Threads.cpp revision cecb30a22f3657483f07f259231b81b02b2a7305
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 audio_input_flags_t flags = mInput->flags; 1276 if (hasFastCapture() || (flags & AUDIO_INPUT_FLAG_FAST)) { 1277 if (flags & AUDIO_INPUT_FLAG_RAW) { 1278 ALOGW("checkEffectCompatibility_l(): effect %s on record thread %s in raw mode", 1279 desc->name, mThreadName); 1280 return BAD_VALUE; 1281 } 1282 if ((desc->flags & EFFECT_FLAG_HW_ACC_TUNNEL) == 0) { 1283 ALOGW("checkEffectCompatibility_l(): non HW effect %s on record thread %s in fast mode", 1284 desc->name, mThreadName); 1285 return BAD_VALUE; 1286 } 1287 } 1288 return NO_ERROR; 1289} 1290 1291// checkEffectCompatibility_l() must be called with ThreadBase::mLock held 1292status_t AudioFlinger::PlaybackThread::checkEffectCompatibility_l( 1293 const effect_descriptor_t *desc, audio_session_t sessionId) 1294{ 1295 // no preprocessing on playback threads 1296 if ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC) { 1297 ALOGW("checkEffectCompatibility_l(): pre processing effect %s created on playback" 1298 " thread %s", desc->name, mThreadName); 1299 return BAD_VALUE; 1300 } 1301 1302 switch (mType) { 1303 case MIXER: { 1304 // Reject any effect on mixer multichannel sinks. 1305 // TODO: fix both format and multichannel issues with effects. 1306 if (mChannelCount != FCC_2) { 1307 ALOGW("checkEffectCompatibility_l(): effect %s for multichannel(%d) on MIXER" 1308 " thread %s", desc->name, mChannelCount, mThreadName); 1309 return BAD_VALUE; 1310 } 1311 audio_output_flags_t flags = mOutput->flags; 1312 if (hasFastMixer() || (flags & AUDIO_OUTPUT_FLAG_FAST)) { 1313 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 1314 // global effects are applied only to non fast tracks if they are SW 1315 if ((desc->flags & EFFECT_FLAG_HW_ACC_TUNNEL) == 0) { 1316 break; 1317 } 1318 } else if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) { 1319 // only post processing on output stage session 1320 if ((desc->flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_POST_PROC) { 1321 ALOGW("checkEffectCompatibility_l(): non post processing effect %s not allowed" 1322 " on output stage session", desc->name); 1323 return BAD_VALUE; 1324 } 1325 } else { 1326 // no restriction on effects applied on non fast tracks 1327 if ((hasAudioSession_l(sessionId) & ThreadBase::FAST_SESSION) == 0) { 1328 break; 1329 } 1330 } 1331 if (flags & AUDIO_OUTPUT_FLAG_RAW) { 1332 ALOGW("checkEffectCompatibility_l(): effect %s on playback thread in raw mode", 1333 desc->name); 1334 return BAD_VALUE; 1335 } 1336 if ((desc->flags & EFFECT_FLAG_HW_ACC_TUNNEL) == 0) { 1337 ALOGW("checkEffectCompatibility_l(): non HW effect %s on playback thread" 1338 " in fast mode", desc->name); 1339 return BAD_VALUE; 1340 } 1341 } 1342 } break; 1343 case OFFLOAD: 1344 // nothing actionable on offload threads, if the effect: 1345 // - is offloadable: the effect can be created 1346 // - is NOT offloadable: the effect should still be created, but EffectHandle::enable() 1347 // will take care of invalidating the tracks of the thread 1348 break; 1349 case DIRECT: 1350 // Reject any effect on Direct output threads for now, since the format of 1351 // mSinkBuffer is not guaranteed to be compatible with effect processing (PCM 16 stereo). 1352 ALOGW("checkEffectCompatibility_l(): effect %s on DIRECT output thread %s", 1353 desc->name, mThreadName); 1354 return BAD_VALUE; 1355 case DUPLICATING: 1356 // Reject any effect on mixer multichannel sinks. 1357 // TODO: fix both format and multichannel issues with effects. 1358 if (mChannelCount != FCC_2) { 1359 ALOGW("checkEffectCompatibility_l(): effect %s for multichannel(%d)" 1360 " on DUPLICATING thread %s", desc->name, mChannelCount, mThreadName); 1361 return BAD_VALUE; 1362 } 1363 if ((sessionId == AUDIO_SESSION_OUTPUT_STAGE) || (sessionId == AUDIO_SESSION_OUTPUT_MIX)) { 1364 ALOGW("checkEffectCompatibility_l(): global effect %s on DUPLICATING" 1365 " thread %s", desc->name, mThreadName); 1366 return BAD_VALUE; 1367 } 1368 if ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC) { 1369 ALOGW("checkEffectCompatibility_l(): post processing effect %s on" 1370 " DUPLICATING thread %s", desc->name, mThreadName); 1371 return BAD_VALUE; 1372 } 1373 if ((desc->flags & EFFECT_FLAG_HW_ACC_TUNNEL) != 0) { 1374 ALOGW("checkEffectCompatibility_l(): HW tunneled effect %s on" 1375 " DUPLICATING thread %s", desc->name, mThreadName); 1376 return BAD_VALUE; 1377 } 1378 break; 1379 default: 1380 LOG_ALWAYS_FATAL("checkEffectCompatibility_l(): wrong thread type %d", mType); 1381 } 1382 1383 return NO_ERROR; 1384} 1385 1386// ThreadBase::createEffect_l() must be called with AudioFlinger::mLock held 1387sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 1388 const sp<AudioFlinger::Client>& client, 1389 const sp<IEffectClient>& effectClient, 1390 int32_t priority, 1391 audio_session_t sessionId, 1392 effect_descriptor_t *desc, 1393 int *enabled, 1394 status_t *status) 1395{ 1396 sp<EffectModule> effect; 1397 sp<EffectHandle> handle; 1398 status_t lStatus; 1399 sp<EffectChain> chain; 1400 bool chainCreated = false; 1401 bool effectCreated = false; 1402 bool effectRegistered = false; 1403 1404 lStatus = initCheck(); 1405 if (lStatus != NO_ERROR) { 1406 ALOGW("createEffect_l() Audio driver not initialized."); 1407 goto Exit; 1408 } 1409 1410 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 1411 1412 { // scope for mLock 1413 Mutex::Autolock _l(mLock); 1414 1415 lStatus = checkEffectCompatibility_l(desc, sessionId); 1416 if (lStatus != NO_ERROR) { 1417 goto Exit; 1418 } 1419 1420 // check for existing effect chain with the requested audio session 1421 chain = getEffectChain_l(sessionId); 1422 if (chain == 0) { 1423 // create a new chain for this session 1424 ALOGV("createEffect_l() new effect chain for session %d", sessionId); 1425 chain = new EffectChain(this, sessionId); 1426 addEffectChain_l(chain); 1427 chain->setStrategy(getStrategyForSession_l(sessionId)); 1428 chainCreated = true; 1429 } else { 1430 effect = chain->getEffectFromDesc_l(desc); 1431 } 1432 1433 ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get()); 1434 1435 if (effect == 0) { 1436 audio_unique_id_t id = mAudioFlinger->nextUniqueId(AUDIO_UNIQUE_ID_USE_EFFECT); 1437 // Check CPU and memory usage 1438 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 1439 if (lStatus != NO_ERROR) { 1440 goto Exit; 1441 } 1442 effectRegistered = true; 1443 // create a new effect module if none present in the chain 1444 effect = new EffectModule(this, chain, desc, id, sessionId); 1445 lStatus = effect->status(); 1446 if (lStatus != NO_ERROR) { 1447 goto Exit; 1448 } 1449 effect->setOffloaded(mType == OFFLOAD, mId); 1450 1451 lStatus = chain->addEffect_l(effect); 1452 if (lStatus != NO_ERROR) { 1453 goto Exit; 1454 } 1455 effectCreated = true; 1456 1457 effect->setDevice(mOutDevice); 1458 effect->setDevice(mInDevice); 1459 effect->setMode(mAudioFlinger->getMode()); 1460 effect->setAudioSource(mAudioSource); 1461 } 1462 // create effect handle and connect it to effect module 1463 handle = new EffectHandle(effect, client, effectClient, priority); 1464 lStatus = handle->initCheck(); 1465 if (lStatus == OK) { 1466 lStatus = effect->addHandle(handle.get()); 1467 } 1468 if (enabled != NULL) { 1469 *enabled = (int)effect->isEnabled(); 1470 } 1471 } 1472 1473Exit: 1474 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 1475 Mutex::Autolock _l(mLock); 1476 if (effectCreated) { 1477 chain->removeEffect_l(effect); 1478 } 1479 if (effectRegistered) { 1480 AudioSystem::unregisterEffect(effect->id()); 1481 } 1482 if (chainCreated) { 1483 removeEffectChain_l(chain); 1484 } 1485 handle.clear(); 1486 } 1487 1488 *status = lStatus; 1489 return handle; 1490} 1491 1492sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect(audio_session_t sessionId, 1493 int effectId) 1494{ 1495 Mutex::Autolock _l(mLock); 1496 return getEffect_l(sessionId, effectId); 1497} 1498 1499sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(audio_session_t sessionId, 1500 int effectId) 1501{ 1502 sp<EffectChain> chain = getEffectChain_l(sessionId); 1503 return chain != 0 ? chain->getEffectFromId_l(effectId) : 0; 1504} 1505 1506// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 1507// PlaybackThread::mLock held 1508status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 1509{ 1510 // check for existing effect chain with the requested audio session 1511 audio_session_t sessionId = effect->sessionId(); 1512 sp<EffectChain> chain = getEffectChain_l(sessionId); 1513 bool chainCreated = false; 1514 1515 ALOGD_IF((mType == OFFLOAD) && !effect->isOffloadable(), 1516 "addEffect_l() on offloaded thread %p: effect %s does not support offload flags %x", 1517 this, effect->desc().name, effect->desc().flags); 1518 1519 if (chain == 0) { 1520 // create a new chain for this session 1521 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 1522 chain = new EffectChain(this, sessionId); 1523 addEffectChain_l(chain); 1524 chain->setStrategy(getStrategyForSession_l(sessionId)); 1525 chainCreated = true; 1526 } 1527 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 1528 1529 if (chain->getEffectFromId_l(effect->id()) != 0) { 1530 ALOGW("addEffect_l() %p effect %s already present in chain %p", 1531 this, effect->desc().name, chain.get()); 1532 return BAD_VALUE; 1533 } 1534 1535 effect->setOffloaded(mType == OFFLOAD, mId); 1536 1537 status_t status = chain->addEffect_l(effect); 1538 if (status != NO_ERROR) { 1539 if (chainCreated) { 1540 removeEffectChain_l(chain); 1541 } 1542 return status; 1543 } 1544 1545 effect->setDevice(mOutDevice); 1546 effect->setDevice(mInDevice); 1547 effect->setMode(mAudioFlinger->getMode()); 1548 effect->setAudioSource(mAudioSource); 1549 return NO_ERROR; 1550} 1551 1552void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 1553 1554 ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 1555 effect_descriptor_t desc = effect->desc(); 1556 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 1557 detachAuxEffect_l(effect->id()); 1558 } 1559 1560 sp<EffectChain> chain = effect->chain().promote(); 1561 if (chain != 0) { 1562 // remove effect chain if removing last effect 1563 if (chain->removeEffect_l(effect) == 0) { 1564 removeEffectChain_l(chain); 1565 } 1566 } else { 1567 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 1568 } 1569} 1570 1571void AudioFlinger::ThreadBase::lockEffectChains_l( 1572 Vector< sp<AudioFlinger::EffectChain> >& effectChains) 1573{ 1574 effectChains = mEffectChains; 1575 for (size_t i = 0; i < mEffectChains.size(); i++) { 1576 mEffectChains[i]->lock(); 1577 } 1578} 1579 1580void AudioFlinger::ThreadBase::unlockEffectChains( 1581 const Vector< sp<AudioFlinger::EffectChain> >& effectChains) 1582{ 1583 for (size_t i = 0; i < effectChains.size(); i++) { 1584 effectChains[i]->unlock(); 1585 } 1586} 1587 1588sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(audio_session_t sessionId) 1589{ 1590 Mutex::Autolock _l(mLock); 1591 return getEffectChain_l(sessionId); 1592} 1593 1594sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(audio_session_t sessionId) 1595 const 1596{ 1597 size_t size = mEffectChains.size(); 1598 for (size_t i = 0; i < size; i++) { 1599 if (mEffectChains[i]->sessionId() == sessionId) { 1600 return mEffectChains[i]; 1601 } 1602 } 1603 return 0; 1604} 1605 1606void AudioFlinger::ThreadBase::setMode(audio_mode_t mode) 1607{ 1608 Mutex::Autolock _l(mLock); 1609 size_t size = mEffectChains.size(); 1610 for (size_t i = 0; i < size; i++) { 1611 mEffectChains[i]->setMode_l(mode); 1612 } 1613} 1614 1615void AudioFlinger::ThreadBase::getAudioPortConfig(struct audio_port_config *config) 1616{ 1617 config->type = AUDIO_PORT_TYPE_MIX; 1618 config->ext.mix.handle = mId; 1619 config->sample_rate = mSampleRate; 1620 config->format = mFormat; 1621 config->channel_mask = mChannelMask; 1622 config->config_mask = AUDIO_PORT_CONFIG_SAMPLE_RATE|AUDIO_PORT_CONFIG_CHANNEL_MASK| 1623 AUDIO_PORT_CONFIG_FORMAT; 1624} 1625 1626void AudioFlinger::ThreadBase::systemReady() 1627{ 1628 Mutex::Autolock _l(mLock); 1629 if (mSystemReady) { 1630 return; 1631 } 1632 mSystemReady = true; 1633 1634 for (size_t i = 0; i < mPendingConfigEvents.size(); i++) { 1635 sendConfigEvent_l(mPendingConfigEvents.editItemAt(i)); 1636 } 1637 mPendingConfigEvents.clear(); 1638} 1639 1640 1641// ---------------------------------------------------------------------------- 1642// Playback 1643// ---------------------------------------------------------------------------- 1644 1645AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, 1646 AudioStreamOut* output, 1647 audio_io_handle_t id, 1648 audio_devices_t device, 1649 type_t type, 1650 bool systemReady) 1651 : ThreadBase(audioFlinger, id, device, AUDIO_DEVICE_NONE, type, systemReady), 1652 mNormalFrameCount(0), mSinkBuffer(NULL), 1653 mMixerBufferEnabled(AudioFlinger::kEnableExtendedPrecision), 1654 mMixerBuffer(NULL), 1655 mMixerBufferSize(0), 1656 mMixerBufferFormat(AUDIO_FORMAT_INVALID), 1657 mMixerBufferValid(false), 1658 mEffectBufferEnabled(AudioFlinger::kEnableExtendedPrecision), 1659 mEffectBuffer(NULL), 1660 mEffectBufferSize(0), 1661 mEffectBufferFormat(AUDIO_FORMAT_INVALID), 1662 mEffectBufferValid(false), 1663 mSuspended(0), mBytesWritten(0), 1664 mFramesWritten(0), 1665 mActiveTracksGeneration(0), 1666 // mStreamTypes[] initialized in constructor body 1667 mOutput(output), 1668 mLastWriteTime(-1), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false), 1669 mMixerStatus(MIXER_IDLE), 1670 mMixerStatusIgnoringFastTracks(MIXER_IDLE), 1671 mStandbyDelayNs(AudioFlinger::mStandbyTimeInNsecs), 1672 mBytesRemaining(0), 1673 mCurrentWriteLength(0), 1674 mUseAsyncWrite(false), 1675 mWriteAckSequence(0), 1676 mDrainSequence(0), 1677 mSignalPending(false), 1678 mScreenState(AudioFlinger::mScreenState), 1679 // index 0 is reserved for normal mixer's submix 1680 mFastTrackAvailMask(((1 << FastMixerState::sMaxFastTracks) - 1) & ~1), 1681 mHwSupportsPause(false), mHwPaused(false), mFlushPending(false) 1682{ 1683 snprintf(mThreadName, kThreadNameLength, "AudioOut_%X", id); 1684 mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mThreadName); 1685 1686 // Assumes constructor is called by AudioFlinger with it's mLock held, but 1687 // it would be safer to explicitly pass initial masterVolume/masterMute as 1688 // parameter. 1689 // 1690 // If the HAL we are using has support for master volume or master mute, 1691 // then do not attenuate or mute during mixing (just leave the volume at 1.0 1692 // and the mute set to false). 1693 mMasterVolume = audioFlinger->masterVolume_l(); 1694 mMasterMute = audioFlinger->masterMute_l(); 1695 if (mOutput && mOutput->audioHwDev) { 1696 if (mOutput->audioHwDev->canSetMasterVolume()) { 1697 mMasterVolume = 1.0; 1698 } 1699 1700 if (mOutput->audioHwDev->canSetMasterMute()) { 1701 mMasterMute = false; 1702 } 1703 } 1704 1705 readOutputParameters_l(); 1706 1707 // ++ operator does not compile 1708 for (audio_stream_type_t stream = AUDIO_STREAM_MIN; stream < AUDIO_STREAM_CNT; 1709 stream = (audio_stream_type_t) (stream + 1)) { 1710 mStreamTypes[stream].volume = mAudioFlinger->streamVolume_l(stream); 1711 mStreamTypes[stream].mute = mAudioFlinger->streamMute_l(stream); 1712 } 1713} 1714 1715AudioFlinger::PlaybackThread::~PlaybackThread() 1716{ 1717 mAudioFlinger->unregisterWriter(mNBLogWriter); 1718 free(mSinkBuffer); 1719 free(mMixerBuffer); 1720 free(mEffectBuffer); 1721} 1722 1723void AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) 1724{ 1725 dumpInternals(fd, args); 1726 dumpTracks(fd, args); 1727 dumpEffectChains(fd, args); 1728} 1729 1730void AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args __unused) 1731{ 1732 const size_t SIZE = 256; 1733 char buffer[SIZE]; 1734 String8 result; 1735 1736 result.appendFormat(" Stream volumes in dB: "); 1737 for (int i = 0; i < AUDIO_STREAM_CNT; ++i) { 1738 const stream_type_t *st = &mStreamTypes[i]; 1739 if (i > 0) { 1740 result.appendFormat(", "); 1741 } 1742 result.appendFormat("%d:%.2g", i, 20.0 * log10(st->volume)); 1743 if (st->mute) { 1744 result.append("M"); 1745 } 1746 } 1747 result.append("\n"); 1748 write(fd, result.string(), result.length()); 1749 result.clear(); 1750 1751 // These values are "raw"; they will wrap around. See prepareTracks_l() for a better way. 1752 FastTrackUnderruns underruns = getFastTrackUnderruns(0); 1753 dprintf(fd, " Normal mixer raw underrun counters: partial=%u empty=%u\n", 1754 underruns.mBitFields.mPartial, underruns.mBitFields.mEmpty); 1755 1756 size_t numtracks = mTracks.size(); 1757 size_t numactive = mActiveTracks.size(); 1758 dprintf(fd, " %zu Tracks", numtracks); 1759 size_t numactiveseen = 0; 1760 if (numtracks) { 1761 dprintf(fd, " of which %zu are active\n", numactive); 1762 Track::appendDumpHeader(result); 1763 for (size_t i = 0; i < numtracks; ++i) { 1764 sp<Track> track = mTracks[i]; 1765 if (track != 0) { 1766 bool active = mActiveTracks.indexOf(track) >= 0; 1767 if (active) { 1768 numactiveseen++; 1769 } 1770 track->dump(buffer, SIZE, active); 1771 result.append(buffer); 1772 } 1773 } 1774 } else { 1775 result.append("\n"); 1776 } 1777 if (numactiveseen != numactive) { 1778 // some tracks in the active list were not in the tracks list 1779 snprintf(buffer, SIZE, " The following tracks are in the active list but" 1780 " not in the track list\n"); 1781 result.append(buffer); 1782 Track::appendDumpHeader(result); 1783 for (size_t i = 0; i < numactive; ++i) { 1784 sp<Track> track = mActiveTracks[i].promote(); 1785 if (track != 0 && mTracks.indexOf(track) < 0) { 1786 track->dump(buffer, SIZE, true); 1787 result.append(buffer); 1788 } 1789 } 1790 } 1791 1792 write(fd, result.string(), result.size()); 1793} 1794 1795void AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 1796{ 1797 dprintf(fd, "\nOutput thread %p type %d (%s):\n", this, type(), threadTypeToString(type())); 1798 1799 dumpBase(fd, args); 1800 1801 dprintf(fd, " Normal frame count: %zu\n", mNormalFrameCount); 1802 dprintf(fd, " Last write occurred (msecs): %llu\n", 1803 (unsigned long long) ns2ms(systemTime() - mLastWriteTime)); 1804 dprintf(fd, " Total writes: %d\n", mNumWrites); 1805 dprintf(fd, " Delayed writes: %d\n", mNumDelayedWrites); 1806 dprintf(fd, " Blocked in write: %s\n", mInWrite ? "yes" : "no"); 1807 dprintf(fd, " Suspend count: %d\n", mSuspended); 1808 dprintf(fd, " Sink buffer : %p\n", mSinkBuffer); 1809 dprintf(fd, " Mixer buffer: %p\n", mMixerBuffer); 1810 dprintf(fd, " Effect buffer: %p\n", mEffectBuffer); 1811 dprintf(fd, " Fast track availMask=%#x\n", mFastTrackAvailMask); 1812 dprintf(fd, " Standby delay ns=%lld\n", (long long)mStandbyDelayNs); 1813 AudioStreamOut *output = mOutput; 1814 audio_output_flags_t flags = output != NULL ? output->flags : AUDIO_OUTPUT_FLAG_NONE; 1815 String8 flagsAsString = outputFlagsToString(flags); 1816 dprintf(fd, " AudioStreamOut: %p flags %#x (%s)\n", output, flags, flagsAsString.string()); 1817} 1818 1819// Thread virtuals 1820 1821void AudioFlinger::PlaybackThread::onFirstRef() 1822{ 1823 run(mThreadName, ANDROID_PRIORITY_URGENT_AUDIO); 1824} 1825 1826// ThreadBase virtuals 1827void AudioFlinger::PlaybackThread::preExit() 1828{ 1829 ALOGV(" preExit()"); 1830 // FIXME this is using hard-coded strings but in the future, this functionality will be 1831 // converted to use audio HAL extensions required to support tunneling 1832 mOutput->stream->common.set_parameters(&mOutput->stream->common, "exiting=1"); 1833} 1834 1835// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 1836sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 1837 const sp<AudioFlinger::Client>& client, 1838 audio_stream_type_t streamType, 1839 uint32_t sampleRate, 1840 audio_format_t format, 1841 audio_channel_mask_t channelMask, 1842 size_t *pFrameCount, 1843 const sp<IMemory>& sharedBuffer, 1844 audio_session_t sessionId, 1845 audio_output_flags_t *flags, 1846 pid_t tid, 1847 int uid, 1848 status_t *status) 1849{ 1850 size_t frameCount = *pFrameCount; 1851 sp<Track> track; 1852 status_t lStatus; 1853 audio_output_flags_t outputFlags = mOutput->flags; 1854 1855 // special case for FAST flag considered OK if fast mixer is present 1856 if (hasFastMixer()) { 1857 outputFlags = (audio_output_flags_t)(outputFlags | AUDIO_OUTPUT_FLAG_FAST); 1858 } 1859 1860 // Check if requested flags are compatible with output stream flags 1861 if ((*flags & outputFlags) != *flags) { 1862 ALOGW("createTrack_l(): mismatch between requested flags (%08x) and output flags (%08x)", 1863 *flags, outputFlags); 1864 *flags = (audio_output_flags_t)(*flags & outputFlags); 1865 } 1866 1867 // client expresses a preference for FAST, but we get the final say 1868 if (*flags & AUDIO_OUTPUT_FLAG_FAST) { 1869 if ( 1870 // PCM data 1871 audio_is_linear_pcm(format) && 1872 // TODO: extract as a data library function that checks that a computationally 1873 // expensive downmixer is not required: isFastOutputChannelConversion() 1874 (channelMask == mChannelMask || 1875 mChannelMask != AUDIO_CHANNEL_OUT_STEREO || 1876 (channelMask == AUDIO_CHANNEL_OUT_MONO 1877 /* && mChannelMask == AUDIO_CHANNEL_OUT_STEREO */)) && 1878 // hardware sample rate 1879 (sampleRate == mSampleRate) && 1880 // normal mixer has an associated fast mixer 1881 hasFastMixer() && 1882 // there are sufficient fast track slots available 1883 (mFastTrackAvailMask != 0) 1884 // FIXME test that MixerThread for this fast track has a capable output HAL 1885 // FIXME add a permission test also? 1886 ) { 1887 // static tracks can have any nonzero framecount, streaming tracks check against minimum. 1888 if (sharedBuffer == 0) { 1889 // read the fast track multiplier property the first time it is needed 1890 int ok = pthread_once(&sFastTrackMultiplierOnce, sFastTrackMultiplierInit); 1891 if (ok != 0) { 1892 ALOGE("%s pthread_once failed: %d", __func__, ok); 1893 } 1894 frameCount = max(frameCount, mFrameCount * sFastTrackMultiplier); // incl framecount 0 1895 } 1896 1897 // check compatibility with audio effects. 1898 { // scope for mLock 1899 Mutex::Autolock _l(mLock); 1900 // do not accept RAW flag if post processing are present. Note that post processing on 1901 // a fast mixer are necessarily hardware 1902 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_STAGE); 1903 if (chain != 0) { 1904 ALOGV_IF((*flags & AUDIO_OUTPUT_FLAG_RAW) != 0, 1905 "AUDIO_OUTPUT_FLAG_RAW denied: post processing effect present"); 1906 *flags = (audio_output_flags_t)(*flags & ~AUDIO_OUTPUT_FLAG_RAW); 1907 } 1908 // Do not accept FAST flag if software global effects are present 1909 chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 1910 if (chain != 0) { 1911 ALOGV_IF((*flags & AUDIO_OUTPUT_FLAG_RAW) != 0, 1912 "AUDIO_OUTPUT_FLAG_RAW denied: global effect present"); 1913 *flags = (audio_output_flags_t)(*flags & ~AUDIO_OUTPUT_FLAG_RAW); 1914 if (chain->hasSoftwareEffect()) { 1915 ALOGV("AUDIO_OUTPUT_FLAG_FAST denied: software global effect present"); 1916 *flags = (audio_output_flags_t)(*flags & ~AUDIO_OUTPUT_FLAG_FAST); 1917 } 1918 } 1919 // Do not accept FAST flag if the session has software effects 1920 chain = getEffectChain_l(sessionId); 1921 if (chain != 0) { 1922 ALOGV_IF((*flags & AUDIO_OUTPUT_FLAG_RAW) != 0, 1923 "AUDIO_OUTPUT_FLAG_RAW denied: effect present on session"); 1924 *flags = (audio_output_flags_t)(*flags & ~AUDIO_OUTPUT_FLAG_RAW); 1925 if (chain->hasSoftwareEffect()) { 1926 ALOGV("AUDIO_OUTPUT_FLAG_FAST denied: software effect present on session"); 1927 *flags = (audio_output_flags_t)(*flags & ~AUDIO_OUTPUT_FLAG_FAST); 1928 } 1929 } 1930 } 1931 ALOGV_IF((*flags & AUDIO_OUTPUT_FLAG_FAST) != 0, 1932 "AUDIO_OUTPUT_FLAG_FAST accepted: frameCount=%zu mFrameCount=%zu", 1933 frameCount, mFrameCount); 1934 } else { 1935 ALOGV("AUDIO_OUTPUT_FLAG_FAST denied: sharedBuffer=%p frameCount=%zu " 1936 "mFrameCount=%zu format=%#x mFormat=%#x isLinear=%d channelMask=%#x " 1937 "sampleRate=%u mSampleRate=%u " 1938 "hasFastMixer=%d tid=%d fastTrackAvailMask=%#x", 1939 sharedBuffer.get(), frameCount, mFrameCount, format, mFormat, 1940 audio_is_linear_pcm(format), 1941 channelMask, sampleRate, mSampleRate, hasFastMixer(), tid, mFastTrackAvailMask); 1942 *flags = (audio_output_flags_t)(*flags & ~AUDIO_OUTPUT_FLAG_FAST); 1943 } 1944 } 1945 // For normal PCM streaming tracks, update minimum frame count. 1946 // For compatibility with AudioTrack calculation, buffer depth is forced 1947 // to be at least 2 x the normal mixer frame count and cover audio hardware latency. 1948 // This is probably too conservative, but legacy application code may depend on it. 1949 // If you change this calculation, also review the start threshold which is related. 1950 if (!(*flags & AUDIO_OUTPUT_FLAG_FAST) 1951 && audio_has_proportional_frames(format) && sharedBuffer == 0) { 1952 // this must match AudioTrack.cpp calculateMinFrameCount(). 1953 // TODO: Move to a common library 1954 uint32_t latencyMs = mOutput->stream->get_latency(mOutput->stream); 1955 uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate); 1956 if (minBufCount < 2) { 1957 minBufCount = 2; 1958 } 1959 // For normal mixing tracks, if speed is > 1.0f (normal), AudioTrack 1960 // or the client should compute and pass in a larger buffer request. 1961 size_t minFrameCount = 1962 minBufCount * sourceFramesNeededWithTimestretch( 1963 sampleRate, mNormalFrameCount, 1964 mSampleRate, AUDIO_TIMESTRETCH_SPEED_NORMAL /*speed*/); 1965 if (frameCount < minFrameCount) { // including frameCount == 0 1966 frameCount = minFrameCount; 1967 } 1968 } 1969 *pFrameCount = frameCount; 1970 1971 switch (mType) { 1972 1973 case DIRECT: 1974 if (audio_is_linear_pcm(format)) { // TODO maybe use audio_has_proportional_frames()? 