(false) 5885{ 5886 snprintf(mThreadName, kThreadNameLength, "AudioIn_%X", id); 5887 mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mThreadName); 5888 5889 readInputParameters_l(); 5890 5891 // create an NBAIO source for the HAL input stream, and negotiate 5892 mInputSource = new AudioStreamInSource(input->stream); 5893 size_t numCounterOffers = 0; 5894 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount, mFormat)}; 5895#if !LOG_NDEBUG 5896 ssize_t index = 5897#else 5898 (void) 5899#endif 5900 mInputSource->negotiate(offers, 1, NULL, numCounterOffers); 5901 ALOG_ASSERT(index == 0); 5902 5903 // initialize fast capture depending on configuration 5904 bool initFastCapture; 5905 switch (kUseFastCapture) { 5906 case FastCapture_Never: 5907 initFastCapture = false; 5908 break; 5909 case FastCapture_Always: 5910 initFastCapture = true; 5911 break; 5912 case FastCapture_Static: 5913 initFastCapture = (mFrameCount * 1000) / mSampleRate < kMinNormalCaptureBufferSizeMs; 5914 break; 5915 // case FastCapture_Dynamic: 5916 } 5917 5918 if (initFastCapture) { 5919 // create a Pipe for FastCapture to write to, and for us and fast tracks to read from 5920 NBAIO_Format format = mInputSource->format(); 5921 size_t pipeFramesP2 = roundup(mSampleRate / 25); // double-buffering of 20 ms each 5922 size_t pipeSize = pipeFramesP2 * Format_frameSize(format); 5923 void *pipeBuffer; 5924 const sp<MemoryDealer> roHeap(readOnlyHeap()); 5925 sp<IMemory> pipeMemory; 5926 if ((roHeap == 0) || 5927 (pipeMemory = roHeap->allocate(pipeSize)) == 0 || 5928 (pipeBuffer = pipeMemory->pointer()) == NULL) { 5929 ALOGE("not enough memory for pipe buffer size=%zu", pipeSize); 5930 goto failed; 5931 } 5932 // pipe will be shared directly with fast clients, so clear to avoid leaking old information 5933 memset(pipeBuffer, 0, pipeSize); 5934 Pipe *pipe = new Pipe(pipeFramesP2, format, pipeBuffer); 5935 const NBAIO_Format offers[1] = {format}; 5936 size_t numCounterOffers = 0; 5937 ssize_t index = pipe->negotiate(offers, 1, NULL, numCounterOffers); 5938 ALOG_ASSERT(index == 0); 5939 mPipeSink = pipe; 5940 PipeReader *pipeReader = new PipeReader(*pipe); 5941 numCounterOffers = 0; 5942 index = pipeReader->negotiate(offers, 1, NULL, numCounterOffers); 5943 ALOG_ASSERT(index == 0); 5944 mPipeSource = pipeReader; 5945 mPipeFramesP2 = pipeFramesP2; 5946 mPipeMemory = pipeMemory; 5947 5948 // create fast capture 5949 mFastCapture = new FastCapture(); 5950 FastCaptureStateQueue *sq = mFastCapture->sq(); 5951#ifdef STATE_QUEUE_DUMP 5952 // FIXME 5953#endif 5954 FastCaptureState *state = sq->begin(); 5955 state->mCblk = NULL; 5956 state->mInputSource = mInputSource.get(); 5957 state->mInputSourceGen++; 5958 state->mPipeSink = pipe; 5959 state->mPipeSinkGen++; 5960 state->mFrameCount = mFrameCount; 5961 state->mCommand = FastCaptureState::COLD_IDLE; 5962 // already done in constructor initialization list 5963 //mFastCaptureFutex = 0; 5964 state->mColdFutexAddr = &mFastCaptureFutex; 5965 state->mColdGen++; 5966 state->mDumpState = &mFastCaptureDumpState; 5967#ifdef TEE_SINK 5968 // FIXME 5969#endif 5970 mFastCaptureNBLogWriter = audioFlinger->newWriter_l(kFastCaptureLogSize, "FastCapture"); 5971 state->mNBLogWriter = mFastCaptureNBLogWriter.get(); 5972 sq->end(); 5973 sq->push(FastCaptureStateQueue::BLOCK_UNTIL_PUSHED); 5974 5975 // start the fast capture 5976 mFastCapture->run("FastCapture", ANDROID_PRIORITY_URGENT_AUDIO); 5977 pid_t tid = mFastCapture->getTid(); 5978 sendPrioConfigEvent(getpid_cached, tid, kPriorityFastCapture); 5979#ifdef AUDIO_WATCHDOG 5980 // FIXME 5981#endif 5982 5983 mFastTrackAvail = true; 5984 } 5985failed: ; 5986 5987 // FIXME mNormalSource 5988} 5989 5990AudioFlinger::RecordThread::~RecordThread() 5991{ 5992 if (mFastCapture != 0) { 5993 FastCaptureStateQueue *sq = mFastCapture->sq(); 5994 FastCaptureState *state = sq->begin(); 5995 if (state->mCommand == FastCaptureState::COLD_IDLE) { 5996 int32_t old = android_atomic_inc(&mFastCaptureFutex); 5997 if (old == -1) { 5998 (void) syscall(__NR_futex, &mFastCaptureFutex, FUTEX_WAKE_PRIVATE, 1); 5999 } 6000 } 6001 state->mCommand = FastCaptureState::EXIT; 6002 sq->end(); 6003 sq->push(FastCaptureStateQueue::BLOCK_UNTIL_PUSHED); 6004 mFastCapture->join(); 6005 mFastCapture.clear(); 6006 } 6007 mAudioFlinger->unregisterWriter(mFastCaptureNBLogWriter); 6008 mAudioFlinger->unregisterWriter(mNBLogWriter); 6009 free(mRsmpInBuffer); 6010} 6011 6012void AudioFlinger::RecordThread::onFirstRef() 6013{ 6014 run(mThreadName, PRIORITY_URGENT_AUDIO); 6015} 6016 6017bool AudioFlinger::RecordThread::threadLoop() 6018{ 6019 nsecs_t lastWarning = 0; 6020 6021 inputStandBy(); 6022 6023reacquire_wakelock: 6024 sp<RecordTrack> activeTrack; 6025 int activeTracksGen; 6026 { 6027 Mutex::Autolock _l(mLock); 6028 size_t size = mActiveTracks.size(); 6029 activeTracksGen = mActiveTracksGen; 6030 if (size > 0) { 6031 // FIXME an arbitrary choice 6032 activeTrack = mActiveTracks[0]; 6033 acquireWakeLock_l(activeTrack->uid()); 6034 if (size > 1) { 6035 SortedVector<int> tmp; 6036 for (size_t i = 0; i < size; i++) { 6037 tmp.add(mActiveTracks[i]->uid()); 6038 } 6039 updateWakeLockUids_l(tmp); 6040 } 6041 } else { 6042 acquireWakeLock_l(-1); 6043 } 6044 } 6045 6046 // used to request a deferred sleep, to be executed later while mutex is unlocked 6047 uint32_t sleepUs = 0; 6048 6049 // loop while there is work to do 6050 for (;;) { 6051 Vector< sp<EffectChain> > effectChains; 6052 6053 // activeTracks accumulates a copy of a subset of mActiveTracks 6054 Vector< sp<RecordTrack> > activeTracks; 6055 6056 // reference to the (first and only) active fast track 6057 sp<RecordTrack> fastTrack; 6058 6059 // reference to a fast track which is about to be removed 6060 sp<RecordTrack> fastTrackToRemove; 6061 6062 { // scope for mLock 6063 Mutex::Autolock _l(mLock); 6064 6065 processConfigEvents_l(); 6066 6067 // check exitPending here because checkForNewParameters_l() and 6068 // checkForNewParameters_l() can temporarily release mLock 6069 if (exitPending()) { 6070 break; 6071 } 6072 6073 // sleep with mutex unlocked 6074 if (sleepUs > 0) { 6075 ATRACE_BEGIN("sleepC"); 6076 mWaitWorkCV.waitRelative(mLock, microseconds((nsecs_t)sleepUs)); 6077 ATRACE_END(); 6078 sleepUs = 0; 6079 continue; 6080 } 6081 6082 // if no active track(s), then standby and release wakelock 6083 size_t size = mActiveTracks.size(); 6084 if (size == 0) { 6085 standbyIfNotAlreadyInStandby(); 6086 // exitPending() can't become true here 6087 releaseWakeLock_l(); 6088 ALOGV("RecordThread: loop stopping"); 6089 // go to sleep 6090 mWaitWorkCV.wait(mLock); 6091 ALOGV("RecordThread: loop starting"); 6092 goto reacquire_wakelock; 6093 } 6094 6095 if (mActiveTracksGen != activeTracksGen) { 6096 activeTracksGen = mActiveTracksGen; 6097 SortedVector<int> tmp; 6098 for (size_t i = 0; i < size; i++) { 6099 tmp.add(mActiveTracks[i]->uid()); 6100 } 6101 updateWakeLockUids_l(tmp); 6102 } 6103 6104 bool doBroadcast = false; 6105 bool allStopped = true; 6106 for (size_t i = 0; i < size; ) { 6107 6108 activeTrack = mActiveTracks[i]; 6109 if (activeTrack->isTerminated()) { 6110 if (activeTrack->isFastTrack()) { 6111 ALOG_ASSERT(fastTrackToRemove == 0); 6112 fastTrackToRemove = activeTrack; 6113 } 6114 removeTrack_l(activeTrack); 6115 mActiveTracks.remove(activeTrack); 6116 mActiveTracksGen++; 6117 size--; 6118 continue; 6119 } 6120 6121 TrackBase::track_state activeTrackState = activeTrack->mState; 6122 switch (activeTrackState) { 6123 6124 case TrackBase::PAUSING: 6125 mActiveTracks.remove(activeTrack); 6126 mActiveTracksGen++; 6127 doBroadcast = true; 6128 size--; 6129 continue; 6130 6131 case TrackBase::STARTING_1: 6132 sleepUs = 10000; 6133 i++; 6134 allStopped = false; 