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)) 93 template <typename T> 94 static 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 102 103 namespace android { 104 105 // retry counts for buffer fill timeout 106 // 50 * ~20msecs = 1 second 107 static const int8_t kMaxTrackRetries = 50; 108 static 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. 112 static const int8_t kMaxTrackRetriesDirect = 2; 113 114 115 116 // don't warn about blocked writes or record buffer overflows more often than this 117 static const nsecs_t kWarningThrottleNs = seconds(5); 118 119 // RecordThread loop sleep time upon application overrun or audio HAL read error 120 static const int kRecordThreadSleepUs = 5000; 121 122 // maximum time to wait in sendConfigEvent_l() for a status to be received 123 static const nsecs_t kConfigEventTimeoutNs = seconds(2); 124 125 // minimum sleep time for the mixer thread loop when tracks are active but in underrun 126 static const uint32_t kMinThreadSleepTimeUs = 5000; 127 // maximum divider applied to the active sleep time in the mixer thread loop 128 static 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 132 static const uint32_t kMinNormalSinkBufferSizeMs = 20; 133 // maximum normal sink buffer size 134 static 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 138 static const uint32_t kMinNormalCaptureBufferSizeMs = 12; 139 140 // Offloaded output thread standby delay: allows track transition without going to standby 141 static const nsecs_t kOffloadStandbyDelayNs = seconds(1); 142 143 // Direct output thread minimum sleep time in idle or active(underrun) state 144 static const nsecs_t kDirectMinSleepTimeUs = 10000; 145 146 147 // Whether to use fast mixer 148 static 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 165 static 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 172 static const int kPriorityAudioApp = 2; 173 static const int kPriorityFastMixer = 3; 174 static 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. 181 static const int kFastTrackMultiplier = 2; 182 183 // The minimum and maximum allowed values 184 static const int kFastTrackMultiplierMin = 1; 185 static const int kFastTrackMultiplierMax = 2; 186 187 // The actual value to use, which can be specified per-device via property af.fast_track_multiplier. 188 static 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. 194 static const size_t kRecordThreadReadOnlyHeapSize = 0x2000; 195 196 // ---------------------------------------------------------------------------- 197 198 static pthread_once_t sFastTrackMultiplierOnce = PTHREAD_ONCE_INIT; 199 200 static 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 216 static 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 228 struct { 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 // ---------------------------------------------------------------------------- 319 320 class CpuStats { 321 public: 322 CpuStats(); 323 void sample(const String8 &title); 324 #ifdef DEBUG_CPU_USAGE 325 private: 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 }; 335 336 CpuStats::CpuStats() 337 #ifdef DEBUG_CPU_USAGE 338 : mCpuNum(-1), mCpukHz(-1) 339 #endif 340 { 341 } 342 343 void 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 427 const 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 } 444 445 String8 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 } 531 532 String8 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 } 567 568 String8 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 } 609 610 const 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 } 628 629 AudioFlinger::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 } 648 649 AudioFlinger::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 } 661 662 status_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 } 672 673 void 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 } 696 697 status_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(). 707 status_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 } 733 734 void 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 741 void 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 } 746 747 void 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 754 void 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 761 status_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 } 778 779 status_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 } 793 794 status_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() 804 void 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 } 861 862 String8 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 } 922 923 void 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 } 966 967 void 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 } 984 985 void AudioFlinger::ThreadBase::acquireWakeLock(int uid) 986 { 987 Mutex::Autolock _l(mLock); 988 acquireWakeLock_l(uid); 989 } 990 991 String16 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 1010 void 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 1045 void AudioFlinger::ThreadBase::releaseWakeLock() 1046 { 1047 Mutex::Autolock _l(mLock); 1048 releaseWakeLock_l(); 1049 } 1050 1051 void 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 1069 void AudioFlinger::ThreadBase::updateWakeLockUids(const SortedVector<int> &uids) { 1070 Mutex::Autolock _l(mLock); 1071 updateWakeLockUids_l(uids); 1072 } 1073 1074 void 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 1088 void 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 1107 void AudioFlinger::ThreadBase::clearPowerManager() 1108 { 1109 Mutex::Autolock _l(mLock); 1110 releaseWakeLock_l(); 1111 mPowerManager.clear(); 1112 } 1113 1114 void 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 1123 void 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 1130 void 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 1145 void 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 1169 void 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 1230 void 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 1238 void 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 1260 status_t AudioFlinger::RecordThread::checkEffectCompatibility_l( 1261 const effect_descriptor_t *desc, audio_session_t sessionId) 1262 { 1263 // No global effect sessions on record threads 1264 if (sessionId == AUDIO_SESSION_OUTPUT_MIX || sessionId == AUDIO_SESSION_OUTPUT_STAGE) { 1265 ALOGW("checkEffectCompatibility_l(): global effect %s on record thread %s", 1266 desc->name, mThreadName); 1267 return BAD_VALUE; 1268 } 1269 // only pre processing effects on record thread 1270 if ((desc->flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_PRE_PROC) { 1271 ALOGW("checkEffectCompatibility_l(): non pre processing effect %s on record thread %s", 1272 desc->name, mThreadName); 1273 return BAD_VALUE; 1274 } 1275 1276 // always allow effects without processing load or latency 1277 if ((desc->flags & EFFECT_FLAG_NO_PROCESS_MASK) == EFFECT_FLAG_NO_PROCESS) { 1278 return NO_ERROR; 1279 } 1280 1281 audio_input_flags_t flags = mInput->flags; 1282 if (hasFastCapture() || (flags & AUDIO_INPUT_FLAG_FAST)) { 1283 if (flags & AUDIO_INPUT_FLAG_RAW) { 1284 ALOGW("checkEffectCompatibility_l(): effect %s on record thread %s in raw mode", 1285 desc->name, mThreadName); 1286 return BAD_VALUE; 1287 } 1288 if ((desc->flags & EFFECT_FLAG_HW_ACC_TUNNEL) == 0) { 1289 ALOGW("checkEffectCompatibility_l(): non HW effect %s on record thread %s in fast mode", 1290 desc->name, mThreadName); 1291 return BAD_VALUE; 1292 } 1293 } 1294 return NO_ERROR; 1295 } 1296 1297 // checkEffectCompatibility_l() must be called with ThreadBase::mLock held 1298 status_t AudioFlinger::PlaybackThread::checkEffectCompatibility_l( 1299 const effect_descriptor_t *desc, audio_session_t sessionId) 1300 { 1301 // no preprocessing on playback threads 1302 if ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC) { 1303 ALOGW("checkEffectCompatibility_l(): pre processing effect %s created on playback" 1304 " thread %s", desc->name, mThreadName); 1305 return BAD_VALUE; 1306 } 1307 1308 switch (mType) { 1309 case MIXER: { 1310 // Reject any effect on mixer multichannel sinks. 1311 // TODO: fix both format and multichannel issues with effects. 1312 if (mChannelCount != FCC_2) { 1313 ALOGW("checkEffectCompatibility_l(): effect %s for multichannel(%d) on MIXER" 1314 " thread %s", desc->name, mChannelCount, mThreadName); 1315 return BAD_VALUE; 1316 } 1317 audio_output_flags_t flags = mOutput->flags; 1318 if (hasFastMixer() || (flags & AUDIO_OUTPUT_FLAG_FAST)) { 1319 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 1320 // global effects are applied only to non fast tracks if they are SW 1321 if ((desc->flags & EFFECT_FLAG_HW_ACC_TUNNEL) == 0) { 1322 break; 1323 } 1324 } else if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) { 1325 // only post processing on output stage session 1326 if ((desc->flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_POST_PROC) { 1327 ALOGW("checkEffectCompatibility_l(): non post processing effect %s not allowed" 1328 " on output stage session", desc->name); 1329 return BAD_VALUE; 1330 } 1331 } else { 1332 // no restriction on effects applied on non fast tracks 1333 if ((hasAudioSession_l(sessionId) & ThreadBase::FAST_SESSION) == 0) { 1334 break; 1335 } 1336 } 1337 1338 // always allow effects without processing load or latency 1339 if ((desc->flags & EFFECT_FLAG_NO_PROCESS_MASK) == EFFECT_FLAG_NO_PROCESS) { 1340 break; 1341 } 1342 if (flags & AUDIO_OUTPUT_FLAG_RAW) { 1343 ALOGW("checkEffectCompatibility_l(): effect %s on playback thread in raw mode", 1344 desc->name); 1345 return BAD_VALUE; 1346 } 1347 if ((desc->flags & EFFECT_FLAG_HW_ACC_TUNNEL) == 0) { 1348 ALOGW("checkEffectCompatibility_l(): non HW effect %s on playback thread" 1349 " in fast mode", desc->name); 1350 return BAD_VALUE; 1351 } 1352 } 1353 } break; 1354 case OFFLOAD: 1355 // nothing actionable on offload threads, if the effect: 1356 // - is offloadable: the effect can be created 1357 // - is NOT offloadable: the effect should still be created, but EffectHandle::enable() 1358 // will take care of invalidating the tracks of the thread 1359 break; 1360 case DIRECT: 1361 // Reject any effect on Direct output threads for now, since the format of 1362 // mSinkBuffer is not guaranteed to be compatible with effect processing (PCM 16 stereo). 1363 ALOGW("checkEffectCompatibility_l(): effect %s on DIRECT output thread %s", 1364 desc->name, mThreadName); 1365 return BAD_VALUE; 1366 case DUPLICATING: 1367 // Reject any effect on mixer multichannel sinks. 1368 // TODO: fix both format and multichannel issues with effects. 1369 if (mChannelCount != FCC_2) { 1370 ALOGW("checkEffectCompatibility_l(): effect %s for multichannel(%d)" 1371 " on DUPLICATING thread %s", desc->name, mChannelCount, mThreadName); 1372 return BAD_VALUE; 1373 } 1374 if ((sessionId == AUDIO_SESSION_OUTPUT_STAGE) || (sessionId == AUDIO_SESSION_OUTPUT_MIX)) { 1375 ALOGW("checkEffectCompatibility_l(): global effect %s on DUPLICATING" 1376 " thread %s", desc->name, mThreadName); 1377 return BAD_VALUE; 1378 } 1379 if ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC) { 1380 ALOGW("checkEffectCompatibility_l(): post processing effect %s on" 1381 " DUPLICATING thread %s", desc->name, mThreadName); 1382 return BAD_VALUE; 1383 } 1384 if ((desc->flags & EFFECT_FLAG_HW_ACC_TUNNEL) != 0) { 1385 ALOGW("checkEffectCompatibility_l(): HW tunneled effect %s on" 1386 " DUPLICATING thread %s", desc->name, mThreadName); 1387 return BAD_VALUE; 1388 } 1389 break; 1390 default: 1391 LOG_ALWAYS_FATAL("checkEffectCompatibility_l(): wrong thread type %d", mType); 1392 } 1393 1394 return NO_ERROR; 1395 } 1396 1397 // ThreadBase::createEffect_l() must be called with AudioFlinger::mLock held 1398 sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 1399 const sp<AudioFlinger::Client>& client, 1400 const sp<IEffectClient>& effectClient, 1401 int32_t priority, 1402 audio_session_t sessionId, 1403 effect_descriptor_t *desc, 1404 int *enabled, 1405 status_t *status) 1406 { 1407 sp<EffectModule> effect; 1408 sp<EffectHandle> handle; 1409 status_t lStatus; 1410 sp<EffectChain> chain; 1411 bool chainCreated = false; 1412 bool effectCreated = false; 1413 bool effectRegistered = false; 1414 1415 lStatus = initCheck(); 1416 if (lStatus != NO_ERROR) { 1417 ALOGW("createEffect_l() Audio driver not initialized."); 1418 goto Exit; 1419 } 1420 1421 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 1422 1423 { // scope for mLock 1424 Mutex::Autolock _l(mLock); 1425 1426 lStatus = checkEffectCompatibility_l(desc, sessionId); 1427 if (lStatus != NO_ERROR) { 1428 goto Exit; 1429 } 1430 1431 // check for existing effect chain with the requested audio session 1432 chain = getEffectChain_l(sessionId); 1433 if (chain == 0) { 1434 // create a new chain for this session 1435 ALOGV("createEffect_l() new effect chain for session %d", sessionId); 1436 chain = new EffectChain(this, sessionId); 1437 addEffectChain_l(chain); 1438 chain->setStrategy(getStrategyForSession_l(sessionId)); 1439 chainCreated = true; 1440 } else { 1441 effect = chain->getEffectFromDesc_l(desc); 1442 } 1443 1444 ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get()); 1445 1446 if (effect == 0) { 1447 audio_unique_id_t id = mAudioFlinger->nextUniqueId(AUDIO_UNIQUE_ID_USE_EFFECT); 1448 // Check CPU and memory usage 1449 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 1450 if (lStatus != NO_ERROR) { 1451 goto Exit; 1452 } 1453 effectRegistered = true; 1454 // create a new effect module if none present in the chain 1455 effect = new EffectModule(this, chain, desc, id, sessionId); 1456 lStatus = effect->status(); 1457 if (lStatus != NO_ERROR) { 1458 goto Exit; 1459 } 1460 effect->setOffloaded(mType == OFFLOAD, mId); 1461 1462 lStatus = chain->addEffect_l(effect); 1463 if (lStatus != NO_ERROR) { 1464 goto Exit; 1465 } 1466 effectCreated = true; 1467 1468 effect->setDevice(mOutDevice); 1469 effect->setDevice(mInDevice); 1470 effect->setMode(mAudioFlinger->getMode()); 1471 effect->setAudioSource(mAudioSource); 1472 } 1473 // create effect handle and connect it to effect module 1474 handle = new EffectHandle(effect, client, effectClient, priority); 1475 lStatus = handle->initCheck(); 1476 if (lStatus == OK) { 1477 lStatus = effect->addHandle(handle.get()); 1478 } 1479 if (enabled != NULL) { 1480 *enabled = (int)effect->isEnabled(); 1481 } 1482 } 1483 1484 Exit: 1485 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 1486 Mutex::Autolock _l(mLock); 1487 if (effectCreated) { 1488 chain->removeEffect_l(effect); 1489 } 1490 if (effectRegistered) { 1491 AudioSystem::unregisterEffect(effect->id()); 1492 } 1493 if (chainCreated) { 1494 removeEffectChain_l(chain); 1495 } 1496 handle.clear(); 1497 } 1498 1499 *status = lStatus; 1500 return handle; 1501 } 1502 1503 sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect(audio_session_t sessionId, 1504 int effectId) 1505 { 1506 Mutex::Autolock _l(mLock); 1507 return getEffect_l(sessionId, effectId); 1508 } 1509 1510 sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(audio_session_t sessionId, 1511 int effectId) 1512 { 1513 sp<EffectChain> chain = getEffectChain_l(sessionId); 1514 return chain != 0 ? chain->getEffectFromId_l(effectId) : 0; 1515 } 1516 1517 // PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 1518 // PlaybackThread::mLock held 1519 status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 1520 { 1521 // check for existing effect chain with the requested audio session 1522 audio_session_t sessionId = effect->sessionId(); 1523 sp<EffectChain> chain = getEffectChain_l(sessionId); 1524 bool chainCreated = false; 1525 1526 ALOGD_IF((mType == OFFLOAD) && !effect->isOffloadable(), 1527 "addEffect_l() on offloaded thread %p: effect %s does not support offload flags %x", 1528 this, effect->desc().name, effect->desc().flags); 1529 1530 if (chain == 0) { 1531 // create a new chain for this session 1532 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 1533 chain = new EffectChain(this, sessionId); 1534 addEffectChain_l(chain); 1535 chain->setStrategy(getStrategyForSession_l(sessionId)); 1536 chainCreated = true; 1537 } 1538 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 1539 1540 if (chain->getEffectFromId_l(effect->id()) != 0) { 1541 ALOGW("addEffect_l() %p effect %s already present in chain %p", 1542 this, effect->desc().name, chain.get()); 1543 return BAD_VALUE; 1544 } 1545 1546 effect->setOffloaded(mType == OFFLOAD, mId); 1547 1548 status_t status = chain->addEffect_l(effect); 1549 if (status != NO_ERROR) { 1550 if (chainCreated) { 1551 removeEffectChain_l(chain); 1552 } 1553 return status; 1554 } 1555 1556 effect->setDevice(mOutDevice); 1557 effect->setDevice(mInDevice); 1558 effect->setMode(mAudioFlinger->getMode()); 1559 effect->setAudioSource(mAudioSource); 1560 return NO_ERROR; 1561 } 1562 1563 void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 1564 1565 ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 1566 effect_descriptor_t desc = effect->desc(); 1567 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 1568 detachAuxEffect_l(effect->id()); 1569 } 1570 1571 sp<EffectChain> chain = effect->chain().promote(); 1572 if (chain != 0) { 1573 // remove effect chain if removing last effect 1574 if (chain->removeEffect_l(effect) == 0) { 1575 removeEffectChain_l(chain); 1576 } 1577 } else { 1578 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 1579 } 1580 } 1581 1582 void AudioFlinger::ThreadBase::lockEffectChains_l( 1583 Vector< sp<AudioFlinger::EffectChain> >& effectChains) 1584 { 1585 effectChains = mEffectChains; 1586 for (size_t i = 0; i < mEffectChains.size(); i++) { 1587 mEffectChains[i]->lock(); 1588 } 1589 } 1590 1591 void AudioFlinger::ThreadBase::unlockEffectChains( 1592 const Vector< sp<AudioFlinger::EffectChain> >& effectChains) 1593 { 1594 for (size_t i = 0; i < effectChains.size(); i++) { 1595 effectChains[i]->unlock(); 1596 } 1597 } 1598 1599 sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(audio_session_t sessionId) 1600 { 1601 Mutex::Autolock _l(mLock); 1602 return getEffectChain_l(sessionId); 1603 } 1604 1605 sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(audio_session_t sessionId) 1606 const 1607 { 1608 size_t size = mEffectChains.size(); 1609 for (size_t i = 0; i < size; i++) { 1610 if (mEffectChains[i]->sessionId() == sessionId) { 1611 return mEffectChains[i]; 1612 } 1613 } 1614 return 0; 1615 } 1616 1617 void AudioFlinger::ThreadBase::setMode(audio_mode_t mode) 1618 { 1619 Mutex::Autolock _l(mLock); 1620 size_t size = mEffectChains.size(); 1621 for (size_t i = 0; i < size; i++) { 1622 mEffectChains[i]->setMode_l(mode); 1623 } 1624 } 1625 1626 void AudioFlinger::ThreadBase::getAudioPortConfig(struct audio_port_config *config) 1627 { 1628 config->type = AUDIO_PORT_TYPE_MIX; 1629 config->ext.mix.handle = mId; 1630 config->sample_rate = mSampleRate; 1631 config->format = mFormat; 1632 config->channel_mask = mChannelMask; 1633 config->config_mask = AUDIO_PORT_CONFIG_SAMPLE_RATE|AUDIO_PORT_CONFIG_CHANNEL_MASK| 1634 AUDIO_PORT_CONFIG_FORMAT; 1635 } 1636 1637 void AudioFlinger::ThreadBase::systemReady() 1638 { 1639 Mutex::Autolock _l(mLock); 1640 if (mSystemReady) { 1641 return; 1642 } 1643 mSystemReady = true; 1644 1645 for (size_t i = 0; i < mPendingConfigEvents.size(); i++) { 1646 sendConfigEvent_l(mPendingConfigEvents.editItemAt(i)); 1647 } 1648 mPendingConfigEvents.clear(); 1649 } 1650 1651 1652 // ---------------------------------------------------------------------------- 1653 // Playback 1654 // ---------------------------------------------------------------------------- 1655 1656 AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, 1657 AudioStreamOut* output, 1658 audio_io_handle_t id, 1659 audio_devices_t device, 1660 type_t type, 1661 bool systemReady) 1662 : ThreadBase(audioFlinger, id, device, AUDIO_DEVICE_NONE, type, systemReady), 1663 mNormalFrameCount(0), mSinkBuffer(NULL), 1664 mMixerBufferEnabled(AudioFlinger::kEnableExtendedPrecision), 1665 mMixerBuffer(NULL), 1666 mMixerBufferSize(0), 1667 mMixerBufferFormat(AUDIO_FORMAT_INVALID), 1668 mMixerBufferValid(false), 1669 mEffectBufferEnabled(AudioFlinger::kEnableExtendedPrecision), 1670 mEffectBuffer(NULL), 1671 mEffectBufferSize(0), 1672 mEffectBufferFormat(AUDIO_FORMAT_INVALID), 1673 mEffectBufferValid(false), 1674 mSuspended(0), mBytesWritten(0), 1675 mFramesWritten(0), 1676 mSuspendedFrames(0), 1677 mActiveTracksGeneration(0), 1678 // mStreamTypes[] initialized in constructor body 1679 mOutput(output), 1680 mLastWriteTime(-1), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false), 1681 mMixerStatus(MIXER_IDLE), 1682 mMixerStatusIgnoringFastTracks(MIXER_IDLE), 1683 mStandbyDelayNs(AudioFlinger::mStandbyTimeInNsecs), 1684 mBytesRemaining(0), 1685 mCurrentWriteLength(0), 1686 mUseAsyncWrite(false), 1687 mWriteAckSequence(0), 1688 mDrainSequence(0), 1689 mSignalPending(false), 1690 mScreenState(AudioFlinger::mScreenState), 1691 // index 0 is reserved for normal mixer's submix 1692 mFastTrackAvailMask(((1 << FastMixerState::sMaxFastTracks) - 1) & ~1), 1693 mHwSupportsPause(false), mHwPaused(false), mFlushPending(false) 1694 { 1695 snprintf(mThreadName, kThreadNameLength, "AudioOut_%X", id); 1696 mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mThreadName); 1697 1698 // Assumes constructor is called by AudioFlinger with it's mLock held, but 1699 // it would be safer to explicitly pass initial masterVolume/masterMute as 1700 // parameter. 1701 // 1702 // If the HAL we are using has support for master volume or master mute, 1703 // then do not attenuate or mute during mixing (just leave the volume at 1.0 1704 // and the mute set to false). 1705 mMasterVolume = audioFlinger->masterVolume_l(); 1706 mMasterMute = audioFlinger->masterMute_l(); 1707 if (mOutput && mOutput->audioHwDev) { 1708 if (mOutput->audioHwDev->canSetMasterVolume()) { 1709 mMasterVolume = 1.0; 1710 } 1711 1712 if (mOutput->audioHwDev->canSetMasterMute()) { 1713 mMasterMute = false; 1714 } 1715 } 1716 1717 readOutputParameters_l(); 1718 1719 // ++ operator does not compile 1720 for (audio_stream_type_t stream = AUDIO_STREAM_MIN; stream < AUDIO_STREAM_CNT; 1721 stream = (audio_stream_type_t) (stream + 1)) { 1722 mStreamTypes[stream].volume = mAudioFlinger->streamVolume_l(stream); 1723 mStreamTypes[stream].mute = mAudioFlinger->streamMute_l(stream); 1724 } 1725 } 1726 1727 AudioFlinger::PlaybackThread::~PlaybackThread() 1728 { 1729 mAudioFlinger->unregisterWriter(mNBLogWriter); 1730 free(mSinkBuffer); 1731 free(mMixerBuffer); 1732 free(mEffectBuffer); 1733 } 1734 1735 void AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) 1736 { 1737 dumpInternals(fd, args); 1738 dumpTracks(fd, args); 1739 dumpEffectChains(fd, args); 1740 } 1741 1742 void AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args __unused) 1743 { 1744 const size_t SIZE = 256; 1745 char buffer[SIZE]; 1746 String8 result; 1747 1748 result.appendFormat(" Stream volumes in dB: "); 1749 for (int i = 0; i < AUDIO_STREAM_CNT; ++i) { 1750 const stream_type_t *st = &mStreamTypes[i]; 1751 if (i > 0) { 1752 result.appendFormat(", "); 1753 } 1754 result.appendFormat("%d:%.2g", i, 20.0 * log10(st->volume)); 1755 if (st->mute) { 1756 result.append("M"); 1757 } 1758 } 1759 result.append("\n"); 1760 write(fd, result.string(), result.length()); 1761 result.clear(); 1762 1763 // These values are "raw"; they will wrap around. See prepareTracks_l() for a better way. 1764 FastTrackUnderruns underruns = getFastTrackUnderruns(0); 1765 dprintf(fd, " Normal mixer raw underrun counters: partial=%u empty=%u\n", 1766 underruns.mBitFields.mPartial, underruns.mBitFields.mEmpty); 1767 1768 size_t numtracks = mTracks.size(); 1769 size_t numactive = mActiveTracks.size(); 1770 dprintf(fd, " %zu Tracks", numtracks); 1771 size_t numactiveseen = 0; 1772 if (numtracks) { 1773 dprintf(fd, " of which %zu are active\n", numactive); 1774 Track::appendDumpHeader(result); 1775 for (size_t i = 0; i < numtracks; ++i) { 1776 sp<Track> track = mTracks[i]; 1777 if (track != 0) { 1778 bool active = mActiveTracks.indexOf(track) >= 0; 1779 if (active) { 1780 numactiveseen++; 1781 } 1782 track->dump(buffer, SIZE, active); 1783 result.append(buffer); 1784 } 1785 } 1786 } else { 1787 result.append("\n"); 1788 } 1789 if (numactiveseen != numactive) { 1790 // some tracks in the active list were not in the tracks list 1791 snprintf(buffer, SIZE, " The following tracks are in the active list but" 1792 " not in the track list\n"); 1793 result.append(buffer); 1794 Track::appendDumpHeader(result); 1795 for (size_t i = 0; i < numactive; ++i) { 1796 sp<Track> track = mActiveTracks[i].promote(); 1797 if (track != 0 && mTracks.indexOf(track) < 0) { 1798 track->dump(buffer, SIZE, true); 1799 result.append(buffer); 1800 } 1801 } 1802 } 1803 1804 write(fd, result.string(), result.size()); 1805 } 1806 1807 void AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 1808 { 1809 dprintf(fd, "\nOutput thread %p type %d (%s):\n", this, type(), threadTypeToString(type())); 1810 1811 dumpBase(fd, args); 1812 1813 dprintf(fd, " Normal frame count: %zu\n", mNormalFrameCount); 1814 dprintf(fd, " Last write occurred (msecs): %llu\n", 1815 (unsigned long long) ns2ms(systemTime() - mLastWriteTime)); 1816 dprintf(fd, " Total writes: %d\n", mNumWrites); 1817 dprintf(fd, " Delayed writes: %d\n", mNumDelayedWrites); 1818 dprintf(fd, " Blocked in write: %s\n", mInWrite ? "yes" : "no"); 1819 dprintf(fd, " Suspend count: %d\n", mSuspended); 1820 dprintf(fd, " Sink buffer : %p\n", mSinkBuffer); 1821 dprintf(fd, " Mixer buffer: %p\n", mMixerBuffer); 1822 dprintf(fd, " Effect buffer: %p\n", mEffectBuffer); 1823 dprintf(fd, " Fast track availMask=%#x\n", mFastTrackAvailMask); 1824 dprintf(fd, " Standby delay ns=%lld\n", (long long)mStandbyDelayNs); 1825 AudioStreamOut *output = mOutput; 1826 audio_output_flags_t flags = output != NULL ? output->flags : AUDIO_OUTPUT_FLAG_NONE; 1827 String8 flagsAsString = outputFlagsToString(flags); 1828 dprintf(fd, " AudioStreamOut: %p flags %#x (%s)\n", output, flags, flagsAsString.string()); 1829 dprintf(fd, " Frames written: %lld\n", (long long)mFramesWritten); 1830 dprintf(fd, " Suspended frames: %lld\n", (long long)mSuspendedFrames); 1831 if (mPipeSink.get() != nullptr) { 1832 dprintf(fd, " PipeSink frames written: %lld\n", (long long)mPipeSink->framesWritten()); 1833 } 1834 if (output != nullptr) { 1835 dprintf(fd, " Hal stream dump:\n"); 1836 (void)output->stream->common.dump(&output->stream->common, fd); 1837 } 1838 } 1839 1840 // Thread virtuals 1841 1842 void AudioFlinger::PlaybackThread::onFirstRef() 1843 { 1844 run(mThreadName, ANDROID_PRIORITY_URGENT_AUDIO); 1845 } 1846 1847 // ThreadBase virtuals 1848 void AudioFlinger::PlaybackThread::preExit() 1849 { 1850 ALOGV(" preExit()"); 1851 // FIXME this is using hard-coded strings but in the future, this functionality will be 1852 // converted to use audio HAL extensions required to support tunneling 1853 mOutput->stream->common.set_parameters(&mOutput->stream->common, "exiting=1"); 1854 } 1855 1856 // PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 1857 sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 1858 const sp<AudioFlinger::Client>& client, 1859 audio_stream_type_t streamType, 1860 uint32_t sampleRate, 1861 audio_format_t format, 1862 audio_channel_mask_t channelMask, 1863 size_t *pFrameCount, 1864 const sp<IMemory>& sharedBuffer, 1865 audio_session_t sessionId, 1866 audio_output_flags_t *flags, 1867 pid_t tid, 1868 int uid, 1869 status_t *status) 1870 { 1871 size_t frameCount = *pFrameCount; 1872 sp<Track> track; 1873 status_t lStatus; 1874 audio_output_flags_t outputFlags = mOutput->flags; 1875 1876 // special case for FAST flag considered OK if fast mixer is present 1877 if (hasFastMixer()) { 1878 outputFlags = (audio_output_flags_t)(outputFlags | AUDIO_OUTPUT_FLAG_FAST); 1879 } 1880 1881 // Check if requested flags are compatible with output stream flags 1882 if ((*flags & outputFlags) != *flags) { 1883 ALOGW("createTrack_l(): mismatch between requested flags (%08x) and output flags (%08x)", 1884 *flags, outputFlags); 1885 *flags = (audio_output_flags_t)(*flags & outputFlags); 1886 } 1887 1888 // client expresses a preference for FAST, but we get the final say 1889 if (*flags & AUDIO_OUTPUT_FLAG_FAST) { 1890 if ( 1891 // PCM data 1892 audio_is_linear_pcm(format) && 1893 // TODO: extract as a data library function that checks that a computationally 1894 // expensive downmixer is not required: isFastOutputChannelConversion() 1895 (channelMask == mChannelMask || 1896 mChannelMask != AUDIO_CHANNEL_OUT_STEREO || 1897 (channelMask == AUDIO_CHANNEL_OUT_MONO 1898 /* && mChannelMask == AUDIO_CHANNEL_OUT_STEREO */)) && 1899 // hardware sample rate 1900 (sampleRate == mSampleRate) && 1901 // normal mixer has an associated fast mixer 1902 hasFastMixer() && 1903 // there are sufficient fast track slots available 1904 (mFastTrackAvailMask != 0) 1905 // FIXME test that MixerThread for this fast track has a capable output HAL 1906 // FIXME add a permission test also? 1907 ) { 1908 // static tracks can have any nonzero framecount, streaming tracks check against minimum. 