1975 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1976 ALOGE("createTrack_l() Bad parameter: sampleRate %u format %#x, channelMask 0x%08x " 1977 "for output %p with format %#x", 1978 sampleRate, format, channelMask, mOutput, mFormat); 1979 lStatus = BAD_VALUE; 1980 goto Exit; 1981 } 1982 } 1983 break; 1984 1985 case OFFLOAD: 1986 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1987 ALOGE("createTrack_l() Bad parameter: sampleRate %d format %#x, channelMask 0x%08x \"" 1988 "for output %p with format %#x", 1989 sampleRate, format, channelMask, mOutput, mFormat); 1990 lStatus = BAD_VALUE; 1991 goto Exit; 1992 } 1993 break; 1994 1995 default: 1996 if (!audio_is_linear_pcm(format)) { 1997 ALOGE("createTrack_l() Bad parameter: format %#x \"" 1998 "for output %p with format %#x", 1999 format, mOutput, mFormat); 2000 lStatus = BAD_VALUE; 2001 goto Exit; 2002 } 2003 if (sampleRate > mSampleRate * AUDIO_RESAMPLER_DOWN_RATIO_MAX) { 2004 ALOGE("Sample rate out of range: %u mSampleRate %u", sampleRate, mSampleRate); 2005 lStatus = BAD_VALUE; 2006 goto Exit; 2007 } 2008 break; 2009 2010 } 2011 2012 lStatus = initCheck(); 2013 if (lStatus != NO_ERROR) { 2014 ALOGE("createTrack_l() audio driver not initialized"); 2015 goto Exit; 2016 } 2017 2018 { // scope for mLock 2019 Mutex::Autolock _l(mLock); 2020 2021 // all tracks in same audio session must share the same routing strategy otherwise 2022 // conflicts will happen when tracks are moved from one output to another by audio policy 2023 // manager 2024 uint32_t strategy = AudioSystem::getStrategyForStream(streamType); 2025 for (size_t i = 0; i < mTracks.size(); ++i) { 2026 sp<Track> t = mTracks[i]; 2027 if (t != 0 && t->isExternalTrack()) { 2028 uint32_t actual = AudioSystem::getStrategyForStream(t->streamType()); 2029 if (sessionId == t->sessionId() && strategy != actual) { 2030 ALOGE("createTrack_l() mismatched strategy; expected %u but found %u", 2031 strategy, actual); 2032 lStatus = BAD_VALUE; 2033 goto Exit; 2034 } 2035 } 2036 } 2037 2038 track = new Track(this, client, streamType, sampleRate, format, 2039 channelMask, frameCount, NULL, sharedBuffer, 2040 sessionId, uid, *flags, TrackBase::TYPE_DEFAULT); 2041 2042 lStatus = track != 0 ? track->initCheck() : (status_t) NO_MEMORY; 2043 if (lStatus != NO_ERROR) { 2044 ALOGE("createTrack_l() initCheck failed %d; no control block?", lStatus); 2045 // track must be cleared from the caller as the caller has the AF lock 2046 goto Exit; 2047 } 2048 mTracks.add(track); 2049 2050 sp<EffectChain> chain = getEffectChain_l(sessionId); 2051 if (chain != 0) { 2052 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 2053 track->setMainBuffer(chain->inBuffer()); 2054 chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType())); 2055 chain->incTrackCnt(); 2056 } 2057 2058 if ((*flags & AUDIO_OUTPUT_FLAG_FAST) && (tid != -1)) { 2059 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 2060 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 2061 // so ask activity manager to do this on our behalf 2062 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 2063 } 2064 } 2065 2066 lStatus = NO_ERROR; 2067 2068Exit: 2069 *status = lStatus; 2070 return track; 2071} 2072 2073uint32_t AudioFlinger::PlaybackThread::correctLatency_l(uint32_t latency) const 2074{ 2075 return latency; 2076} 2077 2078uint32_t AudioFlinger::PlaybackThread::latency() const 2079{ 2080 Mutex::Autolock _l(mLock); 2081 return latency_l(); 2082} 2083uint32_t AudioFlinger::PlaybackThread::latency_l() const 2084{ 2085 if (initCheck() == NO_ERROR) { 2086 return correctLatency_l(mOutput->stream->get_latency(mOutput->stream)); 2087 } else { 2088 return 0; 2089 } 2090} 2091 2092void AudioFlinger::PlaybackThread::setMasterVolume(float value) 2093{ 2094 Mutex::Autolock _l(mLock); 2095 // Don't apply master volume in SW if our HAL can do it for us. 2096 if (mOutput && mOutput->audioHwDev && 2097 mOutput->audioHwDev->canSetMasterVolume()) { 2098 mMasterVolume = 1.0; 2099 } else { 2100 mMasterVolume = value; 2101 } 2102} 2103 2104void AudioFlinger::PlaybackThread::setMasterMute(bool muted) 2105{ 2106 Mutex::Autolock _l(mLock); 2107 // Don't apply master mute in SW if our HAL can do it for us. 2108 if (mOutput && mOutput->audioHwDev && 2109 mOutput->audioHwDev->canSetMasterMute()) { 2110 mMasterMute = false; 2111 } else { 2112 mMasterMute = muted; 2113 } 2114} 2115 2116void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) 2117{ 2118 Mutex::Autolock _l(mLock); 2119 mStreamTypes[stream].volume = value; 2120 broadcast_l(); 2121} 2122 2123void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) 2124{ 2125 Mutex::Autolock _l(mLock); 2126 mStreamTypes[stream].mute = muted; 2127 broadcast_l(); 2128} 2129 2130float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const 2131{ 2132 Mutex::Autolock _l(mLock); 2133 return mStreamTypes[stream].volume; 2134} 2135 2136// addTrack_l() must be called with ThreadBase::mLock held 2137status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 2138{ 2139 status_t status = ALREADY_EXISTS; 2140 2141 if (mActiveTracks.indexOf(track) < 0) { 2142 // the track is newly added, make sure it fills up all its 2143 // buffers before playing. This is to ensure the client will 2144 // effectively get the latency it requested. 2145 if (track->isExternalTrack()) { 2146 TrackBase::track_state state = track->mState; 2147 mLock.unlock(); 2148 status = AudioSystem::startOutput(mId, track->streamType(), 2149 track->sessionId()); 2150 mLock.lock(); 2151 // abort track was stopped/paused while we released the lock 2152 if (state != track->mState) { 2153 if (status == NO_ERROR) { 2154 mLock.unlock(); 2155 AudioSystem::stopOutput(mId, track->streamType(), 2156 track->sessionId()); 2157 mLock.lock(); 2158 } 2159 return INVALID_OPERATION; 2160 } 2161 // abort if start is rejected by audio policy manager 2162 if (status != NO_ERROR) { 2163 return PERMISSION_DENIED; 2164 } 2165#ifdef ADD_BATTERY_DATA 2166 // to track the speaker usage 2167 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 2168#endif 2169 } 2170 2171 // set retry count for buffer fill 2172 if (track->isOffloaded()) { 2173 if (track->isStopping_1()) { 2174 track->mRetryCount = kMaxTrackStopRetriesOffload; 2175 } else { 2176 track->mRetryCount = kMaxTrackStartupRetriesOffload; 2177 } 2178 track->mFillingUpStatus = mStandby ? Track::FS_FILLING : Track::FS_FILLED; 2179 } else { 2180 track->mRetryCount = kMaxTrackStartupRetries; 2181 track->mFillingUpStatus = 2182 track->sharedBuffer() != 0 ? Track::FS_FILLED : Track::FS_FILLING; 2183 } 2184 2185 track->mResetDone = false; 2186 track->mPresentationCompleteFrames = 0; 2187 mActiveTracks.add(track); 2188 mWakeLockUids.add(track->uid()); 2189 mActiveTracksGeneration++; 2190 mLatestActiveTrack = track; 2191 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 2192 if (chain != 0) { 2193 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), 2194 track->sessionId()); 2195 chain->incActiveTrackCnt(); 2196 } 2197 2198 status = NO_ERROR; 2199 } 2200 2201 onAddNewTrack_l(); 2202 return status; 2203} 2204 2205bool AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 2206{ 2207 track->terminate(); 2208 // active tracks are removed by threadLoop() 2209 bool trackActive = (mActiveTracks.indexOf(track) >= 0); 2210 track->mState = TrackBase::STOPPED; 2211 if (!trackActive) { 2212 removeTrack_l(track); 2213 } else if (track->isFastTrack() || track->isOffloaded() || track->isDirect()) { 2214 track->mState = TrackBase::STOPPING_1; 2215 } 2216 2217 return trackActive; 2218} 2219 2220void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 2221{ 2222 track->triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 2223 mTracks.remove(track); 2224 deleteTrackName_l(track->name()); 2225 // redundant as track is about to be destroyed, for dumpsys only 2226 track->mName = -1; 2227 if (track->isFastTrack()) { 2228 int index = track->mFastIndex; 2229 ALOG_ASSERT(0 < index && index < (int)FastMixerState::sMaxFastTracks); 2230 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << index))); 2231 mFastTrackAvailMask |= 1 << index; 2232 // redundant as track is about to be destroyed, for dumpsys only 2233 track->mFastIndex = -1; 2234 } 2235 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 2236 if (chain != 0) { 2237 chain->decTrackCnt(); 2238 } 2239} 2240 2241void AudioFlinger::PlaybackThread::broadcast_l() 2242{ 2243 // Thread could be blocked waiting for async 2244 // so signal it to handle state changes immediately 2245 // If threadLoop is currently unlocked a signal of mWaitWorkCV will 2246 // be lost so we also flag to prevent it blocking on mWaitWorkCV 2247 mSignalPending = true; 2248 mWaitWorkCV.broadcast(); 2249} 2250 2251String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 2252{ 2253 Mutex::Autolock _l(mLock); 2254 if (initCheck() != NO_ERROR) { 2255 return String8(); 2256 } 2257 2258 char *s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 2259 const String8 out_s8(s); 2260 free(s); 2261 return out_s8; 2262} 2263 2264void AudioFlinger::PlaybackThread::ioConfigChanged(audio_io_config_event event, pid_t pid) { 2265 sp<AudioIoDescriptor> desc = new AudioIoDescriptor(); 2266 ALOGV("PlaybackThread::ioConfigChanged, thread %p, event %d", this, event); 2267 2268 desc->mIoHandle = mId; 2269 2270 switch (event) { 2271 case AUDIO_OUTPUT_OPENED: 2272 case AUDIO_OUTPUT_CONFIG_CHANGED: 2273 desc->mPatch = mPatch; 2274 desc->mChannelMask = mChannelMask; 2275 desc->mSamplingRate = mSampleRate; 2276 desc->mFormat = mFormat; 2277 desc->mFrameCount = mNormalFrameCount; // FIXME see 2278 // AudioFlinger::frameCount(audio_io_handle_t) 2279 desc->mFrameCountHAL = mFrameCount; 2280 desc->mLatency = latency_l(); 2281 break; 2282 2283 case AUDIO_OUTPUT_CLOSED: 2284 default: 2285 break; 2286 } 2287 mAudioFlinger->ioConfigChanged(event, desc, pid); 2288} 2289 2290void AudioFlinger::PlaybackThread::writeCallback() 2291{ 2292 ALOG_ASSERT(mCallbackThread != 0); 2293 mCallbackThread->resetWriteBlocked(); 2294} 2295 2296void AudioFlinger::PlaybackThread::drainCallback() 2297{ 2298 ALOG_ASSERT(mCallbackThread != 0); 2299 mCallbackThread->resetDraining(); 2300} 2301 2302void AudioFlinger::PlaybackThread::errorCallback() 2303{ 2304 ALOG_ASSERT(mCallbackThread != 0); 2305 mCallbackThread->setAsyncError(); 2306} 2307 2308void AudioFlinger::PlaybackThread::resetWriteBlocked(uint32_t sequence) 2309{ 2310 Mutex::Autolock _l(mLock); 2311 // reject out of sequence requests 2312 if ((mWriteAckSequence & 1) && (sequence == mWriteAckSequence)) { 2313 mWriteAckSequence &= ~1; 2314 mWaitWorkCV.signal(); 2315 } 2316} 2317 2318void AudioFlinger::PlaybackThread::resetDraining(uint32_t sequence) 2319{ 2320 Mutex::Autolock _l(mLock); 2321 // reject out of sequence requests 2322 if ((mDrainSequence & 1) && (sequence == mDrainSequence)) { 2323 mDrainSequence &= ~1; 2324 mWaitWorkCV.signal(); 2325 } 2326} 2327 2328// static 2329int AudioFlinger::PlaybackThread::asyncCallback(stream_callback_event_t event, 2330 void *param __unused, 2331 void *cookie) 2332{ 2333 AudioFlinger::PlaybackThread *me = (AudioFlinger::PlaybackThread *)cookie; 2334 ALOGV("asyncCallback() event %d", event); 2335 switch (event) { 2336 case STREAM_CBK_EVENT_WRITE_READY: 2337 me->writeCallback(); 2338 break; 2339 case STREAM_CBK_EVENT_DRAIN_READY: 2340 me->drainCallback(); 2341 break; 2342 case STREAM_CBK_EVENT_ERROR: 2343 me->errorCallback(); 2344 break; 2345 default: 2346 ALOGW("asyncCallback() unknown event %d", event); 2347 break; 2348 } 2349 return 0; 2350} 2351 2352void AudioFlinger::PlaybackThread::readOutputParameters_l() 2353{ 2354 // unfortunately we have no way of recovering from errors here, hence the LOG_ALWAYS_FATAL 2355 mSampleRate = mOutput->getSampleRate(); 2356 mChannelMask = mOutput->getChannelMask(); 2357 if (!audio_is_output_channel(mChannelMask)) { 2358 LOG_ALWAYS_FATAL("HAL channel mask %#x not valid for output", mChannelMask); 2359 } 2360 if ((mType == MIXER || mType == DUPLICATING) 2361 && !isValidPcmSinkChannelMask(mChannelMask)) { 2362 LOG_ALWAYS_FATAL("HAL channel mask %#x not supported for mixed output", 2363 mChannelMask); 2364 } 2365 mChannelCount = audio_channel_count_from_out_mask(mChannelMask); 2366 2367 // Get actual HAL format. 2368 mHALFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 2369 // Get format from the shim, which will be different than the HAL format 2370 // if playing compressed audio over HDMI passthrough. 2371 mFormat = mOutput->getFormat(); 2372 if (!audio_is_valid_format(mFormat)) { 2373 LOG_ALWAYS_FATAL("HAL format %#x not valid for output", mFormat); 2374 } 2375 if ((mType == MIXER || mType == DUPLICATING) 2376 && !isValidPcmSinkFormat(mFormat)) { 2377 LOG_FATAL("HAL format %#x not supported for mixed output", 2378 mFormat); 2379 } 2380 mFrameSize = mOutput->getFrameSize(); 2381 mBufferSize = mOutput->stream->common.get_buffer_size(&mOutput->stream->common); 2382 mFrameCount = mBufferSize / mFrameSize; 2383 if (mFrameCount & 15) { 2384 ALOGW("HAL output buffer size is %zu frames but AudioMixer requires multiples of 16 frames", 2385 mFrameCount); 2386 } 2387 2388 if ((mOutput->flags & AUDIO_OUTPUT_FLAG_NON_BLOCKING) && 2389 (mOutput->stream->set_callback != NULL)) { 2390 if (mOutput->stream->set_callback(mOutput->stream, 2391 AudioFlinger::PlaybackThread::asyncCallback, this) == 0) { 2392 mUseAsyncWrite = true; 2393 mCallbackThread = new AudioFlinger::AsyncCallbackThread(this); 2394 } 2395 } 2396 2397 mHwSupportsPause = false; 2398 if (mOutput->flags & AUDIO_OUTPUT_FLAG_DIRECT) { 2399 if (mOutput->stream->pause != NULL) { 2400 if (mOutput->stream->resume != NULL) { 2401 mHwSupportsPause = true; 2402 } else { 2403 ALOGW("direct output implements pause but not resume"); 2404 } 2405 } else if (mOutput->stream->resume != NULL) { 2406 ALOGW("direct output implements resume but not pause"); 2407 } 2408 } 2409 if (!mHwSupportsPause && mOutput->flags & AUDIO_OUTPUT_FLAG_HW_AV_SYNC) { 2410 LOG_ALWAYS_FATAL("HW_AV_SYNC requested but HAL does not implement pause and resume"); 2411 } 2412 2413 if (mType == DUPLICATING && mMixerBufferEnabled && mEffectBufferEnabled) { 2414 // For best precision, we use float instead of the associated output 2415 // device format (typically PCM 16 bit). 2416 2417 mFormat = AUDIO_FORMAT_PCM_FLOAT; 2418 mFrameSize = mChannelCount * audio_bytes_per_sample(mFormat); 2419 mBufferSize = mFrameSize * mFrameCount; 2420 2421 // TODO: We currently use the associated output device channel mask and sample rate. 2422 // (1) Perhaps use the ORed channel mask of all downstream MixerThreads 2423 // (if a valid mask) to avoid premature downmix. 2424 // (2) Perhaps use the maximum sample rate of all downstream MixerThreads 2425 // instead of the output device sample rate to avoid loss of high frequency information. 2426 // This may need to be updated as MixerThread/OutputTracks are added and not here. 2427 } 2428 2429 // Calculate size of normal sink buffer relative to the HAL output buffer size 2430 double multiplier = 1.0; 2431 if (mType == MIXER && (kUseFastMixer == FastMixer_Static || 2432 kUseFastMixer == FastMixer_Dynamic)) { 2433 size_t minNormalFrameCount = (kMinNormalSinkBufferSizeMs * mSampleRate) / 1000; 2434 size_t maxNormalFrameCount = (kMaxNormalSinkBufferSizeMs * mSampleRate) / 1000; 2435 2436 // round up minimum and round down maximum to nearest 16 frames to satisfy AudioMixer 2437 minNormalFrameCount = (minNormalFrameCount + 15) & ~15; 2438 maxNormalFrameCount = maxNormalFrameCount & ~15; 2439 if (maxNormalFrameCount < minNormalFrameCount) { 2440 maxNormalFrameCount = minNormalFrameCount; 2441 } 2442 multiplier = (double) minNormalFrameCount / (double) mFrameCount; 2443 if (multiplier <= 1.0) { 2444 multiplier = 1.0; 2445 } else if (multiplier <= 2.0) { 2446 if (2 * mFrameCount <= maxNormalFrameCount) { 2447 multiplier = 2.0; 2448 } else { 2449 multiplier = (double) maxNormalFrameCount / (double) mFrameCount; 2450 } 2451 } else { 2452 multiplier = floor(multiplier); 2453 } 2454 } 2455 mNormalFrameCount = multiplier * mFrameCount; 2456 // round up to nearest 16 frames to satisfy AudioMixer 2457 if (mType == MIXER || mType == DUPLICATING) { 2458 mNormalFrameCount = (mNormalFrameCount + 15) & ~15; 2459 } 2460 ALOGI("HAL output buffer size %zu frames, normal sink buffer size %zu frames", mFrameCount, 2461 mNormalFrameCount); 2462 2463 // Check if we want to throttle the processing to no more than 2x normal rate 2464 mThreadThrottle = property_get_bool("af.thread.throttle", true /* default_value */); 2465 mThreadThrottleTimeMs = 0; 2466 mThreadThrottleEndMs = 0; 2467 mHalfBufferMs = mNormalFrameCount * 1000 / (2 * mSampleRate); 2468 2469 // mSinkBuffer is the sink buffer. Size is always multiple-of-16 frames. 2470 // Originally this was int16_t[] array, need to remove legacy implications. 2471 free(mSinkBuffer); 2472 mSinkBuffer = NULL; 2473 // For sink buffer size, we use the frame size from the downstream sink to avoid problems 2474 // with non PCM formats for compressed music, e.g. AAC, and Offload threads. 2475 const size_t sinkBufferSize = mNormalFrameCount * mFrameSize; 2476 (void)posix_memalign(&mSinkBuffer, 32, sinkBufferSize); 2477 2478 // We resize the mMixerBuffer according to the requirements of the sink buffer which 2479 // drives the output. 2480 free(mMixerBuffer); 2481 mMixerBuffer = NULL; 2482 if (mMixerBufferEnabled) { 2483 mMixerBufferFormat = AUDIO_FORMAT_PCM_FLOAT; // also valid: AUDIO_FORMAT_PCM_16_BIT. 2484 mMixerBufferSize = mNormalFrameCount * mChannelCount 2485 * audio_bytes_per_sample(mMixerBufferFormat); 2486 (void)posix_memalign(&mMixerBuffer, 32, mMixerBufferSize); 2487 } 2488 free(mEffectBuffer); 2489 mEffectBuffer = NULL; 2490 if (mEffectBufferEnabled) { 2491 mEffectBufferFormat = AUDIO_FORMAT_PCM_16_BIT; // Note: Effects support 16b only 2492 mEffectBufferSize = mNormalFrameCount * mChannelCount 2493 * audio_bytes_per_sample(mEffectBufferFormat); 2494 (void)posix_memalign(&mEffectBuffer, 32, mEffectBufferSize); 2495 } 2496 2497 // force reconfiguration of effect chains and engines to take new buffer size and audio 2498 // parameters into account 2499 // Note that mLock is not held when readOutputParameters_l() is called from the constructor 2500 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 2501 // matter. 2502 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 2503 Vector< sp<EffectChain> > effectChains = mEffectChains; 2504 for (size_t i = 0; i < effectChains.size(); i ++) { 2505 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 2506 } 2507} 2508 2509 2510status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames) 2511{ 2512 if (halFrames == NULL || dspFrames == NULL) { 2513 return BAD_VALUE; 2514 } 2515 Mutex::Autolock _l(mLock); 2516 if (initCheck() != NO_ERROR) { 2517 return INVALID_OPERATION; 2518 } 2519 int64_t framesWritten = mBytesWritten / mFrameSize; 2520 *halFrames = framesWritten; 2521 2522 if (isSuspended()) { 2523 // return an estimation of rendered frames when the output is suspended 2524 size_t latencyFrames = (latency_l() * mSampleRate) / 1000; 2525 *dspFrames = (uint32_t) 2526 (framesWritten >= (int64_t)latencyFrames ? framesWritten - latencyFrames : 0); 2527 return NO_ERROR; 2528 } else { 2529 status_t status; 2530 uint32_t frames; 2531 status = mOutput->getRenderPosition(&frames); 2532 *dspFrames = (size_t)frames; 2533 return status; 2534 } 2535} 2536 2537// hasAudioSession_l() must be called with ThreadBase::mLock held 2538uint32_t AudioFlinger::PlaybackThread::hasAudioSession_l(audio_session_t sessionId) const 2539{ 2540 uint32_t result = 0; 2541 if (getEffectChain_l(sessionId) != 0) { 2542 result = EFFECT_SESSION; 2543 } 2544 2545 for (size_t i = 0; i < mTracks.size(); ++i) { 2546 sp<Track> track = mTracks[i]; 2547 if (sessionId == track->sessionId() && !track->isInvalid()) { 2548 result |= TRACK_SESSION; 2549 if (track->isFastTrack()) { 2550 result |= FAST_SESSION; 2551 } 2552 break; 2553 } 2554 } 2555 2556 return result; 2557} 2558 2559uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(audio_session_t sessionId) 2560{ 2561 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 2562 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 2563 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 2564 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 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 return AudioSystem::getStrategyForStream(track->streamType()); 2570 } 2571 } 2572 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 2573} 2574 2575 2576AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 2577{ 2578 Mutex::Autolock _l(mLock); 2579 return mOutput; 2580} 2581 2582AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 2583{ 2584 Mutex::Autolock _l(mLock); 2585 AudioStreamOut *output = mOutput; 2586 mOutput = NULL; 2587 // FIXME FastMixer might also have a raw ptr to mOutputSink; 2588 // must push a NULL and wait for ack 2589 mOutputSink.clear(); 2590 mPipeSink.clear(); 2591 mNormalSink.clear(); 2592 return output; 2593} 2594 2595// this method must always be called either with ThreadBase mLock held or inside the thread loop 2596audio_stream_t* AudioFlinger::PlaybackThread::stream() const 2597{ 2598 if (mOutput == NULL) { 2599 return NULL; 2600 } 2601 return &mOutput->stream->common; 2602} 2603 2604uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() const 2605{ 2606 return (uint32_t)((uint32_t)((mNormalFrameCount * 1000) / mSampleRate) * 1000); 2607} 2608 2609status_t AudioFlinger::PlaybackThread::setSyncEvent(const sp<SyncEvent>& event) 2610{ 2611 if (!isValidSyncEvent(event)) { 2612 return BAD_VALUE; 2613 } 2614 2615 Mutex::Autolock _l(mLock); 2616 2617 for (size_t i = 0; i < mTracks.size(); ++i) { 2618 sp<Track> track = mTracks[i]; 2619 if (event->triggerSession() == track->sessionId()) { 2620 (void) track->setSyncEvent(event); 2621 return NO_ERROR; 2622 } 2623 } 2624 2625 return NAME_NOT_FOUND; 2626} 2627 2628bool AudioFlinger::PlaybackThread::isValidSyncEvent(const sp<SyncEvent>& event) const 2629{ 2630 return event->type() == AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE; 2631} 2632 2633void AudioFlinger::PlaybackThread::threadLoop_removeTracks( 2634 const Vector< sp<Track> >& tracksToRemove) 2635{ 2636 size_t count = tracksToRemove.size(); 2637 if (count > 0) { 2638 for (size_t i = 0 ; i < count ; i++) { 2639 const sp<Track>& track = tracksToRemove.itemAt(i); 2640 if (track->isExternalTrack()) { 2641 AudioSystem::stopOutput(mId, track->streamType(), 2642 track->sessionId()); 2643#ifdef ADD_BATTERY_DATA 2644 // to track the speaker usage 2645 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 2646#endif 2647 if (track->isTerminated()) { 2648 AudioSystem::releaseOutput(mId, track->streamType(), 2649 track->sessionId()); 2650 } 2651 } 2652 } 2653 } 2654} 2655 2656void AudioFlinger::PlaybackThread::checkSilentMode_l() 2657{ 2658 if (!mMasterMute) { 2659 char value[PROPERTY_VALUE_MAX]; 2660 if (mOutDevice == AUDIO_DEVICE_OUT_REMOTE_SUBMIX) { 2661 ALOGD("ro.audio.silent will be ignored for threads on AUDIO_DEVICE_OUT_REMOTE_SUBMIX"); 2662 return; 2663 } 2664 if (property_get("ro.audio.silent", value, "0") > 0) { 2665 char *endptr; 2666 unsigned long ul = strtoul(value, &endptr, 0); 2667 if (*endptr == '\0' && ul != 0) { 2668 ALOGD("Silence is golden"); 2669 // The setprop command will not allow a property to be changed after 2670 // the first time it is set, so we don't have to worry about un-muting. 2671 setMasterMute_l(true); 2672 } 2673 } 2674 } 2675} 2676 2677// shared by MIXER and DIRECT, overridden by DUPLICATING 2678ssize_t AudioFlinger::PlaybackThread::threadLoop_write() 2679{ 2680 mInWrite = true; 2681 ssize_t bytesWritten; 2682 const size_t offset = mCurrentWriteLength - mBytesRemaining; 2683 2684 // If an NBAIO sink is present, use it to write the normal mixer's submix 2685 if (mNormalSink != 0) { 2686 2687 const size_t count = mBytesRemaining / mFrameSize; 2688 2689 ATRACE_BEGIN("write"); 2690 // update the setpoint when AudioFlinger::mScreenState changes 2691 uint32_t screenState = AudioFlinger::mScreenState; 2692 if (screenState != mScreenState) { 2693 mScreenState = screenState; 2694 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 2695 if (pipe != NULL) { 2696 pipe->setAvgFrames((mScreenState & 1) ? 2697 (pipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2698 } 2699 } 2700 ssize_t framesWritten = mNormalSink->write((char *)mSinkBuffer + offset, count); 2701 ATRACE_END(); 2702 if (framesWritten > 0) { 2703 bytesWritten = framesWritten * mFrameSize; 2704 } else { 2705 bytesWritten = framesWritten; 2706 } 2707 // otherwise use the HAL / AudioStreamOut directly 2708 } else { 2709 // Direct output and offload threads 2710 2711 if (mUseAsyncWrite) { 2712 ALOGW_IF(mWriteAckSequence & 1, "threadLoop_write(): out of sequence write request"); 2713 mWriteAckSequence += 2; 2714 mWriteAckSequence |= 1; 2715 ALOG_ASSERT(mCallbackThread != 0); 2716 mCallbackThread->setWriteBlocked(mWriteAckSequence); 2717 } 2718 // FIXME We should have an implementation of timestamps for direct output threads. 2719 // They are used e.g for multichannel PCM playback over HDMI. 2720 bytesWritten = mOutput->write((char *)mSinkBuffer + offset, mBytesRemaining); 2721 2722 if (mUseAsyncWrite && 2723 ((bytesWritten < 0) || (bytesWritten == (ssize_t)mBytesRemaining))) { 2724 // do not wait for async callback in case of error of full write 2725 mWriteAckSequence &= ~1; 2726 ALOG_ASSERT(mCallbackThread != 0); 2727 mCallbackThread->setWriteBlocked(mWriteAckSequence); 2728 } 2729 } 2730 2731 mNumWrites++; 2732 mInWrite = false; 2733 mStandby = false; 2734 return bytesWritten; 2735} 2736 2737void AudioFlinger::PlaybackThread::threadLoop_drain() 2738{ 2739 if (mOutput->stream->drain) { 2740 ALOGV("draining %s", (mMixerStatus == MIXER_DRAIN_TRACK) ? "early" : "full"); 2741 if (mUseAsyncWrite) { 2742 ALOGW_IF(mDrainSequence & 1, "threadLoop_drain(): out of sequence drain request"); 2743 mDrainSequence |= 1; 2744 ALOG_ASSERT(mCallbackThread != 0); 2745 mCallbackThread->setDraining(mDrainSequence); 2746 } 2747 mOutput->stream->drain(mOutput->stream, 2748 (mMixerStatus == MIXER_DRAIN_TRACK) ? AUDIO_DRAIN_EARLY_NOTIFY 2749 : AUDIO_DRAIN_ALL); 2750 } 2751} 2752 2753void AudioFlinger::PlaybackThread::threadLoop_exit() 2754{ 2755 { 2756 Mutex::Autolock _l(mLock); 2757 for (size_t i = 0; i < mTracks.size(); i++) { 2758 sp<Track> track = mTracks[i]; 2759 track->invalidate(); 2760 } 2761 } 2762} 2763 2764/* 2765The derived values that are cached: 2766 - mSinkBufferSize from frame count * frame size 2767 - mActiveSleepTimeUs from activeSleepTimeUs() 2768 - mIdleSleepTimeUs from idleSleepTimeUs() 2769 - mStandbyDelayNs from mActiveSleepTimeUs (DIRECT only) or forced to at least 2770 kDefaultStandbyTimeInNsecs when connected to an A2DP device. 2771 - maxPeriod from frame count and sample rate (MIXER only) 2772 2773The parameters that affect these derived values are: 2774 - frame count 2775 - frame size 2776 - sample rate 2777 - device type: A2DP or not 2778 - device latency 2779 - format: PCM or not 2780 - active sleep time 2781 - idle sleep time 2782*/ 2783 2784void AudioFlinger::PlaybackThread::cacheParameters_l() 2785{ 2786 mSinkBufferSize = mNormalFrameCount * mFrameSize; 2787 mActiveSleepTimeUs = activeSleepTimeUs(); 2788 mIdleSleepTimeUs = idleSleepTimeUs(); 2789 2790 // make sure standby delay is not too short when connected to an A2DP sink to avoid 2791 // truncating audio when going to standby. 2792 mStandbyDelayNs = AudioFlinger::mStandbyTimeInNsecs; 2793 if ((mOutDevice & AUDIO_DEVICE_OUT_ALL_A2DP) != 0) { 2794 if (mStandbyDelayNs < kDefaultStandbyTimeInNsecs) { 2795 mStandbyDelayNs = kDefaultStandbyTimeInNsecs; 2796 } 2797 } 2798} 2799 2800bool AudioFlinger::PlaybackThread::invalidateTracks_l(audio_stream_type_t streamType) 2801{ 2802 ALOGV("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %zu", 2803 this, streamType, mTracks.size()); 2804 bool trackMatch = false; 2805 size_t size = mTracks.size(); 2806 for (size_t i = 0; i < size; i++) { 2807 sp<Track> t = mTracks[i]; 2808 if (t->streamType() == streamType && t->isExternalTrack()) { 2809 t->invalidate(); 2810 trackMatch = true; 2811 } 2812 } 2813 return trackMatch; 2814} 2815 2816void AudioFlinger::PlaybackThread::invalidateTracks(audio_stream_type_t streamType) 2817{ 2818 Mutex::Autolock _l(mLock); 2819 invalidateTracks_l(streamType); 2820} 2821 2822status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 2823{ 2824 audio_session_t session = chain->sessionId(); 2825 int16_t* buffer = reinterpret_cast<int16_t*>(mEffectBufferEnabled 2826 ? mEffectBuffer : mSinkBuffer); 2827 bool ownsBuffer = false; 2828 2829 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 2830 if (session > AUDIO_SESSION_OUTPUT_MIX) { 2831 // Only one effect chain can be present in direct output thread and it uses 2832 // the sink buffer as input 2833 if (mType != DIRECT) { 2834 size_t numSamples = mNormalFrameCount * mChannelCount; 2835 buffer = new int16_t[numSamples]; 2836 memset(buffer, 0, numSamples * sizeof(int16_t)); 2837 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 2838 ownsBuffer = true; 2839 } 2840 2841 // Attach all tracks with same session ID to this chain. 