Threads.cpp revision 7bafda7a153cda393ce018241fa7304845225ae5
1/* 2** 3** Copyright 2012, The Android Open Source Project 4** 5** Licensed under the Apache License, Version 2.0 (the "License"); 6** you may not use this file except in compliance with the License. 7** You may obtain a copy of the License at 8** 9** http://www.apache.org/licenses/LICENSE-2.0 10** 11** Unless required by applicable law or agreed to in writing, software 12** distributed under the License is distributed on an "AS IS" BASIS, 13** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. 14** See the License for the specific language governing permissions and 15** limitations under the License. 16*/ 17 18 19#define LOG_TAG "AudioFlinger" 20//#define LOG_NDEBUG 0 21#define ATRACE_TAG ATRACE_TAG_AUDIO 22 23#include "Configuration.h" 24#include <math.h> 25#include <fcntl.h> 26#include <linux/futex.h> 27#include <sys/stat.h> 28#include <sys/syscall.h> 29#include <cutils/properties.h> 30#include <media/AudioParameter.h> 31#include <media/AudioResamplerPublic.h> 32#include <utils/Log.h> 33#include <utils/Trace.h> 34 35#include <private/media/AudioTrackShared.h> 36#include <hardware/audio.h> 37#include <audio_effects/effect_ns.h> 38#include <audio_effects/effect_aec.h> 39#include <audio_utils/conversion.h> 40#include <audio_utils/primitives.h> 41#include <audio_utils/format.h> 42#include <audio_utils/minifloat.h> 43 44// NBAIO implementations 45#include <media/nbaio/AudioStreamInSource.h> 46#include <media/nbaio/AudioStreamOutSink.h> 47#include <media/nbaio/MonoPipe.h> 48#include <media/nbaio/MonoPipeReader.h> 49#include <media/nbaio/Pipe.h> 50#include <media/nbaio/PipeReader.h> 51#include <media/nbaio/SourceAudioBufferProvider.h> 52#include <mediautils/BatteryNotifier.h> 53 54#include <powermanager/PowerManager.h> 55 56#include "AudioFlinger.h" 57#include "AudioMixer.h" 58#include "BufferProviders.h" 59#include "FastMixer.h" 60#include "FastCapture.h" 61#include "ServiceUtilities.h" 62#include "mediautils/SchedulingPolicyService.h" 63 64#ifdef ADD_BATTERY_DATA 65#include <media/IMediaPlayerService.h> 66#include <media/IMediaDeathNotifier.h> 67#endif 68 69#ifdef DEBUG_CPU_USAGE 70#include <cpustats/CentralTendencyStatistics.h> 71#include <cpustats/ThreadCpuUsage.h> 72#endif 73 74#include "AutoPark.h" 75 76// ---------------------------------------------------------------------------- 77 78// Note: the following macro is used for extremely verbose logging message. In 79// order to run with ALOG_ASSERT turned on, we need to have LOG_NDEBUG set to 80// 0; but one side effect of this is to turn all LOGV's as well. Some messages 81// are so verbose that we want to suppress them even when we have ALOG_ASSERT 82// turned on. Do not uncomment the #def below unless you really know what you 83// are doing and want to see all of the extremely verbose messages. 84//#define VERY_VERY_VERBOSE_LOGGING 85#ifdef VERY_VERY_VERBOSE_LOGGING 86#define ALOGVV ALOGV 87#else 88#define ALOGVV(a...) do { } while(0) 89#endif 90 91// TODO: Move these macro/inlines to a header file. 92#define max(a, b) ((a) > (b) ? (a) : (b)) 93template <typename T> 94static inline T min(const T& a, const T& b) 95{ 96 return a < b ? a : b; 97} 98 99#ifndef ARRAY_SIZE 100#define ARRAY_SIZE(a) (sizeof(a) / sizeof(a[0])) 101#endif 102 103namespace android { 104 105// retry counts for buffer fill timeout 106// 50 * ~20msecs = 1 second 107static const int8_t kMaxTrackRetries = 50; 108static const int8_t kMaxTrackStartupRetries = 50; 109// allow less retry attempts on direct output thread. 110// direct outputs can be a scarce resource in audio hardware and should 111// be released as quickly as possible. 112static const int8_t kMaxTrackRetriesDirect = 2; 113 114 115 116// don't warn about blocked writes or record buffer overflows more often than this 117static const nsecs_t kWarningThrottleNs = seconds(5); 118 119// RecordThread loop sleep time upon application overrun or audio HAL read error 120static const int kRecordThreadSleepUs = 5000; 121 122// maximum time to wait in sendConfigEvent_l() for a status to be received 123static const nsecs_t kConfigEventTimeoutNs = seconds(2); 124 125// minimum sleep time for the mixer thread loop when tracks are active but in underrun 126static const uint32_t kMinThreadSleepTimeUs = 5000; 127// maximum divider applied to the active sleep time in the mixer thread loop 128static const uint32_t kMaxThreadSleepTimeShift = 2; 129 130// minimum normal sink buffer size, expressed in milliseconds rather than frames 131// FIXME This should be based on experimentally observed scheduling jitter 132static const uint32_t kMinNormalSinkBufferSizeMs = 20; 133// maximum normal sink buffer size 134static const uint32_t kMaxNormalSinkBufferSizeMs = 24; 135 136// minimum capture buffer size in milliseconds to _not_ need a fast capture thread 137// FIXME This should be based on experimentally observed scheduling jitter 138static const uint32_t kMinNormalCaptureBufferSizeMs = 12; 139 140// Offloaded output thread standby delay: allows track transition without going to standby 141static const nsecs_t kOffloadStandbyDelayNs = seconds(1); 142 143// Direct output thread minimum sleep time in idle or active(underrun) state 144static const nsecs_t kDirectMinSleepTimeUs = 10000; 145 146 147// Whether to use fast mixer 148static const enum { 149 FastMixer_Never, // never initialize or use: for debugging only 150 FastMixer_Always, // always initialize and use, even if not needed: for debugging only 151 // normal mixer multiplier is 1 152 FastMixer_Static, // initialize if needed, then use all the time if initialized, 153 // multiplier is calculated based on min & max normal mixer buffer size 154 FastMixer_Dynamic, // initialize if needed, then use dynamically depending on track load, 155 // multiplier is calculated based on min & max normal mixer buffer size 156 // FIXME for FastMixer_Dynamic: 157 // Supporting this option will require fixing HALs that can't handle large writes. 158 // For example, one HAL implementation returns an error from a large write, 159 // and another HAL implementation corrupts memory, possibly in the sample rate converter. 160 // We could either fix the HAL implementations, or provide a wrapper that breaks 161 // up large writes into smaller ones, and the wrapper would need to deal with scheduler. 162} kUseFastMixer = FastMixer_Static; 163 164// Whether to use fast capture 165static const enum { 166 FastCapture_Never, // never initialize or use: for debugging only 167 FastCapture_Always, // always initialize and use, even if not needed: for debugging only 168 FastCapture_Static, // initialize if needed, then use all the time if initialized 169} kUseFastCapture = FastCapture_Static; 170 171// Priorities for requestPriority 172static const int kPriorityAudioApp = 2; 173static const int kPriorityFastMixer = 3; 174static const int kPriorityFastCapture = 3; 175 176// IAudioFlinger::createTrack() has an in/out parameter 'pFrameCount' for the total size of the 177// track buffer in shared memory. Zero on input means to use a default value. For fast tracks, 178// AudioFlinger derives the default from HAL buffer size and 'fast track multiplier'. 179 180// This is the default value, if not specified by property. 181static const int kFastTrackMultiplier = 2; 182 183// The minimum and maximum allowed values 184static const int kFastTrackMultiplierMin = 1; 185static const int kFastTrackMultiplierMax = 2; 186 187// The actual value to use, which can be specified per-device via property af.fast_track_multiplier. 188static int sFastTrackMultiplier = kFastTrackMultiplier; 189 190// See Thread::readOnlyHeap(). 191// Initially this heap is used to allocate client buffers for "fast" AudioRecord. 192// Eventually it will be the single buffer that FastCapture writes into via HAL read(), 193// and that all "fast" AudioRecord clients read from. In either case, the size can be small. 194static const size_t kRecordThreadReadOnlyHeapSize = 0x2000; 195 196// ---------------------------------------------------------------------------- 197 198static pthread_once_t sFastTrackMultiplierOnce = PTHREAD_ONCE_INIT; 199 200static void sFastTrackMultiplierInit() 201{ 202 char value[PROPERTY_VALUE_MAX]; 203 if (property_get("af.fast_track_multiplier", value, NULL) > 0) { 204 char *endptr; 205 unsigned long ul = strtoul(value, &endptr, 0); 206 if (*endptr == '\0' && kFastTrackMultiplierMin <= ul && ul <= kFastTrackMultiplierMax) { 207 sFastTrackMultiplier = (int) ul; 208 } 209 } 210} 211 212// ---------------------------------------------------------------------------- 213 214#ifdef ADD_BATTERY_DATA 215// To collect the amplifier usage 216static void addBatteryData(uint32_t params) { 217 sp<IMediaPlayerService> service = IMediaDeathNotifier::getMediaPlayerService(); 218 if (service == NULL) { 219 // it already logged 220 return; 221 } 222 223 service->addBatteryData(params); 224} 225#endif 226 227// Track the CLOCK_BOOTTIME versus CLOCK_MONOTONIC timebase offset 228struct { 229 // call when you acquire a partial wakelock 230 void acquire(const sp<IBinder> &wakeLockToken) { 231 pthread_mutex_lock(&mLock); 232 if (wakeLockToken.get() == nullptr) { 233 adjustTimebaseOffset(&mBoottimeOffset, ExtendedTimestamp::TIMEBASE_BOOTTIME); 234 } else { 235 if (mCount == 0) { 236 adjustTimebaseOffset(&mBoottimeOffset, ExtendedTimestamp::TIMEBASE_BOOTTIME); 237 } 238 ++mCount; 239 } 240 pthread_mutex_unlock(&mLock); 241 } 242 243 // call when you release a partial wakelock. 244 void release(const sp<IBinder> &wakeLockToken) { 245 if (wakeLockToken.get() == nullptr) { 246 return; 247 } 248 pthread_mutex_lock(&mLock); 249 if (--mCount < 0) { 250 ALOGE("negative wakelock count"); 251 mCount = 0; 252 } 253 pthread_mutex_unlock(&mLock); 254 } 255 256 // retrieves the boottime timebase offset from monotonic. 257 int64_t getBoottimeOffset() { 258 pthread_mutex_lock(&mLock); 259 int64_t boottimeOffset = mBoottimeOffset; 260 pthread_mutex_unlock(&mLock); 261 return boottimeOffset; 262 } 263 264 // Adjusts the timebase offset between TIMEBASE_MONOTONIC 265 // and the selected timebase. 266 // Currently only TIMEBASE_BOOTTIME is allowed. 267 // 268 // This only needs to be called upon acquiring the first partial wakelock 269 // after all other partial wakelocks are released. 270 // 271 // We do an empirical measurement of the offset rather than parsing 272 // /proc/timer_list since the latter is not a formal kernel ABI. 273 static void adjustTimebaseOffset(int64_t *offset, ExtendedTimestamp::Timebase timebase) { 274 int clockbase; 275 switch (timebase) { 276 case ExtendedTimestamp::TIMEBASE_BOOTTIME: 277 clockbase = SYSTEM_TIME_BOOTTIME; 278 break; 279 default: 280 LOG_ALWAYS_FATAL("invalid timebase %d", timebase); 281 break; 282 } 283 // try three times to get the clock offset, choose the one 284 // with the minimum gap in measurements. 285 const int tries = 3; 286 nsecs_t bestGap, measured; 287 for (int i = 0; i < tries; ++i) { 288 const nsecs_t tmono = systemTime(SYSTEM_TIME_MONOTONIC); 289 const nsecs_t tbase = systemTime(clockbase); 290 const nsecs_t tmono2 = systemTime(SYSTEM_TIME_MONOTONIC); 291 const nsecs_t gap = tmono2 - tmono; 292 if (i == 0 || gap < bestGap) { 293 bestGap = gap; 294 measured = tbase - ((tmono + tmono2) >> 1); 295 } 296 } 297 298 // to avoid micro-adjusting, we don't change the timebase 299 // unless it is significantly different. 300 // 301 // Assumption: It probably takes more than toleranceNs to 302 // suspend and resume the device. 303 static int64_t toleranceNs = 10000; // 10 us 304 if (llabs(*offset - measured) > toleranceNs) { 305 ALOGV("Adjusting timebase offset old: %lld new: %lld", 306 (long long)*offset, (long long)measured); 307 *offset = measured; 308 } 309 } 310 311 pthread_mutex_t mLock; 312 int32_t mCount; 313 int64_t mBoottimeOffset; 314} gBoottime = { PTHREAD_MUTEX_INITIALIZER, 0, 0 }; // static, so use POD initialization 315 316// ---------------------------------------------------------------------------- 317// CPU Stats 318// ---------------------------------------------------------------------------- 319 320class CpuStats { 321public: 322 CpuStats(); 323 void sample(const String8 &title); 324#ifdef DEBUG_CPU_USAGE 325private: 326 ThreadCpuUsage mCpuUsage; // instantaneous thread CPU usage in wall clock ns 327 CentralTendencyStatistics mWcStats; // statistics on thread CPU usage in wall clock ns 328 329 CentralTendencyStatistics mHzStats; // statistics on thread CPU usage in cycles 330 331 int mCpuNum; // thread's current CPU number 332 int mCpukHz; // frequency of thread's current CPU in kHz 333#endif 334}; 335 336CpuStats::CpuStats() 337#ifdef DEBUG_CPU_USAGE 338 : mCpuNum(-1), mCpukHz(-1) 339#endif 340{ 341} 342 343void CpuStats::sample(const String8 &title 344#ifndef DEBUG_CPU_USAGE 345 __unused 346#endif 347 ) { 348#ifdef DEBUG_CPU_USAGE 349 // get current thread's delta CPU time in wall clock ns 350 double wcNs; 351 bool valid = mCpuUsage.sampleAndEnable(wcNs); 352 353 // record sample for wall clock statistics 354 if (valid) { 355 mWcStats.sample(wcNs); 356 } 357 358 // get the current CPU number 359 int cpuNum = sched_getcpu(); 360 361 // get the current CPU frequency in kHz 362 int cpukHz = mCpuUsage.getCpukHz(cpuNum); 363 364 // check if either CPU number or frequency changed 365 if (cpuNum != mCpuNum || cpukHz != mCpukHz) { 366 mCpuNum = cpuNum; 367 mCpukHz = cpukHz; 368 // ignore sample for purposes of cycles 369 valid = false; 370 } 371 372 // if no change in CPU number or frequency, then record sample for cycle statistics 373 if (valid && mCpukHz > 0) { 374 double cycles = wcNs * cpukHz * 0.000001; 375 mHzStats.sample(cycles); 376 } 377 378 unsigned n = mWcStats.n(); 379 // mCpuUsage.elapsed() is expensive, so don't call it every loop 380 if ((n & 127) == 1) { 381 long long elapsed = mCpuUsage.elapsed(); 382 if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) { 383 double perLoop = elapsed / (double) n; 384 double perLoop100 = perLoop * 0.01; 385 double perLoop1k = perLoop * 0.001; 386 double mean = mWcStats.mean(); 387 double stddev = mWcStats.stddev(); 388 double minimum = mWcStats.minimum(); 389 double maximum = mWcStats.maximum(); 390 double meanCycles = mHzStats.mean(); 391 double stddevCycles = mHzStats.stddev(); 392 double minCycles = mHzStats.minimum(); 393 double maxCycles = mHzStats.maximum(); 394 mCpuUsage.resetElapsed(); 395 mWcStats.reset(); 396 mHzStats.reset(); 397 ALOGD("CPU usage for %s over past %.1f secs\n" 398 " (%u mixer loops at %.1f mean ms per loop):\n" 399 " us per mix loop: mean=%.0f stddev=%.0f min=%.0f max=%.0f\n" 400 " %% of wall: mean=%.1f stddev=%.1f min=%.1f max=%.1f\n" 401 " MHz: mean=%.1f, stddev=%.1f, min=%.1f max=%.1f", 402 title.string(), 403 elapsed * .000000001, n, perLoop * .000001, 404 mean * .001, 405 stddev * .001, 406 minimum * .001, 407 maximum * .001, 408 mean / perLoop100, 409 stddev / perLoop100, 410 minimum / perLoop100, 411 maximum / perLoop100, 412 meanCycles / perLoop1k, 413 stddevCycles / perLoop1k, 414 minCycles / perLoop1k, 415 maxCycles / perLoop1k); 416 417 } 418 } 419#endif 420}; 421 422// ---------------------------------------------------------------------------- 423// ThreadBase 424// ---------------------------------------------------------------------------- 425 426// static 427const char *AudioFlinger::ThreadBase::threadTypeToString(AudioFlinger::ThreadBase::type_t type) 428{ 429 switch (type) { 430 case MIXER: 431 return "MIXER"; 432 case DIRECT: 433 return "DIRECT"; 434 case DUPLICATING: 435 return "DUPLICATING"; 436 case RECORD: 437 return "RECORD"; 438 case OFFLOAD: 439 return "OFFLOAD"; 440 default: 441 return "unknown"; 442 } 443} 444 445String8 devicesToString(audio_devices_t devices) 446{ 447 static const struct mapping { 448 audio_devices_t mDevices; 449 const char * mString; 450 } mappingsOut[] = { 451 {AUDIO_DEVICE_OUT_EARPIECE, "EARPIECE"}, 452 {AUDIO_DEVICE_OUT_SPEAKER, "SPEAKER"}, 453 {AUDIO_DEVICE_OUT_WIRED_HEADSET, "WIRED_HEADSET"}, 454 {AUDIO_DEVICE_OUT_WIRED_HEADPHONE, "WIRED_HEADPHONE"}, 455 {AUDIO_DEVICE_OUT_BLUETOOTH_SCO, "BLUETOOTH_SCO"}, 456 {AUDIO_DEVICE_OUT_BLUETOOTH_SCO_HEADSET, "BLUETOOTH_SCO_HEADSET"}, 457 {AUDIO_DEVICE_OUT_BLUETOOTH_SCO_CARKIT, "BLUETOOTH_SCO_CARKIT"}, 458 {AUDIO_DEVICE_OUT_BLUETOOTH_A2DP, "BLUETOOTH_A2DP"}, 459 {AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES,"BLUETOOTH_A2DP_HEADPHONES"}, 460 {AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER, "BLUETOOTH_A2DP_SPEAKER"}, 461 {AUDIO_DEVICE_OUT_AUX_DIGITAL, "AUX_DIGITAL"}, 462 {AUDIO_DEVICE_OUT_HDMI, "HDMI"}, 463 {AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET,"ANLG_DOCK_HEADSET"}, 464 {AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET,"DGTL_DOCK_HEADSET"}, 465 {AUDIO_DEVICE_OUT_USB_ACCESSORY, "USB_ACCESSORY"}, 466 {AUDIO_DEVICE_OUT_USB_DEVICE, "USB_DEVICE"}, 467 {AUDIO_DEVICE_OUT_TELEPHONY_TX, "TELEPHONY_TX"}, 468 {AUDIO_DEVICE_OUT_LINE, "LINE"}, 469 {AUDIO_DEVICE_OUT_HDMI_ARC, "HDMI_ARC"}, 470 {AUDIO_DEVICE_OUT_SPDIF, "SPDIF"}, 471 {AUDIO_DEVICE_OUT_FM, "FM"}, 472 {AUDIO_DEVICE_OUT_AUX_LINE, "AUX_LINE"}, 473 {AUDIO_DEVICE_OUT_SPEAKER_SAFE, "SPEAKER_SAFE"}, 474 {AUDIO_DEVICE_OUT_IP, "IP"}, 475 {AUDIO_DEVICE_OUT_BUS, "BUS"}, 476 {AUDIO_DEVICE_NONE, "NONE"}, // must be last 477 }, mappingsIn[] = { 478 {AUDIO_DEVICE_IN_COMMUNICATION, "COMMUNICATION"}, 479 {AUDIO_DEVICE_IN_AMBIENT, "AMBIENT"}, 480 {AUDIO_DEVICE_IN_BUILTIN_MIC, "BUILTIN_MIC"}, 481 {AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET, "BLUETOOTH_SCO_HEADSET"}, 482 {AUDIO_DEVICE_IN_WIRED_HEADSET, "WIRED_HEADSET"}, 483 {AUDIO_DEVICE_IN_AUX_DIGITAL, "AUX_DIGITAL"}, 484 {AUDIO_DEVICE_IN_VOICE_CALL, "VOICE_CALL"}, 485 {AUDIO_DEVICE_IN_TELEPHONY_RX, "TELEPHONY_RX"}, 486 {AUDIO_DEVICE_IN_BACK_MIC, "BACK_MIC"}, 487 {AUDIO_DEVICE_IN_REMOTE_SUBMIX, "REMOTE_SUBMIX"}, 488 {AUDIO_DEVICE_IN_ANLG_DOCK_HEADSET, "ANLG_DOCK_HEADSET"}, 489 {AUDIO_DEVICE_IN_DGTL_DOCK_HEADSET, "DGTL_DOCK_HEADSET"}, 490 {AUDIO_DEVICE_IN_USB_ACCESSORY, "USB_ACCESSORY"}, 491 {AUDIO_DEVICE_IN_USB_DEVICE, "USB_DEVICE"}, 492 {AUDIO_DEVICE_IN_FM_TUNER, "FM_TUNER"}, 493 {AUDIO_DEVICE_IN_TV_TUNER, "TV_TUNER"}, 494 {AUDIO_DEVICE_IN_LINE, "LINE"}, 495 {AUDIO_DEVICE_IN_SPDIF, "SPDIF"}, 496 {AUDIO_DEVICE_IN_BLUETOOTH_A2DP, "BLUETOOTH_A2DP"}, 497 {AUDIO_DEVICE_IN_LOOPBACK, "LOOPBACK"}, 498 {AUDIO_DEVICE_IN_IP, "IP"}, 499 {AUDIO_DEVICE_IN_BUS, "BUS"}, 500 {AUDIO_DEVICE_NONE, "NONE"}, // must be last 501 }; 502 String8 result; 503 audio_devices_t allDevices = AUDIO_DEVICE_NONE; 504 const mapping *entry; 505 if (devices & AUDIO_DEVICE_BIT_IN) { 506 devices &= ~AUDIO_DEVICE_BIT_IN; 507 entry = mappingsIn; 508 } else { 509 entry = mappingsOut; 510 } 511 for ( ; entry->mDevices != AUDIO_DEVICE_NONE; entry++) { 512 allDevices = (audio_devices_t) (allDevices | entry->mDevices); 513 if (devices & entry->mDevices) { 514 if (!result.isEmpty()) { 515 result.append("|"); 516 } 517 result.append(entry->mString); 518 } 519 } 520 if (devices & ~allDevices) { 521 if (!result.isEmpty()) { 522 result.append("|"); 523 } 524 result.appendFormat("0x%X", devices & ~allDevices); 525 } 526 if (result.isEmpty()) { 527 result.append(entry->mString); 528 } 529 return result; 530} 531 532String8 inputFlagsToString(audio_input_flags_t flags) 533{ 534 static const struct mapping { 535 audio_input_flags_t mFlag; 536 const char * mString; 537 } mappings[] = { 538 {AUDIO_INPUT_FLAG_FAST, "FAST"}, 539 {AUDIO_INPUT_FLAG_HW_HOTWORD, "HW_HOTWORD"}, 540 {AUDIO_INPUT_FLAG_RAW, "RAW"}, 541 {AUDIO_INPUT_FLAG_SYNC, "SYNC"}, 542 {AUDIO_INPUT_FLAG_NONE, "NONE"}, // must be last 543 }; 544 String8 result; 545 audio_input_flags_t allFlags = AUDIO_INPUT_FLAG_NONE; 546 const mapping *entry; 547 for (entry = mappings; entry->mFlag != AUDIO_INPUT_FLAG_NONE; entry++) { 548 allFlags = (audio_input_flags_t) (allFlags | entry->mFlag); 549 if (flags & entry->mFlag) { 550 if (!result.isEmpty()) { 551 result.append("|"); 552 } 553 result.append(entry->mString); 554 } 555 } 556 if (flags & ~allFlags) { 557 if (!result.isEmpty()) { 558 result.append("|"); 559 } 560 result.appendFormat("0x%X", flags & ~allFlags); 561 } 562 if (result.isEmpty()) { 563 result.append(entry->mString); 564 } 565 return result; 566} 567 568String8 outputFlagsToString(audio_output_flags_t flags) 569{ 570 static const struct mapping { 571 audio_output_flags_t mFlag; 572 const char * mString; 573 } mappings[] = { 574 {AUDIO_OUTPUT_FLAG_DIRECT, "DIRECT"}, 575 {AUDIO_OUTPUT_FLAG_PRIMARY, "PRIMARY"}, 576 {AUDIO_OUTPUT_FLAG_FAST, "FAST"}, 577 {AUDIO_OUTPUT_FLAG_DEEP_BUFFER, "DEEP_BUFFER"}, 578 {AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD,"COMPRESS_OFFLOAD"}, 579 {AUDIO_OUTPUT_FLAG_NON_BLOCKING, "NON_BLOCKING"}, 580 {AUDIO_OUTPUT_FLAG_HW_AV_SYNC, "HW_AV_SYNC"}, 581 {AUDIO_OUTPUT_FLAG_RAW, "RAW"}, 582 {AUDIO_OUTPUT_FLAG_SYNC, "SYNC"}, 583 {AUDIO_OUTPUT_FLAG_IEC958_NONAUDIO, "IEC958_NONAUDIO"}, 584 {AUDIO_OUTPUT_FLAG_NONE, "NONE"}, // must be last 585 }; 586 String8 result; 587 audio_output_flags_t allFlags = AUDIO_OUTPUT_FLAG_NONE; 588 const mapping *entry; 589 for (entry = mappings; entry->mFlag != AUDIO_OUTPUT_FLAG_NONE; entry++) { 590 allFlags = (audio_output_flags_t) (allFlags | entry->mFlag); 591 if (flags & entry->mFlag) { 592 if (!result.isEmpty()) { 593 result.append("|"); 594 } 595 result.append(entry->mString); 596 } 597 } 598 if (flags & ~allFlags) { 599 if (!result.isEmpty()) { 600 result.append("|"); 601 } 602 result.appendFormat("0x%X", flags & ~allFlags); 603 } 604 if (result.isEmpty()) { 605 result.append(entry->mString); 606 } 607 return result; 608} 609 610const char *sourceToString(audio_source_t source) 611{ 612 switch (source) { 613 case AUDIO_SOURCE_DEFAULT: return "default"; 614 case AUDIO_SOURCE_MIC: return "mic"; 615 case AUDIO_SOURCE_VOICE_UPLINK: return "voice uplink"; 616 case AUDIO_SOURCE_VOICE_DOWNLINK: return "voice downlink"; 617 case AUDIO_SOURCE_VOICE_CALL: return "voice call"; 618 case AUDIO_SOURCE_CAMCORDER: return "camcorder"; 619 case AUDIO_SOURCE_VOICE_RECOGNITION: return "voice recognition"; 620 case AUDIO_SOURCE_VOICE_COMMUNICATION: return "voice communication"; 621 case AUDIO_SOURCE_REMOTE_SUBMIX: return "remote submix"; 622 case AUDIO_SOURCE_UNPROCESSED: return "unprocessed"; 623 case AUDIO_SOURCE_FM_TUNER: return "FM tuner"; 624 case AUDIO_SOURCE_HOTWORD: return "hotword"; 625 default: return "unknown"; 626 } 627} 628 629AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, 630 audio_devices_t outDevice, audio_devices_t inDevice, type_t type, bool systemReady) 631 : Thread(false /*canCallJava*/), 632 mType(type), 633 mAudioFlinger(audioFlinger), 634 // mSampleRate, mFrameCount, mChannelMask, mChannelCount, mFrameSize, mFormat, mBufferSize 635 // are set by PlaybackThread::readOutputParameters_l() or 636 // RecordThread::readInputParameters_l() 637 //FIXME: mStandby should be true here. Is this some kind of hack? 638 mStandby(false), mOutDevice(outDevice), mInDevice(inDevice), 639 mPrevOutDevice(AUDIO_DEVICE_NONE), mPrevInDevice(AUDIO_DEVICE_NONE), 640 mAudioSource(AUDIO_SOURCE_DEFAULT), mId(id), 641 // mName will be set by concrete (non-virtual) subclass 642 mDeathRecipient(new PMDeathRecipient(this)), 643 mSystemReady(systemReady), 644 mNotifiedBatteryStart(false) 645{ 646 memset(&mPatch, 0, sizeof(struct audio_patch)); 647} 648 649AudioFlinger::ThreadBase::~ThreadBase() 650{ 651 // mConfigEvents should be empty, but just in case it isn't, free the memory it owns 652 mConfigEvents.clear(); 653 654 // do not lock the mutex in destructor 655 releaseWakeLock_l(); 656 if (mPowerManager != 0) { 657 sp<IBinder> binder = IInterface::asBinder(mPowerManager); 658 binder->unlinkToDeath(mDeathRecipient); 659 } 660} 661 662status_t AudioFlinger::ThreadBase::readyToRun() 663{ 664 status_t status = initCheck(); 665 if (status == NO_ERROR) { 666 ALOGI("AudioFlinger's thread %p ready to run", this); 667 } else { 668 ALOGE("No working audio driver found."); 669 } 670 return status; 671} 672 673void AudioFlinger::ThreadBase::exit() 674{ 675 ALOGV("ThreadBase::exit"); 676 // do any cleanup required for exit to succeed 677 preExit(); 678 { 679 // This lock prevents the following race in thread (uniprocessor for illustration): 680 // if (!exitPending()) { 681 // // context switch from here to exit() 682 // // exit() calls requestExit(), what exitPending() observes 683 // // exit() calls signal(), which is dropped since no waiters 684 // // context switch back from exit() to here 685 // mWaitWorkCV.wait(...); 686 // // now thread is hung 687 // } 688 AutoMutex lock(mLock); 689 requestExit(); 690 mWaitWorkCV.broadcast(); 691 } 692 // When Thread::requestExitAndWait is made virtual and this method is renamed to 693 // "virtual status_t requestExitAndWait()", replace by "return Thread::requestExitAndWait();" 694 requestExitAndWait(); 695} 696 697status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs) 698{ 699 ALOGV("ThreadBase::setParameters() %s", keyValuePairs.string()); 700 Mutex::Autolock _l(mLock); 701 702 return sendSetParameterConfigEvent_l(keyValuePairs); 703} 704 705// sendConfigEvent_l() must be called with ThreadBase::mLock held 706// Can temporarily release the lock if waiting for a reply from processConfigEvents_l(). 707status_t AudioFlinger::ThreadBase::sendConfigEvent_l(sp<ConfigEvent>& event) 708{ 709 status_t status = NO_ERROR; 710 711 if (event->mRequiresSystemReady && !mSystemReady) { 712 event->mWaitStatus = false; 713 mPendingConfigEvents.add(event); 714 return status; 715 } 716 mConfigEvents.add(event); 717 ALOGV("sendConfigEvent_l() num events %zu event %d", mConfigEvents.size(), event->mType); 718 mWaitWorkCV.signal(); 719 mLock.unlock(); 720 { 721 Mutex::Autolock _l(event->mLock); 722 while (event->mWaitStatus) { 723 if (event->mCond.waitRelative(event->mLock, kConfigEventTimeoutNs) != NO_ERROR) { 724 event->mStatus = TIMED_OUT; 725 event->mWaitStatus = false; 726 } 727 } 728 status = event->mStatus; 729 } 730 mLock.lock(); 731 return status; 732} 733 734void AudioFlinger::ThreadBase::sendIoConfigEvent(audio_io_config_event event, pid_t pid) 735{ 736 Mutex::Autolock _l(mLock); 737 sendIoConfigEvent_l(event, pid); 738} 739 740// sendIoConfigEvent_l() must be called with ThreadBase::mLock held 741void AudioFlinger::ThreadBase::sendIoConfigEvent_l(audio_io_config_event event, pid_t pid) 742{ 743 sp<ConfigEvent> configEvent = (ConfigEvent *)new IoConfigEvent(event, pid); 744 sendConfigEvent_l(configEvent); 745} 746 747void AudioFlinger::ThreadBase::sendPrioConfigEvent(pid_t pid, pid_t tid, int32_t prio) 748{ 749 Mutex::Autolock _l(mLock); 750 sendPrioConfigEvent_l(pid, tid, prio); 751} 752 753// sendPrioConfigEvent_l() must be called with ThreadBase::mLock held 754void AudioFlinger::ThreadBase::sendPrioConfigEvent_l(pid_t pid, pid_t tid, int32_t prio) 755{ 756 sp<ConfigEvent> configEvent = (ConfigEvent *)new PrioConfigEvent(pid, tid, prio); 757 sendConfigEvent_l(configEvent); 758} 759 760// sendSetParameterConfigEvent_l() must be called with ThreadBase::mLock held 761status_t AudioFlinger::ThreadBase::sendSetParameterConfigEvent_l(const String8& keyValuePair) 762{ 763 sp<ConfigEvent> configEvent; 764 AudioParameter param(keyValuePair); 765 int value; 766 if (param.getInt(String8(AUDIO_PARAMETER_MONO_OUTPUT), value) == NO_ERROR) { 767 setMasterMono_l(value != 0); 768 if (param.size() == 1) { 769 return NO_ERROR; // should be a solo parameter - we don't pass down 770 } 771 param.remove(String8(AUDIO_PARAMETER_MONO_OUTPUT)); 772 configEvent = new SetParameterConfigEvent(param.toString()); 773 } else { 774 configEvent = new SetParameterConfigEvent(keyValuePair); 775 } 776 return sendConfigEvent_l(configEvent); 777} 778 779status_t AudioFlinger::ThreadBase::sendCreateAudioPatchConfigEvent( 780 const struct audio_patch *patch, 781 audio_patch_handle_t *handle) 782{ 783 Mutex::Autolock _l(mLock); 784 sp<ConfigEvent> configEvent = (ConfigEvent *)new CreateAudioPatchConfigEvent(*patch, *handle); 785 status_t status = sendConfigEvent_l(configEvent); 786 if (status == NO_ERROR) { 787 CreateAudioPatchConfigEventData *data = 788 (CreateAudioPatchConfigEventData *)configEvent->mData.get(); 789 *handle = data->mHandle; 790 } 791 return status; 792} 793 794status_t AudioFlinger::ThreadBase::sendReleaseAudioPatchConfigEvent( 795 const audio_patch_handle_t handle) 796{ 797 Mutex::Autolock _l(mLock); 798 sp<ConfigEvent> configEvent = (ConfigEvent *)new ReleaseAudioPatchConfigEvent(handle); 799 return sendConfigEvent_l(configEvent); 800} 801 802 803// post condition: mConfigEvents.isEmpty() 804void AudioFlinger::ThreadBase::processConfigEvents_l() 805{ 806 bool configChanged = false; 807 808 while (!mConfigEvents.isEmpty()) { 809 ALOGV("processConfigEvents_l() remaining events %zu", mConfigEvents.size()); 810 sp<ConfigEvent> event = mConfigEvents[0]; 811 mConfigEvents.removeAt(0); 812 switch (event->mType) { 813 case CFG_EVENT_PRIO: { 814 PrioConfigEventData *data = (PrioConfigEventData *)event->mData.get(); 815 // FIXME Need to understand why this has to be done asynchronously 816 int err = requestPriority(data->mPid, data->mTid, data->mPrio, 817 true /*asynchronous*/); 818 if (err != 0) { 819 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 820 data->mPrio, data->mPid, data->mTid, err); 821 } 822 } break; 823 case CFG_EVENT_IO: { 824 IoConfigEventData *data = (IoConfigEventData *)event->mData.get(); 825 ioConfigChanged(data->mEvent, data->mPid); 826 } break; 827 case CFG_EVENT_SET_PARAMETER: { 828 SetParameterConfigEventData *data = (SetParameterConfigEventData *)event->mData.get(); 829 if (checkForNewParameter_l(data->mKeyValuePairs, event->mStatus)) { 830 configChanged = true; 831 } 832 } break; 833 case CFG_EVENT_CREATE_AUDIO_PATCH: { 834 CreateAudioPatchConfigEventData *data = 835 (CreateAudioPatchConfigEventData *)event->mData.get(); 836 event->mStatus = createAudioPatch_l(&data->mPatch, &data->mHandle); 837 } break; 838 case CFG_EVENT_RELEASE_AUDIO_PATCH: { 839 ReleaseAudioPatchConfigEventData *data = 840 (ReleaseAudioPatchConfigEventData *)event->mData.get(); 841 event->mStatus = releaseAudioPatch_l(data->mHandle); 842 } break; 843 default: 844 ALOG_ASSERT(false, "processConfigEvents_l() unknown event type %d", event->mType); 845 break; 846 } 847 { 848 Mutex::Autolock _l(event->mLock); 849 if (event->mWaitStatus) { 850 event->mWaitStatus = false; 851 event->mCond.signal(); 852 } 853 } 854 ALOGV_IF(mConfigEvents.isEmpty(), "processConfigEvents_l() DONE thread %p", this); 855 } 856 857 if (configChanged) { 858 cacheParameters_l(); 859 } 860} 861 862String8 channelMaskToString(audio_channel_mask_t mask, bool output) { 863 String8 s; 864 const audio_channel_representation_t representation = 865 audio_channel_mask_get_representation(mask); 866 867 switch (representation) { 868 case AUDIO_CHANNEL_REPRESENTATION_POSITION: { 869 if (output) { 870 if (mask & AUDIO_CHANNEL_OUT_FRONT_LEFT) s.append("front-left, "); 871 if (mask & AUDIO_CHANNEL_OUT_FRONT_RIGHT) s.append("front-right, "); 872 if (mask & AUDIO_CHANNEL_OUT_FRONT_CENTER) s.append("front-center, "); 873 if (mask & AUDIO_CHANNEL_OUT_LOW_FREQUENCY) s.append("low freq, "); 874 if (mask & AUDIO_CHANNEL_OUT_BACK_LEFT) s.append("back-left, "); 875 if (mask & AUDIO_CHANNEL_OUT_BACK_RIGHT) s.append("back-right, "); 876 if (mask & AUDIO_CHANNEL_OUT_FRONT_LEFT_OF_CENTER) s.append("front-left-of-center, "); 877 if (mask & AUDIO_CHANNEL_OUT_FRONT_RIGHT_OF_CENTER) s.append("front-right-of-center, "); 878 if (mask & AUDIO_CHANNEL_OUT_BACK_CENTER) s.append("back-center, "); 879 if (mask & AUDIO_CHANNEL_OUT_SIDE_LEFT) s.append("side-left, "); 880 if (mask & AUDIO_CHANNEL_OUT_SIDE_RIGHT) s.append("side-right, "); 881 if (mask & AUDIO_CHANNEL_OUT_TOP_CENTER) s.append("top-center ,"); 882 if (mask & AUDIO_CHANNEL_OUT_TOP_FRONT_LEFT) s.append("top-front-left, "); 883 if (mask & AUDIO_CHANNEL_OUT_TOP_FRONT_CENTER) s.append("top-front-center, "); 884 if (mask & AUDIO_CHANNEL_OUT_TOP_FRONT_RIGHT) s.append("top-front-right, "); 885 if (mask & AUDIO_CHANNEL_OUT_TOP_BACK_LEFT) s.append("top-back-left, "); 886 if (mask & AUDIO_CHANNEL_OUT_TOP_BACK_CENTER) s.append("top-back-center, " ); 887 if (mask & AUDIO_CHANNEL_OUT_TOP_BACK_RIGHT) s.append("top-back-right, " ); 888 if (mask & ~AUDIO_CHANNEL_OUT_ALL) s.append("unknown, "); 889 } else { 890 if (mask & AUDIO_CHANNEL_IN_LEFT) s.append("left, "); 891 if (mask & AUDIO_CHANNEL_IN_RIGHT) s.append("right, "); 892 if (mask & AUDIO_CHANNEL_IN_FRONT) s.append("front, "); 893 if (mask & AUDIO_CHANNEL_IN_BACK) s.append("back, "); 894 if (mask & AUDIO_CHANNEL_IN_LEFT_PROCESSED) s.append("left-processed, "); 895 if (mask & AUDIO_CHANNEL_IN_RIGHT_PROCESSED) s.append("right-processed, "); 896 if (mask & AUDIO_CHANNEL_IN_FRONT_PROCESSED) s.append("front-processed, "); 897 if (mask & AUDIO_CHANNEL_IN_BACK_PROCESSED) s.append("back-processed, "); 898 if (mask & AUDIO_CHANNEL_IN_PRESSURE) s.append("pressure, "); 899 if (mask & AUDIO_CHANNEL_IN_X_AXIS) s.append("X, "); 900 if (mask & AUDIO_CHANNEL_IN_Y_AXIS) s.append("Y, "); 901 if (mask & AUDIO_CHANNEL_IN_Z_AXIS) s.append("Z, "); 902 if (mask & AUDIO_CHANNEL_IN_VOICE_UPLINK) s.append("voice-uplink, "); 903 if (mask & AUDIO_CHANNEL_IN_VOICE_DNLINK) s.append("voice-dnlink, "); 904 if (mask & ~AUDIO_CHANNEL_IN_ALL) s.append("unknown, "); 905 } 906 const int len = s.length(); 907 if (len > 2) { 908 (void) s.lockBuffer(len); // needed? 