1909 if (sharedBuffer == 0) { 1910 // read the fast track multiplier property the first time it is needed 1911 int ok = pthread_once(&sFastTrackMultiplierOnce, sFastTrackMultiplierInit); 1912 if (ok != 0) { 1913 ALOGE("%s pthread_once failed: %d", __func__, ok); 1914 } 1915 frameCount = max(frameCount, mFrameCount * sFastTrackMultiplier); // incl framecount 0 1916 } 1917 1918 // check compatibility with audio effects. 1919 { // scope for mLock 1920 Mutex::Autolock _l(mLock); 1921 for (audio_session_t session : { 1922 AUDIO_SESSION_OUTPUT_STAGE, 1923 AUDIO_SESSION_OUTPUT_MIX, 1924 sessionId, 1925 }) { 1926 sp<EffectChain> chain = getEffectChain_l(session); 1927 if (chain.get() != nullptr) { 1928 audio_output_flags_t old = *flags; 1929 chain->checkOutputFlagCompatibility(flags); 1930 if (old != *flags) { 1931 ALOGV("AUDIO_OUTPUT_FLAGS denied by effect, session=%d old=%#x new=%#x", 1932 (int)session, (int)old, (int)*flags); 1933 } 1934 } 1935 } 1936 } 1937 ALOGV_IF((*flags & AUDIO_OUTPUT_FLAG_FAST) != 0, 1938 "AUDIO_OUTPUT_FLAG_FAST accepted: frameCount=%zu mFrameCount=%zu", 1939 frameCount, mFrameCount); 1940 } else { 1941 ALOGV("AUDIO_OUTPUT_FLAG_FAST denied: sharedBuffer=%p frameCount=%zu " 1942 "mFrameCount=%zu format=%#x mFormat=%#x isLinear=%d channelMask=%#x " 1943 "sampleRate=%u mSampleRate=%u " 1944 "hasFastMixer=%d tid=%d fastTrackAvailMask=%#x", 1945 sharedBuffer.get(), frameCount, mFrameCount, format, mFormat, 1946 audio_is_linear_pcm(format), 1947 channelMask, sampleRate, mSampleRate, hasFastMixer(), tid, mFastTrackAvailMask); 1948 *flags = (audio_output_flags_t)(*flags & ~AUDIO_OUTPUT_FLAG_FAST); 1949 } 1950 } 1951 // For normal PCM streaming tracks, update minimum frame count. 1952 // For compatibility with AudioTrack calculation, buffer depth is forced 1953 // to be at least 2 x the normal mixer frame count and cover audio hardware latency. 1954 // This is probably too conservative, but legacy application code may depend on it. 1955 // If you change this calculation, also review the start threshold which is related. 1956 if (!(*flags & AUDIO_OUTPUT_FLAG_FAST) 1957 && audio_has_proportional_frames(format) && sharedBuffer == 0) { 1958 // this must match AudioTrack.cpp calculateMinFrameCount(). 1959 // TODO: Move to a common library 1960 uint32_t latencyMs = mOutput->stream->get_latency(mOutput->stream); 1961 uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate); 1962 if (minBufCount < 2) { 1963 minBufCount = 2; 1964 } 1965 // For normal mixing tracks, if speed is > 1.0f (normal), AudioTrack 1966 // or the client should compute and pass in a larger buffer request. 1967 size_t minFrameCount = 1968 minBufCount * sourceFramesNeededWithTimestretch( 1969 sampleRate, mNormalFrameCount, 1970 mSampleRate, AUDIO_TIMESTRETCH_SPEED_NORMAL /*speed*/); 1971 if (frameCount < minFrameCount) { // including frameCount == 0 1972 frameCount = minFrameCount; 1973 } 1974 } 1975 *pFrameCount = frameCount; 1976 1977 switch (mType) { 1978 1979 case DIRECT: 1980 if (audio_is_linear_pcm(format)) { // TODO maybe use audio_has_proportional_frames()? 1981 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1982 ALOGE("createTrack_l() Bad parameter: sampleRate %u format %#x, channelMask 0x%08x " 1983 "for output %p with format %#x", 1984 sampleRate, format, channelMask, mOutput, mFormat); 1985 lStatus = BAD_VALUE; 1986 goto Exit; 1987 } 1988 } 1989 break; 1990 1991 case OFFLOAD: 1992 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1993 ALOGE("createTrack_l() Bad parameter: sampleRate %d format %#x, channelMask 0x%08x \"" 1994 "for output %p with format %#x", 1995 sampleRate, format, channelMask, mOutput, mFormat); 1996 lStatus = BAD_VALUE; 1997 goto Exit; 1998 } 1999 break; 2000 2001 default: 2002 if (!audio_is_linear_pcm(format)) { 2003 ALOGE("createTrack_l() Bad parameter: format %#x \"" 2004 "for output %p with format %#x", 2005 format, mOutput, mFormat); 2006 lStatus = BAD_VALUE; 2007 goto Exit; 2008 } 2009 if (sampleRate > mSampleRate * AUDIO_RESAMPLER_DOWN_RATIO_MAX) { 2010 ALOGE("Sample rate out of range: %u mSampleRate %u", sampleRate, mSampleRate); 2011 lStatus = BAD_VALUE; 2012 goto Exit; 2013 } 2014 break; 2015 2016 } 2017 2018 lStatus = initCheck(); 2019 if (lStatus != NO_ERROR) { 2020 ALOGE("createTrack_l() audio driver not initialized"); 2021 goto Exit; 2022 } 2023 2024 { // scope for mLock 2025 Mutex::Autolock _l(mLock); 2026 2027 // all tracks in same audio session must share the same routing strategy otherwise 2028 // conflicts will happen when tracks are moved from one output to another by audio policy 2029 // manager 2030 uint32_t strategy = AudioSystem::getStrategyForStream(streamType); 2031 for (size_t i = 0; i < mTracks.size(); ++i) { 2032 sp<Track> t = mTracks[i]; 2033 if (t != 0 && t->isExternalTrack()) { 2034 uint32_t actual = AudioSystem::getStrategyForStream(t->streamType()); 2035 if (sessionId == t->sessionId() && strategy != actual) { 2036 ALOGE("createTrack_l() mismatched strategy; expected %u but found %u", 2037 strategy, actual); 2038 lStatus = BAD_VALUE; 2039 goto Exit; 2040 } 2041 } 2042 } 2043 2044 track = new Track(this, client, streamType, sampleRate, format, 2045 channelMask, frameCount, NULL, sharedBuffer, 2046 sessionId, uid, *flags, TrackBase::TYPE_DEFAULT); 2047 2048 lStatus = track != 0 ? track->initCheck() : (status_t) NO_MEMORY; 2049 if (lStatus != NO_ERROR) { 2050 ALOGE("createTrack_l() initCheck failed %d; no control block?", lStatus); 2051 // track must be cleared from the caller as the caller has the AF lock 2052 goto Exit; 2053 } 2054 mTracks.add(track); 2055 2056 sp<EffectChain> chain = getEffectChain_l(sessionId); 2057 if (chain != 0) { 2058 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 2059 track->setMainBuffer(chain->inBuffer()); 2060 chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType())); 2061 chain->incTrackCnt(); 2062 } 2063 2064 if ((*flags & AUDIO_OUTPUT_FLAG_FAST) && (tid != -1)) { 2065 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 2066 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 2067 // so ask activity manager to do this on our behalf 2068 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 2069 } 2070 } 2071 2072 lStatus = NO_ERROR; 2073 2074 Exit: 2075 *status = lStatus; 2076 return track; 2077 } 2078 2079 uint32_t AudioFlinger::PlaybackThread::correctLatency_l(uint32_t latency) const 2080 { 2081 return latency; 2082 } 2083 2084 uint32_t AudioFlinger::PlaybackThread::latency() const 2085 { 2086 Mutex::Autolock _l(mLock); 2087 return latency_l(); 2088 } 2089 uint32_t AudioFlinger::PlaybackThread::latency_l() const 2090 { 2091 if (initCheck() == NO_ERROR) { 2092 return correctLatency_l(mOutput->stream->get_latency(mOutput->stream)); 2093 } else { 2094 return 0; 2095 } 2096 } 2097 2098 void AudioFlinger::PlaybackThread::setMasterVolume(float value) 2099 { 2100 Mutex::Autolock _l(mLock); 2101 // Don't apply master volume in SW if our HAL can do it for us. 2102 if (mOutput && mOutput->audioHwDev && 2103 mOutput->audioHwDev->canSetMasterVolume()) { 2104 mMasterVolume = 1.0; 2105 } else { 2106 mMasterVolume = value; 2107 } 2108 } 2109 2110 void AudioFlinger::PlaybackThread::setMasterMute(bool muted) 2111 { 2112 Mutex::Autolock _l(mLock); 2113 // Don't apply master mute in SW if our HAL can do it for us. 2114 if (mOutput && mOutput->audioHwDev && 2115 mOutput->audioHwDev->canSetMasterMute()) { 2116 mMasterMute = false; 2117 } else { 2118 mMasterMute = muted; 2119 } 2120 } 2121 2122 void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) 2123 { 2124 Mutex::Autolock _l(mLock); 2125 mStreamTypes[stream].volume = value; 2126 broadcast_l(); 2127 } 2128 2129 void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) 2130 { 2131 Mutex::Autolock _l(mLock); 2132 mStreamTypes[stream].mute = muted; 2133 broadcast_l(); 2134 } 2135 2136 float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const 2137 { 2138 Mutex::Autolock _l(mLock); 2139 return mStreamTypes[stream].volume; 2140 } 2141 2142 // addTrack_l() must be called with ThreadBase::mLock held 2143 status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 2144 { 2145 status_t status = ALREADY_EXISTS; 2146 2147 if (mActiveTracks.indexOf(track) < 0) { 2148 // the track is newly added, make sure it fills up all its 2149 // buffers before playing. This is to ensure the client will 2150 // effectively get the latency it requested. 2151 if (track->isExternalTrack()) { 2152 TrackBase::track_state state = track->mState; 2153 mLock.unlock(); 2154 status = AudioSystem::startOutput(mId, track->streamType(), 2155 track->sessionId()); 2156 mLock.lock(); 2157 // abort track was stopped/paused while we released the lock 2158 if (state != track->mState) { 2159 if (status == NO_ERROR) { 2160 mLock.unlock(); 2161 AudioSystem::stopOutput(mId, track->streamType(), 2162 track->sessionId()); 2163 mLock.lock(); 2164 } 2165 return INVALID_OPERATION; 2166 } 2167 // abort if start is rejected by audio policy manager 2168 if (status != NO_ERROR) { 2169 return PERMISSION_DENIED; 2170 } 2171 #ifdef ADD_BATTERY_DATA 2172 // to track the speaker usage 2173 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 2174 #endif 2175 } 2176 2177 // set retry count for buffer fill 2178 if (track->isOffloaded()) { 2179 if (track->isStopping_1()) { 2180 track->mRetryCount = kMaxTrackStopRetriesOffload; 2181 } else { 2182 track->mRetryCount = kMaxTrackStartupRetriesOffload; 2183 } 2184 track->mFillingUpStatus = mStandby ? Track::FS_FILLING : Track::FS_FILLED; 2185 } else { 2186 track->mRetryCount = kMaxTrackStartupRetries; 2187 track->mFillingUpStatus = 2188 track->sharedBuffer() != 0 ? Track::FS_FILLED : Track::FS_FILLING; 2189 } 2190 2191 track->mResetDone = false; 2192 track->mPresentationCompleteFrames = 0; 2193 mActiveTracks.add(track); 2194 mWakeLockUids.add(track->uid()); 2195 mActiveTracksGeneration++; 2196 mLatestActiveTrack = track; 2197 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 2198 if (chain != 0) { 2199 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), 2200 track->sessionId()); 2201 chain->incActiveTrackCnt(); 2202 } 2203 2204 status = NO_ERROR; 2205 } 2206 2207 onAddNewTrack_l(); 2208 return status; 2209 } 2210 2211 bool AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 2212 { 2213 track->terminate(); 2214 // active tracks are removed by threadLoop() 2215 bool trackActive = (mActiveTracks.indexOf(track) >= 0); 2216 track->mState = TrackBase::STOPPED; 2217 if (!trackActive) { 2218 removeTrack_l(track); 2219 } else if (track->isFastTrack() || track->isOffloaded() || track->isDirect()) { 2220 track->mState = TrackBase::STOPPING_1; 2221 } 2222 2223 return trackActive; 2224 } 2225 2226 void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 2227 { 2228 track->triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 2229 mTracks.remove(track); 2230 deleteTrackName_l(track->name()); 2231 // redundant as track is about to be destroyed, for dumpsys only 2232 track->mName = -1; 2233 if (track->isFastTrack()) { 2234 int index = track->mFastIndex; 2235 ALOG_ASSERT(0 < index && index < (int)FastMixerState::sMaxFastTracks); 2236 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << index))); 2237 mFastTrackAvailMask |= 1 << index; 2238 // redundant as track is about to be destroyed, for dumpsys only 2239 track->mFastIndex = -1; 2240 } 2241 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 2242 if (chain != 0) { 2243 chain->decTrackCnt(); 2244 } 2245 } 2246 2247 void AudioFlinger::PlaybackThread::broadcast_l() 2248 { 2249 // Thread could be blocked waiting for async 2250 // so signal it to handle state changes immediately 2251 // If threadLoop is currently unlocked a signal of mWaitWorkCV will 2252 // be lost so we also flag to prevent it blocking on mWaitWorkCV 2253 mSignalPending = true; 2254 mWaitWorkCV.broadcast(); 2255 } 2256 2257 String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 2258 { 2259 Mutex::Autolock _l(mLock); 2260 if (initCheck() != NO_ERROR) { 2261 return String8(); 2262 } 2263 2264 char *s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 2265 const String8 out_s8(s); 2266 free(s); 2267 return out_s8; 2268 } 2269 2270 void AudioFlinger::PlaybackThread::ioConfigChanged(audio_io_config_event event, pid_t pid) { 2271 sp<AudioIoDescriptor> desc = new AudioIoDescriptor(); 2272 ALOGV("PlaybackThread::ioConfigChanged, thread %p, event %d", this, event); 2273 2274 desc->mIoHandle = mId; 2275 2276 switch (event) { 2277 case AUDIO_OUTPUT_OPENED: 2278 case AUDIO_OUTPUT_CONFIG_CHANGED: 2279 desc->mPatch = mPatch; 2280 desc->mChannelMask = mChannelMask; 2281 desc->mSamplingRate = mSampleRate; 2282 desc->mFormat = mFormat; 2283 desc->mFrameCount = mNormalFrameCount; // FIXME see 2284 // AudioFlinger::frameCount(audio_io_handle_t) 2285 desc->mFrameCountHAL = mFrameCount; 2286 desc->mLatency = latency_l(); 2287 break; 2288 2289 case AUDIO_OUTPUT_CLOSED: 2290 default: 2291 break; 2292 } 2293 mAudioFlinger->ioConfigChanged(event, desc, pid); 2294 } 2295 2296 void AudioFlinger::PlaybackThread::writeCallback() 2297 { 2298 ALOG_ASSERT(mCallbackThread != 0); 2299 mCallbackThread->resetWriteBlocked(); 2300 } 2301 2302 void AudioFlinger::PlaybackThread::drainCallback() 2303 { 2304 ALOG_ASSERT(mCallbackThread != 0); 2305 mCallbackThread->resetDraining(); 2306 } 2307 2308 void AudioFlinger::PlaybackThread::errorCallback() 2309 { 2310 ALOG_ASSERT(mCallbackThread != 0); 2311 mCallbackThread->setAsyncError(); 2312 } 2313 2314 void AudioFlinger::PlaybackThread::resetWriteBlocked(uint32_t sequence) 2315 { 2316 Mutex::Autolock _l(mLock); 2317 // reject out of sequence requests 2318 if ((mWriteAckSequence & 1) && (sequence == mWriteAckSequence)) { 2319 mWriteAckSequence &= ~1; 2320 mWaitWorkCV.signal(); 2321 } 2322 } 2323 2324 void AudioFlinger::PlaybackThread::resetDraining(uint32_t sequence) 2325 { 2326 Mutex::Autolock _l(mLock); 2327 // reject out of sequence requests 2328 if ((mDrainSequence & 1) && (sequence == mDrainSequence)) { 2329 mDrainSequence &= ~1; 2330 mWaitWorkCV.signal(); 2331 } 2332 } 2333 2334 // static 2335 int AudioFlinger::PlaybackThread::asyncCallback(stream_callback_event_t event, 2336 void *param __unused, 2337 void *cookie) 2338 { 2339 AudioFlinger::PlaybackThread *me = (AudioFlinger::PlaybackThread *)cookie; 2340 ALOGV("asyncCallback() event %d", event); 2341 switch (event) { 2342 case STREAM_CBK_EVENT_WRITE_READY: 2343 me->writeCallback(); 2344 break; 2345 case STREAM_CBK_EVENT_DRAIN_READY: 2346 me->drainCallback(); 2347 break; 2348 case STREAM_CBK_EVENT_ERROR: 2349 me->errorCallback(); 2350 break; 2351 default: 2352 ALOGW("asyncCallback() unknown event %d", event); 2353 break; 2354 } 2355 return 0; 2356 } 2357 2358 void AudioFlinger::PlaybackThread::readOutputParameters_l() 2359 { 2360 // unfortunately we have no way of recovering from errors here, hence the LOG_ALWAYS_FATAL 2361 mSampleRate = mOutput->getSampleRate(); 2362 mChannelMask = mOutput->getChannelMask(); 2363 if (!audio_is_output_channel(mChannelMask)) { 2364 LOG_ALWAYS_FATAL("HAL channel mask %#x not valid for output", mChannelMask); 2365 } 2366 if ((mType == MIXER || mType == DUPLICATING) 2367 && !isValidPcmSinkChannelMask(mChannelMask)) { 2368 LOG_ALWAYS_FATAL("HAL channel mask %#x not supported for mixed output", 2369 mChannelMask); 2370 } 2371 mChannelCount = audio_channel_count_from_out_mask(mChannelMask); 2372 2373 // Get actual HAL format. 2374 mHALFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 2375 // Get format from the shim, which will be different than the HAL format 2376 // if playing compressed audio over HDMI passthrough. 2377 mFormat = mOutput->getFormat(); 2378 if (!audio_is_valid_format(mFormat)) { 2379 LOG_ALWAYS_FATAL("HAL format %#x not valid for output", mFormat); 2380 } 2381 if ((mType == MIXER || mType == DUPLICATING) 2382 && !isValidPcmSinkFormat(mFormat)) { 2383 LOG_FATAL("HAL format %#x not supported for mixed output", 2384 mFormat); 2385 } 2386 mFrameSize = mOutput->getFrameSize(); 2387 mBufferSize = mOutput->stream->common.get_buffer_size(&mOutput->stream->common); 2388 mFrameCount = mBufferSize / mFrameSize; 2389 if (mFrameCount & 15) { 2390 ALOGW("HAL output buffer size is %zu frames but AudioMixer requires multiples of 16 frames", 2391 mFrameCount); 2392 } 2393 2394 if ((mOutput->flags & AUDIO_OUTPUT_FLAG_NON_BLOCKING) && 2395 (mOutput->stream->set_callback != NULL)) { 2396 if (mOutput->stream->set_callback(mOutput->stream, 2397 AudioFlinger::PlaybackThread::asyncCallback, this) == 0) { 2398 mUseAsyncWrite = true; 2399 mCallbackThread = new AudioFlinger::AsyncCallbackThread(this); 2400 } 2401 } 2402 2403 mHwSupportsPause = false; 2404 if (mOutput->flags & AUDIO_OUTPUT_FLAG_DIRECT) { 2405 if (mOutput->stream->pause != NULL) { 2406 if (mOutput->stream->resume != NULL) { 2407 mHwSupportsPause = true; 2408 } else { 2409 ALOGW("direct output implements pause but not resume"); 2410 } 2411 } else if (mOutput->stream->resume != NULL) { 2412 ALOGW("direct output implements resume but not pause"); 2413 } 2414 } 2415 if (!mHwSupportsPause && mOutput->flags & AUDIO_OUTPUT_FLAG_HW_AV_SYNC) { 2416 LOG_ALWAYS_FATAL("HW_AV_SYNC requested but HAL does not implement pause and resume"); 2417 } 2418 2419 if (mType == DUPLICATING && mMixerBufferEnabled && mEffectBufferEnabled) { 2420 // For best precision, we use float instead of the associated output 2421 // device format (typically PCM 16 bit). 2422 2423 mFormat = AUDIO_FORMAT_PCM_FLOAT; 2424 mFrameSize = mChannelCount * audio_bytes_per_sample(mFormat); 2425 mBufferSize = mFrameSize * mFrameCount; 2426 2427 // TODO: We currently use the associated output device channel mask and sample rate. 2428 // (1) Perhaps use the ORed channel mask of all downstream MixerThreads 2429 // (if a valid mask) to avoid premature downmix. 2430 // (2) Perhaps use the maximum sample rate of all downstream MixerThreads 2431 // instead of the output device sample rate to avoid loss of high frequency information. 2432 // This may need to be updated as MixerThread/OutputTracks are added and not here. 2433 } 2434 2435 // Calculate size of normal sink buffer relative to the HAL output buffer size 2436 double multiplier = 1.0; 2437 if (mType == MIXER && (kUseFastMixer == FastMixer_Static || 2438 kUseFastMixer == FastMixer_Dynamic)) { 2439 size_t minNormalFrameCount = (kMinNormalSinkBufferSizeMs * mSampleRate) / 1000; 2440 size_t maxNormalFrameCount = (kMaxNormalSinkBufferSizeMs * mSampleRate) / 1000; 2441 2442 // round up minimum and round down maximum to nearest 16 frames to satisfy AudioMixer 2443 minNormalFrameCount = (minNormalFrameCount + 15) & ~15; 2444 maxNormalFrameCount = maxNormalFrameCount & ~15; 2445 if (maxNormalFrameCount < minNormalFrameCount) { 2446 maxNormalFrameCount = minNormalFrameCount; 2447 } 2448 multiplier = (double) minNormalFrameCount / (double) mFrameCount; 2449 if (multiplier <= 1.0) { 2450 multiplier = 1.0; 2451 } else if (multiplier <= 2.0) { 2452 if (2 * mFrameCount <= maxNormalFrameCount) { 2453 multiplier = 2.0; 2454 } else { 2455 multiplier = (double) maxNormalFrameCount / (double) mFrameCount; 2456 } 2457 } else { 2458 multiplier = floor(multiplier); 2459 } 2460 } 2461 mNormalFrameCount = multiplier * mFrameCount; 2462 // round up to nearest 16 frames to satisfy AudioMixer 2463 if (mType == MIXER || mType == DUPLICATING) { 2464 mNormalFrameCount = (mNormalFrameCount + 15) & ~15; 2465 } 2466 ALOGI("HAL output buffer size %zu frames, normal sink buffer size %zu frames", mFrameCount, 2467 mNormalFrameCount); 2468 2469 // Check if we want to throttle the processing to no more than 2x normal rate 2470 mThreadThrottle = property_get_bool("af.thread.throttle", true /* default_value */); 2471 mThreadThrottleTimeMs = 0; 2472 mThreadThrottleEndMs = 0; 2473 mHalfBufferMs = mNormalFrameCount * 1000 / (2 * mSampleRate); 2474 2475 // mSinkBuffer is the sink buffer. Size is always multiple-of-16 frames. 2476 // Originally this was int16_t[] array, need to remove legacy implications. 2477 free(mSinkBuffer); 2478 mSinkBuffer = NULL; 2479 // For sink buffer size, we use the frame size from the downstream sink to avoid problems 2480 // with non PCM formats for compressed music, e.g. AAC, and Offload threads. 2481 const size_t sinkBufferSize = mNormalFrameCount * mFrameSize; 2482 (void)posix_memalign(&mSinkBuffer, 32, sinkBufferSize); 2483 2484 // We resize the mMixerBuffer according to the requirements of the sink buffer which 2485 // drives the output. 2486 free(mMixerBuffer); 2487 mMixerBuffer = NULL; 2488 if (mMixerBufferEnabled) { 2489 mMixerBufferFormat = AUDIO_FORMAT_PCM_FLOAT; // also valid: AUDIO_FORMAT_PCM_16_BIT. 2490 mMixerBufferSize = mNormalFrameCount * mChannelCount 2491 * audio_bytes_per_sample(mMixerBufferFormat); 2492 (void)posix_memalign(&mMixerBuffer, 32, mMixerBufferSize); 2493 } 2494 free(mEffectBuffer); 2495 mEffectBuffer = NULL; 2496 if (mEffectBufferEnabled) { 2497 mEffectBufferFormat = AUDIO_FORMAT_PCM_16_BIT; // Note: Effects support 16b only 2498 mEffectBufferSize = mNormalFrameCount * mChannelCount 2499 * audio_bytes_per_sample(mEffectBufferFormat); 2500 (void)posix_memalign(&mEffectBuffer, 32, mEffectBufferSize); 2501 } 2502 2503 // force reconfiguration of effect chains and engines to take new buffer size and audio 2504 // parameters into account 2505 // Note that mLock is not held when readOutputParameters_l() is called from the constructor 2506 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 2507 // matter. 2508 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 2509 Vector< sp<EffectChain> > effectChains = mEffectChains; 2510 for (size_t i = 0; i < effectChains.size(); i ++) { 2511 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 2512 } 2513 } 2514 2515 2516 status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames) 2517 { 2518 if (halFrames == NULL || dspFrames == NULL) { 2519 return BAD_VALUE; 2520 } 2521 Mutex::Autolock _l(mLock); 2522 if (initCheck() != NO_ERROR) { 2523 return INVALID_OPERATION; 2524 } 2525 int64_t framesWritten = mBytesWritten / mFrameSize; 2526 *halFrames = framesWritten; 2527 2528 if (isSuspended()) { 2529 // return an estimation of rendered frames when the output is suspended 2530 size_t latencyFrames = (latency_l() * mSampleRate) / 1000; 2531 *dspFrames = (uint32_t) 2532 (framesWritten >= (int64_t)latencyFrames ? framesWritten - latencyFrames : 0); 2533 return NO_ERROR; 2534 } else { 2535 status_t status; 2536 uint32_t frames; 2537 status = mOutput->getRenderPosition(&frames); 2538 *dspFrames = (size_t)frames; 2539 return status; 2540 } 2541 } 2542 2543 // hasAudioSession_l() must be called with ThreadBase::mLock held 2544 uint32_t AudioFlinger::PlaybackThread::hasAudioSession_l(audio_session_t sessionId) const 2545 { 2546 uint32_t result = 0; 2547 if (getEffectChain_l(sessionId) != 0) { 2548 result = EFFECT_SESSION; 2549 } 2550 2551 for (size_t i = 0; i < mTracks.size(); ++i) { 2552 sp<Track> track = mTracks[i]; 2553 if (sessionId == track->sessionId() && !track->isInvalid()) { 2554 result |= TRACK_SESSION; 2555 if (track->isFastTrack()) { 2556 result |= FAST_SESSION; 2557 } 2558 break; 2559 } 2560 } 2561 2562 return result; 2563 } 2564 2565 uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(audio_session_t sessionId) 2566 { 2567 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 2568 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 2569 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 2570 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 2571 } 2572 for (size_t i = 0; i < mTracks.size(); i++) { 2573 sp<Track> track = mTracks[i]; 2574 if (sessionId == track->sessionId() && !track->isInvalid()) { 2575 return AudioSystem::getStrategyForStream(track->streamType()); 2576 } 2577 } 2578 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 2579 } 2580 2581 2582 AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 2583 { 2584 Mutex::Autolock _l(mLock); 2585 return mOutput; 2586 } 2587 2588 AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 2589 { 2590 Mutex::Autolock _l(mLock); 2591 AudioStreamOut *output = mOutput; 2592 mOutput = NULL; 2593 // FIXME FastMixer might also have a raw ptr to mOutputSink; 2594 // must push a NULL and wait for ack 2595 mOutputSink.clear(); 2596 mPipeSink.clear(); 2597 mNormalSink.clear(); 2598 return output; 2599 } 2600 2601 // this method must always be called either with ThreadBase mLock held or inside the thread loop 2602 audio_stream_t* AudioFlinger::PlaybackThread::stream() const 2603 { 2604 if (mOutput == NULL) { 2605 return NULL; 2606 } 2607 return &mOutput->stream->common; 2608 } 2609 2610 uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() const 2611 { 2612 return (uint32_t)((uint32_t)((mNormalFrameCount * 1000) / mSampleRate) * 1000); 2613 } 2614 2615 status_t AudioFlinger::PlaybackThread::setSyncEvent(const sp<SyncEvent>& event) 2616 { 2617 if (!isValidSyncEvent(event)) { 2618 return BAD_VALUE; 2619 } 2620 2621 Mutex::Autolock _l(mLock); 2622 2623 for (size_t i = 0; i < mTracks.size(); ++i) { 2624 sp<Track> track = mTracks[i]; 2625 if (event->triggerSession() == track->sessionId()) { 2626 (void) track->setSyncEvent(event); 2627 return NO_ERROR; 2628 } 2629 } 2630 2631 return NAME_NOT_FOUND; 2632 } 2633 2634 bool AudioFlinger::PlaybackThread::isValidSyncEvent(const sp<SyncEvent>& event) const 2635 { 2636 return event->type() == AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE; 2637 } 2638 2639 void AudioFlinger::PlaybackThread::threadLoop_removeTracks( 2640 const Vector< sp<Track> >& tracksToRemove) 2641 { 2642 size_t count = tracksToRemove.size(); 2643 if (count > 0) { 2644 for (size_t i = 0 ; i < count ; i++) { 2645 const sp<Track>& track = tracksToRemove.itemAt(i); 2646 if (track->isExternalTrack()) { 2647 AudioSystem::stopOutput(mId, track->streamType(), 2648 track->sessionId()); 2649 #ifdef ADD_BATTERY_DATA 2650 // to track the speaker usage 2651 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 2652 #endif 2653 if (track->isTerminated()) { 2654 AudioSystem::releaseOutput(mId, track->streamType(), 2655 track->sessionId()); 2656 } 2657 } 2658 } 2659 } 2660 } 2661 2662 void AudioFlinger::PlaybackThread::checkSilentMode_l() 2663 { 2664 if (!mMasterMute) { 2665 char value[PROPERTY_VALUE_MAX]; 2666 if (mOutDevice == AUDIO_DEVICE_OUT_REMOTE_SUBMIX) { 2667 ALOGD("ro.audio.silent will be ignored for threads on AUDIO_DEVICE_OUT_REMOTE_SUBMIX"); 2668 return; 2669 } 2670 if (property_get("ro.audio.silent", value, "0") > 0) { 2671 char *endptr; 2672 unsigned long ul = strtoul(value, &endptr, 0); 2673 if (*endptr == '\0' && ul != 0) { 2674 ALOGD("Silence is golden"); 2675 // The setprop command will not allow a property to be changed after 2676 // the first time it is set, so we don't have to worry about un-muting. 2677 setMasterMute_l(true); 2678 } 2679 } 2680 } 2681 } 2682 2683 // shared by MIXER and DIRECT, overridden by DUPLICATING 2684 ssize_t AudioFlinger::PlaybackThread::threadLoop_write() 2685 { 2686 mInWrite = true; 2687 ssize_t bytesWritten; 2688 const size_t offset = mCurrentWriteLength - mBytesRemaining; 2689 2690 // If an NBAIO sink is present, use it to write the normal mixer's submix 2691 if (mNormalSink != 0) { 2692 2693 const size_t count = mBytesRemaining / mFrameSize; 2694 2695 ATRACE_BEGIN("write"); 2696 // update the setpoint when AudioFlinger::mScreenState changes 2697 uint32_t screenState = AudioFlinger::mScreenState; 2698 if (screenState != mScreenState) { 2699 mScreenState = screenState; 2700 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 2701 if (pipe != NULL) { 2702 pipe->setAvgFrames((mScreenState & 1) ? 