2842 for (size_t i = 0; i < mTracks.size(); ++i) { 2843 sp<Track> track = mTracks[i]; 2844 if (session == track->sessionId()) { 2845 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), 2846 buffer); 2847 track->setMainBuffer(buffer); 2848 chain->incTrackCnt(); 2849 } 2850 } 2851 2852 // indicate all active tracks in the chain 2853 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 2854 sp<Track> track = mActiveTracks[i].promote(); 2855 if (track == 0) { 2856 continue; 2857 } 2858 if (session == track->sessionId()) { 2859 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 2860 chain->incActiveTrackCnt(); 2861 } 2862 } 2863 } 2864 chain->setThread(this); 2865 chain->setInBuffer(buffer, ownsBuffer); 2866 chain->setOutBuffer(reinterpret_cast<int16_t*>(mEffectBufferEnabled 2867 ? mEffectBuffer : mSinkBuffer)); 2868 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 2869 // chains list in order to be processed last as it contains output stage effects. 2870 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 2871 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 2872 // after track specific effects and before output stage. 2873 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 2874 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX. 2875 // Effect chain for other sessions are inserted at beginning of effect 2876 // chains list to be processed before output mix effects. Relative order between other 2877 // sessions is not important. 2878 static_assert(AUDIO_SESSION_OUTPUT_MIX == 0 && 2879 AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX, 2880 "audio_session_t constants misdefined"); 2881 size_t size = mEffectChains.size(); 2882 size_t i = 0; 2883 for (i = 0; i < size; i++) { 2884 if (mEffectChains[i]->sessionId() < session) { 2885 break; 2886 } 2887 } 2888 mEffectChains.insertAt(chain, i); 2889 checkSuspendOnAddEffectChain_l(chain); 2890 2891 return NO_ERROR; 2892} 2893 2894size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 2895{ 2896 audio_session_t session = chain->sessionId(); 2897 2898 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 2899 2900 for (size_t i = 0; i < mEffectChains.size(); i++) { 2901 if (chain == mEffectChains[i]) { 2902 mEffectChains.removeAt(i); 2903 // detach all active tracks from the chain 2904 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 2905 sp<Track> track = mActiveTracks[i].promote(); 2906 if (track == 0) { 2907 continue; 2908 } 2909 if (session == track->sessionId()) { 2910 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 2911 chain.get(), session); 2912 chain->decActiveTrackCnt(); 2913 } 2914 } 2915 2916 // detach all tracks with same session ID from this chain 2917 for (size_t i = 0; i < mTracks.size(); ++i) { 2918 sp<Track> track = mTracks[i]; 2919 if (session == track->sessionId()) { 2920 track->setMainBuffer(reinterpret_cast<int16_t*>(mSinkBuffer)); 2921 chain->decTrackCnt(); 2922 } 2923 } 2924 break; 2925 } 2926 } 2927 return mEffectChains.size(); 2928} 2929 2930status_t AudioFlinger::PlaybackThread::attachAuxEffect( 2931 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2932{ 2933 Mutex::Autolock _l(mLock); 2934 return attachAuxEffect_l(track, EffectId); 2935} 2936 2937status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 2938 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2939{ 2940 status_t status = NO_ERROR; 2941 2942 if (EffectId == 0) { 2943 track->setAuxBuffer(0, NULL); 2944 } else { 2945 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 2946 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 2947 if (effect != 0) { 2948 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 2949 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 2950 } else { 2951 status = INVALID_OPERATION; 2952 } 2953 } else { 2954 status = BAD_VALUE; 2955 } 2956 } 2957 return status; 2958} 2959 2960void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 2961{ 2962 for (size_t i = 0; i < mTracks.size(); ++i) { 2963 sp<Track> track = mTracks[i]; 2964 if (track->auxEffectId() == effectId) { 2965 attachAuxEffect_l(track, 0); 2966 } 2967 } 2968} 2969 2970bool AudioFlinger::PlaybackThread::threadLoop() 2971{ 2972 Vector< sp<Track> > tracksToRemove; 2973 2974 mStandbyTimeNs = systemTime(); 2975 nsecs_t lastWriteFinished = -1; // time last server write completed 2976 int64_t lastFramesWritten = -1; // track changes in timestamp server frames written 2977 2978 // MIXER 2979 nsecs_t lastWarning = 0; 2980 2981 // DUPLICATING 2982 // FIXME could this be made local to while loop? 2983 writeFrames = 0; 2984 2985 int lastGeneration = 0; 2986 2987 cacheParameters_l(); 2988 mSleepTimeUs = mIdleSleepTimeUs; 2989 2990 if (mType == MIXER) { 2991 sleepTimeShift = 0; 2992 } 2993 2994 CpuStats cpuStats; 2995 const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid())); 2996 2997 acquireWakeLock(); 2998 2999 // mNBLogWriter->log can only be called while thread mutex mLock is held. 3000 // So if you need to log when mutex is unlocked, set logString to a non-NULL string, 3001 // and then that string will be logged at the next convenient opportunity. 3002 const char *logString = NULL; 3003 3004 checkSilentMode_l(); 3005 3006 while (!exitPending()) 3007 { 3008 cpuStats.sample(myName); 3009 3010 Vector< sp<EffectChain> > effectChains; 3011 3012 { // scope for mLock 3013 3014 Mutex::Autolock _l(mLock); 3015 3016 processConfigEvents_l(); 3017 3018 if (logString != NULL) { 3019 mNBLogWriter->logTimestamp(); 3020 mNBLogWriter->log(logString); 3021 logString = NULL; 3022 } 3023 3024 // Gather the framesReleased counters for all active tracks, 3025 // and associate with the sink frames written out. We need 3026 // this to convert the sink timestamp to the track timestamp. 3027 bool kernelLocationUpdate = false; 3028 if (mNormalSink != 0) { 3029 // Note: The DuplicatingThread may not have a mNormalSink. 3030 // We always fetch the timestamp here because often the downstream 3031 // sink will block while writing. 3032 ExtendedTimestamp timestamp; // use private copy to fetch 3033 (void) mNormalSink->getTimestamp(timestamp); 3034 3035 // We keep track of the last valid kernel position in case we are in underrun 3036 // and the normal mixer period is the same as the fast mixer period, or there 3037 // is some error from the HAL. 3038 if (mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL] >= 0) { 3039 mTimestamp.mPosition[ExtendedTimestamp::LOCATION_KERNEL_LASTKERNELOK] = 3040 mTimestamp.mPosition[ExtendedTimestamp::LOCATION_KERNEL]; 3041 mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL_LASTKERNELOK] = 3042 mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL]; 3043 3044 mTimestamp.mPosition[ExtendedTimestamp::LOCATION_SERVER_LASTKERNELOK] = 3045 mTimestamp.mPosition[ExtendedTimestamp::LOCATION_SERVER]; 3046 mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_SERVER_LASTKERNELOK] = 3047 mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_SERVER]; 3048 } 3049 3050 if (timestamp.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL] >= 0) { 3051 kernelLocationUpdate = true; 3052 } else { 3053 ALOGVV("getTimestamp error - no valid kernel position"); 3054 } 3055 3056 // copy over kernel info 3057 mTimestamp.mPosition[ExtendedTimestamp::LOCATION_KERNEL] = 3058 timestamp.mPosition[ExtendedTimestamp::LOCATION_KERNEL]; 3059 mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL] = 3060 timestamp.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL]; 3061 } 3062 // mFramesWritten for non-offloaded tracks are contiguous 3063 // even after standby() is called. This is useful for the track frame 3064 // to sink frame mapping. 3065 bool serverLocationUpdate = false; 3066 if (mFramesWritten != lastFramesWritten) { 3067 serverLocationUpdate = true; 3068 lastFramesWritten = mFramesWritten; 3069 } 3070 // Only update timestamps if there is a meaningful change. 3071 // Either the kernel timestamp must be valid or we have written something. 3072 if (kernelLocationUpdate || serverLocationUpdate) { 3073 if (serverLocationUpdate) { 3074 // use the time before we called the HAL write - it is a bit more accurate 3075 // to when the server last read data than the current time here. 3076 // 3077 // If we haven't written anything, mLastWriteTime will be -1 3078 // and we use systemTime(). 3079 mTimestamp.mPosition[ExtendedTimestamp::LOCATION_SERVER] = mFramesWritten; 3080 mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_SERVER] = mLastWriteTime == -1 3081 ? systemTime() : mLastWriteTime; 3082 } 3083 const size_t size = mActiveTracks.size(); 3084 for (size_t i = 0; i < size; ++i) { 3085 sp<Track> t = mActiveTracks[i].promote(); 3086 if (t != 0 && !t->isFastTrack()) { 3087 t->updateTrackFrameInfo( 3088 t->mAudioTrackServerProxy->framesReleased(), 3089 mFramesWritten, 3090 mTimestamp); 3091 } 3092 } 3093 } 3094 3095 saveOutputTracks(); 3096 if (mSignalPending) { 3097 // A signal was raised while we were unlocked 3098 mSignalPending = false; 3099 } else if (waitingAsyncCallback_l()) { 3100 if (exitPending()) { 3101 break; 3102 } 3103 bool released = false; 3104 if (!keepWakeLock()) { 3105 releaseWakeLock_l(); 3106 released = true; 3107 } 3108 mWakeLockUids.clear(); 3109 mActiveTracksGeneration++; 3110 ALOGV("wait async completion"); 3111 mWaitWorkCV.wait(mLock); 3112 ALOGV("async completion/wake"); 3113 if (released) { 3114 acquireWakeLock_l(); 3115 } 3116 mStandbyTimeNs = systemTime() + mStandbyDelayNs; 3117 mSleepTimeUs = 0; 3118 3119 continue; 3120 } 3121 if ((!mActiveTracks.size() && systemTime() > mStandbyTimeNs) || 3122 isSuspended()) { 3123 // put audio hardware into standby after short delay 3124 if (shouldStandby_l()) { 3125 3126 threadLoop_standby(); 3127 3128 mStandby = true; 3129 } 3130 3131 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 3132 // we're about to wait, flush the binder command buffer 3133 IPCThreadState::self()->flushCommands(); 3134 3135 clearOutputTracks(); 3136 3137 if (exitPending()) { 3138 break; 3139 } 3140 3141 releaseWakeLock_l(); 3142 mWakeLockUids.clear(); 3143 mActiveTracksGeneration++; 3144 // wait until we have something to do... 3145 ALOGV("%s going to sleep", myName.string()); 3146 mWaitWorkCV.wait(mLock); 3147 ALOGV("%s waking up", myName.string()); 3148 acquireWakeLock_l(); 3149 3150 mMixerStatus = MIXER_IDLE; 3151 mMixerStatusIgnoringFastTracks = MIXER_IDLE; 3152 mBytesWritten = 0; 3153 mBytesRemaining = 0; 3154 checkSilentMode_l(); 3155 3156 mStandbyTimeNs = systemTime() + mStandbyDelayNs; 3157 mSleepTimeUs = mIdleSleepTimeUs; 3158 if (mType == MIXER) { 3159 sleepTimeShift = 0; 3160 } 3161 3162 continue; 3163 } 3164 } 3165 // mMixerStatusIgnoringFastTracks is also updated internally 3166 mMixerStatus = prepareTracks_l(&tracksToRemove); 3167 3168 // compare with previously applied list 3169 if (lastGeneration != mActiveTracksGeneration) { 3170 // update wakelock 3171 updateWakeLockUids_l(mWakeLockUids); 3172 lastGeneration = mActiveTracksGeneration; 3173 } 3174 3175 // prevent any changes in effect chain list and in each effect chain 3176 // during mixing and effect process as the audio buffers could be deleted 3177 // or modified if an effect is created or deleted 3178 lockEffectChains_l(effectChains); 3179 } // mLock scope ends 3180 3181 if (mBytesRemaining == 0) { 3182 mCurrentWriteLength = 0; 3183 if (mMixerStatus == MIXER_TRACKS_READY) { 3184 // threadLoop_mix() sets mCurrentWriteLength 3185 threadLoop_mix(); 3186 } else if ((mMixerStatus != MIXER_DRAIN_TRACK) 3187 && (mMixerStatus != MIXER_DRAIN_ALL)) { 3188 // threadLoop_sleepTime sets mSleepTimeUs to 0 if data 3189 // must be written to HAL 3190 threadLoop_sleepTime(); 3191 if (mSleepTimeUs == 0) { 3192 mCurrentWriteLength = mSinkBufferSize; 3193 } 3194 } 3195 // Either threadLoop_mix() or threadLoop_sleepTime() should have set 3196 // mMixerBuffer with data if mMixerBufferValid is true and mSleepTimeUs == 0. 3197 // Merge mMixerBuffer data into mEffectBuffer (if any effects are valid) 3198 // or mSinkBuffer (if there are no effects). 3199 // 3200 // This is done pre-effects computation; if effects change to 3201 // support higher precision, this needs to move. 3202 // 3203 // mMixerBufferValid is only set true by MixerThread::prepareTracks_l(). 3204 // TODO use mSleepTimeUs == 0 as an additional condition. 3205 if (mMixerBufferValid) { 3206 void *buffer = mEffectBufferValid ? mEffectBuffer : mSinkBuffer; 3207 audio_format_t format = mEffectBufferValid ? mEffectBufferFormat : mFormat; 3208 3209 // mono blend occurs for mixer threads only (not direct or offloaded) 3210 // and is handled here if we're going directly to the sink. 3211 if (requireMonoBlend() && !mEffectBufferValid) { 3212 mono_blend(mMixerBuffer, mMixerBufferFormat, mChannelCount, mNormalFrameCount, 3213 true /*limit*/); 3214 } 3215 3216 memcpy_by_audio_format(buffer, format, mMixerBuffer, mMixerBufferFormat, 3217 mNormalFrameCount * mChannelCount); 3218 } 3219 3220 mBytesRemaining = mCurrentWriteLength; 3221 if (isSuspended()) { 3222 mSleepTimeUs = suspendSleepTimeUs(); 3223 // simulate write to HAL when suspended 3224 mBytesWritten += mSinkBufferSize; 3225 mFramesWritten += mSinkBufferSize / mFrameSize; 3226 mBytesRemaining = 0; 3227 } 3228 3229 // only process effects if we're going to write 3230 if (mSleepTimeUs == 0 && mType != OFFLOAD) { 3231 for (size_t i = 0; i < effectChains.size(); i ++) { 3232 effectChains[i]->process_l(); 3233 } 3234 } 3235 } 3236 // Process effect chains for offloaded thread even if no audio 3237 // was read from audio track: process only updates effect state 3238 // and thus does have to be synchronized with audio writes but may have 3239 // to be called while waiting for async write callback 3240 if (mType == OFFLOAD) { 3241 for (size_t i = 0; i < effectChains.size(); i ++) { 3242 effectChains[i]->process_l(); 3243 } 3244 } 3245 3246 // Only if the Effects buffer is enabled and there is data in the 3247 // Effects buffer (buffer valid), we need to 3248 // copy into the sink buffer. 3249 // TODO use mSleepTimeUs == 0 as an additional condition. 3250 if (mEffectBufferValid) { 3251 //ALOGV("writing effect buffer to sink buffer format %#x", mFormat); 3252 3253 if (requireMonoBlend()) { 3254 mono_blend(mEffectBuffer, mEffectBufferFormat, mChannelCount, mNormalFrameCount, 3255 true /*limit*/); 3256 } 3257 3258 memcpy_by_audio_format(mSinkBuffer, mFormat, mEffectBuffer, mEffectBufferFormat, 3259 mNormalFrameCount * mChannelCount); 3260 } 3261 3262 // enable changes in effect chain 3263 unlockEffectChains(effectChains); 3264 3265 if (!waitingAsyncCallback()) { 3266 // mSleepTimeUs == 0 means we must write to audio hardware 3267 if (mSleepTimeUs == 0) { 3268 ssize_t ret = 0; 3269 // We save lastWriteFinished here, as previousLastWriteFinished, 3270 // for throttling. On thread start, previousLastWriteFinished will be 3271 // set to -1, which properly results in no throttling after the first write. 3272 nsecs_t previousLastWriteFinished = lastWriteFinished; 3273 nsecs_t delta = 0; 3274 if (mBytesRemaining) { 3275 // FIXME rewrite to reduce number of system calls 3276 mLastWriteTime = systemTime(); // also used for dumpsys 3277 ret = threadLoop_write(); 3278 lastWriteFinished = systemTime(); 3279 delta = lastWriteFinished - mLastWriteTime; 3280 if (ret < 0) { 3281 mBytesRemaining = 0; 3282 } else { 3283 mBytesWritten += ret; 3284 mBytesRemaining -= ret; 3285 mFramesWritten += ret / mFrameSize; 3286 } 3287 } else if ((mMixerStatus == MIXER_DRAIN_TRACK) || 3288 (mMixerStatus == MIXER_DRAIN_ALL)) { 3289 threadLoop_drain(); 3290 } 3291 if (mType == MIXER && !mStandby) { 3292 // write blocked detection 3293 if (delta > maxPeriod) { 3294 mNumDelayedWrites++; 3295 if ((lastWriteFinished - lastWarning) > kWarningThrottleNs) { 3296 ATRACE_NAME("underrun"); 3297 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 3298 (unsigned long long) ns2ms(delta), mNumDelayedWrites, this); 3299 lastWarning = lastWriteFinished; 3300 } 3301 } 3302 3303 if (mThreadThrottle 3304 && mMixerStatus == MIXER_TRACKS_READY // we are mixing (active tracks) 3305 && ret > 0) { // we wrote something 3306 // Limit MixerThread data processing to no more than twice the 3307 // expected processing rate. 3308 // 3309 // This helps prevent underruns with NuPlayer and other applications 3310 // which may set up buffers that are close to the minimum size, or use 3311 // deep buffers, and rely on a double-buffering sleep strategy to fill. 3312 // 3313 // The throttle smooths out sudden large data drains from the device, 3314 // e.g. when it comes out of standby, which often causes problems with 3315 // (1) mixer threads without a fast mixer (which has its own warm-up) 3316 // (2) minimum buffer sized tracks (even if the track is full, 3317 // the app won't fill fast enough to handle the sudden draw). 3318 // 3319 // Total time spent in last processing cycle equals time spent in 3320 // 1. threadLoop_write, as well as time spent in 3321 // 2. threadLoop_mix (significant for heavy mixing, especially 3322 // on low tier processors) 3323 3324 // it's OK if deltaMs is an overestimate. 3325 const int32_t deltaMs = 3326 (lastWriteFinished - previousLastWriteFinished) / 1000000; 3327 const int32_t throttleMs = mHalfBufferMs - deltaMs; 3328 if ((signed)mHalfBufferMs >= throttleMs && throttleMs > 0) { 3329 usleep(throttleMs * 1000); 3330 // notify of throttle start on verbose log 3331 ALOGV_IF(mThreadThrottleEndMs == mThreadThrottleTimeMs, 3332 "mixer(%p) throttle begin:" 3333 " ret(%zd) deltaMs(%d) requires sleep %d ms", 3334 this, ret, deltaMs, throttleMs); 3335 mThreadThrottleTimeMs += throttleMs; 3336 // Throttle must be attributed to the previous mixer loop's write time 3337 // to allow back-to-back throttling. 3338 lastWriteFinished += throttleMs * 1000000; 3339 } else { 3340 uint32_t diff = mThreadThrottleTimeMs - mThreadThrottleEndMs; 3341 if (diff > 0) { 3342 // notify of throttle end on debug log 3343 // but prevent spamming for bluetooth 3344 ALOGD_IF(!audio_is_a2dp_out_device(outDevice()), 3345 "mixer(%p) throttle end: throttle time(%u)", this, diff); 3346 mThreadThrottleEndMs = mThreadThrottleTimeMs; 3347 } 3348 } 3349 } 3350 } 3351 3352 } else { 3353 ATRACE_BEGIN("sleep"); 3354 Mutex::Autolock _l(mLock); 3355 if (!mSignalPending && mConfigEvents.isEmpty() && !exitPending()) { 3356 mWaitWorkCV.waitRelative(mLock, microseconds((nsecs_t)mSleepTimeUs)); 3357 } 3358 ATRACE_END(); 3359 } 3360 } 3361 3362 // Finally let go of removed track(s), without the lock held 3363 // since we can't guarantee the destructors won't acquire that 3364 // same lock. This will also mutate and push a new fast mixer state. 3365 threadLoop_removeTracks(tracksToRemove); 3366 tracksToRemove.clear(); 3367 3368 // FIXME I don't understand the need for this here; 3369 // it was in the original code but maybe the 3370 // assignment in saveOutputTracks() makes this unnecessary? 3371 clearOutputTracks(); 3372 3373 // Effect chains will be actually deleted here if they were removed from 3374 // mEffectChains list during mixing or effects processing 3375 effectChains.clear(); 3376 3377 // FIXME Note that the above .clear() is no longer necessary since effectChains 3378 // is now local to this block, but will keep it for now (at least until merge done). 3379 } 3380 3381 threadLoop_exit(); 3382 3383 if (!mStandby) { 3384 threadLoop_standby(); 3385 mStandby = true; 3386 } 3387 3388 releaseWakeLock(); 3389 mWakeLockUids.clear(); 3390 mActiveTracksGeneration++; 3391 3392 ALOGV("Thread %p type %d exiting", this, mType); 3393 return false; 3394} 3395 3396// removeTracks_l() must be called with ThreadBase::mLock held 3397void AudioFlinger::PlaybackThread::removeTracks_l(const Vector< sp<Track> >& tracksToRemove) 3398{ 3399 size_t count = tracksToRemove.size(); 3400 if (count > 0) { 3401 for (size_t i=0 ; i<count ; i++) { 3402 const sp<Track>& track = tracksToRemove.itemAt(i); 3403 mActiveTracks.remove(track); 3404 mWakeLockUids.remove(track->uid()); 3405 mActiveTracksGeneration++; 3406 ALOGV("removeTracks_l removing track on session %d", track->sessionId()); 3407 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 3408 if (chain != 0) { 3409 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), 3410 track->sessionId()); 3411 chain->decActiveTrackCnt(); 3412 } 3413 if (track->isTerminated()) { 3414 removeTrack_l(track); 3415 } 3416 } 3417 } 3418 3419} 3420 3421status_t AudioFlinger::PlaybackThread::getTimestamp_l(AudioTimestamp& timestamp) 3422{ 3423 if (mNormalSink != 0) { 3424 ExtendedTimestamp ets; 3425 status_t status = mNormalSink->getTimestamp(ets); 3426 if (status == NO_ERROR) { 3427 status = ets.getBestTimestamp(×tamp); 3428 } 3429 return status; 3430 } 3431 if ((mType == OFFLOAD || mType == DIRECT) 3432 && mOutput != NULL && mOutput->stream->get_presentation_position) { 3433 uint64_t position64; 3434 int ret = mOutput->getPresentationPosition(&position64, ×tamp.mTime); 3435 if (ret == 0) { 3436 timestamp.mPosition = (uint32_t)position64; 3437 return NO_ERROR; 3438 } 3439 } 3440 return INVALID_OPERATION; 3441} 3442 3443status_t AudioFlinger::MixerThread::createAudioPatch_l(const struct audio_patch *patch, 3444 audio_patch_handle_t *handle) 3445{ 3446 AutoPark<FastMixer> park(mFastMixer); 3447 3448 status_t status = PlaybackThread::createAudioPatch_l(patch, handle); 3449 3450 return status; 3451} 3452 3453status_t AudioFlinger::PlaybackThread::createAudioPatch_l(const struct audio_patch *patch, 3454 audio_patch_handle_t *handle) 3455{ 3456 status_t status = NO_ERROR; 3457 3458 // store new device and send to effects 3459 audio_devices_t type = AUDIO_DEVICE_NONE; 3460 for (unsigned int i = 0; i < patch->num_sinks; i++) { 3461 type |= patch->sinks[i].ext.device.type; 3462 } 3463 3464#ifdef ADD_BATTERY_DATA 3465 // when changing the audio output device, call addBatteryData to notify 3466 // the change 3467 if (mOutDevice != type) { 3468 uint32_t params = 0; 3469 // check whether speaker is on 3470 if (type & AUDIO_DEVICE_OUT_SPEAKER) { 3471 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 3472 } 3473 3474 audio_devices_t deviceWithoutSpeaker 3475 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 3476 // check if any other device (except speaker) is on 3477 if (type & deviceWithoutSpeaker) { 3478 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 3479 } 3480 3481 if (params != 0) { 3482 addBatteryData(params); 3483 } 3484 } 3485#endif 3486 3487 for (size_t i = 0; i < mEffectChains.size(); i++) { 3488 mEffectChains[i]->setDevice_l(type); 3489 } 3490 3491 // mPrevOutDevice is the latest device set by createAudioPatch_l(). It is not set when 3492 // the thread is created so that the first patch creation triggers an ioConfigChanged callback 3493 bool configChanged = mPrevOutDevice != type; 3494 mOutDevice = type; 3495 mPatch = *patch; 3496 3497 if (mOutput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 3498 audio_hw_device_t *hwDevice = mOutput->audioHwDev->hwDevice(); 3499 status = hwDevice->create_audio_patch(hwDevice, 3500 patch->num_sources, 3501 patch->sources, 3502 patch->num_sinks, 3503 patch->sinks, 3504 handle); 3505 } else { 3506 char *address; 3507 if (strcmp(patch->sinks[0].ext.device.address, "") != 0) { 3508 //FIXME: we only support address on first sink with HAL version < 3.0 3509 address = audio_device_address_to_parameter( 3510 patch->sinks[0].ext.device.type, 3511 patch->sinks[0].ext.device.address); 3512 } else { 3513 address = (char *)calloc(1, 1); 3514 } 3515 AudioParameter param = AudioParameter(String8(address)); 3516 free(address); 3517 param.addInt(String8(AUDIO_PARAMETER_STREAM_ROUTING), (int)type); 3518 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3519 param.toString().string()); 3520 *handle = AUDIO_PATCH_HANDLE_NONE; 3521 } 3522 if (configChanged) { 3523 mPrevOutDevice = type; 3524 sendIoConfigEvent_l(AUDIO_OUTPUT_CONFIG_CHANGED); 3525 } 3526 return status; 3527} 3528 3529status_t AudioFlinger::MixerThread::releaseAudioPatch_l(const audio_patch_handle_t handle) 3530{ 3531 AutoPark<FastMixer> park(mFastMixer); 3532 3533 status_t status = PlaybackThread::releaseAudioPatch_l(handle); 3534 3535 return status; 3536} 3537 3538status_t AudioFlinger::PlaybackThread::releaseAudioPatch_l(const audio_patch_handle_t handle) 3539{ 3540 status_t status = NO_ERROR; 3541 3542 mOutDevice = AUDIO_DEVICE_NONE; 3543 3544 if (mOutput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 3545 audio_hw_device_t *hwDevice = mOutput->audioHwDev->hwDevice(); 3546 status = hwDevice->release_audio_patch(hwDevice, handle); 3547 } else { 3548 AudioParameter param; 3549 param.addInt(String8(AUDIO_PARAMETER_STREAM_ROUTING), 0); 3550 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3551 param.toString().string()); 3552 } 3553 return status; 3554} 3555 3556void AudioFlinger::PlaybackThread::addPatchTrack(const sp<PatchTrack>& track) 3557{ 3558 Mutex::Autolock _l(mLock); 3559 mTracks.add(track); 3560} 3561 3562void AudioFlinger::PlaybackThread::deletePatchTrack(const sp<PatchTrack>& track) 3563{ 3564 Mutex::Autolock _l(mLock); 3565 destroyTrack_l(track); 3566} 3567 3568void AudioFlinger::PlaybackThread::getAudioPortConfig(struct audio_port_config *config) 3569{ 3570 ThreadBase::getAudioPortConfig(config); 3571 config->role = AUDIO_PORT_ROLE_SOURCE; 3572 config->ext.mix.hw_module = mOutput->audioHwDev->handle(); 3573 config->ext.mix.usecase.stream = AUDIO_STREAM_DEFAULT; 3574} 3575 3576// ---------------------------------------------------------------------------- 3577 3578AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 3579 audio_io_handle_t id, audio_devices_t device, bool systemReady, type_t type) 3580 : PlaybackThread(audioFlinger, output, id, device, type, systemReady), 3581 // mAudioMixer below 3582 // mFastMixer below 3583 mFastMixerFutex(0), 3584 mMasterMono(false) 3585 // mOutputSink below 3586 // mPipeSink below 3587 // mNormalSink below 3588{ 3589 ALOGV("MixerThread() id=%d device=%#x type=%d", id, device, type); 3590 ALOGV("mSampleRate=%u, mChannelMask=%#x, mChannelCount=%u, mFormat=%d, mFrameSize=%zu, " 3591 "mFrameCount=%zu, mNormalFrameCount=%zu", 3592 mSampleRate, mChannelMask, mChannelCount, mFormat, mFrameSize, mFrameCount, 3593 mNormalFrameCount); 3594 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 3595 3596 if (type == DUPLICATING) { 3597 // The Duplicating thread uses the AudioMixer and delivers data to OutputTracks 3598 // (downstream MixerThreads) in DuplicatingThread::threadLoop_write(). 3599 // Do not create or use mFastMixer, mOutputSink, mPipeSink, or mNormalSink. 3600 return; 3601 } 3602 // create an NBAIO sink for the HAL output stream, and negotiate 3603 mOutputSink = new AudioStreamOutSink(output->stream); 3604 size_t numCounterOffers = 0; 3605 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount, mFormat)}; 3606#if !LOG_NDEBUG 3607 ssize_t index = 3608#else 3609 (void) 3610#endif 3611 mOutputSink->negotiate(offers, 1, NULL, numCounterOffers); 3612 ALOG_ASSERT(index == 0); 3613 3614 // initialize fast mixer depending on configuration 3615 bool initFastMixer; 3616 switch (kUseFastMixer) { 3617 case FastMixer_Never: 3618 initFastMixer = false; 3619 break; 3620 case FastMixer_Always: 3621 initFastMixer = true; 3622 break; 3623 case FastMixer_Static: 3624 case FastMixer_Dynamic: 3625 initFastMixer = mFrameCount < mNormalFrameCount; 3626 break; 3627 } 3628 if (initFastMixer) { 3629 audio_format_t fastMixerFormat; 3630 if (mMixerBufferEnabled && mEffectBufferEnabled) { 3631 fastMixerFormat = AUDIO_FORMAT_PCM_FLOAT; 3632 } else { 3633 fastMixerFormat = AUDIO_FORMAT_PCM_16_BIT; 3634 } 3635 if (mFormat != fastMixerFormat) { 3636 // change our Sink format to accept our intermediate precision 3637 mFormat = fastMixerFormat; 3638 free(mSinkBuffer); 3639 mFrameSize = mChannelCount * audio_bytes_per_sample(mFormat); 3640 const size_t sinkBufferSize = mNormalFrameCount * mFrameSize; 3641 (void)posix_memalign(&mSinkBuffer, 32, sinkBufferSize); 3642 } 3643 3644 // create a MonoPipe to connect our submix to FastMixer 3645 NBAIO_Format format = mOutputSink->format(); 3646#ifdef TEE_SINK 3647 NBAIO_Format origformat = format; 3648#endif 3649 // adjust format to match that of the Fast Mixer 3650 ALOGV("format changed from %d to %d", format.mFormat, fastMixerFormat); 3651 format.mFormat = fastMixerFormat; 3652 format.mFrameSize = audio_bytes_per_sample(format.mFormat) * format.mChannelCount; 3653 3654 // This pipe depth compensates for scheduling latency of the normal mixer thread. 