909 s.unlockBuffer(len - 2); // remove trailing ", " 910 } 911 return s; 912 } 913 case AUDIO_CHANNEL_REPRESENTATION_INDEX: 914 s.appendFormat("index mask, bits:%#x", audio_channel_mask_get_bits(mask)); 915 return s; 916 default: 917 s.appendFormat("unknown mask, representation:%d bits:%#x", 918 representation, audio_channel_mask_get_bits(mask)); 919 return s; 920 } 921} 922 923void AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args __unused) 924{ 925 const size_t SIZE = 256; 926 char buffer[SIZE]; 927 String8 result; 928 929 bool locked = AudioFlinger::dumpTryLock(mLock); 930 if (!locked) { 931 dprintf(fd, "thread %p may be deadlocked\n", this); 932 } 933 934 dprintf(fd, " Thread name: %s\n", mThreadName); 935 dprintf(fd, " I/O handle: %d\n", mId); 936 dprintf(fd, " TID: %d\n", getTid()); 937 dprintf(fd, " Standby: %s\n", mStandby ? "yes" : "no"); 938 dprintf(fd, " Sample rate: %u Hz\n", mSampleRate); 939 dprintf(fd, " HAL frame count: %zu\n", mFrameCount); 940 dprintf(fd, " HAL format: 0x%x (%s)\n", mHALFormat, formatToString(mHALFormat)); 941 dprintf(fd, " HAL buffer size: %zu bytes\n", mBufferSize); 942 dprintf(fd, " Channel count: %u\n", mChannelCount); 943 dprintf(fd, " Channel mask: 0x%08x (%s)\n", mChannelMask, 944 channelMaskToString(mChannelMask, mType != RECORD).string()); 945 dprintf(fd, " Processing format: 0x%x (%s)\n", mFormat, formatToString(mFormat)); 946 dprintf(fd, " Processing frame size: %zu bytes\n", mFrameSize); 947 dprintf(fd, " Pending config events:"); 948 size_t numConfig = mConfigEvents.size(); 949 if (numConfig) { 950 for (size_t i = 0; i < numConfig; i++) { 951 mConfigEvents[i]->dump(buffer, SIZE); 952 dprintf(fd, "\n %s", buffer); 953 } 954 dprintf(fd, "\n"); 955 } else { 956 dprintf(fd, " none\n"); 957 } 958 dprintf(fd, " Output device: %#x (%s)\n", mOutDevice, devicesToString(mOutDevice).string()); 959 dprintf(fd, " Input device: %#x (%s)\n", mInDevice, devicesToString(mInDevice).string()); 960 dprintf(fd, " Audio source: %d (%s)\n", mAudioSource, sourceToString(mAudioSource)); 961 962 if (locked) { 963 mLock.unlock(); 964 } 965} 966 967void AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args) 968{ 969 const size_t SIZE = 256; 970 char buffer[SIZE]; 971 String8 result; 972 973 size_t numEffectChains = mEffectChains.size(); 974 snprintf(buffer, SIZE, " %zu Effect Chains\n", numEffectChains); 975 write(fd, buffer, strlen(buffer)); 976 977 for (size_t i = 0; i < numEffectChains; ++i) { 978 sp<EffectChain> chain = mEffectChains[i]; 979 if (chain != 0) { 980 chain->dump(fd, args); 981 } 982 } 983} 984 985void AudioFlinger::ThreadBase::acquireWakeLock(int uid) 986{ 987 Mutex::Autolock _l(mLock); 988 acquireWakeLock_l(uid); 989} 990 991String16 AudioFlinger::ThreadBase::getWakeLockTag() 992{ 993 switch (mType) { 994 case MIXER: 995 return String16("AudioMix"); 996 case DIRECT: 997 return String16("AudioDirectOut"); 998 case DUPLICATING: 999 return String16("AudioDup"); 1000 case RECORD: 1001 return String16("AudioIn"); 1002 case OFFLOAD: 1003 return String16("AudioOffload"); 1004 default: 1005 ALOG_ASSERT(false); 1006 return String16("AudioUnknown"); 1007 } 1008} 1009 1010void AudioFlinger::ThreadBase::acquireWakeLock_l(int uid) 1011{ 1012 getPowerManager_l(); 1013 if (mPowerManager != 0) { 1014 sp<IBinder> binder = new BBinder(); 1015 status_t status; 1016 if (uid >= 0) { 1017 status = mPowerManager->acquireWakeLockWithUid(POWERMANAGER_PARTIAL_WAKE_LOCK, 1018 binder, 1019 getWakeLockTag(), 1020 String16("audioserver"), 1021 uid, 1022 true /* FIXME force oneway contrary to .aidl */); 1023 } else { 1024 status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK, 1025 binder, 1026 getWakeLockTag(), 1027 String16("audioserver"), 1028 true /* FIXME force oneway contrary to .aidl */); 1029 } 1030 if (status == NO_ERROR) { 1031 mWakeLockToken = binder; 1032 } 1033 ALOGV("acquireWakeLock_l() %s status %d", mThreadName, status); 1034 } 1035 1036 if (!mNotifiedBatteryStart) { 1037 BatteryNotifier::getInstance().noteStartAudio(); 1038 mNotifiedBatteryStart = true; 1039 } 1040 gBoottime.acquire(mWakeLockToken); 1041 mTimestamp.mTimebaseOffset[ExtendedTimestamp::TIMEBASE_BOOTTIME] = 1042 gBoottime.getBoottimeOffset(); 1043} 1044 1045void AudioFlinger::ThreadBase::releaseWakeLock() 1046{ 1047 Mutex::Autolock _l(mLock); 1048 releaseWakeLock_l(); 1049} 1050 1051void AudioFlinger::ThreadBase::releaseWakeLock_l() 1052{ 1053 gBoottime.release(mWakeLockToken); 1054 if (mWakeLockToken != 0) { 1055 ALOGV("releaseWakeLock_l() %s", mThreadName); 1056 if (mPowerManager != 0) { 1057 mPowerManager->releaseWakeLock(mWakeLockToken, 0, 1058 true /* FIXME force oneway contrary to .aidl */); 1059 } 1060 mWakeLockToken.clear(); 1061 } 1062 1063 if (mNotifiedBatteryStart) { 1064 BatteryNotifier::getInstance().noteStopAudio(); 1065 mNotifiedBatteryStart = false; 1066 } 1067} 1068 1069void AudioFlinger::ThreadBase::updateWakeLockUids(const SortedVector<int> &uids) { 1070 Mutex::Autolock _l(mLock); 1071 updateWakeLockUids_l(uids); 1072} 1073 1074void AudioFlinger::ThreadBase::getPowerManager_l() { 1075 if (mSystemReady && mPowerManager == 0) { 1076 // use checkService() to avoid blocking if power service is not up yet 1077 sp<IBinder> binder = 1078 defaultServiceManager()->checkService(String16("power")); 1079 if (binder == 0) { 1080 ALOGW("Thread %s cannot connect to the power manager service", mThreadName); 1081 } else { 1082 mPowerManager = interface_cast<IPowerManager>(binder); 1083 binder->linkToDeath(mDeathRecipient); 1084 } 1085 } 1086} 1087 1088void AudioFlinger::ThreadBase::updateWakeLockUids_l(const SortedVector<int> &uids) { 1089 getPowerManager_l(); 1090 if (mWakeLockToken == NULL) { // token may be NULL if AudioFlinger::systemReady() not called. 1091 if (mSystemReady) { 1092 ALOGE("no wake lock to update, but system ready!"); 1093 } else { 1094 ALOGW("no wake lock to update, system not ready yet"); 1095 } 1096 return; 1097 } 1098 if (mPowerManager != 0) { 1099 sp<IBinder> binder = new BBinder(); 1100 status_t status; 1101 status = mPowerManager->updateWakeLockUids(mWakeLockToken, uids.size(), uids.array(), 1102 true /* FIXME force oneway contrary to .aidl */); 1103 ALOGV("updateWakeLockUids_l() %s status %d", mThreadName, status); 1104 } 1105} 1106 1107void AudioFlinger::ThreadBase::clearPowerManager() 1108{ 1109 Mutex::Autolock _l(mLock); 1110 releaseWakeLock_l(); 1111 mPowerManager.clear(); 1112} 1113 1114void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who __unused) 1115{ 1116 sp<ThreadBase> thread = mThread.promote(); 1117 if (thread != 0) { 1118 thread->clearPowerManager(); 1119 } 1120 ALOGW("power manager service died !!!"); 1121} 1122 1123void AudioFlinger::ThreadBase::setEffectSuspended( 1124 const effect_uuid_t *type, bool suspend, audio_session_t sessionId) 1125{ 1126 Mutex::Autolock _l(mLock); 1127 setEffectSuspended_l(type, suspend, sessionId); 1128} 1129 1130void AudioFlinger::ThreadBase::setEffectSuspended_l( 1131 const effect_uuid_t *type, bool suspend, audio_session_t sessionId) 1132{ 1133 sp<EffectChain> chain = getEffectChain_l(sessionId); 1134 if (chain != 0) { 1135 if (type != NULL) { 1136 chain->setEffectSuspended_l(type, suspend); 1137 } else { 1138 chain->setEffectSuspendedAll_l(suspend); 1139 } 1140 } 1141 1142 updateSuspendedSessions_l(type, suspend, sessionId); 1143} 1144 1145void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain) 1146{ 1147 ssize_t index = mSuspendedSessions.indexOfKey(chain->sessionId()); 1148 if (index < 0) { 1149 return; 1150 } 1151 1152 const KeyedVector <int, sp<SuspendedSessionDesc> >& sessionEffects = 1153 mSuspendedSessions.valueAt(index); 1154 1155 for (size_t i = 0; i < sessionEffects.size(); i++) { 1156 sp<SuspendedSessionDesc> desc = sessionEffects.valueAt(i); 1157 for (int j = 0; j < desc->mRefCount; j++) { 1158 if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) { 1159 chain->setEffectSuspendedAll_l(true); 1160 } else { 1161 ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x", 1162 desc->mType.timeLow); 1163 chain->setEffectSuspended_l(&desc->mType, true); 1164 } 1165 } 1166 } 1167} 1168 1169void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type, 1170 bool suspend, 1171 audio_session_t sessionId) 1172{ 1173 ssize_t index = mSuspendedSessions.indexOfKey(sessionId); 1174 1175 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects; 1176 1177 if (suspend) { 1178 if (index >= 0) { 1179 sessionEffects = mSuspendedSessions.valueAt(index); 1180 } else { 1181 mSuspendedSessions.add(sessionId, sessionEffects); 1182 } 1183 } else { 1184 if (index < 0) { 1185 return; 1186 } 1187 sessionEffects = mSuspendedSessions.valueAt(index); 1188 } 1189 1190 1191 int key = EffectChain::kKeyForSuspendAll; 1192 if (type != NULL) { 1193 key = type->timeLow; 1194 } 1195 index = sessionEffects.indexOfKey(key); 1196 1197 sp<SuspendedSessionDesc> desc; 1198 if (suspend) { 1199 if (index >= 0) { 1200 desc = sessionEffects.valueAt(index); 1201 } else { 1202 desc = new SuspendedSessionDesc(); 1203 if (type != NULL) { 1204 desc->mType = *type; 1205 } 1206 sessionEffects.add(key, desc); 1207 ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key); 1208 } 1209 desc->mRefCount++; 1210 } else { 1211 if (index < 0) { 1212 return; 1213 } 1214 desc = sessionEffects.valueAt(index); 1215 if (--desc->mRefCount == 0) { 1216 ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key); 1217 sessionEffects.removeItemsAt(index); 1218 if (sessionEffects.isEmpty()) { 1219 ALOGV("updateSuspendedSessions_l() restore removing session %d", 1220 sessionId); 1221 mSuspendedSessions.removeItem(sessionId); 1222 } 1223 } 1224 } 1225 if (!sessionEffects.isEmpty()) { 1226 mSuspendedSessions.replaceValueFor(sessionId, sessionEffects); 1227 } 1228} 1229 1230void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 1231 bool enabled, 1232 audio_session_t sessionId) 1233{ 1234 Mutex::Autolock _l(mLock); 1235 checkSuspendOnEffectEnabled_l(effect, enabled, sessionId); 1236} 1237 1238void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 1239 bool enabled, 1240 audio_session_t sessionId) 1241{ 1242 if (mType != RECORD) { 1243 // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on 1244 // another session. This gives the priority to well behaved effect control panels 1245 // and applications not using global effects. 1246 // Enabling post processing in AUDIO_SESSION_OUTPUT_STAGE session does not affect 1247 // global effects 1248 if ((sessionId != AUDIO_SESSION_OUTPUT_MIX) && (sessionId != AUDIO_SESSION_OUTPUT_STAGE)) { 1249 setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX); 1250 } 1251 } 1252 1253 sp<EffectChain> chain = getEffectChain_l(sessionId); 1254 if (chain != 0) { 1255 chain->checkSuspendOnEffectEnabled(effect, enabled); 1256 } 1257} 1258 1259// 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 1276 // always allow effects without processing load or latency 1277 if ((desc->flags & EFFECT_FLAG_NO_PROCESS_MASK) == EFFECT_FLAG_NO_PROCESS) { 1278 return NO_ERROR; 1279 } 1280 1281 audio_input_flags_t flags = mInput->flags; 1282 if (hasFastCapture() || (flags & AUDIO_INPUT_FLAG_FAST)) { 1283 if (flags & AUDIO_INPUT_FLAG_RAW) { 1284 ALOGW("checkEffectCompatibility_l(): effect %s on record thread %s in raw mode", 1285 desc->name, mThreadName); 1286 return BAD_VALUE; 1287 } 1288 if ((desc->flags & EFFECT_FLAG_HW_ACC_TUNNEL) == 0) { 1289 ALOGW("checkEffectCompatibility_l(): non HW effect %s on record thread %s in fast mode", 1290 desc->name, mThreadName); 1291 return BAD_VALUE; 1292 } 1293 } 1294 return NO_ERROR; 1295} 1296 1297// checkEffectCompatibility_l() must be called with ThreadBase::mLock held 1298status_t AudioFlinger::PlaybackThread::checkEffectCompatibility_l( 1299 const effect_descriptor_t *desc, audio_session_t sessionId) 1300{ 1301 // no preprocessing on playback threads 1302 if ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC) { 1303 ALOGW("checkEffectCompatibility_l(): pre processing effect %s created on playback" 1304 " thread %s", desc->name, mThreadName); 1305 return BAD_VALUE; 1306 } 1307 1308 switch (mType) { 1309 case MIXER: { 1310 // Reject any effect on mixer multichannel sinks. 1311 // TODO: fix both format and multichannel issues with effects. 1312 if (mChannelCount != FCC_2) { 1313 ALOGW("checkEffectCompatibility_l(): effect %s for multichannel(%d) on MIXER" 1314 " thread %s", desc->name, mChannelCount, mThreadName); 1315 return BAD_VALUE; 1316 } 1317 audio_output_flags_t flags = mOutput->flags; 1318 if (hasFastMixer() || (flags & AUDIO_OUTPUT_FLAG_FAST)) { 1319 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 1320 // global effects are applied only to non fast tracks if they are SW 1321 if ((desc->flags & EFFECT_FLAG_HW_ACC_TUNNEL) == 0) { 1322 break; 1323 } 1324 } else if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) { 1325 // only post processing on output stage session 1326 if ((desc->flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_POST_PROC) { 1327 ALOGW("checkEffectCompatibility_l(): non post processing effect %s not allowed" 1328 " on output stage session", desc->name); 1329 return BAD_VALUE; 1330 } 1331 } else { 1332 // no restriction on effects applied on non fast tracks 1333 if ((hasAudioSession_l(sessionId) & ThreadBase::FAST_SESSION) == 0) { 1334 break; 1335 } 1336 } 1337 1338 // always allow effects without processing load or latency 1339 if ((desc->flags & EFFECT_FLAG_NO_PROCESS_MASK) == EFFECT_FLAG_NO_PROCESS) { 1340 break; 1341 } 1342 if (flags & AUDIO_OUTPUT_FLAG_RAW) { 1343 ALOGW("checkEffectCompatibility_l(): effect %s on playback thread in raw mode", 1344 desc->name); 1345 return BAD_VALUE; 1346 } 1347 if ((desc->flags & EFFECT_FLAG_HW_ACC_TUNNEL) == 0) { 1348 ALOGW("checkEffectCompatibility_l(): non HW effect %s on playback thread" 1349 " in fast mode", desc->name); 1350 return BAD_VALUE; 1351 } 1352 } 1353 } break; 1354 case OFFLOAD: 1355 // nothing actionable on offload threads, if the effect: 1356 // - is offloadable: the effect can be created 1357 // - is NOT offloadable: the effect should still be created, but EffectHandle::enable() 1358 // will take care of invalidating the tracks of the thread 1359 break; 1360 case DIRECT: 1361 // Reject any effect on Direct output threads for now, since the format of 1362 // mSinkBuffer is not guaranteed to be compatible with effect processing (PCM 16 stereo). 1363 ALOGW("checkEffectCompatibility_l(): effect %s on DIRECT output thread %s", 1364 desc->name, mThreadName); 1365 return BAD_VALUE; 1366 case DUPLICATING: 1367 // Reject any effect on mixer multichannel sinks. 1368 // TODO: fix both format and multichannel issues with effects. 1369 if (mChannelCount != FCC_2) { 1370 ALOGW("checkEffectCompatibility_l(): effect %s for multichannel(%d)" 1371 " on DUPLICATING thread %s", desc->name, mChannelCount, mThreadName); 1372 return BAD_VALUE; 1373 } 1374 if ((sessionId == AUDIO_SESSION_OUTPUT_STAGE) || (sessionId == AUDIO_SESSION_OUTPUT_MIX)) { 1375 ALOGW("checkEffectCompatibility_l(): global effect %s on DUPLICATING" 1376 " thread %s", desc->name, mThreadName); 1377 return BAD_VALUE; 1378 } 1379 if ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC) { 1380 ALOGW("checkEffectCompatibility_l(): post processing effect %s on" 1381 " DUPLICATING thread %s", desc->name, mThreadName); 1382 return BAD_VALUE; 1383 } 1384 if ((desc->flags & EFFECT_FLAG_HW_ACC_TUNNEL) != 0) { 1385 ALOGW("checkEffectCompatibility_l(): HW tunneled effect %s on" 1386 " DUPLICATING thread %s", desc->name, mThreadName); 1387 return BAD_VALUE; 1388 } 1389 break; 1390 default: 1391 LOG_ALWAYS_FATAL("checkEffectCompatibility_l(): wrong thread type %d", mType); 1392 } 1393 1394 return NO_ERROR; 1395} 1396 1397// ThreadBase::createEffect_l() must be called with AudioFlinger::mLock held 1398sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 1399 const sp<AudioFlinger::Client>& client, 1400 const sp<IEffectClient>& effectClient, 1401 int32_t priority, 1402 audio_session_t sessionId, 1403 effect_descriptor_t *desc, 1404 int *enabled, 1405 status_t *status) 1406{ 1407 sp<EffectModule> effect; 1408 sp<EffectHandle> handle; 1409 status_t lStatus; 1410 sp<EffectChain> chain; 1411 bool chainCreated = false; 1412 bool effectCreated = false; 1413 bool effectRegistered = false; 1414 1415 lStatus = initCheck(); 1416 if (lStatus != NO_ERROR) { 1417 ALOGW("createEffect_l() Audio driver not initialized."); 1418 goto Exit; 1419 } 1420 1421 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 1422 1423 { // scope for mLock 1424 Mutex::Autolock _l(mLock); 1425 1426 lStatus = checkEffectCompatibility_l(desc, sessionId); 1427 if (lStatus != NO_ERROR) { 1428 goto Exit; 1429 } 1430 1431 // check for existing effect chain with the requested audio session 1432 chain = getEffectChain_l(sessionId); 1433 if (chain == 0) { 1434 // create a new chain for this session 1435 ALOGV("createEffect_l() new effect chain for session %d", sessionId); 1436 chain = new EffectChain(this, sessionId); 1437 addEffectChain_l(chain); 1438 chain->setStrategy(getStrategyForSession_l(sessionId)); 1439 chainCreated = true; 1440 } else { 1441 effect = chain->getEffectFromDesc_l(desc); 1442 } 1443 1444 ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get()); 1445 1446 if (effect == 0) { 1447 audio_unique_id_t id = mAudioFlinger->nextUniqueId(AUDIO_UNIQUE_ID_USE_EFFECT); 1448 // Check CPU and memory usage 1449 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 1450 if (lStatus != NO_ERROR) { 1451 goto Exit; 1452 } 1453 effectRegistered = true; 1454 // create a new effect module if none present in the chain 1455 effect = new EffectModule(this, chain, desc, id, sessionId); 1456 lStatus = effect->status(); 1457 if (lStatus != NO_ERROR) { 1458 goto Exit; 1459 } 1460 effect->setOffloaded(mType == OFFLOAD, mId); 1461 1462 lStatus = chain->addEffect_l(effect); 1463 if (lStatus != NO_ERROR) { 1464 goto Exit; 1465 } 1466 effectCreated = true; 1467 1468 effect->setDevice(mOutDevice); 1469 effect->setDevice(mInDevice); 1470 effect->setMode(mAudioFlinger->getMode()); 1471 effect->setAudioSource(mAudioSource); 1472 } 1473 // create effect handle and connect it to effect module 1474 handle = new EffectHandle(effect, client, effectClient, priority); 1475 lStatus = handle->initCheck(); 1476 if (lStatus == OK) { 1477 lStatus = effect->addHandle(handle.get()); 1478 } 1479 if (enabled != NULL) { 1480 *enabled = (int)effect->isEnabled(); 1481 } 1482 } 1483 1484Exit: 1485 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 1486 Mutex::Autolock _l(mLock); 1487 if (effectCreated) { 1488 chain->removeEffect_l(effect); 1489 } 1490 if (effectRegistered) { 1491 AudioSystem::unregisterEffect(effect->id()); 1492 } 1493 if (chainCreated) { 1494 removeEffectChain_l(chain); 1495 } 1496 handle.clear(); 1497 } 1498 1499 *status = lStatus; 1500 return handle; 1501} 1502 1503sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect(audio_session_t sessionId, 1504 int effectId) 1505{ 1506 Mutex::Autolock _l(mLock); 1507 return getEffect_l(sessionId, effectId); 1508} 1509 1510sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(audio_session_t sessionId, 1511 int effectId) 1512{ 1513 sp<EffectChain> chain = getEffectChain_l(sessionId); 1514 return chain != 0 ? chain->getEffectFromId_l(effectId) : 0; 1515} 1516 1517// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 1518// PlaybackThread::mLock held 1519status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 1520{ 1521 // check for existing effect chain with the requested audio session 1522 audio_session_t sessionId = effect->sessionId(); 1523 sp<EffectChain> chain = getEffectChain_l(sessionId); 1524 bool chainCreated = false; 1525 1526 ALOGD_IF((mType == OFFLOAD) && !effect->isOffloadable(), 1527 "addEffect_l() on offloaded thread %p: effect %s does not support offload flags %x", 1528 this, effect->desc().name, effect->desc().flags); 1529 1530 if (chain == 0) { 1531 // create a new chain for this session 1532 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 1533 chain = new EffectChain(this, sessionId); 1534 addEffectChain_l(chain); 1535 chain->setStrategy(getStrategyForSession_l(sessionId)); 1536 chainCreated = true; 1537 } 1538 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 1539 1540 if (chain->getEffectFromId_l(effect->id()) != 0) { 1541 ALOGW("addEffect_l() %p effect %s already present in chain %p", 1542 this, effect->desc().name, chain.get()); 1543 return BAD_VALUE; 1544 } 1545 1546 effect->setOffloaded(mType == OFFLOAD, mId); 1547 1548 status_t status = chain->addEffect_l(effect); 1549 if (status != NO_ERROR) { 1550 if (chainCreated) { 1551 removeEffectChain_l(chain); 1552 } 1553 return status; 1554 } 1555 1556 effect->setDevice(mOutDevice); 1557 effect->setDevice(mInDevice); 1558 effect->setMode(mAudioFlinger->getMode()); 1559 effect->setAudioSource(mAudioSource); 1560 return NO_ERROR; 1561} 1562 1563void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 1564 1565 ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 1566 effect_descriptor_t desc = effect->desc(); 1567 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 1568 detachAuxEffect_l(effect->id()); 1569 } 1570 1571 sp<EffectChain> chain = effect->chain().promote(); 1572 if (chain != 0) { 1573 // remove effect chain if removing last effect 1574 if (chain->removeEffect_l(effect) == 0) { 1575 removeEffectChain_l(chain); 1576 } 1577 } else { 1578 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 1579 } 1580} 1581 1582void AudioFlinger::ThreadBase::lockEffectChains_l( 1583 Vector< sp<AudioFlinger::EffectChain> >& effectChains) 1584{ 1585 effectChains = mEffectChains; 1586 for (size_t i = 0; i < mEffectChains.size(); i++) { 1587 mEffectChains[i]->lock(); 1588 } 1589} 1590 1591void AudioFlinger::ThreadBase::unlockEffectChains( 1592 const Vector< sp<AudioFlinger::EffectChain> >& effectChains) 1593{ 1594 for (size_t i = 0; i < effectChains.size(); i++) { 1595 effectChains[i]->unlock(); 1596 } 1597} 1598 1599sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(audio_session_t sessionId) 1600{ 1601 Mutex::Autolock _l(mLock); 1602 return getEffectChain_l(sessionId); 1603} 1604 1605sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(audio_session_t sessionId) 1606 const 1607{ 1608 size_t size = mEffectChains.size(); 1609 for (size_t i = 0; i < size; i++) { 1610 if (mEffectChains[i]->sessionId() == sessionId) { 1611 return mEffectChains[i]; 1612 } 1613 } 1614 return 0; 1615} 1616 1617void AudioFlinger::ThreadBase::setMode(audio_mode_t mode) 1618{ 1619 Mutex::Autolock _l(mLock); 1620 size_t size = mEffectChains.size(); 1621 for (size_t i = 0; i < size; i++) { 1622 mEffectChains[i]->setMode_l(mode); 1623 } 1624} 1625 1626void AudioFlinger::ThreadBase::getAudioPortConfig(struct audio_port_config *config) 1627{ 1628 config->type = AUDIO_PORT_TYPE_MIX; 1629 config->ext.mix.handle = mId; 1630 config->sample_rate = mSampleRate; 1631 config->format = mFormat; 1632 config->channel_mask = mChannelMask; 1633 config->config_mask = AUDIO_PORT_CONFIG_SAMPLE_RATE|AUDIO_PORT_CONFIG_CHANNEL_MASK| 1634 AUDIO_PORT_CONFIG_FORMAT; 1635} 1636 1637void AudioFlinger::ThreadBase::systemReady() 1638{ 1639 Mutex::Autolock _l(mLock); 1640 if (mSystemReady) { 1641 return; 1642 } 1643 mSystemReady = true; 1644 1645 for (size_t i = 0; i < mPendingConfigEvents.size(); i++) { 1646 sendConfigEvent_l(mPendingConfigEvents.editItemAt(i)); 1647 } 1648 mPendingConfigEvents.clear(); 1649} 1650 1651 1652// ---------------------------------------------------------------------------- 1653// Playback 1654// ---------------------------------------------------------------------------- 1655 1656AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, 1657 AudioStreamOut* output, 1658 audio_io_handle_t id, 1659 audio_devices_t device, 1660 type_t type, 1661 bool systemReady) 1662 : ThreadBase(audioFlinger, id, device, AUDIO_DEVICE_NONE, type, systemReady), 1663 mNormalFrameCount(0), mSinkBuffer(NULL), 1664 mMixerBufferEnabled(AudioFlinger::kEnableExtendedPrecision), 1665 mMixerBuffer(NULL), 1666 mMixerBufferSize(0), 1667 mMixerBufferFormat(AUDIO_FORMAT_INVALID), 1668 mMixerBufferValid(false), 1669 mEffectBufferEnabled(AudioFlinger::kEnableExtendedPrecision), 1670 mEffectBuffer(NULL), 1671 mEffectBufferSize(0), 1672 mEffectBufferFormat(AUDIO_FORMAT_INVALID), 1673 mEffectBufferValid(false), 1674 mSuspended(0), mBytesWritten(0), 1675 mFramesWritten(0), 1676 mSuspendedFrames(0), 1677 mActiveTracksGeneration(0), 1678 // mStreamTypes[] initialized in constructor body 1679 mOutput(output), 1680 mLastWriteTime(-1), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false), 1681 mMixerStatus(MIXER_IDLE), 1682 mMixerStatusIgnoringFastTracks(MIXER_IDLE), 1683 mStandbyDelayNs(AudioFlinger::mStandbyTimeInNsecs), 1684 mBytesRemaining(0), 1685 mCurrentWriteLength(0), 1686 mUseAsyncWrite(false), 1687 mWriteAckSequence(0), 1688 mDrainSequence(0), 1689 mSignalPending(false), 1690 mScreenState(AudioFlinger::mScreenState), 1691 // index 0 is reserved for normal mixer's submix 1692 mFastTrackAvailMask(((1 << FastMixerState::sMaxFastTracks) - 1) & ~1), 1693 mHwSupportsPause(false), mHwPaused(false), mFlushPending(false) 1694{ 1695 snprintf(mThreadName, kThreadNameLength, "AudioOut_%X", id); 1696 mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mThreadName); 1697 1698 // Assumes constructor is called by AudioFlinger with it's mLock held, but 1699 // it would be safer to explicitly pass initial masterVolume/masterMute as 1700 // parameter. 1701 // 1702 // If the HAL we are using has support for master volume or master mute, 1703 // then do not attenuate or mute during mixing (just leave the volume at 1.0 1704 // and the mute set to false). 1705 mMasterVolume = audioFlinger->masterVolume_l(); 1706 mMasterMute = audioFlinger->masterMute_l(); 1707 if (mOutput && mOutput->audioHwDev) { 1708 if (mOutput->audioHwDev->canSetMasterVolume()) { 1709 mMasterVolume = 1.0; 1710 } 1711 1712 if (mOutput->audioHwDev->canSetMasterMute()) { 1713 mMasterMute = false; 1714 } 1715 } 1716 1717 readOutputParameters_l(); 1718 1719 // ++ operator does not compile 1720 for (audio_stream_type_t stream = AUDIO_STREAM_MIN; stream < AUDIO_STREAM_CNT; 1721 stream = (audio_stream_type_t) (stream + 1)) { 1722 mStreamTypes[stream].volume = mAudioFlinger->streamVolume_l(stream); 1723 mStreamTypes[stream].mute = mAudioFlinger->streamMute_l(stream); 1724 } 1725} 1726 1727AudioFlinger::PlaybackThread::~PlaybackThread() 1728{ 1729 mAudioFlinger->unregisterWriter(mNBLogWriter); 1730 free(mSinkBuffer); 1731 free(mMixerBuffer); 1732 free(mEffectBuffer); 1733} 1734 1735void AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) 1736{ 1737 dumpInternals(fd, args); 1738 dumpTracks(fd, args); 1739 dumpEffectChains(fd, args); 1740} 1741 1742void AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args __unused) 1743{ 1744 const size_t SIZE = 256; 1745 char buffer[SIZE]; 1746 String8 result; 1747 1748 result.appendFormat(" Stream volumes in dB: "); 1749 for (int i = 0; i < AUDIO_STREAM_CNT; ++i) { 1750 const stream_type_t *st = &mStreamTypes[i]; 1751 if (i > 0) { 1752 result.appendFormat(", "); 1753 } 1754 result.appendFormat("%d:%.2g", i, 20.0 * log10(st->volume)); 1755 if (st->mute) { 1756 result.append("M"); 1757 } 1758 } 1759 result.append("\n"); 1760 write(fd, result.string(), result.length()); 1761 result.clear(); 1762 1763 // These values are "raw"; they will wrap around. See prepareTracks_l() for a better way. 1764 FastTrackUnderruns underruns = getFastTrackUnderruns(0); 1765 dprintf(fd, " Normal mixer raw underrun counters: partial=%u empty=%u\n", 1766 underruns.mBitFields.mPartial, underruns.mBitFields.mEmpty); 1767 1768 size_t numtracks = mTracks.size(); 1769 size_t numactive = mActiveTracks.size(); 1770 dprintf(fd, " %zu Tracks", numtracks); 1771 size_t numactiveseen = 0; 1772 if (numtracks) { 1773 dprintf(fd, " of which %zu are active\n", numactive); 1774 Track::appendDumpHeader(result); 1775 for (size_t i = 0; i < numtracks; ++i) { 1776 sp<Track> track = mTracks[i]; 1777 if (track != 0) { 1778 bool active = mActiveTracks.indexOf(track) >= 0; 1779 if (active) { 1780 numactiveseen++; 1781 } 1782 track->dump(buffer, SIZE, active); 1783 result.append(buffer); 1784 } 1785 } 1786 } else { 1787 result.append("\n"); 1788 } 1789 if (numactiveseen != numactive) { 1790 // some tracks in the active list were not in the tracks list 1791 snprintf(buffer, SIZE, " The following tracks are in the active list but" 1792 " not in the track list\n"); 1793 result.append(buffer); 1794 Track::appendDumpHeader(result); 1795 for (size_t i = 0; i < numactive; ++i) { 1796 sp<Track> track = mActiveTracks[i].promote(); 1797 if (track != 0 && mTracks.indexOf(track) < 0) { 1798 track->dump(buffer, SIZE, true); 1799 result.append(buffer); 1800 } 1801 } 1802 } 1803 1804 write(fd, result.string(), result.size()); 1805} 1806 1807void AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 1808{ 1809 dprintf(fd, "\nOutput thread %p type %d (%s):\n", this, type(), threadTypeToString(type())); 1810 1811 dumpBase(fd, args); 1812 1813 dprintf(fd, " Normal frame count: %zu\n", mNormalFrameCount); 1814 dprintf(fd, " Last write occurred (msecs): %llu\n", 1815 (unsigned long long) ns2ms(systemTime() - mLastWriteTime)); 1816 dprintf(fd, " Total writes: %d\n", mNumWrites); 1817 dprintf(fd, " Delayed writes: %d\n", mNumDelayedWrites); 1818 dprintf(fd, " Blocked in write: %s\n", mInWrite ? "yes" : "no"); 1819 dprintf(fd, " Suspend count: %d\n", mSuspended); 1820 dprintf(fd, " Sink buffer : %p\n", mSinkBuffer); 1821 dprintf(fd, " Mixer buffer: %p\n", mMixerBuffer); 1822 dprintf(fd, " Effect buffer: %p\n", mEffectBuffer); 1823 dprintf(fd, " Fast track availMask=%#x\n", mFastTrackAvailMask); 1824 dprintf(fd, " Standby delay ns=%lld\n", (long long)mStandbyDelayNs); 1825 AudioStreamOut *output = mOutput; 1826 audio_output_flags_t flags = output != NULL ? output->flags : AUDIO_OUTPUT_FLAG_NONE; 1827 String8 flagsAsString = outputFlagsToString(flags); 1828 dprintf(fd, " AudioStreamOut: %p flags %#x (%s)\n", output, flags, flagsAsString.string()); 1829 dprintf(fd, " Frames written: %lld\n", (long long)mFramesWritten); 1830 dprintf(fd, " Suspended frames: %lld\n", (long long)mSuspendedFrames); 1831 if (mPipeSink.get() != nullptr) { 1832 dprintf(fd, " PipeSink frames written: %lld\n", (long long)mPipeSink->framesWritten()); 1833 } 1834 if (output != nullptr) { 1835 dprintf(fd, " Hal stream dump:\n"); 1836 (void)output->stream->common.dump(&output->stream->common, fd); 1837 } 1838} 1839 1840// Thread virtuals 1841 1842void AudioFlinger::PlaybackThread::onFirstRef() 1843{ 1844 run(mThreadName, ANDROID_PRIORITY_URGENT_AUDIO); 1845} 1846 1847// ThreadBase virtuals 1848void AudioFlinger::PlaybackThread::preExit() 1849{ 1850 ALOGV(" preExit()"); 1851 // FIXME this is using hard-coded strings but in the future, this functionality will be 1852 // converted to use audio HAL extensions required to support tunneling 1853 mOutput->stream->common.set_parameters(&mOutput->stream->common, "exiting=1"); 1854} 1855 1856// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 1857sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 1858 const sp<AudioFlinger::Client>& client, 1859 audio_stream_type_t streamType, 1860 uint32_t sampleRate, 1861 audio_format_t format, 1862 audio_channel_mask_t channelMask, 1863 size_t *pFrameCount, 1864 const sp<IMemory>& sharedBuffer, 1865 audio_session_t sessionId, 1866 audio_output_flags_t *flags, 1867 pid_t tid, 1868 int uid, 1869 status_t *status) 1870{ 1871 size_t frameCount = *pFrameCount; 1872 sp<Track> track; 1873 status_t lStatus; 1874 audio_output_flags_t outputFlags = mOutput->flags; 1875 1876 // special case for FAST flag considered OK if fast mixer is present 1877 if (hasFastMixer()) { 1878 outputFlags = (audio_output_flags_t)(outputFlags | AUDIO_OUTPUT_FLAG_FAST); 1879 } 1880 1881 // Check if requested flags are compatible with output stream flags 1882 if ((*flags & outputFlags) != *flags) { 1883 ALOGW("createTrack_l(): mismatch between requested flags (%08x) and output flags (%08x)", 1884 *flags, outputFlags); 1885 *flags = (audio_output_flags_t)(*flags & outputFlags); 1886 } 1887 1888 // client expresses a preference for FAST, but we get the final say 1889 if (*flags & AUDIO_OUTPUT_FLAG_FAST) { 1890 if ( 1891 // PCM data 1892 audio_is_linear_pcm(format) && 1893 // TODO: extract as a data library function that checks that a computationally 1894 // expensive downmixer is not required: isFastOutputChannelConversion() 1895 (channelMask == mChannelMask || 1896 mChannelMask != AUDIO_CHANNEL_OUT_STEREO || 1897 (channelMask == AUDIO_CHANNEL_OUT_MONO 1898 /* && mChannelMask == AUDIO_CHANNEL_OUT_STEREO */)) && 1899 // hardware sample rate 1900 (sampleRate == mSampleRate) && 1901 // normal mixer has an associated fast mixer 1902 hasFastMixer() && 1903 // there are sufficient fast track slots available 1904 (mFastTrackAvailMask != 0) 1905 // FIXME test that MixerThread for this fast track has a capable output HAL 1906 // FIXME add a permission test also? 1907 ) { 1908 // static tracks can have any nonzero framecount, streaming tracks check against minimum. 1909 if (sharedBuffer == 0) { 1910 // read the fast track multiplier property the first time it is needed 1911 int ok = pthread_once(&sFastTrackMultiplierOnce, sFastTrackMultiplierInit); 1912 if (ok != 0) { 1913 ALOGE("%s pthread_once failed: %d", __func__, ok); 1914 } 1915 frameCount = max(frameCount, mFrameCount * sFastTrackMultiplier); // incl framecount 0 1916 } 1917 1918 // check compatibility with audio effects. 