2703 (pipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2704 } 2705 } 2706 ssize_t framesWritten = mNormalSink->write((char *)mSinkBuffer + offset, count); 2707 ATRACE_END(); 2708 if (framesWritten > 0) { 2709 bytesWritten = framesWritten * mFrameSize; 2710 } else { 2711 bytesWritten = framesWritten; 2712 } 2713 // otherwise use the HAL / AudioStreamOut directly 2714 } else { 2715 // Direct output and offload threads 2716 2717 if (mUseAsyncWrite) { 2718 ALOGW_IF(mWriteAckSequence & 1, "threadLoop_write(): out of sequence write request"); 2719 mWriteAckSequence += 2; 2720 mWriteAckSequence |= 1; 2721 ALOG_ASSERT(mCallbackThread != 0); 2722 mCallbackThread->setWriteBlocked(mWriteAckSequence); 2723 } 2724 // FIXME We should have an implementation of timestamps for direct output threads. 2725 // They are used e.g for multichannel PCM playback over HDMI. 2726 bytesWritten = mOutput->write((char *)mSinkBuffer + offset, mBytesRemaining); 2727 2728 if (mUseAsyncWrite && 2729 ((bytesWritten < 0) || (bytesWritten == (ssize_t)mBytesRemaining))) { 2730 // do not wait for async callback in case of error of full write 2731 mWriteAckSequence &= ~1; 2732 ALOG_ASSERT(mCallbackThread != 0); 2733 mCallbackThread->setWriteBlocked(mWriteAckSequence); 2734 } 2735 } 2736 2737 mNumWrites++; 2738 mInWrite = false; 2739 mStandby = false; 2740 return bytesWritten; 2741 } 2742 2743 void AudioFlinger::PlaybackThread::threadLoop_drain() 2744 { 2745 if (mOutput->stream->drain) { 2746 ALOGV("draining %s", (mMixerStatus == MIXER_DRAIN_TRACK) ? "early" : "full"); 2747 if (mUseAsyncWrite) { 2748 ALOGW_IF(mDrainSequence & 1, "threadLoop_drain(): out of sequence drain request"); 2749 mDrainSequence |= 1; 2750 ALOG_ASSERT(mCallbackThread != 0); 2751 mCallbackThread->setDraining(mDrainSequence); 2752 } 2753 mOutput->stream->drain(mOutput->stream, 2754 (mMixerStatus == MIXER_DRAIN_TRACK) ? AUDIO_DRAIN_EARLY_NOTIFY 2755 : AUDIO_DRAIN_ALL); 2756 } 2757 } 2758 2759 void AudioFlinger::PlaybackThread::threadLoop_exit() 2760 { 2761 { 2762 Mutex::Autolock _l(mLock); 2763 for (size_t i = 0; i < mTracks.size(); i++) { 2764 sp<Track> track = mTracks[i]; 2765 track->invalidate(); 2766 } 2767 } 2768 } 2769 2770 /* 2771 The derived values that are cached: 2772 - mSinkBufferSize from frame count * frame size 2773 - mActiveSleepTimeUs from activeSleepTimeUs() 2774 - mIdleSleepTimeUs from idleSleepTimeUs() 2775 - mStandbyDelayNs from mActiveSleepTimeUs (DIRECT only) or forced to at least 2776 kDefaultStandbyTimeInNsecs when connected to an A2DP device. 2777 - maxPeriod from frame count and sample rate (MIXER only) 2778 2779 The parameters that affect these derived values are: 2780 - frame count 2781 - frame size 2782 - sample rate 2783 - device type: A2DP or not 2784 - device latency 2785 - format: PCM or not 2786 - active sleep time 2787 - idle sleep time 2788 */ 2789 2790 void AudioFlinger::PlaybackThread::cacheParameters_l() 2791 { 2792 mSinkBufferSize = mNormalFrameCount * mFrameSize; 2793 mActiveSleepTimeUs = activeSleepTimeUs(); 2794 mIdleSleepTimeUs = idleSleepTimeUs(); 2795 2796 // make sure standby delay is not too short when connected to an A2DP sink to avoid 2797 // truncating audio when going to standby. 2798 mStandbyDelayNs = AudioFlinger::mStandbyTimeInNsecs; 2799 if ((mOutDevice & AUDIO_DEVICE_OUT_ALL_A2DP) != 0) { 2800 if (mStandbyDelayNs < kDefaultStandbyTimeInNsecs) { 2801 mStandbyDelayNs = kDefaultStandbyTimeInNsecs; 2802 } 2803 } 2804 } 2805 2806 bool AudioFlinger::PlaybackThread::invalidateTracks_l(audio_stream_type_t streamType) 2807 { 2808 ALOGV("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %zu", 2809 this, streamType, mTracks.size()); 2810 bool trackMatch = false; 2811 size_t size = mTracks.size(); 2812 for (size_t i = 0; i < size; i++) { 2813 sp<Track> t = mTracks[i]; 2814 if (t->streamType() == streamType && t->isExternalTrack()) { 2815 t->invalidate(); 2816 trackMatch = true; 2817 } 2818 } 2819 return trackMatch; 2820 } 2821 2822 void AudioFlinger::PlaybackThread::invalidateTracks(audio_stream_type_t streamType) 2823 { 2824 Mutex::Autolock _l(mLock); 2825 invalidateTracks_l(streamType); 2826 } 2827 2828 status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 2829 { 2830 audio_session_t session = chain->sessionId(); 2831 int16_t* buffer = reinterpret_cast<int16_t*>(mEffectBufferEnabled 2832 ? mEffectBuffer : mSinkBuffer); 2833 bool ownsBuffer = false; 2834 2835 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 2836 if (session > AUDIO_SESSION_OUTPUT_MIX) { 2837 // Only one effect chain can be present in direct output thread and it uses 2838 // the sink buffer as input 2839 if (mType != DIRECT) { 2840 size_t numSamples = mNormalFrameCount * mChannelCount; 2841 buffer = new int16_t[numSamples]; 2842 memset(buffer, 0, numSamples * sizeof(int16_t)); 2843 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 2844 ownsBuffer = true; 2845 } 2846 2847 // Attach all tracks with same session ID to this chain. 2848 for (size_t i = 0; i < mTracks.size(); ++i) { 2849 sp<Track> track = mTracks[i]; 2850 if (session == track->sessionId()) { 2851 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), 2852 buffer); 2853 track->setMainBuffer(buffer); 2854 chain->incTrackCnt(); 2855 } 2856 } 2857 2858 // indicate all active tracks in the chain 2859 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 2860 sp<Track> track = mActiveTracks[i].promote(); 2861 if (track == 0) { 2862 continue; 2863 } 2864 if (session == track->sessionId()) { 2865 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 2866 chain->incActiveTrackCnt(); 2867 } 2868 } 2869 } 2870 chain->setThread(this); 2871 chain->setInBuffer(buffer, ownsBuffer); 2872 chain->setOutBuffer(reinterpret_cast<int16_t*>(mEffectBufferEnabled 2873 ? mEffectBuffer : mSinkBuffer)); 2874 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 2875 // chains list in order to be processed last as it contains output stage effects. 2876 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 2877 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 2878 // after track specific effects and before output stage. 2879 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 2880 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX. 2881 // Effect chain for other sessions are inserted at beginning of effect 2882 // chains list to be processed before output mix effects. Relative order between other 2883 // sessions is not important. 2884 static_assert(AUDIO_SESSION_OUTPUT_MIX == 0 && 2885 AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX, 2886 "audio_session_t constants misdefined"); 2887 size_t size = mEffectChains.size(); 2888 size_t i = 0; 2889 for (i = 0; i < size; i++) { 2890 if (mEffectChains[i]->sessionId() < session) { 2891 break; 2892 } 2893 } 2894 mEffectChains.insertAt(chain, i); 2895 checkSuspendOnAddEffectChain_l(chain); 2896 2897 return NO_ERROR; 2898 } 2899 2900 size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 2901 { 2902 audio_session_t session = chain->sessionId(); 2903 2904 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 2905 2906 for (size_t i = 0; i < mEffectChains.size(); i++) { 2907 if (chain == mEffectChains[i]) { 2908 mEffectChains.removeAt(i); 2909 // detach all active tracks from the chain 2910 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 2911 sp<Track> track = mActiveTracks[i].promote(); 2912 if (track == 0) { 2913 continue; 2914 } 2915 if (session == track->sessionId()) { 2916 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 2917 chain.get(), session); 2918 chain->decActiveTrackCnt(); 2919 } 2920 } 2921 2922 // detach all tracks with same session ID from this chain 2923 for (size_t i = 0; i < mTracks.size(); ++i) { 2924 sp<Track> track = mTracks[i]; 2925 if (session == track->sessionId()) { 2926 track->setMainBuffer(reinterpret_cast<int16_t*>(mSinkBuffer)); 2927 chain->decTrackCnt(); 2928 } 2929 } 2930 break; 2931 } 2932 } 2933 return mEffectChains.size(); 2934 } 2935 2936 status_t AudioFlinger::PlaybackThread::attachAuxEffect( 2937 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2938 { 2939 Mutex::Autolock _l(mLock); 2940 return attachAuxEffect_l(track, EffectId); 2941 } 2942 2943 status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 2944 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2945 { 2946 status_t status = NO_ERROR; 2947 2948 if (EffectId == 0) { 2949 track->setAuxBuffer(0, NULL); 2950 } else { 2951 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 2952 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 2953 if (effect != 0) { 2954 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 2955 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 2956 } else { 2957 status = INVALID_OPERATION; 2958 } 2959 } else { 2960 status = BAD_VALUE; 2961 } 2962 } 2963 return status; 2964 } 2965 2966 void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 2967 { 2968 for (size_t i = 0; i < mTracks.size(); ++i) { 2969 sp<Track> track = mTracks[i]; 2970 if (track->auxEffectId() == effectId) { 2971 attachAuxEffect_l(track, 0); 2972 } 2973 } 2974 } 2975 2976 bool AudioFlinger::PlaybackThread::threadLoop() 2977 { 2978 Vector< sp<Track> > tracksToRemove; 2979 2980 mStandbyTimeNs = systemTime(); 2981 nsecs_t lastWriteFinished = -1; // time last server write completed 2982 int64_t lastFramesWritten = -1; // track changes in timestamp server frames written 2983 2984 // MIXER 2985 nsecs_t lastWarning = 0; 2986 2987 // DUPLICATING 2988 // FIXME could this be made local to while loop? 2989 writeFrames = 0; 2990 2991 int lastGeneration = 0; 2992 2993 cacheParameters_l(); 2994 mSleepTimeUs = mIdleSleepTimeUs; 2995 2996 if (mType == MIXER) { 2997 sleepTimeShift = 0; 2998 } 2999 3000 CpuStats cpuStats; 3001 const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid())); 3002 3003 acquireWakeLock(); 3004 3005 // mNBLogWriter->log can only be called while thread mutex mLock is held. 3006 // So if you need to log when mutex is unlocked, set logString to a non-NULL string, 3007 // and then that string will be logged at the next convenient opportunity. 3008 const char *logString = NULL; 3009 3010 checkSilentMode_l(); 3011 3012 while (!exitPending()) 3013 { 3014 cpuStats.sample(myName); 3015 3016 Vector< sp<EffectChain> > effectChains; 3017 3018 { // scope for mLock 3019 3020 Mutex::Autolock _l(mLock); 3021 3022 processConfigEvents_l(); 3023 3024 if (logString != NULL) { 3025 mNBLogWriter->logTimestamp(); 3026 mNBLogWriter->log(logString); 3027 logString = NULL; 3028 } 3029 3030 // Gather the framesReleased counters for all active tracks, 3031 // and associate with the sink frames written out. We need 3032 // this to convert the sink timestamp to the track timestamp. 3033 bool kernelLocationUpdate = false; 3034 if (mNormalSink != 0) { 3035 // Note: The DuplicatingThread may not have a mNormalSink. 3036 // We always fetch the timestamp here because often the downstream 3037 // sink will block while writing. 3038 ExtendedTimestamp timestamp; // use private copy to fetch 3039 (void) mNormalSink->getTimestamp(timestamp); 3040 3041 // We keep track of the last valid kernel position in case we are in underrun 3042 // and the normal mixer period is the same as the fast mixer period, or there 3043 // is some error from the HAL. 3044 if (mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL] >= 0) { 3045 mTimestamp.mPosition[ExtendedTimestamp::LOCATION_KERNEL_LASTKERNELOK] = 3046 mTimestamp.mPosition[ExtendedTimestamp::LOCATION_KERNEL]; 3047 mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL_LASTKERNELOK] = 3048 mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL]; 3049 3050 mTimestamp.mPosition[ExtendedTimestamp::LOCATION_SERVER_LASTKERNELOK] = 3051 mTimestamp.mPosition[ExtendedTimestamp::LOCATION_SERVER]; 3052 mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_SERVER_LASTKERNELOK] = 3053 mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_SERVER]; 3054 } 3055 3056 if (timestamp.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL] >= 0) { 3057 kernelLocationUpdate = true; 3058 } else { 3059 ALOGVV("getTimestamp error - no valid kernel position"); 3060 } 3061 3062 // copy over kernel info 3063 mTimestamp.mPosition[ExtendedTimestamp::LOCATION_KERNEL] = 3064 timestamp.mPosition[ExtendedTimestamp::LOCATION_KERNEL] 3065 + mSuspendedFrames; // add frames discarded when suspended 3066 mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL] = 3067 timestamp.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL]; 3068 } 3069 // mFramesWritten for non-offloaded tracks are contiguous 3070 // even after standby() is called. This is useful for the track frame 3071 // to sink frame mapping. 3072 bool serverLocationUpdate = false; 3073 if (mFramesWritten != lastFramesWritten) { 3074 serverLocationUpdate = true; 3075 lastFramesWritten = mFramesWritten; 3076 } 3077 // Only update timestamps if there is a meaningful change. 3078 // Either the kernel timestamp must be valid or we have written something. 3079 if (kernelLocationUpdate || serverLocationUpdate) { 3080 if (serverLocationUpdate) { 3081 // use the time before we called the HAL write - it is a bit more accurate 3082 // to when the server last read data than the current time here. 3083 // 3084 // If we haven't written anything, mLastWriteTime will be -1 3085 // and we use systemTime(). 3086 mTimestamp.mPosition[ExtendedTimestamp::LOCATION_SERVER] = mFramesWritten; 3087 mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_SERVER] = mLastWriteTime == -1 3088 ? systemTime() : mLastWriteTime; 3089 } 3090 const size_t size = mActiveTracks.size(); 3091 for (size_t i = 0; i < size; ++i) { 3092 sp<Track> t = mActiveTracks[i].promote(); 3093 if (t != 0 && !t->isFastTrack()) { 3094 t->updateTrackFrameInfo( 3095 t->mAudioTrackServerProxy->framesReleased(), 3096 mFramesWritten, 3097 mTimestamp); 3098 } 3099 } 3100 } 3101 3102 saveOutputTracks(); 3103 if (mSignalPending) { 3104 // A signal was raised while we were unlocked 3105 mSignalPending = false; 3106 } else if (waitingAsyncCallback_l()) { 3107 if (exitPending()) { 3108 break; 3109 } 3110 bool released = false; 3111 if (!keepWakeLock()) { 3112 releaseWakeLock_l(); 3113 released = true; 3114 mWakeLockUids.clear(); 3115 mActiveTracksGeneration++; 3116 } 3117 ALOGV("wait async completion"); 3118 mWaitWorkCV.wait(mLock); 3119 ALOGV("async completion/wake"); 3120 if (released) { 3121 acquireWakeLock_l(); 3122 } 3123 mStandbyTimeNs = systemTime() + mStandbyDelayNs; 3124 mSleepTimeUs = 0; 3125 3126 continue; 3127 } 3128 if ((!mActiveTracks.size() && systemTime() > mStandbyTimeNs) || 3129 isSuspended()) { 3130 // put audio hardware into standby after short delay 3131 if (shouldStandby_l()) { 3132 3133 threadLoop_standby(); 3134 3135 mStandby = true; 3136 } 3137 3138 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 3139 // we're about to wait, flush the binder command buffer 3140 IPCThreadState::self()->flushCommands(); 3141 3142 clearOutputTracks(); 3143 3144 if (exitPending()) { 3145 break; 3146 } 3147 3148 releaseWakeLock_l(); 3149 mWakeLockUids.clear(); 3150 mActiveTracksGeneration++; 3151 // wait until we have something to do... 3152 ALOGV("%s going to sleep", myName.string()); 3153 mWaitWorkCV.wait(mLock); 3154 ALOGV("%s waking up", myName.string()); 3155 acquireWakeLock_l(); 3156 3157 mMixerStatus = MIXER_IDLE; 3158 mMixerStatusIgnoringFastTracks = MIXER_IDLE; 3159 mBytesWritten = 0; 3160 mBytesRemaining = 0; 3161 checkSilentMode_l(); 3162 3163 mStandbyTimeNs = systemTime() + mStandbyDelayNs; 3164 mSleepTimeUs = mIdleSleepTimeUs; 3165 if (mType == MIXER) { 3166 sleepTimeShift = 0; 3167 } 3168 3169 continue; 3170 } 3171 } 3172 // mMixerStatusIgnoringFastTracks is also updated internally 3173 mMixerStatus = prepareTracks_l(&tracksToRemove); 3174 3175 // compare with previously applied list 3176 if (lastGeneration != mActiveTracksGeneration) { 3177 // update wakelock 3178 updateWakeLockUids_l(mWakeLockUids); 3179 lastGeneration = mActiveTracksGeneration; 3180 } 3181 3182 // prevent any changes in effect chain list and in each effect chain 3183 // during mixing and effect process as the audio buffers could be deleted 3184 // or modified if an effect is created or deleted 3185 lockEffectChains_l(effectChains); 3186 } // mLock scope ends 3187 3188 if (mBytesRemaining == 0) { 3189 mCurrentWriteLength = 0; 3190 if (mMixerStatus == MIXER_TRACKS_READY) { 3191 // threadLoop_mix() sets mCurrentWriteLength 3192 threadLoop_mix(); 3193 } else if ((mMixerStatus != MIXER_DRAIN_TRACK) 3194 && (mMixerStatus != MIXER_DRAIN_ALL)) { 3195 // threadLoop_sleepTime sets mSleepTimeUs to 0 if data 3196 // must be written to HAL 3197 threadLoop_sleepTime(); 3198 if (mSleepTimeUs == 0) { 3199 mCurrentWriteLength = mSinkBufferSize; 3200 } 3201 } 3202 // Either threadLoop_mix() or threadLoop_sleepTime() should have set 3203 // mMixerBuffer with data if mMixerBufferValid is true and mSleepTimeUs == 0. 3204 // Merge mMixerBuffer data into mEffectBuffer (if any effects are valid) 3205 // or mSinkBuffer (if there are no effects). 3206 // 3207 // This is done pre-effects computation; if effects change to 3208 // support higher precision, this needs to move. 3209 // 3210 // mMixerBufferValid is only set true by MixerThread::prepareTracks_l(). 3211 // TODO use mSleepTimeUs == 0 as an additional condition. 3212 if (mMixerBufferValid) { 3213 void *buffer = mEffectBufferValid ? mEffectBuffer : mSinkBuffer; 3214 audio_format_t format = mEffectBufferValid ? mEffectBufferFormat : mFormat; 3215 3216 // mono blend occurs for mixer threads only (not direct or offloaded) 3217 // and is handled here if we're going directly to the sink. 3218 if (requireMonoBlend() && !mEffectBufferValid) { 3219 mono_blend(mMixerBuffer, mMixerBufferFormat, mChannelCount, mNormalFrameCount, 3220 true /*limit*/); 3221 } 3222 3223 memcpy_by_audio_format(buffer, format, mMixerBuffer, mMixerBufferFormat, 3224 mNormalFrameCount * mChannelCount); 3225 } 3226 3227 mBytesRemaining = mCurrentWriteLength; 3228 if (isSuspended()) { 3229 // Simulate write to HAL when suspended (e.g. BT SCO phone call). 3230 mSleepTimeUs = suspendSleepTimeUs(); // assumes full buffer. 3231 const size_t framesRemaining = mBytesRemaining / mFrameSize; 3232 mBytesWritten += mBytesRemaining; 3233 mFramesWritten += framesRemaining; 3234 mSuspendedFrames += framesRemaining; // to adjust kernel HAL position 3235 mBytesRemaining = 0; 3236 } 3237 3238 // only process effects if we're going to write 3239 if (mSleepTimeUs == 0 && mType != OFFLOAD) { 3240 for (size_t i = 0; i < effectChains.size(); i ++) { 3241 effectChains[i]->process_l(); 3242 } 3243 } 3244 } 3245 // Process effect chains for offloaded thread even if no audio 3246 // was read from audio track: process only updates effect state 3247 // and thus does have to be synchronized with audio writes but may have 3248 // to be called while waiting for async write callback 3249 if (mType == OFFLOAD) { 3250 for (size_t i = 0; i < effectChains.size(); i ++) { 3251 effectChains[i]->process_l(); 3252 } 3253 } 3254 3255 // Only if the Effects buffer is enabled and there is data in the 3256 // Effects buffer (buffer valid), we need to 3257 // copy into the sink buffer. 3258 // TODO use mSleepTimeUs == 0 as an additional condition. 3259 if (mEffectBufferValid) { 3260 //ALOGV("writing effect buffer to sink buffer format %#x", mFormat); 3261 3262 if (requireMonoBlend()) { 3263 mono_blend(mEffectBuffer, mEffectBufferFormat, mChannelCount, mNormalFrameCount, 3264 true /*limit*/); 3265 } 3266 3267 memcpy_by_audio_format(mSinkBuffer, mFormat, mEffectBuffer, mEffectBufferFormat, 3268 mNormalFrameCount * mChannelCount); 3269 } 3270 3271 // enable changes in effect chain 3272 unlockEffectChains(effectChains); 3273 3274 if (!waitingAsyncCallback()) { 3275 // mSleepTimeUs == 0 means we must write to audio hardware 3276 if (mSleepTimeUs == 0) { 3277 ssize_t ret = 0; 3278 // We save lastWriteFinished here, as previousLastWriteFinished, 3279 // for throttling. On thread start, previousLastWriteFinished will be 3280 // set to -1, which properly results in no throttling after the first write. 3281 nsecs_t previousLastWriteFinished = lastWriteFinished; 3282 nsecs_t delta = 0; 3283 if (mBytesRemaining) { 3284 // FIXME rewrite to reduce number of system calls 3285 mLastWriteTime = systemTime(); // also used for dumpsys 3286 ret = threadLoop_write(); 3287 lastWriteFinished = systemTime(); 3288 delta = lastWriteFinished - mLastWriteTime; 3289 if (ret < 0) { 3290 mBytesRemaining = 0; 3291 } else { 3292 mBytesWritten += ret; 3293 mBytesRemaining -= ret; 3294 mFramesWritten += ret / mFrameSize; 3295 } 3296 } else if ((mMixerStatus == MIXER_DRAIN_TRACK) || 3297 (mMixerStatus == MIXER_DRAIN_ALL)) { 3298 threadLoop_drain(); 3299 } 3300 if (mType == MIXER && !mStandby) { 3301 // write blocked detection 3302 if (delta > maxPeriod) { 3303 mNumDelayedWrites++; 3304 if ((lastWriteFinished - lastWarning) > kWarningThrottleNs) { 3305 ATRACE_NAME("underrun"); 3306 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 3307 (unsigned long long) ns2ms(delta), mNumDelayedWrites, this); 3308 lastWarning = lastWriteFinished; 3309 } 3310 } 3311 3312 if (mThreadThrottle 3313 && mMixerStatus == MIXER_TRACKS_READY // we are mixing (active tracks) 3314 && ret > 0) { // we wrote something 3315 // Limit MixerThread data processing to no more than twice the 3316 // expected processing rate. 3317 // 3318 // This helps prevent underruns with NuPlayer and other applications 3319 // which may set up buffers that are close to the minimum size, or use 3320 // deep buffers, and rely on a double-buffering sleep strategy to fill. 3321 // 3322 // The throttle smooths out sudden large data drains from the device, 3323 // e.g. when it comes out of standby, which often causes problems with 3324 // (1) mixer threads without a fast mixer (which has its own warm-up) 3325 // (2) minimum buffer sized tracks (even if the track is full, 3326 // the app won't fill fast enough to handle the sudden draw). 3327 // 3328 // Total time spent in last processing cycle equals time spent in 3329 // 1. threadLoop_write, as well as time spent in 3330 // 2. threadLoop_mix (significant for heavy mixing, especially 3331 // on low tier processors) 3332 3333 // it's OK if deltaMs is an overestimate. 3334 const int32_t deltaMs = 3335 (lastWriteFinished - previousLastWriteFinished) / 1000000; 3336 const int32_t throttleMs = mHalfBufferMs - deltaMs; 3337 if ((signed)mHalfBufferMs >= throttleMs && throttleMs > 0) { 3338 usleep(throttleMs * 1000); 3339 // notify of throttle start on verbose log 3340 ALOGV_IF(mThreadThrottleEndMs == mThreadThrottleTimeMs, 3341 "mixer(%p) throttle begin:" 3342 " ret(%zd) deltaMs(%d) requires sleep %d ms", 3343 this, ret, deltaMs, throttleMs); 3344 mThreadThrottleTimeMs += throttleMs; 3345 // Throttle must be attributed to the previous mixer loop's write time 3346 // to allow back-to-back throttling. 3347 lastWriteFinished += throttleMs * 1000000; 3348 } else { 3349 uint32_t diff = mThreadThrottleTimeMs - mThreadThrottleEndMs; 3350 if (diff > 0) { 3351 // notify of throttle end on debug log 3352 // but prevent spamming for bluetooth 3353 ALOGD_IF(!audio_is_a2dp_out_device(outDevice()), 3354 "mixer(%p) throttle end: throttle time(%u)", this, diff); 3355 mThreadThrottleEndMs = mThreadThrottleTimeMs; 3356 } 3357 } 3358 } 3359 } 3360 3361 } else { 3362 ATRACE_BEGIN("sleep"); 3363 Mutex::Autolock _l(mLock); 3364 if (!mSignalPending && mConfigEvents.isEmpty() && !exitPending()) { 3365 mWaitWorkCV.waitRelative(mLock, microseconds((nsecs_t)mSleepTimeUs)); 3366 } 3367 ATRACE_END(); 3368 } 3369 } 3370 3371 // Finally let go of removed track(s), without the lock held 3372 // since we can't guarantee the destructors won't acquire that 3373 // same lock. This will also mutate and push a new fast mixer state. 3374 threadLoop_removeTracks(tracksToRemove); 3375 tracksToRemove.clear(); 3376 3377 // FIXME I don't understand the need for this here; 3378 // it was in the original code but maybe the 3379 // assignment in saveOutputTracks() makes this unnecessary? 3380 clearOutputTracks(); 3381 3382 // Effect chains will be actually deleted here if they were removed from 3383 // mEffectChains list during mixing or effects processing 3384 effectChains.clear(); 3385 3386 // FIXME Note that the above .clear() is no longer necessary since effectChains 3387 // is now local to this block, but will keep it for now (at least until merge done). 3388 } 3389 3390 threadLoop_exit(); 3391 3392 if (!mStandby) { 3393 threadLoop_standby(); 3394 mStandby = true; 3395 } 3396 3397 releaseWakeLock(); 3398 mWakeLockUids.clear(); 3399 mActiveTracksGeneration++; 3400 3401 ALOGV("Thread %p type %d exiting", this, mType); 3402 return false; 3403 } 3404 3405 // removeTracks_l() must be called with ThreadBase::mLock held 3406 void AudioFlinger::PlaybackThread::removeTracks_l(const Vector< sp<Track> >& tracksToRemove) 3407 { 3408 size_t count = tracksToRemove.size(); 3409 if (count > 0) { 3410 for (size_t i=0 ; i<count ; i++) { 3411 const sp<Track>& track = tracksToRemove.itemAt(i); 3412 mActiveTracks.remove(track); 3413 mWakeLockUids.remove(track->uid()); 3414 mActiveTracksGeneration++; 3415 ALOGV("removeTracks_l removing track on session %d", track->sessionId()); 3416 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 3417 if (chain != 0) { 3418 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), 3419 track->sessionId()); 3420 chain->decActiveTrackCnt(); 3421 } 3422 if (track->isTerminated()) { 3423 removeTrack_l(track); 3424 } 3425 } 3426 } 3427 3428 } 3429 3430 status_t AudioFlinger::PlaybackThread::getTimestamp_l(AudioTimestamp& timestamp) 3431 { 3432 if (mNormalSink != 0) { 3433 ExtendedTimestamp ets; 3434 status_t status = mNormalSink->getTimestamp(ets); 3435 if (status == NO_ERROR) { 3436 status = ets.getBestTimestamp(×tamp); 3437 } 3438 return status; 3439 } 3440 if ((mType == OFFLOAD || mType == DIRECT) 3441 && mOutput != NULL && mOutput->stream->get_presentation_position) { 3442 uint64_t position64; 3443 int ret = mOutput->getPresentationPosition(&position64, ×tamp.mTime); 3444 if (ret == 0) { 3445 timestamp.mPosition = (uint32_t)position64; 3446 return NO_ERROR; 3447 } 3448 } 3449 return INVALID_OPERATION; 3450 } 3451 3452 status_t AudioFlinger::MixerThread::createAudioPatch_l(const struct audio_patch *patch, 3453 audio_patch_handle_t *handle) 3454 { 3455 status_t status; 3456 if (property_get_bool("af.patch_park", false /* default_value */)) { 3457 // Park FastMixer to avoid potential DOS issues with writing to the HAL 3458 // or if HAL does not properly lock against access. 