3655 // When it wakes up after a maximum latency, it runs a few cycles quickly before 3656 // finally blocking. Note the pipe implementation rounds up the request to a power of 2. 3657 MonoPipe *monoPipe = new MonoPipe(mNormalFrameCount * 4, format, true /*writeCanBlock*/); 3658 const NBAIO_Format offers[1] = {format}; 3659 size_t numCounterOffers = 0; 3660#if !LOG_NDEBUG || defined(TEE_SINK) 3661 ssize_t index = 3662#else 3663 (void) 3664#endif 3665 monoPipe->negotiate(offers, 1, NULL, numCounterOffers); 3666 ALOG_ASSERT(index == 0); 3667 monoPipe->setAvgFrames((mScreenState & 1) ? 3668 (monoPipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 3669 mPipeSink = monoPipe; 3670 3671#ifdef TEE_SINK 3672 if (mTeeSinkOutputEnabled) { 3673 // create a Pipe to archive a copy of FastMixer's output for dumpsys 3674 Pipe *teeSink = new Pipe(mTeeSinkOutputFrames, origformat); 3675 const NBAIO_Format offers2[1] = {origformat}; 3676 numCounterOffers = 0; 3677 index = teeSink->negotiate(offers2, 1, NULL, numCounterOffers); 3678 ALOG_ASSERT(index == 0); 3679 mTeeSink = teeSink; 3680 PipeReader *teeSource = new PipeReader(*teeSink); 3681 numCounterOffers = 0; 3682 index = teeSource->negotiate(offers2, 1, NULL, numCounterOffers); 3683 ALOG_ASSERT(index == 0); 3684 mTeeSource = teeSource; 3685 } 3686#endif 3687 3688 // create fast mixer and configure it initially with just one fast track for our submix 3689 mFastMixer = new FastMixer(); 3690 FastMixerStateQueue *sq = mFastMixer->sq(); 3691#ifdef STATE_QUEUE_DUMP 3692 sq->setObserverDump(&mStateQueueObserverDump); 3693 sq->setMutatorDump(&mStateQueueMutatorDump); 3694#endif 3695 FastMixerState *state = sq->begin(); 3696 FastTrack *fastTrack = &state->mFastTracks[0]; 3697 // wrap the source side of the MonoPipe to make it an AudioBufferProvider 3698 fastTrack->mBufferProvider = new SourceAudioBufferProvider(new MonoPipeReader(monoPipe)); 3699 fastTrack->mVolumeProvider = NULL; 3700 fastTrack->mChannelMask = mChannelMask; // mPipeSink channel mask for audio to FastMixer 3701 fastTrack->mFormat = mFormat; // mPipeSink format for audio to FastMixer 3702 fastTrack->mGeneration++; 3703 state->mFastTracksGen++; 3704 state->mTrackMask = 1; 3705 // fast mixer will use the HAL output sink 3706 state->mOutputSink = mOutputSink.get(); 3707 state->mOutputSinkGen++; 3708 state->mFrameCount = mFrameCount; 3709 state->mCommand = FastMixerState::COLD_IDLE; 3710 // already done in constructor initialization list 3711 //mFastMixerFutex = 0; 3712 state->mColdFutexAddr = &mFastMixerFutex; 3713 state->mColdGen++; 3714 state->mDumpState = &mFastMixerDumpState; 3715#ifdef TEE_SINK 3716 state->mTeeSink = mTeeSink.get(); 3717#endif 3718 mFastMixerNBLogWriter = audioFlinger->newWriter_l(kFastMixerLogSize, "FastMixer"); 3719 state->mNBLogWriter = mFastMixerNBLogWriter.get(); 3720 sq->end(); 3721 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3722 3723 // start the fast mixer 3724 mFastMixer->run("FastMixer", PRIORITY_URGENT_AUDIO); 3725 pid_t tid = mFastMixer->getTid(); 3726 sendPrioConfigEvent(getpid_cached, tid, kPriorityFastMixer); 3727 3728#ifdef AUDIO_WATCHDOG 3729 // create and start the watchdog 3730 mAudioWatchdog = new AudioWatchdog(); 3731 mAudioWatchdog->setDump(&mAudioWatchdogDump); 3732 mAudioWatchdog->run("AudioWatchdog", PRIORITY_URGENT_AUDIO); 3733 tid = mAudioWatchdog->getTid(); 3734 sendPrioConfigEvent(getpid_cached, tid, kPriorityFastMixer); 3735#endif 3736 3737 } 3738 3739 switch (kUseFastMixer) { 3740 case FastMixer_Never: 3741 case FastMixer_Dynamic: 3742 mNormalSink = mOutputSink; 3743 break; 3744 case FastMixer_Always: 3745 mNormalSink = mPipeSink; 3746 break; 3747 case FastMixer_Static: 3748 mNormalSink = initFastMixer ? mPipeSink : mOutputSink; 3749 break; 3750 } 3751} 3752 3753AudioFlinger::MixerThread::~MixerThread() 3754{ 3755 if (mFastMixer != 0) { 3756 FastMixerStateQueue *sq = mFastMixer->sq(); 3757 FastMixerState *state = sq->begin(); 3758 if (state->mCommand == FastMixerState::COLD_IDLE) { 3759 int32_t old = android_atomic_inc(&mFastMixerFutex); 3760 if (old == -1) { 3761 (void) syscall(__NR_futex, &mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 3762 } 3763 } 3764 state->mCommand = FastMixerState::EXIT; 3765 sq->end(); 3766 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3767 mFastMixer->join(); 3768 // Though the fast mixer thread has exited, it's state queue is still valid. 3769 // We'll use that extract the final state which contains one remaining fast track 3770 // corresponding to our sub-mix. 3771 state = sq->begin(); 3772 ALOG_ASSERT(state->mTrackMask == 1); 3773 FastTrack *fastTrack = &state->mFastTracks[0]; 3774 ALOG_ASSERT(fastTrack->mBufferProvider != NULL); 3775 delete fastTrack->mBufferProvider; 3776 sq->end(false /*didModify*/); 3777 mFastMixer.clear(); 3778#ifdef AUDIO_WATCHDOG 3779 if (mAudioWatchdog != 0) { 3780 mAudioWatchdog->requestExit(); 3781 mAudioWatchdog->requestExitAndWait(); 3782 mAudioWatchdog.clear(); 3783 } 3784#endif 3785 } 3786 mAudioFlinger->unregisterWriter(mFastMixerNBLogWriter); 3787 delete mAudioMixer; 3788} 3789 3790 3791uint32_t AudioFlinger::MixerThread::correctLatency_l(uint32_t latency) const 3792{ 3793 if (mFastMixer != 0) { 3794 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 3795 latency += (pipe->getAvgFrames() * 1000) / mSampleRate; 3796 } 3797 return latency; 3798} 3799 3800 3801void AudioFlinger::MixerThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) 3802{ 3803 PlaybackThread::threadLoop_removeTracks(tracksToRemove); 3804} 3805 3806ssize_t AudioFlinger::MixerThread::threadLoop_write() 3807{ 3808 // FIXME we should only do one push per cycle; confirm this is true 3809 // Start the fast mixer if it's not already running 3810 if (mFastMixer != 0) { 3811 FastMixerStateQueue *sq = mFastMixer->sq(); 3812 FastMixerState *state = sq->begin(); 3813 if (state->mCommand != FastMixerState::MIX_WRITE && 3814 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)) { 3815 if (state->mCommand == FastMixerState::COLD_IDLE) { 3816 3817 // FIXME workaround for first HAL write being CPU bound on some devices 3818 ATRACE_BEGIN("write"); 3819 mOutput->write((char *)mSinkBuffer, 0); 3820 ATRACE_END(); 3821 3822 int32_t old = android_atomic_inc(&mFastMixerFutex); 3823 if (old == -1) { 3824 (void) syscall(__NR_futex, &mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 3825 } 3826#ifdef AUDIO_WATCHDOG 3827 if (mAudioWatchdog != 0) { 3828 mAudioWatchdog->resume(); 3829 } 3830#endif 3831 } 3832 state->mCommand = FastMixerState::MIX_WRITE; 3833#ifdef FAST_THREAD_STATISTICS 3834 mFastMixerDumpState.increaseSamplingN(mAudioFlinger->isLowRamDevice() ? 3835 FastThreadDumpState::kSamplingNforLowRamDevice : FastThreadDumpState::kSamplingN); 3836#endif 3837 sq->end(); 3838 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3839 if (kUseFastMixer == FastMixer_Dynamic) { 3840 mNormalSink = mPipeSink; 3841 } 3842 } else { 3843 sq->end(false /*didModify*/); 3844 } 3845 } 3846 return PlaybackThread::threadLoop_write(); 3847} 3848 3849void AudioFlinger::MixerThread::threadLoop_standby() 3850{ 3851 // Idle the fast mixer if it's currently running 3852 if (mFastMixer != 0) { 3853 FastMixerStateQueue *sq = mFastMixer->sq(); 3854 FastMixerState *state = sq->begin(); 3855 if (!(state->mCommand & FastMixerState::IDLE)) { 3856 state->mCommand = FastMixerState::COLD_IDLE; 3857 state->mColdFutexAddr = &mFastMixerFutex; 3858 state->mColdGen++; 3859 mFastMixerFutex = 0; 3860 sq->end(); 3861 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 3862 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3863 if (kUseFastMixer == FastMixer_Dynamic) { 3864 mNormalSink = mOutputSink; 3865 } 3866#ifdef AUDIO_WATCHDOG 3867 if (mAudioWatchdog != 0) { 3868 mAudioWatchdog->pause(); 3869 } 3870#endif 3871 } else { 3872 sq->end(false /*didModify*/); 3873 } 3874 } 3875 PlaybackThread::threadLoop_standby(); 3876} 3877 3878bool AudioFlinger::PlaybackThread::waitingAsyncCallback_l() 3879{ 3880 return false; 3881} 3882 3883bool AudioFlinger::PlaybackThread::shouldStandby_l() 3884{ 3885 return !mStandby; 3886} 3887 3888bool AudioFlinger::PlaybackThread::waitingAsyncCallback() 3889{ 3890 Mutex::Autolock _l(mLock); 3891 return waitingAsyncCallback_l(); 3892} 3893 3894// shared by MIXER and DIRECT, overridden by DUPLICATING 3895void AudioFlinger::PlaybackThread::threadLoop_standby() 3896{ 3897 ALOGV("Audio hardware entering standby, mixer %p, suspend count %d", this, mSuspended); 3898 mOutput->standby(); 3899 if (mUseAsyncWrite != 0) { 3900 // discard any pending drain or write ack by incrementing sequence 3901 mWriteAckSequence = (mWriteAckSequence + 2) & ~1; 3902 mDrainSequence = (mDrainSequence + 2) & ~1; 3903 ALOG_ASSERT(mCallbackThread != 0); 3904 mCallbackThread->setWriteBlocked(mWriteAckSequence); 3905 mCallbackThread->setDraining(mDrainSequence); 3906 } 3907 mHwPaused = false; 3908} 3909 3910void AudioFlinger::PlaybackThread::onAddNewTrack_l() 3911{ 3912 ALOGV("signal playback thread"); 3913 broadcast_l(); 3914} 3915 3916void AudioFlinger::PlaybackThread::onAsyncError() 3917{ 3918 for (int i = AUDIO_STREAM_SYSTEM; i < (int)AUDIO_STREAM_CNT; i++) { 3919 invalidateTracks((audio_stream_type_t)i); 3920 } 3921} 3922 3923void AudioFlinger::MixerThread::threadLoop_mix() 3924{ 3925 // mix buffers... 3926 mAudioMixer->process(); 3927 mCurrentWriteLength = mSinkBufferSize; 3928 // increase sleep time progressively when application underrun condition clears. 3929 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 3930 // that a steady state of alternating ready/not ready conditions keeps the sleep time 3931 // such that we would underrun the audio HAL. 3932 if ((mSleepTimeUs == 0) && (sleepTimeShift > 0)) { 3933 sleepTimeShift--; 3934 } 3935 mSleepTimeUs = 0; 3936 mStandbyTimeNs = systemTime() + mStandbyDelayNs; 3937 //TODO: delay standby when effects have a tail 3938 3939} 3940 3941void AudioFlinger::MixerThread::threadLoop_sleepTime() 3942{ 3943 // If no tracks are ready, sleep once for the duration of an output 3944 // buffer size, then write 0s to the output 3945 if (mSleepTimeUs == 0) { 3946 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3947 mSleepTimeUs = mActiveSleepTimeUs >> sleepTimeShift; 3948 if (mSleepTimeUs < kMinThreadSleepTimeUs) { 3949 mSleepTimeUs = kMinThreadSleepTimeUs; 3950 } 3951 // reduce sleep time in case of consecutive application underruns to avoid 3952 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 3953 // duration we would end up writing less data than needed by the audio HAL if 3954 // the condition persists. 3955 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 3956 sleepTimeShift++; 3957 } 3958 } else { 3959 mSleepTimeUs = mIdleSleepTimeUs; 3960 } 3961 } else if (mBytesWritten != 0 || (mMixerStatus == MIXER_TRACKS_ENABLED)) { 3962 // clear out mMixerBuffer or mSinkBuffer, to ensure buffers are cleared 3963 // before effects processing or output. 3964 if (mMixerBufferValid) { 3965 memset(mMixerBuffer, 0, mMixerBufferSize); 3966 } else { 3967 memset(mSinkBuffer, 0, mSinkBufferSize); 3968 } 3969 mSleepTimeUs = 0; 3970 ALOGV_IF(mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED), 3971 "anticipated start"); 3972 } 3973 // TODO add standby time extension fct of effect tail 3974} 3975 3976// prepareTracks_l() must be called with ThreadBase::mLock held 3977AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 3978 Vector< sp<Track> > *tracksToRemove) 3979{ 3980 3981 mixer_state mixerStatus = MIXER_IDLE; 3982 // find out which tracks need to be processed 3983 size_t count = mActiveTracks.size(); 3984 size_t mixedTracks = 0; 3985 size_t tracksWithEffect = 0; 3986 // counts only _active_ fast tracks 3987 size_t fastTracks = 0; 3988 uint32_t resetMask = 0; // bit mask of fast tracks that need to be reset 3989 3990 float masterVolume = mMasterVolume; 3991 bool masterMute = mMasterMute; 3992 3993 if (masterMute) { 3994 masterVolume = 0; 3995 } 3996 // Delegate master volume control to effect in output mix effect chain if needed 3997 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 3998 if (chain != 0) { 3999 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 4000 chain->setVolume_l(&v, &v); 4001 masterVolume = (float)((v + (1 << 23)) >> 24); 4002 chain.clear(); 4003 } 4004 4005 // prepare a new state to push 4006 FastMixerStateQueue *sq = NULL; 4007 FastMixerState *state = NULL; 4008 bool didModify = false; 4009 FastMixerStateQueue::block_t block = FastMixerStateQueue::BLOCK_UNTIL_PUSHED; 4010 if (mFastMixer != 0) { 4011 sq = mFastMixer->sq(); 4012 state = sq->begin(); 4013 } 4014 4015 mMixerBufferValid = false; // mMixerBuffer has no valid data until appropriate tracks found. 4016 mEffectBufferValid = false; // mEffectBuffer has no valid data until tracks found. 4017 4018 for (size_t i=0 ; i<count ; i++) { 4019 const sp<Track> t = mActiveTracks[i].promote(); 4020 if (t == 0) { 4021 continue; 4022 } 4023 4024 // this const just means the local variable doesn't change 4025 Track* const track = t.get(); 4026 4027 // process fast tracks 4028 if (track->isFastTrack()) { 4029 4030 // It's theoretically possible (though unlikely) for a fast track to be created 4031 // and then removed within the same normal mix cycle. This is not a problem, as 4032 // the track never becomes active so it's fast mixer slot is never touched. 4033 // The converse, of removing an (active) track and then creating a new track 4034 // at the identical fast mixer slot within the same normal mix cycle, 4035 // is impossible because the slot isn't marked available until the end of each cycle. 4036 int j = track->mFastIndex; 4037 ALOG_ASSERT(0 < j && j < (int)FastMixerState::sMaxFastTracks); 4038 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << j))); 4039 FastTrack *fastTrack = &state->mFastTracks[j]; 4040 4041 // Determine whether the track is currently in underrun condition, 4042 // and whether it had a recent underrun. 4043 FastTrackDump *ftDump = &mFastMixerDumpState.mTracks[j]; 4044 FastTrackUnderruns underruns = ftDump->mUnderruns; 4045 uint32_t recentFull = (underruns.mBitFields.mFull - 4046 track->mObservedUnderruns.mBitFields.mFull) & UNDERRUN_MASK; 4047 uint32_t recentPartial = (underruns.mBitFields.mPartial - 4048 track->mObservedUnderruns.mBitFields.mPartial) & UNDERRUN_MASK; 4049 uint32_t recentEmpty = (underruns.mBitFields.mEmpty - 4050 track->mObservedUnderruns.mBitFields.mEmpty) & UNDERRUN_MASK; 4051 uint32_t recentUnderruns = recentPartial + recentEmpty; 4052 track->mObservedUnderruns = underruns; 4053 // don't count underruns that occur while stopping or pausing 4054 // or stopped which can occur when flush() is called while active 4055 if (!(track->isStopping() || track->isPausing() || track->isStopped()) && 4056 recentUnderruns > 0) { 4057 // FIXME fast mixer will pull & mix partial buffers, but we count as a full underrun 4058 track->mAudioTrackServerProxy->tallyUnderrunFrames(recentUnderruns * mFrameCount); 4059 } else { 4060 track->mAudioTrackServerProxy->tallyUnderrunFrames(0); 4061 } 4062 4063 // This is similar to the state machine for normal tracks, 4064 // with a few modifications for fast tracks. 4065 bool isActive = true; 4066 switch (track->mState) { 4067 case TrackBase::STOPPING_1: 4068 // track stays active in STOPPING_1 state until first underrun 4069 if (recentUnderruns > 0 || track->isTerminated()) { 4070 track->mState = TrackBase::STOPPING_2; 4071 } 4072 break; 4073 case TrackBase::PAUSING: 4074 // ramp down is not yet implemented 4075 track->setPaused(); 4076 break; 4077 case TrackBase::RESUMING: 4078 // ramp up is not yet implemented 4079 track->mState = TrackBase::ACTIVE; 4080 break; 4081 case TrackBase::ACTIVE: 4082 if (recentFull > 0 || recentPartial > 0) { 4083 // track has provided at least some frames recently: reset retry count 4084 track->mRetryCount = kMaxTrackRetries; 4085 } 4086 if (recentUnderruns == 0) { 4087 // no recent underruns: stay active 4088 break; 4089 } 4090 // there has recently been an underrun of some kind 4091 if (track->sharedBuffer() == 0) { 4092 // were any of the recent underruns "empty" (no frames available)? 4093 if (recentEmpty == 0) { 4094 // no, then ignore the partial underruns as they are allowed indefinitely 4095 break; 4096 } 4097 // there has recently been an "empty" underrun: decrement the retry counter 4098 if (--(track->mRetryCount) > 0) { 4099 break; 4100 } 4101 // indicate to client process that the track was disabled because of underrun; 4102 // it will then automatically call start() when data is available 4103 track->disable(); 4104 // remove from active list, but state remains ACTIVE [confusing but true] 4105 isActive = false; 4106 break; 4107 } 4108 // fall through 4109 case TrackBase::STOPPING_2: 4110 case TrackBase::PAUSED: 4111 case TrackBase::STOPPED: 4112 case TrackBase::FLUSHED: // flush() while active 4113 // Check for presentation complete if track is inactive 4114 // We have consumed all the buffers of this track. 4115 // This would be incomplete if we auto-paused on underrun 4116 { 4117 size_t audioHALFrames = 4118 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 4119 int64_t framesWritten = mBytesWritten / mFrameSize; 4120 if (!(mStandby || track->presentationComplete(framesWritten, audioHALFrames))) { 4121 // track stays in active list until presentation is complete 4122 break; 4123 } 4124 } 4125 if (track->isStopping_2()) { 4126 track->mState = TrackBase::STOPPED; 4127 } 4128 if (track->isStopped()) { 4129 // Can't reset directly, as fast mixer is still polling this track 4130 // track->reset(); 4131 // So instead mark this track as needing to be reset after push with ack 4132 resetMask |= 1 << i; 4133 } 4134 isActive = false; 4135 break; 4136 case TrackBase::IDLE: 4137 default: 4138 LOG_ALWAYS_FATAL("unexpected track state %d", track->mState); 4139 } 4140 4141 if (isActive) { 4142 // was it previously inactive? 4143 if (!(state->mTrackMask & (1 << j))) { 4144 ExtendedAudioBufferProvider *eabp = track; 4145 VolumeProvider *vp = track; 4146 fastTrack->mBufferProvider = eabp; 4147 fastTrack->mVolumeProvider = vp; 4148 fastTrack->mChannelMask = track->mChannelMask; 4149 fastTrack->mFormat = track->mFormat; 4150 fastTrack->mGeneration++; 4151 state->mTrackMask |= 1 << j; 4152 didModify = true; 4153 // no acknowledgement required for newly active tracks 4154 } 4155 // cache the combined master volume and stream type volume for fast mixer; this 4156 // lacks any synchronization or barrier so VolumeProvider may read a stale value 4157 track->mCachedVolume = masterVolume * mStreamTypes[track->streamType()].volume; 4158 ++fastTracks; 4159 } else { 4160 // was it previously active? 4161 if (state->mTrackMask & (1 << j)) { 4162 fastTrack->mBufferProvider = NULL; 4163 fastTrack->mGeneration++; 4164 state->mTrackMask &= ~(1 << j); 4165 didModify = true; 4166 // If any fast tracks were removed, we must wait for acknowledgement 4167 // because we're about to decrement the last sp<> on those tracks. 4168 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 4169 } else { 4170 LOG_ALWAYS_FATAL("fast track %d should have been active; " 4171 "mState=%d, mTrackMask=%#x, recentUnderruns=%u, isShared=%d", 4172 j, track->mState, state->mTrackMask, recentUnderruns, 4173 track->sharedBuffer() != 0); 4174 } 4175 tracksToRemove->add(track); 4176 // Avoids a misleading display in dumpsys 4177 track->mObservedUnderruns.mBitFields.mMostRecent = UNDERRUN_FULL; 4178 } 4179 continue; 4180 } 4181 4182 { // local variable scope to avoid goto warning 4183 4184 audio_track_cblk_t* cblk = track->cblk(); 4185 4186 // The first time a track is added we wait 4187 // for all its buffers to be filled before processing it 4188 int name = track->name(); 4189 // make sure that we have enough frames to mix one full buffer. 4190 // enforce this condition only once to enable draining the buffer in case the client 4191 // app does not call stop() and relies on underrun to stop: 4192 // hence the test on (mMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 4193 // during last round 4194 size_t desiredFrames; 4195 const uint32_t sampleRate = track->mAudioTrackServerProxy->getSampleRate(); 4196 AudioPlaybackRate playbackRate = track->mAudioTrackServerProxy->getPlaybackRate(); 4197 4198 desiredFrames = sourceFramesNeededWithTimestretch( 4199 sampleRate, mNormalFrameCount, mSampleRate, playbackRate.mSpeed); 4200 // TODO: ONLY USED FOR LEGACY RESAMPLERS, remove when they are removed. 4201 // add frames already consumed but not yet released by the resampler 4202 // because mAudioTrackServerProxy->framesReady() will include these frames 4203 desiredFrames += mAudioMixer->getUnreleasedFrames(track->name()); 4204 4205 uint32_t minFrames = 1; 4206 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() && 4207 (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) { 4208 minFrames = desiredFrames; 4209 } 4210 4211 size_t framesReady = track->framesReady(); 4212 if (ATRACE_ENABLED()) { 4213 // I wish we had formatted trace names 4214 char traceName[16]; 4215 strcpy(traceName, "nRdy"); 4216 int name = track->name(); 4217 if (AudioMixer::TRACK0 <= name && 4218 name < (int) (AudioMixer::TRACK0 + AudioMixer::MAX_NUM_TRACKS)) { 4219 name -= AudioMixer::TRACK0; 4220 traceName[4] = (name / 10) + '0'; 4221 traceName[5] = (name % 10) + '0'; 4222 } else { 4223 traceName[4] = '?'; 4224 traceName[5] = '?'; 4225 } 4226 traceName[6] = '\0'; 4227 ATRACE_INT(traceName, framesReady); 4228 } 4229 if ((framesReady >= minFrames) && track->isReady() && 4230 !track->isPaused() && !track->isTerminated()) 4231 { 4232 ALOGVV("track %d s=%08x [OK] on thread %p", name, cblk->mServer, this); 4233 4234 mixedTracks++; 4235 4236 // track->mainBuffer() != mSinkBuffer or mMixerBuffer means 4237 // there is an effect chain connected to the track 4238 chain.clear(); 4239 if (track->mainBuffer() != mSinkBuffer && 4240 track->mainBuffer() != mMixerBuffer) { 4241 if (mEffectBufferEnabled) { 4242 mEffectBufferValid = true; // Later can set directly. 4243 } 4244 chain = getEffectChain_l(track->sessionId()); 4245 // Delegate volume control to effect in track effect chain if needed 4246 if (chain != 0) { 4247 tracksWithEffect++; 4248 } else { 4249 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on " 4250 "session %d", 4251 name, track->sessionId()); 4252 } 4253 } 4254 4255 4256 int param = AudioMixer::VOLUME; 4257 if (track->mFillingUpStatus == Track::FS_FILLED) { 4258 // no ramp for the first volume setting 4259 track->mFillingUpStatus = Track::FS_ACTIVE; 4260 if (track->mState == TrackBase::RESUMING) { 4261 track->mState = TrackBase::ACTIVE; 4262 param = AudioMixer::RAMP_VOLUME; 4263 } 4264 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 4265 // FIXME should not make a decision based on mServer 4266 } else if (cblk->mServer != 0) { 4267 // If the track is stopped before the first frame was mixed, 4268 // do not apply ramp 4269 param = AudioMixer::RAMP_VOLUME; 4270 } 4271 4272 // compute volume for this track 4273 uint32_t vl, vr; // in U8.24 integer format 4274 float vlf, vrf, vaf; // in [0.0, 1.0] float format 4275 if (track->isPausing() || mStreamTypes[track->streamType()].mute) { 4276 vl = vr = 0; 4277 vlf = vrf = vaf = 0.; 4278 if (track->isPausing()) { 4279 track->setPaused(); 4280 } 4281 } else { 4282 4283 // read original volumes with volume control 4284 float typeVolume = mStreamTypes[track->streamType()].volume; 4285 float v = masterVolume * typeVolume; 4286 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 4287 gain_minifloat_packed_t vlr = proxy->getVolumeLR(); 4288 vlf = float_from_gain(gain_minifloat_unpack_left(vlr)); 4289 vrf = float_from_gain(gain_minifloat_unpack_right(vlr)); 4290 // track volumes come from shared memory, so can't be trusted and must be clamped 4291 if (vlf > GAIN_FLOAT_UNITY) { 4292 ALOGV("Track left volume out of range: %.3g", vlf); 4293 vlf = GAIN_FLOAT_UNITY; 4294 } 4295 if (vrf > GAIN_FLOAT_UNITY) { 4296 ALOGV("Track right volume out of range: %.3g", vrf); 4297 vrf = GAIN_FLOAT_UNITY; 4298 } 4299 // now apply the master volume and stream type volume 4300 vlf *= v; 4301 vrf *= v; 4302 // assuming master volume and stream type volume each go up to 1.0, 4303 // then derive vl and vr as U8.24 versions for the effect chain 4304 const float scaleto8_24 = MAX_GAIN_INT * MAX_GAIN_INT; 4305 vl = (uint32_t) (scaleto8_24 * vlf); 4306 vr = (uint32_t) (scaleto8_24 * vrf); 4307 // vl and vr are now in U8.24 format 4308 uint16_t sendLevel = proxy->getSendLevel_U4_12(); 4309 // send level comes from shared memory and so may be corrupt 4310 if (sendLevel > MAX_GAIN_INT) { 4311 ALOGV("Track send level out of range: %04X", sendLevel); 4312 sendLevel = MAX_GAIN_INT; 4313 } 4314 // vaf is represented as [0.0, 1.0] float by rescaling sendLevel 4315 vaf = v * sendLevel * (1. / MAX_GAIN_INT); 4316 } 4317 4318 // Delegate volume control to effect in track effect chain if needed 4319 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 4320 // Do not ramp volume if volume is controlled by effect 4321 param = AudioMixer::VOLUME; 4322 // Update remaining floating point volume levels 4323 vlf = (float)vl / (1 << 24); 4324 vrf = (float)vr / (1 << 24); 4325 track->mHasVolumeController = true; 4326 } else { 4327 // force no volume ramp when volume controller was just disabled or removed 4328 // from effect chain to avoid volume spike 4329 if (track->mHasVolumeController) { 4330 param = AudioMixer::VOLUME; 4331 } 4332 track->mHasVolumeController = false; 4333 } 4334 4335 // XXX: these things DON'T need to be done each time 4336 mAudioMixer->setBufferProvider(name, track); 4337 mAudioMixer->enable(name); 4338 4339 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, &vlf); 4340 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, &vrf); 4341 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, &vaf); 4342 mAudioMixer->setParameter( 4343 name, 4344 AudioMixer::TRACK, 4345 AudioMixer::FORMAT, (void *)track->format()); 4346 mAudioMixer->setParameter( 4347 name, 4348 AudioMixer::TRACK, 4349 AudioMixer::CHANNEL_MASK, (void *)(uintptr_t)track->channelMask()); 4350 mAudioMixer->setParameter( 4351 name, 4352 AudioMixer::TRACK, 4353 AudioMixer::MIXER_CHANNEL_MASK, (void *)(uintptr_t)mChannelMask); 4354 // limit track sample rate to 2 x output sample rate, which changes at re-configuration 4355 uint32_t maxSampleRate = mSampleRate * AUDIO_RESAMPLER_DOWN_RATIO_MAX; 4356 uint32_t reqSampleRate = track->mAudioTrackServerProxy->getSampleRate(); 4357 if (reqSampleRate == 0) { 4358 reqSampleRate = mSampleRate; 4359 } else if (reqSampleRate > maxSampleRate) { 4360 reqSampleRate = maxSampleRate; 4361 } 4362 mAudioMixer->setParameter( 4363 name, 4364 AudioMixer::RESAMPLE, 4365 AudioMixer::SAMPLE_RATE, 4366 (void *)(uintptr_t)reqSampleRate); 4367 4368 AudioPlaybackRate playbackRate = track->mAudioTrackServerProxy->getPlaybackRate(); 4369 mAudioMixer->setParameter( 4370 name, 4371 AudioMixer::TIMESTRETCH, 4372 AudioMixer::PLAYBACK_RATE, 4373 &playbackRate); 4374 4375 /* 4376 * Select the appropriate output buffer for the track. 