1919 { // scope for mLock 1920 Mutex::Autolock _l(mLock); 1921 // do not accept RAW flag if post processing are present. Note that post processing on 1922 // a fast mixer are necessarily hardware 1923 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_STAGE); 1924 if (chain != 0) { 1925 ALOGV_IF((*flags & AUDIO_OUTPUT_FLAG_RAW) != 0, 1926 "AUDIO_OUTPUT_FLAG_RAW denied: post processing effect present"); 1927 *flags = (audio_output_flags_t)(*flags & ~AUDIO_OUTPUT_FLAG_RAW); 1928 } 1929 // Do not accept FAST flag if software global effects are present 1930 chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 1931 if (chain != 0) { 1932 ALOGV_IF((*flags & AUDIO_OUTPUT_FLAG_RAW) != 0, 1933 "AUDIO_OUTPUT_FLAG_RAW denied: global effect present"); 1934 *flags = (audio_output_flags_t)(*flags & ~AUDIO_OUTPUT_FLAG_RAW); 1935 if (chain->hasSoftwareEffect()) { 1936 ALOGV("AUDIO_OUTPUT_FLAG_FAST denied: software global effect present"); 1937 *flags = (audio_output_flags_t)(*flags & ~AUDIO_OUTPUT_FLAG_FAST); 1938 } 1939 } 1940 // Do not accept FAST flag if the session has software effects 1941 chain = getEffectChain_l(sessionId); 1942 if (chain != 0) { 1943 ALOGV_IF((*flags & AUDIO_OUTPUT_FLAG_RAW) != 0, 1944 "AUDIO_OUTPUT_FLAG_RAW denied: effect present on session"); 1945 *flags = (audio_output_flags_t)(*flags & ~AUDIO_OUTPUT_FLAG_RAW); 1946 if (chain->hasSoftwareEffect()) { 1947 ALOGV("AUDIO_OUTPUT_FLAG_FAST denied: software effect present on session"); 1948 *flags = (audio_output_flags_t)(*flags & ~AUDIO_OUTPUT_FLAG_FAST); 1949 } 1950 } 1951 } 1952 ALOGV_IF((*flags & AUDIO_OUTPUT_FLAG_FAST) != 0, 1953 "AUDIO_OUTPUT_FLAG_FAST accepted: frameCount=%zu mFrameCount=%zu", 1954 frameCount, mFrameCount); 1955 } else { 1956 ALOGV("AUDIO_OUTPUT_FLAG_FAST denied: sharedBuffer=%p frameCount=%zu " 1957 "mFrameCount=%zu format=%#x mFormat=%#x isLinear=%d channelMask=%#x " 1958 "sampleRate=%u mSampleRate=%u " 1959 "hasFastMixer=%d tid=%d fastTrackAvailMask=%#x", 1960 sharedBuffer.get(), frameCount, mFrameCount, format, mFormat, 1961 audio_is_linear_pcm(format), 1962 channelMask, sampleRate, mSampleRate, hasFastMixer(), tid, mFastTrackAvailMask); 1963 *flags = (audio_output_flags_t)(*flags & ~AUDIO_OUTPUT_FLAG_FAST); 1964 } 1965 } 1966 // For normal PCM streaming tracks, update minimum frame count. 1967 // For compatibility with AudioTrack calculation, buffer depth is forced 1968 // to be at least 2 x the normal mixer frame count and cover audio hardware latency. 1969 // This is probably too conservative, but legacy application code may depend on it. 1970 // If you change this calculation, also review the start threshold which is related. 1971 if (!(*flags & AUDIO_OUTPUT_FLAG_FAST) 1972 && audio_has_proportional_frames(format) && sharedBuffer == 0) { 1973 // this must match AudioTrack.cpp calculateMinFrameCount(). 1974 // TODO: Move to a common library 1975 uint32_t latencyMs = mOutput->stream->get_latency(mOutput->stream); 1976 uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate); 1977 if (minBufCount < 2) { 1978 minBufCount = 2; 1979 } 1980 // For normal mixing tracks, if speed is > 1.0f (normal), AudioTrack 1981 // or the client should compute and pass in a larger buffer request. 1982 size_t minFrameCount = 1983 minBufCount * sourceFramesNeededWithTimestretch( 1984 sampleRate, mNormalFrameCount, 1985 mSampleRate, AUDIO_TIMESTRETCH_SPEED_NORMAL /*speed*/); 1986 if (frameCount < minFrameCount) { // including frameCount == 0 1987 frameCount = minFrameCount; 1988 } 1989 } 1990 *pFrameCount = frameCount; 1991 1992 switch (mType) { 1993 1994 case DIRECT: 1995 if (audio_is_linear_pcm(format)) { // TODO maybe use audio_has_proportional_frames()? 1996 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1997 ALOGE("createTrack_l() Bad parameter: sampleRate %u format %#x, channelMask 0x%08x " 1998 "for output %p with format %#x", 1999 sampleRate, format, channelMask, mOutput, mFormat); 2000 lStatus = BAD_VALUE; 2001 goto Exit; 2002 } 2003 } 2004 break; 2005 2006 case OFFLOAD: 2007 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 2008 ALOGE("createTrack_l() Bad parameter: sampleRate %d format %#x, channelMask 0x%08x \"" 2009 "for output %p with format %#x", 2010 sampleRate, format, channelMask, mOutput, mFormat); 2011 lStatus = BAD_VALUE; 2012 goto Exit; 2013 } 2014 break; 2015 2016 default: 2017 if (!audio_is_linear_pcm(format)) { 2018 ALOGE("createTrack_l() Bad parameter: format %#x \"" 2019 "for output %p with format %#x", 2020 format, mOutput, mFormat); 2021 lStatus = BAD_VALUE; 2022 goto Exit; 2023 } 2024 if (sampleRate > mSampleRate * AUDIO_RESAMPLER_DOWN_RATIO_MAX) { 2025 ALOGE("Sample rate out of range: %u mSampleRate %u", sampleRate, mSampleRate); 2026 lStatus = BAD_VALUE; 2027 goto Exit; 2028 } 2029 break; 2030 2031 } 2032 2033 lStatus = initCheck(); 2034 if (lStatus != NO_ERROR) { 2035 ALOGE("createTrack_l() audio driver not initialized"); 2036 goto Exit; 2037 } 2038 2039 { // scope for mLock 2040 Mutex::Autolock _l(mLock); 2041 2042 // all tracks in same audio session must share the same routing strategy otherwise 2043 // conflicts will happen when tracks are moved from one output to another by audio policy 2044 // manager 2045 uint32_t strategy = AudioSystem::getStrategyForStream(streamType); 2046 for (size_t i = 0; i < mTracks.size(); ++i) { 2047 sp<Track> t = mTracks[i]; 2048 if (t != 0 && t->isExternalTrack()) { 2049 uint32_t actual = AudioSystem::getStrategyForStream(t->streamType()); 2050 if (sessionId == t->sessionId() && strategy != actual) { 2051 ALOGE("createTrack_l() mismatched strategy; expected %u but found %u", 2052 strategy, actual); 2053 lStatus = BAD_VALUE; 2054 goto Exit; 2055 } 2056 } 2057 } 2058 2059 track = new Track(this, client, streamType, sampleRate, format, 2060 channelMask, frameCount, NULL, sharedBuffer, 2061 sessionId, uid, *flags, TrackBase::TYPE_DEFAULT); 2062 2063 lStatus = track != 0 ? track->initCheck() : (status_t) NO_MEMORY; 2064 if (lStatus != NO_ERROR) { 2065 ALOGE("createTrack_l() initCheck failed %d; no control block?", lStatus); 2066 // track must be cleared from the caller as the caller has the AF lock 2067 goto Exit; 2068 } 2069 mTracks.add(track); 2070 2071 sp<EffectChain> chain = getEffectChain_l(sessionId); 2072 if (chain != 0) { 2073 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 2074 track->setMainBuffer(chain->inBuffer()); 2075 chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType())); 2076 chain->incTrackCnt(); 2077 } 2078 2079 if ((*flags & AUDIO_OUTPUT_FLAG_FAST) && (tid != -1)) { 2080 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 2081 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 2082 // so ask activity manager to do this on our behalf 2083 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 2084 } 2085 } 2086 2087 lStatus = NO_ERROR; 2088 2089Exit: 2090 *status = lStatus; 2091 return track; 2092} 2093 2094uint32_t AudioFlinger::PlaybackThread::correctLatency_l(uint32_t latency) const 2095{ 2096 return latency; 2097} 2098 2099uint32_t AudioFlinger::PlaybackThread::latency() const 2100{ 2101 Mutex::Autolock _l(mLock); 2102 return latency_l(); 2103} 2104uint32_t AudioFlinger::PlaybackThread::latency_l() const 2105{ 2106 if (initCheck() == NO_ERROR) { 2107 return correctLatency_l(mOutput->stream->get_latency(mOutput->stream)); 2108 } else { 2109 return 0; 2110 } 2111} 2112 2113void AudioFlinger::PlaybackThread::setMasterVolume(float value) 2114{ 2115 Mutex::Autolock _l(mLock); 2116 // Don't apply master volume in SW if our HAL can do it for us. 2117 if (mOutput && mOutput->audioHwDev && 2118 mOutput->audioHwDev->canSetMasterVolume()) { 2119 mMasterVolume = 1.0; 2120 } else { 2121 mMasterVolume = value; 2122 } 2123} 2124 2125void AudioFlinger::PlaybackThread::setMasterMute(bool muted) 2126{ 2127 Mutex::Autolock _l(mLock); 2128 // Don't apply master mute in SW if our HAL can do it for us. 2129 if (mOutput && mOutput->audioHwDev && 2130 mOutput->audioHwDev->canSetMasterMute()) { 2131 mMasterMute = false; 2132 } else { 2133 mMasterMute = muted; 2134 } 2135} 2136 2137void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) 2138{ 2139 Mutex::Autolock _l(mLock); 2140 mStreamTypes[stream].volume = value; 2141 broadcast_l(); 2142} 2143 2144void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) 2145{ 2146 Mutex::Autolock _l(mLock); 2147 mStreamTypes[stream].mute = muted; 2148 broadcast_l(); 2149} 2150 2151float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const 2152{ 2153 Mutex::Autolock _l(mLock); 2154 return mStreamTypes[stream].volume; 2155} 2156 2157// addTrack_l() must be called with ThreadBase::mLock held 2158status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 2159{ 2160 status_t status = ALREADY_EXISTS; 2161 2162 if (mActiveTracks.indexOf(track) < 0) { 2163 // the track is newly added, make sure it fills up all its 2164 // buffers before playing. This is to ensure the client will 2165 // effectively get the latency it requested. 2166 if (track->isExternalTrack()) { 2167 TrackBase::track_state state = track->mState; 2168 mLock.unlock(); 2169 status = AudioSystem::startOutput(mId, track->streamType(), 2170 track->sessionId()); 2171 mLock.lock(); 2172 // abort track was stopped/paused while we released the lock 2173 if (state != track->mState) { 2174 if (status == NO_ERROR) { 2175 mLock.unlock(); 2176 AudioSystem::stopOutput(mId, track->streamType(), 2177 track->sessionId()); 2178 mLock.lock(); 2179 } 2180 return INVALID_OPERATION; 2181 } 2182 // abort if start is rejected by audio policy manager 2183 if (status != NO_ERROR) { 2184 return PERMISSION_DENIED; 2185 } 2186#ifdef ADD_BATTERY_DATA 2187 // to track the speaker usage 2188 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 2189#endif 2190 } 2191 2192 // set retry count for buffer fill 2193 if (track->isOffloaded()) { 2194 if (track->isStopping_1()) { 2195 track->mRetryCount = kMaxTrackStopRetriesOffload; 2196 } else { 2197 track->mRetryCount = kMaxTrackStartupRetriesOffload; 2198 } 2199 track->mFillingUpStatus = mStandby ? Track::FS_FILLING : Track::FS_FILLED; 2200 } else { 2201 track->mRetryCount = kMaxTrackStartupRetries; 2202 track->mFillingUpStatus = 2203 track->sharedBuffer() != 0 ? Track::FS_FILLED : Track::FS_FILLING; 2204 } 2205 2206 track->mResetDone = false; 2207 track->mPresentationCompleteFrames = 0; 2208 mActiveTracks.add(track); 2209 mWakeLockUids.add(track->uid()); 2210 mActiveTracksGeneration++; 2211 mLatestActiveTrack = track; 2212 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 2213 if (chain != 0) { 2214 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), 2215 track->sessionId()); 2216 chain->incActiveTrackCnt(); 2217 } 2218 2219 status = NO_ERROR; 2220 } 2221 2222 onAddNewTrack_l(); 2223 return status; 2224} 2225 2226bool AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 2227{ 2228 track->terminate(); 2229 // active tracks are removed by threadLoop() 2230 bool trackActive = (mActiveTracks.indexOf(track) >= 0); 2231 track->mState = TrackBase::STOPPED; 2232 if (!trackActive) { 2233 removeTrack_l(track); 2234 } else if (track->isFastTrack() || track->isOffloaded() || track->isDirect()) { 2235 track->mState = TrackBase::STOPPING_1; 2236 } 2237 2238 return trackActive; 2239} 2240 2241void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 2242{ 2243 track->triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 2244 mTracks.remove(track); 2245 deleteTrackName_l(track->name()); 2246 // redundant as track is about to be destroyed, for dumpsys only 2247 track->mName = -1; 2248 if (track->isFastTrack()) { 2249 int index = track->mFastIndex; 2250 ALOG_ASSERT(0 < index && index < (int)FastMixerState::sMaxFastTracks); 2251 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << index))); 2252 mFastTrackAvailMask |= 1 << index; 2253 // redundant as track is about to be destroyed, for dumpsys only 2254 track->mFastIndex = -1; 2255 } 2256 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 2257 if (chain != 0) { 2258 chain->decTrackCnt(); 2259 } 2260} 2261 2262void AudioFlinger::PlaybackThread::broadcast_l() 2263{ 2264 // Thread could be blocked waiting for async 2265 // so signal it to handle state changes immediately 2266 // If threadLoop is currently unlocked a signal of mWaitWorkCV will 2267 // be lost so we also flag to prevent it blocking on mWaitWorkCV 2268 mSignalPending = true; 2269 mWaitWorkCV.broadcast(); 2270} 2271 2272String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 2273{ 2274 Mutex::Autolock _l(mLock); 2275 if (initCheck() != NO_ERROR) { 2276 return String8(); 2277 } 2278 2279 char *s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 2280 const String8 out_s8(s); 2281 free(s); 2282 return out_s8; 2283} 2284 2285void AudioFlinger::PlaybackThread::ioConfigChanged(audio_io_config_event event, pid_t pid) { 2286 sp<AudioIoDescriptor> desc = new AudioIoDescriptor(); 2287 ALOGV("PlaybackThread::ioConfigChanged, thread %p, event %d", this, event); 2288 2289 desc->mIoHandle = mId; 2290 2291 switch (event) { 2292 case AUDIO_OUTPUT_OPENED: 2293 case AUDIO_OUTPUT_CONFIG_CHANGED: 2294 desc->mPatch = mPatch; 2295 desc->mChannelMask = mChannelMask; 2296 desc->mSamplingRate = mSampleRate; 2297 desc->mFormat = mFormat; 2298 desc->mFrameCount = mNormalFrameCount; // FIXME see 2299 // AudioFlinger::frameCount(audio_io_handle_t) 2300 desc->mFrameCountHAL = mFrameCount; 2301 desc->mLatency = latency_l(); 2302 break; 2303 2304 case AUDIO_OUTPUT_CLOSED: 2305 default: 2306 break; 2307 } 2308 mAudioFlinger->ioConfigChanged(event, desc, pid); 2309} 2310 2311void AudioFlinger::PlaybackThread::writeCallback() 2312{ 2313 ALOG_ASSERT(mCallbackThread != 0); 2314 mCallbackThread->resetWriteBlocked(); 2315} 2316 2317void AudioFlinger::PlaybackThread::drainCallback() 2318{ 2319 ALOG_ASSERT(mCallbackThread != 0); 2320 mCallbackThread->resetDraining(); 2321} 2322 2323void AudioFlinger::PlaybackThread::errorCallback() 2324{ 2325 ALOG_ASSERT(mCallbackThread != 0); 2326 mCallbackThread->setAsyncError(); 2327} 2328 2329void AudioFlinger::PlaybackThread::resetWriteBlocked(uint32_t sequence) 2330{ 2331 Mutex::Autolock _l(mLock); 2332 // reject out of sequence requests 2333 if ((mWriteAckSequence & 1) && (sequence == mWriteAckSequence)) { 2334 mWriteAckSequence &= ~1; 2335 mWaitWorkCV.signal(); 2336 } 2337} 2338 2339void AudioFlinger::PlaybackThread::resetDraining(uint32_t sequence) 2340{ 2341 Mutex::Autolock _l(mLock); 2342 // reject out of sequence requests 2343 if ((mDrainSequence & 1) && (sequence == mDrainSequence)) { 2344 mDrainSequence &= ~1; 2345 mWaitWorkCV.signal(); 2346 } 2347} 2348 2349// static 2350int AudioFlinger::PlaybackThread::asyncCallback(stream_callback_event_t event, 2351 void *param __unused, 2352 void *cookie) 2353{ 2354 AudioFlinger::PlaybackThread *me = (AudioFlinger::PlaybackThread *)cookie; 2355 ALOGV("asyncCallback() event %d", event); 2356 switch (event) { 2357 case STREAM_CBK_EVENT_WRITE_READY: 2358 me->writeCallback(); 2359 break; 2360 case STREAM_CBK_EVENT_DRAIN_READY: 2361 me->drainCallback(); 2362 break; 2363 case STREAM_CBK_EVENT_ERROR: 2364 me->errorCallback(); 2365 break; 2366 default: 2367 ALOGW("asyncCallback() unknown event %d", event); 2368 break; 2369 } 2370 return 0; 2371} 2372 2373void AudioFlinger::PlaybackThread::readOutputParameters_l() 2374{ 2375 // unfortunately we have no way of recovering from errors here, hence the LOG_ALWAYS_FATAL 2376 mSampleRate = mOutput->getSampleRate(); 2377 mChannelMask = mOutput->getChannelMask(); 2378 if (!audio_is_output_channel(mChannelMask)) { 2379 LOG_ALWAYS_FATAL("HAL channel mask %#x not valid for output", mChannelMask); 2380 } 2381 if ((mType == MIXER || mType == DUPLICATING) 2382 && !isValidPcmSinkChannelMask(mChannelMask)) { 2383 LOG_ALWAYS_FATAL("HAL channel mask %#x not supported for mixed output", 2384 mChannelMask); 2385 } 2386 mChannelCount = audio_channel_count_from_out_mask(mChannelMask); 2387 2388 // Get actual HAL format. 2389 mHALFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 2390 // Get format from the shim, which will be different than the HAL format 2391 // if playing compressed audio over HDMI passthrough. 2392 mFormat = mOutput->getFormat(); 2393 if (!audio_is_valid_format(mFormat)) { 2394 LOG_ALWAYS_FATAL("HAL format %#x not valid for output", mFormat); 2395 } 2396 if ((mType == MIXER || mType == DUPLICATING) 2397 && !isValidPcmSinkFormat(mFormat)) { 2398 LOG_FATAL("HAL format %#x not supported for mixed output", 2399 mFormat); 2400 } 2401 mFrameSize = mOutput->getFrameSize(); 2402 mBufferSize = mOutput->stream->common.get_buffer_size(&mOutput->stream->common); 2403 mFrameCount = mBufferSize / mFrameSize; 2404 if (mFrameCount & 15) { 2405 ALOGW("HAL output buffer size is %zu frames but AudioMixer requires multiples of 16 frames", 2406 mFrameCount); 2407 } 2408 2409 if ((mOutput->flags & AUDIO_OUTPUT_FLAG_NON_BLOCKING) && 2410 (mOutput->stream->set_callback != NULL)) { 2411 if (mOutput->stream->set_callback(mOutput->stream, 2412 AudioFlinger::PlaybackThread::asyncCallback, this) == 0) { 2413 mUseAsyncWrite = true; 2414 mCallbackThread = new AudioFlinger::AsyncCallbackThread(this); 2415 } 2416 } 2417 2418 mHwSupportsPause = false; 2419 if (mOutput->flags & AUDIO_OUTPUT_FLAG_DIRECT) { 2420 if (mOutput->stream->pause != NULL) { 2421 if (mOutput->stream->resume != NULL) { 2422 mHwSupportsPause = true; 2423 } else { 2424 ALOGW("direct output implements pause but not resume"); 2425 } 2426 } else if (mOutput->stream->resume != NULL) { 2427 ALOGW("direct output implements resume but not pause"); 2428 } 2429 } 2430 if (!mHwSupportsPause && mOutput->flags & AUDIO_OUTPUT_FLAG_HW_AV_SYNC) { 2431 LOG_ALWAYS_FATAL("HW_AV_SYNC requested but HAL does not implement pause and resume"); 2432 } 2433 2434 if (mType == DUPLICATING && mMixerBufferEnabled && mEffectBufferEnabled) { 2435 // For best precision, we use float instead of the associated output 2436 // device format (typically PCM 16 bit). 2437 2438 mFormat = AUDIO_FORMAT_PCM_FLOAT; 2439 mFrameSize = mChannelCount * audio_bytes_per_sample(mFormat); 2440 mBufferSize = mFrameSize * mFrameCount; 2441 2442 // TODO: We currently use the associated output device channel mask and sample rate. 2443 // (1) Perhaps use the ORed channel mask of all downstream MixerThreads 2444 // (if a valid mask) to avoid premature downmix. 2445 // (2) Perhaps use the maximum sample rate of all downstream MixerThreads 2446 // instead of the output device sample rate to avoid loss of high frequency information. 2447 // This may need to be updated as MixerThread/OutputTracks are added and not here. 2448 } 2449 2450 // Calculate size of normal sink buffer relative to the HAL output buffer size 2451 double multiplier = 1.0; 2452 if (mType == MIXER && (kUseFastMixer == FastMixer_Static || 2453 kUseFastMixer == FastMixer_Dynamic)) { 2454 size_t minNormalFrameCount = (kMinNormalSinkBufferSizeMs * mSampleRate) / 1000; 2455 size_t maxNormalFrameCount = (kMaxNormalSinkBufferSizeMs * mSampleRate) / 1000; 2456 2457 // round up minimum and round down maximum to nearest 16 frames to satisfy AudioMixer 2458 minNormalFrameCount = (minNormalFrameCount + 15) & ~15; 2459 maxNormalFrameCount = maxNormalFrameCount & ~15; 2460 if (maxNormalFrameCount < minNormalFrameCount) { 2461 maxNormalFrameCount = minNormalFrameCount; 2462 } 2463 multiplier = (double) minNormalFrameCount / (double) mFrameCount; 2464 if (multiplier <= 1.0) { 2465 multiplier = 1.0; 2466 } else if (multiplier <= 2.0) { 2467 if (2 * mFrameCount <= maxNormalFrameCount) { 2468 multiplier = 2.0; 2469 } else { 2470 multiplier = (double) maxNormalFrameCount / (double) mFrameCount; 2471 } 2472 } else { 2473 multiplier = floor(multiplier); 2474 } 2475 } 2476 mNormalFrameCount = multiplier * mFrameCount; 2477 // round up to nearest 16 frames to satisfy AudioMixer 2478 if (mType == MIXER || mType == DUPLICATING) { 2479 mNormalFrameCount = (mNormalFrameCount + 15) & ~15; 2480 } 2481 ALOGI("HAL output buffer size %zu frames, normal sink buffer size %zu frames", mFrameCount, 2482 mNormalFrameCount); 2483 2484 // Check if we want to throttle the processing to no more than 2x normal rate 2485 mThreadThrottle = property_get_bool("af.thread.throttle", true /* default_value */); 2486 mThreadThrottleTimeMs = 0; 2487 mThreadThrottleEndMs = 0; 2488 mHalfBufferMs = mNormalFrameCount * 1000 / (2 * mSampleRate); 2489 2490 // mSinkBuffer is the sink buffer. Size is always multiple-of-16 frames. 2491 // Originally this was int16_t[] array, need to remove legacy implications. 2492 free(mSinkBuffer); 2493 mSinkBuffer = NULL; 2494 // For sink buffer size, we use the frame size from the downstream sink to avoid problems 2495 // with non PCM formats for compressed music, e.g. AAC, and Offload threads. 2496 const size_t sinkBufferSize = mNormalFrameCount * mFrameSize; 2497 (void)posix_memalign(&mSinkBuffer, 32, sinkBufferSize); 2498 2499 // We resize the mMixerBuffer according to the requirements of the sink buffer which 2500 // drives the output. 2501 free(mMixerBuffer); 2502 mMixerBuffer = NULL; 2503 if (mMixerBufferEnabled) { 2504 mMixerBufferFormat = AUDIO_FORMAT_PCM_FLOAT; // also valid: AUDIO_FORMAT_PCM_16_BIT. 2505 mMixerBufferSize = mNormalFrameCount * mChannelCount 2506 * audio_bytes_per_sample(mMixerBufferFormat); 2507 (void)posix_memalign(&mMixerBuffer, 32, mMixerBufferSize); 2508 } 2509 free(mEffectBuffer); 2510 mEffectBuffer = NULL; 2511 if (mEffectBufferEnabled) { 2512 mEffectBufferFormat = AUDIO_FORMAT_PCM_16_BIT; // Note: Effects support 16b only 2513 mEffectBufferSize = mNormalFrameCount * mChannelCount 2514 * audio_bytes_per_sample(mEffectBufferFormat); 2515 (void)posix_memalign(&mEffectBuffer, 32, mEffectBufferSize); 2516 } 2517 2518 // force reconfiguration of effect chains and engines to take new buffer size and audio 2519 // parameters into account 2520 // Note that mLock is not held when readOutputParameters_l() is called from the constructor 2521 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 2522 // matter. 2523 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 2524 Vector< sp<EffectChain> > effectChains = mEffectChains; 2525 for (size_t i = 0; i < effectChains.size(); i ++) { 2526 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 2527 } 2528} 2529 2530 2531status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames) 2532{ 2533 if (halFrames == NULL || dspFrames == NULL) { 2534 return BAD_VALUE; 2535 } 2536 Mutex::Autolock _l(mLock); 2537 if (initCheck() != NO_ERROR) { 2538 return INVALID_OPERATION; 2539 } 2540 int64_t framesWritten = mBytesWritten / mFrameSize; 2541 *halFrames = framesWritten; 2542 2543 if (isSuspended()) { 2544 // return an estimation of rendered frames when the output is suspended 2545 size_t latencyFrames = (latency_l() * mSampleRate) / 1000; 2546 *dspFrames = (uint32_t) 2547 (framesWritten >= (int64_t)latencyFrames ? framesWritten - latencyFrames : 0); 2548 return NO_ERROR; 2549 } else { 2550 status_t status; 2551 uint32_t frames; 2552 status = mOutput->getRenderPosition(&frames); 2553 *dspFrames = (size_t)frames; 2554 return status; 2555 } 2556} 2557 2558// hasAudioSession_l() must be called with ThreadBase::mLock held 2559uint32_t AudioFlinger::PlaybackThread::hasAudioSession_l(audio_session_t sessionId) const 2560{ 2561 uint32_t result = 0; 2562 if (getEffectChain_l(sessionId) != 0) { 2563 result = EFFECT_SESSION; 2564 } 2565 2566 for (size_t i = 0; i < mTracks.size(); ++i) { 2567 sp<Track> track = mTracks[i]; 2568 if (sessionId == track->sessionId() && !track->isInvalid()) { 2569 result |= TRACK_SESSION; 2570 if (track->isFastTrack()) { 2571 result |= FAST_SESSION; 2572 } 2573 break; 2574 } 2575 } 2576 2577 return result; 2578} 2579 2580uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(audio_session_t sessionId) 2581{ 2582 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 2583 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 2584 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 2585 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 2586 } 2587 for (size_t i = 0; i < mTracks.size(); i++) { 2588 sp<Track> track = mTracks[i]; 2589 if (sessionId == track->sessionId() && !track->isInvalid()) { 2590 return AudioSystem::getStrategyForStream(track->streamType()); 2591 } 2592 } 2593 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 2594} 2595 2596 2597AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 2598{ 2599 Mutex::Autolock _l(mLock); 2600 return mOutput; 2601} 2602 2603AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 2604{ 2605 Mutex::Autolock _l(mLock); 2606 AudioStreamOut *output = mOutput; 2607 mOutput = NULL; 2608 // FIXME FastMixer might also have a raw ptr to mOutputSink; 2609 // must push a NULL and wait for ack 2610 mOutputSink.clear(); 2611 mPipeSink.clear(); 2612 mNormalSink.clear(); 2613 return output; 2614} 2615 2616// this method must always be called either with ThreadBase mLock held or inside the thread loop 2617audio_stream_t* AudioFlinger::PlaybackThread::stream() const 2618{ 2619 if (mOutput == NULL) { 2620 return NULL; 2621 } 2622 return &mOutput->stream->common; 2623} 2624 2625uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() const 2626{ 2627 return (uint32_t)((uint32_t)((mNormalFrameCount * 1000) / mSampleRate) * 1000); 2628} 2629 2630status_t AudioFlinger::PlaybackThread::setSyncEvent(const sp<SyncEvent>& event) 2631{ 2632 if (!isValidSyncEvent(event)) { 2633 return BAD_VALUE; 2634 } 2635 2636 Mutex::Autolock _l(mLock); 2637 2638 for (size_t i = 0; i < mTracks.size(); ++i) { 2639 sp<Track> track = mTracks[i]; 2640 if (event->triggerSession() == track->sessionId()) { 2641 (void) track->setSyncEvent(event); 2642 return NO_ERROR; 2643 } 2644 } 2645 2646 return NAME_NOT_FOUND; 2647} 2648 2649bool AudioFlinger::PlaybackThread::isValidSyncEvent(const sp<SyncEvent>& event) const 2650{ 2651 return event->type() == AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE; 2652} 2653 2654void AudioFlinger::PlaybackThread::threadLoop_removeTracks( 2655 const Vector< sp<Track> >& tracksToRemove) 2656{ 2657 size_t count = tracksToRemove.size(); 2658 if (count > 0) { 2659 for (size_t i = 0 ; i < count ; i++) { 2660 const sp<Track>& track = tracksToRemove.itemAt(i); 2661 if (track->isExternalTrack()) { 2662 AudioSystem::stopOutput(mId, track->streamType(), 2663 track->sessionId()); 2664#ifdef ADD_BATTERY_DATA 2665 // to track the speaker usage 2666 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 2667#endif 2668 if (track->isTerminated()) { 2669 AudioSystem::releaseOutput(mId, track->streamType(), 2670 track->sessionId()); 2671 } 2672 } 2673 } 2674 } 2675} 2676 2677void AudioFlinger::PlaybackThread::checkSilentMode_l() 2678{ 2679 if (!mMasterMute) { 2680 char value[PROPERTY_VALUE_MAX]; 2681 if (mOutDevice == AUDIO_DEVICE_OUT_REMOTE_SUBMIX) { 2682 ALOGD("ro.audio.silent will be ignored for threads on AUDIO_DEVICE_OUT_REMOTE_SUBMIX"); 2683 return; 2684 } 2685 if (property_get("ro.audio.silent", value, "0") > 0) { 2686 char *endptr; 2687 unsigned long ul = strtoul(value, &endptr, 0); 2688 if (*endptr == '\0' && ul != 0) { 2689 ALOGD("Silence is golden"); 2690 // The setprop command will not allow a property to be changed after 2691 // the first time it is set, so we don't have to worry about un-muting. 2692 setMasterMute_l(true); 2693 } 2694 } 2695 } 2696} 2697 2698// shared by MIXER and DIRECT, overridden by DUPLICATING 2699ssize_t AudioFlinger::PlaybackThread::threadLoop_write() 2700{ 2701 mInWrite = true; 2702 ssize_t bytesWritten; 2703 const size_t offset = mCurrentWriteLength - mBytesRemaining; 2704 2705 // If an NBAIO sink is present, use it to write the normal mixer's submix 2706 if (mNormalSink != 0) { 2707 2708 const size_t count = mBytesRemaining / mFrameSize; 2709 2710 ATRACE_BEGIN("write"); 2711 // update the setpoint when AudioFlinger::mScreenState changes 2712 uint32_t screenState = AudioFlinger::mScreenState; 2713 if (screenState != mScreenState) { 2714 mScreenState = screenState; 2715 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 2716 if (pipe != NULL) { 2717 pipe->setAvgFrames((mScreenState & 1) ? 2718 (pipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2719 } 2720 } 2721 ssize_t framesWritten = mNormalSink->write((char *)mSinkBuffer + offset, count); 2722 ATRACE_END(); 2723 if (framesWritten > 0) { 2724 bytesWritten = framesWritten * mFrameSize; 2725 } else { 2726 bytesWritten = framesWritten; 2727 } 2728 // otherwise use the HAL / AudioStreamOut directly 2729 } else { 2730 // Direct output and offload threads 2731 2732 if (mUseAsyncWrite) { 2733 ALOGW_IF(mWriteAckSequence & 1, "threadLoop_write(): out of sequence write request"); 2734 mWriteAckSequence += 2; 2735 mWriteAckSequence |= 1; 2736 ALOG_ASSERT(mCallbackThread != 0); 2737 mCallbackThread->setWriteBlocked(mWriteAckSequence); 2738 } 2739 // FIXME We should have an implementation of timestamps for direct output threads. 2740 // They are used e.g for multichannel PCM playback over HDMI. 2741 bytesWritten = mOutput->write((char *)mSinkBuffer + offset, mBytesRemaining); 2742 2743 if (mUseAsyncWrite && 2744 ((bytesWritten < 0) || (bytesWritten == (ssize_t)mBytesRemaining))) { 2745 // do not wait for async callback in case of error of full write 2746 mWriteAckSequence &= ~1; 2747 ALOG_ASSERT(mCallbackThread != 0); 2748 mCallbackThread->setWriteBlocked(mWriteAckSequence); 2749 } 2750 } 2751 2752 mNumWrites++; 2753 mInWrite = false; 2754 mStandby = false; 2755 return bytesWritten; 2756} 2757 2758void AudioFlinger::PlaybackThread::threadLoop_drain() 2759{ 2760 if (mOutput->stream->drain) { 2761 ALOGV("draining %s", (mMixerStatus == MIXER_DRAIN_TRACK) ? "early" : "full"); 2762 if (mUseAsyncWrite) { 2763 ALOGW_IF(mDrainSequence & 1, "threadLoop_drain(): out of sequence drain request"); 2764 mDrainSequence |= 1; 2765 ALOG_ASSERT(mCallbackThread != 0); 2766 mCallbackThread->setDraining(mDrainSequence); 2767 } 2768 mOutput->stream->drain(mOutput->stream, 2769 (mMixerStatus == MIXER_DRAIN_TRACK) ? AUDIO_DRAIN_EARLY_NOTIFY 2770 : AUDIO_DRAIN_ALL); 2771 } 2772} 2773 2774void AudioFlinger::PlaybackThread::threadLoop_exit() 2775{ 2776 { 2777 Mutex::Autolock _l(mLock); 2778 for (size_t i = 0; i < mTracks.size(); i++) { 2779 sp<Track> track = mTracks[i]; 2780 track->invalidate(); 2781 } 2782 } 2783} 2784 2785/* 2786The derived values that are cached: 2787 - mSinkBufferSize from frame count * frame size 2788 - mActiveSleepTimeUs from activeSleepTimeUs() 2789 - mIdleSleepTimeUs from idleSleepTimeUs() 2790 - mStandbyDelayNs from mActiveSleepTimeUs (DIRECT only) or forced to at least 2791 kDefaultStandbyTimeInNsecs when connected to an A2DP device. 2792 - maxPeriod from frame count and sample rate (MIXER only) 2793 2794The parameters that affect these derived values are: 2795 - frame count 2796 - frame size 2797 - sample rate 2798 - device type: A2DP or not 2799 - device latency 2800 - format: PCM or not 2801 - active sleep time 2802 - idle sleep time 2803*/ 2804 2805void AudioFlinger::PlaybackThread::cacheParameters_l() 2806{ 2807 mSinkBufferSize = mNormalFrameCount * mFrameSize; 2808 mActiveSleepTimeUs = activeSleepTimeUs(); 2809 mIdleSleepTimeUs = idleSleepTimeUs(); 2810 2811 // make sure standby delay is not too short when connected to an A2DP sink to avoid 2812 // truncating audio when going to standby. 2813 mStandbyDelayNs = AudioFlinger::mStandbyTimeInNsecs; 2814 if ((mOutDevice & AUDIO_DEVICE_OUT_ALL_A2DP) != 0) { 2815 if (mStandbyDelayNs < kDefaultStandbyTimeInNsecs) { 2816 mStandbyDelayNs = kDefaultStandbyTimeInNsecs; 2817 } 2818 } 2819} 2820 2821bool AudioFlinger::PlaybackThread::invalidateTracks_l(audio_stream_type_t streamType) 2822{ 2823 ALOGV("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %zu", 2824 this, streamType, mTracks.size()); 2825 bool trackMatch = false; 2826 size_t size = mTracks.size(); 2827 for (size_t i = 0; i < size; i++) { 2828 sp<Track> t = mTracks[i]; 2829 if (t->streamType() == streamType && t->isExternalTrack()) { 2830 t->invalidate(); 2831 trackMatch = true; 2832 } 2833 } 2834 return trackMatch; 2835} 2836 2837void AudioFlinger::PlaybackThread::invalidateTracks(audio_stream_type_t streamType) 2838{ 2839 Mutex::Autolock _l(mLock); 2840 invalidateTracks_l(streamType); 2841} 2842 2843status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 2844{ 2845 audio_session_t session = chain->sessionId(); 2846 int16_t* buffer = reinterpret_cast<int16_t*>(mEffectBufferEnabled 2847 ? mEffectBuffer : mSinkBuffer); 2848 bool ownsBuffer = false; 2849 2850 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 2851 if (session > AUDIO_SESSION_OUTPUT_MIX) { 2852 // Only one effect chain can be present in direct output thread and it uses 2853 // the sink buffer as input 2854 if (mType != DIRECT) { 2855 size_t numSamples = mNormalFrameCount * mChannelCount; 2856 buffer = new int16_t[numSamples]; 2857 memset(buffer, 0, numSamples * sizeof(int16_t)); 2858 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 2859 ownsBuffer = true; 2860 } 2861 2862 // Attach all tracks with same session ID to this chain. 2863 for (size_t i = 0; i < mTracks.size(); ++i) { 2864 sp<Track> track = mTracks[i]; 2865 if (session == track->sessionId()) { 2866 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), 2867 buffer); 2868 track->setMainBuffer(buffer); 2869 chain->incTrackCnt(); 2870 } 2871 } 2872 2873 // indicate all active tracks in the chain 2874 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 2875 sp<Track> track = mActiveTracks[i].promote(); 2876 if (track == 0) { 2877 continue; 2878 } 2879 if (session == track->sessionId()) { 2880 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 2881 chain->incActiveTrackCnt(); 2882 } 2883 } 2884 } 2885 chain->setThread(this); 2886 chain->setInBuffer(buffer, ownsBuffer); 2887 chain->setOutBuffer(reinterpret_cast<int16_t*>(mEffectBufferEnabled 2888 ? mEffectBuffer : mSinkBuffer)); 2889 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 2890 // chains list in order to be processed last as it contains output stage effects. 