3459 AutoPark<FastMixer> park(mFastMixer); 3460 status = PlaybackThread::createAudioPatch_l(patch, handle); 3461 } else { 3462 status = PlaybackThread::createAudioPatch_l(patch, handle); 3463 } 3464 return status; 3465 } 3466 3467 status_t AudioFlinger::PlaybackThread::createAudioPatch_l(const struct audio_patch *patch, 3468 audio_patch_handle_t *handle) 3469 { 3470 status_t status = NO_ERROR; 3471 3472 // store new device and send to effects 3473 audio_devices_t type = AUDIO_DEVICE_NONE; 3474 for (unsigned int i = 0; i < patch->num_sinks; i++) { 3475 type |= patch->sinks[i].ext.device.type; 3476 } 3477 3478 #ifdef ADD_BATTERY_DATA 3479 // when changing the audio output device, call addBatteryData to notify 3480 // the change 3481 if (mOutDevice != type) { 3482 uint32_t params = 0; 3483 // check whether speaker is on 3484 if (type & AUDIO_DEVICE_OUT_SPEAKER) { 3485 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 3486 } 3487 3488 audio_devices_t deviceWithoutSpeaker 3489 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 3490 // check if any other device (except speaker) is on 3491 if (type & deviceWithoutSpeaker) { 3492 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 3493 } 3494 3495 if (params != 0) { 3496 addBatteryData(params); 3497 } 3498 } 3499 #endif 3500 3501 for (size_t i = 0; i < mEffectChains.size(); i++) { 3502 mEffectChains[i]->setDevice_l(type); 3503 } 3504 3505 // mPrevOutDevice is the latest device set by createAudioPatch_l(). It is not set when 3506 // the thread is created so that the first patch creation triggers an ioConfigChanged callback 3507 bool configChanged = mPrevOutDevice != type; 3508 mOutDevice = type; 3509 mPatch = *patch; 3510 3511 if (mOutput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 3512 audio_hw_device_t *hwDevice = mOutput->audioHwDev->hwDevice(); 3513 status = hwDevice->create_audio_patch(hwDevice, 3514 patch->num_sources, 3515 patch->sources, 3516 patch->num_sinks, 3517 patch->sinks, 3518 handle); 3519 } else { 3520 char *address; 3521 if (strcmp(patch->sinks[0].ext.device.address, "") != 0) { 3522 //FIXME: we only support address on first sink with HAL version < 3.0 3523 address = audio_device_address_to_parameter( 3524 patch->sinks[0].ext.device.type, 3525 patch->sinks[0].ext.device.address); 3526 } else { 3527 address = (char *)calloc(1, 1); 3528 } 3529 AudioParameter param = AudioParameter(String8(address)); 3530 free(address); 3531 param.addInt(String8(AUDIO_PARAMETER_STREAM_ROUTING), (int)type); 3532 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3533 param.toString().string()); 3534 *handle = AUDIO_PATCH_HANDLE_NONE; 3535 } 3536 if (configChanged) { 3537 mPrevOutDevice = type; 3538 sendIoConfigEvent_l(AUDIO_OUTPUT_CONFIG_CHANGED); 3539 } 3540 return status; 3541 } 3542 3543 status_t AudioFlinger::MixerThread::releaseAudioPatch_l(const audio_patch_handle_t handle) 3544 { 3545 status_t status; 3546 if (property_get_bool("af.patch_park", false /* default_value */)) { 3547 // Park FastMixer to avoid potential DOS issues with writing to the HAL 3548 // or if HAL does not properly lock against access. 3549 AutoPark<FastMixer> park(mFastMixer); 3550 status = PlaybackThread::releaseAudioPatch_l(handle); 3551 } else { 3552 status = PlaybackThread::releaseAudioPatch_l(handle); 3553 } 3554 return status; 3555 } 3556 3557 status_t AudioFlinger::PlaybackThread::releaseAudioPatch_l(const audio_patch_handle_t handle) 3558 { 3559 status_t status = NO_ERROR; 3560 3561 mOutDevice = AUDIO_DEVICE_NONE; 3562 3563 if (mOutput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 3564 audio_hw_device_t *hwDevice = mOutput->audioHwDev->hwDevice(); 3565 status = hwDevice->release_audio_patch(hwDevice, handle); 3566 } else { 3567 AudioParameter param; 3568 param.addInt(String8(AUDIO_PARAMETER_STREAM_ROUTING), 0); 3569 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3570 param.toString().string()); 3571 } 3572 return status; 3573 } 3574 3575 void AudioFlinger::PlaybackThread::addPatchTrack(const sp<PatchTrack>& track) 3576 { 3577 Mutex::Autolock _l(mLock); 3578 mTracks.add(track); 3579 } 3580 3581 void AudioFlinger::PlaybackThread::deletePatchTrack(const sp<PatchTrack>& track) 3582 { 3583 Mutex::Autolock _l(mLock); 3584 destroyTrack_l(track); 3585 } 3586 3587 void AudioFlinger::PlaybackThread::getAudioPortConfig(struct audio_port_config *config) 3588 { 3589 ThreadBase::getAudioPortConfig(config); 3590 config->role = AUDIO_PORT_ROLE_SOURCE; 3591 config->ext.mix.hw_module = mOutput->audioHwDev->handle(); 3592 config->ext.mix.usecase.stream = AUDIO_STREAM_DEFAULT; 3593 } 3594 3595 // ---------------------------------------------------------------------------- 3596 3597 AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 3598 audio_io_handle_t id, audio_devices_t device, bool systemReady, type_t type) 3599 : PlaybackThread(audioFlinger, output, id, device, type, systemReady), 3600 // mAudioMixer below 3601 // mFastMixer below 3602 mFastMixerFutex(0), 3603 mMasterMono(false) 3604 // mOutputSink below 3605 // mPipeSink below 3606 // mNormalSink below 3607 { 3608 ALOGV("MixerThread() id=%d device=%#x type=%d", id, device, type); 3609 ALOGV("mSampleRate=%u, mChannelMask=%#x, mChannelCount=%u, mFormat=%d, mFrameSize=%zu, " 3610 "mFrameCount=%zu, mNormalFrameCount=%zu", 3611 mSampleRate, mChannelMask, mChannelCount, mFormat, mFrameSize, mFrameCount, 3612 mNormalFrameCount); 3613 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 3614 3615 if (type == DUPLICATING) { 3616 // The Duplicating thread uses the AudioMixer and delivers data to OutputTracks 3617 // (downstream MixerThreads) in DuplicatingThread::threadLoop_write(). 3618 // Do not create or use mFastMixer, mOutputSink, mPipeSink, or mNormalSink. 3619 return; 3620 } 3621 // create an NBAIO sink for the HAL output stream, and negotiate 3622 mOutputSink = new AudioStreamOutSink(output->stream); 3623 size_t numCounterOffers = 0; 3624 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount, mFormat)}; 3625 #if !LOG_NDEBUG 3626 ssize_t index = 3627 #else 3628 (void) 3629 #endif 3630 mOutputSink->negotiate(offers, 1, NULL, numCounterOffers); 3631 ALOG_ASSERT(index == 0); 3632 3633 // initialize fast mixer depending on configuration 3634 bool initFastMixer; 3635 switch (kUseFastMixer) { 3636 case FastMixer_Never: 3637 initFastMixer = false; 3638 break; 3639 case FastMixer_Always: 3640 initFastMixer = true; 3641 break; 3642 case FastMixer_Static: 3643 case FastMixer_Dynamic: 3644 initFastMixer = mFrameCount < mNormalFrameCount; 3645 break; 3646 } 3647 if (initFastMixer) { 3648 audio_format_t fastMixerFormat; 3649 if (mMixerBufferEnabled && mEffectBufferEnabled) { 3650 fastMixerFormat = AUDIO_FORMAT_PCM_FLOAT; 3651 } else { 3652 fastMixerFormat = AUDIO_FORMAT_PCM_16_BIT; 3653 } 3654 if (mFormat != fastMixerFormat) { 3655 // change our Sink format to accept our intermediate precision 3656 mFormat = fastMixerFormat; 3657 free(mSinkBuffer); 3658 mFrameSize = mChannelCount * audio_bytes_per_sample(mFormat); 3659 const size_t sinkBufferSize = mNormalFrameCount * mFrameSize; 3660 (void)posix_memalign(&mSinkBuffer, 32, sinkBufferSize); 3661 } 3662 3663 // create a MonoPipe to connect our submix to FastMixer 3664 NBAIO_Format format = mOutputSink->format(); 3665 #ifdef TEE_SINK 3666 NBAIO_Format origformat = format; 3667 #endif 3668 // adjust format to match that of the Fast Mixer 3669 ALOGV("format changed from %d to %d", format.mFormat, fastMixerFormat); 3670 format.mFormat = fastMixerFormat; 3671 format.mFrameSize = audio_bytes_per_sample(format.mFormat) * format.mChannelCount; 3672 3673 // This pipe depth compensates for scheduling latency of the normal mixer thread. 3674 // When it wakes up after a maximum latency, it runs a few cycles quickly before 3675 // finally blocking. Note the pipe implementation rounds up the request to a power of 2. 3676 MonoPipe *monoPipe = new MonoPipe(mNormalFrameCount * 4, format, true /*writeCanBlock*/); 3677 const NBAIO_Format offers[1] = {format}; 3678 size_t numCounterOffers = 0; 3679 #if !LOG_NDEBUG || defined(TEE_SINK) 3680 ssize_t index = 3681 #else 3682 (void) 3683 #endif 3684 monoPipe->negotiate(offers, 1, NULL, numCounterOffers); 3685 ALOG_ASSERT(index == 0); 3686 monoPipe->setAvgFrames((mScreenState & 1) ? 3687 (monoPipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 3688 mPipeSink = monoPipe; 3689 3690 #ifdef TEE_SINK 3691 if (mTeeSinkOutputEnabled) { 3692 // create a Pipe to archive a copy of FastMixer's output for dumpsys 3693 Pipe *teeSink = new Pipe(mTeeSinkOutputFrames, origformat); 3694 const NBAIO_Format offers2[1] = {origformat}; 3695 numCounterOffers = 0; 3696 index = teeSink->negotiate(offers2, 1, NULL, numCounterOffers); 3697 ALOG_ASSERT(index == 0); 3698 mTeeSink = teeSink; 3699 PipeReader *teeSource = new PipeReader(*teeSink); 3700 numCounterOffers = 0; 3701 index = teeSource->negotiate(offers2, 1, NULL, numCounterOffers); 3702 ALOG_ASSERT(index == 0); 3703 mTeeSource = teeSource; 3704 } 3705 #endif 3706 3707 // create fast mixer and configure it initially with just one fast track for our submix 3708 mFastMixer = new FastMixer(); 3709 FastMixerStateQueue *sq = mFastMixer->sq(); 3710 #ifdef STATE_QUEUE_DUMP 3711 sq->setObserverDump(&mStateQueueObserverDump); 3712 sq->setMutatorDump(&mStateQueueMutatorDump); 3713 #endif 3714 FastMixerState *state = sq->begin(); 3715 FastTrack *fastTrack = &state->mFastTracks[0]; 3716 // wrap the source side of the MonoPipe to make it an AudioBufferProvider 3717 fastTrack->mBufferProvider = new SourceAudioBufferProvider(new MonoPipeReader(monoPipe)); 3718 fastTrack->mVolumeProvider = NULL; 3719 fastTrack->mChannelMask = mChannelMask; // mPipeSink channel mask for audio to FastMixer 3720 fastTrack->mFormat = mFormat; // mPipeSink format for audio to FastMixer 3721 fastTrack->mGeneration++; 3722 state->mFastTracksGen++; 3723 state->mTrackMask = 1; 3724 // fast mixer will use the HAL output sink 3725 state->mOutputSink = mOutputSink.get(); 3726 state->mOutputSinkGen++; 3727 state->mFrameCount = mFrameCount; 3728 state->mCommand = FastMixerState::COLD_IDLE; 3729 // already done in constructor initialization list 3730 //mFastMixerFutex = 0; 3731 state->mColdFutexAddr = &mFastMixerFutex; 3732 state->mColdGen++; 3733 state->mDumpState = &mFastMixerDumpState; 3734 #ifdef TEE_SINK 3735 state->mTeeSink = mTeeSink.get(); 3736 #endif 3737 mFastMixerNBLogWriter = audioFlinger->newWriter_l(kFastMixerLogSize, "FastMixer"); 3738 state->mNBLogWriter = mFastMixerNBLogWriter.get(); 3739 sq->end(); 3740 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3741 3742 // start the fast mixer 3743 mFastMixer->run("FastMixer", PRIORITY_URGENT_AUDIO); 3744 pid_t tid = mFastMixer->getTid(); 3745 sendPrioConfigEvent(getpid_cached, tid, kPriorityFastMixer); 3746 3747 #ifdef AUDIO_WATCHDOG 3748 // create and start the watchdog 3749 mAudioWatchdog = new AudioWatchdog(); 3750 mAudioWatchdog->setDump(&mAudioWatchdogDump); 3751 mAudioWatchdog->run("AudioWatchdog", PRIORITY_URGENT_AUDIO); 3752 tid = mAudioWatchdog->getTid(); 3753 sendPrioConfigEvent(getpid_cached, tid, kPriorityFastMixer); 3754 #endif 3755 3756 } 3757 3758 switch (kUseFastMixer) { 3759 case FastMixer_Never: 3760 case FastMixer_Dynamic: 3761 mNormalSink = mOutputSink; 3762 break; 3763 case FastMixer_Always: 3764 mNormalSink = mPipeSink; 3765 break; 3766 case FastMixer_Static: 3767 mNormalSink = initFastMixer ? mPipeSink : mOutputSink; 3768 break; 3769 } 3770 } 3771 3772 AudioFlinger::MixerThread::~MixerThread() 3773 { 3774 if (mFastMixer != 0) { 3775 FastMixerStateQueue *sq = mFastMixer->sq(); 3776 FastMixerState *state = sq->begin(); 3777 if (state->mCommand == FastMixerState::COLD_IDLE) { 3778 int32_t old = android_atomic_inc(&mFastMixerFutex); 3779 if (old == -1) { 3780 (void) syscall(__NR_futex, &mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 3781 } 3782 } 3783 state->mCommand = FastMixerState::EXIT; 3784 sq->end(); 3785 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3786 mFastMixer->join(); 3787 // Though the fast mixer thread has exited, it's state queue is still valid. 3788 // We'll use that extract the final state which contains one remaining fast track 3789 // corresponding to our sub-mix. 3790 state = sq->begin(); 3791 ALOG_ASSERT(state->mTrackMask == 1); 3792 FastTrack *fastTrack = &state->mFastTracks[0]; 3793 ALOG_ASSERT(fastTrack->mBufferProvider != NULL); 3794 delete fastTrack->mBufferProvider; 3795 sq->end(false /*didModify*/); 3796 mFastMixer.clear(); 3797 #ifdef AUDIO_WATCHDOG 3798 if (mAudioWatchdog != 0) { 3799 mAudioWatchdog->requestExit(); 3800 mAudioWatchdog->requestExitAndWait(); 3801 mAudioWatchdog.clear(); 3802 } 3803 #endif 3804 } 3805 mAudioFlinger->unregisterWriter(mFastMixerNBLogWriter); 3806 delete mAudioMixer; 3807 } 3808 3809 3810 uint32_t AudioFlinger::MixerThread::correctLatency_l(uint32_t latency) const 3811 { 3812 if (mFastMixer != 0) { 3813 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 3814 latency += (pipe->getAvgFrames() * 1000) / mSampleRate; 3815 } 3816 return latency; 3817 } 3818 3819 3820 void AudioFlinger::MixerThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) 3821 { 3822 PlaybackThread::threadLoop_removeTracks(tracksToRemove); 3823 } 3824 3825 ssize_t AudioFlinger::MixerThread::threadLoop_write() 3826 { 3827 // FIXME we should only do one push per cycle; confirm this is true 3828 // Start the fast mixer if it's not already running 3829 if (mFastMixer != 0) { 3830 FastMixerStateQueue *sq = mFastMixer->sq(); 3831 FastMixerState *state = sq->begin(); 3832 if (state->mCommand != FastMixerState::MIX_WRITE && 3833 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)) { 3834 if (state->mCommand == FastMixerState::COLD_IDLE) { 3835 3836 // FIXME workaround for first HAL write being CPU bound on some devices 3837 ATRACE_BEGIN("write"); 3838 mOutput->write((char *)mSinkBuffer, 0); 3839 ATRACE_END(); 3840 3841 int32_t old = android_atomic_inc(&mFastMixerFutex); 3842 if (old == -1) { 3843 (void) syscall(__NR_futex, &mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 3844 } 3845 #ifdef AUDIO_WATCHDOG 3846 if (mAudioWatchdog != 0) { 3847 mAudioWatchdog->resume(); 3848 } 3849 #endif 3850 } 3851 state->mCommand = FastMixerState::MIX_WRITE; 3852 #ifdef FAST_THREAD_STATISTICS 3853 mFastMixerDumpState.increaseSamplingN(mAudioFlinger->isLowRamDevice() ? 3854 FastThreadDumpState::kSamplingNforLowRamDevice : FastThreadDumpState::kSamplingN); 3855 #endif 3856 sq->end(); 3857 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3858 if (kUseFastMixer == FastMixer_Dynamic) { 3859 mNormalSink = mPipeSink; 3860 } 3861 } else { 3862 sq->end(false /*didModify*/); 3863 } 3864 } 3865 return PlaybackThread::threadLoop_write(); 3866 } 3867 3868 void AudioFlinger::MixerThread::threadLoop_standby() 3869 { 3870 // Idle the fast mixer if it's currently running 3871 if (mFastMixer != 0) { 3872 FastMixerStateQueue *sq = mFastMixer->sq(); 3873 FastMixerState *state = sq->begin(); 3874 if (!(state->mCommand & FastMixerState::IDLE)) { 3875 state->mCommand = FastMixerState::COLD_IDLE; 3876 state->mColdFutexAddr = &mFastMixerFutex; 3877 state->mColdGen++; 3878 mFastMixerFutex = 0; 3879 sq->end(); 3880 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 3881 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3882 if (kUseFastMixer == FastMixer_Dynamic) { 3883 mNormalSink = mOutputSink; 3884 } 3885 #ifdef AUDIO_WATCHDOG 3886 if (mAudioWatchdog != 0) { 3887 mAudioWatchdog->pause(); 3888 } 3889 #endif 3890 } else { 3891 sq->end(false /*didModify*/); 3892 } 3893 } 3894 PlaybackThread::threadLoop_standby(); 3895 } 3896 3897 bool AudioFlinger::PlaybackThread::waitingAsyncCallback_l() 3898 { 3899 return false; 3900 } 3901 3902 bool AudioFlinger::PlaybackThread::shouldStandby_l() 3903 { 3904 return !mStandby; 3905 } 3906 3907 bool AudioFlinger::PlaybackThread::waitingAsyncCallback() 3908 { 3909 Mutex::Autolock _l(mLock); 3910 return waitingAsyncCallback_l(); 3911 } 3912 3913 // shared by MIXER and DIRECT, overridden by DUPLICATING 3914 void AudioFlinger::PlaybackThread::threadLoop_standby() 3915 { 3916 ALOGV("Audio hardware entering standby, mixer %p, suspend count %d", this, mSuspended); 3917 mOutput->standby(); 3918 if (mUseAsyncWrite != 0) { 3919 // discard any pending drain or write ack by incrementing sequence 3920 mWriteAckSequence = (mWriteAckSequence + 2) & ~1; 3921 mDrainSequence = (mDrainSequence + 2) & ~1; 3922 ALOG_ASSERT(mCallbackThread != 0); 3923 mCallbackThread->setWriteBlocked(mWriteAckSequence); 3924 mCallbackThread->setDraining(mDrainSequence); 3925 } 3926 mHwPaused = false; 3927 } 3928 3929 void AudioFlinger::PlaybackThread::onAddNewTrack_l() 3930 { 3931 ALOGV("signal playback thread"); 3932 broadcast_l(); 3933 } 3934 3935 void AudioFlinger::PlaybackThread::onAsyncError() 3936 { 3937 for (int i = AUDIO_STREAM_SYSTEM; i < (int)AUDIO_STREAM_CNT; i++) { 3938 invalidateTracks((audio_stream_type_t)i); 3939 } 3940 } 3941 3942 void AudioFlinger::MixerThread::threadLoop_mix() 3943 { 3944 // mix buffers... 3945 mAudioMixer->process(); 3946 mCurrentWriteLength = mSinkBufferSize; 3947 // increase sleep time progressively when application underrun condition clears. 3948 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 3949 // that a steady state of alternating ready/not ready conditions keeps the sleep time 3950 // such that we would underrun the audio HAL. 3951 if ((mSleepTimeUs == 0) && (sleepTimeShift > 0)) { 3952 sleepTimeShift--; 3953 } 3954 mSleepTimeUs = 0; 3955 mStandbyTimeNs = systemTime() + mStandbyDelayNs; 3956 //TODO: delay standby when effects have a tail 3957 3958 } 3959 3960 void AudioFlinger::MixerThread::threadLoop_sleepTime() 3961 { 3962 // If no tracks are ready, sleep once for the duration of an output 3963 // buffer size, then write 0s to the output 3964 if (mSleepTimeUs == 0) { 3965 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3966 mSleepTimeUs = mActiveSleepTimeUs >> sleepTimeShift; 3967 if (mSleepTimeUs < kMinThreadSleepTimeUs) { 3968 mSleepTimeUs = kMinThreadSleepTimeUs; 3969 } 3970 // reduce sleep time in case of consecutive application underruns to avoid 3971 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 3972 // duration we would end up writing less data than needed by the audio HAL if 3973 // the condition persists. 3974 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 3975 sleepTimeShift++; 3976 } 3977 } else { 3978 mSleepTimeUs = mIdleSleepTimeUs; 3979 } 3980 } else if (mBytesWritten != 0 || (mMixerStatus == MIXER_TRACKS_ENABLED)) { 3981 // clear out mMixerBuffer or mSinkBuffer, to ensure buffers are cleared 3982 // before effects processing or output. 3983 if (mMixerBufferValid) { 3984 memset(mMixerBuffer, 0, mMixerBufferSize); 3985 } else { 3986 memset(mSinkBuffer, 0, mSinkBufferSize); 3987 } 3988 mSleepTimeUs = 0; 3989 ALOGV_IF(mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED), 3990 "anticipated start"); 3991 } 3992 // TODO add standby time extension fct of effect tail 3993 } 3994 3995 // prepareTracks_l() must be called with ThreadBase::mLock held 3996 AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 3997 Vector< sp<Track> > *tracksToRemove) 3998 { 3999 4000 mixer_state mixerStatus = MIXER_IDLE; 4001 // find out which tracks need to be processed 4002 size_t count = mActiveTracks.size(); 4003 size_t mixedTracks = 0; 4004 size_t tracksWithEffect = 0; 4005 // counts only _active_ fast tracks 4006 size_t fastTracks = 0; 4007 uint32_t resetMask = 0; // bit mask of fast tracks that need to be reset 4008 4009 float masterVolume = mMasterVolume; 4010 bool masterMute = mMasterMute; 4011 4012 if (masterMute) { 4013 masterVolume = 0; 4014 } 4015 // Delegate master volume control to effect in output mix effect chain if needed 4016 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 4017 if (chain != 0) { 4018 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 4019 chain->setVolume_l(&v, &v); 4020 masterVolume = (float)((v + (1 << 23)) >> 24); 4021 chain.clear(); 4022 } 4023 4024 // prepare a new state to push 4025 FastMixerStateQueue *sq = NULL; 4026 FastMixerState *state = NULL; 4027 bool didModify = false; 4028 FastMixerStateQueue::block_t block = FastMixerStateQueue::BLOCK_UNTIL_PUSHED; 4029 if (mFastMixer != 0) { 4030 sq = mFastMixer->sq(); 4031 state = sq->begin(); 4032 } 4033 4034 mMixerBufferValid = false; // mMixerBuffer has no valid data until appropriate tracks found. 4035 mEffectBufferValid = false; // mEffectBuffer has no valid data until tracks found. 4036 4037 for (size_t i=0 ; i<count ; i++) { 4038 const sp<Track> t = mActiveTracks[i].promote(); 4039 if (t == 0) { 4040 continue; 4041 } 4042 4043 // this const just means the local variable doesn't change 4044 Track* const track = t.get(); 4045 4046 // process fast tracks 4047 if (track->isFastTrack()) { 4048 4049 // It's theoretically possible (though unlikely) for a fast track to be created 4050 // and then removed within the same normal mix cycle. This is not a problem, as 4051 // the track never becomes active so it's fast mixer slot is never touched. 4052 // The converse, of removing an (active) track and then creating a new track 4053 // at the identical fast mixer slot within the same normal mix cycle, 4054 // is impossible because the slot isn't marked available until the end of each cycle. 4055 int j = track->mFastIndex; 4056 ALOG_ASSERT(0 < j && j < (int)FastMixerState::sMaxFastTracks); 4057 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << j))); 4058 FastTrack *fastTrack = &state->mFastTracks[j]; 4059 4060 // Determine whether the track is currently in underrun condition, 4061 // and whether it had a recent underrun. 4062 FastTrackDump *ftDump = &mFastMixerDumpState.mTracks[j]; 4063 FastTrackUnderruns underruns = ftDump->mUnderruns; 4064 uint32_t recentFull = (underruns.mBitFields.mFull - 4065 track->mObservedUnderruns.mBitFields.mFull) & UNDERRUN_MASK; 4066 uint32_t recentPartial = (underruns.mBitFields.mPartial - 4067 track->mObservedUnderruns.mBitFields.mPartial) & UNDERRUN_MASK; 4068 uint32_t recentEmpty = (underruns.mBitFields.mEmpty - 4069 track->mObservedUnderruns.mBitFields.mEmpty) & UNDERRUN_MASK; 4070 uint32_t recentUnderruns = recentPartial + recentEmpty; 4071 track->mObservedUnderruns = underruns; 4072 // don't count underruns that occur while stopping or pausing 4073 // or stopped which can occur when flush() is called while active 4074 if (!(track->isStopping() || track->isPausing() || track->isStopped()) && 4075 recentUnderruns > 0) { 4076 // FIXME fast mixer will pull & mix partial buffers, but we count as a full underrun 4077 track->mAudioTrackServerProxy->tallyUnderrunFrames(recentUnderruns * mFrameCount); 4078 } else { 4079 track->mAudioTrackServerProxy->tallyUnderrunFrames(0); 4080 } 4081 4082 // This is similar to the state machine for normal tracks, 4083 // with a few modifications for fast tracks. 4084 bool isActive = true; 4085 switch (track->mState) { 4086 case TrackBase::STOPPING_1: 4087 // track stays active in STOPPING_1 state until first underrun 4088 if (recentUnderruns > 0 || track->isTerminated()) { 4089 track->mState = TrackBase::STOPPING_2; 4090 } 4091 break; 4092 case TrackBase::PAUSING: 4093 // ramp down is not yet implemented 4094 track->setPaused(); 4095 break; 4096 case TrackBase::RESUMING: 4097 // ramp up is not yet implemented 4098 track->mState = TrackBase::ACTIVE; 4099 break; 4100 case TrackBase::ACTIVE: 4101 if (recentFull > 0 || recentPartial > 0) { 4102 // track has provided at least some frames recently: reset retry count 4103 track->mRetryCount = kMaxTrackRetries; 4104 } 4105 if (recentUnderruns == 0) { 4106 // no recent underruns: stay active 4107 break; 4108 } 4109 // there has recently been an underrun of some kind 4110 if (track->sharedBuffer() == 0) { 4111 // were any of the recent underruns "empty" (no frames available)? 4112 if (recentEmpty == 0) { 4113 // no, then ignore the partial underruns as they are allowed indefinitely 4114 break; 4115 } 4116 // there has recently been an "empty" underrun: decrement the retry counter 4117 if (--(track->mRetryCount) > 0) { 4118 break; 4119 } 4120 // indicate to client process that the track was disabled because of underrun; 4121 // it will then automatically call start() when data is available 4122 track->disable(); 4123 // remove from active list, but state remains ACTIVE [confusing but true] 4124 isActive = false; 4125 break; 4126 } 4127 // fall through 4128 case TrackBase::STOPPING_2: 4129 case TrackBase::PAUSED: 4130 case TrackBase::STOPPED: 4131 case TrackBase::FLUSHED: // flush() while active 4132 // Check for presentation complete if track is inactive 4133 // We have consumed all the buffers of this track. 4134 // This would be incomplete if we auto-paused on underrun 4135 { 4136 size_t audioHALFrames = 4137 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 4138 int64_t framesWritten = mBytesWritten / mFrameSize; 4139 if (!(mStandby || track->presentationComplete(framesWritten, audioHALFrames))) { 4140 // track stays in active list until presentation is complete 4141 break; 4142 } 4143 } 4144 if (track->isStopping_2()) { 4145 track->mState = TrackBase::STOPPED; 4146 } 4147 if (track->isStopped()) { 4148 // Can't reset directly, as fast mixer is still polling this track 4149 // track->reset(); 4150 // So instead mark this track as needing to be reset after push with ack 4151 resetMask |= 1 << i; 4152 } 4153 isActive = false; 4154 break; 4155 case TrackBase::IDLE: 4156 default: 4157 LOG_ALWAYS_FATAL("unexpected track state %d", track->mState); 4158 } 4159 4160 if (isActive) { 4161 // was it previously inactive? 4162 if (!(state->mTrackMask & (1 << j))) { 4163 ExtendedAudioBufferProvider *eabp = track; 4164 VolumeProvider *vp = track; 4165 fastTrack->mBufferProvider = eabp; 4166 fastTrack->mVolumeProvider = vp; 4167 fastTrack->mChannelMask = track->mChannelMask; 4168 fastTrack->mFormat = track->mFormat; 4169 fastTrack->mGeneration++; 4170 state->mTrackMask |= 1 << j; 4171 didModify = true; 4172 // no acknowledgement required for newly active tracks 4173 } 4174 // cache the combined master volume and stream type volume for fast mixer; this 4175 // lacks any synchronization or barrier so VolumeProvider may read a stale value 4176 track->mCachedVolume = masterVolume * mStreamTypes[track->streamType()].volume; 4177 ++fastTracks; 4178 } else { 4179 // was it previously active? 4180 if (state->mTrackMask & (1 << j)) { 4181 fastTrack->mBufferProvider = NULL; 4182 fastTrack->mGeneration++; 4183 state->mTrackMask &= ~(1 << j); 4184 didModify = true; 4185 // If any fast tracks were removed, we must wait for acknowledgement 4186 // because we're about to decrement the last sp<> on those tracks. 4187 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 4188 } else { 4189 LOG_ALWAYS_FATAL("fast track %d should have been active; " 4190 "mState=%d, mTrackMask=%#x, recentUnderruns=%u, isShared=%d", 4191 j, track->mState, state->mTrackMask, recentUnderruns, 4192 track->sharedBuffer() != 0); 4193 } 4194 tracksToRemove->add(track); 4195 // Avoids a misleading display in dumpsys 4196 track->mObservedUnderruns.mBitFields.mMostRecent = UNDERRUN_FULL; 4197 } 4198 continue; 4199 } 4200 4201 { // local variable scope to avoid goto warning 4202 4203 audio_track_cblk_t* cblk = track->cblk(); 4204 4205 // The first time a track is added we wait 4206 // for all its buffers to be filled before processing it 4207 int name = track->name(); 4208 // make sure that we have enough frames to mix one full buffer. 4209 // enforce this condition only once to enable draining the buffer in case the client 4210 // app does not call stop() and relies on underrun to stop: 4211 // hence the test on (mMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 4212 // during last round 4213 size_t desiredFrames; 4214 const uint32_t sampleRate = track->mAudioTrackServerProxy->getSampleRate(); 4215 AudioPlaybackRate playbackRate = track->mAudioTrackServerProxy->getPlaybackRate(); 4216 4217 desiredFrames = sourceFramesNeededWithTimestretch( 4218 sampleRate, mNormalFrameCount, mSampleRate, playbackRate.mSpeed); 4219 // TODO: ONLY USED FOR LEGACY RESAMPLERS, remove when they are removed. 4220 // add frames already consumed but not yet released by the resampler 4221 // because mAudioTrackServerProxy->framesReady() will include these frames 4222 desiredFrames += mAudioMixer->getUnreleasedFrames(track->name()); 4223 4224 uint32_t minFrames = 1; 4225 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() && 4226 (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) { 4227 minFrames = desiredFrames; 4228 } 4229 4230 size_t framesReady = track->framesReady(); 4231 if (ATRACE_ENABLED()) { 4232 // I wish we had formatted trace names 4233 char traceName[16]; 4234 strcpy(traceName, "nRdy"); 4235 int name = track->name(); 4236 if (AudioMixer::TRACK0 <= name && 4237 name < (int) (AudioMixer::TRACK0 + AudioMixer::MAX_NUM_TRACKS)) { 4238 name -= AudioMixer::TRACK0; 4239 traceName[4] = (name / 10) + '0'; 4240 traceName[5] = (name % 10) + '0'; 4241 } else { 4242 traceName[4] = '?'; 4243 traceName[5] = '?'; 4244 } 4245 traceName[6] = '\0'; 4246 ATRACE_INT(traceName, framesReady); 4247 } 4248 if ((framesReady >= minFrames) && track->isReady() && 4249 !track->isPaused() && !track->isTerminated()) 4250 { 4251 ALOGVV("track %d s=%08x [OK] on thread %p", name, cblk->mServer, this); 4252 4253 mixedTracks++; 4254 4255 // track->mainBuffer() != mSinkBuffer or mMixerBuffer means 4256 // there is an effect chain connected to the track 4257 chain.clear(); 4258 if (track->mainBuffer() != mSinkBuffer && 4259 track->mainBuffer() != mMixerBuffer) { 4260 if (mEffectBufferEnabled) { 4261 mEffectBufferValid = true; // Later can set directly. 