4377 * 4378 * Tracks with effects go into their own effects chain buffer 4379 * and from there into either mEffectBuffer or mSinkBuffer. 4380 * 4381 * Other tracks can use mMixerBuffer for higher precision 4382 * channel accumulation. If this buffer is enabled 4383 * (mMixerBufferEnabled true), then selected tracks will accumulate 4384 * into it. 4385 * 4386 */ 4387 if (mMixerBufferEnabled 4388 && (track->mainBuffer() == mSinkBuffer 4389 || track->mainBuffer() == mMixerBuffer)) { 4390 mAudioMixer->setParameter( 4391 name, 4392 AudioMixer::TRACK, 4393 AudioMixer::MIXER_FORMAT, (void *)mMixerBufferFormat); 4394 mAudioMixer->setParameter( 4395 name, 4396 AudioMixer::TRACK, 4397 AudioMixer::MAIN_BUFFER, (void *)mMixerBuffer); 4398 // TODO: override track->mainBuffer()? 4399 mMixerBufferValid = true; 4400 } else { 4401 mAudioMixer->setParameter( 4402 name, 4403 AudioMixer::TRACK, 4404 AudioMixer::MIXER_FORMAT, (void *)AUDIO_FORMAT_PCM_16_BIT); 4405 mAudioMixer->setParameter( 4406 name, 4407 AudioMixer::TRACK, 4408 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 4409 } 4410 mAudioMixer->setParameter( 4411 name, 4412 AudioMixer::TRACK, 4413 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 4414 4415 // reset retry count 4416 track->mRetryCount = kMaxTrackRetries; 4417 4418 // If one track is ready, set the mixer ready if: 4419 // - the mixer was not ready during previous round OR 4420 // - no other track is not ready 4421 if (mMixerStatusIgnoringFastTracks != MIXER_TRACKS_READY || 4422 mixerStatus != MIXER_TRACKS_ENABLED) { 4423 mixerStatus = MIXER_TRACKS_READY; 4424 } 4425 } else { 4426 if (framesReady < desiredFrames && !track->isStopped() && !track->isPaused()) { 4427 ALOGV("track(%p) underrun, framesReady(%zu) < framesDesired(%zd)", 4428 track, framesReady, desiredFrames); 4429 track->mAudioTrackServerProxy->tallyUnderrunFrames(desiredFrames); 4430 } else { 4431 track->mAudioTrackServerProxy->tallyUnderrunFrames(0); 4432 } 4433 4434 // clear effect chain input buffer if an active track underruns to avoid sending 4435 // previous audio buffer again to effects 4436 chain = getEffectChain_l(track->sessionId()); 4437 if (chain != 0) { 4438 chain->clearInputBuffer(); 4439 } 4440 4441 ALOGVV("track %d s=%08x [NOT READY] on thread %p", name, cblk->mServer, this); 4442 if ((track->sharedBuffer() != 0) || track->isTerminated() || 4443 track->isStopped() || track->isPaused()) { 4444 // We have consumed all the buffers of this track. 4445 // Remove it from the list of active tracks. 4446 // TODO: use actual buffer filling status instead of latency when available from 4447 // audio HAL 4448 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 4449 int64_t framesWritten = mBytesWritten / mFrameSize; 4450 if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) { 4451 if (track->isStopped()) { 4452 track->reset(); 4453 } 4454 tracksToRemove->add(track); 4455 } 4456 } else { 4457 // No buffers for this track. Give it a few chances to 4458 // fill a buffer, then remove it from active list. 4459 if (--(track->mRetryCount) <= 0) { 4460 ALOGI("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 4461 tracksToRemove->add(track); 4462 // indicate to client process that the track was disabled because of underrun; 4463 // it will then automatically call start() when data is available 4464 track->disable(); 4465 // If one track is not ready, mark the mixer also not ready if: 4466 // - the mixer was ready during previous round OR 4467 // - no other track is ready 4468 } else if (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY || 4469 mixerStatus != MIXER_TRACKS_READY) { 4470 mixerStatus = MIXER_TRACKS_ENABLED; 4471 } 4472 } 4473 mAudioMixer->disable(name); 4474 } 4475 4476 } // local variable scope to avoid goto warning 4477 4478 } 4479 4480 // Push the new FastMixer state if necessary 4481 bool pauseAudioWatchdog = false; 4482 if (didModify) { 4483 state->mFastTracksGen++; 4484 // if the fast mixer was active, but now there are no fast tracks, then put it in cold idle 4485 if (kUseFastMixer == FastMixer_Dynamic && 4486 state->mCommand == FastMixerState::MIX_WRITE && state->mTrackMask <= 1) { 4487 state->mCommand = FastMixerState::COLD_IDLE; 4488 state->mColdFutexAddr = &mFastMixerFutex; 4489 state->mColdGen++; 4490 mFastMixerFutex = 0; 4491 if (kUseFastMixer == FastMixer_Dynamic) { 4492 mNormalSink = mOutputSink; 4493 } 4494 // If we go into cold idle, need to wait for acknowledgement 4495 // so that fast mixer stops doing I/O. 4496 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 4497 pauseAudioWatchdog = true; 4498 } 4499 } 4500 if (sq != NULL) { 4501 sq->end(didModify); 4502 sq->push(block); 4503 } 4504#ifdef AUDIO_WATCHDOG 4505 if (pauseAudioWatchdog && mAudioWatchdog != 0) { 4506 mAudioWatchdog->pause(); 4507 } 4508#endif 4509 4510 // Now perform the deferred reset on fast tracks that have stopped 4511 while (resetMask != 0) { 4512 size_t i = __builtin_ctz(resetMask); 4513 ALOG_ASSERT(i < count); 4514 resetMask &= ~(1 << i); 4515 sp<Track> t = mActiveTracks[i].promote(); 4516 if (t == 0) { 4517 continue; 4518 } 4519 Track* track = t.get(); 4520 ALOG_ASSERT(track->isFastTrack() && track->isStopped()); 4521 track->reset(); 4522 } 4523 4524 // remove all the tracks that need to be... 4525 removeTracks_l(*tracksToRemove); 4526 4527 if (getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX) != 0) { 4528 mEffectBufferValid = true; 4529 } 4530 4531 if (mEffectBufferValid) { 4532 // as long as there are effects we should clear the effects buffer, to avoid 4533 // passing a non-clean buffer to the effect chain 4534 memset(mEffectBuffer, 0, mEffectBufferSize); 4535 } 4536 // sink or mix buffer must be cleared if all tracks are connected to an 4537 // effect chain as in this case the mixer will not write to the sink or mix buffer 4538 // and track effects will accumulate into it 4539 if ((mBytesRemaining == 0) && ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 4540 (mixedTracks == 0 && fastTracks > 0))) { 4541 // FIXME as a performance optimization, should remember previous zero status 4542 if (mMixerBufferValid) { 4543 memset(mMixerBuffer, 0, mMixerBufferSize); 4544 // TODO: In testing, mSinkBuffer below need not be cleared because 4545 // the PlaybackThread::threadLoop() copies mMixerBuffer into mSinkBuffer 4546 // after mixing. 4547 // 4548 // To enforce this guarantee: 4549 // ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 4550 // (mixedTracks == 0 && fastTracks > 0)) 4551 // must imply MIXER_TRACKS_READY. 4552 // Later, we may clear buffers regardless, and skip much of this logic. 4553 } 4554 // FIXME as a performance optimization, should remember previous zero status 4555 memset(mSinkBuffer, 0, mNormalFrameCount * mFrameSize); 4556 } 4557 4558 // if any fast tracks, then status is ready 4559 mMixerStatusIgnoringFastTracks = mixerStatus; 4560 if (fastTracks > 0) { 4561 mixerStatus = MIXER_TRACKS_READY; 4562 } 4563 return mixerStatus; 4564} 4565 4566// getTrackName_l() must be called with ThreadBase::mLock held 4567int AudioFlinger::MixerThread::getTrackName_l(audio_channel_mask_t channelMask, 4568 audio_format_t format, audio_session_t sessionId) 4569{ 4570 return mAudioMixer->getTrackName(channelMask, format, sessionId); 4571} 4572 4573// deleteTrackName_l() must be called with ThreadBase::mLock held 4574void AudioFlinger::MixerThread::deleteTrackName_l(int name) 4575{ 4576 ALOGV("remove track (%d) and delete from mixer", name); 4577 mAudioMixer->deleteTrackName(name); 4578} 4579 4580// checkForNewParameter_l() must be called with ThreadBase::mLock held 4581bool AudioFlinger::MixerThread::checkForNewParameter_l(const String8& keyValuePair, 4582 status_t& status) 4583{ 4584 bool reconfig = false; 4585 bool a2dpDeviceChanged = false; 4586 4587 status = NO_ERROR; 4588 4589 AutoPark<FastMixer> park(mFastMixer); 4590 4591 AudioParameter param = AudioParameter(keyValuePair); 4592 int value; 4593 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 4594 reconfig = true; 4595 } 4596 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 4597 if (!isValidPcmSinkFormat((audio_format_t) value)) { 4598 status = BAD_VALUE; 4599 } else { 4600 // no need to save value, since it's constant 4601 reconfig = true; 4602 } 4603 } 4604 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 4605 if (!isValidPcmSinkChannelMask((audio_channel_mask_t) value)) { 4606 status = BAD_VALUE; 4607 } else { 4608 // no need to save value, since it's constant 4609 reconfig = true; 4610 } 4611 } 4612 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 4613 // do not accept frame count changes if tracks are open as the track buffer 4614 // size depends on frame count and correct behavior would not be guaranteed 4615 // if frame count is changed after track creation 4616 if (!mTracks.isEmpty()) { 4617 status = INVALID_OPERATION; 4618 } else { 4619 reconfig = true; 4620 } 4621 } 4622 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 4623#ifdef ADD_BATTERY_DATA 4624 // when changing the audio output device, call addBatteryData to notify 4625 // the change 4626 if (mOutDevice != value) { 4627 uint32_t params = 0; 4628 // check whether speaker is on 4629 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 4630 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 4631 } 4632 4633 audio_devices_t deviceWithoutSpeaker 4634 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 4635 // check if any other device (except speaker) is on 4636 if (value & deviceWithoutSpeaker) { 4637 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 4638 } 4639 4640 if (params != 0) { 4641 addBatteryData(params); 4642 } 4643 } 4644#endif 4645 4646 // forward device change to effects that have requested to be 4647 // aware of attached audio device. 4648 if (value != AUDIO_DEVICE_NONE) { 4649 a2dpDeviceChanged = 4650 (mOutDevice & AUDIO_DEVICE_OUT_ALL_A2DP) != (value & AUDIO_DEVICE_OUT_ALL_A2DP); 4651 mOutDevice = value; 4652 for (size_t i = 0; i < mEffectChains.size(); i++) { 4653 mEffectChains[i]->setDevice_l(mOutDevice); 4654 } 4655 } 4656 } 4657 4658 if (status == NO_ERROR) { 4659 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 4660 keyValuePair.string()); 4661 if (!mStandby && status == INVALID_OPERATION) { 4662 mOutput->standby(); 4663 mStandby = true; 4664 mBytesWritten = 0; 4665 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 4666 keyValuePair.string()); 4667 } 4668 if (status == NO_ERROR && reconfig) { 4669 readOutputParameters_l(); 4670 delete mAudioMixer; 4671 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 4672 for (size_t i = 0; i < mTracks.size() ; i++) { 4673 int name = getTrackName_l(mTracks[i]->mChannelMask, 4674 mTracks[i]->mFormat, mTracks[i]->mSessionId); 4675 if (name < 0) { 4676 break; 4677 } 4678 mTracks[i]->mName = name; 4679 } 4680 sendIoConfigEvent_l(AUDIO_OUTPUT_CONFIG_CHANGED); 4681 } 4682 } 4683 4684 return reconfig || a2dpDeviceChanged; 4685} 4686 4687 4688void AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 4689{ 4690 PlaybackThread::dumpInternals(fd, args); 4691 dprintf(fd, " Thread throttle time (msecs): %u\n", mThreadThrottleTimeMs); 4692 dprintf(fd, " AudioMixer tracks: 0x%08x\n", mAudioMixer->trackNames()); 4693 dprintf(fd, " Master mono: %s\n", mMasterMono ? "on" : "off"); 4694 4695 // Make a non-atomic copy of fast mixer dump state so it won't change underneath us 4696 // while we are dumping it. It may be inconsistent, but it won't mutate! 4697 // This is a large object so we place it on the heap. 4698 // FIXME 25972958: Need an intelligent copy constructor that does not touch unused pages. 4699 const FastMixerDumpState *copy = new FastMixerDumpState(mFastMixerDumpState); 4700 copy->dump(fd); 4701 delete copy; 4702 4703#ifdef STATE_QUEUE_DUMP 4704 // Similar for state queue 4705 StateQueueObserverDump observerCopy = mStateQueueObserverDump; 4706 observerCopy.dump(fd); 4707 StateQueueMutatorDump mutatorCopy = mStateQueueMutatorDump; 4708 mutatorCopy.dump(fd); 4709#endif 4710 4711#ifdef TEE_SINK 4712 // Write the tee output to a .wav file 4713 dumpTee(fd, mTeeSource, mId); 4714#endif 4715 4716#ifdef AUDIO_WATCHDOG 4717 if (mAudioWatchdog != 0) { 4718 // Make a non-atomic copy of audio watchdog dump so it won't change underneath us 4719 AudioWatchdogDump wdCopy = mAudioWatchdogDump; 4720 wdCopy.dump(fd); 4721 } 4722#endif 4723} 4724 4725uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const 4726{ 4727 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000) / 2; 4728} 4729 4730uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() const 4731{ 4732 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000); 4733} 4734 4735void AudioFlinger::MixerThread::cacheParameters_l() 4736{ 4737 PlaybackThread::cacheParameters_l(); 4738 4739 // FIXME: Relaxed timing because of a certain device that can't meet latency 4740 // Should be reduced to 2x after the vendor fixes the driver issue 4741 // increase threshold again due to low power audio mode. The way this warning 4742 // threshold is calculated and its usefulness should be reconsidered anyway. 4743 maxPeriod = seconds(mNormalFrameCount) / mSampleRate * 15; 4744} 4745 4746// ---------------------------------------------------------------------------- 4747 4748AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 4749 AudioStreamOut* output, audio_io_handle_t id, audio_devices_t device, bool systemReady) 4750 : PlaybackThread(audioFlinger, output, id, device, DIRECT, systemReady) 4751 // mLeftVolFloat, mRightVolFloat 4752{ 4753} 4754 4755AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 4756 AudioStreamOut* output, audio_io_handle_t id, uint32_t device, 4757 ThreadBase::type_t type, bool systemReady) 4758 : PlaybackThread(audioFlinger, output, id, device, type, systemReady) 4759 // mLeftVolFloat, mRightVolFloat 4760{ 4761} 4762 4763AudioFlinger::DirectOutputThread::~DirectOutputThread() 4764{ 4765} 4766 4767void AudioFlinger::DirectOutputThread::processVolume_l(Track *track, bool lastTrack) 4768{ 4769 float left, right; 4770 4771 if (mMasterMute || mStreamTypes[track->streamType()].mute) { 4772 left = right = 0; 4773 } else { 4774 float typeVolume = mStreamTypes[track->streamType()].volume; 4775 float v = mMasterVolume * typeVolume; 4776 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 4777 gain_minifloat_packed_t vlr = proxy->getVolumeLR(); 4778 left = float_from_gain(gain_minifloat_unpack_left(vlr)); 4779 if (left > GAIN_FLOAT_UNITY) { 4780 left = GAIN_FLOAT_UNITY; 4781 } 4782 left *= v; 4783 right = float_from_gain(gain_minifloat_unpack_right(vlr)); 4784 if (right > GAIN_FLOAT_UNITY) { 4785 right = GAIN_FLOAT_UNITY; 4786 } 4787 right *= v; 4788 } 4789 4790 if (lastTrack) { 4791 if (left != mLeftVolFloat || right != mRightVolFloat) { 4792 mLeftVolFloat = left; 4793 mRightVolFloat = right; 4794 4795 // Convert volumes from float to 8.24 4796 uint32_t vl = (uint32_t)(left * (1 << 24)); 4797 uint32_t vr = (uint32_t)(right * (1 << 24)); 4798 4799 // Delegate volume control to effect in track effect chain if needed 4800 // only one effect chain can be present on DirectOutputThread, so if 4801 // there is one, the track is connected to it 4802 if (!mEffectChains.isEmpty()) { 4803 mEffectChains[0]->setVolume_l(&vl, &vr); 4804 left = (float)vl / (1 << 24); 4805 right = (float)vr / (1 << 24); 4806 } 4807 if (mOutput->stream->set_volume) { 4808 mOutput->stream->set_volume(mOutput->stream, left, right); 4809 } 4810 } 4811 } 4812} 4813 4814void AudioFlinger::DirectOutputThread::onAddNewTrack_l() 4815{ 4816 sp<Track> previousTrack = mPreviousTrack.promote(); 4817 sp<Track> latestTrack = mLatestActiveTrack.promote(); 4818 4819 if (previousTrack != 0 && latestTrack != 0) { 4820 if (mType == DIRECT) { 4821 if (previousTrack.get() != latestTrack.get()) { 4822 mFlushPending = true; 4823 } 4824 } else /* mType == OFFLOAD */ { 4825 if (previousTrack->sessionId() != latestTrack->sessionId()) { 4826 mFlushPending = true; 4827 } 4828 } 4829 } 4830 PlaybackThread::onAddNewTrack_l(); 4831} 4832 4833AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l( 4834 Vector< sp<Track> > *tracksToRemove 4835) 4836{ 4837 size_t count = mActiveTracks.size(); 4838 mixer_state mixerStatus = MIXER_IDLE; 4839 bool doHwPause = false; 4840 bool doHwResume = false; 4841 4842 // find out which tracks need to be processed 4843 for (size_t i = 0; i < count; i++) { 4844 sp<Track> t = mActiveTracks[i].promote(); 4845 // The track died recently 4846 if (t == 0) { 4847 continue; 4848 } 4849 4850 if (t->isInvalid()) { 4851 ALOGW("An invalidated track shouldn't be in active list"); 4852 tracksToRemove->add(t); 4853 continue; 4854 } 4855 4856 Track* const track = t.get(); 4857#ifdef VERY_VERY_VERBOSE_LOGGING 4858 audio_track_cblk_t* cblk = track->cblk(); 4859#endif 4860 // Only consider last track started for volume and mixer state control. 4861 // In theory an older track could underrun and restart after the new one starts 4862 // but as we only care about the transition phase between two tracks on a 4863 // direct output, it is not a problem to ignore the underrun case. 4864 sp<Track> l = mLatestActiveTrack.promote(); 4865 bool last = l.get() == track; 4866 4867 if (track->isPausing()) { 4868 track->setPaused(); 4869 if (mHwSupportsPause && last && !mHwPaused) { 4870 doHwPause = true; 4871 mHwPaused = true; 4872 } 4873 tracksToRemove->add(track); 4874 } else if (track->isFlushPending()) { 4875 track->flushAck(); 4876 if (last) { 4877 mFlushPending = true; 4878 } 4879 } else if (track->isResumePending()) { 4880 track->resumeAck(); 4881 if (last) { 4882 mLeftVolFloat = mRightVolFloat = -1.0; 4883 if (mHwPaused) { 4884 doHwResume = true; 4885 mHwPaused = false; 4886 } 4887 } 4888 } 4889 4890 // The first time a track is added we wait 4891 // for all its buffers to be filled before processing it. 4892 // Allow draining the buffer in case the client 4893 // app does not call stop() and relies on underrun to stop: 4894 // hence the test on (track->mRetryCount > 1). 4895 // If retryCount<=1 then track is about to underrun and be removed. 4896 // Do not use a high threshold for compressed audio. 4897 uint32_t minFrames; 4898 if ((track->sharedBuffer() == 0) && !track->isStopping_1() && !track->isPausing() 4899 && (track->mRetryCount > 1) && audio_has_proportional_frames(mFormat)) { 4900 minFrames = mNormalFrameCount; 4901 } else { 4902 minFrames = 1; 4903 } 4904 4905 if ((track->framesReady() >= minFrames) && track->isReady() && !track->isPaused() && 4906 !track->isStopping_2() && !track->isStopped()) 4907 { 4908 ALOGVV("track %d s=%08x [OK]", track->name(), cblk->mServer); 4909 4910 if (track->mFillingUpStatus == Track::FS_FILLED) { 4911 track->mFillingUpStatus = Track::FS_ACTIVE; 4912 if (last) { 4913 // make sure processVolume_l() will apply new volume even if 0 4914 mLeftVolFloat = mRightVolFloat = -1.0; 4915 } 4916 if (!mHwSupportsPause) { 4917 track->resumeAck(); 4918 } 4919 } 4920 4921 // compute volume for this track 4922 processVolume_l(track, last); 4923 if (last) { 4924 sp<Track> previousTrack = mPreviousTrack.promote(); 4925 if (previousTrack != 0) { 4926 if (track != previousTrack.get()) { 4927 // Flush any data still being written from last track 4928 mBytesRemaining = 0; 4929 // Invalidate previous track to force a seek when resuming. 4930 previousTrack->invalidate(); 4931 } 4932 } 4933 mPreviousTrack = track; 4934 4935 // reset retry count 4936 track->mRetryCount = kMaxTrackRetriesDirect; 4937 mActiveTrack = t; 4938 mixerStatus = MIXER_TRACKS_READY; 4939 if (mHwPaused) { 4940 doHwResume = true; 4941 mHwPaused = false; 4942 } 4943 } 4944 } else { 4945 // clear effect chain input buffer if the last active track started underruns 4946 // to avoid sending previous audio buffer again to effects 4947 if (!mEffectChains.isEmpty() && last) { 4948 mEffectChains[0]->clearInputBuffer(); 4949 } 4950 if (track->isStopping_1()) { 4951 track->mState = TrackBase::STOPPING_2; 4952 if (last && mHwPaused) { 4953 doHwResume = true; 4954 mHwPaused = false; 4955 } 4956 } 4957 if ((track->sharedBuffer() != 0) || track->isStopped() || 4958 track->isStopping_2() || track->isPaused()) { 4959 // We have consumed all the buffers of this track. 4960 // Remove it from the list of active tracks. 4961 size_t audioHALFrames; 4962 if (audio_has_proportional_frames(mFormat)) { 4963 audioHALFrames = (latency_l() * mSampleRate) / 1000; 4964 } else { 4965 audioHALFrames = 0; 4966 } 4967 4968 int64_t framesWritten = mBytesWritten / mFrameSize; 4969 if (mStandby || !last || 4970 track->presentationComplete(framesWritten, audioHALFrames)) { 4971 if (track->isStopping_2()) { 4972 track->mState = TrackBase::STOPPED; 4973 } 4974 if (track->isStopped()) { 4975 track->reset(); 4976 } 4977 tracksToRemove->add(track); 4978 } 4979 } else { 4980 // No buffers for this track. Give it a few chances to 4981 // fill a buffer, then remove it from active list. 4982 // Only consider last track started for mixer state control 4983 if (--(track->mRetryCount) <= 0) { 4984 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 4985 tracksToRemove->add(track); 4986 // indicate to client process that the track was disabled because of underrun; 4987 // it will then automatically call start() when data is available 4988 track->disable(); 4989 } else if (last) { 4990 ALOGW("pause because of UNDERRUN, framesReady = %zu," 4991 "minFrames = %u, mFormat = %#x", 4992 track->framesReady(), minFrames, mFormat); 4993 mixerStatus = MIXER_TRACKS_ENABLED; 4994 if (mHwSupportsPause && !mHwPaused && !mStandby) { 4995 doHwPause = true; 4996 mHwPaused = true; 4997 } 4998 } 4999 } 5000 } 5001 } 5002 5003 // if an active track did not command a flush, check for pending flush on stopped tracks 5004 if (!mFlushPending) { 5005 for (size_t i = 0; i < mTracks.size(); i++) { 5006 if (mTracks[i]->isFlushPending()) { 5007 mTracks[i]->flushAck(); 5008 mFlushPending = true; 5009 } 5010 } 5011 } 5012 5013 // make sure the pause/flush/resume sequence is executed in the right order. 5014 // If a flush is pending and a track is active but the HW is not paused, force a HW pause 5015 // before flush and then resume HW. This can happen in case of pause/flush/resume 5016 // if resume is received before pause is executed. 5017 if (mHwSupportsPause && !mStandby && 5018 (doHwPause || (mFlushPending && !mHwPaused && (count != 0)))) { 5019 mOutput->stream->pause(mOutput->stream); 5020 } 5021 if (mFlushPending) { 5022 flushHw_l(); 5023 } 5024 if (mHwSupportsPause && !mStandby && doHwResume) { 5025 mOutput->stream->resume(mOutput->stream); 5026 } 5027 // remove all the tracks that need to be... 5028 removeTracks_l(*tracksToRemove); 5029 5030 return mixerStatus; 5031} 5032 5033void AudioFlinger::DirectOutputThread::threadLoop_mix() 5034{ 5035 size_t frameCount = mFrameCount; 5036 int8_t *curBuf = (int8_t *)mSinkBuffer; 5037 // output audio to hardware 5038 while (frameCount) { 5039 AudioBufferProvider::Buffer buffer; 5040 buffer.frameCount = frameCount; 5041 status_t status = mActiveTrack->getNextBuffer(&buffer); 5042 if (status != NO_ERROR || buffer.raw == NULL) { 5043 // no need to pad with 0 for compressed audio 5044 if (audio_has_proportional_frames(mFormat)) { 5045 memset(curBuf, 0, frameCount * mFrameSize); 5046 } 5047 break; 5048 } 5049 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 5050 frameCount -= buffer.frameCount; 5051 curBuf += buffer.frameCount * mFrameSize; 5052 mActiveTrack->releaseBuffer(&buffer); 5053 } 5054 mCurrentWriteLength = curBuf - (int8_t *)mSinkBuffer; 5055 mSleepTimeUs = 0; 5056 mStandbyTimeNs = systemTime() + mStandbyDelayNs; 5057 mActiveTrack.clear(); 5058} 5059 5060void AudioFlinger::DirectOutputThread::threadLoop_sleepTime() 5061{ 5062 // do not write to HAL when paused 5063 if (mHwPaused || (usesHwAvSync() && mStandby)) { 5064 mSleepTimeUs = mIdleSleepTimeUs; 5065 return; 5066 } 5067 if (mSleepTimeUs == 0) { 5068 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 5069 mSleepTimeUs = mActiveSleepTimeUs; 5070 } else { 5071 mSleepTimeUs = mIdleSleepTimeUs; 5072 } 5073 } else if (mBytesWritten != 0 && audio_has_proportional_frames(mFormat)) { 5074 memset(mSinkBuffer, 0, mFrameCount * mFrameSize); 5075 mSleepTimeUs = 0; 5076 } 5077} 5078 5079void AudioFlinger::DirectOutputThread::threadLoop_exit() 5080{ 5081 { 5082 Mutex::Autolock _l(mLock); 5083 for (size_t i = 0; i < mTracks.size(); i++) { 5084 if (mTracks[i]->isFlushPending()) { 5085 mTracks[i]->flushAck(); 5086 mFlushPending = true; 5087 } 5088 } 5089 if (mFlushPending) { 5090 flushHw_l(); 5091 } 5092 } 5093 PlaybackThread::threadLoop_exit(); 5094} 5095 5096// must be called with thread mutex locked 5097bool AudioFlinger::DirectOutputThread::shouldStandby_l() 5098{ 5099 bool trackPaused = false; 5100 bool trackStopped = false; 5101 5102 if ((mType == DIRECT) && audio_is_linear_pcm(mFormat) && !usesHwAvSync()) { 5103 return !mStandby; 5104 } 5105 5106 // do not put the HAL in standby when paused. AwesomePlayer clear the offloaded AudioTrack 5107 // after a timeout and we will enter standby then. 5108 if (mTracks.size() > 0) { 5109 trackPaused = mTracks[mTracks.size() - 1]->isPaused(); 5110 trackStopped = mTracks[mTracks.size() - 1]->isStopped() || 5111 mTracks[mTracks.size() - 1]->mState == TrackBase::IDLE; 5112 } 5113 5114 return !mStandby && !(trackPaused || (mHwPaused && !trackStopped)); 5115} 5116 5117// getTrackName_l() must be called with ThreadBase::mLock held 5118int AudioFlinger::DirectOutputThread::getTrackName_l(audio_channel_mask_t channelMask __unused, 5119 audio_format_t format __unused, audio_session_t sessionId __unused) 5120{ 5121 return 0; 5122} 5123 5124// deleteTrackName_l() must be called with ThreadBase::mLock held 5125void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name __unused) 5126{ 5127} 5128 5129// checkForNewParameter_l() must be called with ThreadBase::mLock held 5130bool AudioFlinger::DirectOutputThread::checkForNewParameter_l(const String8& keyValuePair, 5131 status_t& status) 5132{ 5133 bool reconfig = false; 5134 bool a2dpDeviceChanged = false; 5135 5136 status = NO_ERROR; 5137 5138 AudioParameter param = AudioParameter(keyValuePair); 5139 int value; 5140 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 5141 // forward device change to effects that have requested to be 5142 // aware of attached audio device. 