2891 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 2892 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 2893 // after track specific effects and before output stage. 2894 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 2895 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX. 2896 // Effect chain for other sessions are inserted at beginning of effect 2897 // chains list to be processed before output mix effects. Relative order between other 2898 // sessions is not important. 2899 static_assert(AUDIO_SESSION_OUTPUT_MIX == 0 && 2900 AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX, 2901 "audio_session_t constants misdefined"); 2902 size_t size = mEffectChains.size(); 2903 size_t i = 0; 2904 for (i = 0; i < size; i++) { 2905 if (mEffectChains[i]->sessionId() < session) { 2906 break; 2907 } 2908 } 2909 mEffectChains.insertAt(chain, i); 2910 checkSuspendOnAddEffectChain_l(chain); 2911 2912 return NO_ERROR; 2913} 2914 2915size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 2916{ 2917 audio_session_t session = chain->sessionId(); 2918 2919 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 2920 2921 for (size_t i = 0; i < mEffectChains.size(); i++) { 2922 if (chain == mEffectChains[i]) { 2923 mEffectChains.removeAt(i); 2924 // detach all active tracks from the chain 2925 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 2926 sp<Track> track = mActiveTracks[i].promote(); 2927 if (track == 0) { 2928 continue; 2929 } 2930 if (session == track->sessionId()) { 2931 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 2932 chain.get(), session); 2933 chain->decActiveTrackCnt(); 2934 } 2935 } 2936 2937 // detach all tracks with same session ID from this chain 2938 for (size_t i = 0; i < mTracks.size(); ++i) { 2939 sp<Track> track = mTracks[i]; 2940 if (session == track->sessionId()) { 2941 track->setMainBuffer(reinterpret_cast<int16_t*>(mSinkBuffer)); 2942 chain->decTrackCnt(); 2943 } 2944 } 2945 break; 2946 } 2947 } 2948 return mEffectChains.size(); 2949} 2950 2951status_t AudioFlinger::PlaybackThread::attachAuxEffect( 2952 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2953{ 2954 Mutex::Autolock _l(mLock); 2955 return attachAuxEffect_l(track, EffectId); 2956} 2957 2958status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 2959 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2960{ 2961 status_t status = NO_ERROR; 2962 2963 if (EffectId == 0) { 2964 track->setAuxBuffer(0, NULL); 2965 } else { 2966 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 2967 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 2968 if (effect != 0) { 2969 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 2970 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 2971 } else { 2972 status = INVALID_OPERATION; 2973 } 2974 } else { 2975 status = BAD_VALUE; 2976 } 2977 } 2978 return status; 2979} 2980 2981void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 2982{ 2983 for (size_t i = 0; i < mTracks.size(); ++i) { 2984 sp<Track> track = mTracks[i]; 2985 if (track->auxEffectId() == effectId) { 2986 attachAuxEffect_l(track, 0); 2987 } 2988 } 2989} 2990 2991bool AudioFlinger::PlaybackThread::threadLoop() 2992{ 2993 Vector< sp<Track> > tracksToRemove; 2994 2995 mStandbyTimeNs = systemTime(); 2996 nsecs_t lastWriteFinished = -1; // time last server write completed 2997 int64_t lastFramesWritten = -1; // track changes in timestamp server frames written 2998 2999 // MIXER 3000 nsecs_t lastWarning = 0; 3001 3002 // DUPLICATING 3003 // FIXME could this be made local to while loop? 3004 writeFrames = 0; 3005 3006 int lastGeneration = 0; 3007 3008 cacheParameters_l(); 3009 mSleepTimeUs = mIdleSleepTimeUs; 3010 3011 if (mType == MIXER) { 3012 sleepTimeShift = 0; 3013 } 3014 3015 CpuStats cpuStats; 3016 const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid())); 3017 3018 acquireWakeLock(); 3019 3020 // mNBLogWriter->log can only be called while thread mutex mLock is held. 3021 // So if you need to log when mutex is unlocked, set logString to a non-NULL string, 3022 // and then that string will be logged at the next convenient opportunity. 3023 const char *logString = NULL; 3024 3025 checkSilentMode_l(); 3026 3027 while (!exitPending()) 3028 { 3029 cpuStats.sample(myName); 3030 3031 Vector< sp<EffectChain> > effectChains; 3032 3033 { // scope for mLock 3034 3035 Mutex::Autolock _l(mLock); 3036 3037 processConfigEvents_l(); 3038 3039 if (logString != NULL) { 3040 mNBLogWriter->logTimestamp(); 3041 mNBLogWriter->log(logString); 3042 logString = NULL; 3043 } 3044 3045 // Gather the framesReleased counters for all active tracks, 3046 // and associate with the sink frames written out. We need 3047 // this to convert the sink timestamp to the track timestamp. 3048 bool kernelLocationUpdate = false; 3049 if (mNormalSink != 0) { 3050 // Note: The DuplicatingThread may not have a mNormalSink. 3051 // We always fetch the timestamp here because often the downstream 3052 // sink will block while writing. 3053 ExtendedTimestamp timestamp; // use private copy to fetch 3054 (void) mNormalSink->getTimestamp(timestamp); 3055 3056 // We keep track of the last valid kernel position in case we are in underrun 3057 // and the normal mixer period is the same as the fast mixer period, or there 3058 // is some error from the HAL. 3059 if (mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL] >= 0) { 3060 mTimestamp.mPosition[ExtendedTimestamp::LOCATION_KERNEL_LASTKERNELOK] = 3061 mTimestamp.mPosition[ExtendedTimestamp::LOCATION_KERNEL]; 3062 mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL_LASTKERNELOK] = 3063 mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL]; 3064 3065 mTimestamp.mPosition[ExtendedTimestamp::LOCATION_SERVER_LASTKERNELOK] = 3066 mTimestamp.mPosition[ExtendedTimestamp::LOCATION_SERVER]; 3067 mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_SERVER_LASTKERNELOK] = 3068 mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_SERVER]; 3069 } 3070 3071 if (timestamp.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL] >= 0) { 3072 kernelLocationUpdate = true; 3073 } else { 3074 ALOGVV("getTimestamp error - no valid kernel position"); 3075 } 3076 3077 // copy over kernel info 3078 mTimestamp.mPosition[ExtendedTimestamp::LOCATION_KERNEL] = 3079 timestamp.mPosition[ExtendedTimestamp::LOCATION_KERNEL] 3080 + mSuspendedFrames; // add frames discarded when suspended 3081 mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL] = 3082 timestamp.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL]; 3083 } 3084 // mFramesWritten for non-offloaded tracks are contiguous 3085 // even after standby() is called. This is useful for the track frame 3086 // to sink frame mapping. 3087 bool serverLocationUpdate = false; 3088 if (mFramesWritten != lastFramesWritten) { 3089 serverLocationUpdate = true; 3090 lastFramesWritten = mFramesWritten; 3091 } 3092 // Only update timestamps if there is a meaningful change. 3093 // Either the kernel timestamp must be valid or we have written something. 3094 if (kernelLocationUpdate || serverLocationUpdate) { 3095 if (serverLocationUpdate) { 3096 // use the time before we called the HAL write - it is a bit more accurate 3097 // to when the server last read data than the current time here. 3098 // 3099 // If we haven't written anything, mLastWriteTime will be -1 3100 // and we use systemTime(). 3101 mTimestamp.mPosition[ExtendedTimestamp::LOCATION_SERVER] = mFramesWritten; 3102 mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_SERVER] = mLastWriteTime == -1 3103 ? systemTime() : mLastWriteTime; 3104 } 3105 const size_t size = mActiveTracks.size(); 3106 for (size_t i = 0; i < size; ++i) { 3107 sp<Track> t = mActiveTracks[i].promote(); 3108 if (t != 0 && !t->isFastTrack()) { 3109 t->updateTrackFrameInfo( 3110 t->mAudioTrackServerProxy->framesReleased(), 3111 mFramesWritten, 3112 mTimestamp); 3113 } 3114 } 3115 } 3116 3117 saveOutputTracks(); 3118 if (mSignalPending) { 3119 // A signal was raised while we were unlocked 3120 mSignalPending = false; 3121 } else if (waitingAsyncCallback_l()) { 3122 if (exitPending()) { 3123 break; 3124 } 3125 bool released = false; 3126 if (!keepWakeLock()) { 3127 releaseWakeLock_l(); 3128 released = true; 3129 mWakeLockUids.clear(); 3130 mActiveTracksGeneration++; 3131 } 3132 ALOGV("wait async completion"); 3133 mWaitWorkCV.wait(mLock); 3134 ALOGV("async completion/wake"); 3135 if (released) { 3136 acquireWakeLock_l(); 3137 } 3138 mStandbyTimeNs = systemTime() + mStandbyDelayNs; 3139 mSleepTimeUs = 0; 3140 3141 continue; 3142 } 3143 if ((!mActiveTracks.size() && systemTime() > mStandbyTimeNs) || 3144 isSuspended()) { 3145 // put audio hardware into standby after short delay 3146 if (shouldStandby_l()) { 3147 3148 threadLoop_standby(); 3149 3150 mStandby = true; 3151 } 3152 3153 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 3154 // we're about to wait, flush the binder command buffer 3155 IPCThreadState::self()->flushCommands(); 3156 3157 clearOutputTracks(); 3158 3159 if (exitPending()) { 3160 break; 3161 } 3162 3163 releaseWakeLock_l(); 3164 mWakeLockUids.clear(); 3165 mActiveTracksGeneration++; 3166 // wait until we have something to do... 3167 ALOGV("%s going to sleep", myName.string()); 3168 mWaitWorkCV.wait(mLock); 3169 ALOGV("%s waking up", myName.string()); 3170 acquireWakeLock_l(); 3171 3172 mMixerStatus = MIXER_IDLE; 3173 mMixerStatusIgnoringFastTracks = MIXER_IDLE; 3174 mBytesWritten = 0; 3175 mBytesRemaining = 0; 3176 checkSilentMode_l(); 3177 3178 mStandbyTimeNs = systemTime() + mStandbyDelayNs; 3179 mSleepTimeUs = mIdleSleepTimeUs; 3180 if (mType == MIXER) { 3181 sleepTimeShift = 0; 3182 } 3183 3184 continue; 3185 } 3186 } 3187 // mMixerStatusIgnoringFastTracks is also updated internally 3188 mMixerStatus = prepareTracks_l(&tracksToRemove); 3189 3190 // compare with previously applied list 3191 if (lastGeneration != mActiveTracksGeneration) { 3192 // update wakelock 3193 updateWakeLockUids_l(mWakeLockUids); 3194 lastGeneration = mActiveTracksGeneration; 3195 } 3196 3197 // prevent any changes in effect chain list and in each effect chain 3198 // during mixing and effect process as the audio buffers could be deleted 3199 // or modified if an effect is created or deleted 3200 lockEffectChains_l(effectChains); 3201 } // mLock scope ends 3202 3203 if (mBytesRemaining == 0) { 3204 mCurrentWriteLength = 0; 3205 if (mMixerStatus == MIXER_TRACKS_READY) { 3206 // threadLoop_mix() sets mCurrentWriteLength 3207 threadLoop_mix(); 3208 } else if ((mMixerStatus != MIXER_DRAIN_TRACK) 3209 && (mMixerStatus != MIXER_DRAIN_ALL)) { 3210 // threadLoop_sleepTime sets mSleepTimeUs to 0 if data 3211 // must be written to HAL 3212 threadLoop_sleepTime(); 3213 if (mSleepTimeUs == 0) { 3214 mCurrentWriteLength = mSinkBufferSize; 3215 } 3216 } 3217 // Either threadLoop_mix() or threadLoop_sleepTime() should have set 3218 // mMixerBuffer with data if mMixerBufferValid is true and mSleepTimeUs == 0. 3219 // Merge mMixerBuffer data into mEffectBuffer (if any effects are valid) 3220 // or mSinkBuffer (if there are no effects). 3221 // 3222 // This is done pre-effects computation; if effects change to 3223 // support higher precision, this needs to move. 3224 // 3225 // mMixerBufferValid is only set true by MixerThread::prepareTracks_l(). 3226 // TODO use mSleepTimeUs == 0 as an additional condition. 3227 if (mMixerBufferValid) { 3228 void *buffer = mEffectBufferValid ? mEffectBuffer : mSinkBuffer; 3229 audio_format_t format = mEffectBufferValid ? mEffectBufferFormat : mFormat; 3230 3231 // mono blend occurs for mixer threads only (not direct or offloaded) 3232 // and is handled here if we're going directly to the sink. 3233 if (requireMonoBlend() && !mEffectBufferValid) { 3234 mono_blend(mMixerBuffer, mMixerBufferFormat, mChannelCount, mNormalFrameCount, 3235 true /*limit*/); 3236 } 3237 3238 memcpy_by_audio_format(buffer, format, mMixerBuffer, mMixerBufferFormat, 3239 mNormalFrameCount * mChannelCount); 3240 } 3241 3242 mBytesRemaining = mCurrentWriteLength; 3243 if (isSuspended()) { 3244 // Simulate write to HAL when suspended (e.g. BT SCO phone call). 3245 mSleepTimeUs = suspendSleepTimeUs(); // assumes full buffer. 3246 const size_t framesRemaining = mBytesRemaining / mFrameSize; 3247 mBytesWritten += mBytesRemaining; 3248 mFramesWritten += framesRemaining; 3249 mSuspendedFrames += framesRemaining; // to adjust kernel HAL position 3250 mBytesRemaining = 0; 3251 } 3252 3253 // only process effects if we're going to write 3254 if (mSleepTimeUs == 0 && mType != OFFLOAD) { 3255 for (size_t i = 0; i < effectChains.size(); i ++) { 3256 effectChains[i]->process_l(); 3257 } 3258 } 3259 } 3260 // Process effect chains for offloaded thread even if no audio 3261 // was read from audio track: process only updates effect state 3262 // and thus does have to be synchronized with audio writes but may have 3263 // to be called while waiting for async write callback 3264 if (mType == OFFLOAD) { 3265 for (size_t i = 0; i < effectChains.size(); i ++) { 3266 effectChains[i]->process_l(); 3267 } 3268 } 3269 3270 // Only if the Effects buffer is enabled and there is data in the 3271 // Effects buffer (buffer valid), we need to 3272 // copy into the sink buffer. 3273 // TODO use mSleepTimeUs == 0 as an additional condition. 3274 if (mEffectBufferValid) { 3275 //ALOGV("writing effect buffer to sink buffer format %#x", mFormat); 3276 3277 if (requireMonoBlend()) { 3278 mono_blend(mEffectBuffer, mEffectBufferFormat, mChannelCount, mNormalFrameCount, 3279 true /*limit*/); 3280 } 3281 3282 memcpy_by_audio_format(mSinkBuffer, mFormat, mEffectBuffer, mEffectBufferFormat, 3283 mNormalFrameCount * mChannelCount); 3284 } 3285 3286 // enable changes in effect chain 3287 unlockEffectChains(effectChains); 3288 3289 if (!waitingAsyncCallback()) { 3290 // mSleepTimeUs == 0 means we must write to audio hardware 3291 if (mSleepTimeUs == 0) { 3292 ssize_t ret = 0; 3293 // We save lastWriteFinished here, as previousLastWriteFinished, 3294 // for throttling. On thread start, previousLastWriteFinished will be 3295 // set to -1, which properly results in no throttling after the first write. 3296 nsecs_t previousLastWriteFinished = lastWriteFinished; 3297 nsecs_t delta = 0; 3298 if (mBytesRemaining) { 3299 // FIXME rewrite to reduce number of system calls 3300 mLastWriteTime = systemTime(); // also used for dumpsys 3301 ret = threadLoop_write(); 3302 lastWriteFinished = systemTime(); 3303 delta = lastWriteFinished - mLastWriteTime; 3304 if (ret < 0) { 3305 mBytesRemaining = 0; 3306 } else { 3307 mBytesWritten += ret; 3308 mBytesRemaining -= ret; 3309 mFramesWritten += ret / mFrameSize; 3310 } 3311 } else if ((mMixerStatus == MIXER_DRAIN_TRACK) || 3312 (mMixerStatus == MIXER_DRAIN_ALL)) { 3313 threadLoop_drain(); 3314 } 3315 if (mType == MIXER && !mStandby) { 3316 // write blocked detection 3317 if (delta > maxPeriod) { 3318 mNumDelayedWrites++; 3319 if ((lastWriteFinished - lastWarning) > kWarningThrottleNs) { 3320 ATRACE_NAME("underrun"); 3321 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 3322 (unsigned long long) ns2ms(delta), mNumDelayedWrites, this); 3323 lastWarning = lastWriteFinished; 3324 } 3325 } 3326 3327 if (mThreadThrottle 3328 && mMixerStatus == MIXER_TRACKS_READY // we are mixing (active tracks) 3329 && ret > 0) { // we wrote something 3330 // Limit MixerThread data processing to no more than twice the 3331 // expected processing rate. 3332 // 3333 // This helps prevent underruns with NuPlayer and other applications 3334 // which may set up buffers that are close to the minimum size, or use 3335 // deep buffers, and rely on a double-buffering sleep strategy to fill. 3336 // 3337 // The throttle smooths out sudden large data drains from the device, 3338 // e.g. when it comes out of standby, which often causes problems with 3339 // (1) mixer threads without a fast mixer (which has its own warm-up) 3340 // (2) minimum buffer sized tracks (even if the track is full, 3341 // the app won't fill fast enough to handle the sudden draw). 3342 // 3343 // Total time spent in last processing cycle equals time spent in 3344 // 1. threadLoop_write, as well as time spent in 3345 // 2. threadLoop_mix (significant for heavy mixing, especially 3346 // on low tier processors) 3347 3348 // it's OK if deltaMs is an overestimate. 3349 const int32_t deltaMs = 3350 (lastWriteFinished - previousLastWriteFinished) / 1000000; 3351 const int32_t throttleMs = mHalfBufferMs - deltaMs; 3352 if ((signed)mHalfBufferMs >= throttleMs && throttleMs > 0) { 3353 usleep(throttleMs * 1000); 3354 // notify of throttle start on verbose log 3355 ALOGV_IF(mThreadThrottleEndMs == mThreadThrottleTimeMs, 3356 "mixer(%p) throttle begin:" 3357 " ret(%zd) deltaMs(%d) requires sleep %d ms", 3358 this, ret, deltaMs, throttleMs); 3359 mThreadThrottleTimeMs += throttleMs; 3360 // Throttle must be attributed to the previous mixer loop's write time 3361 // to allow back-to-back throttling. 3362 lastWriteFinished += throttleMs * 1000000; 3363 } else { 3364 uint32_t diff = mThreadThrottleTimeMs - mThreadThrottleEndMs; 3365 if (diff > 0) { 3366 // notify of throttle end on debug log 3367 // but prevent spamming for bluetooth 3368 ALOGD_IF(!audio_is_a2dp_out_device(outDevice()), 3369 "mixer(%p) throttle end: throttle time(%u)", this, diff); 3370 mThreadThrottleEndMs = mThreadThrottleTimeMs; 3371 } 3372 } 3373 } 3374 } 3375 3376 } else { 3377 ATRACE_BEGIN("sleep"); 3378 Mutex::Autolock _l(mLock); 3379 if (!mSignalPending && mConfigEvents.isEmpty() && !exitPending()) { 3380 mWaitWorkCV.waitRelative(mLock, microseconds((nsecs_t)mSleepTimeUs)); 3381 } 3382 ATRACE_END(); 3383 } 3384 } 3385 3386 // Finally let go of removed track(s), without the lock held 3387 // since we can't guarantee the destructors won't acquire that 3388 // same lock. This will also mutate and push a new fast mixer state. 3389 threadLoop_removeTracks(tracksToRemove); 3390 tracksToRemove.clear(); 3391 3392 // FIXME I don't understand the need for this here; 3393 // it was in the original code but maybe the 3394 // assignment in saveOutputTracks() makes this unnecessary? 3395 clearOutputTracks(); 3396 3397 // Effect chains will be actually deleted here if they were removed from 3398 // mEffectChains list during mixing or effects processing 3399 effectChains.clear(); 3400 3401 // FIXME Note that the above .clear() is no longer necessary since effectChains 3402 // is now local to this block, but will keep it for now (at least until merge done). 3403 } 3404 3405 threadLoop_exit(); 3406 3407 if (!mStandby) { 3408 threadLoop_standby(); 3409 mStandby = true; 3410 } 3411 3412 releaseWakeLock(); 3413 mWakeLockUids.clear(); 3414 mActiveTracksGeneration++; 3415 3416 ALOGV("Thread %p type %d exiting", this, mType); 3417 return false; 3418} 3419 3420// removeTracks_l() must be called with ThreadBase::mLock held 3421void AudioFlinger::PlaybackThread::removeTracks_l(const Vector< sp<Track> >& tracksToRemove) 3422{ 3423 size_t count = tracksToRemove.size(); 3424 if (count > 0) { 3425 for (size_t i=0 ; i<count ; i++) { 3426 const sp<Track>& track = tracksToRemove.itemAt(i); 3427 mActiveTracks.remove(track); 3428 mWakeLockUids.remove(track->uid()); 3429 mActiveTracksGeneration++; 3430 ALOGV("removeTracks_l removing track on session %d", track->sessionId()); 3431 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 3432 if (chain != 0) { 3433 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), 3434 track->sessionId()); 3435 chain->decActiveTrackCnt(); 3436 } 3437 if (track->isTerminated()) { 3438 removeTrack_l(track); 3439 } 3440 } 3441 } 3442 3443} 3444 3445status_t AudioFlinger::PlaybackThread::getTimestamp_l(AudioTimestamp& timestamp) 3446{ 3447 if (mNormalSink != 0) { 3448 ExtendedTimestamp ets; 3449 status_t status = mNormalSink->getTimestamp(ets); 3450 if (status == NO_ERROR) { 3451 status = ets.getBestTimestamp(×tamp); 3452 } 3453 return status; 3454 } 3455 if ((mType == OFFLOAD || mType == DIRECT) 3456 && mOutput != NULL && mOutput->stream->get_presentation_position) { 3457 uint64_t position64; 3458 int ret = mOutput->getPresentationPosition(&position64, ×tamp.mTime); 3459 if (ret == 0) { 3460 timestamp.mPosition = (uint32_t)position64; 3461 return NO_ERROR; 3462 } 3463 } 3464 return INVALID_OPERATION; 3465} 3466 3467status_t AudioFlinger::MixerThread::createAudioPatch_l(const struct audio_patch *patch, 3468 audio_patch_handle_t *handle) 3469{ 3470 status_t status; 3471 if (property_get_bool("af.patch_park", false /* default_value */)) { 3472 // Park FastMixer to avoid potential DOS issues with writing to the HAL 3473 // or if HAL does not properly lock against access. 3474 AutoPark<FastMixer> park(mFastMixer); 3475 status = PlaybackThread::createAudioPatch_l(patch, handle); 3476 } else { 3477 status = PlaybackThread::createAudioPatch_l(patch, handle); 3478 } 3479 return status; 3480} 3481 3482status_t AudioFlinger::PlaybackThread::createAudioPatch_l(const struct audio_patch *patch, 3483 audio_patch_handle_t *handle) 3484{ 3485 status_t status = NO_ERROR; 3486 3487 // store new device and send to effects 3488 audio_devices_t type = AUDIO_DEVICE_NONE; 3489 for (unsigned int i = 0; i < patch->num_sinks; i++) { 3490 type |= patch->sinks[i].ext.device.type; 3491 } 3492 3493#ifdef ADD_BATTERY_DATA 3494 // when changing the audio output device, call addBatteryData to notify 3495 // the change 3496 if (mOutDevice != type) { 3497 uint32_t params = 0; 3498 // check whether speaker is on 3499 if (type & AUDIO_DEVICE_OUT_SPEAKER) { 3500 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 3501 } 3502 3503 audio_devices_t deviceWithoutSpeaker 3504 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 3505 // check if any other device (except speaker) is on 3506 if (type & deviceWithoutSpeaker) { 3507 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 3508 } 3509 3510 if (params != 0) { 3511 addBatteryData(params); 3512 } 3513 } 3514#endif 3515 3516 for (size_t i = 0; i < mEffectChains.size(); i++) { 3517 mEffectChains[i]->setDevice_l(type); 3518 } 3519 3520 // mPrevOutDevice is the latest device set by createAudioPatch_l(). It is not set when 3521 // the thread is created so that the first patch creation triggers an ioConfigChanged callback 3522 bool configChanged = mPrevOutDevice != type; 3523 mOutDevice = type; 3524 mPatch = *patch; 3525 3526 if (mOutput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 3527 audio_hw_device_t *hwDevice = mOutput->audioHwDev->hwDevice(); 3528 status = hwDevice->create_audio_patch(hwDevice, 3529 patch->num_sources, 3530 patch->sources, 3531 patch->num_sinks, 3532 patch->sinks, 3533 handle); 3534 } else { 3535 char *address; 3536 if (strcmp(patch->sinks[0].ext.device.address, "") != 0) { 3537 //FIXME: we only support address on first sink with HAL version < 3.0 3538 address = audio_device_address_to_parameter( 3539 patch->sinks[0].ext.device.type, 3540 patch->sinks[0].ext.device.address); 3541 } else { 3542 address = (char *)calloc(1, 1); 3543 } 3544 AudioParameter param = AudioParameter(String8(address)); 3545 free(address); 3546 param.addInt(String8(AUDIO_PARAMETER_STREAM_ROUTING), (int)type); 3547 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3548 param.toString().string()); 3549 *handle = AUDIO_PATCH_HANDLE_NONE; 3550 } 3551 if (configChanged) { 3552 mPrevOutDevice = type; 3553 sendIoConfigEvent_l(AUDIO_OUTPUT_CONFIG_CHANGED); 3554 } 3555 return status; 3556} 3557 3558status_t AudioFlinger::MixerThread::releaseAudioPatch_l(const audio_patch_handle_t handle) 3559{ 3560 status_t status; 3561 if (property_get_bool("af.patch_park", false /* default_value */)) { 3562 // Park FastMixer to avoid potential DOS issues with writing to the HAL 3563 // or if HAL does not properly lock against access. 3564 AutoPark<FastMixer> park(mFastMixer); 3565 status = PlaybackThread::releaseAudioPatch_l(handle); 3566 } else { 3567 status = PlaybackThread::releaseAudioPatch_l(handle); 3568 } 3569 return status; 3570} 3571 3572status_t AudioFlinger::PlaybackThread::releaseAudioPatch_l(const audio_patch_handle_t handle) 3573{ 3574 status_t status = NO_ERROR; 3575 3576 mOutDevice = AUDIO_DEVICE_NONE; 3577 3578 if (mOutput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 3579 audio_hw_device_t *hwDevice = mOutput->audioHwDev->hwDevice(); 3580 status = hwDevice->release_audio_patch(hwDevice, handle); 3581 } else { 3582 AudioParameter param; 3583 param.addInt(String8(AUDIO_PARAMETER_STREAM_ROUTING), 0); 3584 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3585 param.toString().string()); 3586 } 3587 return status; 3588} 3589 3590void AudioFlinger::PlaybackThread::addPatchTrack(const sp<PatchTrack>& track) 3591{ 3592 Mutex::Autolock _l(mLock); 3593 mTracks.add(track); 3594} 3595 3596void AudioFlinger::PlaybackThread::deletePatchTrack(const sp<PatchTrack>& track) 3597{ 3598 Mutex::Autolock _l(mLock); 3599 destroyTrack_l(track); 3600} 3601 3602void AudioFlinger::PlaybackThread::getAudioPortConfig(struct audio_port_config *config) 3603{ 3604 ThreadBase::getAudioPortConfig(config); 3605 config->role = AUDIO_PORT_ROLE_SOURCE; 3606 config->ext.mix.hw_module = mOutput->audioHwDev->handle(); 3607 config->ext.mix.usecase.stream = AUDIO_STREAM_DEFAULT; 3608} 3609 3610// ---------------------------------------------------------------------------- 3611 3612AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 3613 audio_io_handle_t id, audio_devices_t device, bool systemReady, type_t type) 3614 : PlaybackThread(audioFlinger, output, id, device, type, systemReady), 3615 // mAudioMixer below 3616 // mFastMixer below 3617 mFastMixerFutex(0), 3618 mMasterMono(false) 3619 // mOutputSink below 3620 // mPipeSink below 3621 // mNormalSink below 3622{ 3623 ALOGV("MixerThread() id=%d device=%#x type=%d", id, device, type); 3624 ALOGV("mSampleRate=%u, mChannelMask=%#x, mChannelCount=%u, mFormat=%d, mFrameSize=%zu, " 3625 "mFrameCount=%zu, mNormalFrameCount=%zu", 3626 mSampleRate, mChannelMask, mChannelCount, mFormat, mFrameSize, mFrameCount, 3627 mNormalFrameCount); 3628 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 3629 3630 if (type == DUPLICATING) { 3631 // The Duplicating thread uses the AudioMixer and delivers data to OutputTracks 3632 // (downstream MixerThreads) in DuplicatingThread::threadLoop_write(). 3633 // Do not create or use mFastMixer, mOutputSink, mPipeSink, or mNormalSink. 3634 return; 3635 } 3636 // create an NBAIO sink for the HAL output stream, and negotiate 3637 mOutputSink = new AudioStreamOutSink(output->stream); 3638 size_t numCounterOffers = 0; 3639 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount, mFormat)}; 3640#if !LOG_NDEBUG 3641 ssize_t index = 3642#else 3643 (void) 3644#endif 3645 mOutputSink->negotiate(offers, 1, NULL, numCounterOffers); 3646 ALOG_ASSERT(index == 0); 3647 3648 // initialize fast mixer depending on configuration 3649 bool initFastMixer; 3650 switch (kUseFastMixer) { 3651 case FastMixer_Never: 3652 initFastMixer = false; 3653 break; 3654 case FastMixer_Always: 3655 initFastMixer = true; 3656 break; 3657 case FastMixer_Static: 3658 case FastMixer_Dynamic: 3659 initFastMixer = mFrameCount < mNormalFrameCount; 3660 break; 3661 } 3662 if (initFastMixer) { 3663 audio_format_t fastMixerFormat; 3664 if (mMixerBufferEnabled && mEffectBufferEnabled) { 3665 fastMixerFormat = AUDIO_FORMAT_PCM_FLOAT; 3666 } else { 3667 fastMixerFormat = AUDIO_FORMAT_PCM_16_BIT; 3668 } 3669 if (mFormat != fastMixerFormat) { 3670 // change our Sink format to accept our intermediate precision 3671 mFormat = fastMixerFormat; 3672 free(mSinkBuffer); 3673 mFrameSize = mChannelCount * audio_bytes_per_sample(mFormat); 3674 const size_t sinkBufferSize = mNormalFrameCount * mFrameSize; 3675 (void)posix_memalign(&mSinkBuffer, 32, sinkBufferSize); 3676 } 3677 3678 // create a MonoPipe to connect our submix to FastMixer 3679 NBAIO_Format format = mOutputSink->format(); 3680#ifdef TEE_SINK 3681 NBAIO_Format origformat = format; 3682#endif 3683 // adjust format to match that of the Fast Mixer 3684 ALOGV("format changed from %d to %d", format.mFormat, fastMixerFormat); 3685 format.mFormat = fastMixerFormat; 3686 format.mFrameSize = audio_bytes_per_sample(format.mFormat) * format.mChannelCount; 3687 3688 // This pipe depth compensates for scheduling latency of the normal mixer thread. 3689 // When it wakes up after a maximum latency, it runs a few cycles quickly before 3690 // finally blocking. Note the pipe implementation rounds up the request to a power of 2. 3691 MonoPipe *monoPipe = new MonoPipe(mNormalFrameCount * 4, format, true /*writeCanBlock*/); 3692 const NBAIO_Format offers[1] = {format}; 3693 size_t numCounterOffers = 0; 3694#if !LOG_NDEBUG || defined(TEE_SINK) 3695 ssize_t index = 3696#else 3697 (void) 3698#endif 3699 monoPipe->negotiate(offers, 1, NULL, numCounterOffers); 3700 ALOG_ASSERT(index == 0); 3701 monoPipe->setAvgFrames((mScreenState & 1) ? 3702 (monoPipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 3703 mPipeSink = monoPipe; 3704 3705#ifdef TEE_SINK 3706 if (mTeeSinkOutputEnabled) { 3707 // create a Pipe to archive a copy of FastMixer's output for dumpsys 3708 Pipe *teeSink = new Pipe(mTeeSinkOutputFrames, origformat); 3709 const NBAIO_Format offers2[1] = {origformat}; 3710 numCounterOffers = 0; 3711 index = teeSink->negotiate(offers2, 1, NULL, numCounterOffers); 3712 ALOG_ASSERT(index == 0); 3713 mTeeSink = teeSink; 3714 PipeReader *teeSource = new PipeReader(*teeSink); 3715 numCounterOffers = 0; 3716 index = teeSource->negotiate(offers2, 1, NULL, numCounterOffers); 3717 ALOG_ASSERT(index == 0); 3718 mTeeSource = teeSource; 3719 } 3720#endif 3721 3722 // create fast mixer and configure it initially with just one fast track for our submix 3723 mFastMixer = new FastMixer(); 3724 FastMixerStateQueue *sq = mFastMixer->sq(); 3725#ifdef STATE_QUEUE_DUMP 3726 sq->setObserverDump(&mStateQueueObserverDump); 3727 sq->setMutatorDump(&mStateQueueMutatorDump); 3728#endif 3729 FastMixerState *state = sq->begin(); 3730 FastTrack *fastTrack = &state->mFastTracks[0]; 3731 // wrap the source side of the MonoPipe to make it an AudioBufferProvider 3732 fastTrack->mBufferProvider = new SourceAudioBufferProvider(new MonoPipeReader(monoPipe)); 3733 fastTrack->mVolumeProvider = NULL; 3734 fastTrack->mChannelMask = mChannelMask; // mPipeSink channel mask for audio to FastMixer 3735 fastTrack->mFormat = mFormat; // mPipeSink format for audio to FastMixer 3736 fastTrack->mGeneration++; 3737 state->mFastTracksGen++; 3738 state->mTrackMask = 1; 3739 // fast mixer will use the HAL output sink 3740 state->mOutputSink = mOutputSink.get(); 3741 state->mOutputSinkGen++; 3742 state->mFrameCount = mFrameCount; 3743 state->mCommand = FastMixerState::COLD_IDLE; 3744 // already done in constructor initialization list 3745 //mFastMixerFutex = 0; 3746 state->mColdFutexAddr = &mFastMixerFutex; 3747 state->mColdGen++; 3748 state->mDumpState = &mFastMixerDumpState; 3749#ifdef TEE_SINK 3750 state->mTeeSink = mTeeSink.get(); 3751#endif 3752 mFastMixerNBLogWriter = audioFlinger->newWriter_l(kFastMixerLogSize, "FastMixer"); 3753 state->mNBLogWriter = mFastMixerNBLogWriter.get(); 3754 sq->end(); 3755 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3756 3757 // start the fast mixer 3758 mFastMixer->run("FastMixer", PRIORITY_URGENT_AUDIO); 3759 pid_t tid = mFastMixer->getTid(); 3760 sendPrioConfigEvent(getpid_cached, tid, kPriorityFastMixer); 3761 3762#ifdef AUDIO_WATCHDOG 3763 // create and start the watchdog 3764 mAudioWatchdog = new AudioWatchdog(); 3765 mAudioWatchdog->setDump(&mAudioWatchdogDump); 3766 mAudioWatchdog->run("AudioWatchdog", PRIORITY_URGENT_AUDIO); 3767 tid = mAudioWatchdog->getTid(); 3768 sendPrioConfigEvent(getpid_cached, tid, kPriorityFastMixer); 3769#endif 3770 3771 } 3772 3773 switch (kUseFastMixer) { 3774 case FastMixer_Never: 3775 case FastMixer_Dynamic: 3776 mNormalSink = mOutputSink; 3777 break; 3778 case FastMixer_Always: 3779 mNormalSink = mPipeSink; 3780 break; 3781 case FastMixer_Static: 3782 mNormalSink = initFastMixer ? mPipeSink : mOutputSink; 3783 break; 3784 } 3785} 3786 3787AudioFlinger::MixerThread::~MixerThread() 3788{ 3789 if (mFastMixer != 0) { 3790 FastMixerStateQueue *sq = mFastMixer->sq(); 3791 FastMixerState *state = sq->begin(); 3792 if (state->mCommand == FastMixerState::COLD_IDLE) { 3793 int32_t old = android_atomic_inc(&mFastMixerFutex); 3794 if (old == -1) { 3795 (void) syscall(__NR_futex, &mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 3796 } 3797 } 3798 state->mCommand = FastMixerState::EXIT; 3799 sq->end(); 3800 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3801 mFastMixer->join(); 3802 // Though the fast mixer thread has exited, it's state queue is still valid. 