4262 } 4263 chain = getEffectChain_l(track->sessionId()); 4264 // Delegate volume control to effect in track effect chain if needed 4265 if (chain != 0) { 4266 tracksWithEffect++; 4267 } else { 4268 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on " 4269 "session %d", 4270 name, track->sessionId()); 4271 } 4272 } 4273 4274 4275 int param = AudioMixer::VOLUME; 4276 if (track->mFillingUpStatus == Track::FS_FILLED) { 4277 // no ramp for the first volume setting 4278 track->mFillingUpStatus = Track::FS_ACTIVE; 4279 if (track->mState == TrackBase::RESUMING) { 4280 track->mState = TrackBase::ACTIVE; 4281 param = AudioMixer::RAMP_VOLUME; 4282 } 4283 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 4284 // FIXME should not make a decision based on mServer 4285 } else if (cblk->mServer != 0) { 4286 // If the track is stopped before the first frame was mixed, 4287 // do not apply ramp 4288 param = AudioMixer::RAMP_VOLUME; 4289 } 4290 4291 // compute volume for this track 4292 uint32_t vl, vr; // in U8.24 integer format 4293 float vlf, vrf, vaf; // in [0.0, 1.0] float format 4294 if (track->isPausing() || mStreamTypes[track->streamType()].mute) { 4295 vl = vr = 0; 4296 vlf = vrf = vaf = 0.; 4297 if (track->isPausing()) { 4298 track->setPaused(); 4299 } 4300 } else { 4301 4302 // read original volumes with volume control 4303 float typeVolume = mStreamTypes[track->streamType()].volume; 4304 float v = masterVolume * typeVolume; 4305 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 4306 gain_minifloat_packed_t vlr = proxy->getVolumeLR(); 4307 vlf = float_from_gain(gain_minifloat_unpack_left(vlr)); 4308 vrf = float_from_gain(gain_minifloat_unpack_right(vlr)); 4309 // track volumes come from shared memory, so can't be trusted and must be clamped 4310 if (vlf > GAIN_FLOAT_UNITY) { 4311 ALOGV("Track left volume out of range: %.3g", vlf); 4312 vlf = GAIN_FLOAT_UNITY; 4313 } 4314 if (vrf > GAIN_FLOAT_UNITY) { 4315 ALOGV("Track right volume out of range: %.3g", vrf); 4316 vrf = GAIN_FLOAT_UNITY; 4317 } 4318 // now apply the master volume and stream type volume 4319 vlf *= v; 4320 vrf *= v; 4321 // assuming master volume and stream type volume each go up to 1.0, 4322 // then derive vl and vr as U8.24 versions for the effect chain 4323 const float scaleto8_24 = MAX_GAIN_INT * MAX_GAIN_INT; 4324 vl = (uint32_t) (scaleto8_24 * vlf); 4325 vr = (uint32_t) (scaleto8_24 * vrf); 4326 // vl and vr are now in U8.24 format 4327 uint16_t sendLevel = proxy->getSendLevel_U4_12(); 4328 // send level comes from shared memory and so may be corrupt 4329 if (sendLevel > MAX_GAIN_INT) { 4330 ALOGV("Track send level out of range: %04X", sendLevel); 4331 sendLevel = MAX_GAIN_INT; 4332 } 4333 // vaf is represented as [0.0, 1.0] float by rescaling sendLevel 4334 vaf = v * sendLevel * (1. / MAX_GAIN_INT); 4335 } 4336 4337 // Delegate volume control to effect in track effect chain if needed 4338 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 4339 // Do not ramp volume if volume is controlled by effect 4340 param = AudioMixer::VOLUME; 4341 // Update remaining floating point volume levels 4342 vlf = (float)vl / (1 << 24); 4343 vrf = (float)vr / (1 << 24); 4344 track->mHasVolumeController = true; 4345 } else { 4346 // force no volume ramp when volume controller was just disabled or removed 4347 // from effect chain to avoid volume spike 4348 if (track->mHasVolumeController) { 4349 param = AudioMixer::VOLUME; 4350 } 4351 track->mHasVolumeController = false; 4352 } 4353 4354 // XXX: these things DON'T need to be done each time 4355 mAudioMixer->setBufferProvider(name, track); 4356 mAudioMixer->enable(name); 4357 4358 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, &vlf); 4359 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, &vrf); 4360 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, &vaf); 4361 mAudioMixer->setParameter( 4362 name, 4363 AudioMixer::TRACK, 4364 AudioMixer::FORMAT, (void *)track->format()); 4365 mAudioMixer->setParameter( 4366 name, 4367 AudioMixer::TRACK, 4368 AudioMixer::CHANNEL_MASK, (void *)(uintptr_t)track->channelMask()); 4369 mAudioMixer->setParameter( 4370 name, 4371 AudioMixer::TRACK, 4372 AudioMixer::MIXER_CHANNEL_MASK, (void *)(uintptr_t)mChannelMask); 4373 // limit track sample rate to 2 x output sample rate, which changes at re-configuration 4374 uint32_t maxSampleRate = mSampleRate * AUDIO_RESAMPLER_DOWN_RATIO_MAX; 4375 uint32_t reqSampleRate = track->mAudioTrackServerProxy->getSampleRate(); 4376 if (reqSampleRate == 0) { 4377 reqSampleRate = mSampleRate; 4378 } else if (reqSampleRate > maxSampleRate) { 4379 reqSampleRate = maxSampleRate; 4380 } 4381 mAudioMixer->setParameter( 4382 name, 4383 AudioMixer::RESAMPLE, 4384 AudioMixer::SAMPLE_RATE, 4385 (void *)(uintptr_t)reqSampleRate); 4386 4387 AudioPlaybackRate playbackRate = track->mAudioTrackServerProxy->getPlaybackRate(); 4388 mAudioMixer->setParameter( 4389 name, 4390 AudioMixer::TIMESTRETCH, 4391 AudioMixer::PLAYBACK_RATE, 4392 &playbackRate); 4393 4394 /* 4395 * Select the appropriate output buffer for the track. 4396 * 4397 * Tracks with effects go into their own effects chain buffer 4398 * and from there into either mEffectBuffer or mSinkBuffer. 4399 * 4400 * Other tracks can use mMixerBuffer for higher precision 4401 * channel accumulation. If this buffer is enabled 4402 * (mMixerBufferEnabled true), then selected tracks will accumulate 4403 * into it. 4404 * 4405 */ 4406 if (mMixerBufferEnabled 4407 && (track->mainBuffer() == mSinkBuffer 4408 || track->mainBuffer() == mMixerBuffer)) { 4409 mAudioMixer->setParameter( 4410 name, 4411 AudioMixer::TRACK, 4412 AudioMixer::MIXER_FORMAT, (void *)mMixerBufferFormat); 4413 mAudioMixer->setParameter( 4414 name, 4415 AudioMixer::TRACK, 4416 AudioMixer::MAIN_BUFFER, (void *)mMixerBuffer); 4417 // TODO: override track->mainBuffer()? 4418 mMixerBufferValid = true; 4419 } else { 4420 mAudioMixer->setParameter( 4421 name, 4422 AudioMixer::TRACK, 4423 AudioMixer::MIXER_FORMAT, (void *)AUDIO_FORMAT_PCM_16_BIT); 4424 mAudioMixer->setParameter( 4425 name, 4426 AudioMixer::TRACK, 4427 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 4428 } 4429 mAudioMixer->setParameter( 4430 name, 4431 AudioMixer::TRACK, 4432 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 4433 4434 // reset retry count 4435 track->mRetryCount = kMaxTrackRetries; 4436 4437 // If one track is ready, set the mixer ready if: 4438 // - the mixer was not ready during previous round OR 4439 // - no other track is not ready 4440 if (mMixerStatusIgnoringFastTracks != MIXER_TRACKS_READY || 4441 mixerStatus != MIXER_TRACKS_ENABLED) { 4442 mixerStatus = MIXER_TRACKS_READY; 4443 } 4444 } else { 4445 if (framesReady < desiredFrames && !track->isStopped() && !track->isPaused()) { 4446 ALOGV("track(%p) underrun, framesReady(%zu) < framesDesired(%zd)", 4447 track, framesReady, desiredFrames); 4448 track->mAudioTrackServerProxy->tallyUnderrunFrames(desiredFrames); 4449 } else { 4450 track->mAudioTrackServerProxy->tallyUnderrunFrames(0); 4451 } 4452 4453 // clear effect chain input buffer if an active track underruns to avoid sending 4454 // previous audio buffer again to effects 4455 chain = getEffectChain_l(track->sessionId()); 4456 if (chain != 0) { 4457 chain->clearInputBuffer(); 4458 } 4459 4460 ALOGVV("track %d s=%08x [NOT READY] on thread %p", name, cblk->mServer, this); 4461 if ((track->sharedBuffer() != 0) || track->isTerminated() || 4462 track->isStopped() || track->isPaused()) { 4463 // We have consumed all the buffers of this track. 4464 // Remove it from the list of active tracks. 4465 // TODO: use actual buffer filling status instead of latency when available from 4466 // audio HAL 4467 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 4468 int64_t framesWritten = mBytesWritten / mFrameSize; 4469 if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) { 4470 if (track->isStopped()) { 4471 track->reset(); 4472 } 4473 tracksToRemove->add(track); 4474 } 4475 } else { 4476 // No buffers for this track. Give it a few chances to 4477 // fill a buffer, then remove it from active list. 4478 if (--(track->mRetryCount) <= 0) { 4479 ALOGI("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 4480 tracksToRemove->add(track); 4481 // indicate to client process that the track was disabled because of underrun; 4482 // it will then automatically call start() when data is available 4483 track->disable(); 4484 // If one track is not ready, mark the mixer also not ready if: 4485 // - the mixer was ready during previous round OR 4486 // - no other track is ready 4487 } else if (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY || 4488 mixerStatus != MIXER_TRACKS_READY) { 4489 mixerStatus = MIXER_TRACKS_ENABLED; 4490 } 4491 } 4492 mAudioMixer->disable(name); 4493 } 4494 4495 } // local variable scope to avoid goto warning 4496 4497 } 4498 4499 // Push the new FastMixer state if necessary 4500 bool pauseAudioWatchdog = false; 4501 if (didModify) { 4502 state->mFastTracksGen++; 4503 // if the fast mixer was active, but now there are no fast tracks, then put it in cold idle 4504 if (kUseFastMixer == FastMixer_Dynamic && 4505 state->mCommand == FastMixerState::MIX_WRITE && state->mTrackMask <= 1) { 4506 state->mCommand = FastMixerState::COLD_IDLE; 4507 state->mColdFutexAddr = &mFastMixerFutex; 4508 state->mColdGen++; 4509 mFastMixerFutex = 0; 4510 if (kUseFastMixer == FastMixer_Dynamic) { 4511 mNormalSink = mOutputSink; 4512 } 4513 // If we go into cold idle, need to wait for acknowledgement 4514 // so that fast mixer stops doing I/O. 4515 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 4516 pauseAudioWatchdog = true; 4517 } 4518 } 4519 if (sq != NULL) { 4520 sq->end(didModify); 4521 sq->push(block); 4522 } 4523 #ifdef AUDIO_WATCHDOG 4524 if (pauseAudioWatchdog && mAudioWatchdog != 0) { 4525 mAudioWatchdog->pause(); 4526 } 4527 #endif 4528 4529 // Now perform the deferred reset on fast tracks that have stopped 4530 while (resetMask != 0) { 4531 size_t i = __builtin_ctz(resetMask); 4532 ALOG_ASSERT(i < count); 4533 resetMask &= ~(1 << i); 4534 sp<Track> t = mActiveTracks[i].promote(); 4535 if (t == 0) { 4536 continue; 4537 } 4538 Track* track = t.get(); 4539 ALOG_ASSERT(track->isFastTrack() && track->isStopped()); 4540 track->reset(); 4541 } 4542 4543 // remove all the tracks that need to be... 4544 removeTracks_l(*tracksToRemove); 4545 4546 if (getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX) != 0) { 4547 mEffectBufferValid = true; 4548 } 4549 4550 if (mEffectBufferValid) { 4551 // as long as there are effects we should clear the effects buffer, to avoid 4552 // passing a non-clean buffer to the effect chain 4553 memset(mEffectBuffer, 0, mEffectBufferSize); 4554 } 4555 // sink or mix buffer must be cleared if all tracks are connected to an 4556 // effect chain as in this case the mixer will not write to the sink or mix buffer 4557 // and track effects will accumulate into it 4558 if ((mBytesRemaining == 0) && ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 4559 (mixedTracks == 0 && fastTracks > 0))) { 4560 // FIXME as a performance optimization, should remember previous zero status 4561 if (mMixerBufferValid) { 4562 memset(mMixerBuffer, 0, mMixerBufferSize); 4563 // TODO: In testing, mSinkBuffer below need not be cleared because 4564 // the PlaybackThread::threadLoop() copies mMixerBuffer into mSinkBuffer 4565 // after mixing. 4566 // 4567 // To enforce this guarantee: 4568 // ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 4569 // (mixedTracks == 0 && fastTracks > 0)) 4570 // must imply MIXER_TRACKS_READY. 4571 // Later, we may clear buffers regardless, and skip much of this logic. 4572 } 4573 // FIXME as a performance optimization, should remember previous zero status 4574 memset(mSinkBuffer, 0, mNormalFrameCount * mFrameSize); 4575 } 4576 4577 // if any fast tracks, then status is ready 4578 mMixerStatusIgnoringFastTracks = mixerStatus; 4579 if (fastTracks > 0) { 4580 mixerStatus = MIXER_TRACKS_READY; 4581 } 4582 return mixerStatus; 4583 } 4584 4585 // trackCountForUid_l() must be called with ThreadBase::mLock held 4586 uint32_t AudioFlinger::PlaybackThread::trackCountForUid_l(uid_t uid) 4587 { 4588 uint32_t trackCount = 0; 4589 for (size_t i = 0; i < mTracks.size() ; i++) { 4590 if (mTracks[i]->uid() == (int)uid) { 4591 trackCount++; 4592 } 4593 } 4594 return trackCount; 4595 } 4596 4597 // getTrackName_l() must be called with ThreadBase::mLock held 4598 int AudioFlinger::MixerThread::getTrackName_l(audio_channel_mask_t channelMask, 4599 audio_format_t format, audio_session_t sessionId, uid_t uid) 4600 { 4601 if (trackCountForUid_l(uid) > (PlaybackThread::kMaxTracksPerUid - 1)) { 4602 return -1; 4603 } 4604 return mAudioMixer->getTrackName(channelMask, format, sessionId); 4605 } 4606 4607 // deleteTrackName_l() must be called with ThreadBase::mLock held 4608 void AudioFlinger::MixerThread::deleteTrackName_l(int name) 4609 { 4610 ALOGV("remove track (%d) and delete from mixer", name); 4611 mAudioMixer->deleteTrackName(name); 4612 } 4613 4614 // checkForNewParameter_l() must be called with ThreadBase::mLock held 4615 bool AudioFlinger::MixerThread::checkForNewParameter_l(const String8& keyValuePair, 4616 status_t& status) 4617 { 4618 bool reconfig = false; 4619 bool a2dpDeviceChanged = false; 4620 4621 status = NO_ERROR; 4622 4623 AutoPark<FastMixer> park(mFastMixer); 4624 4625 AudioParameter param = AudioParameter(keyValuePair); 4626 int value; 4627 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 4628 reconfig = true; 4629 } 4630 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 4631 if (!isValidPcmSinkFormat((audio_format_t) value)) { 4632 status = BAD_VALUE; 4633 } else { 4634 // no need to save value, since it's constant 4635 reconfig = true; 4636 } 4637 } 4638 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 4639 if (!isValidPcmSinkChannelMask((audio_channel_mask_t) value)) { 4640 status = BAD_VALUE; 4641 } else { 4642 // no need to save value, since it's constant 4643 reconfig = true; 4644 } 4645 } 4646 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 4647 // do not accept frame count changes if tracks are open as the track buffer 4648 // size depends on frame count and correct behavior would not be guaranteed 4649 // if frame count is changed after track creation 4650 if (!mTracks.isEmpty()) { 4651 status = INVALID_OPERATION; 4652 } else { 4653 reconfig = true; 4654 } 4655 } 4656 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 4657 #ifdef ADD_BATTERY_DATA 4658 // when changing the audio output device, call addBatteryData to notify 4659 // the change 4660 if (mOutDevice != value) { 4661 uint32_t params = 0; 4662 // check whether speaker is on 4663 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 4664 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 4665 } 4666 4667 audio_devices_t deviceWithoutSpeaker 4668 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 4669 // check if any other device (except speaker) is on 4670 if (value & deviceWithoutSpeaker) { 4671 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 4672 } 4673 4674 if (params != 0) { 4675 addBatteryData(params); 4676 } 4677 } 4678 #endif 4679 4680 // forward device change to effects that have requested to be 4681 // aware of attached audio device. 4682 if (value != AUDIO_DEVICE_NONE) { 4683 a2dpDeviceChanged = 4684 (mOutDevice & AUDIO_DEVICE_OUT_ALL_A2DP) != (value & AUDIO_DEVICE_OUT_ALL_A2DP); 4685 mOutDevice = value; 4686 for (size_t i = 0; i < mEffectChains.size(); i++) { 4687 mEffectChains[i]->setDevice_l(mOutDevice); 4688 } 4689 } 4690 } 4691 4692 if (status == NO_ERROR) { 4693 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 4694 keyValuePair.string()); 4695 if (!mStandby && status == INVALID_OPERATION) { 4696 mOutput->standby(); 4697 mStandby = true; 4698 mBytesWritten = 0; 4699 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 4700 keyValuePair.string()); 4701 } 4702 if (status == NO_ERROR && reconfig) { 4703 readOutputParameters_l(); 4704 delete mAudioMixer; 4705 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 4706 for (size_t i = 0; i < mTracks.size() ; i++) { 4707 int name = getTrackName_l(mTracks[i]->mChannelMask, 4708 mTracks[i]->mFormat, mTracks[i]->mSessionId, mTracks[i]->uid()); 4709 if (name < 0) { 4710 break; 4711 } 4712 mTracks[i]->mName = name; 4713 } 4714 sendIoConfigEvent_l(AUDIO_OUTPUT_CONFIG_CHANGED); 4715 } 4716 } 4717 4718 return reconfig || a2dpDeviceChanged; 4719 } 4720 4721 4722 void AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 4723 { 4724 PlaybackThread::dumpInternals(fd, args); 4725 dprintf(fd, " Thread throttle time (msecs): %u\n", mThreadThrottleTimeMs); 4726 dprintf(fd, " AudioMixer tracks: 0x%08x\n", mAudioMixer->trackNames()); 4727 dprintf(fd, " Master mono: %s\n", mMasterMono ? "on" : "off"); 4728 4729 // Make a non-atomic copy of fast mixer dump state so it won't change underneath us 4730 // while we are dumping it. It may be inconsistent, but it won't mutate! 4731 // This is a large object so we place it on the heap. 4732 // FIXME 25972958: Need an intelligent copy constructor that does not touch unused pages. 4733 const FastMixerDumpState *copy = new FastMixerDumpState(mFastMixerDumpState); 4734 copy->dump(fd); 4735 delete copy; 4736 4737 #ifdef STATE_QUEUE_DUMP 4738 // Similar for state queue 4739 StateQueueObserverDump observerCopy = mStateQueueObserverDump; 4740 observerCopy.dump(fd); 4741 StateQueueMutatorDump mutatorCopy = mStateQueueMutatorDump; 4742 mutatorCopy.dump(fd); 4743 #endif 4744 4745 #ifdef TEE_SINK 4746 // Write the tee output to a .wav file 4747 dumpTee(fd, mTeeSource, mId); 4748 #endif 4749 4750 #ifdef AUDIO_WATCHDOG 4751 if (mAudioWatchdog != 0) { 4752 // Make a non-atomic copy of audio watchdog dump so it won't change underneath us 4753 AudioWatchdogDump wdCopy = mAudioWatchdogDump; 4754 wdCopy.dump(fd); 4755 } 4756 #endif 4757 } 4758 4759 uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const 4760 { 4761 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000) / 2; 4762 } 4763 4764 uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() const 4765 { 4766 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000); 4767 } 4768 4769 void AudioFlinger::MixerThread::cacheParameters_l() 4770 { 4771 PlaybackThread::cacheParameters_l(); 4772 4773 // FIXME: Relaxed timing because of a certain device that can't meet latency 4774 // Should be reduced to 2x after the vendor fixes the driver issue 4775 // increase threshold again due to low power audio mode. The way this warning 4776 // threshold is calculated and its usefulness should be reconsidered anyway. 4777 maxPeriod = seconds(mNormalFrameCount) / mSampleRate * 15; 4778 } 4779 4780 // ---------------------------------------------------------------------------- 4781 4782 AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 4783 AudioStreamOut* output, audio_io_handle_t id, audio_devices_t device, bool systemReady) 4784 : PlaybackThread(audioFlinger, output, id, device, DIRECT, systemReady) 4785 // mLeftVolFloat, mRightVolFloat 4786 { 4787 } 4788 4789 AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 4790 AudioStreamOut* output, audio_io_handle_t id, uint32_t device, 4791 ThreadBase::type_t type, bool systemReady) 4792 : PlaybackThread(audioFlinger, output, id, device, type, systemReady) 4793 // mLeftVolFloat, mRightVolFloat 4794 { 4795 } 4796 4797 AudioFlinger::DirectOutputThread::~DirectOutputThread() 4798 { 4799 } 4800 4801 void AudioFlinger::DirectOutputThread::processVolume_l(Track *track, bool lastTrack) 4802 { 4803 float left, right; 4804 4805 if (mMasterMute || mStreamTypes[track->streamType()].mute) { 4806 left = right = 0; 4807 } else { 4808 float typeVolume = mStreamTypes[track->streamType()].volume; 4809 float v = mMasterVolume * typeVolume; 4810 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 4811 gain_minifloat_packed_t vlr = proxy->getVolumeLR(); 4812 left = float_from_gain(gain_minifloat_unpack_left(vlr)); 4813 if (left > GAIN_FLOAT_UNITY) { 4814 left = GAIN_FLOAT_UNITY; 4815 } 4816 left *= v; 4817 right = float_from_gain(gain_minifloat_unpack_right(vlr)); 4818 if (right > GAIN_FLOAT_UNITY) { 4819 right = GAIN_FLOAT_UNITY; 4820 } 4821 right *= v; 4822 } 4823 4824 if (lastTrack) { 4825 if (left != mLeftVolFloat || right != mRightVolFloat) { 4826 mLeftVolFloat = left; 4827 mRightVolFloat = right; 4828 4829 // Convert volumes from float to 8.24 4830 uint32_t vl = (uint32_t)(left * (1 << 24)); 4831 uint32_t vr = (uint32_t)(right * (1 << 24)); 4832 4833 // Delegate volume control to effect in track effect chain if needed 4834 // only one effect chain can be present on DirectOutputThread, so if 4835 // there is one, the track is connected to it 4836 if (!mEffectChains.isEmpty()) { 4837 mEffectChains[0]->setVolume_l(&vl, &vr); 4838 left = (float)vl / (1 << 24); 4839 right = (float)vr / (1 << 24); 4840 } 4841 if (mOutput->stream->set_volume) { 4842 mOutput->stream->set_volume(mOutput->stream, left, right); 4843 } 4844 } 4845 } 4846 } 4847 4848 void AudioFlinger::DirectOutputThread::onAddNewTrack_l() 4849 { 4850 sp<Track> previousTrack = mPreviousTrack.promote(); 4851 sp<Track> latestTrack = mLatestActiveTrack.promote(); 4852 4853 if (previousTrack != 0 && latestTrack != 0) { 4854 if (mType == DIRECT) { 4855 if (previousTrack.get() != latestTrack.get()) { 4856 mFlushPending = true; 4857 } 4858 } else /* mType == OFFLOAD */ { 4859 if (previousTrack->sessionId() != latestTrack->sessionId()) { 4860 mFlushPending = true; 4861 } 4862 } 4863 } 4864 PlaybackThread::onAddNewTrack_l(); 4865 } 4866 4867 AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l( 4868 Vector< sp<Track> > *tracksToRemove 4869 ) 4870 { 4871 size_t count = mActiveTracks.size(); 4872 mixer_state mixerStatus = MIXER_IDLE; 4873 bool doHwPause = false; 4874 bool doHwResume = false; 4875 4876 // find out which tracks need to be processed 4877 for (size_t i = 0; i < count; i++) { 4878 sp<Track> t = mActiveTracks[i].promote(); 4879 // The track died recently 4880 if (t == 0) { 4881 continue; 4882 } 4883 4884 if (t->isInvalid()) { 4885 ALOGW("An invalidated track shouldn't be in active list"); 4886 tracksToRemove->add(t); 4887 continue; 4888 } 4889 4890 Track* const track = t.get(); 4891 #ifdef VERY_VERY_VERBOSE_LOGGING 4892 audio_track_cblk_t* cblk = track->cblk(); 4893 #endif 4894 // Only consider last track started for volume and mixer state control. 4895 // In theory an older track could underrun and restart after the new one starts 4896 // but as we only care about the transition phase between two tracks on a 4897 // direct output, it is not a problem to ignore the underrun case. 4898 sp<Track> l = mLatestActiveTrack.promote(); 4899 bool last = l.get() == track; 4900 4901 if (track->isPausing()) { 4902 track->setPaused(); 4903 if (mHwSupportsPause && last && !mHwPaused) { 4904 doHwPause = true; 4905 mHwPaused = true; 4906 } 4907 tracksToRemove->add(track); 4908 } else if (track->isFlushPending()) { 4909 track->flushAck(); 4910 if (last) { 4911 mFlushPending = true; 4912 } 4913 } else if (track->isResumePending()) { 4914 track->resumeAck(); 4915 if (last) { 4916 mLeftVolFloat = mRightVolFloat = -1.0; 4917 if (mHwPaused) { 4918 doHwResume = true; 4919 mHwPaused = false; 4920 } 4921 } 4922 } 4923 4924 // The first time a track is added we wait 4925 // for all its buffers to be filled before processing it. 4926 // Allow draining the buffer in case the client 4927 // app does not call stop() and relies on underrun to stop: 4928 // hence the test on (track->mRetryCount > 1). 4929 // If retryCount<=1 then track is about to underrun and be removed. 4930 // Do not use a high threshold for compressed audio. 4931 uint32_t minFrames; 4932 if ((track->sharedBuffer() == 0) && !track->isStopping_1() && !track->isPausing() 4933 && (track->mRetryCount > 1) && audio_has_proportional_frames(mFormat)) { 4934 minFrames = mNormalFrameCount; 4935 } else { 4936 minFrames = 1; 4937 } 4938 4939 if ((track->framesReady() >= minFrames) && track->isReady() && !track->isPaused() && 4940 !track->isStopping_2() && !track->isStopped()) 4941 { 4942 ALOGVV("track %d s=%08x [OK]", track->name(), cblk->mServer); 4943 4944 if (track->mFillingUpStatus == Track::FS_FILLED) { 4945 track->mFillingUpStatus = Track::FS_ACTIVE; 4946 if (last) { 4947 // make sure processVolume_l() will apply new volume even if 0 4948 mLeftVolFloat = mRightVolFloat = -1.0; 4949 } 4950 if (!mHwSupportsPause) { 4951 track->resumeAck(); 4952 } 4953 } 4954 4955 // compute volume for this track 4956 processVolume_l(track, last); 4957 if (last) { 4958 sp<Track> previousTrack = mPreviousTrack.promote(); 4959 if (previousTrack != 0) { 4960 if (track != previousTrack.get()) { 4961 // Flush any data still being written from last track 4962 mBytesRemaining = 0; 4963 // Invalidate previous track to force a seek when resuming. 4964 previousTrack->invalidate(); 4965 } 4966 } 4967 mPreviousTrack = track; 4968 4969 // reset retry count 4970 track->mRetryCount = kMaxTrackRetriesDirect; 4971 mActiveTrack = t; 4972 mixerStatus = MIXER_TRACKS_READY; 4973 if (mHwPaused) { 4974 doHwResume = true; 4975 mHwPaused = false; 4976 } 4977 } 4978 } else { 4979 // clear effect chain input buffer if the last active track started underruns 4980 // to avoid sending previous audio buffer again to effects 4981 if (!mEffectChains.isEmpty() && last) { 4982 mEffectChains[0]->clearInputBuffer(); 4983 } 4984 if (track->isStopping_1()) { 4985 track->mState = TrackBase::STOPPING_2; 4986 if (last && mHwPaused) { 4987 doHwResume = true; 4988 mHwPaused = false; 4989 } 4990 } 4991 if ((track->sharedBuffer() != 0) || track->isStopped() || 4992 track->isStopping_2() || track->isPaused()) { 4993 // We have consumed all the buffers of this track. 4994 // Remove it from the list of active tracks. 4995 size_t audioHALFrames; 4996 if (audio_has_proportional_frames(mFormat)) { 4997 audioHALFrames = (latency_l() * mSampleRate) / 1000; 4998 } else { 4999 audioHALFrames = 0; 5000 } 5001 5002 int64_t framesWritten = mBytesWritten / mFrameSize; 5003 if (mStandby || !last || 5004 track->presentationComplete(framesWritten, audioHALFrames)) { 5005 if (track->isStopping_2()) { 5006 track->mState = TrackBase::STOPPED; 5007 } 5008 if (track->isStopped()) { 5009 track->reset(); 5010 } 5011 tracksToRemove->add(track); 5012 } 5013 } else { 5014 // No buffers for this track. Give it a few chances to 5015 // fill a buffer, then remove it from active list. 5016 // Only consider last track started for mixer state control 5017 if (--(track->mRetryCount) <= 0) { 5018 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 5019 tracksToRemove->add(track); 5020 // indicate to client process that the track was disabled because of underrun; 5021 // it will then automatically call start() when data is available 5022 track->disable(); 5023 } else if (last) { 5024 ALOGW("pause because of UNDERRUN, framesReady = %zu," 5025 "minFrames = %u, mFormat = %#x", 5026 track->framesReady(), minFrames, mFormat); 5027 mixerStatus = MIXER_TRACKS_ENABLED; 5028 if (mHwSupportsPause && !mHwPaused && !mStandby) { 5029 doHwPause = true; 5030 mHwPaused = true; 5031 } 5032 } 5033 } 5034 } 5035 } 5036 5037 // if an active track did not command a flush, check for pending flush on stopped tracks 5038 if (!mFlushPending) { 5039 for (size_t i = 0; i < mTracks.size(); i++) { 5040 if (mTracks[i]->isFlushPending()) { 5041 mTracks[i]->flushAck(); 5042 mFlushPending = true; 5043 } 5044 } 5045 } 5046 5047 // make sure the pause/flush/resume sequence is executed in the right order. 5048 // If a flush is pending and a track is active but the HW is not paused, force a HW pause 5049 // before flush and then resume HW. This can happen in case of pause/flush/resume 5050 // if resume is received before pause is executed. 