5143 if (value != AUDIO_DEVICE_NONE) { 5144 a2dpDeviceChanged = 5145 (mOutDevice & AUDIO_DEVICE_OUT_ALL_A2DP) != (value & AUDIO_DEVICE_OUT_ALL_A2DP); 5146 mOutDevice = value; 5147 for (size_t i = 0; i < mEffectChains.size(); i++) { 5148 mEffectChains[i]->setDevice_l(mOutDevice); 5149 } 5150 } 5151 } 5152 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 5153 // do not accept frame count changes if tracks are open as the track buffer 5154 // size depends on frame count and correct behavior would not be garantied 5155 // if frame count is changed after track creation 5156 if (!mTracks.isEmpty()) { 5157 status = INVALID_OPERATION; 5158 } else { 5159 reconfig = true; 5160 } 5161 } 5162 if (status == NO_ERROR) { 5163 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 5164 keyValuePair.string()); 5165 if (!mStandby && status == INVALID_OPERATION) { 5166 mOutput->standby(); 5167 mStandby = true; 5168 mBytesWritten = 0; 5169 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 5170 keyValuePair.string()); 5171 } 5172 if (status == NO_ERROR && reconfig) { 5173 readOutputParameters_l(); 5174 sendIoConfigEvent_l(AUDIO_OUTPUT_CONFIG_CHANGED); 5175 } 5176 } 5177 5178 return reconfig || a2dpDeviceChanged; 5179} 5180 5181uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() const 5182{ 5183 uint32_t time; 5184 if (audio_has_proportional_frames(mFormat)) { 5185 time = PlaybackThread::activeSleepTimeUs(); 5186 } else { 5187 time = kDirectMinSleepTimeUs; 5188 } 5189 return time; 5190} 5191 5192uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() const 5193{ 5194 uint32_t time; 5195 if (audio_has_proportional_frames(mFormat)) { 5196 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 5197 } else { 5198 time = kDirectMinSleepTimeUs; 5199 } 5200 return time; 5201} 5202 5203uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() const 5204{ 5205 uint32_t time; 5206 if (audio_has_proportional_frames(mFormat)) { 5207 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 5208 } else { 5209 time = kDirectMinSleepTimeUs; 5210 } 5211 return time; 5212} 5213 5214void AudioFlinger::DirectOutputThread::cacheParameters_l() 5215{ 5216 PlaybackThread::cacheParameters_l(); 5217 5218 // use shorter standby delay as on normal output to release 5219 // hardware resources as soon as possible 5220 // no delay on outputs with HW A/V sync 5221 if (usesHwAvSync()) { 5222 mStandbyDelayNs = 0; 5223 } else if ((mType == OFFLOAD) && !audio_has_proportional_frames(mFormat)) { 5224 mStandbyDelayNs = kOffloadStandbyDelayNs; 5225 } else { 5226 mStandbyDelayNs = microseconds(mActiveSleepTimeUs*2); 5227 } 5228} 5229 5230void AudioFlinger::DirectOutputThread::flushHw_l() 5231{ 5232 mOutput->flush(); 5233 mHwPaused = false; 5234 mFlushPending = false; 5235} 5236 5237// ---------------------------------------------------------------------------- 5238 5239AudioFlinger::AsyncCallbackThread::AsyncCallbackThread( 5240 const wp<AudioFlinger::PlaybackThread>& playbackThread) 5241 : Thread(false /*canCallJava*/), 5242 mPlaybackThread(playbackThread), 5243 mWriteAckSequence(0), 5244 mDrainSequence(0), 5245 mAsyncError(false) 5246{ 5247} 5248 5249AudioFlinger::AsyncCallbackThread::~AsyncCallbackThread() 5250{ 5251} 5252 5253void AudioFlinger::AsyncCallbackThread::onFirstRef() 5254{ 5255 run("Offload Cbk", ANDROID_PRIORITY_URGENT_AUDIO); 5256} 5257 5258bool AudioFlinger::AsyncCallbackThread::threadLoop() 5259{ 5260 while (!exitPending()) { 5261 uint32_t writeAckSequence; 5262 uint32_t drainSequence; 5263 bool asyncError; 5264 5265 { 5266 Mutex::Autolock _l(mLock); 5267 while (!((mWriteAckSequence & 1) || 5268 (mDrainSequence & 1) || 5269 mAsyncError || 5270 exitPending())) { 5271 mWaitWorkCV.wait(mLock); 5272 } 5273 5274 if (exitPending()) { 5275 break; 5276 } 5277 ALOGV("AsyncCallbackThread mWriteAckSequence %d mDrainSequence %d", 5278 mWriteAckSequence, mDrainSequence); 5279 writeAckSequence = mWriteAckSequence; 5280 mWriteAckSequence &= ~1; 5281 drainSequence = mDrainSequence; 5282 mDrainSequence &= ~1; 5283 asyncError = mAsyncError; 5284 mAsyncError = false; 5285 } 5286 { 5287 sp<AudioFlinger::PlaybackThread> playbackThread = mPlaybackThread.promote(); 5288 if (playbackThread != 0) { 5289 if (writeAckSequence & 1) { 5290 playbackThread->resetWriteBlocked(writeAckSequence >> 1); 5291 } 5292 if (drainSequence & 1) { 5293 playbackThread->resetDraining(drainSequence >> 1); 5294 } 5295 if (asyncError) { 5296 playbackThread->onAsyncError(); 5297 } 5298 } 5299 } 5300 } 5301 return false; 5302} 5303 5304void AudioFlinger::AsyncCallbackThread::exit() 5305{ 5306 ALOGV("AsyncCallbackThread::exit"); 5307 Mutex::Autolock _l(mLock); 5308 requestExit(); 5309 mWaitWorkCV.broadcast(); 5310} 5311 5312void AudioFlinger::AsyncCallbackThread::setWriteBlocked(uint32_t sequence) 5313{ 5314 Mutex::Autolock _l(mLock); 5315 // bit 0 is cleared 5316 mWriteAckSequence = sequence << 1; 5317} 5318 5319void AudioFlinger::AsyncCallbackThread::resetWriteBlocked() 5320{ 5321 Mutex::Autolock _l(mLock); 5322 // ignore unexpected callbacks 5323 if (mWriteAckSequence & 2) { 5324 mWriteAckSequence |= 1; 5325 mWaitWorkCV.signal(); 5326 } 5327} 5328 5329void AudioFlinger::AsyncCallbackThread::setDraining(uint32_t sequence) 5330{ 5331 Mutex::Autolock _l(mLock); 5332 // bit 0 is cleared 5333 mDrainSequence = sequence << 1; 5334} 5335 5336void AudioFlinger::AsyncCallbackThread::resetDraining() 5337{ 5338 Mutex::Autolock _l(mLock); 5339 // ignore unexpected callbacks 5340 if (mDrainSequence & 2) { 5341 mDrainSequence |= 1; 5342 mWaitWorkCV.signal(); 5343 } 5344} 5345 5346void AudioFlinger::AsyncCallbackThread::setAsyncError() 5347{ 5348 Mutex::Autolock _l(mLock); 5349 mAsyncError = true; 5350 mWaitWorkCV.signal(); 5351} 5352 5353 5354// ---------------------------------------------------------------------------- 5355AudioFlinger::OffloadThread::OffloadThread(const sp<AudioFlinger>& audioFlinger, 5356 AudioStreamOut* output, audio_io_handle_t id, uint32_t device, bool systemReady) 5357 : DirectOutputThread(audioFlinger, output, id, device, OFFLOAD, systemReady), 5358 mPausedWriteLength(0), mPausedBytesRemaining(0), mKeepWakeLock(true) 5359{ 5360 //FIXME: mStandby should be set to true by ThreadBase constructor 5361 mStandby = true; 5362 mKeepWakeLock = property_get_bool("ro.audio.offload_wakelock", true /* default_value */); 5363} 5364 5365void AudioFlinger::OffloadThread::threadLoop_exit() 5366{ 5367 if (mFlushPending || mHwPaused) { 5368 // If a flush is pending or track was paused, just discard buffered data 5369 flushHw_l(); 5370 } else { 5371 mMixerStatus = MIXER_DRAIN_ALL; 5372 threadLoop_drain(); 5373 } 5374 if (mUseAsyncWrite) { 5375 ALOG_ASSERT(mCallbackThread != 0); 5376 mCallbackThread->exit(); 5377 } 5378 PlaybackThread::threadLoop_exit(); 5379} 5380 5381AudioFlinger::PlaybackThread::mixer_state AudioFlinger::OffloadThread::prepareTracks_l( 5382 Vector< sp<Track> > *tracksToRemove 5383) 5384{ 5385 size_t count = mActiveTracks.size(); 5386 5387 mixer_state mixerStatus = MIXER_IDLE; 5388 bool doHwPause = false; 5389 bool doHwResume = false; 5390 5391 ALOGV("OffloadThread::prepareTracks_l active tracks %zu", count); 5392 5393 // find out which tracks need to be processed 5394 for (size_t i = 0; i < count; i++) { 5395 sp<Track> t = mActiveTracks[i].promote(); 5396 // The track died recently 5397 if (t == 0) { 5398 continue; 5399 } 5400 Track* const track = t.get(); 5401#ifdef VERY_VERY_VERBOSE_LOGGING 5402 audio_track_cblk_t* cblk = track->cblk(); 5403#endif 5404 // Only consider last track started for volume and mixer state control. 5405 // In theory an older track could underrun and restart after the new one starts 5406 // but as we only care about the transition phase between two tracks on a 5407 // direct output, it is not a problem to ignore the underrun case. 5408 sp<Track> l = mLatestActiveTrack.promote(); 5409 bool last = l.get() == track; 5410 5411 if (track->isInvalid()) { 5412 ALOGW("An invalidated track shouldn't be in active list"); 5413 tracksToRemove->add(track); 5414 continue; 5415 } 5416 5417 if (track->mState == TrackBase::IDLE) { 5418 ALOGW("An idle track shouldn't be in active list"); 5419 continue; 5420 } 5421 5422 if (track->isPausing()) { 5423 track->setPaused(); 5424 if (last) { 5425 if (mHwSupportsPause && !mHwPaused) { 5426 doHwPause = true; 5427 mHwPaused = true; 5428 } 5429 // If we were part way through writing the mixbuffer to 5430 // the HAL we must save this until we resume 5431 // BUG - this will be wrong if a different track is made active, 5432 // in that case we want to discard the pending data in the 5433 // mixbuffer and tell the client to present it again when the 5434 // track is resumed 5435 mPausedWriteLength = mCurrentWriteLength; 5436 mPausedBytesRemaining = mBytesRemaining; 5437 mBytesRemaining = 0; // stop writing 5438 } 5439 tracksToRemove->add(track); 5440 } else if (track->isFlushPending()) { 5441 if (track->isStopping_1()) { 5442 track->mRetryCount = kMaxTrackStopRetriesOffload; 5443 } else { 5444 track->mRetryCount = kMaxTrackRetriesOffload; 5445 } 5446 track->flushAck(); 5447 if (last) { 5448 mFlushPending = true; 5449 } 5450 } else if (track->isResumePending()){ 5451 track->resumeAck(); 5452 if (last) { 5453 if (mPausedBytesRemaining) { 5454 // Need to continue write that was interrupted 5455 mCurrentWriteLength = mPausedWriteLength; 5456 mBytesRemaining = mPausedBytesRemaining; 5457 mPausedBytesRemaining = 0; 5458 } 5459 if (mHwPaused) { 5460 doHwResume = true; 5461 mHwPaused = false; 5462 // threadLoop_mix() will handle the case that we need to 5463 // resume an interrupted write 5464 } 5465 // enable write to audio HAL 5466 mSleepTimeUs = 0; 5467 5468 mLeftVolFloat = mRightVolFloat = -1.0; 5469 5470 // Do not handle new data in this iteration even if track->framesReady() 5471 mixerStatus = MIXER_TRACKS_ENABLED; 5472 } 5473 } else if (track->framesReady() && track->isReady() && 5474 !track->isPaused() && !track->isTerminated() && !track->isStopping_2()) { 5475 ALOGVV("OffloadThread: track %d s=%08x [OK]", track->name(), cblk->mServer); 5476 if (track->mFillingUpStatus == Track::FS_FILLED) { 5477 track->mFillingUpStatus = Track::FS_ACTIVE; 5478 if (last) { 5479 // make sure processVolume_l() will apply new volume even if 0 5480 mLeftVolFloat = mRightVolFloat = -1.0; 5481 } 5482 } 5483 5484 if (last) { 5485 sp<Track> previousTrack = mPreviousTrack.promote(); 5486 if (previousTrack != 0) { 5487 if (track != previousTrack.get()) { 5488 // Flush any data still being written from last track 5489 mBytesRemaining = 0; 5490 if (mPausedBytesRemaining) { 5491 // Last track was paused so we also need to flush saved 5492 // mixbuffer state and invalidate track so that it will 5493 // re-submit that unwritten data when it is next resumed 5494 mPausedBytesRemaining = 0; 5495 // Invalidate is a bit drastic - would be more efficient 5496 // to have a flag to tell client that some of the 5497 // previously written data was lost 5498 previousTrack->invalidate(); 5499 } 5500 // flush data already sent to the DSP if changing audio session as audio 5501 // comes from a different source. Also invalidate previous track to force a 5502 // seek when resuming. 5503 if (previousTrack->sessionId() != track->sessionId()) { 5504 previousTrack->invalidate(); 5505 } 5506 } 5507 } 5508 mPreviousTrack = track; 5509 // reset retry count 5510 if (track->isStopping_1()) { 5511 track->mRetryCount = kMaxTrackStopRetriesOffload; 5512 } else { 5513 track->mRetryCount = kMaxTrackRetriesOffload; 5514 } 5515 mActiveTrack = t; 5516 mixerStatus = MIXER_TRACKS_READY; 5517 } 5518 } else { 5519 ALOGVV("OffloadThread: track %d s=%08x [NOT READY]", track->name(), cblk->mServer); 5520 if (track->isStopping_1()) { 5521 if (--(track->mRetryCount) <= 0) { 5522 // Hardware buffer can hold a large amount of audio so we must 5523 // wait for all current track's data to drain before we say 5524 // that the track is stopped. 5525 if (mBytesRemaining == 0) { 5526 // Only start draining when all data in mixbuffer 5527 // has been written 5528 ALOGV("OffloadThread: underrun and STOPPING_1 -> draining, STOPPING_2"); 5529 track->mState = TrackBase::STOPPING_2; // so presentation completes after 5530 // drain do not drain if no data was ever sent to HAL (mStandby == true) 5531 if (last && !mStandby) { 5532 // do not modify drain sequence if we are already draining. This happens 5533 // when resuming from pause after drain. 5534 if ((mDrainSequence & 1) == 0) { 5535 mSleepTimeUs = 0; 5536 mStandbyTimeNs = systemTime() + mStandbyDelayNs; 5537 mixerStatus = MIXER_DRAIN_TRACK; 5538 mDrainSequence += 2; 5539 } 5540 if (mHwPaused) { 5541 // It is possible to move from PAUSED to STOPPING_1 without 5542 // a resume so we must ensure hardware is running 5543 doHwResume = true; 5544 mHwPaused = false; 5545 } 5546 } 5547 } 5548 } else if (last) { 5549 ALOGV("stopping1 underrun retries left %d", track->mRetryCount); 5550 mixerStatus = MIXER_TRACKS_ENABLED; 5551 } 5552 } else if (track->isStopping_2()) { 5553 // Drain has completed or we are in standby, signal presentation complete 5554 if (!(mDrainSequence & 1) || !last || mStandby) { 5555 track->mState = TrackBase::STOPPED; 5556 size_t audioHALFrames = 5557 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 5558 int64_t framesWritten = 5559 mBytesWritten / mOutput->getFrameSize(); 5560 track->presentationComplete(framesWritten, audioHALFrames); 5561 track->reset(); 5562 tracksToRemove->add(track); 5563 } 5564 } else { 5565 // No buffers for this track. Give it a few chances to 5566 // fill a buffer, then remove it from active list. 5567 if (--(track->mRetryCount) <= 0) { 5568 ALOGV("OffloadThread: BUFFER TIMEOUT: remove(%d) from active list", 5569 track->name()); 5570 tracksToRemove->add(track); 5571 // indicate to client process that the track was disabled because of underrun; 5572 // it will then automatically call start() when data is available 5573 track->disable(); 5574 } else if (last){ 5575 mixerStatus = MIXER_TRACKS_ENABLED; 5576 } 5577 } 5578 } 5579 // compute volume for this track 5580 processVolume_l(track, last); 5581 } 5582 5583 // make sure the pause/flush/resume sequence is executed in the right order. 5584 // If a flush is pending and a track is active but the HW is not paused, force a HW pause 5585 // before flush and then resume HW. This can happen in case of pause/flush/resume 5586 // if resume is received before pause is executed. 5587 if (!mStandby && (doHwPause || (mFlushPending && !mHwPaused && (count != 0)))) { 5588 mOutput->stream->pause(mOutput->stream); 5589 } 5590 if (mFlushPending) { 5591 flushHw_l(); 5592 } 5593 if (!mStandby && doHwResume) { 5594 mOutput->stream->resume(mOutput->stream); 5595 } 5596 5597 // remove all the tracks that need to be... 5598 removeTracks_l(*tracksToRemove); 5599 5600 return mixerStatus; 5601} 5602 5603// must be called with thread mutex locked 5604bool AudioFlinger::OffloadThread::waitingAsyncCallback_l() 5605{ 5606 ALOGVV("waitingAsyncCallback_l mWriteAckSequence %d mDrainSequence %d", 5607 mWriteAckSequence, mDrainSequence); 5608 if (mUseAsyncWrite && ((mWriteAckSequence & 1) || (mDrainSequence & 1))) { 5609 return true; 5610 } 5611 return false; 5612} 5613 5614bool AudioFlinger::OffloadThread::waitingAsyncCallback() 5615{ 5616 Mutex::Autolock _l(mLock); 5617 return waitingAsyncCallback_l(); 5618} 5619 5620void AudioFlinger::OffloadThread::flushHw_l() 5621{ 5622 DirectOutputThread::flushHw_l(); 5623 // Flush anything still waiting in the mixbuffer 5624 mCurrentWriteLength = 0; 5625 mBytesRemaining = 0; 5626 mPausedWriteLength = 0; 5627 mPausedBytesRemaining = 0; 5628 // reset bytes written count to reflect that DSP buffers are empty after flush. 5629 mBytesWritten = 0; 5630 5631 if (mUseAsyncWrite) { 5632 // discard any pending drain or write ack by incrementing sequence 5633 mWriteAckSequence = (mWriteAckSequence + 2) & ~1; 5634 mDrainSequence = (mDrainSequence + 2) & ~1; 5635 ALOG_ASSERT(mCallbackThread != 0); 5636 mCallbackThread->setWriteBlocked(mWriteAckSequence); 5637 mCallbackThread->setDraining(mDrainSequence); 5638 } 5639} 5640 5641void AudioFlinger::OffloadThread::invalidateTracks(audio_stream_type_t streamType) 5642{ 5643 Mutex::Autolock _l(mLock); 5644 if (PlaybackThread::invalidateTracks_l(streamType)) { 5645 mFlushPending = true; 5646 } 5647} 5648 5649// ---------------------------------------------------------------------------- 5650 5651AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 5652 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id, bool systemReady) 5653 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->outDevice(), 5654 systemReady, DUPLICATING), 5655 mWaitTimeMs(UINT_MAX) 5656{ 5657 addOutputTrack(mainThread); 5658} 5659 5660AudioFlinger::DuplicatingThread::~DuplicatingThread() 5661{ 5662 for (size_t i = 0; i < mOutputTracks.size(); i++) { 5663 mOutputTracks[i]->destroy(); 5664 } 5665} 5666 5667void AudioFlinger::DuplicatingThread::threadLoop_mix() 5668{ 5669 // mix buffers... 5670 if (outputsReady(outputTracks)) { 5671 mAudioMixer->process(); 5672 } else { 5673 if (mMixerBufferValid) { 5674 memset(mMixerBuffer, 0, mMixerBufferSize); 5675 } else { 5676 memset(mSinkBuffer, 0, mSinkBufferSize); 5677 } 5678 } 5679 mSleepTimeUs = 0; 5680 writeFrames = mNormalFrameCount; 5681 mCurrentWriteLength = mSinkBufferSize; 5682 mStandbyTimeNs = systemTime() + mStandbyDelayNs; 5683} 5684 5685void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() 5686{ 5687 if (mSleepTimeUs == 0) { 5688 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 5689 mSleepTimeUs = mActiveSleepTimeUs; 5690 } else { 5691 mSleepTimeUs = mIdleSleepTimeUs; 5692 } 5693 } else if (mBytesWritten != 0) { 5694 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 5695 writeFrames = mNormalFrameCount; 5696 memset(mSinkBuffer, 0, mSinkBufferSize); 5697 } else { 5698 // flush remaining overflow buffers in output tracks 5699 writeFrames = 0; 5700 } 5701 mSleepTimeUs = 0; 5702 } 5703} 5704 5705ssize_t AudioFlinger::DuplicatingThread::threadLoop_write() 5706{ 5707 for (size_t i = 0; i < outputTracks.size(); i++) { 5708 outputTracks[i]->write(mSinkBuffer, writeFrames); 5709 } 5710 mStandby = false; 5711 return (ssize_t)mSinkBufferSize; 5712} 5713 5714void AudioFlinger::DuplicatingThread::threadLoop_standby() 5715{ 5716 // DuplicatingThread implements standby by stopping all tracks 5717 for (size_t i = 0; i < outputTracks.size(); i++) { 5718 outputTracks[i]->stop(); 5719 } 5720} 5721 5722void AudioFlinger::DuplicatingThread::saveOutputTracks() 5723{ 5724 outputTracks = mOutputTracks; 5725} 5726 5727void AudioFlinger::DuplicatingThread::clearOutputTracks() 5728{ 5729 outputTracks.clear(); 5730} 5731 5732void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 5733{ 5734 Mutex::Autolock _l(mLock); 5735 // The downstream MixerThread consumes thread->frameCount() amount of frames per mix pass. 5736 // Adjust for thread->sampleRate() to determine minimum buffer frame count. 5737 // Then triple buffer because Threads do not run synchronously and may not be clock locked. 5738 const size_t frameCount = 5739 3 * sourceFramesNeeded(mSampleRate, thread->frameCount(), thread->sampleRate()); 5740 // TODO: Consider asynchronous sample rate conversion to handle clock disparity 5741 // from different OutputTracks and their associated MixerThreads (e.g. one may 5742 // nearly empty and the other may be dropping data). 5743 5744 sp<OutputTrack> outputTrack = new OutputTrack(thread, 5745 this, 5746 mSampleRate, 5747 mFormat, 5748 mChannelMask, 5749 frameCount, 5750 IPCThreadState::self()->getCallingUid()); 5751 if (outputTrack->cblk() != NULL) { 5752 thread->setStreamVolume(AUDIO_STREAM_PATCH, 1.0f); 5753 mOutputTracks.add(outputTrack); 5754 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack.get(), thread); 5755 updateWaitTime_l(); 5756 } 5757} 5758 5759void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 5760{ 5761 Mutex::Autolock _l(mLock); 5762 for (size_t i = 0; i < mOutputTracks.size(); i++) { 5763 if (mOutputTracks[i]->thread() == thread) { 5764 mOutputTracks[i]->destroy(); 5765 mOutputTracks.removeAt(i); 5766 updateWaitTime_l(); 5767 if (thread->getOutput() == mOutput) { 5768 mOutput = NULL; 5769 } 5770 return; 5771 } 5772 } 5773 ALOGV("removeOutputTrack(): unknown thread: %p", thread); 5774} 5775 5776// caller must hold mLock 5777void AudioFlinger::DuplicatingThread::updateWaitTime_l() 5778{ 5779 mWaitTimeMs = UINT_MAX; 5780 for (size_t i = 0; i < mOutputTracks.size(); i++) { 5781 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 5782 if (strong != 0) { 5783 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 5784 if (waitTimeMs < mWaitTimeMs) { 5785 mWaitTimeMs = waitTimeMs; 5786 } 5787 } 5788 } 5789} 5790 5791 5792bool AudioFlinger::DuplicatingThread::outputsReady( 5793 const SortedVector< sp<OutputTrack> > &outputTracks) 5794{ 5795 for (size_t i = 0; i < outputTracks.size(); i++) { 5796 sp<ThreadBase> thread = outputTracks[i]->thread().promote(); 5797 if (thread == 0) { 5798 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", 5799 outputTracks[i].get()); 5800 return false; 5801 } 5802 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 5803 // see note at standby() declaration 5804 if (playbackThread->standby() && !playbackThread->isSuspended()) { 5805 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), 5806 thread.get()); 5807 return false; 5808 } 5809 } 5810 return true; 5811} 5812 5813uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() const 5814{ 5815 return (mWaitTimeMs * 1000) / 2; 5816} 5817 5818void AudioFlinger::DuplicatingThread::cacheParameters_l() 5819{ 5820 // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first 5821 updateWaitTime_l(); 5822 5823 MixerThread::cacheParameters_l(); 5824} 5825 5826// ---------------------------------------------------------------------------- 5827// Record 5828// ---------------------------------------------------------------------------- 5829 5830AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 5831 AudioStreamIn *input, 5832 audio_io_handle_t id, 5833 audio_devices_t outDevice, 5834 audio_devices_t inDevice, 5835 bool systemReady 5836#ifdef TEE_SINK 5837 , const sp<NBAIO_Sink>& teeSink 5838#endif 5839 ) : 5840 ThreadBase(audioFlinger, id, outDevice, inDevice, RECORD, systemReady), 5841 mInput(input), mActiveTracksGen(0), mRsmpInBuffer(NULL), 5842 // mRsmpInFrames and mRsmpInFramesP2 are set by readInputParameters_l() 5843 mRsmpInRear(0) 5844#ifdef TEE_SINK 5845 , mTeeSink(teeSink) 5846#endif 5847 , mReadOnlyHeap(new MemoryDealer(kRecordThreadReadOnlyHeapSize, 5848 "RecordThreadRO", MemoryHeapBase::READ_ONLY)) 5849 // mFastCapture below 5850 , mFastCaptureFutex(0) 5851 // mInputSource 5852 // mPipeSink 5853 // mPipeSource 5854 , mPipeFramesP2(0) 5855 // mPipeMemory 5856 // mFastCaptureNBLogWriter 5857 , mFastTrackAvail(false) 5858{ 5859 snprintf(mThreadName, kThreadNameLength, "AudioIn_%X", id); 5860 mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mThreadName); 5861 5862 readInputParameters_l(); 5863 5864 // create an NBAIO source for the HAL input stream, and negotiate 5865 mInputSource = new AudioStreamInSource(input->stream); 5866 size_t numCounterOffers = 0; 5867 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount, mFormat)}; 5868#if !LOG_NDEBUG 5869 ssize_t index = 5870#else 5871 (void) 5872#endif 5873 mInputSource->negotiate(offers, 1, NULL, numCounterOffers); 5874 ALOG_ASSERT(index == 0); 5875 5876 // initialize fast capture depending on configuration 5877 bool initFastCapture; 5878 switch (kUseFastCapture) { 5879 case FastCapture_Never: 5880 initFastCapture = false; 5881 break; 5882 case FastCapture_Always: 5883 initFastCapture = true; 5884 break; 5885 case FastCapture_Static: 5886 initFastCapture = (mFrameCount * 1000) / mSampleRate < kMinNormalCaptureBufferSizeMs; 5887 break; 5888 // case FastCapture_Dynamic: 5889 } 5890 5891 if (initFastCapture) { 5892 // create a Pipe for FastCapture to write to, and for us and fast tracks to read from 5893 NBAIO_Format format = mInputSource->format(); 5894 size_t pipeFramesP2 = roundup(mSampleRate / 25); // double-buffering of 20 ms each 5895 size_t pipeSize = pipeFramesP2 * Format_frameSize(format); 5896 void *pipeBuffer; 5897 const sp<MemoryDealer> roHeap(readOnlyHeap()); 5898 sp<IMemory> pipeMemory; 5899 if ((roHeap == 0) || 5900 (pipeMemory = roHeap->allocate(pipeSize)) == 0 || 5901 (pipeBuffer = pipeMemory->pointer()) == NULL) { 5902 ALOGE("not enough memory for pipe buffer size=%zu", pipeSize); 5903 goto failed; 5904 } 5905 // pipe will be shared directly with fast clients, so clear to avoid leaking old information 5906 memset(pipeBuffer, 0, pipeSize); 5907 Pipe *pipe = new Pipe(pipeFramesP2, format, pipeBuffer); 5908 const NBAIO_Format offers[1] = {format}; 5909 size_t numCounterOffers = 0; 5910 ssize_t index = pipe->negotiate(offers, 1, NULL, numCounterOffers); 5911 ALOG_ASSERT(index == 0); 5912 mPipeSink = pipe; 5913 PipeReader *pipeReader = new PipeReader(*pipe); 5914 numCounterOffers = 0; 5915 index = pipeReader->negotiate(offers, 1, NULL, numCounterOffers); 5916 ALOG_ASSERT(index == 0); 5917 mPipeSource = pipeReader; 5918 mPipeFramesP2 = pipeFramesP2; 5919 mPipeMemory = pipeMemory; 5920 5921 // create fast capture 5922 mFastCapture = new FastCapture(); 5923 FastCaptureStateQueue *sq = mFastCapture->sq(); 5924#ifdef STATE_QUEUE_DUMP 5925 // FIXME 5926#endif 5927 FastCaptureState *state = sq->begin(); 5928 state->mCblk = NULL; 5929 state->mInputSource = mInputSource.get(); 5930 state->mInputSourceGen++; 5931 state->mPipeSink = pipe; 5932 state->mPipeSinkGen++; 5933 state->mFrameCount = mFrameCount; 5934 state->mCommand = FastCaptureState::COLD_IDLE; 5935 // already done in constructor initialization list 5936 //mFastCaptureFutex = 0; 5937 state->mColdFutexAddr = &mFastCaptureFutex; 5938 state->mColdGen++; 5939 state->mDumpState = &mFastCaptureDumpState; 5940#ifdef TEE_SINK 5941 // FIXME 5942#endif 5943 mFastCaptureNBLogWriter = audioFlinger->newWriter_l(kFastCaptureLogSize, "FastCapture"); 5944 state->mNBLogWriter = mFastCaptureNBLogWriter.get(); 5945 sq->end(); 5946 sq->push(FastCaptureStateQueue::BLOCK_UNTIL_PUSHED); 5947 5948 // start the fast capture 5949 mFastCapture->run("FastCapture", ANDROID_PRIORITY_URGENT_AUDIO); 5950 pid_t tid = mFastCapture->getTid(); 5951 sendPrioConfigEvent(getpid_cached, tid, kPriorityFastCapture); 5952#ifdef AUDIO_WATCHDOG 5953 // FIXME 5954#endif 5955 5956 mFastTrackAvail = true; 5957 } 5958failed: ; 5959 5960 // FIXME mNormalSource 5961} 5962 5963AudioFlinger::RecordThread::~RecordThread() 5964{ 5965 if (mFastCapture != 0) { 5966 FastCaptureStateQueue *sq = mFastCapture->sq(); 5967 FastCaptureState *state = sq->begin(); 5968 if (state->mCommand == FastCaptureState::COLD_IDLE) { 5969 int32_t old = android_atomic_inc(&mFastCaptureFutex); 5970 if (old == -1) { 5971 (void) syscall(__NR_futex, &mFastCaptureFutex, FUTEX_WAKE_PRIVATE, 1); 5972 } 5973 } 5974 state->mCommand = FastCaptureState::EXIT; 5975 sq->end(); 5976 sq->push(FastCaptureStateQueue::BLOCK_UNTIL_PUSHED); 5977 mFastCapture->join(); 5978 mFastCapture.clear(); 5979 } 5980 mAudioFlinger->unregisterWriter(mFastCaptureNBLogWriter); 5981 mAudioFlinger->unregisterWriter(mNBLogWriter); 5982 free(mRsmpInBuffer); 5983} 5984 5985void AudioFlinger::RecordThread::onFirstRef() 5986{ 5987 run(mThreadName, PRIORITY_URGENT_AUDIO); 5988} 5989 5990bool AudioFlinger::RecordThread::threadLoop() 5991{ 5992 nsecs_t lastWarning = 0; 5993 5994 inputStandBy(); 5995 5996reacquire_wakelock: 5997 sp<RecordTrack> activeTrack; 5998 int activeTracksGen; 5999 { 6000 Mutex::Autolock _l(mLock); 6001 size_t size = mActiveTracks.size(); 6002 activeTracksGen = mActiveTracksGen; 6003 if (size > 0) { 6004 // FIXME an arbitrary choice 6005 activeTrack = mActiveTracks[0]; 6006 acquireWakeLock_l(activeTrack->uid()); 6007 if (size > 1) { 6008 SortedVector<int> tmp; 6009 for (size_t i = 0; i < size; i++) { 6010 tmp.add(mActiveTracks[i]->uid()); 6011 } 6012 updateWakeLockUids_l(tmp); 6013 } 6014 } else { 6015 acquireWakeLock_l(-1); 6016 } 6017 } 6018 6019 // used to request a deferred sleep, to be executed later while mutex is unlocked 6020 uint32_t sleepUs = 0; 6021 6022 // loop while there is work to do 6023 for (;;) { 6024 Vector< sp<EffectChain> > effectChains; 6025 6026 // activeTracks accumulates a copy of a subset of mActiveTracks 6027 Vector< sp<RecordTrack> > activeTracks; 6028 6029 // reference to the (first and only) active fast track 6030 sp<RecordTrack> fastTrack; 6031 6032 // reference to a fast track which is about to be removed 6033 sp<RecordTrack> fastTrackToRemove; 6034 6035 { // scope for mLock 6036 Mutex::Autolock _l(mLock); 6037 6038 processConfigEvents_l(); 6039 6040 // check exitPending here because checkForNewParameters_l() and 6041 // checkForNewParameters_l() can temporarily release mLock 6042 if (exitPending()) { 6043 break; 6044 } 6045 6046 // sleep with mutex unlocked 6047 if (sleepUs > 0) { 6048 ATRACE_BEGIN("sleepC"); 6049 mWaitWorkCV.waitRelative(mLock, microseconds((nsecs_t)sleepUs)); 6050 ATRACE_END(); 6051 sleepUs = 0; 6052 continue; 6053 } 6054 6055 // if no active track(s), then standby and release wakelock 6056 size_t size = mActiveTracks.size(); 6057 if (size == 0) { 6058 standbyIfNotAlreadyInStandby(); 6059 // exitPending() can't become true here 6060 releaseWakeLock_l(); 6061 ALOGV("RecordThread: loop stopping"); 6062 // go to sleep 6063 mWaitWorkCV.wait(mLock); 6064 ALOGV("RecordThread: loop starting"); 6065 goto reacquire_wakelock; 6066 } 6067 6068 if (mActiveTracksGen != activeTracksGen) { 6069 activeTracksGen = mActiveTracksGen; 6070 SortedVector<int> tmp; 6071 for (size_t i = 0; i < size; i++) { 6072 tmp.add(mActiveTracks[i]->uid()); 6073 } 6074 updateWakeLockUids_l(tmp); 6075 } 6076 6077 bool doBroadcast = false; 6078 bool allStopped = true; 6079 for (size_t i = 0; i < size; ) { 6080 6081 activeTrack = mActiveTracks[i]; 6082 if (activeTrack->isTerminated()) { 6083 if (activeTrack->isFastTrack()) { 6084 ALOG_ASSERT(fastTrackToRemove == 0); 6085 fastTrackToRemove = activeTrack; 6086 } 6087 removeTrack_l(activeTrack); 6088 mActiveTracks.remove(activeTrack); 6089 mActiveTracksGen++; 6090 size--; 6091 continue; 6092 } 6093 6094 TrackBase::track_state activeTrackState = activeTrack->mState; 6095 switch (activeTrackState) { 6096 6097 case TrackBase::PAUSING: 6098 mActiveTracks.remove(activeTrack); 6099 mActiveTracksGen++; 6100 doBroadcast = true; 6101 size--; 6102 continue; 6103 6104 case TrackBase::STARTING_1: 6105 sleepUs = 10000; 6106 i++; 6107 allStopped = false; 6108 continue; 6109 6110 case TrackBase::STARTING_2: 6111 doBroadcast = true; 6112 mStandby = false; 6113 activeTrack->mState = TrackBase::ACTIVE; 6114 allStopped = false; 6115 break; 6116 6117 case TrackBase::ACTIVE: 6118 allStopped = false; 6119 break; 6120 6121 case TrackBase::IDLE: 6122 i++; 6123 continue; 6124 6125 default: 6126 LOG_ALWAYS_FATAL("Unexpected activeTrackState %d", activeTrackState); 6127 } 6128 6129 activeTracks.add(activeTrack); 6130 i++; 6131 6132 if (activeTrack->isFastTrack()) { 6133 ALOG_ASSERT(!mFastTrackAvail); 6134 ALOG_ASSERT(fastTrack == 0); 6135 fastTrack = activeTrack; 6136 } 6137 } 6138 6139 if (allStopped) { 6140 standbyIfNotAlreadyInStandby(); 6141 } 6142 if (doBroadcast) { 6143 mStartStopCond.broadcast(); 6144 } 6145 6146 // sleep if there are no active tracks to process 6147 if (activeTracks.size() == 0) { 6148 if (sleepUs == 0) { 6149 sleepUs = kRecordThreadSleepUs; 6150 } 6151 continue; 6152 } 6153 sleepUs = 0; 6154 6155 lockEffectChains_l(effectChains); 6156 } 6157 6158 // thread mutex is now unlocked, mActiveTracks unknown, activeTracks.size() > 0 6159 6160 size_t size = effectChains.size(); 6161 for (size_t i = 0; i < size; i++) { 6162 // thread mutex is not locked, but effect chain is locked 6163 effectChains[i]->process_l(); 6164 } 6165 6166 // Push a new fast capture state if fast capture is not already running, or cblk change 6167 if (mFastCapture != 0) { 6168 FastCaptureStateQueue *sq = mFastCapture->sq(); 6169 FastCaptureState *state = sq->begin(); 6170 bool didModify = false; 6171 FastCaptureStateQueue::block_t block = FastCaptureStateQueue::BLOCK_UNTIL_PUSHED; 6172 if (state->mCommand != FastCaptureState::READ_WRITE /* FIXME && 6173 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)*/) { 6174 if (state->mCommand == FastCaptureState::COLD_IDLE) { 6175 int32_t old = android_atomic_inc(&mFastCaptureFutex); 6176 if (old == -1) { 6177 (void) syscall(__NR_futex, &mFastCaptureFutex, FUTEX_WAKE_PRIVATE, 1); 6178 } 6179 } 6180 state->mCommand = FastCaptureState::READ_WRITE; 6181#if 0 // FIXME 6182 mFastCaptureDumpState.increaseSamplingN(mAudioFlinger->isLowRamDevice() ? 