6135 continue; 6136 6137 case TrackBase::STARTING_2: 6138 doBroadcast = true; 6139 mStandby = false; 6140 activeTrack->mState = TrackBase::ACTIVE; 6141 allStopped = false; 6142 break; 6143 6144 case TrackBase::ACTIVE: 6145 allStopped = false; 6146 break; 6147 6148 case TrackBase::IDLE: 6149 i++; 6150 continue; 6151 6152 default: 6153 LOG_ALWAYS_FATAL("Unexpected activeTrackState %d", activeTrackState); 6154 } 6155 6156 activeTracks.add(activeTrack); 6157 i++; 6158 6159 if (activeTrack->isFastTrack()) { 6160 ALOG_ASSERT(!mFastTrackAvail); 6161 ALOG_ASSERT(fastTrack == 0); 6162 fastTrack = activeTrack; 6163 } 6164 } 6165 6166 if (allStopped) { 6167 standbyIfNotAlreadyInStandby(); 6168 } 6169 if (doBroadcast) { 6170 mStartStopCond.broadcast(); 6171 } 6172 6173 // sleep if there are no active tracks to process 6174 if (activeTracks.size() == 0) { 6175 if (sleepUs == 0) { 6176 sleepUs = kRecordThreadSleepUs; 6177 } 6178 continue; 6179 } 6180 sleepUs = 0; 6181 6182 lockEffectChains_l(effectChains); 6183 } 6184 6185 // thread mutex is now unlocked, mActiveTracks unknown, activeTracks.size() > 0 6186 6187 size_t size = effectChains.size(); 6188 for (size_t i = 0; i < size; i++) { 6189 // thread mutex is not locked, but effect chain is locked 6190 effectChains[i]->process_l(); 6191 } 6192 6193 // Push a new fast capture state if fast capture is not already running, or cblk change 6194 if (mFastCapture != 0) { 6195 FastCaptureStateQueue *sq = mFastCapture->sq(); 6196 FastCaptureState *state = sq->begin(); 6197 bool didModify = false; 6198 FastCaptureStateQueue::block_t block = FastCaptureStateQueue::BLOCK_UNTIL_PUSHED; 6199 if (state->mCommand != FastCaptureState::READ_WRITE /* FIXME && 6200 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)*/) { 6201 if (state->mCommand == FastCaptureState::COLD_IDLE) { 6202 int32_t old = android_atomic_inc(&mFastCaptureFutex); 6203 if (old == -1) { 6204 (void) syscall(__NR_futex, &mFastCaptureFutex, FUTEX_WAKE_PRIVATE, 1); 6205 } 6206 } 6207 state->mCommand = FastCaptureState::READ_WRITE; 6208#if 0 // FIXME 6209 mFastCaptureDumpState.increaseSamplingN(mAudioFlinger->isLowRamDevice() ? 6210 FastThreadDumpState::kSamplingNforLowRamDevice : 6211 FastThreadDumpState::kSamplingN); 6212#endif 6213 didModify = true; 6214 } 6215 audio_track_cblk_t *cblkOld = state->mCblk; 6216 audio_track_cblk_t *cblkNew = fastTrack != 0 ? fastTrack->cblk() : NULL; 6217 if (cblkNew != cblkOld) { 6218 state->mCblk = cblkNew; 6219 // block until acked if removing a fast track 6220 if (cblkOld != NULL) { 6221 block = FastCaptureStateQueue::BLOCK_UNTIL_ACKED; 6222 } 6223 didModify = true; 6224 } 6225 sq->end(didModify); 6226 if (didModify) { 6227 sq->push(block); 6228#if 0 6229 if (kUseFastCapture == FastCapture_Dynamic) { 6230 mNormalSource = mPipeSource; 6231 } 6232#endif 6233 } 6234 } 6235 6236 // now run the fast track destructor with thread mutex unlocked 6237 fastTrackToRemove.clear(); 6238 6239 // Read from HAL to keep up with fastest client if multiple active tracks, not slowest one. 6240 // Only the client(s) that are too slow will overrun. But if even the fastest client is too 6241 // slow, then this RecordThread will overrun by not calling HAL read often enough. 6242 // If destination is non-contiguous, first read past the nominal end of buffer, then 6243 // copy to the right place. Permitted because mRsmpInBuffer was over-allocated. 6244 6245 int32_t rear = mRsmpInRear & (mRsmpInFramesP2 - 1); 6246 ssize_t framesRead; 6247 6248 // If an NBAIO source is present, use it to read the normal capture's data 6249 if (mPipeSource != 0) { 6250 size_t framesToRead = mBufferSize / mFrameSize; 6251 framesRead = mPipeSource->read((uint8_t*)mRsmpInBuffer + rear * mFrameSize, 6252 framesToRead); 6253 if (framesRead == 0) { 6254 // since pipe is non-blocking, simulate blocking input 6255 sleepUs = (framesToRead * 1000000LL) / mSampleRate; 6256 } 6257 // otherwise use the HAL / AudioStreamIn directly 6258 } else { 6259 ATRACE_BEGIN("read"); 6260 ssize_t bytesRead = mInput->stream->read(mInput->stream, 6261 (uint8_t*)mRsmpInBuffer + rear * mFrameSize, mBufferSize); 6262 ATRACE_END(); 6263 if (bytesRead < 0) { 6264 framesRead = bytesRead; 6265 } else { 6266 framesRead = bytesRead / mFrameSize; 6267 } 6268 } 6269 6270 // Update server timestamp with server stats 6271 // systemTime() is optional if the hardware supports timestamps. 6272 mTimestamp.mPosition[ExtendedTimestamp::LOCATION_SERVER] += framesRead; 6273 mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_SERVER] = systemTime(); 6274 6275 // Update server timestamp with kernel stats 6276 if (mInput->stream->get_capture_position != nullptr 6277 && mPipeSource.get() == nullptr /* don't obtain for FastCapture, could block */) { 6278 int64_t position, time; 6279 int ret = mInput->stream->get_capture_position(mInput->stream, &position, &time); 6280 if (ret == NO_ERROR) { 6281 mTimestamp.mPosition[ExtendedTimestamp::LOCATION_KERNEL] = position; 6282 mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL] = time; 6283 // Note: In general record buffers should tend to be empty in 6284 // a properly running pipeline. 6285 // 6286 // Also, it is not advantageous to call get_presentation_position during the read 6287 // as the read obtains a lock, preventing the timestamp call from executing. 6288 } 6289 } 6290 // Use this to track timestamp information 6291 // ALOGD("%s", mTimestamp.toString().c_str()); 6292 6293 if (framesRead < 0 || (framesRead == 0 && mPipeSource == 0)) { 6294 ALOGE("read failed: framesRead=%zd", framesRead); 6295 // Force input into standby so that it tries to recover at next read attempt 6296 inputStandBy(); 6297 sleepUs = kRecordThreadSleepUs; 6298 } 6299 if (framesRead <= 0) { 6300 goto unlock; 6301 } 6302 ALOG_ASSERT(framesRead > 0); 6303 6304 if (mTeeSink != 0) { 6305 (void) mTeeSink->write((uint8_t*)mRsmpInBuffer + rear * mFrameSize, framesRead); 6306 } 6307 // If destination is non-contiguous, we now correct for reading past end of buffer. 6308 { 6309 size_t part1 = mRsmpInFramesP2 - rear; 6310 if ((size_t) framesRead > part1) { 6311 memcpy(mRsmpInBuffer, (uint8_t*)mRsmpInBuffer + mRsmpInFramesP2 * mFrameSize, 6312 (framesRead - part1) * mFrameSize); 6313 } 6314 } 6315 rear = mRsmpInRear += framesRead; 6316 6317 size = activeTracks.size(); 6318 // loop over each active track 6319 for (size_t i = 0; i < size; i++) { 6320 activeTrack = activeTracks[i]; 6321 6322 // skip fast tracks, as those are handled directly by FastCapture 6323 if (activeTrack->isFastTrack()) { 6324 continue; 6325 } 6326 6327 // TODO: This code probably should be moved to RecordTrack. 6328 // TODO: Update the activeTrack buffer converter in case of reconfigure. 6329 6330 enum { 6331 OVERRUN_UNKNOWN, 6332 OVERRUN_TRUE, 6333 OVERRUN_FALSE 6334 } overrun = OVERRUN_UNKNOWN; 6335 6336 // loop over getNextBuffer to handle circular sink 6337 for (;;) { 6338 6339 activeTrack->mSink.frameCount = ~0; 6340 status_t status = activeTrack->getNextBuffer(&activeTrack->mSink); 6341 size_t framesOut = activeTrack->mSink.frameCount; 6342 LOG_ALWAYS_FATAL_IF((status == OK) != (framesOut > 0)); 6343 6344 // check available frames and handle overrun conditions 6345 // if the record track isn't draining fast enough. 6346 bool hasOverrun; 6347 size_t framesIn; 6348 activeTrack->mResamplerBufferProvider->sync(&framesIn, &hasOverrun); 6349 if (hasOverrun) { 6350 overrun = OVERRUN_TRUE; 6351 } 6352 if (framesOut == 0 || framesIn == 0) { 6353 break; 6354 } 6355 6356 // Don't allow framesOut to be larger than what is possible with resampling 6357 // from framesIn. 6358 // This isn't strictly necessary but helps limit buffer resizing in 6359 // RecordBufferConverter. TODO: remove when no longer needed. 