5051 if (mHwSupportsPause && !mStandby && 5052 (doHwPause || (mFlushPending && !mHwPaused && (count != 0)))) { 5053 mOutput->stream->pause(mOutput->stream); 5054 } 5055 if (mFlushPending) { 5056 flushHw_l(); 5057 } 5058 if (mHwSupportsPause && !mStandby && doHwResume) { 5059 mOutput->stream->resume(mOutput->stream); 5060 } 5061 // remove all the tracks that need to be... 5062 removeTracks_l(*tracksToRemove); 5063 5064 return mixerStatus; 5065 } 5066 5067 void AudioFlinger::DirectOutputThread::threadLoop_mix() 5068 { 5069 size_t frameCount = mFrameCount; 5070 int8_t *curBuf = (int8_t *)mSinkBuffer; 5071 // output audio to hardware 5072 while (frameCount) { 5073 AudioBufferProvider::Buffer buffer; 5074 buffer.frameCount = frameCount; 5075 status_t status = mActiveTrack->getNextBuffer(&buffer); 5076 if (status != NO_ERROR || buffer.raw == NULL) { 5077 // no need to pad with 0 for compressed audio 5078 if (audio_has_proportional_frames(mFormat)) { 5079 memset(curBuf, 0, frameCount * mFrameSize); 5080 } 5081 break; 5082 } 5083 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 5084 frameCount -= buffer.frameCount; 5085 curBuf += buffer.frameCount * mFrameSize; 5086 mActiveTrack->releaseBuffer(&buffer); 5087 } 5088 mCurrentWriteLength = curBuf - (int8_t *)mSinkBuffer; 5089 mSleepTimeUs = 0; 5090 mStandbyTimeNs = systemTime() + mStandbyDelayNs; 5091 mActiveTrack.clear(); 5092 } 5093 5094 void AudioFlinger::DirectOutputThread::threadLoop_sleepTime() 5095 { 5096 // do not write to HAL when paused 5097 if (mHwPaused || (usesHwAvSync() && mStandby)) { 5098 mSleepTimeUs = mIdleSleepTimeUs; 5099 return; 5100 } 5101 if (mSleepTimeUs == 0) { 5102 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 5103 mSleepTimeUs = mActiveSleepTimeUs; 5104 } else { 5105 mSleepTimeUs = mIdleSleepTimeUs; 5106 } 5107 } else if (mBytesWritten != 0 && audio_has_proportional_frames(mFormat)) { 5108 memset(mSinkBuffer, 0, mFrameCount * mFrameSize); 5109 mSleepTimeUs = 0; 5110 } 5111 } 5112 5113 void AudioFlinger::DirectOutputThread::threadLoop_exit() 5114 { 5115 { 5116 Mutex::Autolock _l(mLock); 5117 for (size_t i = 0; i < mTracks.size(); i++) { 5118 if (mTracks[i]->isFlushPending()) { 5119 mTracks[i]->flushAck(); 5120 mFlushPending = true; 5121 } 5122 } 5123 if (mFlushPending) { 5124 flushHw_l(); 5125 } 5126 } 5127 PlaybackThread::threadLoop_exit(); 5128 } 5129 5130 // must be called with thread mutex locked 5131 bool AudioFlinger::DirectOutputThread::shouldStandby_l() 5132 { 5133 bool trackPaused = false; 5134 bool trackStopped = false; 5135 5136 if ((mType == DIRECT) && audio_is_linear_pcm(mFormat) && !usesHwAvSync()) { 5137 return !mStandby; 5138 } 5139 5140 // do not put the HAL in standby when paused. AwesomePlayer clear the offloaded AudioTrack 5141 // after a timeout and we will enter standby then. 5142 if (mTracks.size() > 0) { 5143 trackPaused = mTracks[mTracks.size() - 1]->isPaused(); 5144 trackStopped = mTracks[mTracks.size() - 1]->isStopped() || 5145 mTracks[mTracks.size() - 1]->mState == TrackBase::IDLE; 5146 } 5147 5148 return !mStandby && !(trackPaused || (mHwPaused && !trackStopped)); 5149 } 5150 5151 // getTrackName_l() must be called with ThreadBase::mLock held 5152 int AudioFlinger::DirectOutputThread::getTrackName_l(audio_channel_mask_t channelMask __unused, 5153 audio_format_t format __unused, audio_session_t sessionId __unused, uid_t uid) 5154 { 5155 if (trackCountForUid_l(uid) > (PlaybackThread::kMaxTracksPerUid - 1)) { 5156 return -1; 5157 } 5158 return 0; 5159 } 5160 5161 // deleteTrackName_l() must be called with ThreadBase::mLock held 5162 void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name __unused) 5163 { 5164 } 5165 5166 // checkForNewParameter_l() must be called with ThreadBase::mLock held 5167 bool AudioFlinger::DirectOutputThread::checkForNewParameter_l(const String8& keyValuePair, 5168 status_t& status) 5169 { 5170 bool reconfig = false; 5171 bool a2dpDeviceChanged = false; 5172 5173 status = NO_ERROR; 5174 5175 AudioParameter param = AudioParameter(keyValuePair); 5176 int value; 5177 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 5178 // forward device change to effects that have requested to be 5179 // aware of attached audio device. 5180 if (value != AUDIO_DEVICE_NONE) { 5181 a2dpDeviceChanged = 5182 (mOutDevice & AUDIO_DEVICE_OUT_ALL_A2DP) != (value & AUDIO_DEVICE_OUT_ALL_A2DP); 5183 mOutDevice = value; 5184 for (size_t i = 0; i < mEffectChains.size(); i++) { 5185 mEffectChains[i]->setDevice_l(mOutDevice); 5186 } 5187 } 5188 } 5189 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 5190 // do not accept frame count changes if tracks are open as the track buffer 5191 // size depends on frame count and correct behavior would not be garantied 5192 // if frame count is changed after track creation 5193 if (!mTracks.isEmpty()) { 5194 status = INVALID_OPERATION; 5195 } else { 5196 reconfig = true; 5197 } 5198 } 5199 if (status == NO_ERROR) { 5200 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 5201 keyValuePair.string()); 5202 if (!mStandby && status == INVALID_OPERATION) { 5203 mOutput->standby(); 5204 mStandby = true; 5205 mBytesWritten = 0; 5206 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 5207 keyValuePair.string()); 5208 } 5209 if (status == NO_ERROR && reconfig) { 5210 readOutputParameters_l(); 5211 sendIoConfigEvent_l(AUDIO_OUTPUT_CONFIG_CHANGED); 5212 } 5213 } 5214 5215 return reconfig || a2dpDeviceChanged; 5216 } 5217 5218 uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() const 5219 { 5220 uint32_t time; 5221 if (audio_has_proportional_frames(mFormat)) { 5222 time = PlaybackThread::activeSleepTimeUs(); 5223 } else { 5224 time = kDirectMinSleepTimeUs; 5225 } 5226 return time; 5227 } 5228 5229 uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() const 5230 { 5231 uint32_t time; 5232 if (audio_has_proportional_frames(mFormat)) { 5233 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 5234 } else { 5235 time = kDirectMinSleepTimeUs; 5236 } 5237 return time; 5238 } 5239 5240 uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() const 5241 { 5242 uint32_t time; 5243 if (audio_has_proportional_frames(mFormat)) { 5244 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 5245 } else { 5246 time = kDirectMinSleepTimeUs; 5247 } 5248 return time; 5249 } 5250 5251 void AudioFlinger::DirectOutputThread::cacheParameters_l() 5252 { 5253 PlaybackThread::cacheParameters_l(); 5254 5255 // use shorter standby delay as on normal output to release 5256 // hardware resources as soon as possible 5257 // no delay on outputs with HW A/V sync 5258 if (usesHwAvSync()) { 5259 mStandbyDelayNs = 0; 5260 } else if ((mType == OFFLOAD) && !audio_has_proportional_frames(mFormat)) { 5261 mStandbyDelayNs = kOffloadStandbyDelayNs; 5262 } else { 5263 mStandbyDelayNs = microseconds(mActiveSleepTimeUs*2); 5264 } 5265 } 5266 5267 void AudioFlinger::DirectOutputThread::flushHw_l() 5268 { 5269 mOutput->flush(); 5270 mHwPaused = false; 5271 mFlushPending = false; 5272 } 5273 5274 // ---------------------------------------------------------------------------- 5275 5276 AudioFlinger::AsyncCallbackThread::AsyncCallbackThread( 5277 const wp<AudioFlinger::PlaybackThread>& playbackThread) 5278 : Thread(false /*canCallJava*/), 5279 mPlaybackThread(playbackThread), 5280 mWriteAckSequence(0), 5281 mDrainSequence(0), 5282 mAsyncError(false) 5283 { 5284 } 5285 5286 AudioFlinger::AsyncCallbackThread::~AsyncCallbackThread() 5287 { 5288 } 5289 5290 void AudioFlinger::AsyncCallbackThread::onFirstRef() 5291 { 5292 run("Offload Cbk", ANDROID_PRIORITY_URGENT_AUDIO); 5293 } 5294 5295 bool AudioFlinger::AsyncCallbackThread::threadLoop() 5296 { 5297 while (!exitPending()) { 5298 uint32_t writeAckSequence; 5299 uint32_t drainSequence; 5300 bool asyncError; 5301 5302 { 5303 Mutex::Autolock _l(mLock); 5304 while (!((mWriteAckSequence & 1) || 5305 (mDrainSequence & 1) || 5306 mAsyncError || 5307 exitPending())) { 5308 mWaitWorkCV.wait(mLock); 5309 } 5310 5311 if (exitPending()) { 5312 break; 5313 } 5314 ALOGV("AsyncCallbackThread mWriteAckSequence %d mDrainSequence %d", 5315 mWriteAckSequence, mDrainSequence); 5316 writeAckSequence = mWriteAckSequence; 5317 mWriteAckSequence &= ~1; 5318 drainSequence = mDrainSequence; 5319 mDrainSequence &= ~1; 5320 asyncError = mAsyncError; 5321 mAsyncError = false; 5322 } 5323 { 5324 sp<AudioFlinger::PlaybackThread> playbackThread = mPlaybackThread.promote(); 5325 if (playbackThread != 0) { 5326 if (writeAckSequence & 1) { 5327 playbackThread->resetWriteBlocked(writeAckSequence >> 1); 5328 } 5329 if (drainSequence & 1) { 5330 playbackThread->resetDraining(drainSequence >> 1); 5331 } 5332 if (asyncError) { 5333 playbackThread->onAsyncError(); 5334 } 5335 } 5336 } 5337 } 5338 return false; 5339 } 5340 5341 void AudioFlinger::AsyncCallbackThread::exit() 5342 { 5343 ALOGV("AsyncCallbackThread::exit"); 5344 Mutex::Autolock _l(mLock); 5345 requestExit(); 5346 mWaitWorkCV.broadcast(); 5347 } 5348 5349 void AudioFlinger::AsyncCallbackThread::setWriteBlocked(uint32_t sequence) 5350 { 5351 Mutex::Autolock _l(mLock); 5352 // bit 0 is cleared 5353 mWriteAckSequence = sequence << 1; 5354 } 5355 5356 void AudioFlinger::AsyncCallbackThread::resetWriteBlocked() 5357 { 5358 Mutex::Autolock _l(mLock); 5359 // ignore unexpected callbacks 5360 if (mWriteAckSequence & 2) { 5361 mWriteAckSequence |= 1; 5362 mWaitWorkCV.signal(); 5363 } 5364 } 5365 5366 void AudioFlinger::AsyncCallbackThread::setDraining(uint32_t sequence) 5367 { 5368 Mutex::Autolock _l(mLock); 5369 // bit 0 is cleared 5370 mDrainSequence = sequence << 1; 5371 } 5372 5373 void AudioFlinger::AsyncCallbackThread::resetDraining() 5374 { 5375 Mutex::Autolock _l(mLock); 5376 // ignore unexpected callbacks 5377 if (mDrainSequence & 2) { 5378 mDrainSequence |= 1; 5379 mWaitWorkCV.signal(); 5380 } 5381 } 5382 5383 void AudioFlinger::AsyncCallbackThread::setAsyncError() 5384 { 5385 Mutex::Autolock _l(mLock); 5386 mAsyncError = true; 5387 mWaitWorkCV.signal(); 5388 } 5389 5390 5391 // ---------------------------------------------------------------------------- 5392 AudioFlinger::OffloadThread::OffloadThread(const sp<AudioFlinger>& audioFlinger, 5393 AudioStreamOut* output, audio_io_handle_t id, uint32_t device, bool systemReady) 5394 : DirectOutputThread(audioFlinger, output, id, device, OFFLOAD, systemReady), 5395 mPausedWriteLength(0), mPausedBytesRemaining(0), mKeepWakeLock(true), 5396 mOffloadUnderrunPosition(~0LL) 5397 { 5398 //FIXME: mStandby should be set to true by ThreadBase constructor 5399 mStandby = true; 5400 mKeepWakeLock = property_get_bool("ro.audio.offload_wakelock", true /* default_value */); 5401 } 5402 5403 void AudioFlinger::OffloadThread::threadLoop_exit() 5404 { 5405 if (mFlushPending || mHwPaused) { 5406 // If a flush is pending or track was paused, just discard buffered data 5407 flushHw_l(); 5408 } else { 5409 mMixerStatus = MIXER_DRAIN_ALL; 5410 threadLoop_drain(); 5411 } 5412 if (mUseAsyncWrite) { 5413 ALOG_ASSERT(mCallbackThread != 0); 5414 mCallbackThread->exit(); 5415 } 5416 PlaybackThread::threadLoop_exit(); 5417 } 5418 5419 AudioFlinger::PlaybackThread::mixer_state AudioFlinger::OffloadThread::prepareTracks_l( 5420 Vector< sp<Track> > *tracksToRemove 5421 ) 5422 { 5423 size_t count = mActiveTracks.size(); 5424 5425 mixer_state mixerStatus = MIXER_IDLE; 5426 bool doHwPause = false; 5427 bool doHwResume = false; 5428 5429 ALOGV("OffloadThread::prepareTracks_l active tracks %zu", count); 5430 5431 // find out which tracks need to be processed 5432 for (size_t i = 0; i < count; i++) { 5433 sp<Track> t = mActiveTracks[i].promote(); 5434 // The track died recently 5435 if (t == 0) { 5436 continue; 5437 } 5438 Track* const track = t.get(); 5439 #ifdef VERY_VERY_VERBOSE_LOGGING 5440 audio_track_cblk_t* cblk = track->cblk(); 5441 #endif 5442 // Only consider last track started for volume and mixer state control. 5443 // In theory an older track could underrun and restart after the new one starts 5444 // but as we only care about the transition phase between two tracks on a 5445 // direct output, it is not a problem to ignore the underrun case. 5446 sp<Track> l = mLatestActiveTrack.promote(); 5447 bool last = l.get() == track; 5448 5449 if (track->isInvalid()) { 5450 ALOGW("An invalidated track shouldn't be in active list"); 5451 tracksToRemove->add(track); 5452 continue; 5453 } 5454 5455 if (track->mState == TrackBase::IDLE) { 5456 ALOGW("An idle track shouldn't be in active list"); 5457 continue; 5458 } 5459 5460 if (track->isPausing()) { 5461 track->setPaused(); 5462 if (last) { 5463 if (mHwSupportsPause && !mHwPaused) { 5464 doHwPause = true; 5465 mHwPaused = true; 5466 } 5467 // If we were part way through writing the mixbuffer to 5468 // the HAL we must save this until we resume 5469 // BUG - this will be wrong if a different track is made active, 5470 // in that case we want to discard the pending data in the 5471 // mixbuffer and tell the client to present it again when the 5472 // track is resumed 5473 mPausedWriteLength = mCurrentWriteLength; 5474 mPausedBytesRemaining = mBytesRemaining; 5475 mBytesRemaining = 0; // stop writing 5476 } 5477 tracksToRemove->add(track); 5478 } else if (track->isFlushPending()) { 5479 if (track->isStopping_1()) { 5480 track->mRetryCount = kMaxTrackStopRetriesOffload; 5481 } else { 5482 track->mRetryCount = kMaxTrackRetriesOffload; 5483 } 5484 track->flushAck(); 5485 if (last) { 5486 mFlushPending = true; 5487 } 5488 } else if (track->isResumePending()){ 5489 track->resumeAck(); 5490 if (last) { 5491 if (mPausedBytesRemaining) { 5492 // Need to continue write that was interrupted 5493 mCurrentWriteLength = mPausedWriteLength; 5494 mBytesRemaining = mPausedBytesRemaining; 5495 mPausedBytesRemaining = 0; 5496 } 5497 if (mHwPaused) { 5498 doHwResume = true; 5499 mHwPaused = false; 5500 // threadLoop_mix() will handle the case that we need to 5501 // resume an interrupted write 5502 } 5503 // enable write to audio HAL 5504 mSleepTimeUs = 0; 5505 5506 mLeftVolFloat = mRightVolFloat = -1.0; 5507 5508 // Do not handle new data in this iteration even if track->framesReady() 5509 mixerStatus = MIXER_TRACKS_ENABLED; 5510 } 5511 } else if (track->framesReady() && track->isReady() && 5512 !track->isPaused() && !track->isTerminated() && !track->isStopping_2()) { 5513 ALOGVV("OffloadThread: track %d s=%08x [OK]", track->name(), cblk->mServer); 5514 if (track->mFillingUpStatus == Track::FS_FILLED) { 5515 track->mFillingUpStatus = Track::FS_ACTIVE; 5516 if (last) { 5517 // make sure processVolume_l() will apply new volume even if 0 5518 mLeftVolFloat = mRightVolFloat = -1.0; 5519 } 5520 } 5521 5522 if (last) { 5523 sp<Track> previousTrack = mPreviousTrack.promote(); 5524 if (previousTrack != 0) { 5525 if (track != previousTrack.get()) { 5526 // Flush any data still being written from last track 5527 mBytesRemaining = 0; 5528 if (mPausedBytesRemaining) { 5529 // Last track was paused so we also need to flush saved 5530 // mixbuffer state and invalidate track so that it will 5531 // re-submit that unwritten data when it is next resumed 5532 mPausedBytesRemaining = 0; 5533 // Invalidate is a bit drastic - would be more efficient 5534 // to have a flag to tell client that some of the 5535 // previously written data was lost 5536 previousTrack->invalidate(); 5537 } 5538 // flush data already sent to the DSP if changing audio session as audio 5539 // comes from a different source. Also invalidate previous track to force a 5540 // seek when resuming. 5541 if (previousTrack->sessionId() != track->sessionId()) { 5542 previousTrack->invalidate(); 5543 } 5544 } 5545 } 5546 mPreviousTrack = track; 5547 // reset retry count 5548 if (track->isStopping_1()) { 5549 track->mRetryCount = kMaxTrackStopRetriesOffload; 5550 } else { 5551 track->mRetryCount = kMaxTrackRetriesOffload; 5552 } 5553 mActiveTrack = t; 5554 mixerStatus = MIXER_TRACKS_READY; 5555 } 5556 } else { 5557 ALOGVV("OffloadThread: track %d s=%08x [NOT READY]", track->name(), cblk->mServer); 5558 if (track->isStopping_1()) { 5559 if (--(track->mRetryCount) <= 0) { 5560 // Hardware buffer can hold a large amount of audio so we must 5561 // wait for all current track's data to drain before we say 5562 // that the track is stopped. 5563 if (mBytesRemaining == 0) { 5564 // Only start draining when all data in mixbuffer 5565 // has been written 5566 ALOGV("OffloadThread: underrun and STOPPING_1 -> draining, STOPPING_2"); 5567 track->mState = TrackBase::STOPPING_2; // so presentation completes after 5568 // drain do not drain if no data was ever sent to HAL (mStandby == true) 5569 if (last && !mStandby) { 5570 // do not modify drain sequence if we are already draining. This happens 5571 // when resuming from pause after drain. 5572 if ((mDrainSequence & 1) == 0) { 5573 mSleepTimeUs = 0; 5574 mStandbyTimeNs = systemTime() + mStandbyDelayNs; 5575 mixerStatus = MIXER_DRAIN_TRACK; 5576 mDrainSequence += 2; 5577 } 5578 if (mHwPaused) { 5579 // It is possible to move from PAUSED to STOPPING_1 without 5580 // a resume so we must ensure hardware is running 5581 doHwResume = true; 5582 mHwPaused = false; 5583 } 5584 } 5585 } 5586 } else if (last) { 5587 ALOGV("stopping1 underrun retries left %d", track->mRetryCount); 5588 mixerStatus = MIXER_TRACKS_ENABLED; 5589 } 5590 } else if (track->isStopping_2()) { 5591 // Drain has completed or we are in standby, signal presentation complete 5592 if (!(mDrainSequence & 1) || !last || mStandby) { 5593 track->mState = TrackBase::STOPPED; 5594 size_t audioHALFrames = 5595 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 5596 int64_t framesWritten = 5597 mBytesWritten / mOutput->getFrameSize(); 5598 track->presentationComplete(framesWritten, audioHALFrames); 5599 track->reset(); 5600 tracksToRemove->add(track); 5601 } 5602 } else { 5603 // No buffers for this track. Give it a few chances to 5604 // fill a buffer, then remove it from active list. 5605 if (--(track->mRetryCount) <= 0) { 5606 bool running = false; 5607 if (mOutput->stream->get_presentation_position != nullptr) { 5608 uint64_t position = 0; 5609 struct timespec unused; 5610 // The running check restarts the retry counter at least once. 5611 int ret = mOutput->stream->get_presentation_position( 5612 mOutput->stream, &position, &unused); 5613 if (ret == NO_ERROR && position != mOffloadUnderrunPosition) { 5614 running = true; 5615 mOffloadUnderrunPosition = position; 5616 } 5617 ALOGVV("underrun counter, running(%d): %lld vs %lld", running, 5618 (long long)position, (long long)mOffloadUnderrunPosition); 5619 } 5620 if (running) { // still running, give us more time. 5621 track->mRetryCount = kMaxTrackRetriesOffload; 5622 } else { 5623 ALOGV("OffloadThread: BUFFER TIMEOUT: remove(%d) from active list", 5624 track->name()); 5625 tracksToRemove->add(track); 5626 // indicate to client process that the track was disabled because of underrun; 5627 // it will then automatically call start() when data is available 5628 track->disable(); 5629 } 5630 } else if (last){ 5631 mixerStatus = MIXER_TRACKS_ENABLED; 5632 } 5633 } 5634 } 5635 // compute volume for this track 5636 processVolume_l(track, last); 5637 } 5638 5639 // make sure the pause/flush/resume sequence is executed in the right order. 5640 // If a flush is pending and a track is active but the HW is not paused, force a HW pause 5641 // before flush and then resume HW. This can happen in case of pause/flush/resume 5642 // if resume is received before pause is executed. 5643 if (!mStandby && (doHwPause || (mFlushPending && !mHwPaused && (count != 0)))) { 5644 mOutput->stream->pause(mOutput->stream); 5645 } 5646 if (mFlushPending) { 5647 flushHw_l(); 5648 } 5649 if (!mStandby && doHwResume) { 5650 mOutput->stream->resume(mOutput->stream); 5651 } 5652 5653 // remove all the tracks that need to be... 5654 removeTracks_l(*tracksToRemove); 5655 5656 return mixerStatus; 5657 } 5658 5659 // must be called with thread mutex locked 5660 bool AudioFlinger::OffloadThread::waitingAsyncCallback_l() 5661 { 5662 ALOGVV("waitingAsyncCallback_l mWriteAckSequence %d mDrainSequence %d", 5663 mWriteAckSequence, mDrainSequence); 5664 if (mUseAsyncWrite && ((mWriteAckSequence & 1) || (mDrainSequence & 1))) { 5665 return true; 5666 } 5667 return false; 5668 } 5669 5670 bool AudioFlinger::OffloadThread::waitingAsyncCallback() 5671 { 5672 Mutex::Autolock _l(mLock); 5673 return waitingAsyncCallback_l(); 5674 } 5675 5676 void AudioFlinger::OffloadThread::flushHw_l() 5677 { 5678 DirectOutputThread::flushHw_l(); 5679 // Flush anything still waiting in the mixbuffer 5680 mCurrentWriteLength = 0; 5681 mBytesRemaining = 0; 5682 mPausedWriteLength = 0; 5683 mPausedBytesRemaining = 0; 5684 // reset bytes written count to reflect that DSP buffers are empty after flush. 5685 mBytesWritten = 0; 5686 mOffloadUnderrunPosition = ~0LL; 5687 5688 if (mUseAsyncWrite) { 5689 // discard any pending drain or write ack by incrementing sequence 5690 mWriteAckSequence = (mWriteAckSequence + 2) & ~1; 5691 mDrainSequence = (mDrainSequence + 2) & ~1; 5692 ALOG_ASSERT(mCallbackThread != 0); 5693 mCallbackThread->setWriteBlocked(mWriteAckSequence); 5694 mCallbackThread->setDraining(mDrainSequence); 5695 } 5696 } 5697 5698 void AudioFlinger::OffloadThread::invalidateTracks(audio_stream_type_t streamType) 5699 { 5700 Mutex::Autolock _l(mLock); 5701 if (PlaybackThread::invalidateTracks_l(streamType)) { 5702 mFlushPending = true; 5703 } 5704 } 5705 5706 // ---------------------------------------------------------------------------- 5707 5708 AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 5709 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id, bool systemReady) 5710 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->outDevice(), 5711 systemReady, DUPLICATING), 5712 mWaitTimeMs(UINT_MAX) 5713 { 5714 addOutputTrack(mainThread); 5715 } 5716 5717 AudioFlinger::DuplicatingThread::~DuplicatingThread() 5718 { 5719 for (size_t i = 0; i < mOutputTracks.size(); i++) { 5720 mOutputTracks[i]->destroy(); 5721 } 5722 } 5723 5724 void AudioFlinger::DuplicatingThread::threadLoop_mix() 5725 { 5726 // mix buffers... 5727 if (outputsReady(outputTracks)) { 5728 mAudioMixer->process(); 5729 } else { 5730 if (mMixerBufferValid) { 5731 memset(mMixerBuffer, 0, mMixerBufferSize); 5732 } else { 5733 memset(mSinkBuffer, 0, mSinkBufferSize); 5734 } 5735 } 5736 mSleepTimeUs = 0; 5737 writeFrames = mNormalFrameCount; 5738 mCurrentWriteLength = mSinkBufferSize; 5739 mStandbyTimeNs = systemTime() + mStandbyDelayNs; 5740 } 5741 5742 void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() 5743 { 5744 if (mSleepTimeUs == 0) { 5745 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 5746 mSleepTimeUs = mActiveSleepTimeUs; 5747 } else { 5748 mSleepTimeUs = mIdleSleepTimeUs; 5749 } 5750 } else if (mBytesWritten != 0) { 5751 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 5752 writeFrames = mNormalFrameCount; 5753 memset(mSinkBuffer, 0, mSinkBufferSize); 5754 } else { 5755 // flush remaining overflow buffers in output tracks 5756 writeFrames = 0; 5757 } 5758 mSleepTimeUs = 0; 5759 } 5760 } 5761 5762 ssize_t AudioFlinger::DuplicatingThread::threadLoop_write() 5763 { 5764 for (size_t i = 0; i < outputTracks.size(); i++) { 5765 outputTracks[i]->write(mSinkBuffer, writeFrames); 5766 } 5767 mStandby = false; 5768 return (ssize_t)mSinkBufferSize; 5769 } 5770 5771 void AudioFlinger::DuplicatingThread::threadLoop_standby() 5772 { 5773 // DuplicatingThread implements standby by stopping all tracks 5774 for (size_t i = 0; i < outputTracks.size(); i++) { 5775 outputTracks[i]->stop(); 5776 } 5777 } 5778 5779 void AudioFlinger::DuplicatingThread::saveOutputTracks() 5780 { 5781 outputTracks = mOutputTracks; 5782 } 5783 5784 void AudioFlinger::DuplicatingThread::clearOutputTracks() 5785 { 5786 outputTracks.clear(); 5787 } 5788 5789 void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 5790 { 5791 Mutex::Autolock _l(mLock); 5792 // The downstream MixerThread consumes thread->frameCount() amount of frames per mix pass. 5793 // Adjust for thread->sampleRate() to determine minimum buffer frame count. 5794 // Then triple buffer because Threads do not run synchronously and may not be clock locked. 5795 const size_t frameCount = 5796 3 * sourceFramesNeeded(mSampleRate, thread->frameCount(), thread->sampleRate()); 5797 // TODO: Consider asynchronous sample rate conversion to handle clock disparity 5798 // from different OutputTracks and their associated MixerThreads (e.g. one may 5799 // nearly empty and the other may be dropping data). 