3803 // We'll use that extract the final state which contains one remaining fast track 3804 // corresponding to our sub-mix. 3805 state = sq->begin(); 3806 ALOG_ASSERT(state->mTrackMask == 1); 3807 FastTrack *fastTrack = &state->mFastTracks[0]; 3808 ALOG_ASSERT(fastTrack->mBufferProvider != NULL); 3809 delete fastTrack->mBufferProvider; 3810 sq->end(false /*didModify*/); 3811 mFastMixer.clear(); 3812#ifdef AUDIO_WATCHDOG 3813 if (mAudioWatchdog != 0) { 3814 mAudioWatchdog->requestExit(); 3815 mAudioWatchdog->requestExitAndWait(); 3816 mAudioWatchdog.clear(); 3817 } 3818#endif 3819 } 3820 mAudioFlinger->unregisterWriter(mFastMixerNBLogWriter); 3821 delete mAudioMixer; 3822} 3823 3824 3825uint32_t AudioFlinger::MixerThread::correctLatency_l(uint32_t latency) const 3826{ 3827 if (mFastMixer != 0) { 3828 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 3829 latency += (pipe->getAvgFrames() * 1000) / mSampleRate; 3830 } 3831 return latency; 3832} 3833 3834 3835void AudioFlinger::MixerThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) 3836{ 3837 PlaybackThread::threadLoop_removeTracks(tracksToRemove); 3838} 3839 3840ssize_t AudioFlinger::MixerThread::threadLoop_write() 3841{ 3842 // FIXME we should only do one push per cycle; confirm this is true 3843 // Start the fast mixer if it's not already running 3844 if (mFastMixer != 0) { 3845 FastMixerStateQueue *sq = mFastMixer->sq(); 3846 FastMixerState *state = sq->begin(); 3847 if (state->mCommand != FastMixerState::MIX_WRITE && 3848 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)) { 3849 if (state->mCommand == FastMixerState::COLD_IDLE) { 3850 3851 // FIXME workaround for first HAL write being CPU bound on some devices 3852 ATRACE_BEGIN("write"); 3853 mOutput->write((char *)mSinkBuffer, 0); 3854 ATRACE_END(); 3855 3856 int32_t old = android_atomic_inc(&mFastMixerFutex); 3857 if (old == -1) { 3858 (void) syscall(__NR_futex, &mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 3859 } 3860#ifdef AUDIO_WATCHDOG 3861 if (mAudioWatchdog != 0) { 3862 mAudioWatchdog->resume(); 3863 } 3864#endif 3865 } 3866 state->mCommand = FastMixerState::MIX_WRITE; 3867#ifdef FAST_THREAD_STATISTICS 3868 mFastMixerDumpState.increaseSamplingN(mAudioFlinger->isLowRamDevice() ? 3869 FastThreadDumpState::kSamplingNforLowRamDevice : FastThreadDumpState::kSamplingN); 3870#endif 3871 sq->end(); 3872 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3873 if (kUseFastMixer == FastMixer_Dynamic) { 3874 mNormalSink = mPipeSink; 3875 } 3876 } else { 3877 sq->end(false /*didModify*/); 3878 } 3879 } 3880 return PlaybackThread::threadLoop_write(); 3881} 3882 3883void AudioFlinger::MixerThread::threadLoop_standby() 3884{ 3885 // Idle the fast mixer if it's currently running 3886 if (mFastMixer != 0) { 3887 FastMixerStateQueue *sq = mFastMixer->sq(); 3888 FastMixerState *state = sq->begin(); 3889 if (!(state->mCommand & FastMixerState::IDLE)) { 3890 state->mCommand = FastMixerState::COLD_IDLE; 3891 state->mColdFutexAddr = &mFastMixerFutex; 3892 state->mColdGen++; 3893 mFastMixerFutex = 0; 3894 sq->end(); 3895 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 3896 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3897 if (kUseFastMixer == FastMixer_Dynamic) { 3898 mNormalSink = mOutputSink; 3899 } 3900#ifdef AUDIO_WATCHDOG 3901 if (mAudioWatchdog != 0) { 3902 mAudioWatchdog->pause(); 3903 } 3904#endif 3905 } else { 3906 sq->end(false /*didModify*/); 3907 } 3908 } 3909 PlaybackThread::threadLoop_standby(); 3910} 3911 3912bool AudioFlinger::PlaybackThread::waitingAsyncCallback_l() 3913{ 3914 return false; 3915} 3916 3917bool AudioFlinger::PlaybackThread::shouldStandby_l() 3918{ 3919 return !mStandby; 3920} 3921 3922bool AudioFlinger::PlaybackThread::waitingAsyncCallback() 3923{ 3924 Mutex::Autolock _l(mLock); 3925 return waitingAsyncCallback_l(); 3926} 3927 3928// shared by MIXER and DIRECT, overridden by DUPLICATING 3929void AudioFlinger::PlaybackThread::threadLoop_standby() 3930{ 3931 ALOGV("Audio hardware entering standby, mixer %p, suspend count %d", this, mSuspended); 3932 mOutput->standby(); 3933 if (mUseAsyncWrite != 0) { 3934 // discard any pending drain or write ack by incrementing sequence 3935 mWriteAckSequence = (mWriteAckSequence + 2) & ~1; 3936 mDrainSequence = (mDrainSequence + 2) & ~1; 3937 ALOG_ASSERT(mCallbackThread != 0); 3938 mCallbackThread->setWriteBlocked(mWriteAckSequence); 3939 mCallbackThread->setDraining(mDrainSequence); 3940 } 3941 mHwPaused = false; 3942} 3943 3944void AudioFlinger::PlaybackThread::onAddNewTrack_l() 3945{ 3946 ALOGV("signal playback thread"); 3947 broadcast_l(); 3948} 3949 3950void AudioFlinger::PlaybackThread::onAsyncError() 3951{ 3952 for (int i = AUDIO_STREAM_SYSTEM; i < (int)AUDIO_STREAM_CNT; i++) { 3953 invalidateTracks((audio_stream_type_t)i); 3954 } 3955} 3956 3957void AudioFlinger::MixerThread::threadLoop_mix() 3958{ 3959 // mix buffers... 3960 mAudioMixer->process(); 3961 mCurrentWriteLength = mSinkBufferSize; 3962 // increase sleep time progressively when application underrun condition clears. 3963 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 3964 // that a steady state of alternating ready/not ready conditions keeps the sleep time 3965 // such that we would underrun the audio HAL. 3966 if ((mSleepTimeUs == 0) && (sleepTimeShift > 0)) { 3967 sleepTimeShift--; 3968 } 3969 mSleepTimeUs = 0; 3970 mStandbyTimeNs = systemTime() + mStandbyDelayNs; 3971 //TODO: delay standby when effects have a tail 3972 3973} 3974 3975void AudioFlinger::MixerThread::threadLoop_sleepTime() 3976{ 3977 // If no tracks are ready, sleep once for the duration of an output 3978 // buffer size, then write 0s to the output 3979 if (mSleepTimeUs == 0) { 3980 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3981 mSleepTimeUs = mActiveSleepTimeUs >> sleepTimeShift; 3982 if (mSleepTimeUs < kMinThreadSleepTimeUs) { 3983 mSleepTimeUs = kMinThreadSleepTimeUs; 3984 } 3985 // reduce sleep time in case of consecutive application underruns to avoid 3986 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 3987 // duration we would end up writing less data than needed by the audio HAL if 3988 // the condition persists. 3989 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 3990 sleepTimeShift++; 3991 } 3992 } else { 3993 mSleepTimeUs = mIdleSleepTimeUs; 3994 } 3995 } else if (mBytesWritten != 0 || (mMixerStatus == MIXER_TRACKS_ENABLED)) { 3996 // clear out mMixerBuffer or mSinkBuffer, to ensure buffers are cleared 3997 // before effects processing or output. 3998 if (mMixerBufferValid) { 3999 memset(mMixerBuffer, 0, mMixerBufferSize); 4000 } else { 4001 memset(mSinkBuffer, 0, mSinkBufferSize); 4002 } 4003 mSleepTimeUs = 0; 4004 ALOGV_IF(mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED), 4005 "anticipated start"); 4006 } 4007 // TODO add standby time extension fct of effect tail 4008} 4009 4010// prepareTracks_l() must be called with ThreadBase::mLock held 4011AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 4012 Vector< sp<Track> > *tracksToRemove) 4013{ 4014 4015 mixer_state mixerStatus = MIXER_IDLE; 4016 // find out which tracks need to be processed 4017 size_t count = mActiveTracks.size(); 4018 size_t mixedTracks = 0; 4019 size_t tracksWithEffect = 0; 4020 // counts only _active_ fast tracks 4021 size_t fastTracks = 0; 4022 uint32_t resetMask = 0; // bit mask of fast tracks that need to be reset 4023 4024 float masterVolume = mMasterVolume; 4025 bool masterMute = mMasterMute; 4026 4027 if (masterMute) { 4028 masterVolume = 0; 4029 } 4030 // Delegate master volume control to effect in output mix effect chain if needed 4031 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 4032 if (chain != 0) { 4033 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 4034 chain->setVolume_l(&v, &v); 4035 masterVolume = (float)((v + (1 << 23)) >> 24); 4036 chain.clear(); 4037 } 4038 4039 // prepare a new state to push 4040 FastMixerStateQueue *sq = NULL; 4041 FastMixerState *state = NULL; 4042 bool didModify = false; 4043 FastMixerStateQueue::block_t block = FastMixerStateQueue::BLOCK_UNTIL_PUSHED; 4044 if (mFastMixer != 0) { 4045 sq = mFastMixer->sq(); 4046 state = sq->begin(); 4047 } 4048 4049 mMixerBufferValid = false; // mMixerBuffer has no valid data until appropriate tracks found. 4050 mEffectBufferValid = false; // mEffectBuffer has no valid data until tracks found. 4051 4052 for (size_t i=0 ; i<count ; i++) { 4053 const sp<Track> t = mActiveTracks[i].promote(); 4054 if (t == 0) { 4055 continue; 4056 } 4057 4058 // this const just means the local variable doesn't change 4059 Track* const track = t.get(); 4060 4061 // process fast tracks 4062 if (track->isFastTrack()) { 4063 4064 // It's theoretically possible (though unlikely) for a fast track to be created 4065 // and then removed within the same normal mix cycle. This is not a problem, as 4066 // the track never becomes active so it's fast mixer slot is never touched. 4067 // The converse, of removing an (active) track and then creating a new track 4068 // at the identical fast mixer slot within the same normal mix cycle, 4069 // is impossible because the slot isn't marked available until the end of each cycle. 4070 int j = track->mFastIndex; 4071 ALOG_ASSERT(0 < j && j < (int)FastMixerState::sMaxFastTracks); 4072 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << j))); 4073 FastTrack *fastTrack = &state->mFastTracks[j]; 4074 4075 // Determine whether the track is currently in underrun condition, 4076 // and whether it had a recent underrun. 4077 FastTrackDump *ftDump = &mFastMixerDumpState.mTracks[j]; 4078 FastTrackUnderruns underruns = ftDump->mUnderruns; 4079 uint32_t recentFull = (underruns.mBitFields.mFull - 4080 track->mObservedUnderruns.mBitFields.mFull) & UNDERRUN_MASK; 4081 uint32_t recentPartial = (underruns.mBitFields.mPartial - 4082 track->mObservedUnderruns.mBitFields.mPartial) & UNDERRUN_MASK; 4083 uint32_t recentEmpty = (underruns.mBitFields.mEmpty - 4084 track->mObservedUnderruns.mBitFields.mEmpty) & UNDERRUN_MASK; 4085 uint32_t recentUnderruns = recentPartial + recentEmpty; 4086 track->mObservedUnderruns = underruns; 4087 // don't count underruns that occur while stopping or pausing 4088 // or stopped which can occur when flush() is called while active 4089 if (!(track->isStopping() || track->isPausing() || track->isStopped()) && 4090 recentUnderruns > 0) { 4091 // FIXME fast mixer will pull & mix partial buffers, but we count as a full underrun 4092 track->mAudioTrackServerProxy->tallyUnderrunFrames(recentUnderruns * mFrameCount); 4093 } else { 4094 track->mAudioTrackServerProxy->tallyUnderrunFrames(0); 4095 } 4096 4097 // This is similar to the state machine for normal tracks, 4098 // with a few modifications for fast tracks. 4099 bool isActive = true; 4100 switch (track->mState) { 4101 case TrackBase::STOPPING_1: 4102 // track stays active in STOPPING_1 state until first underrun 4103 if (recentUnderruns > 0 || track->isTerminated()) { 4104 track->mState = TrackBase::STOPPING_2; 4105 } 4106 break; 4107 case TrackBase::PAUSING: 4108 // ramp down is not yet implemented 4109 track->setPaused(); 4110 break; 4111 case TrackBase::RESUMING: 4112 // ramp up is not yet implemented 4113 track->mState = TrackBase::ACTIVE; 4114 break; 4115 case TrackBase::ACTIVE: 4116 if (recentFull > 0 || recentPartial > 0) { 4117 // track has provided at least some frames recently: reset retry count 4118 track->mRetryCount = kMaxTrackRetries; 4119 } 4120 if (recentUnderruns == 0) { 4121 // no recent underruns: stay active 4122 break; 4123 } 4124 // there has recently been an underrun of some kind 4125 if (track->sharedBuffer() == 0) { 4126 // were any of the recent underruns "empty" (no frames available)? 4127 if (recentEmpty == 0) { 4128 // no, then ignore the partial underruns as they are allowed indefinitely 4129 break; 4130 } 4131 // there has recently been an "empty" underrun: decrement the retry counter 4132 if (--(track->mRetryCount) > 0) { 4133 break; 4134 } 4135 // indicate to client process that the track was disabled because of underrun; 4136 // it will then automatically call start() when data is available 4137 track->disable(); 4138 // remove from active list, but state remains ACTIVE [confusing but true] 4139 isActive = false; 4140 break; 4141 } 4142 // fall through 4143 case TrackBase::STOPPING_2: 4144 case TrackBase::PAUSED: 4145 case TrackBase::STOPPED: 4146 case TrackBase::FLUSHED: // flush() while active 4147 // Check for presentation complete if track is inactive 4148 // We have consumed all the buffers of this track. 4149 // This would be incomplete if we auto-paused on underrun 4150 { 4151 size_t audioHALFrames = 4152 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 4153 int64_t framesWritten = mBytesWritten / mFrameSize; 4154 if (!(mStandby || track->presentationComplete(framesWritten, audioHALFrames))) { 4155 // track stays in active list until presentation is complete 4156 break; 4157 } 4158 } 4159 if (track->isStopping_2()) { 4160 track->mState = TrackBase::STOPPED; 4161 } 4162 if (track->isStopped()) { 4163 // Can't reset directly, as fast mixer is still polling this track 4164 // track->reset(); 4165 // So instead mark this track as needing to be reset after push with ack 4166 resetMask |= 1 << i; 4167 } 4168 isActive = false; 4169 break; 4170 case TrackBase::IDLE: 4171 default: 4172 LOG_ALWAYS_FATAL("unexpected track state %d", track->mState); 4173 } 4174 4175 if (isActive) { 4176 // was it previously inactive? 4177 if (!(state->mTrackMask & (1 << j))) { 4178 ExtendedAudioBufferProvider *eabp = track; 4179 VolumeProvider *vp = track; 4180 fastTrack->mBufferProvider = eabp; 4181 fastTrack->mVolumeProvider = vp; 4182 fastTrack->mChannelMask = track->mChannelMask; 4183 fastTrack->mFormat = track->mFormat; 4184 fastTrack->mGeneration++; 4185 state->mTrackMask |= 1 << j; 4186 didModify = true; 4187 // no acknowledgement required for newly active tracks 4188 } 4189 // cache the combined master volume and stream type volume for fast mixer; this 4190 // lacks any synchronization or barrier so VolumeProvider may read a stale value 4191 track->mCachedVolume = masterVolume * mStreamTypes[track->streamType()].volume; 4192 ++fastTracks; 4193 } else { 4194 // was it previously active? 4195 if (state->mTrackMask & (1 << j)) { 4196 fastTrack->mBufferProvider = NULL; 4197 fastTrack->mGeneration++; 4198 state->mTrackMask &= ~(1 << j); 4199 didModify = true; 4200 // If any fast tracks were removed, we must wait for acknowledgement 4201 // because we're about to decrement the last sp<> on those tracks. 4202 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 4203 } else { 4204 LOG_ALWAYS_FATAL("fast track %d should have been active; " 4205 "mState=%d, mTrackMask=%#x, recentUnderruns=%u, isShared=%d", 4206 j, track->mState, state->mTrackMask, recentUnderruns, 4207 track->sharedBuffer() != 0); 4208 } 4209 tracksToRemove->add(track); 4210 // Avoids a misleading display in dumpsys 4211 track->mObservedUnderruns.mBitFields.mMostRecent = UNDERRUN_FULL; 4212 } 4213 continue; 4214 } 4215 4216 { // local variable scope to avoid goto warning 4217 4218 audio_track_cblk_t* cblk = track->cblk(); 4219 4220 // The first time a track is added we wait 4221 // for all its buffers to be filled before processing it 4222 int name = track->name(); 4223 // make sure that we have enough frames to mix one full buffer. 4224 // enforce this condition only once to enable draining the buffer in case the client 4225 // app does not call stop() and relies on underrun to stop: 4226 // hence the test on (mMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 4227 // during last round 4228 size_t desiredFrames; 4229 const uint32_t sampleRate = track->mAudioTrackServerProxy->getSampleRate(); 4230 AudioPlaybackRate playbackRate = track->mAudioTrackServerProxy->getPlaybackRate(); 4231 4232 desiredFrames = sourceFramesNeededWithTimestretch( 4233 sampleRate, mNormalFrameCount, mSampleRate, playbackRate.mSpeed); 4234 // TODO: ONLY USED FOR LEGACY RESAMPLERS, remove when they are removed. 4235 // add frames already consumed but not yet released by the resampler 4236 // because mAudioTrackServerProxy->framesReady() will include these frames 4237 desiredFrames += mAudioMixer->getUnreleasedFrames(track->name()); 4238 4239 uint32_t minFrames = 1; 4240 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() && 4241 (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) { 4242 minFrames = desiredFrames; 4243 } 4244 4245 size_t framesReady = track->framesReady(); 4246 if (ATRACE_ENABLED()) { 4247 // I wish we had formatted trace names 4248 char traceName[16]; 4249 strcpy(traceName, "nRdy"); 4250 int name = track->name(); 4251 if (AudioMixer::TRACK0 <= name && 4252 name < (int) (AudioMixer::TRACK0 + AudioMixer::MAX_NUM_TRACKS)) { 4253 name -= AudioMixer::TRACK0; 4254 traceName[4] = (name / 10) + '0'; 4255 traceName[5] = (name % 10) + '0'; 4256 } else { 4257 traceName[4] = '?'; 4258 traceName[5] = '?'; 4259 } 4260 traceName[6] = '\0'; 4261 ATRACE_INT(traceName, framesReady); 4262 } 4263 if ((framesReady >= minFrames) && track->isReady() && 4264 !track->isPaused() && !track->isTerminated()) 4265 { 4266 ALOGVV("track %d s=%08x [OK] on thread %p", name, cblk->mServer, this); 4267 4268 mixedTracks++; 4269 4270 // track->mainBuffer() != mSinkBuffer or mMixerBuffer means 4271 // there is an effect chain connected to the track 4272 chain.clear(); 4273 if (track->mainBuffer() != mSinkBuffer && 4274 track->mainBuffer() != mMixerBuffer) { 4275 if (mEffectBufferEnabled) { 4276 mEffectBufferValid = true; // Later can set directly. 4277 } 4278 chain = getEffectChain_l(track->sessionId()); 4279 // Delegate volume control to effect in track effect chain if needed 4280 if (chain != 0) { 4281 tracksWithEffect++; 4282 } else { 4283 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on " 4284 "session %d", 4285 name, track->sessionId()); 4286 } 4287 } 4288 4289 4290 int param = AudioMixer::VOLUME; 4291 if (track->mFillingUpStatus == Track::FS_FILLED) { 4292 // no ramp for the first volume setting 4293 track->mFillingUpStatus = Track::FS_ACTIVE; 4294 if (track->mState == TrackBase::RESUMING) { 4295 track->mState = TrackBase::ACTIVE; 4296 param = AudioMixer::RAMP_VOLUME; 4297 } 4298 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 4299 // FIXME should not make a decision based on mServer 4300 } else if (cblk->mServer != 0) { 4301 // If the track is stopped before the first frame was mixed, 4302 // do not apply ramp 4303 param = AudioMixer::RAMP_VOLUME; 4304 } 4305 4306 // compute volume for this track 4307 uint32_t vl, vr; // in U8.24 integer format 4308 float vlf, vrf, vaf; // in [0.0, 1.0] float format 4309 if (track->isPausing() || mStreamTypes[track->streamType()].mute) { 4310 vl = vr = 0; 4311 vlf = vrf = vaf = 0.; 4312 if (track->isPausing()) { 4313 track->setPaused(); 4314 } 4315 } else { 4316 4317 // read original volumes with volume control 4318 float typeVolume = mStreamTypes[track->streamType()].volume; 4319 float v = masterVolume * typeVolume; 4320 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 4321 gain_minifloat_packed_t vlr = proxy->getVolumeLR(); 4322 vlf = float_from_gain(gain_minifloat_unpack_left(vlr)); 4323 vrf = float_from_gain(gain_minifloat_unpack_right(vlr)); 4324 // track volumes come from shared memory, so can't be trusted and must be clamped 4325 if (vlf > GAIN_FLOAT_UNITY) { 4326 ALOGV("Track left volume out of range: %.3g", vlf); 4327 vlf = GAIN_FLOAT_UNITY; 4328 } 4329 if (vrf > GAIN_FLOAT_UNITY) { 4330 ALOGV("Track right volume out of range: %.3g", vrf); 4331 vrf = GAIN_FLOAT_UNITY; 4332 } 4333 // now apply the master volume and stream type volume 4334 vlf *= v; 4335 vrf *= v; 4336 // assuming master volume and stream type volume each go up to 1.0, 4337 // then derive vl and vr as U8.24 versions for the effect chain 4338 const float scaleto8_24 = MAX_GAIN_INT * MAX_GAIN_INT; 4339 vl = (uint32_t) (scaleto8_24 * vlf); 4340 vr = (uint32_t) (scaleto8_24 * vrf); 4341 // vl and vr are now in U8.24 format 4342 uint16_t sendLevel = proxy->getSendLevel_U4_12(); 4343 // send level comes from shared memory and so may be corrupt 4344 if (sendLevel > MAX_GAIN_INT) { 4345 ALOGV("Track send level out of range: %04X", sendLevel); 4346 sendLevel = MAX_GAIN_INT; 4347 } 4348 // vaf is represented as [0.0, 1.0] float by rescaling sendLevel 4349 vaf = v * sendLevel * (1. / MAX_GAIN_INT); 4350 } 4351 4352 // Delegate volume control to effect in track effect chain if needed 4353 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 4354 // Do not ramp volume if volume is controlled by effect 4355 param = AudioMixer::VOLUME; 4356 // Update remaining floating point volume levels 4357 vlf = (float)vl / (1 << 24); 4358 vrf = (float)vr / (1 << 24); 4359 track->mHasVolumeController = true; 4360 } else { 4361 // force no volume ramp when volume controller was just disabled or removed 4362 // from effect chain to avoid volume spike 4363 if (track->mHasVolumeController) { 4364 param = AudioMixer::VOLUME; 4365 } 4366 track->mHasVolumeController = false; 4367 } 4368 4369 // XXX: these things DON'T need to be done each time 4370 mAudioMixer->setBufferProvider(name, track); 4371 mAudioMixer->enable(name); 4372 4373 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, &vlf); 4374 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, &vrf); 4375 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, &vaf); 4376 mAudioMixer->setParameter( 4377 name, 4378 AudioMixer::TRACK, 4379 AudioMixer::FORMAT, (void *)track->format()); 4380 mAudioMixer->setParameter( 4381 name, 4382 AudioMixer::TRACK, 4383 AudioMixer::CHANNEL_MASK, (void *)(uintptr_t)track->channelMask()); 4384 mAudioMixer->setParameter( 4385 name, 4386 AudioMixer::TRACK, 4387 AudioMixer::MIXER_CHANNEL_MASK, (void *)(uintptr_t)mChannelMask); 4388 // limit track sample rate to 2 x output sample rate, which changes at re-configuration 4389 uint32_t maxSampleRate = mSampleRate * AUDIO_RESAMPLER_DOWN_RATIO_MAX; 4390 uint32_t reqSampleRate = track->mAudioTrackServerProxy->getSampleRate(); 4391 if (reqSampleRate == 0) { 4392 reqSampleRate = mSampleRate; 4393 } else if (reqSampleRate > maxSampleRate) { 4394 reqSampleRate = maxSampleRate; 4395 } 4396 mAudioMixer->setParameter( 4397 name, 4398 AudioMixer::RESAMPLE, 4399 AudioMixer::SAMPLE_RATE, 4400 (void *)(uintptr_t)reqSampleRate); 4401 4402 AudioPlaybackRate playbackRate = track->mAudioTrackServerProxy->getPlaybackRate(); 4403 mAudioMixer->setParameter( 4404 name, 4405 AudioMixer::TIMESTRETCH, 4406 AudioMixer::PLAYBACK_RATE, 4407 &playbackRate); 4408 4409 /* 4410 * Select the appropriate output buffer for the track. 4411 * 4412 * Tracks with effects go into their own effects chain buffer 4413 * and from there into either mEffectBuffer or mSinkBuffer. 4414 * 4415 * Other tracks can use mMixerBuffer for higher precision 4416 * channel accumulation. If this buffer is enabled 4417 * (mMixerBufferEnabled true), then selected tracks will accumulate 4418 * into it. 4419 * 4420 */ 4421 if (mMixerBufferEnabled 4422 && (track->mainBuffer() == mSinkBuffer 4423 || track->mainBuffer() == mMixerBuffer)) { 4424 mAudioMixer->setParameter( 4425 name, 4426 AudioMixer::TRACK, 4427 AudioMixer::MIXER_FORMAT, (void *)mMixerBufferFormat); 4428 mAudioMixer->setParameter( 4429 name, 4430 AudioMixer::TRACK, 4431 AudioMixer::MAIN_BUFFER, (void *)mMixerBuffer); 4432 // TODO: override track->mainBuffer()? 4433 mMixerBufferValid = true; 4434 } else { 4435 mAudioMixer->setParameter( 4436 name, 4437 AudioMixer::TRACK, 4438 AudioMixer::MIXER_FORMAT, (void *)AUDIO_FORMAT_PCM_16_BIT); 4439 mAudioMixer->setParameter( 4440 name, 4441 AudioMixer::TRACK, 4442 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 4443 } 4444 mAudioMixer->setParameter( 4445 name, 4446 AudioMixer::TRACK, 4447 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 4448 4449 // reset retry count 4450 track->mRetryCount = kMaxTrackRetries; 4451 4452 // If one track is ready, set the mixer ready if: 4453 // - the mixer was not ready during previous round OR 4454 // - no other track is not ready 4455 if (mMixerStatusIgnoringFastTracks != MIXER_TRACKS_READY || 4456 mixerStatus != MIXER_TRACKS_ENABLED) { 4457 mixerStatus = MIXER_TRACKS_READY; 4458 } 4459 } else { 4460 if (framesReady < desiredFrames && !track->isStopped() && !track->isPaused()) { 4461 ALOGV("track(%p) underrun, framesReady(%zu) < framesDesired(%zd)", 4462 track, framesReady, desiredFrames); 4463 track->mAudioTrackServerProxy->tallyUnderrunFrames(desiredFrames); 4464 } else { 4465 track->mAudioTrackServerProxy->tallyUnderrunFrames(0); 4466 } 4467 4468 // clear effect chain input buffer if an active track underruns to avoid sending 4469 // previous audio buffer again to effects 4470 chain = getEffectChain_l(track->sessionId()); 4471 if (chain != 0) { 4472 chain->clearInputBuffer(); 4473 } 4474 4475 ALOGVV("track %d s=%08x [NOT READY] on thread %p", name, cblk->mServer, this); 4476 if ((track->sharedBuffer() != 0) || track->isTerminated() || 4477 track->isStopped() || track->isPaused()) { 4478 // We have consumed all the buffers of this track. 4479 // Remove it from the list of active tracks. 4480 // TODO: use actual buffer filling status instead of latency when available from 4481 // audio HAL 4482 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 4483 int64_t framesWritten = mBytesWritten / mFrameSize; 4484 if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) { 4485 if (track->isStopped()) { 4486 track->reset(); 4487 } 4488 tracksToRemove->add(track); 4489 } 4490 } else { 4491 // No buffers for this track. Give it a few chances to 4492 // fill a buffer, then remove it from active list. 4493 if (--(track->mRetryCount) <= 0) { 4494 ALOGI("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 4495 tracksToRemove->add(track); 4496 // indicate to client process that the track was disabled because of underrun; 4497 // it will then automatically call start() when data is available 4498 track->disable(); 4499 // If one track is not ready, mark the mixer also not ready if: 4500 // - the mixer was ready during previous round OR 4501 // - no other track is ready 4502 } else if (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY || 4503 mixerStatus != MIXER_TRACKS_READY) { 4504 mixerStatus = MIXER_TRACKS_ENABLED; 4505 } 4506 } 4507 mAudioMixer->disable(name); 4508 } 4509 4510 } // local variable scope to avoid goto warning 4511 4512 } 4513 4514 // Push the new FastMixer state if necessary 4515 bool pauseAudioWatchdog = false; 4516 if (didModify) { 4517 state->mFastTracksGen++; 4518 // if the fast mixer was active, but now there are no fast tracks, then put it in cold idle 4519 if (kUseFastMixer == FastMixer_Dynamic && 4520 state->mCommand == FastMixerState::MIX_WRITE && state->mTrackMask <= 1) { 4521 state->mCommand = FastMixerState::COLD_IDLE; 4522 state->mColdFutexAddr = &mFastMixerFutex; 4523 state->mColdGen++; 4524 mFastMixerFutex = 0; 4525 if (kUseFastMixer == FastMixer_Dynamic) { 4526 mNormalSink = mOutputSink; 4527 } 4528 // If we go into cold idle, need to wait for acknowledgement 4529 // so that fast mixer stops doing I/O. 4530 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 4531 pauseAudioWatchdog = true; 4532 } 4533 } 4534 if (sq != NULL) { 4535 sq->end(didModify); 4536 sq->push(block); 4537 } 4538#ifdef AUDIO_WATCHDOG 4539 if (pauseAudioWatchdog && mAudioWatchdog != 0) { 4540 mAudioWatchdog->pause(); 4541 } 4542#endif 4543 4544 // Now perform the deferred reset on fast tracks that have stopped 4545 while (resetMask != 0) { 4546 size_t i = __builtin_ctz(resetMask); 4547 ALOG_ASSERT(i < count); 4548 resetMask &= ~(1 << i); 4549 sp<Track> t = mActiveTracks[i].promote(); 4550 if (t == 0) { 4551 continue; 4552 } 4553 Track* track = t.get(); 4554 ALOG_ASSERT(track->isFastTrack() && track->isStopped()); 4555 track->reset(); 4556 } 4557 4558 // remove all the tracks that need to be... 4559 removeTracks_l(*tracksToRemove); 4560 4561 if (getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX) != 0) { 4562 mEffectBufferValid = true; 4563 } 4564 4565 if (mEffectBufferValid) { 4566 // as long as there are effects we should clear the effects buffer, to avoid 4567 // passing a non-clean buffer to the effect chain 4568 memset(mEffectBuffer, 0, mEffectBufferSize); 4569 } 4570 // sink or mix buffer must be cleared if all tracks are connected to an 4571 // effect chain as in this case the mixer will not write to the sink or mix buffer 4572 // and track effects will accumulate into it 4573 if ((mBytesRemaining == 0) && ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 4574 (mixedTracks == 0 && fastTracks > 0))) { 4575 // FIXME as a performance optimization, should remember previous zero status 4576 if (mMixerBufferValid) { 4577 memset(mMixerBuffer, 0, mMixerBufferSize); 4578 // TODO: In testing, mSinkBuffer below need not be cleared because 4579 // the PlaybackThread::threadLoop() copies mMixerBuffer into mSinkBuffer 4580 // after mixing. 4581 // 4582 // To enforce this guarantee: 4583 // ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 4584 // (mixedTracks == 0 && fastTracks > 0)) 4585 // must imply MIXER_TRACKS_READY. 4586 // Later, we may clear buffers regardless, and skip much of this logic. 4587 } 4588 // FIXME as a performance optimization, should remember previous zero status 4589 memset(mSinkBuffer, 0, mNormalFrameCount * mFrameSize); 4590 } 4591 4592 // if any fast tracks, then status is ready 4593 mMixerStatusIgnoringFastTracks = mixerStatus; 4594 if (fastTracks > 0) { 4595 mixerStatus = MIXER_TRACKS_READY; 4596 } 4597 return mixerStatus; 4598} 4599 4600// trackCountForUid_l() must be called with ThreadBase::mLock held 4601uint32_t AudioFlinger::PlaybackThread::trackCountForUid_l(uid_t uid) 4602{ 4603 uint32_t trackCount = 0; 4604 for (size_t i = 0; i < mTracks.size() ; i++) { 4605 if (mTracks[i]->uid() == (int)uid) { 4606 trackCount++; 4607 } 4608 } 4609 return trackCount; 4610} 4611 4612// getTrackName_l() must be called with ThreadBase::mLock held 4613int AudioFlinger::MixerThread::getTrackName_l(audio_channel_mask_t channelMask, 4614 audio_format_t format, audio_session_t sessionId, uid_t uid) 4615{ 4616 if (trackCountForUid_l(uid) > (PlaybackThread::kMaxTracksPerUid - 1)) { 4617 return -1; 4618 } 4619 return mAudioMixer->getTrackName(channelMask, format, sessionId); 4620} 4621 4622// deleteTrackName_l() must be called with ThreadBase::mLock held 4623void AudioFlinger::MixerThread::deleteTrackName_l(int name) 4624{ 4625 ALOGV("remove track (%d) and delete from mixer", name); 4626 mAudioMixer->deleteTrackName(name); 4627} 4628 4629// checkForNewParameter_l() must be called with ThreadBase::mLock held 4630bool AudioFlinger::MixerThread::checkForNewParameter_l(const String8& keyValuePair, 4631 status_t& status) 4632{ 4633 bool reconfig = false; 4634 bool a2dpDeviceChanged = false; 4635 4636 status = NO_ERROR; 4637 4638 AutoPark<FastMixer> park(mFastMixer); 4639 4640 AudioParameter param = AudioParameter(keyValuePair); 4641 int value; 4642 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 4643 reconfig = true; 4644 } 4645 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 4646 if (!isValidPcmSinkFormat((audio_format_t) value)) { 4647 status = BAD_VALUE; 4648 } else { 4649 // no need to save value, since it's constant 4650 reconfig = true; 4651 } 4652 } 4653 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 4654 if (!isValidPcmSinkChannelMask((audio_channel_mask_t) value)) { 4655 status = BAD_VALUE; 4656 } else { 4657 // no need to save value, since it's constant 4658 reconfig = true; 4659 } 4660 } 4661 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 4662 // do not accept frame count changes if tracks are open as the track buffer 4663 // size depends on frame count and correct behavior would not be guaranteed 4664 // if frame count is changed after track creation 4665 if (!mTracks.isEmpty()) { 4666 status = INVALID_OPERATION; 4667 } else { 4668 reconfig = true; 4669 } 4670 } 4671 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 4672#ifdef ADD_BATTERY_DATA 4673 // when changing the audio output device, call addBatteryData to notify 4674 // the change 4675 if (mOutDevice != value) { 4676 uint32_t params = 0; 4677 // check whether speaker is on 4678 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 4679 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 4680 } 4681 4682 audio_devices_t deviceWithoutSpeaker 4683 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 4684 // check if any other device (except speaker) is on 4685 if (value & deviceWithoutSpeaker) { 4686 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 4687 } 4688 4689 if (params != 0) { 4690 addBatteryData(params); 4691 } 4692 } 4693#endif 4694 4695 // forward device change to effects that have requested to be 4696 // aware of attached audio device. 4697 if (value != AUDIO_DEVICE_NONE) { 4698 a2dpDeviceChanged = 4699 (mOutDevice & AUDIO_DEVICE_OUT_ALL_A2DP) != (value & AUDIO_DEVICE_OUT_ALL_A2DP); 4700 mOutDevice = value; 4701 for (size_t i = 0; i < mEffectChains.size(); i++) { 4702 mEffectChains[i]->setDevice_l(mOutDevice); 4703 } 4704 } 4705 } 4706 4707 if (status == NO_ERROR) { 4708 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 4709 keyValuePair.string()); 4710 if (!mStandby && status == INVALID_OPERATION) { 4711 mOutput->standby(); 4712 mStandby = true; 4713 mBytesWritten = 0; 4714 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 4715 keyValuePair.string()); 4716 } 4717 if (status == NO_ERROR && reconfig) { 4718 readOutputParameters_l(); 4719 delete mAudioMixer; 4720 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 4721 for (size_t i = 0; i < mTracks.size() ; i++) { 4722 int name = getTrackName_l(mTracks[i]->mChannelMask, 4723 mTracks[i]->mFormat, mTracks[i]->mSessionId, mTracks[i]->uid()); 4724 if (name < 0) { 4725 break; 4726 } 4727 mTracks[i]->mName = name; 4728 } 4729 sendIoConfigEvent_l(AUDIO_OUTPUT_CONFIG_CHANGED); 4730 } 4731 } 4732 4733 return reconfig || a2dpDeviceChanged; 4734} 4735 4736 4737void AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 4738{ 4739 PlaybackThread::dumpInternals(fd, args); 4740 dprintf(fd, " Thread throttle time (msecs): %u\n", mThreadThrottleTimeMs); 4741 dprintf(fd, " AudioMixer tracks: 0x%08x\n", mAudioMixer->trackNames()); 4742 dprintf(fd, " Master mono: %s\n", mMasterMono ? "on" : "off"); 4743 4744 // Make a non-atomic copy of fast mixer dump state so it won't change underneath us 4745 // while we are dumping it. It may be inconsistent, but it won't mutate! 