6360 framesOut = min(framesOut, 6361 destinationFramesPossible( 6362 framesIn, mSampleRate, activeTrack->mSampleRate)); 6363 // process frames from the RecordThread buffer provider to the RecordTrack buffer 6364 framesOut = activeTrack->mRecordBufferConverter->convert( 6365 activeTrack->mSink.raw, activeTrack->mResamplerBufferProvider, framesOut); 6366 6367 if (framesOut > 0 && (overrun == OVERRUN_UNKNOWN)) { 6368 overrun = OVERRUN_FALSE; 6369 } 6370 6371 if (activeTrack->mFramesToDrop == 0) { 6372 if (framesOut > 0) { 6373 activeTrack->mSink.frameCount = framesOut; 6374 activeTrack->releaseBuffer(&activeTrack->mSink); 6375 } 6376 } else { 6377 // FIXME could do a partial drop of framesOut 6378 if (activeTrack->mFramesToDrop > 0) { 6379 activeTrack->mFramesToDrop -= framesOut; 6380 if (activeTrack->mFramesToDrop <= 0) { 6381 activeTrack->clearSyncStartEvent(); 6382 } 6383 } else { 6384 activeTrack->mFramesToDrop += framesOut; 6385 if (activeTrack->mFramesToDrop >= 0 || activeTrack->mSyncStartEvent == 0 || 6386 activeTrack->mSyncStartEvent->isCancelled()) { 6387 ALOGW("Synced record %s, session %d, trigger session %d", 6388 (activeTrack->mFramesToDrop >= 0) ? "timed out" : "cancelled", 6389 activeTrack->sessionId(), 6390 (activeTrack->mSyncStartEvent != 0) ? 6391 activeTrack->mSyncStartEvent->triggerSession() : 6392 AUDIO_SESSION_NONE); 6393 activeTrack->clearSyncStartEvent(); 6394 } 6395 } 6396 } 6397 6398 if (framesOut == 0) { 6399 break; 6400 } 6401 } 6402 6403 switch (overrun) { 6404 case OVERRUN_TRUE: 6405 // client isn't retrieving buffers fast enough 6406 if (!activeTrack->setOverflow()) { 6407 nsecs_t now = systemTime(); 6408 // FIXME should lastWarning per track? 6409 if ((now - lastWarning) > kWarningThrottleNs) { 6410 ALOGW("RecordThread: buffer overflow"); 6411 lastWarning = now; 6412 } 6413 } 6414 break; 6415 case OVERRUN_FALSE: 6416 activeTrack->clearOverflow(); 6417 break; 6418 case OVERRUN_UNKNOWN: 6419 break; 6420 } 6421 6422 // update frame information and push timestamp out 6423 activeTrack->updateTrackFrameInfo( 6424 activeTrack->mServerProxy->framesReleased(), 6425 mTimestamp.mPosition[ExtendedTimestamp::LOCATION_SERVER], 6426 mSampleRate, mTimestamp); 6427 } 6428 6429unlock: 6430 // enable changes in effect chain 6431 unlockEffectChains(effectChains); 6432 // effectChains doesn't need to be cleared, since it is cleared by destructor at scope end 6433 } 6434 6435 standbyIfNotAlreadyInStandby(); 6436 6437 { 6438 Mutex::Autolock _l(mLock); 6439 for (size_t i = 0; i < mTracks.size(); i++) { 6440 sp<RecordTrack> track = mTracks[i]; 6441 track->invalidate(); 6442 } 6443 mActiveTracks.clear(); 6444 mActiveTracksGen++; 6445 mStartStopCond.broadcast(); 6446 } 6447 6448 releaseWakeLock(); 6449 6450 ALOGV("RecordThread %p exiting", this); 6451 return false; 6452} 6453 6454void AudioFlinger::RecordThread::standbyIfNotAlreadyInStandby() 6455{ 6456 if (!mStandby) { 6457 inputStandBy(); 6458 mStandby = true; 6459 } 6460} 6461 6462void AudioFlinger::RecordThread::inputStandBy() 6463{ 6464 // Idle the fast capture if it's currently running 6465 if (mFastCapture != 0) { 6466 FastCaptureStateQueue *sq = mFastCapture->sq(); 6467 FastCaptureState *state = sq->begin(); 6468 if (!(state->mCommand & FastCaptureState::IDLE)) { 6469 state->mCommand = FastCaptureState::COLD_IDLE; 6470 state->mColdFutexAddr = &mFastCaptureFutex; 6471 state->mColdGen++; 6472 mFastCaptureFutex = 0; 6473 sq->end(); 6474 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 6475 sq->push(FastCaptureStateQueue::BLOCK_UNTIL_ACKED); 6476#if 0 6477 if (kUseFastCapture == FastCapture_Dynamic) { 6478 // FIXME 6479 } 6480#endif 6481#ifdef AUDIO_WATCHDOG 6482 // FIXME 6483#endif 6484 } else { 6485 sq->end(false /*didModify*/); 6486 } 6487 } 6488 mInput->stream->common.standby(&mInput->stream->common); 6489 6490 // If going into standby, flush the pipe source. 6491 if (mPipeSource.get() != nullptr) { 6492 const ssize_t flushed = mPipeSource->flush(); 6493 if (flushed > 0) { 6494 ALOGV("Input standby flushed PipeSource %zd frames", flushed); 6495 mTimestamp.mPosition[ExtendedTimestamp::LOCATION_SERVER] += flushed; 6496 mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_SERVER] = systemTime(); 6497 } 6498 } 6499} 6500 6501// RecordThread::createRecordTrack_l() must be called with AudioFlinger::mLock held 6502sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 6503 const sp<AudioFlinger::Client>& client, 6504 uint32_t sampleRate, 6505 audio_format_t format, 6506 audio_channel_mask_t channelMask, 6507 size_t *pFrameCount, 6508 audio_session_t sessionId, 6509 size_t *notificationFrames, 6510 int uid, 6511 audio_input_flags_t *flags, 6512 pid_t tid, 6513 status_t *status) 6514{ 6515 size_t frameCount = *pFrameCount; 6516 sp<RecordTrack> track; 6517 status_t lStatus; 6518 audio_input_flags_t inputFlags = mInput->flags; 6519 6520 // special case for FAST flag considered OK if fast capture is present 6521 if (hasFastCapture()) { 6522 inputFlags = (audio_input_flags_t)(inputFlags | AUDIO_INPUT_FLAG_FAST); 6523 } 6524 6525 // Check if requested flags are compatible with output stream flags 6526 if ((*flags & inputFlags) != *flags) { 6527 ALOGW("createRecordTrack_l(): mismatch between requested flags (%08x) and" 6528 " input flags (%08x)", 6529 *flags, inputFlags); 6530 *flags = (audio_input_flags_t)(*flags & inputFlags); 6531 } 6532 6533 // client expresses a preference for FAST, but we get the final say 6534 if (*flags & AUDIO_INPUT_FLAG_FAST) { 6535 if ( 6536 // we formerly checked for a callback handler (non-0 tid), 6537 // but that is no longer required for TRANSFER_OBTAIN mode 6538 // 6539 // frame count is not specified, or is exactly the pipe depth 6540 ((frameCount == 0) || (frameCount == mPipeFramesP2)) && 6541 // PCM data 6542 audio_is_linear_pcm(format) && 6543 // hardware format 6544 (format == mFormat) && 6545 // hardware channel mask 6546 (channelMask == mChannelMask) && 6547 // hardware sample rate 6548 (sampleRate == mSampleRate) && 6549 // record thread has an associated fast capture 6550 hasFastCapture() && 6551 // there are sufficient fast track slots available 6552 mFastTrackAvail 6553 ) { 6554 // check compatibility with audio effects. 6555 Mutex::Autolock _l(mLock); 6556 // Do not accept FAST flag if the session has software effects 6557 sp<EffectChain> chain = getEffectChain_l(sessionId); 6558 if (chain != 0) { 6559 ALOGV_IF((*flags & AUDIO_INPUT_FLAG_RAW) != 0, 6560 "AUDIO_INPUT_FLAG_RAW denied: effect present on session"); 6561 *flags = (audio_input_flags_t)(*flags & ~AUDIO_INPUT_FLAG_RAW); 6562 if (chain->hasSoftwareEffect()) { 6563 ALOGV("AUDIO_INPUT_FLAG_FAST denied: software effect present on session"); 6564 *flags = (audio_input_flags_t)(*flags & ~AUDIO_INPUT_FLAG_FAST); 6565 } 6566 } 6567 ALOGV_IF((*flags & AUDIO_INPUT_FLAG_FAST) != 0, 6568 "AUDIO_INPUT_FLAG_FAST accepted: frameCount=%zu mFrameCount=%zu", 6569 frameCount, mFrameCount); 6570 } else { 6571 ALOGV("AUDIO_INPUT_FLAG_FAST denied: frameCount=%zu mFrameCount=%zu mPipeFramesP2=%zu " 6572 "format=%#x isLinear=%d channelMask=%#x sampleRate=%u mSampleRate=%u " 6573 "hasFastCapture=%d tid=%d mFastTrackAvail=%d", 6574 frameCount, mFrameCount, mPipeFramesP2, 6575 format, audio_is_linear_pcm(format), channelMask, sampleRate, mSampleRate, 6576 hasFastCapture(), tid, mFastTrackAvail); 6577 *flags = (audio_input_flags_t)(*flags & ~AUDIO_INPUT_FLAG_FAST); 6578 } 6579 } 6580 6581 // compute track buffer size in frames, and suggest the notification frame count 6582 if (*flags & AUDIO_INPUT_FLAG_FAST) { 6583 // fast track: frame count is exactly the pipe depth 6584 frameCount = mPipeFramesP2; 6585 // ignore requested notificationFrames, and always notify exactly once every