5800 5801 sp<OutputTrack> outputTrack = new OutputTrack(thread, 5802 this, 5803 mSampleRate, 5804 mFormat, 5805 mChannelMask, 5806 frameCount, 5807 IPCThreadState::self()->getCallingUid()); 5808 status_t status = outputTrack != 0 ? outputTrack->initCheck() : (status_t) NO_MEMORY; 5809 if (status != NO_ERROR) { 5810 ALOGE("addOutputTrack() initCheck failed %d", status); 5811 return; 5812 } 5813 thread->setStreamVolume(AUDIO_STREAM_PATCH, 1.0f); 5814 mOutputTracks.add(outputTrack); 5815 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack.get(), thread); 5816 updateWaitTime_l(); 5817 } 5818 5819 void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 5820 { 5821 Mutex::Autolock _l(mLock); 5822 for (size_t i = 0; i < mOutputTracks.size(); i++) { 5823 if (mOutputTracks[i]->thread() == thread) { 5824 mOutputTracks[i]->destroy(); 5825 mOutputTracks.removeAt(i); 5826 updateWaitTime_l(); 5827 if (thread->getOutput() == mOutput) { 5828 mOutput = NULL; 5829 } 5830 return; 5831 } 5832 } 5833 ALOGV("removeOutputTrack(): unknown thread: %p", thread); 5834 } 5835 5836 // caller must hold mLock 5837 void AudioFlinger::DuplicatingThread::updateWaitTime_l() 5838 { 5839 mWaitTimeMs = UINT_MAX; 5840 for (size_t i = 0; i < mOutputTracks.size(); i++) { 5841 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 5842 if (strong != 0) { 5843 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 5844 if (waitTimeMs < mWaitTimeMs) { 5845 mWaitTimeMs = waitTimeMs; 5846 } 5847 } 5848 } 5849 } 5850 5851 5852 bool AudioFlinger::DuplicatingThread::outputsReady( 5853 const SortedVector< sp<OutputTrack> > &outputTracks) 5854 { 5855 for (size_t i = 0; i < outputTracks.size(); i++) { 5856 sp<ThreadBase> thread = outputTracks[i]->thread().promote(); 5857 if (thread == 0) { 5858 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", 5859 outputTracks[i].get()); 5860 return false; 5861 } 5862 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 5863 // see note at standby() declaration 5864 if (playbackThread->standby() && !playbackThread->isSuspended()) { 5865 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), 5866 thread.get()); 5867 return false; 5868 } 5869 } 5870 return true; 5871 } 5872 5873 uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() const 5874 { 5875 return (mWaitTimeMs * 1000) / 2; 5876 } 5877 5878 void AudioFlinger::DuplicatingThread::cacheParameters_l() 5879 { 5880 // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first 5881 updateWaitTime_l(); 5882 5883 MixerThread::cacheParameters_l(); 5884 } 5885 5886 // ---------------------------------------------------------------------------- 5887 // Record 5888 // ---------------------------------------------------------------------------- 5889 5890 AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 5891 AudioStreamIn *input, 5892 audio_io_handle_t id, 5893 audio_devices_t outDevice, 5894 audio_devices_t inDevice, 5895 bool systemReady 5896 #ifdef TEE_SINK 5897 , const sp<NBAIO_Sink>& teeSink 5898 #endif 5899 ) : 5900 ThreadBase(audioFlinger, id, outDevice, inDevice, RECORD, systemReady), 5901 mInput(input), mActiveTracksGen(0), mRsmpInBuffer(NULL), 5902 // mRsmpInFrames and mRsmpInFramesP2 are set by readInputParameters_l() 5903 mRsmpInRear(0) 5904 #ifdef TEE_SINK 5905 , mTeeSink(teeSink) 5906 #endif 5907 , mReadOnlyHeap(new MemoryDealer(kRecordThreadReadOnlyHeapSize, 5908 "RecordThreadRO", MemoryHeapBase::READ_ONLY)) 5909 // mFastCapture below 5910 , mFastCaptureFutex(0) 5911 // mInputSource 5912 // mPipeSink 5913 // mPipeSource 5914 , mPipeFramesP2(0) 5915 // mPipeMemory 5916 // mFastCaptureNBLogWriter 5917 , mFastTrackAvail(false) 5918 { 5919 snprintf(mThreadName, kThreadNameLength, "AudioIn_%X", id); 5920 mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mThreadName); 5921 5922 readInputParameters_l(); 5923 5924 // create an NBAIO source for the HAL input stream, and negotiate 5925 mInputSource = new AudioStreamInSource(input->stream); 5926 size_t numCounterOffers = 0; 5927 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount, mFormat)}; 5928 #if !LOG_NDEBUG 5929 ssize_t index = 5930 #else 5931 (void) 5932 #endif 5933 mInputSource->negotiate(offers, 1, NULL, numCounterOffers); 5934 ALOG_ASSERT(index == 0); 5935 5936 // initialize fast capture depending on configuration 5937 bool initFastCapture; 5938 switch (kUseFastCapture) { 5939 case FastCapture_Never: 5940 initFastCapture = false; 5941 break; 5942 case FastCapture_Always: 5943 initFastCapture = true; 5944 break; 5945 case FastCapture_Static: 5946 initFastCapture = (mFrameCount * 1000) / mSampleRate < kMinNormalCaptureBufferSizeMs; 5947 break; 5948 // case FastCapture_Dynamic: 5949 } 5950 5951 if (initFastCapture) { 5952 // create a Pipe for FastCapture to write to, and for us and fast tracks to read from 5953 NBAIO_Format format = mInputSource->format(); 5954 size_t pipeFramesP2 = roundup(mSampleRate / 25); // double-buffering of 20 ms each 5955 size_t pipeSize = pipeFramesP2 * Format_frameSize(format); 5956 void *pipeBuffer; 5957 const sp<MemoryDealer> roHeap(readOnlyHeap()); 5958 sp<IMemory> pipeMemory; 5959 if ((roHeap == 0) || 5960 (pipeMemory = roHeap->allocate(pipeSize)) == 0 || 5961 (pipeBuffer = pipeMemory->pointer()) == NULL) { 5962 ALOGE("not enough memory for pipe buffer size=%zu", pipeSize); 5963 goto failed; 5964 } 5965 // pipe will be shared directly with fast clients, so clear to avoid leaking old information 5966 memset(pipeBuffer, 0, pipeSize); 5967 Pipe *pipe = new Pipe(pipeFramesP2, format, pipeBuffer); 5968 const NBAIO_Format offers[1] = {format}; 5969 size_t numCounterOffers = 0; 5970 ssize_t index = pipe->negotiate(offers, 1, NULL, numCounterOffers); 5971 ALOG_ASSERT(index == 0); 5972 mPipeSink = pipe; 5973 PipeReader *pipeReader = new PipeReader(*pipe); 5974 numCounterOffers = 0; 5975 index = pipeReader->negotiate(offers, 1, NULL, numCounterOffers); 5976 ALOG_ASSERT(index == 0); 5977 mPipeSource = pipeReader; 5978 mPipeFramesP2 = pipeFramesP2; 5979 mPipeMemory = pipeMemory; 5980 5981 // create fast capture 5982 mFastCapture = new FastCapture(); 5983 FastCaptureStateQueue *sq = mFastCapture->sq(); 5984 #ifdef STATE_QUEUE_DUMP 5985 // FIXME 5986 #endif 5987 FastCaptureState *state = sq->begin(); 5988 state->mCblk = NULL; 5989 state->mInputSource = mInputSource.get(); 5990 state->mInputSourceGen++; 5991 state->mPipeSink = pipe; 5992 state->mPipeSinkGen++; 5993 state->mFrameCount = mFrameCount; 5994 state->mCommand = FastCaptureState::COLD_IDLE; 5995 // already done in constructor initialization list 5996 //mFastCaptureFutex = 0; 5997 state->mColdFutexAddr = &mFastCaptureFutex; 5998 state->mColdGen++; 5999 state->mDumpState = &mFastCaptureDumpState; 6000 #ifdef TEE_SINK 6001 // FIXME 6002 #endif 6003 mFastCaptureNBLogWriter = audioFlinger->newWriter_l(kFastCaptureLogSize, "FastCapture"); 6004 state->mNBLogWriter = mFastCaptureNBLogWriter.get(); 6005 sq->end(); 6006 sq->push(FastCaptureStateQueue::BLOCK_UNTIL_PUSHED); 6007 6008 // start the fast capture 6009 mFastCapture->run("FastCapture", ANDROID_PRIORITY_URGENT_AUDIO); 6010 pid_t tid = mFastCapture->getTid(); 6011 sendPrioConfigEvent(getpid_cached, tid, kPriorityFastCapture); 6012 #ifdef AUDIO_WATCHDOG 6013 // FIXME 6014 #endif 6015 6016 mFastTrackAvail = true; 6017 } 6018 failed: ; 6019 6020 // FIXME mNormalSource 6021 } 6022 6023 AudioFlinger::RecordThread::~RecordThread() 6024 { 6025 if (mFastCapture != 0) { 6026 FastCaptureStateQueue *sq = mFastCapture->sq(); 6027 FastCaptureState *state = sq->begin(); 6028 if (state->mCommand == FastCaptureState::COLD_IDLE) { 6029 int32_t old = android_atomic_inc(&mFastCaptureFutex); 6030 if (old == -1) { 6031 (void) syscall(__NR_futex, &mFastCaptureFutex, FUTEX_WAKE_PRIVATE, 1); 6032 } 6033 } 6034 state->mCommand = FastCaptureState::EXIT; 6035 sq->end(); 6036 sq->push(FastCaptureStateQueue::BLOCK_UNTIL_PUSHED); 6037 mFastCapture->join(); 6038 mFastCapture.clear(); 6039 } 6040 mAudioFlinger->unregisterWriter(mFastCaptureNBLogWriter); 6041 mAudioFlinger->unregisterWriter(mNBLogWriter); 6042 free(mRsmpInBuffer); 6043 } 6044 6045 void AudioFlinger::RecordThread::onFirstRef() 6046 { 6047 run(mThreadName, PRIORITY_URGENT_AUDIO); 6048 } 6049 6050 bool AudioFlinger::RecordThread::threadLoop() 6051 { 6052 nsecs_t lastWarning = 0; 6053 6054 inputStandBy(); 6055 6056 reacquire_wakelock: 6057 sp<RecordTrack> activeTrack; 6058 int activeTracksGen; 6059 { 6060 Mutex::Autolock _l(mLock); 6061 size_t size = mActiveTracks.size(); 6062 activeTracksGen = mActiveTracksGen; 6063 if (size > 0) { 6064 // FIXME an arbitrary choice 6065 activeTrack = mActiveTracks[0]; 6066 acquireWakeLock_l(activeTrack->uid()); 6067 if (size > 1) { 6068 SortedVector<int> tmp; 6069 for (size_t i = 0; i < size; i++) { 6070 tmp.add(mActiveTracks[i]->uid()); 6071 } 6072 updateWakeLockUids_l(tmp); 6073 } 6074 } else { 6075 acquireWakeLock_l(-1); 6076 } 6077 } 6078 6079 // used to request a deferred sleep, to be executed later while mutex is unlocked 6080 uint32_t sleepUs = 0; 6081 6082 // loop while there is work to do 6083 for (;;) { 6084 Vector< sp<EffectChain> > effectChains; 6085 6086 // activeTracks accumulates a copy of a subset of mActiveTracks 6087 Vector< sp<RecordTrack> > activeTracks; 6088 6089 // reference to the (first and only) active fast track 6090 sp<RecordTrack> fastTrack; 6091 6092 // reference to a fast track which is about to be removed 6093 sp<RecordTrack> fastTrackToRemove; 6094 6095 { // scope for mLock 6096 Mutex::Autolock _l(mLock); 6097 6098 processConfigEvents_l(); 6099 6100 // check exitPending here because checkForNewParameters_l() and 6101 // checkForNewParameters_l() can temporarily release mLock 6102 if (exitPending()) { 6103 break; 6104 } 6105 6106 // sleep with mutex unlocked 6107 if (sleepUs > 0) { 6108 ATRACE_BEGIN("sleepC"); 6109 mWaitWorkCV.waitRelative(mLock, microseconds((nsecs_t)sleepUs)); 6110 ATRACE_END(); 6111 sleepUs = 0; 6112 continue; 6113 } 6114 6115 // if no active track(s), then standby and release wakelock 6116 size_t size = mActiveTracks.size(); 6117 if (size == 0) { 6118 standbyIfNotAlreadyInStandby(); 6119 // exitPending() can't become true here 6120 releaseWakeLock_l(); 6121 ALOGV("RecordThread: loop stopping"); 6122 // go to sleep 6123 mWaitWorkCV.wait(mLock); 6124 ALOGV("RecordThread: loop starting"); 6125 goto reacquire_wakelock; 6126 } 6127 6128 if (mActiveTracksGen != activeTracksGen) { 6129 activeTracksGen = mActiveTracksGen; 6130 SortedVector<int> tmp; 6131 for (size_t i = 0; i < size; i++) { 6132 tmp.add(mActiveTracks[i]->uid()); 6133 } 6134 updateWakeLockUids_l(tmp); 6135 } 6136 6137 bool doBroadcast = false; 6138 bool allStopped = true; 6139 for (size_t i = 0; i < size; ) { 6140 6141 activeTrack = mActiveTracks[i]; 6142 if (activeTrack->isTerminated()) { 6143 if (activeTrack->isFastTrack()) { 6144 ALOG_ASSERT(fastTrackToRemove == 0); 6145 fastTrackToRemove = activeTrack; 6146 } 6147 removeTrack_l(activeTrack); 6148 mActiveTracks.remove(activeTrack); 6149 mActiveTracksGen++; 6150 size--; 6151 continue; 6152 } 6153 6154 TrackBase::track_state activeTrackState = activeTrack->mState; 6155 switch (activeTrackState) { 6156 6157 case TrackBase::PAUSING: 6158 mActiveTracks.remove(activeTrack); 6159 mActiveTracksGen++; 6160 doBroadcast = true; 6161 size--; 6162 continue; 6163 6164 case TrackBase::STARTING_1: 6165 sleepUs = 10000; 6166 i++; 6167 allStopped = false; 6168 continue; 6169 6170 case TrackBase::STARTING_2: 6171 doBroadcast = true; 6172 mStandby = false; 6173 activeTrack->mState = TrackBase::ACTIVE; 6174 allStopped = false; 6175 break; 6176 6177 case TrackBase::ACTIVE: 6178 allStopped = false; 6179 break; 6180 6181 case TrackBase::IDLE: 6182 i++; 6183 continue; 6184 6185 default: 6186 LOG_ALWAYS_FATAL("Unexpected activeTrackState %d", activeTrackState); 6187 } 6188 6189 activeTracks.add(activeTrack); 6190 i++; 6191 6192 if (activeTrack->isFastTrack()) { 6193 ALOG_ASSERT(!mFastTrackAvail); 6194 ALOG_ASSERT(fastTrack == 0); 6195 fastTrack = activeTrack; 6196 } 6197 } 6198 6199 if (allStopped) { 6200 standbyIfNotAlreadyInStandby(); 6201 } 6202 if (doBroadcast) { 6203 mStartStopCond.broadcast(); 6204 } 6205 6206 // sleep if there are no active tracks to process 6207 if (activeTracks.size() == 0) { 6208 if (sleepUs == 0) { 6209 sleepUs = kRecordThreadSleepUs; 6210 } 6211 continue; 6212 } 6213 sleepUs = 0; 6214 6215 lockEffectChains_l(effectChains); 6216 } 6217 6218 // thread mutex is now unlocked, mActiveTracks unknown, activeTracks.size() > 0 6219 6220 size_t size = effectChains.size(); 6221 for (size_t i = 0; i < size; i++) { 6222 // thread mutex is not locked, but effect chain is locked 6223 effectChains[i]->process_l(); 6224 } 6225 6226 // Push a new fast capture state if fast capture is not already running, or cblk change 6227 if (mFastCapture != 0) { 6228 FastCaptureStateQueue *sq = mFastCapture->sq(); 6229 FastCaptureState *state = sq->begin(); 6230 bool didModify = false; 6231 FastCaptureStateQueue::block_t block = FastCaptureStateQueue::BLOCK_UNTIL_PUSHED; 6232 if (state->mCommand != FastCaptureState::READ_WRITE /* FIXME && 6233 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)*/) { 6234 if (state->mCommand == FastCaptureState::COLD_IDLE) { 6235 int32_t old = android_atomic_inc(&mFastCaptureFutex); 6236 if (old == -1) { 6237 (void) syscall(__NR_futex, &mFastCaptureFutex, FUTEX_WAKE_PRIVATE, 1); 6238 } 6239 } 6240 state->mCommand = FastCaptureState::READ_WRITE; 6241 #if 0 // FIXME 6242 mFastCaptureDumpState.increaseSamplingN(mAudioFlinger->isLowRamDevice() ? 6243 FastThreadDumpState::kSamplingNforLowRamDevice : 6244 FastThreadDumpState::kSamplingN); 6245 #endif 6246 didModify = true; 6247 } 6248 audio_track_cblk_t *cblkOld = state->mCblk; 6249 audio_track_cblk_t *cblkNew = fastTrack != 0 ? fastTrack->cblk() : NULL; 6250 if (cblkNew != cblkOld) { 6251 state->mCblk = cblkNew; 6252 // block until acked if removing a fast track 6253 if (cblkOld != NULL) { 6254 block = FastCaptureStateQueue::BLOCK_UNTIL_ACKED; 6255 } 6256 didModify = true; 6257 } 6258 sq->end(didModify); 6259 if (didModify) { 6260 sq->push(block); 6261 #if 0 6262 if (kUseFastCapture == FastCapture_Dynamic) { 6263 mNormalSource = mPipeSource; 6264 } 6265 #endif 6266 } 6267 } 6268 6269 // now run the fast track destructor with thread mutex unlocked 6270 fastTrackToRemove.clear(); 6271 6272 // Read from HAL to keep up with fastest client if multiple active tracks, not slowest one. 6273 // Only the client(s) that are too slow will overrun. But if even the fastest client is too 6274 // slow, then this RecordThread will overrun by not calling HAL read often enough. 6275 // If destination is non-contiguous, first read past the nominal end of buffer, then 6276 // copy to the right place. Permitted because mRsmpInBuffer was over-allocated. 6277 6278 int32_t rear = mRsmpInRear & (mRsmpInFramesP2 - 1); 6279 ssize_t framesRead; 6280 6281 // If an NBAIO source is present, use it to read the normal capture's data 6282 if (mPipeSource != 0) { 6283 size_t framesToRead = mBufferSize / mFrameSize; 6284 framesRead = mPipeSource->read((uint8_t*)mRsmpInBuffer + rear * mFrameSize, 6285 framesToRead); 6286 if (framesRead == 0) { 6287 // since pipe is non-blocking, simulate blocking input 6288 sleepUs = (framesToRead * 1000000LL) / mSampleRate; 6289 } 6290 // otherwise use the HAL / AudioStreamIn directly 6291 } else { 6292 ATRACE_BEGIN("read"); 6293 ssize_t bytesRead = mInput->stream->read(mInput->stream, 6294 (uint8_t*)mRsmpInBuffer + rear * mFrameSize, mBufferSize); 6295 ATRACE_END(); 6296 if (bytesRead < 0) { 6297 framesRead = bytesRead; 6298 } else { 6299 framesRead = bytesRead / mFrameSize; 6300 } 6301 } 6302 6303 // Update server timestamp with server stats 6304 // systemTime() is optional if the hardware supports timestamps. 6305 mTimestamp.mPosition[ExtendedTimestamp::LOCATION_SERVER] += framesRead; 6306 mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_SERVER] = systemTime(); 6307 6308 // Update server timestamp with kernel stats 6309 if (mInput->stream->get_capture_position != nullptr 6310 && mPipeSource.get() == nullptr /* don't obtain for FastCapture, could block */) { 6311 int64_t position, time; 6312 int ret = mInput->stream->get_capture_position(mInput->stream, &position, &time); 6313 if (ret == NO_ERROR) { 6314 mTimestamp.mPosition[ExtendedTimestamp::LOCATION_KERNEL] = position; 6315 mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL] = time; 6316 // Note: In general record buffers should tend to be empty in 6317 // a properly running pipeline. 6318 // 6319 // Also, it is not advantageous to call get_presentation_position during the read 6320 // as the read obtains a lock, preventing the timestamp call from executing. 6321 } 6322 } 6323 // Use this to track timestamp information 6324 // ALOGD("%s", mTimestamp.toString().c_str()); 6325 6326 if (framesRead < 0 || (framesRead == 0 && mPipeSource == 0)) { 6327 ALOGE("read failed: framesRead=%zd", framesRead); 6328 // Force input into standby so that it tries to recover at next read attempt 6329 inputStandBy(); 6330 sleepUs = kRecordThreadSleepUs; 6331 } 6332 if (framesRead <= 0) { 6333 goto unlock; 6334 } 6335 ALOG_ASSERT(framesRead > 0); 6336 6337 if (mTeeSink != 0) { 6338 (void) mTeeSink->write((uint8_t*)mRsmpInBuffer + rear * mFrameSize, framesRead); 6339 } 6340 // If destination is non-contiguous, we now correct for reading past end of buffer. 6341 { 6342 size_t part1 = mRsmpInFramesP2 - rear; 6343 if ((size_t) framesRead > part1) { 6344 memcpy(mRsmpInBuffer, (uint8_t*)mRsmpInBuffer + mRsmpInFramesP2 * mFrameSize, 6345 (framesRead - part1) * mFrameSize); 6346 } 6347 } 6348 rear = mRsmpInRear += framesRead; 6349 6350 size = activeTracks.size(); 6351 // loop over each active track 6352 for (size_t i = 0; i < size; i++) { 6353 activeTrack = activeTracks[i]; 6354 6355 // skip fast tracks, as those are handled directly by FastCapture 6356 if (activeTrack->isFastTrack()) { 6357 continue; 6358 } 6359 6360 // TODO: This code probably should be moved to RecordTrack. 6361 // TODO: Update the activeTrack buffer converter in case of reconfigure. 6362 6363 enum { 6364 OVERRUN_UNKNOWN, 6365 OVERRUN_TRUE, 6366 OVERRUN_FALSE 6367 } overrun = OVERRUN_UNKNOWN; 6368 6369 // loop over getNextBuffer to handle circular sink 6370 for (;;) { 6371 6372 activeTrack->mSink.frameCount = ~0; 6373 status_t status = activeTrack->getNextBuffer(&activeTrack->mSink); 6374 size_t framesOut = activeTrack->mSink.frameCount; 6375 LOG_ALWAYS_FATAL_IF((status == OK) != (framesOut > 0)); 6376 6377 // check available frames and handle overrun conditions 6378 // if the record track isn't draining fast enough. 6379 bool hasOverrun; 6380 size_t framesIn; 6381 activeTrack->mResamplerBufferProvider->sync(&framesIn, &hasOverrun); 6382 if (hasOverrun) { 6383 overrun = OVERRUN_TRUE; 6384 } 6385 if (framesOut == 0 || framesIn == 0) { 6386 break; 6387 } 6388 6389 // Don't allow framesOut to be larger than what is possible with resampling 6390 // from framesIn. 6391 // This isn't strictly necessary but helps limit buffer resizing in 6392 // RecordBufferConverter. TODO: remove when no longer needed. 6393 framesOut = min(framesOut, 6394 destinationFramesPossible( 6395 framesIn, mSampleRate, activeTrack->mSampleRate)); 6396 // process frames from the RecordThread buffer provider to the RecordTrack buffer 6397 framesOut = activeTrack->mRecordBufferConverter->convert( 6398 activeTrack->mSink.raw, activeTrack->mResamplerBufferProvider, framesOut); 6399 6400 if (framesOut > 0 && (overrun == OVERRUN_UNKNOWN)) { 6401 overrun = OVERRUN_FALSE; 6402 } 6403 6404 if (activeTrack->mFramesToDrop == 0) { 6405 if (framesOut > 0) { 6406 activeTrack->mSink.frameCount = framesOut; 6407 activeTrack->releaseBuffer(&activeTrack->mSink); 6408 } 6409 } else { 6410 // FIXME could do a partial drop of framesOut 6411 if (activeTrack->mFramesToDrop > 0) { 6412 activeTrack->mFramesToDrop -= framesOut; 6413 if (activeTrack->mFramesToDrop <= 0) { 6414 activeTrack->clearSyncStartEvent(); 6415 } 6416 } else { 6417 activeTrack->mFramesToDrop += framesOut; 6418 if (activeTrack->mFramesToDrop >= 0 || activeTrack->mSyncStartEvent == 0 || 6419 activeTrack->mSyncStartEvent->isCancelled()) { 6420 ALOGW("Synced record %s, session %d, trigger session %d", 6421 (activeTrack->mFramesToDrop >= 0) ? "timed out" : "cancelled", 6422 activeTrack->sessionId(), 6423 (activeTrack->mSyncStartEvent != 0) ? 6424 activeTrack->mSyncStartEvent->triggerSession() : 6425 AUDIO_SESSION_NONE); 6426 activeTrack->clearSyncStartEvent(); 6427 } 6428 } 6429 } 6430 6431 if (framesOut == 0) { 6432 break; 6433 } 6434 } 6435 6436 switch (overrun) { 6437 case OVERRUN_TRUE: 6438 // client isn't retrieving buffers fast enough 6439 if (!activeTrack->setOverflow()) { 6440 nsecs_t now = systemTime(); 6441 // FIXME should lastWarning per track? 6442 if ((now - lastWarning) > kWarningThrottleNs) { 6443 ALOGW("RecordThread: buffer overflow"); 6444 lastWarning = now; 6445 } 6446 } 6447 break; 6448 case OVERRUN_FALSE: 6449 activeTrack->clearOverflow(); 6450 break; 6451 case OVERRUN_UNKNOWN: 6452 break; 6453 } 6454 6455 // update frame information and push timestamp out 6456 activeTrack->updateTrackFrameInfo( 6457 activeTrack->mServerProxy->framesReleased(), 6458 mTimestamp.mPosition[ExtendedTimestamp::LOCATION_SERVER], 6459 mSampleRate, mTimestamp); 6460 } 6461 6462 unlock: 6463 // enable changes in effect chain 6464 unlockEffectChains(effectChains); 6465 // effectChains doesn't need to be cleared, since it is cleared by destructor at scope end 6466 } 6467 6468 standbyIfNotAlreadyInStandby(); 6469 6470 { 6471 Mutex::Autolock _l(mLock); 6472 for (size_t i = 0; i < mTracks.size(); i++) { 6473 sp<RecordTrack> track = mTracks[i]; 6474 track->invalidate(); 6475 } 6476 mActiveTracks.clear(); 6477 mActiveTracksGen++; 6478 mStartStopCond.broadcast(); 6479 } 6480 6481 releaseWakeLock(); 6482 6483 ALOGV("RecordThread %p exiting", this); 6484 return false; 6485 } 6486 6487 void AudioFlinger::RecordThread::standbyIfNotAlreadyInStandby() 6488 { 6489 if (!mStandby) { 6490 inputStandBy(); 6491 mStandby = true; 6492 } 6493 } 6494 6495 void AudioFlinger::RecordThread::inputStandBy() 6496 { 6497 // Idle the fast capture if it's currently running 6498 if (mFastCapture != 0) { 6499 FastCaptureStateQueue *sq = mFastCapture->sq(); 6500 FastCaptureState *state = sq->begin(); 6501 if (!(state->mCommand & FastCaptureState::IDLE)) { 6502 state->mCommand = FastCaptureState::COLD_IDLE; 6503 state->mColdFutexAddr = &mFastCaptureFutex; 6504 state->mColdGen++; 6505 mFastCaptureFutex = 0; 6506 sq->end(); 6507 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 6508 sq->push(FastCaptureStateQueue::BLOCK_UNTIL_ACKED); 6509 #if 0 6510 if (kUseFastCapture == FastCapture_Dynamic) { 6511 // FIXME 6512 } 6513 #endif 6514 #ifdef AUDIO_WATCHDOG 6515 // FIXME 6516 #endif 6517 } else { 6518 sq->end(false /*didModify*/); 6519 } 6520 } 6521 mInput->stream->common.standby(&mInput->stream->common); 6522 } 6523 6524 // RecordThread::createRecordTrack_l() must be called with AudioFlinger::mLock held 6525 sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 6526 const sp<AudioFlinger::Client>& client, 6527 uint32_t sampleRate, 6528 audio_format_t format, 6529 audio_channel_mask_t channelMask, 6530 size_t *pFrameCount, 6531 audio_session_t sessionId, 6532 size_t *notificationFrames, 6533 int uid, 6534 audio_input_flags_t *flags, 6535 pid_t tid, 6536 status_t *status) 6537 { 6538 size_t frameCount = *pFrameCount; 6539 sp<RecordTrack> track; 6540 status_t lStatus; 6541 audio_input_flags_t inputFlags = mInput->flags; 6542 6543 // special case for FAST flag considered OK if fast capture is present 6544 if (hasFastCapture()) { 6545 inputFlags = (audio_input_flags_t)(inputFlags | AUDIO_INPUT_FLAG_FAST); 6546 } 6547 6548 // Check if requested flags are compatible with output stream flags 6549 if ((*flags & inputFlags) != *flags) { 6550 ALOGW("createRecordTrack_l(): mismatch between requested flags (%08x) and" 6551 " input flags (%08x)", 6552 *flags, inputFlags); 6553 *flags = (audio_input_flags_t)(*flags & inputFlags); 6554 } 6555 6556 // client expresses a preference for FAST, but we get the final say 6557 if (*flags & AUDIO_INPUT_FLAG_FAST) { 6558 if ( 6559 // we formerly checked for a callback handler (non-0 tid), 6560 // but that is no longer required for TRANSFER_OBTAIN mode 6561 // 6562 // frame count is not specified, or is exactly the pipe depth 6563 ((frameCount == 0) || (frameCount == mPipeFramesP2)) && 6564 // PCM data 6565 audio_is_linear_pcm(format) && 6566 // hardware format 6567 (format == mFormat) && 6568 // hardware channel mask 6569 (channelMask == mChannelMask) && 6570 // hardware sample rate 6571 (sampleRate == mSampleRate) && 6572 // record thread has an associated fast capture 6573 hasFastCapture() && 6574 // there are sufficient fast track slots available 6575 mFastTrackAvail 6576 ) { 6577 // check compatibility with audio effects. 6578 Mutex::Autolock _l(mLock); 6579 // Do not accept FAST flag if the session has software effects 6580 sp<EffectChain> chain = getEffectChain_l(sessionId); 6581 if (chain != 0) { 6582 audio_input_flags_t old = *flags; 6583 chain->checkInputFlagCompatibility(flags); 6584 if (old != *flags) { 6585 ALOGV("AUDIO_INPUT_FLAGS denied by effect old=%#x new=%#x", 6586 (int)old, (int)*flags); 6587 } 6588 } 6589 ALOGV_IF((*flags & AUDIO_INPUT_FLAG_FAST) != 0, 6590 "AUDIO_INPUT_FLAG_FAST accepted: frameCount=%zu mFrameCount=%zu", 6591 frameCount, mFrameCount); 6592 } else { 6593 ALOGV("AUDIO_INPUT_FLAG_FAST denied: frameCount=%zu mFrameCount=%zu mPipeFramesP2=%zu " 6594 "format=%#x isLinear=%d channelMask=%#x sampleRate=%u mSampleRate=%u " 6595 "hasFastCapture=%d tid=%d mFastTrackAvail=%d", 6596 frameCount, mFrameCount, mPipeFramesP2, 6597 format, audio_is_linear_pcm(format), channelMask, sampleRate, mSampleRate, 6598 hasFastCapture(), tid, mFastTrackAvail); 6599 *flags = (audio_input_flags_t)(*flags & ~AUDIO_INPUT_FLAG_FAST); 6600 } 6601 } 6602 6603 // compute track buffer size in frames, and suggest the notification frame count 6604 if (*flags & AUDIO_INPUT_FLAG_FAST) { 6605 // fast track: frame count is exactly the pipe depth 6606 frameCount = mPipeFramesP2; 6607 // ignore requested notificationFrames, and always notify exactly once every HAL buffer 6608 *notificationFrames = mFrameCount; 6609 } else { 6610 // not fast track: max notification period is resampled equivalent of one HAL buffer time 6611 // or 20 ms if there is a fast capture 6612 // TODO This could be a roundupRatio inline, and const 6613 size_t maxNotificationFrames = ((int64_t) (hasFastCapture() ? mSampleRate/50 : mFrameCount) 6614 * sampleRate + mSampleRate - 1) / mSampleRate; 6615 // minimum number of notification periods is at least kMinNotifications, 6616 // and at least kMinMs rounded up to a whole notification period (minNotificationsByMs) 6617 static const size_t kMinNotifications = 3; 6618 static const uint32_t kMinMs = 30; 6619 // TODO This could be a roundupRatio inline 6620 const size_t minFramesByMs = (sampleRate * kMinMs + 1000 - 1) / 1000; 6621 // TODO This could be a roundupRatio inline 6622 const size_t minNotificationsByMs = (minFramesByMs + maxNotificationFrames - 1) / 6623 maxNotificationFrames; 6624 const size_t minFrameCount = maxNotificationFrames * 6625 max(kMinNotifications, minNotificationsByMs); 6626 frameCount = max(frameCount, minFrameCount); 6627 if (*notificationFrames == 0 || *notificationFrames > maxNotificationFrames) { 6628 *notificationFrames = maxNotificationFrames; 6629 } 6630 } 6631 *pFrameCount = frameCount; 6632 6633 lStatus = initCheck(); 6634 if (lStatus != NO_ERROR) { 6635 ALOGE("createRecordTrack_l() audio driver not initialized"); 6636 goto Exit; 6637 } 6638 6639 { // scope for mLock 6640 Mutex::Autolock _l(mLock); 6641 6642 track = new RecordTrack(this, client, sampleRate, 6643 format, channelMask, frameCount, NULL, sessionId, uid, 6644 *flags, TrackBase::TYPE_DEFAULT); 6645 6646 lStatus = track->initCheck(); 6647 if (lStatus != NO_ERROR) { 6648 ALOGE("createRecordTrack_l() initCheck failed %d; no control block?", lStatus); 6649 // track must be cleared from the caller as the caller has the AF lock 6650 goto Exit; 6651 } 6652 mTracks.add(track); 6653 6654 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 6655 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 6656 mAudioFlinger->btNrecIsOff(); 6657 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 6658 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 6659 6660 if ((*flags & AUDIO_INPUT_FLAG_FAST) && (tid != -1)) { 6661 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 6662 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 6663 // so ask activity manager to do this on our behalf 6664 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 6665 } 6666 } 6667 6668 lStatus = NO_ERROR; 6669 6670 Exit: 6671 *status = lStatus; 6672 return track; 6673 } 6674 6675 status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, 6676 AudioSystem::sync_event_t event, 6677 audio_session_t triggerSession) 6678 { 6679 ALOGV("RecordThread::start event %d, triggerSession %d", event, triggerSession); 6680 sp<ThreadBase> strongMe = this; 6681 status_t status = NO_ERROR; 6682 6683 if (event == AudioSystem::SYNC_EVENT_NONE) { 6684 recordTrack->clearSyncStartEvent(); 6685 } else if (event != AudioSystem::SYNC_EVENT_SAME) { 6686 recordTrack->mSyncStartEvent = mAudioFlinger->createSyncEvent(event, 6687 triggerSession, 6688 recordTrack->sessionId(), 6689 syncStartEventCallback, 6690 recordTrack); 6691 // Sync event can be cancelled by the trigger session if the track is not in a 6692 // compatible state in which case we start record immediately 6693 if (recordTrack->mSyncStartEvent->isCancelled()) { 6694 recordTrack->clearSyncStartEvent(); 6695 } else { 6696 // do not wait for the event for more than AudioSystem::kSyncRecordStartTimeOutMs 6697 recordTrack->mFramesToDrop = - 6698 ((AudioSystem::kSyncRecordStartTimeOutMs * recordTrack->mSampleRate) / 1000); 6699 } 6700 } 6701 6702 { 6703 // This section is a rendezvous between binder thread executing start() and RecordThread 6704 AutoMutex lock(mLock); 6705 if (mActiveTracks.indexOf(recordTrack) >= 0) { 6706 if (recordTrack->mState == TrackBase::PAUSING) { 6707 ALOGV("active record track PAUSING -> ACTIVE"); 6708 recordTrack->mState = TrackBase::ACTIVE; 6709 } else { 6710 ALOGV("active record track state %d", recordTrack->mState); 6711 } 6712 return status; 6713 } 6714 6715 // TODO consider other ways of handling this, such as changing the state to :STARTING and 6716 // adding the track to mActiveTracks after returning from AudioSystem::startInput(), 6717 // or using a separate command thread 6718 recordTrack->mState = TrackBase::STARTING_1; 6719 mActiveTracks.add(recordTrack); 6720 mActiveTracksGen++; 6721 status_t status = NO_ERROR; 6722 if (recordTrack->isExternalTrack()) { 6723 mLock.unlock(); 6724 status = AudioSystem::startInput(mId, recordTrack->sessionId()); 6725 mLock.lock(); 6726 // FIXME should verify that recordTrack is still in mActiveTracks 6727 if (status != NO_ERROR) { 6728 mActiveTracks.remove(recordTrack); 6729 mActiveTracksGen++; 6730 recordTrack->clearSyncStartEvent(); 6731 ALOGV("RecordThread::start error %d", status); 6732 return status; 6733 } 6734 } 6735 // Catch up with current buffer indices if thread is already running. 6736 // This is what makes a new client discard all buffered data. If the track's mRsmpInFront 6737 // was initialized to some value closer to the thread's mRsmpInFront, then the track could 6738 // see previously buffered data before it called start(), but with greater risk of overrun. 