4746 // This is a large object so we place it on the heap. 4747 // FIXME 25972958: Need an intelligent copy constructor that does not touch unused pages. 4748 const FastMixerDumpState *copy = new FastMixerDumpState(mFastMixerDumpState); 4749 copy->dump(fd); 4750 delete copy; 4751 4752#ifdef STATE_QUEUE_DUMP 4753 // Similar for state queue 4754 StateQueueObserverDump observerCopy = mStateQueueObserverDump; 4755 observerCopy.dump(fd); 4756 StateQueueMutatorDump mutatorCopy = mStateQueueMutatorDump; 4757 mutatorCopy.dump(fd); 4758#endif 4759 4760#ifdef TEE_SINK 4761 // Write the tee output to a .wav file 4762 dumpTee(fd, mTeeSource, mId); 4763#endif 4764 4765#ifdef AUDIO_WATCHDOG 4766 if (mAudioWatchdog != 0) { 4767 // Make a non-atomic copy of audio watchdog dump so it won't change underneath us 4768 AudioWatchdogDump wdCopy = mAudioWatchdogDump; 4769 wdCopy.dump(fd); 4770 } 4771#endif 4772} 4773 4774uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const 4775{ 4776 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000) / 2; 4777} 4778 4779uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() const 4780{ 4781 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000); 4782} 4783 4784void AudioFlinger::MixerThread::cacheParameters_l() 4785{ 4786 PlaybackThread::cacheParameters_l(); 4787 4788 // FIXME: Relaxed timing because of a certain device that can't meet latency 4789 // Should be reduced to 2x after the vendor fixes the driver issue 4790 // increase threshold again due to low power audio mode. The way this warning 4791 // threshold is calculated and its usefulness should be reconsidered anyway. 4792 maxPeriod = seconds(mNormalFrameCount) / mSampleRate * 15; 4793} 4794 4795// ---------------------------------------------------------------------------- 4796 4797AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 4798 AudioStreamOut* output, audio_io_handle_t id, audio_devices_t device, bool systemReady) 4799 : PlaybackThread(audioFlinger, output, id, device, DIRECT, systemReady) 4800 // mLeftVolFloat, mRightVolFloat 4801{ 4802} 4803 4804AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 4805 AudioStreamOut* output, audio_io_handle_t id, uint32_t device, 4806 ThreadBase::type_t type, bool systemReady) 4807 : PlaybackThread(audioFlinger, output, id, device, type, systemReady) 4808 // mLeftVolFloat, mRightVolFloat 4809{ 4810} 4811 4812AudioFlinger::DirectOutputThread::~DirectOutputThread() 4813{ 4814} 4815 4816void AudioFlinger::DirectOutputThread::processVolume_l(Track *track, bool lastTrack) 4817{ 4818 float left, right; 4819 4820 if (mMasterMute || mStreamTypes[track->streamType()].mute) { 4821 left = right = 0; 4822 } else { 4823 float typeVolume = mStreamTypes[track->streamType()].volume; 4824 float v = mMasterVolume * typeVolume; 4825 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 4826 gain_minifloat_packed_t vlr = proxy->getVolumeLR(); 4827 left = float_from_gain(gain_minifloat_unpack_left(vlr)); 4828 if (left > GAIN_FLOAT_UNITY) { 4829 left = GAIN_FLOAT_UNITY; 4830 } 4831 left *= v; 4832 right = float_from_gain(gain_minifloat_unpack_right(vlr)); 4833 if (right > GAIN_FLOAT_UNITY) { 4834 right = GAIN_FLOAT_UNITY; 4835 } 4836 right *= v; 4837 } 4838 4839 if (lastTrack) { 4840 if (left != mLeftVolFloat || right != mRightVolFloat) { 4841 mLeftVolFloat = left; 4842 mRightVolFloat = right; 4843 4844 // Convert volumes from float to 8.24 4845 uint32_t vl = (uint32_t)(left * (1 << 24)); 4846 uint32_t vr = (uint32_t)(right * (1 << 24)); 4847 4848 // Delegate volume control to effect in track effect chain if needed 4849 // only one effect chain can be present on DirectOutputThread, so if 4850 // there is one, the track is connected to it 4851 if (!mEffectChains.isEmpty()) { 4852 mEffectChains[0]->setVolume_l(&vl, &vr); 4853 left = (float)vl / (1 << 24); 4854 right = (float)vr / (1 << 24); 4855 } 4856 if (mOutput->stream->set_volume) { 4857 mOutput->stream->set_volume(mOutput->stream, left, right); 4858 } 4859 } 4860 } 4861} 4862 4863void AudioFlinger::DirectOutputThread::onAddNewTrack_l() 4864{ 4865 sp<Track> previousTrack = mPreviousTrack.promote(); 4866 sp<Track> latestTrack = mLatestActiveTrack.promote(); 4867 4868 if (previousTrack != 0 && latestTrack != 0) { 4869 if (mType == DIRECT) { 4870 if (previousTrack.get() != latestTrack.get()) { 4871 mFlushPending = true; 4872 } 4873 } else /* mType == OFFLOAD */ { 4874 if (previousTrack->sessionId() != latestTrack->sessionId()) { 4875 mFlushPending = true; 4876 } 4877 } 4878 } 4879 PlaybackThread::onAddNewTrack_l(); 4880} 4881 4882AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l( 4883 Vector< sp<Track> > *tracksToRemove 4884) 4885{ 4886 size_t count = mActiveTracks.size(); 4887 mixer_state mixerStatus = MIXER_IDLE; 4888 bool doHwPause = false; 4889 bool doHwResume = false; 4890 4891 // find out which tracks need to be processed 4892 for (size_t i = 0; i < count; i++) { 4893 sp<Track> t = mActiveTracks[i].promote(); 4894 // The track died recently 4895 if (t == 0) { 4896 continue; 4897 } 4898 4899 if (t->isInvalid()) { 4900 ALOGW("An invalidated track shouldn't be in active list"); 4901 tracksToRemove->add(t); 4902 continue; 4903 } 4904 4905 Track* const track = t.get(); 4906#ifdef VERY_VERY_VERBOSE_LOGGING 4907 audio_track_cblk_t* cblk = track->cblk(); 4908#endif 4909 // Only consider last track started for volume and mixer state control. 4910 // In theory an older track could underrun and restart after the new one starts 4911 // but as we only care about the transition phase between two tracks on a 4912 // direct output, it is not a problem to ignore the underrun case. 4913 sp<Track> l = mLatestActiveTrack.promote(); 4914 bool last = l.get() == track; 4915 4916 if (track->isPausing()) { 4917 track->setPaused(); 4918 if (mHwSupportsPause && last && !mHwPaused) { 4919 doHwPause = true; 4920 mHwPaused = true; 4921 } 4922 tracksToRemove->add(track); 4923 } else if (track->isFlushPending()) { 4924 track->flushAck(); 4925 if (last) { 4926 mFlushPending = true; 4927 } 4928 } else if (track->isResumePending()) { 4929 track->resumeAck(); 4930 if (last) { 4931 mLeftVolFloat = mRightVolFloat = -1.0; 4932 if (mHwPaused) { 4933 doHwResume = true; 4934 mHwPaused = false; 4935 } 4936 } 4937 } 4938 4939 // The first time a track is added we wait 4940 // for all its buffers to be filled before processing it. 4941 // Allow draining the buffer in case the client 4942 // app does not call stop() and relies on underrun to stop: 4943 // hence the test on (track->mRetryCount > 1). 4944 // If retryCount<=1 then track is about to underrun and be removed. 4945 // Do not use a high threshold for compressed audio. 4946 uint32_t minFrames; 4947 if ((track->sharedBuffer() == 0) && !track->isStopping_1() && !track->isPausing() 4948 && (track->mRetryCount > 1) && audio_has_proportional_frames(mFormat)) { 4949 minFrames = mNormalFrameCount; 4950 } else { 4951 minFrames = 1; 4952 } 4953 4954 if ((track->framesReady() >= minFrames) && track->isReady() && !track->isPaused() && 4955 !track->isStopping_2() && !track->isStopped()) 4956 { 4957 ALOGVV("track %d s=%08x [OK]", track->name(), cblk->mServer); 4958 4959 if (track->mFillingUpStatus == Track::FS_FILLED) { 4960 track->mFillingUpStatus = Track::FS_ACTIVE; 4961 if (last) { 4962 // make sure processVolume_l() will apply new volume even if 0 4963 mLeftVolFloat = mRightVolFloat = -1.0; 4964 } 4965 if (!mHwSupportsPause) { 4966 track->resumeAck(); 4967 } 4968 } 4969 4970 // compute volume for this track 4971 processVolume_l(track, last); 4972 if (last) { 4973 sp<Track> previousTrack = mPreviousTrack.promote(); 4974 if (previousTrack != 0) { 4975 if (track != previousTrack.get()) { 4976 // Flush any data still being written from last track 4977 mBytesRemaining = 0; 4978 // Invalidate previous track to force a seek when resuming. 4979 previousTrack->invalidate(); 4980 } 4981 } 4982 mPreviousTrack = track; 4983 4984 // reset retry count 4985 track->mRetryCount = kMaxTrackRetriesDirect; 4986 mActiveTrack = t; 4987 mixerStatus = MIXER_TRACKS_READY; 4988 if (mHwPaused) { 4989 doHwResume = true; 4990 mHwPaused = false; 4991 } 4992 } 4993 } else { 4994 // clear effect chain input buffer if the last active track started underruns 4995 // to avoid sending previous audio buffer again to effects 4996 if (!mEffectChains.isEmpty() && last) { 4997 mEffectChains[0]->clearInputBuffer(); 4998 } 4999 if (track->isStopping_1()) { 5000 track->mState = TrackBase::STOPPING_2; 5001 if (last && mHwPaused) { 5002 doHwResume = true; 5003 mHwPaused = false; 5004 } 5005 } 5006 if ((track->sharedBuffer() != 0) || track->isStopped() || 5007 track->isStopping_2() || track->isPaused()) { 5008 // We have consumed all the buffers of this track. 5009 // Remove it from the list of active tracks. 5010 size_t audioHALFrames; 5011 if (audio_has_proportional_frames(mFormat)) { 5012 audioHALFrames = (latency_l() * mSampleRate) / 1000; 5013 } else { 5014 audioHALFrames = 0; 5015 } 5016 5017 int64_t framesWritten = mBytesWritten / mFrameSize; 5018 if (mStandby || !last || 5019 track->presentationComplete(framesWritten, audioHALFrames)) { 5020 if (track->isStopping_2()) { 5021 track->mState = TrackBase::STOPPED; 5022 } 5023 if (track->isStopped()) { 5024 track->reset(); 5025 } 5026 tracksToRemove->add(track); 5027 } 5028 } else { 5029 // No buffers for this track. Give it a few chances to 5030 // fill a buffer, then remove it from active list. 5031 // Only consider last track started for mixer state control 5032 if (--(track->mRetryCount) <= 0) { 5033 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 5034 tracksToRemove->add(track); 5035 // indicate to client process that the track was disabled because of underrun; 5036 // it will then automatically call start() when data is available 5037 track->disable(); 5038 } else if (last) { 5039 ALOGW("pause because of UNDERRUN, framesReady = %zu," 5040 "minFrames = %u, mFormat = %#x", 5041 track->framesReady(), minFrames, mFormat); 5042 mixerStatus = MIXER_TRACKS_ENABLED; 5043 if (mHwSupportsPause && !mHwPaused && !mStandby) { 5044 doHwPause = true; 5045 mHwPaused = true; 5046 } 5047 } 5048 } 5049 } 5050 } 5051 5052 // if an active track did not command a flush, check for pending flush on stopped tracks 5053 if (!mFlushPending) { 5054 for (size_t i = 0; i < mTracks.size(); i++) { 5055 if (mTracks[i]->isFlushPending()) { 5056 mTracks[i]->flushAck(); 5057 mFlushPending = true; 5058 } 5059 } 5060 } 5061 5062 // make sure the pause/flush/resume sequence is executed in the right order. 5063 // If a flush is pending and a track is active but the HW is not paused, force a HW pause 5064 // before flush and then resume HW. This can happen in case of pause/flush/resume 5065 // if resume is received before pause is executed. 5066 if (mHwSupportsPause && !mStandby && 5067 (doHwPause || (mFlushPending && !mHwPaused && (count != 0)))) { 5068 mOutput->stream->pause(mOutput->stream); 5069 } 5070 if (mFlushPending) { 5071 flushHw_l(); 5072 } 5073 if (mHwSupportsPause && !mStandby && doHwResume) { 5074 mOutput->stream->resume(mOutput->stream); 5075 } 5076 // remove all the tracks that need to be... 5077 removeTracks_l(*tracksToRemove); 5078 5079 return mixerStatus; 5080} 5081 5082void AudioFlinger::DirectOutputThread::threadLoop_mix() 5083{ 5084 size_t frameCount = mFrameCount; 5085 int8_t *curBuf = (int8_t *)mSinkBuffer; 5086 // output audio to hardware 5087 while (frameCount) { 5088 AudioBufferProvider::Buffer buffer; 5089 buffer.frameCount = frameCount; 5090 status_t status = mActiveTrack->getNextBuffer(&buffer); 5091 if (status != NO_ERROR || buffer.raw == NULL) { 5092 // no need to pad with 0 for compressed audio 5093 if (audio_has_proportional_frames(mFormat)) { 5094 memset(curBuf, 0, frameCount * mFrameSize); 5095 } 5096 break; 5097 } 5098 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 5099 frameCount -= buffer.frameCount; 5100 curBuf += buffer.frameCount * mFrameSize; 5101 mActiveTrack->releaseBuffer(&buffer); 5102 } 5103 mCurrentWriteLength = curBuf - (int8_t *)mSinkBuffer; 5104 mSleepTimeUs = 0; 5105 mStandbyTimeNs = systemTime() + mStandbyDelayNs; 5106 mActiveTrack.clear(); 5107} 5108 5109void AudioFlinger::DirectOutputThread::threadLoop_sleepTime() 5110{ 5111 // do not write to HAL when paused 5112 if (mHwPaused || (usesHwAvSync() && mStandby)) { 5113 mSleepTimeUs = mIdleSleepTimeUs; 5114 return; 5115 } 5116 if (mSleepTimeUs == 0) { 5117 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 5118 mSleepTimeUs = mActiveSleepTimeUs; 5119 } else { 5120 mSleepTimeUs = mIdleSleepTimeUs; 5121 } 5122 } else if (mBytesWritten != 0 && audio_has_proportional_frames(mFormat)) { 5123 memset(mSinkBuffer, 0, mFrameCount * mFrameSize); 5124 mSleepTimeUs = 0; 5125 } 5126} 5127 5128void AudioFlinger::DirectOutputThread::threadLoop_exit() 5129{ 5130 { 5131 Mutex::Autolock _l(mLock); 5132 for (size_t i = 0; i < mTracks.size(); i++) { 5133 if (mTracks[i]->isFlushPending()) { 5134 mTracks[i]->flushAck(); 5135 mFlushPending = true; 5136 } 5137 } 5138 if (mFlushPending) { 5139 flushHw_l(); 5140 } 5141 } 5142 PlaybackThread::threadLoop_exit(); 5143} 5144 5145// must be called with thread mutex locked 5146bool AudioFlinger::DirectOutputThread::shouldStandby_l() 5147{ 5148 bool trackPaused = false; 5149 bool trackStopped = false; 5150 5151 if ((mType == DIRECT) && audio_is_linear_pcm(mFormat) && !usesHwAvSync()) { 5152 return !mStandby; 5153 } 5154 5155 // do not put the HAL in standby when paused. AwesomePlayer clear the offloaded AudioTrack 5156 // after a timeout and we will enter standby then. 5157 if (mTracks.size() > 0) { 5158 trackPaused = mTracks[mTracks.size() - 1]->isPaused(); 5159 trackStopped = mTracks[mTracks.size() - 1]->isStopped() || 5160 mTracks[mTracks.size() - 1]->mState == TrackBase::IDLE; 5161 } 5162 5163 return !mStandby && !(trackPaused || (mHwPaused && !trackStopped)); 5164} 5165 5166// getTrackName_l() must be called with ThreadBase::mLock held 5167int AudioFlinger::DirectOutputThread::getTrackName_l(audio_channel_mask_t channelMask __unused, 5168 audio_format_t format __unused, audio_session_t sessionId __unused, uid_t uid) 5169{ 5170 if (trackCountForUid_l(uid) > (PlaybackThread::kMaxTracksPerUid - 1)) { 5171 return -1; 5172 } 5173 return 0; 5174} 5175 5176// deleteTrackName_l() must be called with ThreadBase::mLock held 5177void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name __unused) 5178{ 5179} 5180 5181// checkForNewParameter_l() must be called with ThreadBase::mLock held 5182bool AudioFlinger::DirectOutputThread::checkForNewParameter_l(const String8& keyValuePair, 5183 status_t& status) 5184{ 5185 bool reconfig = false; 5186 bool a2dpDeviceChanged = false; 5187 5188 status = NO_ERROR; 5189 5190 AudioParameter param = AudioParameter(keyValuePair); 5191 int value; 5192 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 5193 // forward device change to effects that have requested to be 5194 // aware of attached audio device. 5195 if (value != AUDIO_DEVICE_NONE) { 5196 a2dpDeviceChanged = 5197 (mOutDevice & AUDIO_DEVICE_OUT_ALL_A2DP) != (value & AUDIO_DEVICE_OUT_ALL_A2DP); 5198 mOutDevice = value; 5199 for (size_t i = 0; i < mEffectChains.size(); i++) { 5200 mEffectChains[i]->setDevice_l(mOutDevice); 5201 } 5202 } 5203 } 5204 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 5205 // do not accept frame count changes if tracks are open as the track buffer 5206 // size depends on frame count and correct behavior would not be garantied 5207 // if frame count is changed after track creation 5208 if (!mTracks.isEmpty()) { 5209 status = INVALID_OPERATION; 5210 } else { 5211 reconfig = true; 5212 } 5213 } 5214 if (status == NO_ERROR) { 5215 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 5216 keyValuePair.string()); 5217 if (!mStandby && status == INVALID_OPERATION) { 5218 mOutput->standby(); 5219 mStandby = true; 5220 mBytesWritten = 0; 5221 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 5222 keyValuePair.string()); 5223 } 5224 if (status == NO_ERROR && reconfig) { 5225 readOutputParameters_l(); 5226 sendIoConfigEvent_l(AUDIO_OUTPUT_CONFIG_CHANGED); 5227 } 5228 } 5229 5230 return reconfig || a2dpDeviceChanged; 5231} 5232 5233uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() const 5234{ 5235 uint32_t time; 5236 if (audio_has_proportional_frames(mFormat)) { 5237 time = PlaybackThread::activeSleepTimeUs(); 5238 } else { 5239 time = kDirectMinSleepTimeUs; 5240 } 5241 return time; 5242} 5243 5244uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() const 5245{ 5246 uint32_t time; 5247 if (audio_has_proportional_frames(mFormat)) { 5248 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 5249 } else { 5250 time = kDirectMinSleepTimeUs; 5251 } 5252 return time; 5253} 5254 5255uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() const 5256{ 5257 uint32_t time; 5258 if (audio_has_proportional_frames(mFormat)) { 5259 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 5260 } else { 5261 time = kDirectMinSleepTimeUs; 5262 } 5263 return time; 5264} 5265 5266void AudioFlinger::DirectOutputThread::cacheParameters_l() 5267{ 5268 PlaybackThread::cacheParameters_l(); 5269 5270 // use shorter standby delay as on normal output to release 5271 // hardware resources as soon as possible 5272 // no delay on outputs with HW A/V sync 5273 if (usesHwAvSync()) { 5274 mStandbyDelayNs = 0; 5275 } else if ((mType == OFFLOAD) && !audio_has_proportional_frames(mFormat)) { 5276 mStandbyDelayNs = kOffloadStandbyDelayNs; 5277 } else { 5278 mStandbyDelayNs = microseconds(mActiveSleepTimeUs*2); 5279 } 5280} 5281 5282void AudioFlinger::DirectOutputThread::flushHw_l() 5283{ 5284 mOutput->flush(); 5285 mHwPaused = false; 5286 mFlushPending = false; 5287} 5288 5289// ---------------------------------------------------------------------------- 5290 5291AudioFlinger::AsyncCallbackThread::AsyncCallbackThread( 5292 const wp<AudioFlinger::PlaybackThread>& playbackThread) 5293 : Thread(false /*canCallJava*/), 5294 mPlaybackThread(playbackThread), 5295 mWriteAckSequence(0), 5296 mDrainSequence(0), 5297 mAsyncError(false) 5298{ 5299} 5300 5301AudioFlinger::AsyncCallbackThread::~AsyncCallbackThread() 5302{ 5303} 5304 5305void AudioFlinger::AsyncCallbackThread::onFirstRef() 5306{ 5307 run("Offload Cbk", ANDROID_PRIORITY_URGENT_AUDIO); 5308} 5309 5310bool AudioFlinger::AsyncCallbackThread::threadLoop() 5311{ 5312 while (!exitPending()) { 5313 uint32_t writeAckSequence; 5314 uint32_t drainSequence; 5315 bool asyncError; 5316 5317 { 5318 Mutex::Autolock _l(mLock); 5319 while (!((mWriteAckSequence & 1) || 5320 (mDrainSequence & 1) || 5321 mAsyncError || 5322 exitPending())) { 5323 mWaitWorkCV.wait(mLock); 5324 } 5325 5326 if (exitPending()) { 5327 break; 5328 } 5329 ALOGV("AsyncCallbackThread mWriteAckSequence %d mDrainSequence %d", 5330 mWriteAckSequence, mDrainSequence); 5331 writeAckSequence = mWriteAckSequence; 5332 mWriteAckSequence &= ~1; 5333 drainSequence = mDrainSequence; 5334 mDrainSequence &= ~1; 5335 asyncError = mAsyncError; 5336 mAsyncError = false; 5337 } 5338 { 5339 sp<AudioFlinger::PlaybackThread> playbackThread = mPlaybackThread.promote(); 5340 if (playbackThread != 0) { 5341 if (writeAckSequence & 1) { 5342 playbackThread->resetWriteBlocked(writeAckSequence >> 1); 5343 } 5344 if (drainSequence & 1) { 5345 playbackThread->resetDraining(drainSequence >> 1); 5346 } 5347 if (asyncError) { 5348 playbackThread->onAsyncError(); 5349 } 5350 } 5351 } 5352 } 5353 return false; 5354} 5355 5356void AudioFlinger::AsyncCallbackThread::exit() 5357{ 5358 ALOGV("AsyncCallbackThread::exit"); 5359 Mutex::Autolock _l(mLock); 5360 requestExit(); 5361 mWaitWorkCV.broadcast(); 5362} 5363 5364void AudioFlinger::AsyncCallbackThread::setWriteBlocked(uint32_t sequence) 5365{ 5366 Mutex::Autolock _l(mLock); 5367 // bit 0 is cleared 5368 mWriteAckSequence = sequence << 1; 5369} 5370 5371void AudioFlinger::AsyncCallbackThread::resetWriteBlocked() 5372{ 5373 Mutex::Autolock _l(mLock); 5374 // ignore unexpected callbacks 5375 if (mWriteAckSequence & 2) { 5376 mWriteAckSequence |= 1; 5377 mWaitWorkCV.signal(); 5378 } 5379} 5380 5381void AudioFlinger::AsyncCallbackThread::setDraining(uint32_t sequence) 5382{ 5383 Mutex::Autolock _l(mLock); 5384 // bit 0 is cleared 5385 mDrainSequence = sequence << 1; 5386} 5387 5388void AudioFlinger::AsyncCallbackThread::resetDraining() 5389{ 5390 Mutex::Autolock _l(mLock); 5391 // ignore unexpected callbacks 5392 if (mDrainSequence & 2) { 5393 mDrainSequence |= 1; 5394 mWaitWorkCV.signal(); 5395 } 5396} 5397 5398void AudioFlinger::AsyncCallbackThread::setAsyncError() 5399{ 5400 Mutex::Autolock _l(mLock); 5401 mAsyncError = true; 5402 mWaitWorkCV.signal(); 5403} 5404 5405 5406// ---------------------------------------------------------------------------- 5407AudioFlinger::OffloadThread::OffloadThread(const sp<AudioFlinger>& audioFlinger, 5408 AudioStreamOut* output, audio_io_handle_t id, uint32_t device, bool systemReady) 5409 : DirectOutputThread(audioFlinger, output, id, device, OFFLOAD, systemReady), 5410 mPausedWriteLength(0), mPausedBytesRemaining(0), mKeepWakeLock(true), 5411 mOffloadUnderrunPosition(~0LL) 5412{ 5413 //FIXME: mStandby should be set to true by ThreadBase constructor 5414 mStandby = true; 5415 mKeepWakeLock = property_get_bool("ro.audio.offload_wakelock", true /* default_value */); 5416} 5417 5418void AudioFlinger::OffloadThread::threadLoop_exit() 5419{ 5420 if (mFlushPending || mHwPaused) { 5421 // If a flush is pending or track was paused, just discard buffered data 5422 flushHw_l(); 5423 } else { 5424 mMixerStatus = MIXER_DRAIN_ALL; 5425 threadLoop_drain(); 5426 } 5427 if (mUseAsyncWrite) { 5428 ALOG_ASSERT(mCallbackThread != 0); 5429 mCallbackThread->exit(); 5430 } 5431 PlaybackThread::threadLoop_exit(); 5432} 5433 5434AudioFlinger::PlaybackThread::mixer_state AudioFlinger::OffloadThread::prepareTracks_l( 5435 Vector< sp<Track> > *tracksToRemove 5436) 5437{ 5438 size_t count = mActiveTracks.size(); 5439 5440 mixer_state mixerStatus = MIXER_IDLE; 5441 bool doHwPause = false; 5442 bool doHwResume = false; 5443 5444 ALOGV("OffloadThread::prepareTracks_l active tracks %zu", count); 5445 5446 // find out which tracks need to be processed 5447 for (size_t i = 0; i < count; i++) { 5448 sp<Track> t = mActiveTracks[i].promote(); 5449 // The track died recently 5450 if (t == 0) { 5451 continue; 5452 } 5453 Track* const track = t.get(); 5454#ifdef VERY_VERY_VERBOSE_LOGGING 5455 audio_track_cblk_t* cblk = track->cblk(); 5456#endif 5457 // Only consider last track started for volume and mixer state control. 5458 // In theory an older track could underrun and restart after the new one starts 5459 // but as we only care about the transition phase between two tracks on a 5460 // direct output, it is not a problem to ignore the underrun case. 5461 sp<Track> l = mLatestActiveTrack.promote(); 5462 bool last = l.get() == track; 5463 5464 if (track->isInvalid()) { 5465 ALOGW("An invalidated track shouldn't be in active list"); 5466 tracksToRemove->add(track); 5467 continue; 5468 } 5469 5470 if (track->mState == TrackBase::IDLE) { 5471 ALOGW("An idle track shouldn't be in active list"); 5472 continue; 5473 } 5474 5475 if (track->isPausing()) { 5476 track->setPaused(); 5477 if (last) { 5478 if (mHwSupportsPause && !mHwPaused) { 5479 doHwPause = true; 5480 mHwPaused = true; 5481 } 5482 // If we were part way through writing the mixbuffer to 5483 // the HAL we must save this until we resume 5484 // BUG - this will be wrong if a different track is made active, 5485 // in that case we want to discard the pending data in the 5486 // mixbuffer and tell the client to present it again when the 5487 // track is resumed 5488 mPausedWriteLength = mCurrentWriteLength; 5489 mPausedBytesRemaining = mBytesRemaining; 5490 mBytesRemaining = 0; // stop writing 5491 } 5492 tracksToRemove->add(track); 5493 } else if (track->isFlushPending()) { 5494 if (track->isStopping_1()) { 5495 track->mRetryCount = kMaxTrackStopRetriesOffload; 5496 } else { 5497 track->mRetryCount = kMaxTrackRetriesOffload; 5498 } 5499 track->flushAck(); 5500 if (last) { 5501 mFlushPending = true; 5502 } 5503 } else if (track->isResumePending()){ 5504 track->resumeAck(); 5505 if (last) { 5506 if (mPausedBytesRemaining) { 5507 // Need to continue write that was interrupted 5508 mCurrentWriteLength = mPausedWriteLength; 5509 mBytesRemaining = mPausedBytesRemaining; 5510 mPausedBytesRemaining = 0; 5511 } 5512 if (mHwPaused) { 5513 doHwResume = true; 5514 mHwPaused = false; 5515 // threadLoop_mix() will handle the case that we need to 5516 // resume an interrupted write 5517 } 5518 // enable write to audio HAL 5519 mSleepTimeUs = 0; 5520 5521 mLeftVolFloat = mRightVolFloat = -1.0; 5522 5523 // Do not handle new data in this iteration even if track->framesReady() 5524 mixerStatus = MIXER_TRACKS_ENABLED; 5525 } 5526 } else if (track->framesReady() && track->isReady() && 5527 !track->isPaused() && !track->isTerminated() && !track->isStopping_2()) { 5528 ALOGVV("OffloadThread: track %d s=%08x [OK]", track->name(), cblk->mServer); 5529 if (track->mFillingUpStatus == Track::FS_FILLED) { 5530 track->mFillingUpStatus = Track::FS_ACTIVE; 5531 if (last) { 5532 // make sure processVolume_l() will apply new volume even if 0 5533 mLeftVolFloat = mRightVolFloat = -1.0; 5534 } 5535 } 5536 5537 if (last) { 5538 sp<Track> previousTrack = mPreviousTrack.promote(); 5539 if (previousTrack != 0) { 5540 if (track != previousTrack.get()) { 5541 // Flush any data still being written from last track 5542 mBytesRemaining = 0; 5543 if (mPausedBytesRemaining) { 5544 // Last track was paused so we also need to flush saved 5545 // mixbuffer state and invalidate track so that it will 5546 // re-submit that unwritten data when it is next resumed 5547 mPausedBytesRemaining = 0; 5548 // Invalidate is a bit drastic - would be more efficient 5549 // to have a flag to tell client that some of the 5550 // previously written data was lost 5551 previousTrack->invalidate(); 5552 } 5553 // flush data already sent to the DSP if changing audio session as audio 5554 // comes from a different source. Also invalidate previous track to force a 5555 // seek when resuming. 5556 if (previousTrack->sessionId() != track->sessionId()) { 5557 previousTrack->invalidate(); 5558 } 5559 } 5560 } 5561 mPreviousTrack = track; 5562 // reset retry count 5563 if (track->isStopping_1()) { 5564 track->mRetryCount = kMaxTrackStopRetriesOffload; 5565 } else { 5566 track->mRetryCount = kMaxTrackRetriesOffload; 5567 } 5568 mActiveTrack = t; 5569 mixerStatus = MIXER_TRACKS_READY; 5570 } 5571 } else { 5572 ALOGVV("OffloadThread: track %d s=%08x [NOT READY]", track->name(), cblk->mServer); 5573 if (track->isStopping_1()) { 5574 if (--(track->mRetryCount) <= 0) { 5575 // Hardware buffer can hold a large amount of audio so we must 5576 // wait for all current track's data to drain before we say 5577 // that the track is stopped. 5578 if (mBytesRemaining == 0) { 5579 // Only start draining when all data in mixbuffer 5580 // has been written 5581 ALOGV("OffloadThread: underrun and STOPPING_1 -> draining, STOPPING_2"); 5582 track->mState = TrackBase::STOPPING_2; // so presentation completes after 5583 // drain do not drain if no data was ever sent to HAL (mStandby == true) 5584 if (last && !mStandby) { 5585 // do not modify drain sequence if we are already draining. This happens 5586 // when resuming from pause after drain. 5587 if ((mDrainSequence & 1) == 0) { 5588 mSleepTimeUs = 0; 5589 mStandbyTimeNs = systemTime() + mStandbyDelayNs; 5590 mixerStatus = MIXER_DRAIN_TRACK; 5591 mDrainSequence += 2; 5592 } 5593 if (mHwPaused) { 5594 // It is possible to move from PAUSED to STOPPING_1 without 5595 // a resume so we must ensure hardware is running 5596 doHwResume = true; 5597 mHwPaused = false; 5598 } 5599 } 5600 } 5601 } else if (last) { 5602 ALOGV("stopping1 underrun retries left %d", track->mRetryCount); 5603 mixerStatus = MIXER_TRACKS_ENABLED; 5604 } 5605 } else if (track->isStopping_2()) { 5606 // Drain has completed or we are in standby, signal presentation complete 5607 if (!(mDrainSequence & 1) || !last || mStandby) { 5608 track->mState = TrackBase::STOPPED; 5609 size_t audioHALFrames = 5610 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 5611 int64_t framesWritten = 5612 mBytesWritten / mOutput->getFrameSize(); 5613 track->presentationComplete(framesWritten, audioHALFrames); 5614 track->reset(); 5615 tracksToRemove->add(track); 5616 } 5617 } else { 5618 // No buffers for this track. Give it a few chances to 5619 // fill a buffer, then remove it from active list. 5620 if (--(track->mRetryCount) <= 0) { 5621 bool running = false; 5622 if (mOutput->stream->get_presentation_position != nullptr) { 5623 uint64_t position = 0; 5624 struct timespec unused; 5625 // The running check restarts the retry counter at least once. 5626 int ret = mOutput->stream->get_presentation_position( 5627 mOutput->stream, &position, &unused); 5628 if (ret == NO_ERROR && position != mOffloadUnderrunPosition) { 5629 running = true; 5630 mOffloadUnderrunPosition = position; 5631 } 5632 ALOGVV("underrun counter, running(%d): %lld vs %lld", running, 5633 (long long)position, (long long)mOffloadUnderrunPosition); 5634 } 5635 if (running) { // still running, give us more time. 5636 track->mRetryCount = kMaxTrackRetriesOffload; 5637 } else { 5638 ALOGV("OffloadThread: BUFFER TIMEOUT: remove(%d) from active list", 5639 track->name()); 5640 tracksToRemove->add(track); 5641 // indicate to client process that the track was disabled because of underrun; 5642 // it will then automatically call start() when data is available 5643 track->disable(); 5644 } 5645 } else if (last){ 5646 mixerStatus = MIXER_TRACKS_ENABLED; 5647 } 5648 } 5649 } 5650 // compute volume for this track 5651 processVolume_l(track, last); 5652 } 5653 5654 // make sure the pause/flush/resume sequence is executed in the right order. 5655 // If a flush is pending and a track is active but the HW is not paused, force a HW pause 5656 // before flush and then resume HW. This can happen in case of pause/flush/resume 5657 // if resume is received before pause is executed. 5658 if (!mStandby && (doHwPause || (mFlushPending && !mHwPaused && (count != 0)))) { 5659 mOutput->stream->pause(mOutput->stream); 5660 } 5661 if (mFlushPending) { 5662 flushHw_l(); 5663 } 5664 if (!mStandby && doHwResume) { 5665 mOutput->stream->resume(mOutput->stream); 5666 } 5667 5668 // remove all the tracks that need to be... 5669 removeTracks_l(*tracksToRemove); 5670 5671 return mixerStatus; 5672} 5673 5674// must be called with thread mutex locked 5675bool AudioFlinger::OffloadThread::waitingAsyncCallback_l() 5676{ 5677 ALOGVV("waitingAsyncCallback_l mWriteAckSequence %d mDrainSequence %d", 5678 mWriteAckSequence, mDrainSequence); 5679 if (mUseAsyncWrite && ((mWriteAckSequence & 1) || (mDrainSequence & 1))) { 5680 return true; 5681 } 5682 return false; 5683} 5684 5685bool AudioFlinger::OffloadThread::waitingAsyncCallback() 5686{ 5687 Mutex::Autolock _l(mLock); 5688 return waitingAsyncCallback_l(); 5689} 5690 5691void AudioFlinger::OffloadThread::flushHw_l() 5692{ 5693 DirectOutputThread::flushHw_l(); 5694 // Flush anything still waiting in the mixbuffer 5695 mCurrentWriteLength = 0; 5696 mBytesRemaining = 0; 5697 mPausedWriteLength = 0; 5698 mPausedBytesRemaining = 0; 5699 // reset bytes written count to reflect that DSP buffers are empty after flush. 5700 mBytesWritten = 0; 5701 mOffloadUnderrunPosition = ~0LL; 5702 5703 if (mUseAsyncWrite) { 5704 // discard any pending drain or write ack by incrementing sequence 5705 mWriteAckSequence = (mWriteAckSequence + 2) & ~1; 5706 mDrainSequence = (mDrainSequence + 2) & ~1; 5707 ALOG_ASSERT(mCallbackThread != 0); 5708 mCallbackThread->setWriteBlocked(mWriteAckSequence); 5709 mCallbackThread->setDraining(mDrainSequence); 5710 } 5711} 5712 5713void AudioFlinger::OffloadThread::invalidateTracks(audio_stream_type_t streamType) 5714{ 5715 Mutex::Autolock _l(mLock); 5716 if (PlaybackThread::invalidateTracks_l(streamType)) { 5717 mFlushPending = true; 5718 } 5719} 5720 5721// ---------------------------------------------------------------------------- 5722 5723AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 5724 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id, bool systemReady) 5725 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->outDevice(), 5726 systemReady, DUPLICATING), 5727 mWaitTimeMs(UINT_MAX) 5728{ 5729 addOutputTrack(mainThread); 5730} 5731 5732AudioFlinger::DuplicatingThread::~DuplicatingThread() 5733{ 5734 for (size_t i = 0; i < mOutputTracks.size(); i++) { 5735 mOutputTracks[i]->destroy(); 5736 } 5737} 5738 5739void AudioFlinger::DuplicatingThread::threadLoop_mix() 5740{ 5741 // mix buffers... 