HAL buffer 6586 *notificationFrames = mFrameCount; 6587 } else { 6588 // not fast track: max notification period is resampled equivalent of one HAL buffer time 6589 // or 20 ms if there is a fast capture 6590 // TODO This could be a roundupRatio inline, and const 6591 size_t maxNotificationFrames = ((int64_t) (hasFastCapture() ? mSampleRate/50 : mFrameCount) 6592 * sampleRate + mSampleRate - 1) / mSampleRate; 6593 // minimum number of notification periods is at least kMinNotifications, 6594 // and at least kMinMs rounded up to a whole notification period (minNotificationsByMs) 6595 static const size_t kMinNotifications = 3; 6596 static const uint32_t kMinMs = 30; 6597 // TODO This could be a roundupRatio inline 6598 const size_t minFramesByMs = (sampleRate * kMinMs + 1000 - 1) / 1000; 6599 // TODO This could be a roundupRatio inline 6600 const size_t minNotificationsByMs = (minFramesByMs + maxNotificationFrames - 1) / 6601 maxNotificationFrames; 6602 const size_t minFrameCount = maxNotificationFrames * 6603 max(kMinNotifications, minNotificationsByMs); 6604 frameCount = max(frameCount, minFrameCount); 6605 if (*notificationFrames == 0 || *notificationFrames > maxNotificationFrames) { 6606 *notificationFrames = maxNotificationFrames; 6607 } 6608 } 6609 *pFrameCount = frameCount; 6610 6611 lStatus = initCheck(); 6612 if (lStatus != NO_ERROR) { 6613 ALOGE("createRecordTrack_l() audio driver not initialized"); 6614 goto Exit; 6615 } 6616 6617 { // scope for mLock 6618 Mutex::Autolock _l(mLock); 6619 6620 track = new RecordTrack(this, client, sampleRate, 6621 format, channelMask, frameCount, NULL, sessionId, uid, 6622 *flags, TrackBase::TYPE_DEFAULT); 6623 6624 lStatus = track->initCheck(); 6625 if (lStatus != NO_ERROR) { 6626 ALOGE("createRecordTrack_l() initCheck failed %d; no control block?", lStatus); 6627 // track must be cleared from the caller as the caller has the AF lock 6628 goto Exit; 6629 } 6630 mTracks.add(track); 6631 6632 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 6633 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 6634 mAudioFlinger->btNrecIsOff(); 6635 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 6636 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 6637 6638 if ((*flags & AUDIO_INPUT_FLAG_FAST) && (tid != -1)) { 6639 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 6640 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 6641 // so ask activity manager to do this on our behalf 6642 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 6643 } 6644 } 6645 6646 lStatus = NO_ERROR; 6647 6648Exit: 6649 *status = lStatus; 6650 return track; 6651} 6652 6653status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, 6654 AudioSystem::sync_event_t event, 6655 audio_session_t triggerSession) 6656{ 6657 ALOGV("RecordThread::start event %d, triggerSession %d", event, triggerSession); 6658 sp<ThreadBase> strongMe = this; 6659 status_t status = NO_ERROR; 6660 6661 if (event == AudioSystem::SYNC_EVENT_NONE) { 6662 recordTrack->clearSyncStartEvent(); 6663 } else if (event != AudioSystem::SYNC_EVENT_SAME) { 6664 recordTrack->mSyncStartEvent = mAudioFlinger->createSyncEvent(event, 6665 triggerSession, 6666 recordTrack->sessionId(), 6667 syncStartEventCallback, 6668 recordTrack); 6669 // Sync event can be cancelled by the trigger session if the track is not in a 6670 // compatible state in which case we start record immediately 6671 if (recordTrack->mSyncStartEvent->isCancelled()) { 6672 recordTrack->clearSyncStartEvent(); 6673 } else { 6674 // do not wait for the event for more than AudioSystem::kSyncRecordStartTimeOutMs 6675 recordTrack->mFramesToDrop = - 6676 ((AudioSystem::kSyncRecordStartTimeOutMs * recordTrack->mSampleRate) / 1000); 6677 } 6678 } 6679 6680 { 6681 // This section is a rendezvous between binder thread executing start() and RecordThread 6682 AutoMutex lock(mLock); 6683 if (mActiveTracks.indexOf(recordTrack) >= 0) { 6684 if (recordTrack->mState == TrackBase::PAUSING) { 6685 ALOGV("active record track PAUSING -> ACTIVE"); 6686 recordTrack->mState = TrackBase::ACTIVE; 6687 } else { 6688 ALOGV("active record track state %d", recordTrack->mState); 6689 } 6690 return status; 6691 } 6692 6693 // TODO consider other ways of handling this, such as changing the state to :STARTING and 6694 // adding the track to mActiveTracks after returning from AudioSystem::startInput(), 6695 // or using a separate command thread 6696 recordTrack->mState = TrackBase::STARTING_1; 6697 mActiveTracks.add(recordTrack); 6698 mActiveTracksGen++; 6699 status_t status = NO_ERROR; 6700 if (recordTrack->isExternalTrack()) { 6701 mLock.unlock(); 6702 status = AudioSystem::startInput(mId, recordTrack->sessionId()); 6703 mLock.lock(); 6704 // FIXME should verify that recordTrack is still in mActiveTracks 6705 if (status != NO_ERROR) { 6706 mActiveTracks.remove(recordTrack); 6707 mActiveTracksGen++; 6708 recordTrack->clearSyncStartEvent(); 6709 ALOGV("RecordThread::start error %d", status); 6710 return status; 6711 } 6712 } 6713 // Catch up with current buffer indices if thread is already running. 6714 // This is what makes a new client discard all buffered data. If the track's mRsmpInFront 6715 // was initialized to some value closer to the thread's mRsmpInFront, then the track could 6716 // see previously buffered data before it called start(), but with greater risk of overrun. 6717 6718 recordTrack->mResamplerBufferProvider->reset(); 6719 // clear any converter state as new data will be discontinuous 6720 recordTrack->mRecordBufferConverter->reset(); 6721 recordTrack->mState = TrackBase::STARTING_2; 6722 // signal thread to start 6723 mWaitWorkCV.broadcast(); 6724 if (mActiveTracks.indexOf(recordTrack) < 0) { 6725 ALOGV("Record failed to start"); 6726 status = BAD_VALUE; 6727 goto startError; 6728 } 6729 return status; 6730 } 6731 6732startError: 6733 if (recordTrack->isExternalTrack()) { 6734 AudioSystem::stopInput(mId, recordTrack->sessionId()); 6735 } 6736 recordTrack->clearSyncStartEvent(); 6737 // FIXME I wonder why we do not reset the state here? 6738 return status; 6739} 6740 6741void AudioFlinger::RecordThread::syncStartEventCallback(const wp<SyncEvent>& event) 6742{ 6743 sp<SyncEvent> strongEvent = event.promote(); 6744 6745 if (strongEvent != 0) { 6746 sp<RefBase> ptr = strongEvent->cookie().promote(); 6747 if (ptr != 0) { 6748 RecordTrack *recordTrack = (RecordTrack *)ptr.get(); 6749 recordTrack->handleSyncStartEvent(strongEvent); 6750 } 6751 } 6752} 6753 6754bool AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 6755 ALOGV("RecordThread::stop"); 6756 AutoMutex _l(mLock); 6757 if (mActiveTracks.indexOf(recordTrack) != 0 || recordTrack->mState == TrackBase::PAUSING) { 6758 return false; 6759 } 6760 // note that threadLoop may still be processing the track at this point [without lock] 6761 recordTrack->mState = TrackBase::PAUSING; 6762 // signal thread to stop 6763 mWaitWorkCV.broadcast(); 6764 // do not wait for mStartStopCond if exiting 6765 if (exitPending()) { 6766 return true; 6767 } 6768 // FIXME incorrect usage of wait: no explicit predicate or loop 6769 mStartStopCond.wait(mLock); 6770 // if we have been restarted, recordTrack is in mActiveTracks here 6771 if (exitPending() || mActiveTracks.indexOf(recordTrack) != 0) { 6772 ALOGV("Record stopped OK"); 6773 return true; 6774 } 6775 return false; 6776} 6777 6778bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event __unused) const 6779{ 6780 return false; 6781} 6782 6783status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event __unused) 6784{ 6785#if 0 // This branch is currently dead code, but is preserved in case it will be needed in future 6786 if (!isValidSyncEvent(event)) { 6787 return BAD_VALUE; 6788 } 6789 