6183 FastThreadDumpState::kSamplingNforLowRamDevice : 6184 FastThreadDumpState::kSamplingN); 6185#endif 6186 didModify = true; 6187 } 6188 audio_track_cblk_t *cblkOld = state->mCblk; 6189 audio_track_cblk_t *cblkNew = fastTrack != 0 ? fastTrack->cblk() : NULL; 6190 if (cblkNew != cblkOld) { 6191 state->mCblk = cblkNew; 6192 // block until acked if removing a fast track 6193 if (cblkOld != NULL) { 6194 block = FastCaptureStateQueue::BLOCK_UNTIL_ACKED; 6195 } 6196 didModify = true; 6197 } 6198 sq->end(didModify); 6199 if (didModify) { 6200 sq->push(block); 6201#if 0 6202 if (kUseFastCapture == FastCapture_Dynamic) { 6203 mNormalSource = mPipeSource; 6204 } 6205#endif 6206 } 6207 } 6208 6209 // now run the fast track destructor with thread mutex unlocked 6210 fastTrackToRemove.clear(); 6211 6212 // Read from HAL to keep up with fastest client if multiple active tracks, not slowest one. 6213 // Only the client(s) that are too slow will overrun. But if even the fastest client is too 6214 // slow, then this RecordThread will overrun by not calling HAL read often enough. 6215 // If destination is non-contiguous, first read past the nominal end of buffer, then 6216 // copy to the right place. Permitted because mRsmpInBuffer was over-allocated. 6217 6218 int32_t rear = mRsmpInRear & (mRsmpInFramesP2 - 1); 6219 ssize_t framesRead; 6220 6221 // If an NBAIO source is present, use it to read the normal capture's data 6222 if (mPipeSource != 0) { 6223 size_t framesToRead = mBufferSize / mFrameSize; 6224 framesRead = mPipeSource->read((uint8_t*)mRsmpInBuffer + rear * mFrameSize, 6225 framesToRead); 6226 if (framesRead == 0) { 6227 // since pipe is non-blocking, simulate blocking input 6228 sleepUs = (framesToRead * 1000000LL) / mSampleRate; 6229 } 6230 // otherwise use the HAL / AudioStreamIn directly 6231 } else { 6232 ATRACE_BEGIN("read"); 6233 ssize_t bytesRead = mInput->stream->read(mInput->stream, 6234 (uint8_t*)mRsmpInBuffer + rear * mFrameSize, mBufferSize); 6235 ATRACE_END(); 6236 if (bytesRead < 0) { 6237 framesRead = bytesRead; 6238 } else { 6239 framesRead = bytesRead / mFrameSize; 6240 } 6241 } 6242 6243 // Update server timestamp with server stats 6244 // systemTime() is optional if the hardware supports timestamps. 6245 mTimestamp.mPosition[ExtendedTimestamp::LOCATION_SERVER] += framesRead; 6246 mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_SERVER] = systemTime(); 6247 6248 // Update server timestamp with kernel stats 6249 if (mInput->stream->get_capture_position != nullptr) { 6250 int64_t position, time; 6251 int ret = mInput->stream->get_capture_position(mInput->stream, &position, &time); 6252 if (ret == NO_ERROR) { 6253 mTimestamp.mPosition[ExtendedTimestamp::LOCATION_KERNEL] = position; 6254 mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL] = time; 6255 // Note: In general record buffers should tend to be empty in 6256 // a properly running pipeline. 6257 // 6258 // Also, it is not advantageous to call get_presentation_position during the read 6259 // as the read obtains a lock, preventing the timestamp call from executing. 6260 } 6261 } 6262 // Use this to track timestamp information 6263 // ALOGD("%s", mTimestamp.toString().c_str()); 6264 6265 if (framesRead < 0 || (framesRead == 0 && mPipeSource == 0)) { 6266 ALOGE("read failed: framesRead=%zd", framesRead); 6267 // Force input into standby so that it tries to recover at next read attempt 6268 inputStandBy(); 6269 sleepUs = kRecordThreadSleepUs; 6270 } 6271 if (framesRead <= 0) { 6272 goto unlock; 6273 } 6274 ALOG_ASSERT(framesRead > 0); 6275 6276 if (mTeeSink != 0) { 6277 (void) mTeeSink->write((uint8_t*)mRsmpInBuffer + rear * mFrameSize, framesRead); 6278 } 6279 // If destination is non-contiguous, we now correct for reading past end of buffer. 6280 { 6281 size_t part1 = mRsmpInFramesP2 - rear; 6282 if ((size_t) framesRead > part1) { 6283 memcpy(mRsmpInBuffer, (uint8_t*)mRsmpInBuffer + mRsmpInFramesP2 * mFrameSize, 6284 (framesRead - part1) * mFrameSize); 6285 } 6286 } 6287 rear = mRsmpInRear += framesRead; 6288 6289 size = activeTracks.size(); 6290 // loop over each active track 6291 for (size_t i = 0; i < size; i++) { 6292 activeTrack = activeTracks[i]; 6293 6294 // skip fast tracks, as those are handled directly by FastCapture 6295 if (activeTrack->isFastTrack()) { 6296 continue; 6297 } 6298 6299 // TODO: This code probably should be moved to RecordTrack. 6300 // TODO: Update the activeTrack buffer converter in case of reconfigure. 6301 6302 enum { 6303 OVERRUN_UNKNOWN, 6304 OVERRUN_TRUE, 6305 OVERRUN_FALSE 6306 } overrun = OVERRUN_UNKNOWN; 6307 6308 // loop over getNextBuffer to handle circular sink 6309 for (;;) { 6310 6311 activeTrack->mSink.frameCount = ~0; 6312 status_t status = activeTrack->getNextBuffer(&activeTrack->mSink); 6313 size_t framesOut = activeTrack->mSink.frameCount; 6314 LOG_ALWAYS_FATAL_IF((status == OK) != (framesOut > 0)); 6315 6316 // check available frames and handle overrun conditions 6317 // if the record track isn't draining fast enough. 6318 bool hasOverrun; 6319 size_t framesIn; 6320 activeTrack->mResamplerBufferProvider->sync(&framesIn, &hasOverrun); 6321 if (hasOverrun) { 6322 overrun = OVERRUN_TRUE; 6323 } 6324 if (framesOut == 0 || framesIn == 0) { 6325 break; 6326 } 6327 6328 // Don't allow framesOut to be larger than what is possible with resampling 6329 // from framesIn. 6330 // This isn't strictly necessary but helps limit buffer resizing in 6331 // RecordBufferConverter. TODO: remove when no longer needed. 6332 framesOut = min(framesOut, 6333 destinationFramesPossible( 6334 framesIn, mSampleRate, activeTrack->mSampleRate)); 6335 // process frames from the RecordThread buffer provider to the RecordTrack buffer 6336 framesOut = activeTrack->mRecordBufferConverter->convert( 6337 activeTrack->mSink.raw, activeTrack->mResamplerBufferProvider, framesOut); 6338 6339 if (framesOut > 0 && (overrun == OVERRUN_UNKNOWN)) { 6340 overrun = OVERRUN_FALSE; 6341 } 6342 6343 if (activeTrack->mFramesToDrop == 0) { 6344 if (framesOut > 0) { 6345 activeTrack->mSink.frameCount = framesOut; 6346 activeTrack->releaseBuffer(&activeTrack->mSink); 6347 } 6348 } else { 6349 // FIXME could do a partial drop of framesOut 6350 if (activeTrack->mFramesToDrop > 0) { 6351 activeTrack->mFramesToDrop -= framesOut; 6352 if (activeTrack->mFramesToDrop <= 0) { 6353 activeTrack->clearSyncStartEvent(); 6354 } 6355 } else { 6356 activeTrack->mFramesToDrop += framesOut; 6357 if (activeTrack->mFramesToDrop >= 0 || activeTrack->mSyncStartEvent == 0 || 6358 activeTrack->mSyncStartEvent->isCancelled()) { 6359 ALOGW("Synced record %s, session %d, trigger session %d", 6360 (activeTrack->mFramesToDrop >= 0) ? "timed out" : "cancelled", 6361 activeTrack->sessionId(), 6362 (activeTrack->mSyncStartEvent != 0) ? 6363 activeTrack->mSyncStartEvent->triggerSession() : 6364 AUDIO_SESSION_NONE); 6365 activeTrack->clearSyncStartEvent(); 6366 } 6367 } 6368 } 6369 6370 if (framesOut == 0) { 6371 break; 6372 } 6373 } 6374 6375 switch (overrun) { 6376 case OVERRUN_TRUE: 6377 // client isn't retrieving buffers fast enough 6378 if (!activeTrack->setOverflow()) { 6379 nsecs_t now = systemTime(); 6380 // FIXME should lastWarning per track? 6381 if ((now - lastWarning) > kWarningThrottleNs) { 6382 ALOGW("RecordThread: buffer overflow"); 6383 lastWarning = now; 6384 } 6385 } 6386 break; 6387 case OVERRUN_FALSE: 6388 activeTrack->clearOverflow(); 6389 break; 6390 case OVERRUN_UNKNOWN: 6391 break; 6392 } 6393 6394 // update frame information and push timestamp out 6395 activeTrack->updateTrackFrameInfo( 6396 activeTrack->mServerProxy->framesReleased(), 6397 mTimestamp.mPosition[ExtendedTimestamp::LOCATION_SERVER], 6398 mSampleRate, mTimestamp); 6399 } 6400 6401unlock: 6402 // enable changes in effect chain 6403 unlockEffectChains(effectChains); 6404 // effectChains doesn't need to be cleared, since it is cleared by destructor at scope end 6405 } 6406 6407 standbyIfNotAlreadyInStandby(); 6408 6409 { 6410 Mutex::Autolock _l(mLock); 6411 for (size_t i = 0; i < mTracks.size(); i++) { 6412 sp<RecordTrack> track = mTracks[i]; 6413 track->invalidate(); 6414 } 6415 mActiveTracks.clear(); 6416 mActiveTracksGen++; 6417 mStartStopCond.broadcast(); 6418 } 6419 6420 releaseWakeLock(); 6421 6422 ALOGV("RecordThread %p exiting", this); 6423 return false; 6424} 6425 6426void AudioFlinger::RecordThread::standbyIfNotAlreadyInStandby() 6427{ 6428 if (!mStandby) { 6429 inputStandBy(); 6430 mStandby = true; 6431 } 6432} 6433 6434void AudioFlinger::RecordThread::inputStandBy() 6435{ 6436 // Idle the fast capture if it's currently running 6437 if (mFastCapture != 0) { 6438 FastCaptureStateQueue *sq = mFastCapture->sq(); 6439 FastCaptureState *state = sq->begin(); 6440 if (!(state->mCommand & FastCaptureState::IDLE)) { 6441 state->mCommand = FastCaptureState::COLD_IDLE; 6442 state->mColdFutexAddr = &mFastCaptureFutex; 6443 state->mColdGen++; 6444 mFastCaptureFutex = 0; 6445 sq->end(); 6446 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 6447 sq->push(FastCaptureStateQueue::BLOCK_UNTIL_ACKED); 6448#if 0 6449 if (kUseFastCapture == FastCapture_Dynamic) { 6450 // FIXME 6451 } 6452#endif 6453#ifdef AUDIO_WATCHDOG 6454 // FIXME 6455#endif 6456 } else { 6457 sq->end(false /*didModify*/); 6458 } 6459 } 6460 mInput->stream->common.standby(&mInput->stream->common); 6461} 6462 6463// RecordThread::createRecordTrack_l() must be called with AudioFlinger::mLock held 6464sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 6465 const sp<AudioFlinger::Client>& client, 6466 uint32_t sampleRate, 6467 audio_format_t format, 6468 audio_channel_mask_t channelMask, 6469 size_t *pFrameCount, 6470 audio_session_t sessionId, 6471 size_t *notificationFrames, 6472 int uid, 6473 audio_input_flags_t *flags, 6474 pid_t tid, 6475 status_t *status) 6476{ 6477 size_t frameCount = *pFrameCount; 6478 sp<RecordTrack> track; 6479 status_t lStatus; 6480 audio_input_flags_t inputFlags = mInput->flags; 6481 6482 // special case for FAST flag considered OK if fast capture is present 6483 if (hasFastCapture()) { 6484 inputFlags = (audio_input_flags_t)(inputFlags | AUDIO_INPUT_FLAG_FAST); 6485 } 6486 6487 // Check if requested flags are compatible with output stream flags 6488 if ((*flags & inputFlags) != *flags) { 6489 ALOGW("createRecordTrack_l(): mismatch between requested flags (%08x) and" 6490 " input flags (%08x)", 6491 *flags, inputFlags); 6492 *flags = (audio_input_flags_t)(*flags & inputFlags); 6493 } 6494 6495 // client expresses a preference for FAST, but we get the final say 6496 if (*flags & AUDIO_INPUT_FLAG_FAST) { 6497 if ( 6498 // we formerly checked for a callback handler (non-0 tid), 6499 // but that is no longer required for TRANSFER_OBTAIN mode 6500 // 6501 // frame count is not specified, or is exactly the pipe depth 6502 ((frameCount == 0) || (frameCount == mPipeFramesP2)) && 6503 // PCM data 6504 audio_is_linear_pcm(format) && 6505 // hardware format 6506 (format == mFormat) && 6507 // hardware channel mask 6508 (channelMask == mChannelMask) && 6509 // hardware sample rate 6510 (sampleRate == mSampleRate) && 6511 // record thread has an associated fast capture 6512 hasFastCapture() && 6513 // there are sufficient fast track slots available 6514 mFastTrackAvail 6515 ) { 6516 // check compatibility with audio effects. 6517 Mutex::Autolock _l(mLock); 6518 // Do not accept FAST flag if the session has software effects 6519 sp<EffectChain> chain = getEffectChain_l(sessionId); 6520 if (chain != 0) { 6521 ALOGV_IF((*flags & AUDIO_INPUT_FLAG_RAW) != 0, 6522 "AUDIO_INPUT_FLAG_RAW denied: effect present on session"); 6523 *flags = (audio_input_flags_t)(*flags & ~AUDIO_INPUT_FLAG_RAW); 6524 if (chain->hasSoftwareEffect()) { 6525 ALOGV("AUDIO_INPUT_FLAG_FAST denied: software effect present on session"); 6526 *flags = (audio_input_flags_t)(*flags & ~AUDIO_INPUT_FLAG_FAST); 6527 } 6528 } 6529 ALOGV_IF((*flags & AUDIO_INPUT_FLAG_FAST) != 0, 6530 "AUDIO_INPUT_FLAG_FAST accepted: frameCount=%zu mFrameCount=%zu", 6531 frameCount, mFrameCount); 6532 } else { 6533 ALOGV("AUDIO_INPUT_FLAG_FAST denied: frameCount=%zu mFrameCount=%zu mPipeFramesP2=%zu " 6534 "format=%#x isLinear=%d channelMask=%#x sampleRate=%u mSampleRate=%u " 6535 "hasFastCapture=%d tid=%d mFastTrackAvail=%d", 6536 frameCount, mFrameCount, mPipeFramesP2, 6537 format, audio_is_linear_pcm(format), channelMask, sampleRate, mSampleRate, 6538 hasFastCapture(), tid, mFastTrackAvail); 6539 *flags = (audio_input_flags_t)(*flags & ~AUDIO_INPUT_FLAG_FAST); 6540 } 6541 } 6542 6543 // compute track buffer size in frames, and suggest the notification frame count 6544 if (*flags & AUDIO_INPUT_FLAG_FAST) { 6545 // fast track: frame count is exactly the pipe depth 6546 frameCount = mPipeFramesP2; 6547 // ignore requested notificationFrames, and always notify exactly once every HAL buffer 6548 *notificationFrames = mFrameCount; 6549 } else { 6550 // not fast track: max notification period is resampled equivalent of one HAL buffer time 6551 // or 20 ms if there is a fast capture 6552 // TODO This could be a roundupRatio inline, and const 6553 size_t maxNotificationFrames = ((int64_t) (hasFastCapture() ? mSampleRate/50 : mFrameCount) 6554 * sampleRate + mSampleRate - 1) / mSampleRate; 6555 // minimum number of notification periods is at least kMinNotifications, 6556 // and at least kMinMs rounded up to a whole notification period (minNotificationsByMs) 6557 static const size_t kMinNotifications = 3; 6558 static const uint32_t kMinMs = 30; 6559 // TODO This could be a roundupRatio inline 6560 const size_t minFramesByMs = (sampleRate * kMinMs + 1000 - 1) / 1000; 6561 // TODO This could be a roundupRatio inline 6562 const size_t minNotificationsByMs = (minFramesByMs + maxNotificationFrames - 1) / 6563 maxNotificationFrames; 6564 const size_t minFrameCount = maxNotificationFrames * 6565 max(kMinNotifications, minNotificationsByMs); 6566 frameCount = max(frameCount, minFrameCount); 6567 if (*notificationFrames == 0 || *notificationFrames > maxNotificationFrames) { 6568 *notificationFrames = maxNotificationFrames; 6569 } 6570 } 6571 *pFrameCount = frameCount; 6572 6573 lStatus = initCheck(); 6574 if (lStatus != NO_ERROR) { 6575 ALOGE("createRecordTrack_l() audio driver not initialized"); 6576 goto Exit; 6577 } 6578 6579 { // scope for mLock 6580 Mutex::Autolock _l(mLock); 6581 6582 track = new RecordTrack(this, client, sampleRate, 6583 format, channelMask, frameCount, NULL, sessionId, uid, 6584 *flags, TrackBase::TYPE_DEFAULT); 6585 6586 lStatus = track->initCheck(); 6587 if (lStatus != NO_ERROR) { 6588 ALOGE("createRecordTrack_l() initCheck failed %d; no control block?", lStatus); 6589 // track must be cleared from the caller as the caller has the AF lock 6590 goto Exit; 6591 } 6592 mTracks.add(track); 6593 6594 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 6595 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 6596 mAudioFlinger->btNrecIsOff(); 6597 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 6598 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 6599 6600 if ((*flags & AUDIO_INPUT_FLAG_FAST) && (tid != -1)) { 6601 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 6602 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 6603 // so ask activity manager to do this on our behalf 6604 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 6605 } 6606 } 6607 6608 lStatus = NO_ERROR; 6609 6610Exit: 6611 *status = lStatus; 6612 return track; 6613} 6614 6615status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, 6616 AudioSystem::sync_event_t event, 6617 audio_session_t triggerSession) 6618{ 6619 ALOGV("RecordThread::start event %d, triggerSession %d", event, triggerSession); 6620 sp<ThreadBase> strongMe = this; 6621 status_t status = NO_ERROR; 6622 6623 if (event == AudioSystem::SYNC_EVENT_NONE) { 6624 recordTrack->clearSyncStartEvent(); 6625 } else if (event != AudioSystem::SYNC_EVENT_SAME) { 6626 recordTrack->mSyncStartEvent = mAudioFlinger->createSyncEvent(event, 6627 triggerSession, 6628 recordTrack->sessionId(), 6629 syncStartEventCallback, 6630 recordTrack); 6631 // Sync event can be cancelled by the trigger session if the track is not in a 6632 // compatible state in which case we start record immediately 6633 if (recordTrack->mSyncStartEvent->isCancelled()) { 6634 recordTrack->clearSyncStartEvent(); 6635 } else { 6636 // do not wait for the event for more than AudioSystem::kSyncRecordStartTimeOutMs 6637 recordTrack->mFramesToDrop = - 6638 ((AudioSystem::kSyncRecordStartTimeOutMs * recordTrack->mSampleRate) / 1000); 6639 } 6640 } 6641 6642 { 6643 // This section is a rendezvous between binder thread executing start() and RecordThread 6644 AutoMutex lock(mLock); 6645 if (mActiveTracks.indexOf(recordTrack) >= 0) { 6646 if (recordTrack->mState == TrackBase::PAUSING) { 6647 ALOGV("active record track PAUSING -> ACTIVE"); 6648 recordTrack->mState = TrackBase::ACTIVE; 6649 } else { 6650 ALOGV("active record track state %d", recordTrack->mState); 6651 } 6652 return status; 6653 } 6654 6655 // TODO consider other ways of handling this, such as changing the state to :STARTING and 6656 // adding the track to mActiveTracks after returning from AudioSystem::startInput(), 6657 // or using a separate command thread 6658 recordTrack->mState = TrackBase::STARTING_1; 6659 mActiveTracks.add(recordTrack); 6660 mActiveTracksGen++; 6661 status_t status = NO_ERROR; 6662 if (recordTrack->isExternalTrack()) { 6663 mLock.unlock(); 6664 status = AudioSystem::startInput(mId, recordTrack->sessionId()); 6665 mLock.lock(); 6666 // FIXME should verify that recordTrack is still in mActiveTracks 6667 if (status != NO_ERROR) { 6668 mActiveTracks.remove(recordTrack); 6669 mActiveTracksGen++; 6670 recordTrack->clearSyncStartEvent(); 6671 ALOGV("RecordThread::start error %d", status); 6672 return status; 6673 } 6674 } 6675 // Catch up with current buffer indices if thread is already running. 6676 // This is what makes a new client discard all buffered data. If the track's mRsmpInFront 6677 // was initialized to some value closer to the thread's mRsmpInFront, then the track could 6678 // see previously buffered data before it called start(), but with greater risk of overrun. 6679 6680 recordTrack->mResamplerBufferProvider->reset(); 6681 // clear any converter state as new data will be discontinuous 6682 recordTrack->mRecordBufferConverter->reset(); 6683 recordTrack->mState = TrackBase::STARTING_2; 6684 // signal thread to start 6685 mWaitWorkCV.broadcast(); 6686 if (mActiveTracks.indexOf(recordTrack) < 0) { 6687 ALOGV("Record failed to start"); 6688 status = BAD_VALUE; 6689 goto startError; 6690 } 6691 return status; 6692 } 6693 6694startError: 6695 if (recordTrack->isExternalTrack()) { 6696 AudioSystem::stopInput(mId, recordTrack->sessionId()); 6697 } 6698 recordTrack->clearSyncStartEvent(); 6699 // FIXME I wonder why we do not reset the state here? 6700 return status; 6701} 6702 6703void AudioFlinger::RecordThread::syncStartEventCallback(const wp<SyncEvent>& event) 6704{ 6705 sp<SyncEvent> strongEvent = event.promote(); 6706 6707 if (strongEvent != 0) { 6708 sp<RefBase> ptr = strongEvent->cookie().promote(); 6709 if (ptr != 0) { 6710 RecordTrack *recordTrack = (RecordTrack *)ptr.get(); 6711 recordTrack->handleSyncStartEvent(strongEvent); 6712 } 6713 } 6714} 6715 6716bool AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 6717 ALOGV("RecordThread::stop"); 6718 AutoMutex _l(mLock); 6719 if (mActiveTracks.indexOf(recordTrack) != 0 || recordTrack->mState == TrackBase::PAUSING) { 6720 return false; 6721 } 6722 // note that threadLoop may still be processing the track at this point [without lock] 6723 recordTrack->mState = TrackBase::PAUSING; 6724 // signal thread to stop 6725 mWaitWorkCV.broadcast(); 6726 // do not wait for mStartStopCond if exiting 6727 if (exitPending()) { 6728 return true; 6729 } 6730 // FIXME incorrect usage of wait: no explicit predicate or loop 6731 mStartStopCond.wait(mLock); 6732 // if we have been restarted, recordTrack is in mActiveTracks here 6733 if (exitPending() || mActiveTracks.indexOf(recordTrack) != 0) { 6734 ALOGV("Record stopped OK"); 6735 return true; 6736 } 6737 return false; 6738} 6739 6740bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event __unused) const 6741{ 6742 return false; 6743} 6744 6745status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event __unused) 6746{ 6747#if 0 // This branch is currently dead code, but is preserved in case it will be needed in future 6748 if (!isValidSyncEvent(event)) { 6749 return BAD_VALUE; 6750 } 6751 6752 audio_session_t eventSession = event->triggerSession(); 6753 status_t ret = NAME_NOT_FOUND; 6754 6755 Mutex::Autolock _l(mLock); 6756 6757 for (size_t i = 0; i < mTracks.size(); i++) { 6758 sp<RecordTrack> track = mTracks[i]; 6759 if (eventSession == track->sessionId()) { 6760 (void) track->setSyncEvent(event); 6761 ret = NO_ERROR; 6762 } 6763 } 6764 return ret; 6765#else 6766 return BAD_VALUE; 6767#endif 6768} 6769 6770// destroyTrack_l() must be called with ThreadBase::mLock held 6771void AudioFlinger::RecordThread::destroyTrack_l(const sp<RecordTrack>& track) 6772{ 6773 track->terminate(); 6774 track->mState = TrackBase::STOPPED; 6775 // active tracks are removed by threadLoop() 6776 if (mActiveTracks.indexOf(track) < 0) { 6777 removeTrack_l(track); 6778 } 6779} 6780 6781void AudioFlinger::RecordThread::removeTrack_l(const sp<RecordTrack>& track) 6782{ 6783 mTracks.remove(track); 6784 // need anything related to effects here? 6785 if (track->isFastTrack()) { 6786 ALOG_ASSERT(!mFastTrackAvail); 6787 mFastTrackAvail = true; 6788 } 6789} 6790 6791void AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 6792{ 6793 dumpInternals(fd, args); 6794 dumpTracks(fd, args); 6795 dumpEffectChains(fd, args); 6796} 6797 6798void AudioFlinger::RecordThread::dumpInternals(int fd, const Vector<String16>& args) 6799{ 6800 dprintf(fd, "\nInput thread %p:\n", this); 6801 6802 dumpBase(fd, args); 6803 6804 if (mActiveTracks.size() == 0) { 6805 dprintf(fd, " No active record clients\n"); 6806 } 6807 dprintf(fd, " Fast capture thread: %s\n", hasFastCapture() ? "yes" : "no"); 6808 dprintf(fd, " Fast track available: %s\n", mFastTrackAvail ? "yes" : "no"); 6809 6810 // Make a non-atomic copy of fast capture dump state so it won't change underneath us 6811 // while we are dumping it. It may be inconsistent, but it won't mutate! 6812 // This is a large object so we place it on the heap. 6813 // FIXME 25972958: Need an intelligent copy constructor that does not touch unused pages. 