5742 if (outputsReady(outputTracks)) { 5743 mAudioMixer->process(); 5744 } else { 5745 if (mMixerBufferValid) { 5746 memset(mMixerBuffer, 0, mMixerBufferSize); 5747 } else { 5748 memset(mSinkBuffer, 0, mSinkBufferSize); 5749 } 5750 } 5751 mSleepTimeUs = 0; 5752 writeFrames = mNormalFrameCount; 5753 mCurrentWriteLength = mSinkBufferSize; 5754 mStandbyTimeNs = systemTime() + mStandbyDelayNs; 5755} 5756 5757void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() 5758{ 5759 if (mSleepTimeUs == 0) { 5760 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 5761 mSleepTimeUs = mActiveSleepTimeUs; 5762 } else { 5763 mSleepTimeUs = mIdleSleepTimeUs; 5764 } 5765 } else if (mBytesWritten != 0) { 5766 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 5767 writeFrames = mNormalFrameCount; 5768 memset(mSinkBuffer, 0, mSinkBufferSize); 5769 } else { 5770 // flush remaining overflow buffers in output tracks 5771 writeFrames = 0; 5772 } 5773 mSleepTimeUs = 0; 5774 } 5775} 5776 5777ssize_t AudioFlinger::DuplicatingThread::threadLoop_write() 5778{ 5779 for (size_t i = 0; i < outputTracks.size(); i++) { 5780 outputTracks[i]->write(mSinkBuffer, writeFrames); 5781 } 5782 mStandby = false; 5783 return (ssize_t)mSinkBufferSize; 5784} 5785 5786void AudioFlinger::DuplicatingThread::threadLoop_standby() 5787{ 5788 // DuplicatingThread implements standby by stopping all tracks 5789 for (size_t i = 0; i < outputTracks.size(); i++) { 5790 outputTracks[i]->stop(); 5791 } 5792} 5793 5794void AudioFlinger::DuplicatingThread::saveOutputTracks() 5795{ 5796 outputTracks = mOutputTracks; 5797} 5798 5799void AudioFlinger::DuplicatingThread::clearOutputTracks() 5800{ 5801 outputTracks.clear(); 5802} 5803 5804void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 5805{ 5806 Mutex::Autolock _l(mLock); 5807 // The downstream MixerThread consumes thread->frameCount() amount of frames per mix pass. 5808 // Adjust for thread->sampleRate() to determine minimum buffer frame count. 5809 // Then triple buffer because Threads do not run synchronously and may not be clock locked. 5810 const size_t frameCount = 5811 3 * sourceFramesNeeded(mSampleRate, thread->frameCount(), thread->sampleRate()); 5812 // TODO: Consider asynchronous sample rate conversion to handle clock disparity 5813 // from different OutputTracks and their associated MixerThreads (e.g. one may 5814 // nearly empty and the other may be dropping data). 5815 5816 sp<OutputTrack> outputTrack = new OutputTrack(thread, 5817 this, 5818 mSampleRate, 5819 mFormat, 5820 mChannelMask, 5821 frameCount, 5822 IPCThreadState::self()->getCallingUid()); 5823 status_t status = outputTrack != 0 ? outputTrack->initCheck() : (status_t) NO_MEMORY; 5824 if (status != NO_ERROR) { 5825 ALOGE("addOutputTrack() initCheck failed %d", status); 5826 return; 5827 } 5828 thread->setStreamVolume(AUDIO_STREAM_PATCH, 1.0f); 5829 mOutputTracks.add(outputTrack); 5830 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack.get(), thread); 5831 updateWaitTime_l(); 5832} 5833 5834void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 5835{ 5836 Mutex::Autolock _l(mLock); 5837 for (size_t i = 0; i < mOutputTracks.size(); i++) { 5838 if (mOutputTracks[i]->thread() == thread) { 5839 mOutputTracks[i]->destroy(); 5840 mOutputTracks.removeAt(i); 5841 updateWaitTime_l(); 5842 if (thread->getOutput() == mOutput) { 5843 mOutput = NULL; 5844 } 5845 return; 5846 } 5847 } 5848 ALOGV("removeOutputTrack(): unknown thread: %p", thread); 5849} 5850 5851// caller must hold mLock 5852void AudioFlinger::DuplicatingThread::updateWaitTime_l() 5853{ 5854 mWaitTimeMs = UINT_MAX; 5855 for (size_t i = 0; i < mOutputTracks.size(); i++) { 5856 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 5857 if (strong != 0) { 5858 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 5859 if (waitTimeMs < mWaitTimeMs) { 5860 mWaitTimeMs = waitTimeMs; 5861 } 5862 } 5863 } 5864} 5865 5866 5867bool AudioFlinger::DuplicatingThread::outputsReady( 5868 const SortedVector< sp<OutputTrack> > &outputTracks) 5869{ 5870 for (size_t i = 0; i < outputTracks.size(); i++) { 5871 sp<ThreadBase> thread = outputTracks[i]->thread().promote(); 5872 if (thread == 0) { 5873 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", 5874 outputTracks[i].get()); 5875 return false; 5876 } 5877 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 5878 // see note at standby() declaration 5879 if (playbackThread->standby() && !playbackThread->isSuspended()) { 5880 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), 5881 thread.get()); 5882 return false; 5883 } 5884 } 5885 return true; 5886} 5887 5888uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() const 5889{ 5890 return (mWaitTimeMs * 1000) / 2; 5891} 5892 5893void AudioFlinger::DuplicatingThread::cacheParameters_l() 5894{ 5895 // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first 5896 updateWaitTime_l(); 5897 5898 MixerThread::cacheParameters_l(); 5899} 5900 5901// ---------------------------------------------------------------------------- 5902// Record 5903// ---------------------------------------------------------------------------- 5904 5905AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 5906 AudioStreamIn *input, 5907 audio_io_handle_t id, 5908 audio_devices_t outDevice, 5909 audio_devices_t inDevice, 5910 bool systemReady 5911#ifdef TEE_SINK 5912 , const sp<NBAIO_Sink>& teeSink 5913#endif 5914 ) : 5915 ThreadBase(audioFlinger, id, outDevice, inDevice, RECORD, systemReady), 5916 mInput(input), mActiveTracksGen(0), mRsmpInBuffer(NULL), 5917 // mRsmpInFrames and mRsmpInFramesP2 are set by readInputParameters_l() 5918 mRsmpInRear(0) 5919#ifdef TEE_SINK 5920 , mTeeSink(teeSink) 5921#endif 5922 , mReadOnlyHeap(new MemoryDealer(kRecordThreadReadOnlyHeapSize, 5923 "RecordThreadRO", MemoryHeapBase::READ_ONLY)) 5924 // mFastCapture below 5925 , mFastCaptureFutex(0) 5926 // mInputSource 5927 // mPipeSink 5928 // mPipeSource 5929 , mPipeFramesP2(0) 5930 // mPipeMemory 5931 // mFastCaptureNBLogWriter 5932 , mFastTrackAvail(false) 5933{ 5934 snprintf(mThreadName, kThreadNameLength, "AudioIn_%X", id); 5935 mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mThreadName); 5936 5937 readInputParameters_l(); 5938 5939 // create an NBAIO source for the HAL input stream, and negotiate 5940 mInputSource = new AudioStreamInSource(input->stream); 5941 size_t numCounterOffers = 0; 5942 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount, mFormat)}; 5943#if !LOG_NDEBUG 5944 ssize_t index = 5945#else 5946 (void) 5947#endif 5948 mInputSource->negotiate(offers, 1, NULL, numCounterOffers); 5949 ALOG_ASSERT(index == 0); 5950 5951 // initialize fast capture depending on configuration 5952 bool initFastCapture; 5953 switch (kUseFastCapture) { 5954 case FastCapture_Never: 5955 initFastCapture = false; 5956 break; 5957 case FastCapture_Always: 5958 initFastCapture = true; 5959 break; 5960 case FastCapture_Static: 5961 initFastCapture = (mFrameCount * 1000) / mSampleRate < kMinNormalCaptureBufferSizeMs; 5962 break; 5963 // case FastCapture_Dynamic: 5964 } 5965 5966 if (initFastCapture) { 5967 // create a Pipe for FastCapture to write to, and for us and fast tracks to read from 5968 NBAIO_Format format = mInputSource->format(); 5969 size_t pipeFramesP2 = roundup(mSampleRate / 25); // double-buffering of 20 ms each 5970 size_t pipeSize = pipeFramesP2 * Format_frameSize(format); 5971 void *pipeBuffer; 5972 const sp<MemoryDealer> roHeap(readOnlyHeap()); 5973 sp<IMemory> pipeMemory; 5974 if ((roHeap == 0) || 5975 (pipeMemory = roHeap->allocate(pipeSize)) == 0 || 5976 (pipeBuffer = pipeMemory->pointer()) == NULL) { 5977 ALOGE("not enough memory for pipe buffer size=%zu", pipeSize); 5978 goto failed; 5979 } 5980 // pipe will be shared directly with fast clients, so clear to avoid leaking old information 5981 memset(pipeBuffer, 0, pipeSize); 5982 Pipe *pipe = new Pipe(pipeFramesP2, format, pipeBuffer); 5983 const NBAIO_Format offers[1] = {format}; 5984 size_t numCounterOffers = 0; 5985 ssize_t index = pipe->negotiate(offers, 1, NULL, numCounterOffers); 5986 ALOG_ASSERT(index == 0); 5987 mPipeSink = pipe; 5988 PipeReader *pipeReader = new PipeReader(*pipe); 5989 numCounterOffers = 0; 5990 index = pipeReader->negotiate(offers, 1, NULL, numCounterOffers); 5991 ALOG_ASSERT(index == 0); 5992 mPipeSource = pipeReader; 5993 mPipeFramesP2 = pipeFramesP2; 5994 mPipeMemory = pipeMemory; 5995 5996 // create fast capture 5997 mFastCapture = new FastCapture(); 5998 FastCaptureStateQueue *sq = mFastCapture->sq(); 5999#ifdef STATE_QUEUE_DUMP 6000 // FIXME 6001#endif 6002 FastCaptureState *state = sq->begin(); 6003 state->mCblk = NULL; 6004 state->mInputSource = mInputSource.get(); 6005 state->mInputSourceGen++; 6006 state->mPipeSink = pipe; 6007 state->mPipeSinkGen++; 6008 state->mFrameCount = mFrameCount; 6009 state->mCommand = FastCaptureState::COLD_IDLE; 6010 // already done in constructor initialization list 6011 //mFastCaptureFutex = 0; 6012 state->mColdFutexAddr = &mFastCaptureFutex; 6013 state->mColdGen++; 6014 state->mDumpState = &mFastCaptureDumpState; 6015#ifdef TEE_SINK 6016 // FIXME 6017#endif 6018 mFastCaptureNBLogWriter = audioFlinger->newWriter_l(kFastCaptureLogSize, "FastCapture"); 6019 state->mNBLogWriter = mFastCaptureNBLogWriter.get(); 6020 sq->end(); 6021 sq->push(FastCaptureStateQueue::BLOCK_UNTIL_PUSHED); 6022 6023 // start the fast capture 6024 mFastCapture->run("FastCapture", ANDROID_PRIORITY_URGENT_AUDIO); 6025 pid_t tid = mFastCapture->getTid(); 6026 sendPrioConfigEvent(getpid_cached, tid, kPriorityFastCapture); 6027#ifdef AUDIO_WATCHDOG 6028 // FIXME 6029#endif 6030 6031 mFastTrackAvail = true; 6032 } 6033failed: ; 6034 6035 // FIXME mNormalSource 6036} 6037 6038AudioFlinger::RecordThread::~RecordThread() 6039{ 6040 if (mFastCapture != 0) { 6041 FastCaptureStateQueue *sq = mFastCapture->sq(); 6042 FastCaptureState *state = sq->begin(); 6043 if (state->mCommand == FastCaptureState::COLD_IDLE) { 6044 int32_t old = android_atomic_inc(&mFastCaptureFutex); 6045 if (old == -1) { 6046 (void) syscall(__NR_futex, &mFastCaptureFutex, FUTEX_WAKE_PRIVATE, 1); 6047 } 6048 } 6049 state->mCommand = FastCaptureState::EXIT; 6050 sq->end(); 6051 sq->push(FastCaptureStateQueue::BLOCK_UNTIL_PUSHED); 6052 mFastCapture->join(); 6053 mFastCapture.clear(); 6054 } 6055 mAudioFlinger->unregisterWriter(mFastCaptureNBLogWriter); 6056 mAudioFlinger->unregisterWriter(mNBLogWriter); 6057 free(mRsmpInBuffer); 6058} 6059 6060void AudioFlinger::RecordThread::onFirstRef() 6061{ 6062 run(mThreadName, PRIORITY_URGENT_AUDIO); 6063} 6064 6065bool AudioFlinger::RecordThread::threadLoop() 6066{ 6067 nsecs_t lastWarning = 0; 6068 6069 inputStandBy(); 6070 6071reacquire_wakelock: 6072 sp<RecordTrack> activeTrack; 6073 int activeTracksGen; 6074 { 6075 Mutex::Autolock _l(mLock); 6076 size_t size = mActiveTracks.size(); 6077 activeTracksGen = mActiveTracksGen; 6078 if (size > 0) { 6079 // FIXME an arbitrary choice 6080 activeTrack = mActiveTracks[0]; 6081 acquireWakeLock_l(activeTrack->uid()); 6082 if (size > 1) { 6083 SortedVector<int> tmp; 6084 for (size_t i = 0; i < size; i++) { 6085 tmp.add(mActiveTracks[i]->uid()); 6086 } 6087 updateWakeLockUids_l(tmp); 6088 } 6089 } else { 6090 acquireWakeLock_l(-1); 6091 } 6092 } 6093 6094 // used to request a deferred sleep, to be executed later while mutex is unlocked 6095 uint32_t sleepUs = 0; 6096 6097 // loop while there is work to do 6098 for (;;) { 6099 Vector< sp<EffectChain> > effectChains; 6100 6101 // activeTracks accumulates a copy of a subset of mActiveTracks 6102 Vector< sp<RecordTrack> > activeTracks; 6103 6104 // reference to the (first and only) active fast track 6105 sp<RecordTrack> fastTrack; 6106 6107 // reference to a fast track which is about to be removed 6108 sp<RecordTrack> fastTrackToRemove; 6109 6110 { // scope for mLock 6111 Mutex::Autolock _l(mLock); 6112 6113 processConfigEvents_l(); 6114 6115 // check exitPending here because checkForNewParameters_l() and 6116 // checkForNewParameters_l() can temporarily release mLock 6117 if (exitPending()) { 6118 break; 6119 } 6120 6121 // sleep with mutex unlocked 6122 if (sleepUs > 0) { 6123 ATRACE_BEGIN("sleepC"); 6124 mWaitWorkCV.waitRelative(mLock, microseconds((nsecs_t)sleepUs)); 6125 ATRACE_END(); 6126 sleepUs = 0; 6127 continue; 6128 } 6129 6130 // if no active track(s), then standby and release wakelock 6131 size_t size = mActiveTracks.size(); 6132 if (size == 0) { 6133 standbyIfNotAlreadyInStandby(); 6134 // exitPending() can't become true here 6135 releaseWakeLock_l(); 6136 ALOGV("RecordThread: loop stopping"); 6137 // go to sleep 6138 mWaitWorkCV.wait(mLock); 6139 ALOGV("RecordThread: loop starting"); 6140 goto reacquire_wakelock; 6141 } 6142 6143 if (mActiveTracksGen != activeTracksGen) { 6144 activeTracksGen = mActiveTracksGen; 6145 SortedVector<int> tmp; 6146 for (size_t i = 0; i < size; i++) { 6147 tmp.add(mActiveTracks[i]->uid()); 6148 } 6149 updateWakeLockUids_l(tmp); 6150 } 6151 6152 bool doBroadcast = false; 6153 bool allStopped = true; 6154 for (size_t i = 0; i < size; ) { 6155 6156 activeTrack = mActiveTracks[i]; 6157 if (activeTrack->isTerminated()) { 6158 if (activeTrack->isFastTrack()) { 6159 ALOG_ASSERT(fastTrackToRemove == 0); 6160 fastTrackToRemove = activeTrack; 6161 } 6162 removeTrack_l(activeTrack); 6163 mActiveTracks.remove(activeTrack); 6164 mActiveTracksGen++; 6165 size--; 6166 continue; 6167 } 6168 6169 TrackBase::track_state activeTrackState = activeTrack->mState; 6170 switch (activeTrackState) { 6171 6172 case TrackBase::PAUSING: 6173 mActiveTracks.remove(activeTrack); 6174 mActiveTracksGen++; 6175 doBroadcast = true; 6176 size--; 6177 continue; 6178 6179 case TrackBase::STARTING_1: 6180 sleepUs = 10000; 6181 i++; 6182 allStopped = false; 6183 continue; 6184 6185 case TrackBase::STARTING_2: 6186 doBroadcast = true; 6187 mStandby = false; 6188 activeTrack->mState = TrackBase::ACTIVE; 6189 allStopped = false; 6190 break; 6191 6192 case TrackBase::ACTIVE: 6193 allStopped = false; 6194 break; 6195 6196 case TrackBase::IDLE: 6197 i++; 6198 continue; 6199 6200 default: 6201 LOG_ALWAYS_FATAL("Unexpected activeTrackState %d", activeTrackState); 6202 } 6203 6204 activeTracks.add(activeTrack); 6205 i++; 6206 6207 if (activeTrack->isFastTrack()) { 6208 ALOG_ASSERT(!mFastTrackAvail); 6209 ALOG_ASSERT(fastTrack == 0); 6210 fastTrack = activeTrack; 6211 } 6212 } 6213 6214 if (allStopped) { 6215 standbyIfNotAlreadyInStandby(); 6216 } 6217 if (doBroadcast) { 6218 mStartStopCond.broadcast(); 6219 } 6220 6221 // sleep if there are no active tracks to process 6222 if (activeTracks.size() == 0) { 6223 if (sleepUs == 0) { 6224 sleepUs = kRecordThreadSleepUs; 6225 } 6226 continue; 6227 } 6228 sleepUs = 0; 6229 6230 lockEffectChains_l(effectChains); 6231 } 6232 6233 // thread mutex is now unlocked, mActiveTracks unknown, activeTracks.size() > 0 6234 6235 size_t size = effectChains.size(); 6236 for (size_t i = 0; i < size; i++) { 6237 // thread mutex is not locked, but effect chain is locked 6238 effectChains[i]->process_l(); 6239 } 6240 6241 // Push a new fast capture state if fast capture is not already running, or cblk change 6242 if (mFastCapture != 0) { 6243 FastCaptureStateQueue *sq = mFastCapture->sq(); 6244 FastCaptureState *state = sq->begin(); 6245 bool didModify = false; 6246 FastCaptureStateQueue::block_t block = FastCaptureStateQueue::BLOCK_UNTIL_PUSHED; 6247 if (state->mCommand != FastCaptureState::READ_WRITE /* FIXME && 6248 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)*/) { 6249 if (state->mCommand == FastCaptureState::COLD_IDLE) { 6250 int32_t old = android_atomic_inc(&mFastCaptureFutex); 6251 if (old == -1) { 6252 (void) syscall(__NR_futex, &mFastCaptureFutex, FUTEX_WAKE_PRIVATE, 1); 6253 } 6254 } 6255 state->mCommand = FastCaptureState::READ_WRITE; 6256#if 0 // FIXME 6257 mFastCaptureDumpState.increaseSamplingN(mAudioFlinger->isLowRamDevice() ? 6258 FastThreadDumpState::kSamplingNforLowRamDevice : 6259 FastThreadDumpState::kSamplingN); 6260#endif 6261 didModify = true; 6262 } 6263 audio_track_cblk_t *cblkOld = state->mCblk; 6264 audio_track_cblk_t *cblkNew = fastTrack != 0 ? fastTrack->cblk() : NULL; 6265 if (cblkNew != cblkOld) { 6266 state->mCblk = cblkNew; 6267 // block until acked if removing a fast track 6268 if (cblkOld != NULL) { 6269 block = FastCaptureStateQueue::BLOCK_UNTIL_ACKED; 6270 } 6271 didModify = true; 6272 } 6273 sq->end(didModify); 6274 if (didModify) { 6275 sq->push(block); 6276#if 0 6277 if (kUseFastCapture == FastCapture_Dynamic) { 6278 mNormalSource = mPipeSource; 6279 } 6280#endif 6281 } 6282 } 6283 6284 // now run the fast track destructor with thread mutex unlocked 6285 fastTrackToRemove.clear(); 6286 6287 // Read from HAL to keep up with fastest client if multiple active tracks, not slowest one. 6288 // Only the client(s) that are too slow will overrun. But if even the fastest client is too 6289 // slow, then this RecordThread will overrun by not calling HAL read often enough. 6290 // If destination is non-contiguous, first read past the nominal end of buffer, then 6291 // copy to the right place. Permitted because mRsmpInBuffer was over-allocated. 6292 6293 int32_t rear = mRsmpInRear & (mRsmpInFramesP2 - 1); 6294 ssize_t framesRead; 6295 6296 // If an NBAIO source is present, use it to read the normal capture's data 6297 if (mPipeSource != 0) { 6298 size_t framesToRead = mBufferSize / mFrameSize; 6299 framesRead = mPipeSource->read((uint8_t*)mRsmpInBuffer + rear * mFrameSize, 6300 framesToRead); 6301 if (framesRead == 0) { 6302 // since pipe is non-blocking, simulate blocking input 6303 sleepUs = (framesToRead * 1000000LL) / mSampleRate; 6304 } 6305 // otherwise use the HAL / AudioStreamIn directly 6306 } else { 6307 ATRACE_BEGIN("read"); 6308 ssize_t bytesRead = mInput->stream->read(mInput->stream, 6309 (uint8_t*)mRsmpInBuffer + rear * mFrameSize, mBufferSize); 6310 ATRACE_END(); 6311 if (bytesRead < 0) { 6312 framesRead = bytesRead; 6313 } else { 6314 framesRead = bytesRead / mFrameSize; 6315 } 6316 } 6317 6318 // Update server timestamp with server stats 6319 // systemTime() is optional if the hardware supports timestamps. 6320 mTimestamp.mPosition[ExtendedTimestamp::LOCATION_SERVER] += framesRead; 6321 mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_SERVER] = systemTime(); 6322 6323 // Update server timestamp with kernel stats 6324 if (mInput->stream->get_capture_position != nullptr 6325 && mPipeSource.get() == nullptr /* don't obtain for FastCapture, could block */) { 6326 int64_t position, time; 6327 int ret = mInput->stream->get_capture_position(mInput->stream, &position, &time); 6328 if (ret == NO_ERROR) { 6329 mTimestamp.mPosition[ExtendedTimestamp::LOCATION_KERNEL] = position; 6330 mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL] = time; 6331 // Note: In general record buffers should tend to be empty in 6332 // a properly running pipeline. 6333 // 6334 // Also, it is not advantageous to call get_presentation_position during the read 6335 // as the read obtains a lock, preventing the timestamp call from executing. 6336 } 6337 } 6338 // Use this to track timestamp information 6339 // ALOGD("%s", mTimestamp.toString().c_str()); 6340 6341 if (framesRead < 0 || (framesRead == 0 && mPipeSource == 0)) { 6342 ALOGE("read failed: framesRead=%zd", framesRead); 6343 // Force input into standby so that it tries to recover at next read attempt 6344 inputStandBy(); 6345 sleepUs = kRecordThreadSleepUs; 6346 } 6347 if (framesRead <= 0) { 6348 goto unlock; 6349 } 6350 ALOG_ASSERT(framesRead > 0); 6351 6352 if (mTeeSink != 0) { 6353 (void) mTeeSink->write((uint8_t*)mRsmpInBuffer + rear * mFrameSize, framesRead); 6354 } 6355 // If destination is non-contiguous, we now correct for reading past end of buffer. 6356 { 6357 size_t part1 = mRsmpInFramesP2 - rear; 6358 if ((size_t) framesRead > part1) { 6359 memcpy(mRsmpInBuffer, (uint8_t*)mRsmpInBuffer + mRsmpInFramesP2 * mFrameSize, 6360 (framesRead - part1) * mFrameSize); 6361 } 6362 } 6363 rear = mRsmpInRear += framesRead; 6364 6365 size = activeTracks.size(); 6366 // loop over each active track 6367 for (size_t i = 0; i < size; i++) { 6368 activeTrack = activeTracks[i]; 6369 6370 // skip fast tracks, as those are handled directly by FastCapture 6371 if (activeTrack->isFastTrack()) { 6372 continue; 6373 } 6374 6375 // TODO: This code probably should be moved to RecordTrack. 6376 // TODO: Update the activeTrack buffer converter in case of reconfigure. 6377 6378 enum { 6379 OVERRUN_UNKNOWN, 6380 OVERRUN_TRUE, 6381 OVERRUN_FALSE 6382 } overrun = OVERRUN_UNKNOWN; 6383 6384 // loop over getNextBuffer to handle circular sink 6385 for (;;) { 6386 6387 activeTrack->mSink.frameCount = ~0; 6388 status_t status = activeTrack->getNextBuffer(&activeTrack->mSink); 6389 size_t framesOut = activeTrack->mSink.frameCount; 6390 LOG_ALWAYS_FATAL_IF((status == OK) != (framesOut > 0)); 6391 6392 // check available frames and handle overrun conditions 6393 // if the record track isn't draining fast enough. 6394 bool hasOverrun; 6395 size_t framesIn; 6396 activeTrack->mResamplerBufferProvider->sync(&framesIn, &hasOverrun); 6397 if (hasOverrun) { 6398 overrun = OVERRUN_TRUE; 6399 } 6400 if (framesOut == 0 || framesIn == 0) { 6401 break; 6402 } 6403 6404 // Don't allow framesOut to be larger than what is possible with resampling 6405 // from framesIn. 6406 // This isn't strictly necessary but helps limit buffer resizing in 6407 // RecordBufferConverter. TODO: remove when no longer needed. 6408 framesOut = min(framesOut, 6409 destinationFramesPossible( 6410 framesIn, mSampleRate, activeTrack->mSampleRate)); 6411 // process frames from the RecordThread buffer provider to the RecordTrack buffer 6412 framesOut = activeTrack->mRecordBufferConverter->convert( 6413 activeTrack->mSink.raw, activeTrack->mResamplerBufferProvider, framesOut); 6414 6415 if (framesOut > 0 && (overrun == OVERRUN_UNKNOWN)) { 6416 overrun = OVERRUN_FALSE; 6417 } 6418 6419 if (activeTrack->mFramesToDrop == 0) { 6420 if (framesOut > 0) { 6421 activeTrack->mSink.frameCount = framesOut; 6422 activeTrack->releaseBuffer(&activeTrack->mSink); 6423 } 6424 } else { 6425 // FIXME could do a partial drop of framesOut 6426 if (activeTrack->mFramesToDrop > 0) { 6427 activeTrack->mFramesToDrop -= framesOut; 6428 if (activeTrack->mFramesToDrop <= 0) { 6429 activeTrack->clearSyncStartEvent(); 6430 } 6431 } else { 6432 activeTrack->mFramesToDrop += framesOut; 6433 if (activeTrack->mFramesToDrop >= 0 || activeTrack->mSyncStartEvent == 0 || 6434 activeTrack->mSyncStartEvent->isCancelled()) { 6435 ALOGW("Synced record %s, session %d, trigger session %d", 6436 (activeTrack->mFramesToDrop >= 0) ? "timed out" : "cancelled", 6437 activeTrack->sessionId(), 6438 (activeTrack->mSyncStartEvent != 0) ? 6439 activeTrack->mSyncStartEvent->triggerSession() : 6440 AUDIO_SESSION_NONE); 6441 activeTrack->clearSyncStartEvent(); 6442 } 6443 } 6444 } 6445 6446 if (framesOut == 0) { 6447 break; 6448 } 6449 } 6450 6451 switch (overrun) { 6452 case OVERRUN_TRUE: 6453 // client isn't retrieving buffers fast enough 6454 if (!activeTrack->setOverflow()) { 6455 nsecs_t now = systemTime(); 6456 // FIXME should lastWarning per track? 6457 if ((now - lastWarning) > kWarningThrottleNs) { 6458 ALOGW("RecordThread: buffer overflow"); 6459 lastWarning = now; 6460 } 6461 } 6462 break; 6463 case OVERRUN_FALSE: 6464 activeTrack->clearOverflow(); 6465 break; 6466 case OVERRUN_UNKNOWN: 6467 break; 6468 } 6469 6470 // update frame information and push timestamp out 6471 activeTrack->updateTrackFrameInfo( 6472 activeTrack->mServerProxy->framesReleased(), 6473 mTimestamp.mPosition[ExtendedTimestamp::LOCATION_SERVER], 6474 mSampleRate, mTimestamp); 6475 } 6476 6477unlock: 6478 // enable changes in effect chain 6479 unlockEffectChains(effectChains); 6480 // effectChains doesn't need to be cleared, since it is cleared by destructor at scope end 6481 } 6482 6483 standbyIfNotAlreadyInStandby(); 6484 6485 { 6486 Mutex::Autolock _l(mLock); 6487 for (size_t i = 0; i < mTracks.size(); i++) { 6488 sp<RecordTrack> track = mTracks[i]; 6489 track->invalidate(); 6490 } 6491 mActiveTracks.clear(); 6492 mActiveTracksGen++; 6493 mStartStopCond.broadcast(); 6494 } 6495 6496 releaseWakeLock(); 6497 6498 ALOGV("RecordThread %p exiting", this); 6499 return false; 6500} 6501 6502void AudioFlinger::RecordThread::standbyIfNotAlreadyInStandby() 6503{ 6504 if (!mStandby) { 6505 inputStandBy(); 6506 mStandby = true; 6507 } 6508} 6509 6510void AudioFlinger::RecordThread::inputStandBy() 6511{ 6512 // Idle the fast capture if it's currently running 6513 if (mFastCapture != 0) { 6514 FastCaptureStateQueue *sq = mFastCapture->sq(); 6515 FastCaptureState *state = sq->begin(); 6516 if (!(state->mCommand & FastCaptureState::IDLE)) { 6517 state->mCommand = FastCaptureState::COLD_IDLE; 6518 state->mColdFutexAddr = &mFastCaptureFutex; 6519 state->mColdGen++; 6520 mFastCaptureFutex = 0; 6521 sq->end(); 6522 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 6523 sq->push(FastCaptureStateQueue::BLOCK_UNTIL_ACKED); 6524#if 0 6525 if (kUseFastCapture == FastCapture_Dynamic) { 6526 // FIXME 6527 } 6528#endif 6529#ifdef AUDIO_WATCHDOG 6530 // FIXME 6531#endif 6532 } else { 6533 sq->end(false /*didModify*/); 6534 } 6535 } 6536 mInput->stream->common.standby(&mInput->stream->common); 6537} 6538 6539// RecordThread::createRecordTrack_l() must be called with AudioFlinger::mLock held 6540sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 6541 const sp<AudioFlinger::Client>& client, 6542 uint32_t sampleRate, 6543 audio_format_t format, 6544 audio_channel_mask_t channelMask, 6545 size_t *pFrameCount, 6546 audio_session_t sessionId, 6547 size_t *notificationFrames, 6548 int uid, 6549 audio_input_flags_t *flags, 6550 pid_t tid, 6551 status_t *status) 6552{ 6553 size_t frameCount = *pFrameCount; 6554 sp<RecordTrack> track; 6555 status_t lStatus; 6556 audio_input_flags_t inputFlags = mInput->flags; 6557 6558 // special case for FAST flag considered OK if fast capture is present 6559 if (hasFastCapture()) { 6560 inputFlags = (audio_input_flags_t)(inputFlags | AUDIO_INPUT_FLAG_FAST); 6561 } 6562 6563 // Check if requested flags are compatible with output stream flags 6564 if ((*flags & inputFlags) != *flags) { 6565 ALOGW("createRecordTrack_l(): mismatch between requested flags (%08x) and" 6566 " input flags (%08x)", 6567 *flags, inputFlags); 6568 *flags = (audio_input_flags_t)(*flags & inputFlags); 6569 } 6570 6571 // client expresses a preference for FAST, but we get the final say 6572 if (*flags & AUDIO_INPUT_FLAG_FAST) { 6573 if ( 6574 // we formerly checked for a callback handler (non-0 tid), 6575 // but that is no longer required for TRANSFER_OBTAIN mode 6576 // 6577 // frame count is not specified, or is exactly the pipe depth 6578 ((frameCount == 0) || (frameCount == mPipeFramesP2)) && 6579 // PCM data 6580 audio_is_linear_pcm(format) && 6581 // hardware format 6582 (format == mFormat) && 6583 // hardware channel mask 6584 (channelMask == mChannelMask) && 6585 // hardware sample rate 6586 (sampleRate == mSampleRate) && 6587 // record thread has an associated fast capture 6588 hasFastCapture() && 6589 // there are sufficient fast track slots available 6590 mFastTrackAvail 6591 ) { 6592 // check compatibility with audio effects. 6593 Mutex::Autolock _l(mLock); 6594 // Do not accept FAST flag if the session has software effects 6595 sp<EffectChain> chain = getEffectChain_l(sessionId); 6596 if (chain != 0) { 6597 ALOGV_IF((*flags & AUDIO_INPUT_FLAG_RAW) != 0, 6598 "AUDIO_INPUT_FLAG_RAW denied: effect present on session"); 6599 *flags = (audio_input_flags_t)(*flags & ~AUDIO_INPUT_FLAG_RAW); 6600 if (chain->hasSoftwareEffect()) { 6601 ALOGV("AUDIO_INPUT_FLAG_FAST denied: software effect present on session"); 6602 *flags = (audio_input_flags_t)(*flags & ~AUDIO_INPUT_FLAG_FAST); 6603 } 6604 } 6605 ALOGV_IF((*flags & AUDIO_INPUT_FLAG_FAST) != 0, 6606 "AUDIO_INPUT_FLAG_FAST accepted: frameCount=%zu mFrameCount=%zu", 6607 frameCount, mFrameCount); 6608 } else { 6609 ALOGV("AUDIO_INPUT_FLAG_FAST denied: frameCount=%zu mFrameCount=%zu mPipeFramesP2=%zu " 6610 "format=%#x isLinear=%d channelMask=%#x sampleRate=%u mSampleRate=%u " 6611 "hasFastCapture=%d tid=%d mFastTrackAvail=%d", 6612 frameCount, mFrameCount, mPipeFramesP2, 6613 format, audio_is_linear_pcm(format), channelMask, sampleRate, mSampleRate, 6614 hasFastCapture(), tid, mFastTrackAvail); 6615 *flags = (audio_input_flags_t)(*flags & ~AUDIO_INPUT_FLAG_FAST); 6616 } 6617 } 6618 6619 // compute track buffer size in frames, and suggest the notification frame count 6620 if (*flags & AUDIO_INPUT_FLAG_FAST) { 6621 // fast track: frame count is exactly the pipe depth 6622 frameCount = mPipeFramesP2; 6623 // ignore requested notificationFrames, and always notify exactly once every HAL buffer 6624 *notificationFrames = mFrameCount; 6625 } else { 6626 // not fast track: max notification period is resampled equivalent of one HAL buffer time 6627 // or 20 ms if there is a fast capture 6628 // TODO This could be a roundupRatio inline, and const 6629 size_t maxNotificationFrames = ((int64_t) (hasFastCapture() ? mSampleRate/50 : mFrameCount) 6630 * sampleRate + mSampleRate - 1) / mSampleRate; 6631 // minimum number of notification periods is at least kMinNotifications, 6632 // and at least kMinMs rounded up to a whole notification period (minNotificationsByMs) 6633 static const size_t kMinNotifications = 3; 6634 static const uint32_t kMinMs = 30; 6635 // TODO This could be a roundupRatio inline 6636 const size_t minFramesByMs = (sampleRate * kMinMs + 1000 - 1) / 1000; 6637 // TODO This could be a roundupRatio inline 6638 const size_t minNotificationsByMs = (minFramesByMs + maxNotificationFrames - 1) / 6639 maxNotificationFrames; 6640 const size_t minFrameCount = maxNotificationFrames * 6641 max(kMinNotifications, minNotificationsByMs); 6642 frameCount = max(frameCount, minFrameCount); 6643 if (*notificationFrames == 0 || *notificationFrames > maxNotificationFrames) { 6644 *notificationFrames = maxNotificationFrames; 6645 } 6646 } 6647 *pFrameCount = frameCount; 6648 6649 lStatus = initCheck(); 6650 if (lStatus != NO_ERROR) { 6651 ALOGE("createRecordTrack_l() audio driver not initialized"); 6652 goto Exit; 6653 } 6654 6655 { // scope for mLock 6656 Mutex::Autolock _l(mLock); 6657 6658 track = new RecordTrack(this, client, sampleRate, 6659 format, channelMask, frameCount, NULL, sessionId, uid, 6660 *flags, TrackBase::TYPE_DEFAULT); 6661 6662 lStatus = track->initCheck(); 6663 if (lStatus != NO_ERROR) { 6664 ALOGE("createRecordTrack_l() initCheck failed %d; no control block?", lStatus); 6665 // track must be cleared from the caller as the caller has the AF lock 6666 goto Exit; 6667 } 6668 mTracks.add(track); 6669 6670 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 6671 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 6672 mAudioFlinger->btNrecIsOff(); 6673 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 6674 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 6675 6676 if ((*flags & AUDIO_INPUT_FLAG_FAST) && (tid != -1)) { 6677 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 6678 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 6679 // so ask activity manager to do this on our behalf 6680 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 6681 } 6682 } 6683 6684 lStatus = NO_ERROR; 6685 6686Exit: 6687 *status = lStatus; 6688 return track; 6689} 6690 6691status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, 6692 AudioSystem::sync_event_t event, 6693 audio_session_t triggerSession) 6694{ 6695 ALOGV("RecordThread::start event %d, triggerSession %d", event, triggerSession); 6696 sp<ThreadBase> strongMe = this; 6697 status_t status = NO_ERROR; 6698 6699 if (event == AudioSystem::SYNC_EVENT_NONE) { 6700 recordTrack->clearSyncStartEvent(); 6701 } else if (event != AudioSystem::SYNC_EVENT_SAME) { 6702 recordTrack->mSyncStartEvent = mAudioFlinger->createSyncEvent(event, 6703 triggerSession, 6704 recordTrack->sessionId(), 6705 syncStartEventCallback, 6706 recordTrack); 6707 // Sync event can be cancelled by the trigger session if the track is not in a 6708 // compatible state in which case we start record immediately 6709 if (recordTrack->mSyncStartEvent->isCancelled()) { 6710 recordTrack->clearSyncStartEvent(); 6711 } else { 6712 // do not wait for the event for more than AudioSystem::kSyncRecordStartTimeOutMs 6713 recordTrack->mFramesToDrop = - 6714 ((AudioSystem::kSyncRecordStartTimeOutMs * recordTrack->mSampleRate) / 1000); 6715 } 6716 } 6717 6718 { 6719 // This section is a rendezvous between binder thread executing start() and RecordThread 6720 AutoMutex lock(mLock); 6721 if (mActiveTracks.indexOf(recordTrack) >= 0) { 6722 if (recordTrack->mState == TrackBase::PAUSING) { 6723 ALOGV("active record track PAUSING -> ACTIVE"); 6724 recordTrack->mState = TrackBase::ACTIVE; 6725 } else { 6726 ALOGV("active record track state %d", recordTrack->mState); 6727 } 6728 return status; 6729 } 6730 6731 // TODO consider other ways of handling this, such as changing the state to :STARTING and 6732 // adding the track to mActiveTracks after returning from AudioSystem::startInput(), 6733 // or using a separate command thread 6734 recordTrack->mState = TrackBase::STARTING_1; 6735 mActiveTracks.add(recordTrack); 6736 mActiveTracksGen++; 6737 status_t status = NO_ERROR; 6738 if (recordTrack->isExternalTrack()) { 6739 mLock.unlock(); 6740 status = AudioSystem::startInput(mId, recordTrack->sessionId()); 6741 mLock.lock(); 6742 // FIXME should verify that recordTrack is still in mActiveTracks 6743 if (status != NO_ERROR) { 6744 mActiveTracks.remove(recordTrack); 6745 mActiveTracksGen++; 6746 recordTrack->clearSyncStartEvent(); 6747 ALOGV("RecordThread::start error %d", status); 6748 return status; 6749 } 6750 } 6751 // Catch up with current buffer indices if thread is already running. 