6790 audio_session_t eventSession = event->triggerSession(); 6791 status_t ret = NAME_NOT_FOUND; 6792 6793 Mutex::Autolock _l(mLock); 6794 6795 for (size_t i = 0; i < mTracks.size(); i++) { 6796 sp<RecordTrack> track = mTracks[i]; 6797 if (eventSession == track->sessionId()) { 6798 (void) track->setSyncEvent(event); 6799 ret = NO_ERROR; 6800 } 6801 } 6802 return ret; 6803#else 6804 return BAD_VALUE; 6805#endif 6806} 6807 6808// destroyTrack_l() must be called with ThreadBase::mLock held 6809void AudioFlinger::RecordThread::destroyTrack_l(const sp<RecordTrack>& track) 6810{ 6811 track->terminate(); 6812 track->mState = TrackBase::STOPPED; 6813 // active tracks are removed by threadLoop() 6814 if (mActiveTracks.indexOf(track) < 0) { 6815 removeTrack_l(track); 6816 } 6817} 6818 6819void AudioFlinger::RecordThread::removeTrack_l(const sp<RecordTrack>& track) 6820{ 6821 mTracks.remove(track); 6822 // need anything related to effects here? 6823 if (track->isFastTrack()) { 6824 ALOG_ASSERT(!mFastTrackAvail); 6825 mFastTrackAvail = true; 6826 } 6827} 6828 6829void AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 6830{ 6831 dumpInternals(fd, args); 6832 dumpTracks(fd, args); 6833 dumpEffectChains(fd, args); 6834} 6835 6836void AudioFlinger::RecordThread::dumpInternals(int fd, const Vector<String16>& args) 6837{ 6838 dprintf(fd, "\nInput thread %p:\n", this); 6839 6840 dumpBase(fd, args); 6841 6842 if (mActiveTracks.size() == 0) { 6843 dprintf(fd, " No active record clients\n"); 6844 } 6845 dprintf(fd, " Fast capture thread: %s\n", hasFastCapture() ? "yes" : "no"); 6846 dprintf(fd, " Fast track available: %s\n", mFastTrackAvail ? "yes" : "no"); 6847 6848 // Make a non-atomic copy of fast capture dump state so it won't change underneath us 6849 // while we are dumping it. It may be inconsistent, but it won't mutate! 6850 // This is a large object so we place it on the heap. 6851 // FIXME 25972958: Need an intelligent copy constructor that does not touch unused pages. 6852 const FastCaptureDumpState *copy = new FastCaptureDumpState(mFastCaptureDumpState); 6853 copy->dump(fd); 6854 delete copy; 6855} 6856 6857void AudioFlinger::RecordThread::dumpTracks(int fd, const Vector<String16>& args __unused) 6858{ 6859 const size_t SIZE = 256; 6860 char buffer[SIZE]; 6861 String8 result; 6862 6863 size_t numtracks = mTracks.size(); 6864 size_t numactive = mActiveTracks.size(); 6865 size_t numactiveseen = 0; 6866 dprintf(fd, " %zu Tracks", numtracks); 6867 if (numtracks) { 6868 dprintf(fd, " of which %zu are active\n", numactive); 6869 RecordTrack::appendDumpHeader(result); 6870 for (size_t i = 0; i < numtracks ; ++i) { 6871 sp<RecordTrack> track = mTracks[i]; 6872 if (track != 0) { 6873 bool active = mActiveTracks.indexOf(track) >= 0; 6874 if (active) { 6875 numactiveseen++; 6876 } 6877 track->dump(buffer, SIZE, active); 6878 result.append(buffer); 6879 } 6880 } 6881 } else { 6882 dprintf(fd, "\n"); 6883 } 6884 6885 if (numactiveseen != numactive) { 6886 snprintf(buffer, SIZE, " The following tracks are in the active list but" 6887 " not in the track list\n"); 6888 result.append(buffer); 6889 RecordTrack::appendDumpHeader(result); 6890 for (size_t i = 0; i < numactive; ++i) { 6891 sp<RecordTrack> track = mActiveTracks[i]; 6892 if (mTracks.indexOf(track) < 0) { 6893 track->dump(buffer, SIZE, true); 6894 result.append(buffer); 6895 } 6896 } 6897 6898 } 6899 write(fd, result.string(), result.size()); 6900} 6901 6902 6903void AudioFlinger::RecordThread::ResamplerBufferProvider::reset() 6904{ 6905 sp<ThreadBase> threadBase = mRecordTrack->mThread.promote(); 6906 RecordThread *recordThread = (RecordThread *) threadBase.get(); 6907 mRsmpInFront = recordThread->mRsmpInRear; 6908 mRsmpInUnrel = 0; 6909} 6910 6911void AudioFlinger::RecordThread::ResamplerBufferProvider::sync( 6912 size_t *framesAvailable, bool *hasOverrun) 6913{ 6914 sp<ThreadBase> threadBase = mRecordTrack->mThread.promote(); 6915 RecordThread *recordThread = (RecordThread *) threadBase.get(); 6916 const int32_t rear = recordThread->mRsmpInRear; 6917 const int32_t front = mRsmpInFront; 6918 const ssize_t filled = rear - front; 6919 6920 size_t framesIn; 6921 bool overrun = false; 6922 if (filled < 0) { 6923 // should not happen, but treat like a massive overrun and re-sync 6924 framesIn = 0; 6925 mRsmpInFront = rear; 6926 overrun = true; 6927 } else if ((size_t) filled <= recordThread->mRsmpInFrames) { 6928 framesIn = (size_t) filled; 6929 } else { 6930 // client is not keeping up with server, but give it latest data 6931 framesIn = recordThread->mRsmpInFrames; 6932 mRsmpInFront = /* front = */ rear - framesIn; 6933 overrun = true; 6934 } 6935 if (framesAvailable != NULL) { 6936 *framesAvailable = framesIn; 6937 } 6938 if (hasOverrun != NULL) { 6939 *hasOverrun = overrun; 6940 } 6941} 6942 6943// AudioBufferProvider interface 6944status_t AudioFlinger::RecordThread::ResamplerBufferProvider::getNextBuffer( 6945 AudioBufferProvider::Buffer* buffer) 6946{ 6947 sp<ThreadBase> threadBase = mRecordTrack->mThread.promote(); 6948 if (threadBase == 0) { 6949 buffer->frameCount = 0; 6950 buffer->raw = NULL; 6951 return NOT_ENOUGH_DATA; 6952 } 6953 RecordThread *recordThread = (RecordThread *) threadBase.get(); 6954 int32_t rear = recordThread->mRsmpInRear; 6955 int32_t front = mRsmpInFront; 6956 ssize_t filled = rear - front; 6957 // FIXME should not be P2 (don't want to increase latency) 6958 // FIXME if client not keeping up, discard 6959 LOG_ALWAYS_FATAL_IF(!(0 <= filled && (size_t) filled <= recordThread->mRsmpInFrames)); 6960 // 'filled' may be non-contiguous, so return only the first contiguous chunk 6961 front &= recordThread->mRsmpInFramesP2 - 1; 6962 size_t part1 = recordThread->mRsmpInFramesP2 - front; 6963 if (part1 > (size_t) filled) { 6964 part1 = filled; 6965 } 6966 size_t ask = buffer->frameCount; 6967 ALOG_ASSERT(ask > 0); 6968 if (part1 > ask) { 6969 part1 = ask; 6970 } 6971 if (part1 == 0) { 6972 // out of data is fine since the resampler will return a short-count. 6973 buffer->raw = NULL; 6974 buffer->frameCount = 0; 6975 mRsmpInUnrel = 0; 6976 return NOT_ENOUGH_DATA; 6977 } 6978 6979 buffer->raw = (uint8_t*)recordThread->mRsmpInBuffer + front * recordThread->mFrameSize; 6980 buffer->frameCount = part1; 6981 mRsmpInUnrel = part1; 6982 return NO_ERROR; 6983} 6984 6985// AudioBufferProvider interface 6986void AudioFlinger::RecordThread::ResamplerBufferProvider::releaseBuffer( 6987 AudioBufferProvider::Buffer* buffer) 6988{ 6989 size_t stepCount = buffer->frameCount; 6990 if (stepCount == 0) { 6991 return; 6992 } 6993 ALOG_ASSERT(stepCount <= mRsmpInUnrel); 6994 mRsmpInUnrel -= stepCount; 6995 mRsmpInFront += stepCount; 6996 buffer->raw = NULL; 6997 buffer->frameCount = 0; 6998} 6999 7000AudioFlinger::RecordThread::RecordBufferConverter::RecordBufferConverter( 7001 audio_channel_mask_t srcChannelMask, audio_format_t srcFormat, 7002 uint32_t srcSampleRate, 7003 audio_channel_mask_t dstChannelMask, audio_format_t dstFormat, 7004 uint32_t dstSampleRate) : 7005 mSrcChannelMask(AUDIO_CHANNEL_INVALID), // updateParameters will set following vars 7006 // mSrcFormat 7007 // mSrcSampleRate 7008 // mDstChannelMask 7009 // mDstFormat 7010 // mDstSampleRate 7011 // mSrcChannelCount 7012 // mDstChannelCount 7013 // mDstFrameSize 7014 mBuf(NULL), mBufFrames(0), mBufFrameSize(0), 7015 mResampler(NULL), 7016 mIsLegacyDownmix(false), 7017 mIsLegacyUpmix(false), 7018 mRequiresFloat(false), 7019 mInputConverterProvider(NULL) 7020{ 7021 (void)updateParameters(srcChannelMask, srcFormat, srcSampleRate, 7022 dstChannelMask, dstFormat, dstSampleRate); 7023} 7024 7025AudioFlinger::RecordThread::RecordBufferConverter::~RecordBufferConverter() { 7026 free(mBuf); 7027 delete mResampler; 7028 delete mInputConverterProvider; 7029} 7030 7031size_t AudioFlinger::RecordThread::RecordBufferConverter::convert(void *dst, 7032 AudioBufferProvider *provider, size_t frames) 7033{ 7034 if (mInputConverterProvider != NULL) { 7035 mInputConverterProvider->setBufferProvider(provider); 7036 provider = mInputConverterProvider; 7037 } 7038 7039 if (mResampler == NULL) { 7040 ALOGVV("NO RESAMPLING sampleRate:%u mSrcFormat:%#x mDstFormat:%#x", 7041 mSrcSampleRate, mSrcFormat, mDstFormat); 7042 7043 AudioBufferProvider::Buffer buffer; 7044 for (size_t i = frames; i > 0; ) { 7045 buffer.frameCount = i; 7046 status_t status = provider->getNextBuffer(&buffer); 7047 if (status != OK || buffer.frameCount == 0) { 7048 frames -= i; // cannot fill request. 