6814 const FastCaptureDumpState *copy = new FastCaptureDumpState(mFastCaptureDumpState); 6815 copy->dump(fd); 6816 delete copy; 6817} 6818 6819void AudioFlinger::RecordThread::dumpTracks(int fd, const Vector<String16>& args __unused) 6820{ 6821 const size_t SIZE = 256; 6822 char buffer[SIZE]; 6823 String8 result; 6824 6825 size_t numtracks = mTracks.size(); 6826 size_t numactive = mActiveTracks.size(); 6827 size_t numactiveseen = 0; 6828 dprintf(fd, " %zu Tracks", numtracks); 6829 if (numtracks) { 6830 dprintf(fd, " of which %zu are active\n", numactive); 6831 RecordTrack::appendDumpHeader(result); 6832 for (size_t i = 0; i < numtracks ; ++i) { 6833 sp<RecordTrack> track = mTracks[i]; 6834 if (track != 0) { 6835 bool active = mActiveTracks.indexOf(track) >= 0; 6836 if (active) { 6837 numactiveseen++; 6838 } 6839 track->dump(buffer, SIZE, active); 6840 result.append(buffer); 6841 } 6842 } 6843 } else { 6844 dprintf(fd, "\n"); 6845 } 6846 6847 if (numactiveseen != numactive) { 6848 snprintf(buffer, SIZE, " The following tracks are in the active list but" 6849 " not in the track list\n"); 6850 result.append(buffer); 6851 RecordTrack::appendDumpHeader(result); 6852 for (size_t i = 0; i < numactive; ++i) { 6853 sp<RecordTrack> track = mActiveTracks[i]; 6854 if (mTracks.indexOf(track) < 0) { 6855 track->dump(buffer, SIZE, true); 6856 result.append(buffer); 6857 } 6858 } 6859 6860 } 6861 write(fd, result.string(), result.size()); 6862} 6863 6864 6865void AudioFlinger::RecordThread::ResamplerBufferProvider::reset() 6866{ 6867 sp<ThreadBase> threadBase = mRecordTrack->mThread.promote(); 6868 RecordThread *recordThread = (RecordThread *) threadBase.get(); 6869 mRsmpInFront = recordThread->mRsmpInRear; 6870 mRsmpInUnrel = 0; 6871} 6872 6873void AudioFlinger::RecordThread::ResamplerBufferProvider::sync( 6874 size_t *framesAvailable, bool *hasOverrun) 6875{ 6876 sp<ThreadBase> threadBase = mRecordTrack->mThread.promote(); 6877 RecordThread *recordThread = (RecordThread *) threadBase.get(); 6878 const int32_t rear = recordThread->mRsmpInRear; 6879 const int32_t front = mRsmpInFront; 6880 const ssize_t filled = rear - front; 6881 6882 size_t framesIn; 6883 bool overrun = false; 6884 if (filled < 0) { 6885 // should not happen, but treat like a massive overrun and re-sync 6886 framesIn = 0; 6887 mRsmpInFront = rear; 6888 overrun = true; 6889 } else if ((size_t) filled <= recordThread->mRsmpInFrames) { 6890 framesIn = (size_t) filled; 6891 } else { 6892 // client is not keeping up with server, but give it latest data 6893 framesIn = recordThread->mRsmpInFrames; 6894 mRsmpInFront = /* front = */ rear - framesIn; 6895 overrun = true; 6896 } 6897 if (framesAvailable != NULL) { 6898 *framesAvailable = framesIn; 6899 } 6900 if (hasOverrun != NULL) { 6901 *hasOverrun = overrun; 6902 } 6903} 6904 6905// AudioBufferProvider interface 6906status_t AudioFlinger::RecordThread::ResamplerBufferProvider::getNextBuffer( 6907 AudioBufferProvider::Buffer* buffer) 6908{ 6909 sp<ThreadBase> threadBase = mRecordTrack->mThread.promote(); 6910 if (threadBase == 0) { 6911 buffer->frameCount = 0; 6912 buffer->raw = NULL; 6913 return NOT_ENOUGH_DATA; 6914 } 6915 RecordThread *recordThread = (RecordThread *) threadBase.get(); 6916 int32_t rear = recordThread->mRsmpInRear; 6917 int32_t front = mRsmpInFront; 6918 ssize_t filled = rear - front; 6919 // FIXME should not be P2 (don't want to increase latency) 6920 // FIXME if client not keeping up, discard 6921 LOG_ALWAYS_FATAL_IF(!(0 <= filled && (size_t) filled <= recordThread->mRsmpInFrames)); 6922 // 'filled' may be non-contiguous, so return only the first contiguous chunk 6923 front &= recordThread->mRsmpInFramesP2 - 1; 6924 size_t part1 = recordThread->mRsmpInFramesP2 - front; 6925 if (part1 > (size_t) filled) { 6926 part1 = filled; 6927 } 6928 size_t ask = buffer->frameCount; 6929 ALOG_ASSERT(ask > 0); 6930 if (part1 > ask) { 6931 part1 = ask; 6932 } 6933 if (part1 == 0) { 6934 // out of data is fine since the resampler will return a short-count. 6935 buffer->raw = NULL; 6936 buffer->frameCount = 0; 6937 mRsmpInUnrel = 0; 6938 return NOT_ENOUGH_DATA; 6939 } 6940 6941 buffer->raw = (uint8_t*)recordThread->mRsmpInBuffer + front * recordThread->mFrameSize; 6942 buffer->frameCount = part1; 6943 mRsmpInUnrel = part1; 6944 return NO_ERROR; 6945} 6946 6947// AudioBufferProvider interface 6948void AudioFlinger::RecordThread::ResamplerBufferProvider::releaseBuffer( 6949 AudioBufferProvider::Buffer* buffer) 6950{ 6951 size_t stepCount = buffer->frameCount; 6952 if (stepCount == 0) { 6953 return; 6954 } 6955 ALOG_ASSERT(stepCount <= mRsmpInUnrel); 6956 mRsmpInUnrel -= stepCount; 6957 mRsmpInFront += stepCount; 6958 buffer->raw = NULL; 6959 buffer->frameCount = 0; 6960} 6961 6962AudioFlinger::RecordThread::RecordBufferConverter::RecordBufferConverter( 6963 audio_channel_mask_t srcChannelMask, audio_format_t srcFormat, 6964 uint32_t srcSampleRate, 6965 audio_channel_mask_t dstChannelMask, audio_format_t dstFormat, 6966 uint32_t dstSampleRate) : 6967 mSrcChannelMask(AUDIO_CHANNEL_INVALID), // updateParameters will set following vars 6968 // mSrcFormat 6969 // mSrcSampleRate 6970 // mDstChannelMask 6971 // mDstFormat 6972 // mDstSampleRate 6973 // mSrcChannelCount 6974 // mDstChannelCount 6975 // mDstFrameSize 6976 mBuf(NULL), mBufFrames(0), mBufFrameSize(0), 6977 mResampler(NULL), 6978 mIsLegacyDownmix(false), 6979 mIsLegacyUpmix(false), 6980 mRequiresFloat(false), 6981 mInputConverterProvider(NULL) 6982{ 6983 (void)updateParameters(srcChannelMask, srcFormat, srcSampleRate, 6984 dstChannelMask, dstFormat, dstSampleRate); 6985} 6986 6987AudioFlinger::RecordThread::RecordBufferConverter::~RecordBufferConverter() { 6988 free(mBuf); 6989 delete mResampler; 6990 delete mInputConverterProvider; 6991} 6992 6993size_t AudioFlinger::RecordThread::RecordBufferConverter::convert(void *dst, 6994 AudioBufferProvider *provider, size_t frames) 6995{ 6996 if (mInputConverterProvider != NULL) { 6997 mInputConverterProvider->setBufferProvider(provider); 6998 provider = mInputConverterProvider; 6999 } 7000 7001 if (mResampler == NULL) { 7002 ALOGVV("NO RESAMPLING sampleRate:%u mSrcFormat:%#x mDstFormat:%#x", 7003 mSrcSampleRate, mSrcFormat, mDstFormat); 7004 7005 AudioBufferProvider::Buffer buffer; 7006 for (size_t i = frames; i > 0; ) { 7007 buffer.frameCount = i; 7008 status_t status = provider->getNextBuffer(&buffer); 7009 if (status != OK || buffer.frameCount == 0) { 7010 frames -= i; // cannot fill request. 7011 break; 7012 } 7013 // format convert to destination buffer 7014 convertNoResampler(dst, buffer.raw, buffer.frameCount); 7015 7016 dst = (int8_t*)dst + buffer.frameCount * mDstFrameSize; 7017 i -= buffer.frameCount; 7018 provider->releaseBuffer(&buffer); 7019 } 7020 } else { 7021 ALOGVV("RESAMPLING mSrcSampleRate:%u mDstSampleRate:%u mSrcFormat:%#x mDstFormat:%#x", 7022 mSrcSampleRate, mDstSampleRate, mSrcFormat, mDstFormat); 7023 7024 // reallocate buffer if needed 7025 if (mBufFrameSize != 0 && mBufFrames < frames) { 7026 free(mBuf); 7027 mBufFrames = frames; 7028 (void)posix_memalign(&mBuf, 32, mBufFrames * mBufFrameSize); 7029 } 7030 // resampler accumulates, but we only have one source track 7031 memset(mBuf, 0, frames * mBufFrameSize); 7032 frames = mResampler->resample((int32_t*)mBuf, frames, provider); 7033 // format convert to destination buffer 7034 convertResampler(dst, mBuf, frames); 7035 } 7036 return frames; 7037} 7038 7039status_t AudioFlinger::RecordThread::RecordBufferConverter::updateParameters( 7040 audio_channel_mask_t srcChannelMask, audio_format_t srcFormat, 7041 uint32_t srcSampleRate, 7042 audio_channel_mask_t dstChannelMask, audio_format_t dstFormat, 7043 uint32_t dstSampleRate) 7044{ 7045 // quick evaluation if there is any change. 7046 if (mSrcFormat == srcFormat 7047 && mSrcChannelMask == srcChannelMask 7048 && mSrcSampleRate == srcSampleRate 7049 && mDstFormat == dstFormat 7050 && mDstChannelMask == dstChannelMask 7051 && mDstSampleRate == dstSampleRate) { 7052 return NO_ERROR; 7053 } 7054 7055 ALOGV("RecordBufferConverter updateParameters srcMask:%#x dstMask:%#x" 7056 " srcFormat:%#x dstFormat:%#x srcRate:%u dstRate:%u", 7057 srcChannelMask, dstChannelMask, srcFormat, dstFormat, srcSampleRate, dstSampleRate); 7058 const bool valid = 7059 audio_is_input_channel(srcChannelMask) 7060 && audio_is_input_channel(dstChannelMask) 7061 && audio_is_valid_format(srcFormat) && audio_is_linear_pcm(srcFormat) 7062 && audio_is_valid_format(dstFormat) && audio_is_linear_pcm(dstFormat) 7063 && (srcSampleRate <= dstSampleRate * AUDIO_RESAMPLER_DOWN_RATIO_MAX) 7064 ; // no upsampling checks for now 7065 if (!valid) { 7066 return BAD_VALUE; 7067 } 7068 7069 mSrcFormat = srcFormat; 7070 mSrcChannelMask = srcChannelMask; 7071 mSrcSampleRate = srcSampleRate; 7072 mDstFormat = dstFormat; 7073 mDstChannelMask = dstChannelMask; 7074 mDstSampleRate = dstSampleRate; 7075 7076 // compute derived parameters 7077 mSrcChannelCount = audio_channel_count_from_in_mask(srcChannelMask); 7078 mDstChannelCount = audio_channel_count_from_in_mask(dstChannelMask); 7079 mDstFrameSize = mDstChannelCount * audio_bytes_per_sample(mDstFormat); 7080 7081 // do we need to resample? 7082 delete mResampler; 7083 mResampler = NULL; 7084 if (mSrcSampleRate != mDstSampleRate) { 7085 mResampler = AudioResampler::create(AUDIO_FORMAT_PCM_FLOAT, 7086 mSrcChannelCount, mDstSampleRate); 7087 mResampler->setSampleRate(mSrcSampleRate); 7088 mResampler->setVolume(AudioMixer::UNITY_GAIN_FLOAT, AudioMixer::UNITY_GAIN_FLOAT); 7089 } 7090 7091 // are we running legacy channel conversion modes? 7092 mIsLegacyDownmix = (mSrcChannelMask == AUDIO_CHANNEL_IN_STEREO 7093 || mSrcChannelMask == AUDIO_CHANNEL_IN_FRONT_BACK) 7094 && mDstChannelMask == AUDIO_CHANNEL_IN_MONO; 7095 mIsLegacyUpmix = mSrcChannelMask == AUDIO_CHANNEL_IN_MONO 7096 && (mDstChannelMask == AUDIO_CHANNEL_IN_STEREO 7097 || mDstChannelMask == AUDIO_CHANNEL_IN_FRONT_BACK); 7098 7099 // do we need to process in float? 7100 mRequiresFloat = mResampler != NULL || mIsLegacyDownmix || mIsLegacyUpmix; 7101 7102 // do we need a staging buffer to convert for destination (we can still optimize this)? 7103 // we use mBufFrameSize > 0 to indicate both frame size as well as buffer necessity 7104 if (mResampler != NULL) { 7105 mBufFrameSize = max(mSrcChannelCount, FCC_2) 7106 * audio_bytes_per_sample(AUDIO_FORMAT_PCM_FLOAT); 7107 } else if (mIsLegacyUpmix || mIsLegacyDownmix) { // legacy modes always float 7108 mBufFrameSize = mDstChannelCount * audio_bytes_per_sample(AUDIO_FORMAT_PCM_FLOAT); 7109 } else if (mSrcChannelMask != mDstChannelMask && mDstFormat != mSrcFormat) { 7110 mBufFrameSize = mDstChannelCount * audio_bytes_per_sample(mSrcFormat); 7111 } else { 7112 mBufFrameSize = 0; 7113 } 7114 mBufFrames = 0; // force the buffer to be resized. 7115 7116 // do we need an input converter buffer provider to give us float? 7117 delete mInputConverterProvider; 7118 mInputConverterProvider = NULL; 7119 if (mRequiresFloat && mSrcFormat != AUDIO_FORMAT_PCM_FLOAT) { 7120 mInputConverterProvider = new ReformatBufferProvider( 7121 audio_channel_count_from_in_mask(mSrcChannelMask), 7122 mSrcFormat, 7123 AUDIO_FORMAT_PCM_FLOAT, 7124 256 /* provider buffer frame count */); 7125 } 7126 7127 // do we need a remixer to do channel mask conversion 7128 if (!mIsLegacyDownmix && !mIsLegacyUpmix && mSrcChannelMask != mDstChannelMask) { 7129 (void) memcpy_by_index_array_initialization_from_channel_mask( 7130 mIdxAry, ARRAY_SIZE(mIdxAry), mDstChannelMask, mSrcChannelMask); 7131 } 7132 return NO_ERROR; 7133} 7134 7135void AudioFlinger::RecordThread::RecordBufferConverter::convertNoResampler( 7136 void *dst, const void *src, size_t frames) 7137{ 7138 // src is native type unless there is legacy upmix or downmix, whereupon it is float. 7139 if (mBufFrameSize != 0 && mBufFrames < frames) { 7140 free(mBuf); 7141 mBufFrames = frames; 7142 (void)posix_memalign(&mBuf, 32, mBufFrames * mBufFrameSize); 7143 } 7144 // do we need to do legacy upmix and downmix? 7145 if (mIsLegacyUpmix || mIsLegacyDownmix) { 7146 void *dstBuf = mBuf != NULL ? mBuf : dst; 7147 if (mIsLegacyUpmix) { 7148 upmix_to_stereo_float_from_mono_float((float *)dstBuf, 7149 (const float *)src, frames); 7150 } else /*mIsLegacyDownmix */ { 7151 downmix_to_mono_float_from_stereo_float((float *)dstBuf, 7152 (const float *)src, frames); 7153 } 7154 if (mBuf != NULL) { 7155 memcpy_by_audio_format(dst, mDstFormat, mBuf, AUDIO_FORMAT_PCM_FLOAT, 7156 frames * mDstChannelCount); 7157 } 7158 return; 7159 } 7160 // do we need to do channel mask conversion? 7161 if (mSrcChannelMask != mDstChannelMask) { 7162 void *dstBuf = mBuf != NULL ? mBuf : dst; 7163 memcpy_by_index_array(dstBuf, mDstChannelCount, 7164 src, mSrcChannelCount, mIdxAry, audio_bytes_per_sample(mSrcFormat), frames); 7165 if (dstBuf == dst) { 7166 return; // format is the same 7167 } 7168 } 7169 // convert to destination buffer 7170 const void *convertBuf = mBuf != NULL ? mBuf : src; 7171 memcpy_by_audio_format(dst, mDstFormat, convertBuf, mSrcFormat, 7172 frames * mDstChannelCount); 7173} 7174 7175void AudioFlinger::RecordThread::RecordBufferConverter::convertResampler( 7176 void *dst, /*not-a-const*/ void *src, size_t frames) 7177{ 7178 // src buffer format is ALWAYS float when entering this routine 7179 if (mIsLegacyUpmix) { 7180 ; // mono to stereo already handled by resampler 7181 } else if (mIsLegacyDownmix 7182 || (mSrcChannelMask == mDstChannelMask && mSrcChannelCount == 1)) { 7183 // the resampler outputs stereo for mono input channel (a feature?) 7184 // must convert to mono 7185 downmix_to_mono_float_from_stereo_float((float *)src, 7186 (const float *)src, frames); 7187 } else if (mSrcChannelMask != mDstChannelMask) { 7188 // convert to mono channel again for channel mask conversion (could be skipped 7189 // with further optimization). 7190 if (mSrcChannelCount == 1) { 7191 downmix_to_mono_float_from_stereo_float((float *)src, 7192 (const float *)src, frames); 7193 } 7194 // convert to destination format (in place, OK as float is larger than other types) 7195 if (mDstFormat != AUDIO_FORMAT_PCM_FLOAT) { 7196 memcpy_by_audio_format(src, mDstFormat, src, AUDIO_FORMAT_PCM_FLOAT, 7197 frames * mSrcChannelCount); 7198 } 7199 // channel convert and save to dst 7200 memcpy_by_index_array(dst, mDstChannelCount, 7201 src, mSrcChannelCount, mIdxAry, audio_bytes_per_sample(mDstFormat), frames); 7202 return; 7203 } 7204 // convert to destination format and save to dst 7205 memcpy_by_audio_format(dst, mDstFormat, src, AUDIO_FORMAT_PCM_FLOAT, 7206 frames * mDstChannelCount); 7207} 7208 7209bool AudioFlinger::RecordThread::checkForNewParameter_l(const String8& keyValuePair, 7210 status_t& status) 7211{ 7212 bool reconfig = false; 7213 7214 status = NO_ERROR; 7215 7216 audio_format_t reqFormat = mFormat; 7217 uint32_t samplingRate = mSampleRate; 7218 // TODO this may change if we want to support capture from HDMI PCM multi channel (e.g on TVs). 7219 audio_channel_mask_t channelMask = audio_channel_in_mask_from_count(mChannelCount); 7220 7221 AudioParameter param = AudioParameter(keyValuePair); 7222 int value; 7223 7224 // scope for AutoPark extends to end of method 7225 AutoPark<FastCapture> park(mFastCapture); 7226 7227 // TODO Investigate when this code runs. Check with audio policy when a sample rate and 7228 // channel count change can be requested. Do we mandate the first client defines the 7229 // HAL sampling rate and channel count or do we allow changes on the fly? 7230 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 7231 samplingRate = value; 7232 reconfig = true; 7233 } 7234 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 7235 if (!audio_is_linear_pcm((audio_format_t) value)) { 7236 status = BAD_VALUE; 7237 } else { 7238 reqFormat = (audio_format_t) value; 7239 reconfig = true; 7240 } 7241 } 7242 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 7243 audio_channel_mask_t mask = (audio_channel_mask_t) value; 7244 if (!audio_is_input_channel(mask) || 7245 audio_channel_count_from_in_mask(mask) > FCC_8) { 7246 status = BAD_VALUE; 7247 } else { 7248 channelMask = mask; 7249 reconfig = true; 7250 } 7251 } 7252 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 7253 // do not accept frame count changes if tracks are open as the track buffer 7254 // size depends on frame count and correct behavior would not be guaranteed 7255 // if frame count is changed after track creation 7256 if (mActiveTracks.size() > 0) { 7257 status = INVALID_OPERATION; 7258 } else { 7259 reconfig = true; 7260 } 7261 } 7262 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 7263 // forward device change to effects that have requested to be 7264 // aware of attached audio device. 7265 for (size_t i = 0; i < mEffectChains.size(); i++) { 7266 mEffectChains[i]->setDevice_l(value); 7267 } 7268 7269 // store input device and output device but do not forward output device to audio HAL. 7270 // Note that status is ignored by the caller for output device 7271 // (see AudioFlinger::setParameters() 7272 if (audio_is_output_devices(value)) { 7273 mOutDevice = value; 7274 status = BAD_VALUE; 7275 } else { 7276 mInDevice = value; 7277 if (value != AUDIO_DEVICE_NONE) { 7278 mPrevInDevice = value; 7279 } 7280 // disable AEC and NS if the device is a BT SCO headset supporting those 7281 // pre processings 7282 if (mTracks.size() > 0) { 7283 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 7284 mAudioFlinger->btNrecIsOff(); 7285 for (size_t i = 0; i < mTracks.size(); i++) { 7286 sp<RecordTrack> track = mTracks[i]; 7287 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId()); 7288 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId()); 7289 } 7290 } 7291 } 7292 } 7293 if (param.getInt(String8(AudioParameter::keyInputSource), value) == NO_ERROR && 7294 mAudioSource != (audio_source_t)value) { 7295 // forward device change to effects that have requested to be 7296 // aware of attached audio device. 7297 for (size_t i = 0; i < mEffectChains.size(); i++) { 7298 mEffectChains[i]->setAudioSource_l((audio_source_t)value); 7299 } 7300 mAudioSource = (audio_source_t)value; 7301 } 7302 7303 if (status == NO_ERROR) { 7304 status = mInput->stream->common.set_parameters(&mInput->stream->common, 7305 keyValuePair.string()); 7306 if (status == INVALID_OPERATION) { 7307 inputStandBy(); 7308 status = mInput->stream->common.set_parameters(&mInput->stream->common, 7309 keyValuePair.string()); 7310 } 7311 if (reconfig) { 7312 if (status == BAD_VALUE && 7313 audio_is_linear_pcm(mInput->stream->common.get_format(&mInput->stream->common)) && 7314 audio_is_linear_pcm(reqFormat) && 7315 (mInput->stream->common.get_sample_rate(&mInput->stream->common) 7316 <= (AUDIO_RESAMPLER_DOWN_RATIO_MAX * samplingRate)) && 7317 audio_channel_count_from_in_mask( 7318 mInput->stream->common.get_channels(&mInput->stream->common)) <= FCC_8) { 7319 status = NO_ERROR; 7320 } 7321 if (status == NO_ERROR) { 7322 readInputParameters_l(); 7323 sendIoConfigEvent_l(AUDIO_INPUT_CONFIG_CHANGED); 7324 } 7325 } 7326 } 7327 7328 return reconfig; 7329} 7330 7331String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 7332{ 7333 Mutex::Autolock _l(mLock); 7334 if (initCheck() != NO_ERROR) { 7335 return String8(); 7336 } 7337 7338 char *s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 7339 const String8 out_s8(s); 7340 free(s); 7341 return out_s8; 7342} 7343 7344void AudioFlinger::RecordThread::ioConfigChanged(audio_io_config_event event, pid_t pid) { 7345 sp<AudioIoDescriptor> desc = new AudioIoDescriptor(); 7346 7347 desc->mIoHandle = mId; 7348 7349 switch (event) { 7350 case AUDIO_INPUT_OPENED: 7351 case AUDIO_INPUT_CONFIG_CHANGED: 7352 desc->mPatch = mPatch; 7353 desc->mChannelMask = mChannelMask; 7354 desc->mSamplingRate = mSampleRate; 7355 desc->mFormat = mFormat; 7356 desc->mFrameCount = mFrameCount; 7357 desc->mFrameCountHAL = mFrameCount; 7358 desc->mLatency = 0; 7359 break; 7360 7361 case AUDIO_INPUT_CLOSED: 7362 default: 7363 break; 7364 } 7365 mAudioFlinger->ioConfigChanged(event, desc, pid); 7366} 7367 7368void AudioFlinger::RecordThread::readInputParameters_l() 7369{ 7370 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 7371 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 7372 mChannelCount = audio_channel_count_from_in_mask(mChannelMask); 7373 if (mChannelCount > FCC_8) { 7374 ALOGE("HAL channel count %d > %d", mChannelCount, FCC_8); 7375 } 7376 mHALFormat = mInput->stream->common.get_format(&mInput->stream->common); 7377 mFormat = mHALFormat; 7378 if (!audio_is_linear_pcm(mFormat)) { 7379 ALOGE("HAL format %#x is not linear pcm", mFormat); 7380 } 7381 mFrameSize = audio_stream_in_frame_size(mInput->stream); 7382 mBufferSize = mInput->stream->common.get_buffer_size(&mInput->stream->common); 7383 mFrameCount = mBufferSize / mFrameSize; 7384 // This is the formula for calculating the temporary buffer size. 7385 // With 7 HAL buffers, we can guarantee ability to down-sample the input by ratio of 6:1 to 7386 // 1 full output buffer, regardless of the alignment of the available input. 7387 // The value is somewhat arbitrary, and could probably be even larger. 7388 // A larger value should allow more old data to be read after a track calls start(), 7389 // without increasing latency. 7390 // 7391 // Note this is independent of the maximum downsampling ratio permitted for capture. 7392 mRsmpInFrames = mFrameCount * 7; 7393 mRsmpInFramesP2 = roundup(mRsmpInFrames); 7394 free(mRsmpInBuffer); 7395 mRsmpInBuffer = NULL; 7396 7397 // TODO optimize audio capture buffer sizes ... 7398 // Here we calculate the size of the sliding buffer used as a source 7399 // for resampling. mRsmpInFramesP2 is currently roundup(mFrameCount * 7). 7400 // For current HAL frame counts, this is usually 2048 = 40 ms. It would 7401 // be better to have it derived from the pipe depth in the long term. 7402 // The current value is higher than necessary. However it should not add to latency. 7403 7404 // Over-allocate beyond mRsmpInFramesP2 to permit a HAL read past end of buffer 7405 size_t bufferSize = (mRsmpInFramesP2 + mFrameCount - 1) * mFrameSize; 7406 (void)posix_memalign(&mRsmpInBuffer, 32, bufferSize); 7407 memset(mRsmpInBuffer, 0, bufferSize); // if posix_memalign fails, will segv here. 7408 7409 // AudioRecord mSampleRate and mChannelCount are constant due to AudioRecord API constraints. 7410 // But if thread's mSampleRate or mChannelCount changes, how will that affect active tracks? 7411} 7412 7413uint32_t AudioFlinger::RecordThread::getInputFramesLost() 7414{ 7415 Mutex::Autolock _l(mLock); 7416 if (initCheck() != NO_ERROR) { 7417 return 0; 7418 } 7419 7420 return mInput->stream->get_input_frames_lost(mInput->stream); 7421} 7422 7423// hasAudioSession_l() must be called with ThreadBase::mLock held 7424uint32_t AudioFlinger::RecordThread::hasAudioSession_l(audio_session_t sessionId) const 7425{ 7426 uint32_t result = 0; 7427 if (getEffectChain_l(sessionId) != 0) { 7428 result = EFFECT_SESSION; 7429 } 7430 7431 for (size_t i = 0; i < mTracks.size(); ++i) { 7432 if (sessionId == mTracks[i]->sessionId()) { 7433 result |= TRACK_SESSION; 7434 if (mTracks[i]->isFastTrack()) { 7435 result |= FAST_SESSION; 7436 } 7437 break; 7438 } 7439 } 7440 7441 return result; 7442} 7443 7444KeyedVector<audio_session_t, bool> AudioFlinger::RecordThread::sessionIds() const 7445{ 7446 KeyedVector<audio_session_t, bool> ids; 7447 Mutex::Autolock _l(mLock); 7448 for (size_t j = 0; j < mTracks.size(); ++j) { 7449 sp<RecordThread::RecordTrack> track = mTracks[j]; 7450 audio_session_t sessionId = track->sessionId(); 7451 if (ids.indexOfKey(sessionId) < 0) { 7452 ids.add(sessionId, true); 7453 } 7454 } 7455 return ids; 7456} 7457 7458AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 7459{ 7460 Mutex::Autolock _l(mLock); 7461 AudioStreamIn *input = mInput; 7462 mInput = NULL; 7463 return input; 7464} 7465 7466// this method must always be called either with ThreadBase mLock held or inside the thread loop 7467audio_stream_t* AudioFlinger::RecordThread::stream() const 7468{ 7469 if (mInput == NULL) { 7470 return NULL; 7471 } 7472 return &mInput->stream->common; 7473} 7474 7475status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 7476{ 7477 // only one chain per input thread 7478 if (mEffectChains.size() != 0) { 7479 ALOGW("addEffectChain_l() already one chain %p on thread %p", chain.get(), this); 7480 return INVALID_OPERATION; 7481 } 7482 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 7483 chain->setThread(this); 7484 chain->setInBuffer(NULL); 7485 chain->setOutBuffer(NULL); 7486 7487 checkSuspendOnAddEffectChain_l(chain); 7488 7489 // make sure enabled pre processing effects state is communicated to the HAL as we 7490 // just moved them to a new input stream. 7491 chain->syncHalEffectsState(); 7492 7493 mEffectChains.add(chain); 7494 7495 return NO_ERROR; 7496} 7497 7498size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 7499{ 7500 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 7501 ALOGW_IF(mEffectChains.size() != 1, 7502 "removeEffectChain_l() %p invalid chain size %zu on thread %p", 7503 chain.get(), mEffectChains.size(), this); 7504 if (mEffectChains.size() == 1) { 7505 mEffectChains.removeAt(0); 7506 } 7507 return 0; 7508} 7509 7510status_t AudioFlinger::RecordThread::createAudioPatch_l(const struct audio_patch *patch, 7511 audio_patch_handle_t *handle) 7512{ 7513 status_t status = NO_ERROR; 7514 7515 // store new device and send to effects 7516 mInDevice = patch->sources[0].ext.device.type; 7517 mPatch = *patch; 7518 for (size_t i = 0; i < mEffectChains.size(); i++) { 7519 mEffectChains[i]->setDevice_l(mInDevice); 7520 } 7521 7522 // disable AEC and NS if the device is a BT SCO headset supporting those 7523 // pre processings 7524 if (mTracks.size() > 0) { 7525 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 7526 mAudioFlinger->btNrecIsOff(); 7527 for (size_t i = 0; i < mTracks.size(); i++) { 7528 sp<RecordTrack> track = mTracks[i]; 7529 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId()); 7530 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId()); 7531 } 7532 } 7533 7534 // store new source and send to effects 7535 if (mAudioSource != patch->sinks[0].ext.mix.usecase.source) { 7536 mAudioSource = patch->sinks[0].ext.mix.usecase.source; 7537 for (size_t i = 0; i < mEffectChains.size(); i++) { 7538 mEffectChains[i]->setAudioSource_l(mAudioSource); 7539 } 7540 } 7541 7542 if (mInput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 7543 audio_hw_device_t *hwDevice = mInput->audioHwDev->hwDevice(); 7544 status = hwDevice->create_audio_patch(hwDevice, 7545 patch->num_sources, 7546 patch->sources, 7547 patch->num_sinks, 7548 patch->sinks, 7549 handle); 7550 } else { 7551 char *address; 7552 if (strcmp(patch->sources[0].ext.device.address, "") != 0) { 7553 address = audio_device_address_to_parameter( 7554 patch->sources[0].ext.device.type, 7555 patch->sources[0].ext.device.address); 7556 } else { 7557 address = (char *)calloc(1, 1); 7558 } 7559 AudioParameter param = AudioParameter(String8(address)); 7560 free(address); 7561 param.addInt(String8(AUDIO_PARAMETER_STREAM_ROUTING), 7562 (int)patch->sources[0].ext.device.type); 7563 param.addInt(String8(AUDIO_PARAMETER_STREAM_INPUT_SOURCE), 7564 (int)patch->sinks[0].ext.mix.usecase.source); 7565 status = mInput->stream->common.set_parameters(&mInput->stream->common, 7566 param.toString().string()); 7567 *handle = AUDIO_PATCH_HANDLE_NONE; 7568 } 7569 7570 if (mInDevice != mPrevInDevice) { 7571 sendIoConfigEvent_l(AUDIO_INPUT_CONFIG_CHANGED); 7572 mPrevInDevice = mInDevice; 7573 } 7574 7575 return status; 7576} 7577 7578status_t AudioFlinger::RecordThread::releaseAudioPatch_l(const audio_patch_handle_t handle) 7579{ 7580 status_t status = NO_ERROR; 7581 7582 mInDevice = AUDIO_DEVICE_NONE; 7583 7584 if (mInput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 7585 audio_hw_device_t *hwDevice = mInput->audioHwDev->hwDevice(); 7586 status = hwDevice->release_audio_patch(hwDevice, handle); 7587 } else { 7588 AudioParameter param; 7589 param.addInt(String8(AUDIO_PARAMETER_STREAM_ROUTING), 0); 7590 status = mInput->stream->common.set_parameters(&mInput->stream->common, 7591 param.toString().string()); 7592 } 7593 return status; 7594} 7595 7596void AudioFlinger::RecordThread::addPatchRecord(const sp<PatchRecord>& record) 7597{ 7598 Mutex::Autolock _l(mLock); 7599 mTracks.add(record); 7600} 7601 7602void AudioFlinger::RecordThread::deletePatchRecord(const sp<PatchRecord>& record) 7603{ 7604 Mutex::Autolock _l(mLock); 7605 destroyTrack_l(record); 7606} 7607 7608void AudioFlinger::RecordThread::getAudioPortConfig(struct audio_port_config *config) 7609{ 7610 ThreadBase::getAudioPortConfig(config); 7611 config->role = AUDIO_PORT_ROLE_SINK; 7612 config->ext.mix.hw_module = mInput->audioHwDev->handle(); 7613 config->ext.mix.usecase.source = mAudioSource; 7614} 7615 7616} // namespace android 7617