6752 // This is what makes a new client discard all buffered data. If the track's mRsmpInFront 6753 // was initialized to some value closer to the thread's mRsmpInFront, then the track could 6754 // see previously buffered data before it called start(), but with greater risk of overrun. 6755 6756 recordTrack->mResamplerBufferProvider->reset(); 6757 // clear any converter state as new data will be discontinuous 6758 recordTrack->mRecordBufferConverter->reset(); 6759 recordTrack->mState = TrackBase::STARTING_2; 6760 // signal thread to start 6761 mWaitWorkCV.broadcast(); 6762 if (mActiveTracks.indexOf(recordTrack) < 0) { 6763 ALOGV("Record failed to start"); 6764 status = BAD_VALUE; 6765 goto startError; 6766 } 6767 return status; 6768 } 6769 6770startError: 6771 if (recordTrack->isExternalTrack()) { 6772 AudioSystem::stopInput(mId, recordTrack->sessionId()); 6773 } 6774 recordTrack->clearSyncStartEvent(); 6775 // FIXME I wonder why we do not reset the state here? 6776 return status; 6777} 6778 6779void AudioFlinger::RecordThread::syncStartEventCallback(const wp<SyncEvent>& event) 6780{ 6781 sp<SyncEvent> strongEvent = event.promote(); 6782 6783 if (strongEvent != 0) { 6784 sp<RefBase> ptr = strongEvent->cookie().promote(); 6785 if (ptr != 0) { 6786 RecordTrack *recordTrack = (RecordTrack *)ptr.get(); 6787 recordTrack->handleSyncStartEvent(strongEvent); 6788 } 6789 } 6790} 6791 6792bool AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 6793 ALOGV("RecordThread::stop"); 6794 AutoMutex _l(mLock); 6795 if (mActiveTracks.indexOf(recordTrack) != 0 || recordTrack->mState == TrackBase::PAUSING) { 6796 return false; 6797 } 6798 // note that threadLoop may still be processing the track at this point [without lock] 6799 recordTrack->mState = TrackBase::PAUSING; 6800 // signal thread to stop 6801 mWaitWorkCV.broadcast(); 6802 // do not wait for mStartStopCond if exiting 6803 if (exitPending()) { 6804 return true; 6805 } 6806 // FIXME incorrect usage of wait: no explicit predicate or loop 6807 mStartStopCond.wait(mLock); 6808 // if we have been restarted, recordTrack is in mActiveTracks here 6809 if (exitPending() || mActiveTracks.indexOf(recordTrack) != 0) { 6810 ALOGV("Record stopped OK"); 6811 return true; 6812 } 6813 return false; 6814} 6815 6816bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event __unused) const 6817{ 6818 return false; 6819} 6820 6821status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event __unused) 6822{ 6823#if 0 // This branch is currently dead code, but is preserved in case it will be needed in future 6824 if (!isValidSyncEvent(event)) { 6825 return BAD_VALUE; 6826 } 6827 6828 audio_session_t eventSession = event->triggerSession(); 6829 status_t ret = NAME_NOT_FOUND; 6830 6831 Mutex::Autolock _l(mLock); 6832 6833 for (size_t i = 0; i < mTracks.size(); i++) { 6834 sp<RecordTrack> track = mTracks[i]; 6835 if (eventSession == track->sessionId()) { 6836 (void) track->setSyncEvent(event); 6837 ret = NO_ERROR; 6838 } 6839 } 6840 return ret; 6841#else 6842 return BAD_VALUE; 6843#endif 6844} 6845 6846// destroyTrack_l() must be called with ThreadBase::mLock held 6847void AudioFlinger::RecordThread::destroyTrack_l(const sp<RecordTrack>& track) 6848{ 6849 track->terminate(); 6850 track->mState = TrackBase::STOPPED; 6851 // active tracks are removed by threadLoop() 6852 if (mActiveTracks.indexOf(track) < 0) { 6853 removeTrack_l(track); 6854 } 6855} 6856 6857void AudioFlinger::RecordThread::removeTrack_l(const sp<RecordTrack>& track) 6858{ 6859 mTracks.remove(track); 6860 // need anything related to effects here? 6861 if (track->isFastTrack()) { 6862 ALOG_ASSERT(!mFastTrackAvail); 6863 mFastTrackAvail = true; 6864 } 6865} 6866 6867void AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 6868{ 6869 dumpInternals(fd, args); 6870 dumpTracks(fd, args); 6871 dumpEffectChains(fd, args); 6872} 6873 6874void AudioFlinger::RecordThread::dumpInternals(int fd, const Vector<String16>& args) 6875{ 6876 dprintf(fd, "\nInput thread %p:\n", this); 6877 6878 dumpBase(fd, args); 6879 6880 if (mActiveTracks.size() == 0) { 6881 dprintf(fd, " No active record clients\n"); 6882 } 6883 dprintf(fd, " Fast capture thread: %s\n", hasFastCapture() ? "yes" : "no"); 6884 dprintf(fd, " Fast track available: %s\n", mFastTrackAvail ? "yes" : "no"); 6885 6886 // Make a non-atomic copy of fast capture dump state so it won't change underneath us 6887 // while we are dumping it. It may be inconsistent, but it won't mutate! 6888 // This is a large object so we place it on the heap. 6889 // FIXME 25972958: Need an intelligent copy constructor that does not touch unused pages. 6890 const FastCaptureDumpState *copy = new FastCaptureDumpState(mFastCaptureDumpState); 6891 copy->dump(fd); 6892 delete copy; 6893} 6894 6895void AudioFlinger::RecordThread::dumpTracks(int fd, const Vector<String16>& args __unused) 6896{ 6897 const size_t SIZE = 256; 6898 char buffer[SIZE]; 6899 String8 result; 6900 6901 size_t numtracks = mTracks.size(); 6902 size_t numactive = mActiveTracks.size(); 6903 size_t numactiveseen = 0; 6904 dprintf(fd, " %zu Tracks", numtracks); 6905 if (numtracks) { 6906 dprintf(fd, " of which %zu are active\n", numactive); 6907 RecordTrack::appendDumpHeader(result); 6908 for (size_t i = 0; i < numtracks ; ++i) { 6909 sp<RecordTrack> track = mTracks[i]; 6910 if (track != 0) { 6911 bool active = mActiveTracks.indexOf(track) >= 0; 6912 if (active) { 6913 numactiveseen++; 6914 } 6915 track->dump(buffer, SIZE, active); 6916 result.append(buffer); 6917 } 6918 } 6919 } else { 6920 dprintf(fd, "\n"); 6921 } 6922 6923 if (numactiveseen != numactive) { 6924 snprintf(buffer, SIZE, " The following tracks are in the active list but" 6925 " not in the track list\n"); 6926 result.append(buffer); 6927 RecordTrack::appendDumpHeader(result); 6928 for (size_t i = 0; i < numactive; ++i) { 6929 sp<RecordTrack> track = mActiveTracks[i]; 6930 if (mTracks.indexOf(track) < 0) { 6931 track->dump(buffer, SIZE, true); 6932 result.append(buffer); 6933 } 6934 } 6935 6936 } 6937 write(fd, result.string(), result.size()); 6938} 6939 6940 6941void AudioFlinger::RecordThread::ResamplerBufferProvider::reset() 6942{ 6943 sp<ThreadBase> threadBase = mRecordTrack->mThread.promote(); 6944 RecordThread *recordThread = (RecordThread *) threadBase.get(); 6945 mRsmpInFront = recordThread->mRsmpInRear; 6946 mRsmpInUnrel = 0; 6947} 6948 6949void AudioFlinger::RecordThread::ResamplerBufferProvider::sync( 6950 size_t *framesAvailable, bool *hasOverrun) 6951{ 6952 sp<ThreadBase> threadBase = mRecordTrack->mThread.promote(); 6953 RecordThread *recordThread = (RecordThread *) threadBase.get(); 6954 const int32_t rear = recordThread->mRsmpInRear; 6955 const int32_t front = mRsmpInFront; 6956 const ssize_t filled = rear - front; 6957 6958 size_t framesIn; 6959 bool overrun = false; 6960 if (filled < 0) { 6961 // should not happen, but treat like a massive overrun and re-sync 6962 framesIn = 0; 6963 mRsmpInFront = rear; 6964 overrun = true; 6965 } else if ((size_t) filled <= recordThread->mRsmpInFrames) { 6966 framesIn = (size_t) filled; 6967 } else { 6968 // client is not keeping up with server, but give it latest data 6969 framesIn = recordThread->mRsmpInFrames; 6970 mRsmpInFront = /* front = */ rear - framesIn; 6971 overrun = true; 6972 } 6973 if (framesAvailable != NULL) { 6974 *framesAvailable = framesIn; 6975 } 6976 if (hasOverrun != NULL) { 6977 *hasOverrun = overrun; 6978 } 6979} 6980 6981// AudioBufferProvider interface 6982status_t AudioFlinger::RecordThread::ResamplerBufferProvider::getNextBuffer( 6983 AudioBufferProvider::Buffer* buffer) 6984{ 6985 sp<ThreadBase> threadBase = mRecordTrack->mThread.promote(); 6986 if (threadBase == 0) { 6987 buffer->frameCount = 0; 6988 buffer->raw = NULL; 6989 return NOT_ENOUGH_DATA; 6990 } 6991 RecordThread *recordThread = (RecordThread *) threadBase.get(); 6992 int32_t rear = recordThread->mRsmpInRear; 6993 int32_t front = mRsmpInFront; 6994 ssize_t filled = rear - front; 6995 // FIXME should not be P2 (don't want to increase latency) 6996 // FIXME if client not keeping up, discard 6997 LOG_ALWAYS_FATAL_IF(!(0 <= filled && (size_t) filled <= recordThread->mRsmpInFrames)); 6998 // 'filled' may be non-contiguous, so return only the first contiguous chunk 6999 front &= recordThread->mRsmpInFramesP2 - 1; 7000 size_t part1 = recordThread->mRsmpInFramesP2 - front; 7001 if (part1 > (size_t) filled) { 7002 part1 = filled; 7003 } 7004 size_t ask = buffer->frameCount; 7005 ALOG_ASSERT(ask > 0); 7006 if (part1 > ask) { 7007 part1 = ask; 7008 } 7009 if (part1 == 0) { 7010 // out of data is fine since the resampler will return a short-count. 7011 buffer->raw = NULL; 7012 buffer->frameCount = 0; 7013 mRsmpInUnrel = 0; 7014 return NOT_ENOUGH_DATA; 7015 } 7016 7017 buffer->raw = (uint8_t*)recordThread->mRsmpInBuffer + front * recordThread->mFrameSize; 7018 buffer->frameCount = part1; 7019 mRsmpInUnrel = part1; 7020 return NO_ERROR; 7021} 7022 7023// AudioBufferProvider interface 7024void AudioFlinger::RecordThread::ResamplerBufferProvider::releaseBuffer( 7025 AudioBufferProvider::Buffer* buffer) 7026{ 7027 size_t stepCount = buffer->frameCount; 7028 if (stepCount == 0) { 7029 return; 7030 } 7031 ALOG_ASSERT(stepCount <= mRsmpInUnrel); 7032 mRsmpInUnrel -= stepCount; 7033 mRsmpInFront += stepCount; 7034 buffer->raw = NULL; 7035 buffer->frameCount = 0; 7036} 7037 7038AudioFlinger::RecordThread::RecordBufferConverter::RecordBufferConverter( 7039 audio_channel_mask_t srcChannelMask, audio_format_t srcFormat, 7040 uint32_t srcSampleRate, 7041 audio_channel_mask_t dstChannelMask, audio_format_t dstFormat, 7042 uint32_t dstSampleRate) : 7043 mSrcChannelMask(AUDIO_CHANNEL_INVALID), // updateParameters will set following vars 7044 // mSrcFormat 7045 // mSrcSampleRate 7046 // mDstChannelMask 7047 // mDstFormat 7048 // mDstSampleRate 7049 // mSrcChannelCount 7050 // mDstChannelCount 7051 // mDstFrameSize 7052 mBuf(NULL), mBufFrames(0), mBufFrameSize(0), 7053 mResampler(NULL), 7054 mIsLegacyDownmix(false), 7055 mIsLegacyUpmix(false), 7056 mRequiresFloat(false), 7057 mInputConverterProvider(NULL) 7058{ 7059 (void)updateParameters(srcChannelMask, srcFormat, srcSampleRate, 7060 dstChannelMask, dstFormat, dstSampleRate); 7061} 7062 7063AudioFlinger::RecordThread::RecordBufferConverter::~RecordBufferConverter() { 7064 free(mBuf); 7065 delete mResampler; 7066 delete mInputConverterProvider; 7067} 7068 7069size_t AudioFlinger::RecordThread::RecordBufferConverter::convert(void *dst, 7070 AudioBufferProvider *provider, size_t frames) 7071{ 7072 if (mInputConverterProvider != NULL) { 7073 mInputConverterProvider->setBufferProvider(provider); 7074 provider = mInputConverterProvider; 7075 } 7076 7077 if (mResampler == NULL) { 7078 ALOGVV("NO RESAMPLING sampleRate:%u mSrcFormat:%#x mDstFormat:%#x", 7079 mSrcSampleRate, mSrcFormat, mDstFormat); 7080 7081 AudioBufferProvider::Buffer buffer; 7082 for (size_t i = frames; i > 0; ) { 7083 buffer.frameCount = i; 7084 status_t status = provider->getNextBuffer(&buffer); 7085 if (status != OK || buffer.frameCount == 0) { 7086 frames -= i; // cannot fill request. 7087 break; 7088 } 7089 // format convert to destination buffer 7090 convertNoResampler(dst, buffer.raw, buffer.frameCount); 7091 7092 dst = (int8_t*)dst + buffer.frameCount * mDstFrameSize; 7093 i -= buffer.frameCount; 7094 provider->releaseBuffer(&buffer); 7095 } 7096 } else { 7097 ALOGVV("RESAMPLING mSrcSampleRate:%u mDstSampleRate:%u mSrcFormat:%#x mDstFormat:%#x", 7098 mSrcSampleRate, mDstSampleRate, mSrcFormat, mDstFormat); 7099 7100 // reallocate buffer if needed 7101 if (mBufFrameSize != 0 && mBufFrames < frames) { 7102 free(mBuf); 7103 mBufFrames = frames; 7104 (void)posix_memalign(&mBuf, 32, mBufFrames * mBufFrameSize); 7105 } 7106 // resampler accumulates, but we only have one source track 7107 memset(mBuf, 0, frames * mBufFrameSize); 7108 frames = mResampler->resample((int32_t*)mBuf, frames, provider); 7109 // format convert to destination buffer 7110 convertResampler(dst, mBuf, frames); 7111 } 7112 return frames; 7113} 7114 7115status_t AudioFlinger::RecordThread::RecordBufferConverter::updateParameters( 7116 audio_channel_mask_t srcChannelMask, audio_format_t srcFormat, 7117 uint32_t srcSampleRate, 7118 audio_channel_mask_t dstChannelMask, audio_format_t dstFormat, 7119 uint32_t dstSampleRate) 7120{ 7121 // quick evaluation if there is any change. 7122 if (mSrcFormat == srcFormat 7123 && mSrcChannelMask == srcChannelMask 7124 && mSrcSampleRate == srcSampleRate 7125 && mDstFormat == dstFormat 7126 && mDstChannelMask == dstChannelMask 7127 && mDstSampleRate == dstSampleRate) { 7128 return NO_ERROR; 7129 } 7130 7131 ALOGV("RecordBufferConverter updateParameters srcMask:%#x dstMask:%#x" 7132 " srcFormat:%#x dstFormat:%#x srcRate:%u dstRate:%u", 7133 srcChannelMask, dstChannelMask, srcFormat, dstFormat, srcSampleRate, dstSampleRate); 7134 const bool valid = 7135 audio_is_input_channel(srcChannelMask) 7136 && audio_is_input_channel(dstChannelMask) 7137 && audio_is_valid_format(srcFormat) && audio_is_linear_pcm(srcFormat) 7138 && audio_is_valid_format(dstFormat) && audio_is_linear_pcm(dstFormat) 7139 && (srcSampleRate <= dstSampleRate * AUDIO_RESAMPLER_DOWN_RATIO_MAX) 7140 ; // no upsampling checks for now 7141 if (!valid) { 7142 return BAD_VALUE; 7143 } 7144 7145 mSrcFormat = srcFormat; 7146 mSrcChannelMask = srcChannelMask; 7147 mSrcSampleRate = srcSampleRate; 7148 mDstFormat = dstFormat; 7149 mDstChannelMask = dstChannelMask; 7150 mDstSampleRate = dstSampleRate; 7151 7152 // compute derived parameters 7153 mSrcChannelCount = audio_channel_count_from_in_mask(srcChannelMask); 7154 mDstChannelCount = audio_channel_count_from_in_mask(dstChannelMask); 7155 mDstFrameSize = mDstChannelCount * audio_bytes_per_sample(mDstFormat); 7156 7157 // do we need to resample? 7158 delete mResampler; 7159 mResampler = NULL; 7160 if (mSrcSampleRate != mDstSampleRate) { 7161 mResampler = AudioResampler::create(AUDIO_FORMAT_PCM_FLOAT, 7162 mSrcChannelCount, mDstSampleRate); 7163 mResampler->setSampleRate(mSrcSampleRate); 7164 mResampler->setVolume(AudioMixer::UNITY_GAIN_FLOAT, AudioMixer::UNITY_GAIN_FLOAT); 7165 } 7166 7167 // are we running legacy channel conversion modes? 7168 mIsLegacyDownmix = (mSrcChannelMask == AUDIO_CHANNEL_IN_STEREO 7169 || mSrcChannelMask == AUDIO_CHANNEL_IN_FRONT_BACK) 7170 && mDstChannelMask == AUDIO_CHANNEL_IN_MONO; 7171 mIsLegacyUpmix = mSrcChannelMask == AUDIO_CHANNEL_IN_MONO 7172 && (mDstChannelMask == AUDIO_CHANNEL_IN_STEREO 7173 || mDstChannelMask == AUDIO_CHANNEL_IN_FRONT_BACK); 7174 7175 // do we need to process in float? 7176 mRequiresFloat = mResampler != NULL || mIsLegacyDownmix || mIsLegacyUpmix; 7177 7178 // do we need a staging buffer to convert for destination (we can still optimize this)? 7179 // we use mBufFrameSize > 0 to indicate both frame size as well as buffer necessity 7180 if (mResampler != NULL) { 7181 mBufFrameSize = max(mSrcChannelCount, FCC_2) 7182 * audio_bytes_per_sample(AUDIO_FORMAT_PCM_FLOAT); 7183 } else if (mIsLegacyUpmix || mIsLegacyDownmix) { // legacy modes always float 7184 mBufFrameSize = mDstChannelCount * audio_bytes_per_sample(AUDIO_FORMAT_PCM_FLOAT); 7185 } else if (mSrcChannelMask != mDstChannelMask && mDstFormat != mSrcFormat) { 7186 mBufFrameSize = mDstChannelCount * audio_bytes_per_sample(mSrcFormat); 7187 } else { 7188 mBufFrameSize = 0; 7189 } 7190 mBufFrames = 0; // force the buffer to be resized. 7191 7192 // do we need an input converter buffer provider to give us float? 7193 delete mInputConverterProvider; 7194 mInputConverterProvider = NULL; 7195 if (mRequiresFloat && mSrcFormat != AUDIO_FORMAT_PCM_FLOAT) { 7196 mInputConverterProvider = new ReformatBufferProvider( 7197 audio_channel_count_from_in_mask(mSrcChannelMask), 7198 mSrcFormat, 7199 AUDIO_FORMAT_PCM_FLOAT, 7200 256 /* provider buffer frame count */); 7201 } 7202 7203 // do we need a remixer to do channel mask conversion 7204 if (!mIsLegacyDownmix && !mIsLegacyUpmix && mSrcChannelMask != mDstChannelMask) { 7205 (void) memcpy_by_index_array_initialization_from_channel_mask( 7206 mIdxAry, ARRAY_SIZE(mIdxAry), mDstChannelMask, mSrcChannelMask); 7207 } 7208 return NO_ERROR; 7209} 7210 7211void AudioFlinger::RecordThread::RecordBufferConverter::convertNoResampler( 7212 void *dst, const void *src, size_t frames) 7213{ 7214 // src is native type unless there is legacy upmix or downmix, whereupon it is float. 7215 if (mBufFrameSize != 0 && mBufFrames < frames) { 7216 free(mBuf); 7217 mBufFrames = frames; 7218 (void)posix_memalign(&mBuf, 32, mBufFrames * mBufFrameSize); 7219 } 7220 // do we need to do legacy upmix and downmix? 7221 if (mIsLegacyUpmix || mIsLegacyDownmix) { 7222 void *dstBuf = mBuf != NULL ? mBuf : dst; 7223 if (mIsLegacyUpmix) { 7224 upmix_to_stereo_float_from_mono_float((float *)dstBuf, 7225 (const float *)src, frames); 7226 } else /*mIsLegacyDownmix */ { 7227 downmix_to_mono_float_from_stereo_float((float *)dstBuf, 7228 (const float *)src, frames); 7229 } 7230 if (mBuf != NULL) { 7231 memcpy_by_audio_format(dst, mDstFormat, mBuf, AUDIO_FORMAT_PCM_FLOAT, 7232 frames * mDstChannelCount); 7233 } 7234 return; 7235 } 7236 // do we need to do channel mask conversion? 7237 if (mSrcChannelMask != mDstChannelMask) { 7238 void *dstBuf = mBuf != NULL ? mBuf : dst; 7239 memcpy_by_index_array(dstBuf, mDstChannelCount, 7240 src, mSrcChannelCount, mIdxAry, audio_bytes_per_sample(mSrcFormat), frames); 7241 if (dstBuf == dst) { 7242 return; // format is the same 7243 } 7244 } 7245 // convert to destination buffer 7246 const void *convertBuf = mBuf != NULL ? mBuf : src; 7247 memcpy_by_audio_format(dst, mDstFormat, convertBuf, mSrcFormat, 7248 frames * mDstChannelCount); 7249} 7250 7251void AudioFlinger::RecordThread::RecordBufferConverter::convertResampler( 7252 void *dst, /*not-a-const*/ void *src, size_t frames) 7253{ 7254 // src buffer format is ALWAYS float when entering this routine 7255 if (mIsLegacyUpmix) { 7256 ; // mono to stereo already handled by resampler 7257 } else if (mIsLegacyDownmix 7258 || (mSrcChannelMask == mDstChannelMask && mSrcChannelCount == 1)) { 7259 // the resampler outputs stereo for mono input channel (a feature?) 7260 // must convert to mono 7261 downmix_to_mono_float_from_stereo_float((float *)src, 7262 (const float *)src, frames); 7263 } else if (mSrcChannelMask != mDstChannelMask) { 7264 // convert to mono channel again for channel mask conversion (could be skipped 7265 // with further optimization). 7266 if (mSrcChannelCount == 1) { 7267 downmix_to_mono_float_from_stereo_float((float *)src, 7268 (const float *)src, frames); 7269 } 7270 // convert to destination format (in place, OK as float is larger than other types) 7271 if (mDstFormat != AUDIO_FORMAT_PCM_FLOAT) { 7272 memcpy_by_audio_format(src, mDstFormat, src, AUDIO_FORMAT_PCM_FLOAT, 7273 frames * mSrcChannelCount); 7274 } 7275 // channel convert and save to dst 7276 memcpy_by_index_array(dst, mDstChannelCount, 7277 src, mSrcChannelCount, mIdxAry, audio_bytes_per_sample(mDstFormat), frames); 7278 return; 7279 } 7280 // convert to destination format and save to dst 7281 memcpy_by_audio_format(dst, mDstFormat, src, AUDIO_FORMAT_PCM_FLOAT, 7282 frames * mDstChannelCount); 7283} 7284 7285bool AudioFlinger::RecordThread::checkForNewParameter_l(const String8& keyValuePair, 7286 status_t& status) 7287{ 7288 bool reconfig = false; 7289 7290 status = NO_ERROR; 7291 7292 audio_format_t reqFormat = mFormat; 7293 uint32_t samplingRate = mSampleRate; 7294 // TODO this may change if we want to support capture from HDMI PCM multi channel (e.g on TVs). 7295 audio_channel_mask_t channelMask = audio_channel_in_mask_from_count(mChannelCount); 7296 7297 AudioParameter param = AudioParameter(keyValuePair); 7298 int value; 7299 7300 // scope for AutoPark extends to end of method 7301 AutoPark<FastCapture> park(mFastCapture); 7302 7303 // TODO Investigate when this code runs. Check with audio policy when a sample rate and 7304 // channel count change can be requested. Do we mandate the first client defines the 7305 // HAL sampling rate and channel count or do we allow changes on the fly? 7306 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 7307 samplingRate = value; 7308 reconfig = true; 7309 } 7310 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 7311 if (!audio_is_linear_pcm((audio_format_t) value)) { 7312 status = BAD_VALUE; 7313 } else { 7314 reqFormat = (audio_format_t) value; 7315 reconfig = true; 7316 } 7317 } 7318 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 7319 audio_channel_mask_t mask = (audio_channel_mask_t) value; 7320 if (!audio_is_input_channel(mask) || 7321 audio_channel_count_from_in_mask(mask) > FCC_8) { 7322 status = BAD_VALUE; 7323 } else { 7324 channelMask = mask; 7325 reconfig = true; 7326 } 7327 } 7328 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 7329 // do not accept frame count changes if tracks are open as the track buffer 7330 // size depends on frame count and correct behavior would not be guaranteed 7331 // if frame count is changed after track creation 7332 if (mActiveTracks.size() > 0) { 7333 status = INVALID_OPERATION; 7334 } else { 7335 reconfig = true; 7336 } 7337 } 7338 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 7339 // forward device change to effects that have requested to be 7340 // aware of attached audio device. 7341 for (size_t i = 0; i < mEffectChains.size(); i++) { 7342 mEffectChains[i]->setDevice_l(value); 7343 } 7344 7345 // store input device and output device but do not forward output device to audio HAL. 7346 // Note that status is ignored by the caller for output device 7347 // (see AudioFlinger::setParameters() 7348 if (audio_is_output_devices(value)) { 7349 mOutDevice = value; 7350 status = BAD_VALUE; 7351 } else { 7352 mInDevice = value; 7353 if (value != AUDIO_DEVICE_NONE) { 7354 mPrevInDevice = value; 7355 } 7356 // disable AEC and NS if the device is a BT SCO headset supporting those 7357 // pre processings 7358 if (mTracks.size() > 0) { 7359 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 7360 mAudioFlinger->btNrecIsOff(); 7361 for (size_t i = 0; i < mTracks.size(); i++) { 7362 sp<RecordTrack> track = mTracks[i]; 7363 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId()); 7364 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId()); 7365 } 7366 } 7367 } 7368 } 7369 if (param.getInt(String8(AudioParameter::keyInputSource), value) == NO_ERROR && 7370 mAudioSource != (audio_source_t)value) { 7371 // forward device change to effects that have requested to be 7372 // aware of attached audio device. 7373 for (size_t i = 0; i < mEffectChains.size(); i++) { 7374 mEffectChains[i]->setAudioSource_l((audio_source_t)value); 7375 } 7376 mAudioSource = (audio_source_t)value; 7377 } 7378 7379 if (status == NO_ERROR) { 7380 status = mInput->stream->common.set_parameters(&mInput->stream->common, 7381 keyValuePair.string()); 7382 if (status == INVALID_OPERATION) { 7383 inputStandBy(); 7384 status = mInput->stream->common.set_parameters(&mInput->stream->common, 7385 keyValuePair.string()); 7386 } 7387 if (reconfig) { 7388 if (status == BAD_VALUE && 7389 audio_is_linear_pcm(mInput->stream->common.get_format(&mInput->stream->common)) && 7390 audio_is_linear_pcm(reqFormat) && 7391 (mInput->stream->common.get_sample_rate(&mInput->stream->common) 7392 <= (AUDIO_RESAMPLER_DOWN_RATIO_MAX * samplingRate)) && 7393 audio_channel_count_from_in_mask( 7394 mInput->stream->common.get_channels(&mInput->stream->common)) <= FCC_8) { 7395 status = NO_ERROR; 7396 } 7397 if (status == NO_ERROR) { 7398 readInputParameters_l(); 7399 sendIoConfigEvent_l(AUDIO_INPUT_CONFIG_CHANGED); 7400 } 7401 } 7402 } 7403 7404 return reconfig; 7405} 7406 7407String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 7408{ 7409 Mutex::Autolock _l(mLock); 7410 if (initCheck() != NO_ERROR) { 7411 return String8(); 7412 } 7413 7414 char *s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 7415 const String8 out_s8(s); 7416 free(s); 7417 return out_s8; 7418} 7419 7420void AudioFlinger::RecordThread::ioConfigChanged(audio_io_config_event event, pid_t pid) { 7421 sp<AudioIoDescriptor> desc = new AudioIoDescriptor(); 7422 7423 desc->mIoHandle = mId; 7424 7425 switch (event) { 7426 case AUDIO_INPUT_OPENED: 7427 case AUDIO_INPUT_CONFIG_CHANGED: 7428 desc->mPatch = mPatch; 7429 desc->mChannelMask = mChannelMask; 7430 desc->mSamplingRate = mSampleRate; 7431 desc->mFormat = mFormat; 7432 desc->mFrameCount = mFrameCount; 7433 desc->mFrameCountHAL = mFrameCount; 7434 desc->mLatency = 0; 7435 break; 7436 7437 case AUDIO_INPUT_CLOSED: 7438 default: 7439 break; 7440 } 7441 mAudioFlinger->ioConfigChanged(event, desc, pid); 7442} 7443 7444void AudioFlinger::RecordThread::readInputParameters_l() 7445{ 7446 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 7447 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 7448 mChannelCount = audio_channel_count_from_in_mask(mChannelMask); 7449 if (mChannelCount > FCC_8) { 7450 ALOGE("HAL channel count %d > %d", mChannelCount, FCC_8); 7451 } 7452 mHALFormat = mInput->stream->common.get_format(&mInput->stream->common); 7453 mFormat = mHALFormat; 7454 if (!audio_is_linear_pcm(mFormat)) { 7455 ALOGE("HAL format %#x is not linear pcm", mFormat); 7456 } 7457 mFrameSize = audio_stream_in_frame_size(mInput->stream); 7458 mBufferSize = mInput->stream->common.get_buffer_size(&mInput->stream->common); 7459 mFrameCount = mBufferSize / mFrameSize; 7460 // This is the formula for calculating the temporary buffer size. 7461 // With 7 HAL buffers, we can guarantee ability to down-sample the input by ratio of 6:1 to 7462 // 1 full output buffer, regardless of the alignment of the available input. 7463 // The value is somewhat arbitrary, and could probably be even larger. 7464 // A larger value should allow more old data to be read after a track calls start(), 7465 // without increasing latency. 7466 // 7467 // Note this is independent of the maximum downsampling ratio permitted for capture. 7468 mRsmpInFrames = mFrameCount * 7; 7469 mRsmpInFramesP2 = roundup(mRsmpInFrames); 7470 free(mRsmpInBuffer); 7471 mRsmpInBuffer = NULL; 7472 7473 // TODO optimize audio capture buffer sizes ... 7474 // Here we calculate the size of the sliding buffer used as a source 7475 // for resampling. mRsmpInFramesP2 is currently roundup(mFrameCount * 7). 7476 // For current HAL frame counts, this is usually 2048 = 40 ms. It would 7477 // be better to have it derived from the pipe depth in the long term. 7478 // The current value is higher than necessary. However it should not add to latency. 7479 7480 // Over-allocate beyond mRsmpInFramesP2 to permit a HAL read past end of buffer 7481 size_t bufferSize = (mRsmpInFramesP2 + mFrameCount - 1) * mFrameSize; 7482 (void)posix_memalign(&mRsmpInBuffer, 32, bufferSize); 7483 memset(mRsmpInBuffer, 0, bufferSize); // if posix_memalign fails, will segv here. 7484 7485 // AudioRecord mSampleRate and mChannelCount are constant due to AudioRecord API constraints. 7486 // But if thread's mSampleRate or mChannelCount changes, how will that affect active tracks? 7487} 7488 7489uint32_t AudioFlinger::RecordThread::getInputFramesLost() 7490{ 7491 Mutex::Autolock _l(mLock); 7492 if (initCheck() != NO_ERROR) { 7493 return 0; 7494 } 7495 7496 return mInput->stream->get_input_frames_lost(mInput->stream); 7497} 7498 7499// hasAudioSession_l() must be called with ThreadBase::mLock held 7500uint32_t AudioFlinger::RecordThread::hasAudioSession_l(audio_session_t sessionId) const 7501{ 7502 uint32_t result = 0; 7503 if (getEffectChain_l(sessionId) != 0) { 7504 result = EFFECT_SESSION; 7505 } 7506 7507 for (size_t i = 0; i < mTracks.size(); ++i) { 7508 if (sessionId == mTracks[i]->sessionId()) { 7509 result |= TRACK_SESSION; 7510 if (mTracks[i]->isFastTrack()) { 7511 result |= FAST_SESSION; 7512 } 7513 break; 7514 } 7515 } 7516 7517 return result; 7518} 7519 7520KeyedVector<audio_session_t, bool> AudioFlinger::RecordThread::sessionIds() const 7521{ 7522 KeyedVector<audio_session_t, bool> ids; 7523 Mutex::Autolock _l(mLock); 7524 for (size_t j = 0; j < mTracks.size(); ++j) { 7525 sp<RecordThread::RecordTrack> track = mTracks[j]; 7526 audio_session_t sessionId = track->sessionId(); 7527 if (ids.indexOfKey(sessionId) < 0) { 7528 ids.add(sessionId, true); 7529 } 7530 } 7531 return ids; 7532} 7533 7534AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 7535{ 7536 Mutex::Autolock _l(mLock); 7537 AudioStreamIn *input = mInput; 7538 mInput = NULL; 7539 return input; 7540} 7541 7542// this method must always be called either with ThreadBase mLock held or inside the thread loop 7543audio_stream_t* AudioFlinger::RecordThread::stream() const 7544{ 7545 if (mInput == NULL) { 7546 return NULL; 7547 } 7548 return &mInput->stream->common; 7549} 7550 7551status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 7552{ 7553 // only one chain per input thread 7554 if (mEffectChains.size() != 0) { 7555 ALOGW("addEffectChain_l() already one chain %p on thread %p", chain.get(), this); 7556 return INVALID_OPERATION; 7557 } 7558 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 7559 chain->setThread(this); 7560 chain->setInBuffer(NULL); 7561 chain->setOutBuffer(NULL); 7562 7563 checkSuspendOnAddEffectChain_l(chain); 7564 7565 // make sure enabled pre processing effects state is communicated to the HAL as we 7566 // just moved them to a new input stream. 7567 chain->syncHalEffectsState(); 7568 7569 mEffectChains.add(chain); 7570 7571 return NO_ERROR; 7572} 7573 7574size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 7575{ 7576 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 7577 ALOGW_IF(mEffectChains.size() != 1, 7578 "removeEffectChain_l() %p invalid chain size %zu on thread %p", 7579 chain.get(), mEffectChains.size(), this); 7580 if (mEffectChains.size() == 1) { 7581 mEffectChains.removeAt(0); 7582 } 7583 return 0; 7584} 7585 7586status_t AudioFlinger::RecordThread::createAudioPatch_l(const struct audio_patch *patch, 7587 audio_patch_handle_t *handle) 7588{ 7589 status_t status = NO_ERROR; 7590 7591 // store new device and send to effects 7592 mInDevice = patch->sources[0].ext.device.type; 7593 mPatch = *patch; 7594 for (size_t i = 0; i < mEffectChains.size(); i++) { 7595 mEffectChains[i]->setDevice_l(mInDevice); 7596 } 7597 7598 // disable AEC and NS if the device is a BT SCO headset supporting those 7599 // pre processings 7600 if (mTracks.size() > 0) { 7601 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 7602 mAudioFlinger->btNrecIsOff(); 7603 for (size_t i = 0; i < mTracks.size(); i++) { 7604 sp<RecordTrack> track = mTracks[i]; 7605 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId()); 7606 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId()); 7607 } 7608 } 7609 7610 // store new source and send to effects 7611 if (mAudioSource != patch->sinks[0].ext.mix.usecase.source) { 7612 mAudioSource = patch->sinks[0].ext.mix.usecase.source; 7613 for (size_t i = 0; i < mEffectChains.size(); i++) { 7614 mEffectChains[i]->setAudioSource_l(mAudioSource); 7615 } 7616 } 7617 7618 if (mInput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 7619 audio_hw_device_t *hwDevice = mInput->audioHwDev->hwDevice(); 7620 status = hwDevice->create_audio_patch(hwDevice, 7621 patch->num_sources, 7622 patch->sources, 7623 patch->num_sinks, 7624 patch->sinks, 7625 handle); 7626 } else { 7627 char *address; 7628 if (strcmp(patch->sources[0].ext.device.address, "") != 0) { 7629 address = audio_device_address_to_parameter( 7630 patch->sources[0].ext.device.type, 7631 patch->sources[0].ext.device.address); 7632 } else { 7633 address = (char *)calloc(1, 1); 7634 } 7635 AudioParameter param = AudioParameter(String8(address)); 7636 free(address); 7637 param.addInt(String8(AUDIO_PARAMETER_STREAM_ROUTING), 7638 (int)patch->sources[0].ext.device.type); 7639 param.addInt(String8(AUDIO_PARAMETER_STREAM_INPUT_SOURCE), 7640 (int)patch->sinks[0].ext.mix.usecase.source); 7641 status = mInput->stream->common.set_parameters(&mInput->stream->common, 7642 param.toString().string()); 7643 *handle = AUDIO_PATCH_HANDLE_NONE; 7644 } 7645 7646 if (mInDevice != mPrevInDevice) { 7647 sendIoConfigEvent_l(AUDIO_INPUT_CONFIG_CHANGED); 7648 mPrevInDevice = mInDevice; 7649 } 7650 7651 return status; 7652} 7653 7654status_t AudioFlinger::RecordThread::releaseAudioPatch_l(const audio_patch_handle_t handle) 7655{ 7656 status_t status = NO_ERROR; 7657 7658 mInDevice = AUDIO_DEVICE_NONE; 7659 7660 if (mInput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 7661 audio_hw_device_t *hwDevice = mInput->audioHwDev->hwDevice(); 7662 status = hwDevice->release_audio_patch(hwDevice, handle); 7663 } else { 7664 AudioParameter param; 7665 param.addInt(String8(AUDIO_PARAMETER_STREAM_ROUTING), 0); 7666 status = mInput->stream->common.set_parameters(&mInput->stream->common, 7667 param.toString().string()); 7668 } 7669 return status; 7670} 7671 7672void AudioFlinger::RecordThread::addPatchRecord(const sp<PatchRecord>& record) 7673{ 7674 Mutex::Autolock _l(mLock); 7675 mTracks.add(record); 7676} 7677 7678void AudioFlinger::RecordThread::deletePatchRecord(const sp<PatchRecord>& record) 7679{ 7680 Mutex::Autolock _l(mLock); 7681 destroyTrack_l(record); 7682} 7683 7684void AudioFlinger::RecordThread::getAudioPortConfig(struct audio_port_config *config) 7685{ 7686 ThreadBase::getAudioPortConfig(config); 7687 config->role = AUDIO_PORT_ROLE_SINK; 7688 config->ext.mix.hw_module = mInput->audioHwDev->handle(); 7689 config->ext.mix.usecase.source = mAudioSource; 7690} 7691 7692} // namespace android 7693