7049 break; 7050 } 7051 // format convert to destination buffer 7052 convertNoResampler(dst, buffer.raw, buffer.frameCount); 7053 7054 dst = (int8_t*)dst + buffer.frameCount * mDstFrameSize; 7055 i -= buffer.frameCount; 7056 provider->releaseBuffer(&buffer); 7057 } 7058 } else { 7059 ALOGVV("RESAMPLING mSrcSampleRate:%u mDstSampleRate:%u mSrcFormat:%#x mDstFormat:%#x", 7060 mSrcSampleRate, mDstSampleRate, mSrcFormat, mDstFormat); 7061 7062 // reallocate buffer if needed 7063 if (mBufFrameSize != 0 && mBufFrames < frames) { 7064 free(mBuf); 7065 mBufFrames = frames; 7066 (void)posix_memalign(&mBuf, 32, mBufFrames * mBufFrameSize); 7067 } 7068 // resampler accumulates, but we only have one source track 7069 memset(mBuf, 0, frames * mBufFrameSize); 7070 frames = mResampler->resample((int32_t*)mBuf, frames, provider); 7071 // format convert to destination buffer 7072 convertResampler(dst, mBuf, frames); 7073 } 7074 return frames; 7075} 7076 7077status_t AudioFlinger::RecordThread::RecordBufferConverter::updateParameters( 7078 audio_channel_mask_t srcChannelMask, audio_format_t srcFormat, 7079 uint32_t srcSampleRate, 7080 audio_channel_mask_t dstChannelMask, audio_format_t dstFormat, 7081 uint32_t dstSampleRate) 7082{ 7083 // quick evaluation if there is any change. 7084 if (mSrcFormat == srcFormat 7085 && mSrcChannelMask == srcChannelMask 7086 && mSrcSampleRate == srcSampleRate 7087 && mDstFormat == dstFormat 7088 && mDstChannelMask == dstChannelMask 7089 && mDstSampleRate == dstSampleRate) { 7090 return NO_ERROR; 7091 } 7092 7093 ALOGV("RecordBufferConverter updateParameters srcMask:%#x dstMask:%#x" 7094 " srcFormat:%#x dstFormat:%#x srcRate:%u dstRate:%u", 7095 srcChannelMask, dstChannelMask, srcFormat, dstFormat, srcSampleRate, dstSampleRate); 7096 const bool valid = 7097 audio_is_input_channel(srcChannelMask) 7098 && audio_is_input_channel(dstChannelMask) 7099 && audio_is_valid_format(srcFormat) && audio_is_linear_pcm(srcFormat) 7100 && audio_is_valid_format(dstFormat) && audio_is_linear_pcm(dstFormat) 7101 && (srcSampleRate <= dstSampleRate * AUDIO_RESAMPLER_DOWN_RATIO_MAX) 7102 ; // no upsampling checks for now 7103 if (!valid) { 7104 return BAD_VALUE; 7105 } 7106 7107 mSrcFormat = srcFormat; 7108 mSrcChannelMask = srcChannelMask; 7109 mSrcSampleRate = srcSampleRate; 7110 mDstFormat = dstFormat; 7111 mDstChannelMask = dstChannelMask; 7112 mDstSampleRate = dstSampleRate; 7113 7114 // compute derived parameters 7115 mSrcChannelCount = audio_channel_count_from_in_mask(srcChannelMask); 7116 mDstChannelCount = audio_channel_count_from_in_mask(dstChannelMask); 7117 mDstFrameSize = mDstChannelCount * audio_bytes_per_sample(mDstFormat); 7118 7119 // do we need to resample? 7120 delete mResampler; 7121 mResampler = NULL; 7122 if (mSrcSampleRate != mDstSampleRate) { 7123 mResampler = AudioResampler::create(AUDIO_FORMAT_PCM_FLOAT, 7124 mSrcChannelCount, mDstSampleRate); 7125 mResampler->setSampleRate(mSrcSampleRate); 7126 mResampler->setVolume(AudioMixer::UNITY_GAIN_FLOAT, AudioMixer::UNITY_GAIN_FLOAT); 7127 } 7128 7129 // are we running legacy channel conversion modes? 7130 mIsLegacyDownmix = (mSrcChannelMask == AUDIO_CHANNEL_IN_STEREO 7131 || mSrcChannelMask == AUDIO_CHANNEL_IN_FRONT_BACK) 7132 && mDstChannelMask == AUDIO_CHANNEL_IN_MONO; 7133 mIsLegacyUpmix = mSrcChannelMask == AUDIO_CHANNEL_IN_MONO 7134 && (mDstChannelMask == AUDIO_CHANNEL_IN_STEREO 7135 || mDstChannelMask == AUDIO_CHANNEL_IN_FRONT_BACK); 7136 7137 // do we need to process in float? 7138 mRequiresFloat = mResampler != NULL || mIsLegacyDownmix || mIsLegacyUpmix; 7139 7140 // do we need a staging buffer to convert for destination (we can still optimize this)? 7141 // we use mBufFrameSize > 0 to indicate both frame size as well as buffer necessity 7142 if (mResampler != NULL) { 7143 mBufFrameSize = max(mSrcChannelCount, FCC_2) 7144 * audio_bytes_per_sample(AUDIO_FORMAT_PCM_FLOAT); 7145 } else if (mIsLegacyUpmix || mIsLegacyDownmix) { // legacy modes always float 7146 mBufFrameSize = mDstChannelCount * audio_bytes_per_sample(AUDIO_FORMAT_PCM_FLOAT); 7147 } else if (mSrcChannelMask != mDstChannelMask && mDstFormat != mSrcFormat) { 7148 mBufFrameSize = mDstChannelCount * audio_bytes_per_sample(mSrcFormat); 7149 } else { 7150 mBufFrameSize = 0; 7151 } 7152 mBufFrames = 0; // force the buffer to be resized. 7153 7154 // do we need an input converter buffer provider to give us float? 7155 delete mInputConverterProvider; 7156 mInputConverterProvider = NULL; 7157 if (mRequiresFloat && mSrcFormat != AUDIO_FORMAT_PCM_FLOAT) { 7158 mInputConverterProvider = new ReformatBufferProvider( 7159 audio_channel_count_from_in_mask(mSrcChannelMask), 7160 mSrcFormat, 7161 AUDIO_FORMAT_PCM_FLOAT, 7162 256 /* provider buffer frame count */); 7163 } 7164 7165 // do we need a remixer to do channel mask conversion 7166 if (!mIsLegacyDownmix && !mIsLegacyUpmix && mSrcChannelMask != mDstChannelMask) { 7167 (void) memcpy_by_index_array_initialization_from_channel_mask( 7168 mIdxAry, ARRAY_SIZE(mIdxAry), mDstChannelMask, mSrcChannelMask); 7169 } 7170 return NO_ERROR; 7171} 7172 7173void AudioFlinger::RecordThread::RecordBufferConverter::convertNoResampler( 7174 void *dst, const void *src, size_t frames) 7175{ 7176 // src is native type unless there is legacy upmix or downmix, whereupon it is float. 7177 if (mBufFrameSize != 0 && mBufFrames < frames) { 7178 free(mBuf); 7179 mBufFrames = frames; 7180 (void)posix_memalign(&mBuf, 32, mBufFrames * mBufFrameSize); 7181 } 7182 // do we need to do legacy upmix and downmix? 7183 if (mIsLegacyUpmix || mIsLegacyDownmix) { 7184 void *dstBuf = mBuf != NULL ? mBuf : dst; 7185 if (mIsLegacyUpmix) { 7186 upmix_to_stereo_float_from_mono_float((float *)dstBuf, 7187 (const float *)src, frames); 7188 } else /*mIsLegacyDownmix */ { 7189 downmix_to_mono_float_from_stereo_float((float *)dstBuf, 7190 (const float *)src, frames); 7191 } 7192 if (mBuf != NULL) { 7193 memcpy_by_audio_format(dst, mDstFormat, mBuf, AUDIO_FORMAT_PCM_FLOAT, 7194 frames * mDstChannelCount); 7195 } 7196 return; 7197 } 7198 // do we need to do channel mask conversion? 7199 if (mSrcChannelMask != mDstChannelMask) { 7200 void *dstBuf = mBuf != NULL ? mBuf : dst; 7201 memcpy_by_index_array(dstBuf, mDstChannelCount, 7202 src, mSrcChannelCount, mIdxAry, audio_bytes_per_sample(mSrcFormat), frames); 7203 if (dstBuf == dst) { 7204 return; // format is the same 7205 } 7206 } 7207 // convert to destination buffer 7208 const void *convertBuf = mBuf != NULL ? mBuf : src; 7209 memcpy_by_audio_format(dst, mDstFormat, convertBuf, mSrcFormat, 7210 frames * mDstChannelCount); 7211} 7212 7213void AudioFlinger::RecordThread::RecordBufferConverter::convertResampler( 7214 void *dst, /*not-a-const*/ void *src, size_t frames) 7215{ 7216 // src buffer format is ALWAYS float when entering this routine 7217 if (mIsLegacyUpmix) { 7218 ; // mono to stereo already handled by resampler 7219 } else if (mIsLegacyDownmix 7220 || (mSrcChannelMask == mDstChannelMask && mSrcChannelCount == 1)) { 7221 // the resampler outputs stereo for mono input channel (a feature?) 7222 // must convert to mono 7223 downmix_to_mono_float_from_stereo_float((float *)src, 7224 (const float *)src, frames); 7225 } else if (mSrcChannelMask != mDstChannelMask) { 7226 // convert to mono channel again for channel mask conversion (could be skipped 7227 // with further optimization). 7228 if (mSrcChannelCount == 1) { 7229 downmix_to_mono_float_from_stereo_float((float *)src, 7230 (const float *)src, frames); 7231 } 7232 // convert to destination format (in place, OK as float is larger than other types) 7233 if (mDstFormat != AUDIO_FORMAT_PCM_FLOAT) { 7234 memcpy_by_audio_format(src, mDstFormat, src, AUDIO_FORMAT_PCM_FLOAT, 7235 frames * mSrcChannelCount); 7236 } 7237 // channel convert and save to dst 7238 memcpy_by_index_array(dst, mDstChannelCount, 7239 src, mSrcChannelCount, mIdxAry, audio_bytes_per_sample(mDstFormat), frames); 7240 return; 7241 } 7242 // convert to destination format and save to dst 7243 memcpy_by_audio_format(dst, mDstFormat, src, AUDIO_FORMAT_PCM_FLOAT, 7244 frames * mDstChannelCount); 7245} 7246 7247bool AudioFlinger::RecordThread::checkForNewParameter_l(const String8& keyValuePair, 7248 status_t& status) 7249{ 7250 bool reconfig = false; 7251 7252 status = NO_ERROR; 7253 7254 audio_format_t reqFormat = mFormat; 7255 uint32_t samplingRate = mSampleRate; 7256 // TODO this may change if we want to support capture from HDMI PCM multi channel (e.g on TVs). 7257 audio_channel_mask_t channelMask = audio_channel_in_mask_from_count(mChannelCount); 7258 7259 AudioParameter param = AudioParameter(keyValuePair); 7260 int value; 7261 7262 // scope for AutoPark extends to end of method 7263 AutoPark<FastCapture> park(mFastCapture); 7264 7265 // TODO Investigate when this code runs. Check with audio policy when a sample rate and 7266 // channel count change can be requested. Do we mandate the first client defines the 7267 // HAL sampling rate and channel count or do we allow changes on the fly? 7268 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 7269 samplingRate = value; 7270 reconfig = true; 7271 } 7272 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 7273 if (!audio_is_linear_pcm((audio_format_t) value)) { 7274 status = BAD_VALUE; 7275 } else { 7276 reqFormat = (audio_format_t) value; 7277 reconfig = true; 7278 } 7279 } 7280 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 7281 audio_channel_mask_t mask = (audio_channel_mask_t) value; 7282 if (!audio_is_input_channel(mask) || 7283 audio_channel_count_from_in_mask(mask) > FCC_8) { 7284 status = BAD_VALUE; 7285 } else { 7286 channelMask = mask; 7287 reconfig = true; 7288 } 7289 } 7290 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 7291 // do not accept frame count changes if tracks are open as the track buffer 7292 // size depends on frame count and correct behavior would not be guaranteed 7293 // if frame count is changed after track creation 7294 if (mActiveTracks.size() > 0) { 7295 status = INVALID_OPERATION; 7296 } else { 7297 reconfig = true; 7298 } 7299 } 7300 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 7301 // forward device change to effects that have requested to be 7302 // aware of attached audio device. 7303 for (size_t i = 0; i < mEffectChains.size(); i++) { 7304 mEffectChains[i]->setDevice_l(value); 7305 } 7306 7307 // store input device and output device but do not forward output device to audio HAL. 7308 // Note that status is ignored by the caller for output device 7309 // (see AudioFlinger::setParameters() 7310 if (audio_is_output_devices(value)) { 7311 mOutDevice = value; 7312 status = BAD_VALUE; 7313 } else { 7314 mInDevice = value; 7315 if (value != AUDIO_DEVICE_NONE) { 7316 mPrevInDevice = value; 7317 } 7318 // disable AEC and NS if the device is a BT SCO headset supporting those 7319 // pre processings 7320 if (mTracks.size() > 0) { 7321 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 7322 mAudioFlinger->btNrecIsOff(); 7323 for (size_t i = 0; i < mTracks.size(); i++) { 7324 sp<RecordTrack> track = mTracks[i]; 7325 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId()); 7326 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId()); 7327 } 7328 } 7329 } 7330 } 7331 if (param.getInt(String8(AudioParameter::keyInputSource), value) == NO_ERROR && 7332 mAudioSource != (audio_source_t)value) { 7333 // forward device change to effects that have requested to be 7334 // aware of attached audio device. 7335 for (size_t i = 0; i < mEffectChains.size(); i++) { 7336 mEffectChains[i]->setAudioSource_l((audio_source_t)value); 7337 } 7338 mAudioSource = (audio_source_t)value; 7339 } 7340 7341 if (status == NO_ERROR) { 7342 status = mInput->stream->common.set_parameters(&mInput->stream->common, 7343 keyValuePair.string()); 7344 if (status == INVALID_OPERATION) { 7345 inputStandBy(); 7346 status = mInput->stream->common.set_parameters(&mInput->stream->common, 7347 keyValuePair.string()); 7348 } 7349 if (reconfig) { 7350 if (status == BAD_VALUE && 7351 audio_is_linear_pcm(mInput->stream->common.get_format(&mInput->stream->common)) && 7352 audio_is_linear_pcm(reqFormat) && 7353 (mInput->stream->common.get_sample_rate(&mInput->stream->common) 7354 <= (AUDIO_RESAMPLER_DOWN_RATIO_MAX * samplingRate)) && 7355 audio_channel_count_from_in_mask( 7356 mInput->stream->common.get_channels(&mInput->stream->common)) <= FCC_8) { 7357 status = NO_ERROR; 7358 } 7359 if (status == NO_ERROR) { 7360 readInputParameters_l(); 7361 sendIoConfigEvent_l(AUDIO_INPUT_CONFIG_CHANGED); 7362 } 7363 } 7364 } 7365 7366 return reconfig; 7367} 7368 7369String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 7370{ 7371 Mutex::Autolock _l(mLock); 7372 if (initCheck() != NO_ERROR) { 7373 return String8(); 7374 } 7375 7376 char *s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 7377 const String8 out_s8(s); 7378 free(s); 7379 return out_s8; 7380} 7381 7382void AudioFlinger::RecordThread::ioConfigChanged(audio_io_config_event event, pid_t pid) { 7383 sp<AudioIoDescriptor> desc = new AudioIoDescriptor(); 7384 7385 desc->mIoHandle = mId; 7386 7387 switch (event) { 7388 case AUDIO_INPUT_OPENED: 7389 case AUDIO_INPUT_CONFIG_CHANGED: 7390 desc->mPatch = mPatch; 7391 desc->mChannelMask = mChannelMask; 7392 desc->mSamplingRate = mSampleRate; 7393 desc->mFormat = mFormat; 7394 desc->mFrameCount = mFrameCount; 7395 desc->mFrameCountHAL = mFrameCount; 7396 desc->mLatency = 0; 7397 break; 7398 7399 case AUDIO_INPUT_CLOSED: 7400 default: 7401 break; 7402 } 7403 mAudioFlinger->ioConfigChanged(event, desc, pid); 7404} 7405 7406void AudioFlinger::RecordThread::readInputParameters_l() 7407{ 7408 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 7409 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 7410 mChannelCount = audio_channel_count_from_in_mask(mChannelMask); 7411 if (mChannelCount > FCC_8) { 7412 ALOGE("HAL channel count %d > %d", mChannelCount, FCC_8); 7413 } 7414 mHALFormat = mInput->stream->common.get_format(&mInput->stream->common); 7415 mFormat = mHALFormat; 7416 if (!audio_is_linear_pcm(mFormat)) { 7417 ALOGE("HAL format %#x is not linear pcm", mFormat); 7418 } 7419 mFrameSize = audio_stream_in_frame_size(mInput->stream); 7420 mBufferSize = mInput->stream->common.get_buffer_size(&mInput->stream->common); 7421 mFrameCount = mBufferSize / mFrameSize; 7422 // This is the formula for calculating the temporary buffer size. 7423 // With 7 HAL buffers, we can guarantee ability to down-sample the input by ratio of 6:1 to 7424 // 1 full output buffer, regardless of the alignment of the available input. 7425 // The value is somewhat arbitrary, and could probably be even larger. 7426 // A larger value should allow more old data to be read after a track calls start(), 7427 // without increasing latency. 7428 // 7429 // Note this is independent of the maximum downsampling ratio permitted for capture. 7430 mRsmpInFrames = mFrameCount * 7; 7431 mRsmpInFramesP2 = roundup(mRsmpInFrames); 7432 free(mRsmpInBuffer); 7433 mRsmpInBuffer = NULL; 7434 7435 // TODO optimize audio capture buffer sizes ... 7436 // Here we calculate the size of the sliding buffer used as a source 7437 // for resampling. mRsmpInFramesP2 is currently roundup(mFrameCount * 7). 7438 // For current HAL frame counts, this is usually 2048 = 40 ms. It would 7439 // be better to have it derived from the pipe depth in the long term. 7440 // The current value is higher than necessary. However it should not add to latency. 7441 7442 // Over-allocate beyond mRsmpInFramesP2 to permit a HAL read past end of buffer 7443 size_t bufferSize = (mRsmpInFramesP2 + mFrameCount - 1) * mFrameSize; 7444 (void)posix_memalign(&mRsmpInBuffer, 32, bufferSize); 7445 memset(mRsmpInBuffer, 0, bufferSize); // if posix_memalign fails, will segv here. 7446 7447 // AudioRecord mSampleRate and mChannelCount are constant due to AudioRecord API constraints. 7448 // But if thread's mSampleRate or mChannelCount changes, how will that affect active tracks? 7449} 7450 7451uint32_t AudioFlinger::RecordThread::getInputFramesLost() 7452{ 7453 Mutex::Autolock _l(mLock); 7454 if (initCheck() != NO_ERROR) { 7455 return 0; 7456 } 7457 7458 return mInput->stream->get_input_frames_lost(mInput->stream); 7459} 7460 7461// hasAudioSession_l() must be called with ThreadBase::mLock held 7462uint32_t AudioFlinger::RecordThread::hasAudioSession_l(audio_session_t sessionId) const 7463{ 7464 uint32_t result = 0; 7465 if (getEffectChain_l(sessionId) != 0) { 7466 result = EFFECT_SESSION; 7467 } 7468 7469 for (size_t i = 0; i < mTracks.size(); ++i) { 7470 if (sessionId == mTracks[i]->sessionId()) { 7471 result |= TRACK_SESSION; 7472 if (mTracks[i]->isFastTrack()) { 7473 result |= FAST_SESSION; 7474 } 7475 break; 7476 } 7477 } 7478 7479 return result; 7480} 7481 7482KeyedVector<audio_session_t, bool> AudioFlinger::RecordThread::sessionIds() const 7483{ 7484 KeyedVector<audio_session_t, bool> ids; 7485 Mutex::Autolock _l(mLock); 7486 for (size_t j = 0; j < mTracks.size(); ++j) { 7487 sp<RecordThread::RecordTrack> track = mTracks[j]; 7488 audio_session_t sessionId = track->sessionId(); 7489 if (ids.indexOfKey(sessionId) < 0) { 7490 ids.add(sessionId, true); 7491 } 7492 } 7493 return ids; 7494} 7495 7496AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 7497{ 7498 Mutex::Autolock _l(mLock); 7499 AudioStreamIn *input = mInput; 7500 mInput = NULL; 7501 return input; 7502} 7503 7504// this method must always be called either with ThreadBase mLock held or inside the thread loop 7505audio_stream_t* AudioFlinger::RecordThread::stream() const 7506{ 7507 if (mInput == NULL) { 7508 return NULL; 7509 } 7510 return &mInput->stream->common; 7511} 7512 7513status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 7514{ 7515 // only one chain per input thread 7516 if (mEffectChains.size() != 0) { 7517 ALOGW("addEffectChain_l() already one chain %p on thread %p", chain.get(), this); 7518 return INVALID_OPERATION; 7519 } 7520 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 7521 chain->setThread(this); 7522 chain->setInBuffer(NULL); 7523 chain->setOutBuffer(NULL); 7524 7525 checkSuspendOnAddEffectChain_l(chain); 7526 7527 // make sure enabled pre processing effects state is communicated to the HAL as we 7528 // just moved them to a new input stream. 7529 chain->syncHalEffectsState(); 7530 7531 mEffectChains.add(chain); 7532 7533 return NO_ERROR; 7534} 7535 7536size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 7537{ 7538 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 7539 ALOGW_IF(mEffectChains.size() != 1, 7540 "removeEffectChain_l() %p invalid chain size %zu on thread %p", 7541 chain.get(), mEffectChains.size(), this); 7542 if (mEffectChains.size() == 1) { 7543 mEffectChains.removeAt(0); 7544 } 7545 return 0; 7546} 7547 7548status_t AudioFlinger::RecordThread::createAudioPatch_l(const struct audio_patch *patch, 7549 audio_patch_handle_t *handle) 7550{ 7551 status_t status = NO_ERROR; 7552 7553 // store new device and send to effects 7554 mInDevice = patch->sources[0].ext.device.type; 7555 mPatch = *patch; 7556 for (size_t i = 0; i < mEffectChains.size(); i++) { 7557 mEffectChains[i]->setDevice_l(mInDevice); 7558 } 7559 7560 // disable AEC and NS if the device is a BT SCO headset supporting those 7561 // pre processings 7562 if (mTracks.size() > 0) { 7563 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 7564 mAudioFlinger->btNrecIsOff(); 7565 for (size_t i = 0; i < mTracks.size(); i++) { 7566 sp<RecordTrack> track = mTracks[i]; 7567 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId()); 7568 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId()); 7569 } 7570 } 7571 7572 // store new source and send to effects 7573 if (mAudioSource != patch->sinks[0].ext.mix.usecase.source) { 7574 mAudioSource = patch->sinks[0].ext.mix.usecase.source; 7575 for (size_t i = 0; i < mEffectChains.size(); i++) { 7576 mEffectChains[i]->setAudioSource_l(mAudioSource); 7577 } 7578 } 7579 7580 if (mInput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 7581 audio_hw_device_t *hwDevice = mInput->audioHwDev->hwDevice(); 7582 status = hwDevice->create_audio_patch(hwDevice, 7583 patch->num_sources, 7584 patch->sources, 7585 patch->num_sinks, 7586 patch->sinks, 7587 handle); 7588 } else { 7589 char *address; 7590 if (strcmp(patch->sources[0].ext.device.address, "") != 0) { 7591 address = audio_device_address_to_parameter( 7592 patch->sources[0].ext.device.type, 7593 patch->sources[0].ext.device.address); 7594 } else { 7595 address = (char *)calloc(1, 1); 7596 } 7597 AudioParameter param = AudioParameter(String8(address)); 7598 free(address); 7599 param.addInt(String8(AUDIO_PARAMETER_STREAM_ROUTING), 7600 (int)patch->sources[0].ext.device.type); 7601 param.addInt(String8(AUDIO_PARAMETER_STREAM_INPUT_SOURCE), 7602 (int)patch->sinks[0].ext.mix.usecase.source); 7603 status = mInput->stream->common.set_parameters(&mInput->stream->common, 7604 param.toString().string()); 7605 *handle = AUDIO_PATCH_HANDLE_NONE; 7606 } 7607 7608 if (mInDevice != mPrevInDevice) { 7609 sendIoConfigEvent_l(AUDIO_INPUT_CONFIG_CHANGED); 7610 mPrevInDevice = mInDevice; 7611 } 7612 7613 return status; 7614} 7615 7616status_t AudioFlinger::RecordThread::releaseAudioPatch_l(const audio_patch_handle_t handle) 7617{ 7618 status_t status = NO_ERROR; 7619 7620 mInDevice = AUDIO_DEVICE_NONE; 7621 7622 if (mInput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 7623 audio_hw_device_t *hwDevice = mInput->audioHwDev->hwDevice(); 7624 status = hwDevice->release_audio_patch(hwDevice, handle); 7625 } else { 7626 AudioParameter param; 7627 param.addInt(String8(AUDIO_PARAMETER_STREAM_ROUTING), 0); 7628 status = mInput->stream->common.set_parameters(&mInput->stream->common, 7629 param.toString().string()); 7630 } 7631 return status; 7632} 7633 7634void AudioFlinger::RecordThread::addPatchRecord(const sp<PatchRecord>& record) 7635{ 7636 Mutex::Autolock _l(mLock); 7637 mTracks.add(record); 7638} 7639 7640void AudioFlinger::RecordThread::deletePatchRecord(const sp<PatchRecord>& record) 7641{ 7642 Mutex::Autolock _l(mLock); 7643 destroyTrack_l(record); 7644} 7645 7646void AudioFlinger::RecordThread::getAudioPortConfig(struct audio_port_config *config) 7647{ 7648 ThreadBase::getAudioPortConfig(config); 7649 config->role = AUDIO_PORT_ROLE_SINK; 7650 config->ext.mix.hw_module = mInput->audioHwDev->handle(); 7651 config->ext.mix.usecase.source = mAudioSource; 7652} 7653 7654} // namespace android 7655