6739 6740 recordTrack->mResamplerBufferProvider->reset(); 6741 // clear any converter state as new data will be discontinuous 6742 recordTrack->mRecordBufferConverter->reset(); 6743 recordTrack->mState = TrackBase::STARTING_2; 6744 // signal thread to start 6745 mWaitWorkCV.broadcast(); 6746 if (mActiveTracks.indexOf(recordTrack) < 0) { 6747 ALOGV("Record failed to start"); 6748 status = BAD_VALUE; 6749 goto startError; 6750 } 6751 return status; 6752 } 6753 6754 startError: 6755 if (recordTrack->isExternalTrack()) { 6756 AudioSystem::stopInput(mId, recordTrack->sessionId()); 6757 } 6758 recordTrack->clearSyncStartEvent(); 6759 // FIXME I wonder why we do not reset the state here? 6760 return status; 6761 } 6762 6763 void AudioFlinger::RecordThread::syncStartEventCallback(const wp<SyncEvent>& event) 6764 { 6765 sp<SyncEvent> strongEvent = event.promote(); 6766 6767 if (strongEvent != 0) { 6768 sp<RefBase> ptr = strongEvent->cookie().promote(); 6769 if (ptr != 0) { 6770 RecordTrack *recordTrack = (RecordTrack *)ptr.get(); 6771 recordTrack->handleSyncStartEvent(strongEvent); 6772 } 6773 } 6774 } 6775 6776 bool AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 6777 ALOGV("RecordThread::stop"); 6778 AutoMutex _l(mLock); 6779 if (mActiveTracks.indexOf(recordTrack) != 0 || recordTrack->mState == TrackBase::PAUSING) { 6780 return false; 6781 } 6782 // note that threadLoop may still be processing the track at this point [without lock] 6783 recordTrack->mState = TrackBase::PAUSING; 6784 // signal thread to stop 6785 mWaitWorkCV.broadcast(); 6786 // do not wait for mStartStopCond if exiting 6787 if (exitPending()) { 6788 return true; 6789 } 6790 // FIXME incorrect usage of wait: no explicit predicate or loop 6791 mStartStopCond.wait(mLock); 6792 // if we have been restarted, recordTrack is in mActiveTracks here 6793 if (exitPending() || mActiveTracks.indexOf(recordTrack) != 0) { 6794 ALOGV("Record stopped OK"); 6795 return true; 6796 } 6797 return false; 6798 } 6799 6800 bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event __unused) const 6801 { 6802 return false; 6803 } 6804 6805 status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event __unused) 6806 { 6807 #if 0 // This branch is currently dead code, but is preserved in case it will be needed in future 6808 if (!isValidSyncEvent(event)) { 6809 return BAD_VALUE; 6810 } 6811 6812 audio_session_t eventSession = event->triggerSession(); 6813 status_t ret = NAME_NOT_FOUND; 6814 6815 Mutex::Autolock _l(mLock); 6816 6817 for (size_t i = 0; i < mTracks.size(); i++) { 6818 sp<RecordTrack> track = mTracks[i]; 6819 if (eventSession == track->sessionId()) { 6820 (void) track->setSyncEvent(event); 6821 ret = NO_ERROR; 6822 } 6823 } 6824 return ret; 6825 #else 6826 return BAD_VALUE; 6827 #endif 6828 } 6829 6830 // destroyTrack_l() must be called with ThreadBase::mLock held 6831 void AudioFlinger::RecordThread::destroyTrack_l(const sp<RecordTrack>& track) 6832 { 6833 track->terminate(); 6834 track->mState = TrackBase::STOPPED; 6835 // active tracks are removed by threadLoop() 6836 if (mActiveTracks.indexOf(track) < 0) { 6837 removeTrack_l(track); 6838 } 6839 } 6840 6841 void AudioFlinger::RecordThread::removeTrack_l(const sp<RecordTrack>& track) 6842 { 6843 mTracks.remove(track); 6844 // need anything related to effects here? 6845 if (track->isFastTrack()) { 6846 ALOG_ASSERT(!mFastTrackAvail); 6847 mFastTrackAvail = true; 6848 } 6849 } 6850 6851 void AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 6852 { 6853 dumpInternals(fd, args); 6854 dumpTracks(fd, args); 6855 dumpEffectChains(fd, args); 6856 } 6857 6858 void AudioFlinger::RecordThread::dumpInternals(int fd, const Vector<String16>& args) 6859 { 6860 dprintf(fd, "\nInput thread %p:\n", this); 6861 6862 dumpBase(fd, args); 6863 6864 if (mActiveTracks.size() == 0) { 6865 dprintf(fd, " No active record clients\n"); 6866 } 6867 dprintf(fd, " Fast capture thread: %s\n", hasFastCapture() ? "yes" : "no"); 6868 dprintf(fd, " Fast track available: %s\n", mFastTrackAvail ? "yes" : "no"); 6869 6870 // Make a non-atomic copy of fast capture dump state so it won't change underneath us 6871 // while we are dumping it. It may be inconsistent, but it won't mutate! 6872 // This is a large object so we place it on the heap. 6873 // FIXME 25972958: Need an intelligent copy constructor that does not touch unused pages. 6874 const FastCaptureDumpState *copy = new FastCaptureDumpState(mFastCaptureDumpState); 6875 copy->dump(fd); 6876 delete copy; 6877 } 6878 6879 void AudioFlinger::RecordThread::dumpTracks(int fd, const Vector<String16>& args __unused) 6880 { 6881 const size_t SIZE = 256; 6882 char buffer[SIZE]; 6883 String8 result; 6884 6885 size_t numtracks = mTracks.size(); 6886 size_t numactive = mActiveTracks.size(); 6887 size_t numactiveseen = 0; 6888 dprintf(fd, " %zu Tracks", numtracks); 6889 if (numtracks) { 6890 dprintf(fd, " of which %zu are active\n", numactive); 6891 RecordTrack::appendDumpHeader(result); 6892 for (size_t i = 0; i < numtracks ; ++i) { 6893 sp<RecordTrack> track = mTracks[i]; 6894 if (track != 0) { 6895 bool active = mActiveTracks.indexOf(track) >= 0; 6896 if (active) { 6897 numactiveseen++; 6898 } 6899 track->dump(buffer, SIZE, active); 6900 result.append(buffer); 6901 } 6902 } 6903 } else { 6904 dprintf(fd, "\n"); 6905 } 6906 6907 if (numactiveseen != numactive) { 6908 snprintf(buffer, SIZE, " The following tracks are in the active list but" 6909 " not in the track list\n"); 6910 result.append(buffer); 6911 RecordTrack::appendDumpHeader(result); 6912 for (size_t i = 0; i < numactive; ++i) { 6913 sp<RecordTrack> track = mActiveTracks[i]; 6914 if (mTracks.indexOf(track) < 0) { 6915 track->dump(buffer, SIZE, true); 6916 result.append(buffer); 6917 } 6918 } 6919 6920 } 6921 write(fd, result.string(), result.size()); 6922 } 6923 6924 6925 void AudioFlinger::RecordThread::ResamplerBufferProvider::reset() 6926 { 6927 sp<ThreadBase> threadBase = mRecordTrack->mThread.promote(); 6928 RecordThread *recordThread = (RecordThread *) threadBase.get(); 6929 mRsmpInFront = recordThread->mRsmpInRear; 6930 mRsmpInUnrel = 0; 6931 } 6932 6933 void AudioFlinger::RecordThread::ResamplerBufferProvider::sync( 6934 size_t *framesAvailable, bool *hasOverrun) 6935 { 6936 sp<ThreadBase> threadBase = mRecordTrack->mThread.promote(); 6937 RecordThread *recordThread = (RecordThread *) threadBase.get(); 6938 const int32_t rear = recordThread->mRsmpInRear; 6939 const int32_t front = mRsmpInFront; 6940 const ssize_t filled = rear - front; 6941 6942 size_t framesIn; 6943 bool overrun = false; 6944 if (filled < 0) { 6945 // should not happen, but treat like a massive overrun and re-sync 6946 framesIn = 0; 6947 mRsmpInFront = rear; 6948 overrun = true; 6949 } else if ((size_t) filled <= recordThread->mRsmpInFrames) { 6950 framesIn = (size_t) filled; 6951 } else { 6952 // client is not keeping up with server, but give it latest data 6953 framesIn = recordThread->mRsmpInFrames; 6954 mRsmpInFront = /* front = */ rear - framesIn; 6955 overrun = true; 6956 } 6957 if (framesAvailable != NULL) { 6958 *framesAvailable = framesIn; 6959 } 6960 if (hasOverrun != NULL) { 6961 *hasOverrun = overrun; 6962 } 6963 } 6964 6965 // AudioBufferProvider interface 6966 status_t AudioFlinger::RecordThread::ResamplerBufferProvider::getNextBuffer( 6967 AudioBufferProvider::Buffer* buffer) 6968 { 6969 sp<ThreadBase> threadBase = mRecordTrack->mThread.promote(); 6970 if (threadBase == 0) { 6971 buffer->frameCount = 0; 6972 buffer->raw = NULL; 6973 return NOT_ENOUGH_DATA; 6974 } 6975 RecordThread *recordThread = (RecordThread *) threadBase.get(); 6976 int32_t rear = recordThread->mRsmpInRear; 6977 int32_t front = mRsmpInFront; 6978 ssize_t filled = rear - front; 6979 // FIXME should not be P2 (don't want to increase latency) 6980 // FIXME if client not keeping up, discard 6981 LOG_ALWAYS_FATAL_IF(!(0 <= filled && (size_t) filled <= recordThread->mRsmpInFrames)); 6982 // 'filled' may be non-contiguous, so return only the first contiguous chunk 6983 front &= recordThread->mRsmpInFramesP2 - 1; 6984 size_t part1 = recordThread->mRsmpInFramesP2 - front; 6985 if (part1 > (size_t) filled) { 6986 part1 = filled; 6987 } 6988 size_t ask = buffer->frameCount; 6989 ALOG_ASSERT(ask > 0); 6990 if (part1 > ask) { 6991 part1 = ask; 6992 } 6993 if (part1 == 0) { 6994 // out of data is fine since the resampler will return a short-count. 6995 buffer->raw = NULL; 6996 buffer->frameCount = 0; 6997 mRsmpInUnrel = 0; 6998 return NOT_ENOUGH_DATA; 6999 } 7000 7001 buffer->raw = (uint8_t*)recordThread->mRsmpInBuffer + front * recordThread->mFrameSize; 7002 buffer->frameCount = part1; 7003 mRsmpInUnrel = part1; 7004 return NO_ERROR; 7005 } 7006 7007 // AudioBufferProvider interface 7008 void AudioFlinger::RecordThread::ResamplerBufferProvider::releaseBuffer( 7009 AudioBufferProvider::Buffer* buffer) 7010 { 7011 size_t stepCount = buffer->frameCount; 7012 if (stepCount == 0) { 7013 return; 7014 } 7015 ALOG_ASSERT(stepCount <= mRsmpInUnrel); 7016 mRsmpInUnrel -= stepCount; 7017 mRsmpInFront += stepCount; 7018 buffer->raw = NULL; 7019 buffer->frameCount = 0; 7020 } 7021 7022 AudioFlinger::RecordThread::RecordBufferConverter::RecordBufferConverter( 7023 audio_channel_mask_t srcChannelMask, audio_format_t srcFormat, 7024 uint32_t srcSampleRate, 7025 audio_channel_mask_t dstChannelMask, audio_format_t dstFormat, 7026 uint32_t dstSampleRate) : 7027 mSrcChannelMask(AUDIO_CHANNEL_INVALID), // updateParameters will set following vars 7028 // mSrcFormat 7029 // mSrcSampleRate 7030 // mDstChannelMask 7031 // mDstFormat 7032 // mDstSampleRate 7033 // mSrcChannelCount 7034 // mDstChannelCount 7035 // mDstFrameSize 7036 mBuf(NULL), mBufFrames(0), mBufFrameSize(0), 7037 mResampler(NULL), 7038 mIsLegacyDownmix(false), 7039 mIsLegacyUpmix(false), 7040 mRequiresFloat(false), 7041 mInputConverterProvider(NULL) 7042 { 7043 (void)updateParameters(srcChannelMask, srcFormat, srcSampleRate, 7044 dstChannelMask, dstFormat, dstSampleRate); 7045 } 7046 7047 AudioFlinger::RecordThread::RecordBufferConverter::~RecordBufferConverter() { 7048 free(mBuf); 7049 delete mResampler; 7050 delete mInputConverterProvider; 7051 } 7052 7053 size_t AudioFlinger::RecordThread::RecordBufferConverter::convert(void *dst, 7054 AudioBufferProvider *provider, size_t frames) 7055 { 7056 if (mInputConverterProvider != NULL) { 7057 mInputConverterProvider->setBufferProvider(provider); 7058 provider = mInputConverterProvider; 7059 } 7060 7061 if (mResampler == NULL) { 7062 ALOGVV("NO RESAMPLING sampleRate:%u mSrcFormat:%#x mDstFormat:%#x", 7063 mSrcSampleRate, mSrcFormat, mDstFormat); 7064 7065 AudioBufferProvider::Buffer buffer; 7066 for (size_t i = frames; i > 0; ) { 7067 buffer.frameCount = i; 7068 status_t status = provider->getNextBuffer(&buffer); 7069 if (status != OK || buffer.frameCount == 0) { 7070 frames -= i; // cannot fill request. 7071 break; 7072 } 7073 // format convert to destination buffer 7074 convertNoResampler(dst, buffer.raw, buffer.frameCount); 7075 7076 dst = (int8_t*)dst + buffer.frameCount * mDstFrameSize; 7077 i -= buffer.frameCount; 7078 provider->releaseBuffer(&buffer); 7079 } 7080 } else { 7081 ALOGVV("RESAMPLING mSrcSampleRate:%u mDstSampleRate:%u mSrcFormat:%#x mDstFormat:%#x", 7082 mSrcSampleRate, mDstSampleRate, mSrcFormat, mDstFormat); 7083 7084 // reallocate buffer if needed 7085 if (mBufFrameSize != 0 && mBufFrames < frames) { 7086 free(mBuf); 7087 mBufFrames = frames; 7088 (void)posix_memalign(&mBuf, 32, mBufFrames * mBufFrameSize); 7089 } 7090 // resampler accumulates, but we only have one source track 7091 memset(mBuf, 0, frames * mBufFrameSize); 7092 frames = mResampler->resample((int32_t*)mBuf, frames, provider); 7093 // format convert to destination buffer 7094 convertResampler(dst, mBuf, frames); 7095 } 7096 return frames; 7097 } 7098 7099 status_t AudioFlinger::RecordThread::RecordBufferConverter::updateParameters( 7100 audio_channel_mask_t srcChannelMask, audio_format_t srcFormat, 7101 uint32_t srcSampleRate, 7102 audio_channel_mask_t dstChannelMask, audio_format_t dstFormat, 7103 uint32_t dstSampleRate) 7104 { 7105 // quick evaluation if there is any change. 7106 if (mSrcFormat == srcFormat 7107 && mSrcChannelMask == srcChannelMask 7108 && mSrcSampleRate == srcSampleRate 7109 && mDstFormat == dstFormat 7110 && mDstChannelMask == dstChannelMask 7111 && mDstSampleRate == dstSampleRate) { 7112 return NO_ERROR; 7113 } 7114 7115 ALOGV("RecordBufferConverter updateParameters srcMask:%#x dstMask:%#x" 7116 " srcFormat:%#x dstFormat:%#x srcRate:%u dstRate:%u", 7117 srcChannelMask, dstChannelMask, srcFormat, dstFormat, srcSampleRate, dstSampleRate); 7118 const bool valid = 7119 audio_is_input_channel(srcChannelMask) 7120 && audio_is_input_channel(dstChannelMask) 7121 && audio_is_valid_format(srcFormat) && audio_is_linear_pcm(srcFormat) 7122 && audio_is_valid_format(dstFormat) && audio_is_linear_pcm(dstFormat) 7123 && (srcSampleRate <= dstSampleRate * AUDIO_RESAMPLER_DOWN_RATIO_MAX) 7124 ; // no upsampling checks for now 7125 if (!valid) { 7126 return BAD_VALUE; 7127 } 7128 7129 mSrcFormat = srcFormat; 7130 mSrcChannelMask = srcChannelMask; 7131 mSrcSampleRate = srcSampleRate; 7132 mDstFormat = dstFormat; 7133 mDstChannelMask = dstChannelMask; 7134 mDstSampleRate = dstSampleRate; 7135 7136 // compute derived parameters 7137 mSrcChannelCount = audio_channel_count_from_in_mask(srcChannelMask); 7138 mDstChannelCount = audio_channel_count_from_in_mask(dstChannelMask); 7139 mDstFrameSize = mDstChannelCount * audio_bytes_per_sample(mDstFormat); 7140 7141 // do we need to resample? 7142 delete mResampler; 7143 mResampler = NULL; 7144 if (mSrcSampleRate != mDstSampleRate) { 7145 mResampler = AudioResampler::create(AUDIO_FORMAT_PCM_FLOAT, 7146 mSrcChannelCount, mDstSampleRate); 7147 mResampler->setSampleRate(mSrcSampleRate); 7148 mResampler->setVolume(AudioMixer::UNITY_GAIN_FLOAT, AudioMixer::UNITY_GAIN_FLOAT); 7149 } 7150 7151 // are we running legacy channel conversion modes? 7152 mIsLegacyDownmix = (mSrcChannelMask == AUDIO_CHANNEL_IN_STEREO 7153 || mSrcChannelMask == AUDIO_CHANNEL_IN_FRONT_BACK) 7154 && mDstChannelMask == AUDIO_CHANNEL_IN_MONO; 7155 mIsLegacyUpmix = mSrcChannelMask == AUDIO_CHANNEL_IN_MONO 7156 && (mDstChannelMask == AUDIO_CHANNEL_IN_STEREO 7157 || mDstChannelMask == AUDIO_CHANNEL_IN_FRONT_BACK); 7158 7159 // do we need to process in float? 7160 mRequiresFloat = mResampler != NULL || mIsLegacyDownmix || mIsLegacyUpmix; 7161 7162 // do we need a staging buffer to convert for destination (we can still optimize this)? 7163 // we use mBufFrameSize > 0 to indicate both frame size as well as buffer necessity 7164 if (mResampler != NULL) { 7165 mBufFrameSize = max(mSrcChannelCount, FCC_2) 7166 * audio_bytes_per_sample(AUDIO_FORMAT_PCM_FLOAT); 7167 } else if (mIsLegacyUpmix || mIsLegacyDownmix) { // legacy modes always float 7168 mBufFrameSize = mDstChannelCount * audio_bytes_per_sample(AUDIO_FORMAT_PCM_FLOAT); 7169 } else if (mSrcChannelMask != mDstChannelMask && mDstFormat != mSrcFormat) { 7170 mBufFrameSize = mDstChannelCount * audio_bytes_per_sample(mSrcFormat); 7171 } else { 7172 mBufFrameSize = 0; 7173 } 7174 mBufFrames = 0; // force the buffer to be resized. 7175 7176 // do we need an input converter buffer provider to give us float? 7177 delete mInputConverterProvider; 7178 mInputConverterProvider = NULL; 7179 if (mRequiresFloat && mSrcFormat != AUDIO_FORMAT_PCM_FLOAT) { 7180 mInputConverterProvider = new ReformatBufferProvider( 7181 audio_channel_count_from_in_mask(mSrcChannelMask), 7182 mSrcFormat, 7183 AUDIO_FORMAT_PCM_FLOAT, 7184 256 /* provider buffer frame count */); 7185 } 7186 7187 // do we need a remixer to do channel mask conversion 7188 if (!mIsLegacyDownmix && !mIsLegacyUpmix && mSrcChannelMask != mDstChannelMask) { 7189 (void) memcpy_by_index_array_initialization_from_channel_mask( 7190 mIdxAry, ARRAY_SIZE(mIdxAry), mDstChannelMask, mSrcChannelMask); 7191 } 7192 return NO_ERROR; 7193 } 7194 7195 void AudioFlinger::RecordThread::RecordBufferConverter::convertNoResampler( 7196 void *dst, const void *src, size_t frames) 7197 { 7198 // src is native type unless there is legacy upmix or downmix, whereupon it is float. 7199 if (mBufFrameSize != 0 && mBufFrames < frames) { 7200 free(mBuf); 7201 mBufFrames = frames; 7202 (void)posix_memalign(&mBuf, 32, mBufFrames * mBufFrameSize); 7203 } 7204 // do we need to do legacy upmix and downmix? 7205 if (mIsLegacyUpmix || mIsLegacyDownmix) { 7206 void *dstBuf = mBuf != NULL ? mBuf : dst; 7207 if (mIsLegacyUpmix) { 7208 upmix_to_stereo_float_from_mono_float((float *)dstBuf, 7209 (const float *)src, frames); 7210 } else /*mIsLegacyDownmix */ { 7211 downmix_to_mono_float_from_stereo_float((float *)dstBuf, 7212 (const float *)src, frames); 7213 } 7214 if (mBuf != NULL) { 7215 memcpy_by_audio_format(dst, mDstFormat, mBuf, AUDIO_FORMAT_PCM_FLOAT, 7216 frames * mDstChannelCount); 7217 } 7218 return; 7219 } 7220 // do we need to do channel mask conversion? 7221 if (mSrcChannelMask != mDstChannelMask) { 7222 void *dstBuf = mBuf != NULL ? mBuf : dst; 7223 memcpy_by_index_array(dstBuf, mDstChannelCount, 7224 src, mSrcChannelCount, mIdxAry, audio_bytes_per_sample(mSrcFormat), frames); 7225 if (dstBuf == dst) { 7226 return; // format is the same 7227 } 7228 } 7229 // convert to destination buffer 7230 const void *convertBuf = mBuf != NULL ? mBuf : src; 7231 memcpy_by_audio_format(dst, mDstFormat, convertBuf, mSrcFormat, 7232 frames * mDstChannelCount); 7233 } 7234 7235 void AudioFlinger::RecordThread::RecordBufferConverter::convertResampler( 7236 void *dst, /*not-a-const*/ void *src, size_t frames) 7237 { 7238 // src buffer format is ALWAYS float when entering this routine 7239 if (mIsLegacyUpmix) { 7240 ; // mono to stereo already handled by resampler 7241 } else if (mIsLegacyDownmix 7242 || (mSrcChannelMask == mDstChannelMask && mSrcChannelCount == 1)) { 7243 // the resampler outputs stereo for mono input channel (a feature?) 7244 // must convert to mono 7245 downmix_to_mono_float_from_stereo_float((float *)src, 7246 (const float *)src, frames); 7247 } else if (mSrcChannelMask != mDstChannelMask) { 7248 // convert to mono channel again for channel mask conversion (could be skipped 7249 // with further optimization). 7250 if (mSrcChannelCount == 1) { 7251 downmix_to_mono_float_from_stereo_float((float *)src, 7252 (const float *)src, frames); 7253 } 7254 // convert to destination format (in place, OK as float is larger than other types) 7255 if (mDstFormat != AUDIO_FORMAT_PCM_FLOAT) { 7256 memcpy_by_audio_format(src, mDstFormat, src, AUDIO_FORMAT_PCM_FLOAT, 7257 frames * mSrcChannelCount); 7258 } 7259 // channel convert and save to dst 7260 memcpy_by_index_array(dst, mDstChannelCount, 7261 src, mSrcChannelCount, mIdxAry, audio_bytes_per_sample(mDstFormat), frames); 7262 return; 7263 } 7264 // convert to destination format and save to dst 7265 memcpy_by_audio_format(dst, mDstFormat, src, AUDIO_FORMAT_PCM_FLOAT, 7266 frames * mDstChannelCount); 7267 } 7268 7269 bool AudioFlinger::RecordThread::checkForNewParameter_l(const String8& keyValuePair, 7270 status_t& status) 7271 { 7272 bool reconfig = false; 7273 7274 status = NO_ERROR; 7275 7276 audio_format_t reqFormat = mFormat; 7277 uint32_t samplingRate = mSampleRate; 7278 // TODO this may change if we want to support capture from HDMI PCM multi channel (e.g on TVs). 7279 audio_channel_mask_t channelMask = audio_channel_in_mask_from_count(mChannelCount); 7280 7281 AudioParameter param = AudioParameter(keyValuePair); 7282 int value; 7283 7284 // scope for AutoPark extends to end of method 7285 AutoPark<FastCapture> park(mFastCapture); 7286 7287 // TODO Investigate when this code runs. Check with audio policy when a sample rate and 7288 // channel count change can be requested. Do we mandate the first client defines the 7289 // HAL sampling rate and channel count or do we allow changes on the fly? 7290 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 7291 samplingRate = value; 7292 reconfig = true; 7293 } 7294 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 7295 if (!audio_is_linear_pcm((audio_format_t) value)) { 7296 status = BAD_VALUE; 7297 } else { 7298 reqFormat = (audio_format_t) value; 7299 reconfig = true; 7300 } 7301 } 7302 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 7303 audio_channel_mask_t mask = (audio_channel_mask_t) value; 7304 if (!audio_is_input_channel(mask) || 7305 audio_channel_count_from_in_mask(mask) > FCC_8) { 7306 status = BAD_VALUE; 7307 } else { 7308 channelMask = mask; 7309 reconfig = true; 7310 } 7311 } 7312 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 7313 // do not accept frame count changes if tracks are open as the track buffer 7314 // size depends on frame count and correct behavior would not be guaranteed 7315 // if frame count is changed after track creation 7316 if (mActiveTracks.size() > 0) { 7317 status = INVALID_OPERATION; 7318 } else { 7319 reconfig = true; 7320 } 7321 } 7322 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 7323 // forward device change to effects that have requested to be 7324 // aware of attached audio device. 7325 for (size_t i = 0; i < mEffectChains.size(); i++) { 7326 mEffectChains[i]->setDevice_l(value); 7327 } 7328 7329 // store input device and output device but do not forward output device to audio HAL. 7330 // Note that status is ignored by the caller for output device 7331 // (see AudioFlinger::setParameters() 7332 if (audio_is_output_devices(value)) { 7333 mOutDevice = value; 7334 status = BAD_VALUE; 7335 } else { 7336 mInDevice = value; 7337 if (value != AUDIO_DEVICE_NONE) { 7338 mPrevInDevice = value; 7339 } 7340 // disable AEC and NS if the device is a BT SCO headset supporting those 7341 // pre processings 7342 if (mTracks.size() > 0) { 7343 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 7344 mAudioFlinger->btNrecIsOff(); 7345 for (size_t i = 0; i < mTracks.size(); i++) { 7346 sp<RecordTrack> track = mTracks[i]; 7347 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId()); 7348 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId()); 7349 } 7350 } 7351 } 7352 } 7353 if (param.getInt(String8(AudioParameter::keyInputSource), value) == NO_ERROR && 7354 mAudioSource != (audio_source_t)value) { 7355 // forward device change to effects that have requested to be 7356 // aware of attached audio device. 7357 for (size_t i = 0; i < mEffectChains.size(); i++) { 7358 mEffectChains[i]->setAudioSource_l((audio_source_t)value); 7359 } 7360 mAudioSource = (audio_source_t)value; 7361 } 7362 7363 if (status == NO_ERROR) { 7364 status = mInput->stream->common.set_parameters(&mInput->stream->common, 7365 keyValuePair.string()); 7366 if (status == INVALID_OPERATION) { 7367 inputStandBy(); 7368 status = mInput->stream->common.set_parameters(&mInput->stream->common, 7369 keyValuePair.string()); 7370 } 7371 if (reconfig) { 7372 if (status == BAD_VALUE && 7373 audio_is_linear_pcm(mInput->stream->common.get_format(&mInput->stream->common)) && 7374 audio_is_linear_pcm(reqFormat) && 7375 (mInput->stream->common.get_sample_rate(&mInput->stream->common) 7376 <= (AUDIO_RESAMPLER_DOWN_RATIO_MAX * samplingRate)) && 7377 audio_channel_count_from_in_mask( 7378 mInput->stream->common.get_channels(&mInput->stream->common)) <= FCC_8) { 7379 status = NO_ERROR; 7380 } 7381 if (status == NO_ERROR) { 7382 readInputParameters_l(); 7383 sendIoConfigEvent_l(AUDIO_INPUT_CONFIG_CHANGED); 7384 } 7385 } 7386 } 7387 7388 return reconfig; 7389 } 7390 7391 String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 7392 { 7393 Mutex::Autolock _l(mLock); 7394 if (initCheck() != NO_ERROR) { 7395 return String8(); 7396 } 7397 7398 char *s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 7399 const String8 out_s8(s); 7400 free(s); 7401 return out_s8; 7402 } 7403 7404 void AudioFlinger::RecordThread::ioConfigChanged(audio_io_config_event event, pid_t pid) { 7405 sp<AudioIoDescriptor> desc = new AudioIoDescriptor(); 7406 7407 desc->mIoHandle = mId; 7408 7409 switch (event) { 7410 case AUDIO_INPUT_OPENED: 7411 case AUDIO_INPUT_CONFIG_CHANGED: 7412 desc->mPatch = mPatch; 7413 desc->mChannelMask = mChannelMask; 7414 desc->mSamplingRate = mSampleRate; 7415 desc->mFormat = mFormat; 7416 desc->mFrameCount = mFrameCount; 7417 desc->mFrameCountHAL = mFrameCount; 7418 desc->mLatency = 0; 7419 break; 7420 7421 case AUDIO_INPUT_CLOSED: 7422 default: 7423 break; 7424 } 7425 mAudioFlinger->ioConfigChanged(event, desc, pid); 7426 } 7427 7428 void AudioFlinger::RecordThread::readInputParameters_l() 7429 { 7430 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 7431 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 7432 mChannelCount = audio_channel_count_from_in_mask(mChannelMask); 7433 if (mChannelCount > FCC_8) { 7434 ALOGE("HAL channel count %d > %d", mChannelCount, FCC_8); 7435 } 7436 mHALFormat = mInput->stream->common.get_format(&mInput->stream->common); 7437 mFormat = mHALFormat; 7438 if (!audio_is_linear_pcm(mFormat)) { 7439 ALOGE("HAL format %#x is not linear pcm", mFormat); 7440 } 7441 mFrameSize = audio_stream_in_frame_size(mInput->stream); 7442 mBufferSize = mInput->stream->common.get_buffer_size(&mInput->stream->common); 7443 mFrameCount = mBufferSize / mFrameSize; 7444 // This is the formula for calculating the temporary buffer size. 7445 // With 7 HAL buffers, we can guarantee ability to down-sample the input by ratio of 6:1 to 7446 // 1 full output buffer, regardless of the alignment of the available input. 7447 // The value is somewhat arbitrary, and could probably be even larger. 7448 // A larger value should allow more old data to be read after a track calls start(), 7449 // without increasing latency. 7450 // 7451 // Note this is independent of the maximum downsampling ratio permitted for capture. 7452 mRsmpInFrames = mFrameCount * 7; 7453 mRsmpInFramesP2 = roundup(mRsmpInFrames); 7454 free(mRsmpInBuffer); 7455 mRsmpInBuffer = NULL; 7456 7457 // TODO optimize audio capture buffer sizes ... 7458 // Here we calculate the size of the sliding buffer used as a source 7459 // for resampling. mRsmpInFramesP2 is currently roundup(mFrameCount * 7). 7460 // For current HAL frame counts, this is usually 2048 = 40 ms. It would 7461 // be better to have it derived from the pipe depth in the long term. 7462 // The current value is higher than necessary. However it should not add to latency. 7463 7464 // Over-allocate beyond mRsmpInFramesP2 to permit a HAL read past end of buffer 7465 size_t bufferSize = (mRsmpInFramesP2 + mFrameCount - 1) * mFrameSize; 7466 (void)posix_memalign(&mRsmpInBuffer, 32, bufferSize); 7467 memset(mRsmpInBuffer, 0, bufferSize); // if posix_memalign fails, will segv here. 7468 7469 // AudioRecord mSampleRate and mChannelCount are constant due to AudioRecord API constraints. 7470 // But if thread's mSampleRate or mChannelCount changes, how will that affect active tracks? 7471 } 7472 7473 uint32_t AudioFlinger::RecordThread::getInputFramesLost() 7474 { 7475 Mutex::Autolock _l(mLock); 7476 if (initCheck() != NO_ERROR) { 7477 return 0; 7478 } 7479 7480 return mInput->stream->get_input_frames_lost(mInput->stream); 7481 } 7482 7483 // hasAudioSession_l() must be called with ThreadBase::mLock held 7484 uint32_t AudioFlinger::RecordThread::hasAudioSession_l(audio_session_t sessionId) const 7485 { 7486 uint32_t result = 0; 7487 if (getEffectChain_l(sessionId) != 0) { 7488 result = EFFECT_SESSION; 7489 } 7490 7491 for (size_t i = 0; i < mTracks.size(); ++i) { 7492 if (sessionId == mTracks[i]->sessionId()) { 7493 result |= TRACK_SESSION; 7494 if (mTracks[i]->isFastTrack()) { 7495 result |= FAST_SESSION; 7496 } 7497 break; 7498 } 7499 } 7500 7501 return result; 7502 } 7503 7504 KeyedVector<audio_session_t, bool> AudioFlinger::RecordThread::sessionIds() const 7505 { 7506 KeyedVector<audio_session_t, bool> ids; 7507 Mutex::Autolock _l(mLock); 7508 for (size_t j = 0; j < mTracks.size(); ++j) { 7509 sp<RecordThread::RecordTrack> track = mTracks[j]; 7510 audio_session_t sessionId = track->sessionId(); 7511 if (ids.indexOfKey(sessionId) < 0) { 7512 ids.add(sessionId, true); 7513 } 7514 } 7515 return ids; 7516 } 7517 7518 AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 7519 { 7520 Mutex::Autolock _l(mLock); 7521 AudioStreamIn *input = mInput; 7522 mInput = NULL; 7523 return input; 7524 } 7525 7526 // this method must always be called either with ThreadBase mLock held or inside the thread loop 7527 audio_stream_t* AudioFlinger::RecordThread::stream() const 7528 { 7529 if (mInput == NULL) { 7530 return NULL; 7531 } 7532 return &mInput->stream->common; 7533 } 7534 7535 status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 7536 { 7537 // only one chain per input thread 7538 if (mEffectChains.size() != 0) { 7539 ALOGW("addEffectChain_l() already one chain %p on thread %p", chain.get(), this); 7540 return INVALID_OPERATION; 7541 } 7542 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 7543 chain->setThread(this); 7544 chain->setInBuffer(NULL); 7545 chain->setOutBuffer(NULL); 7546 7547 checkSuspendOnAddEffectChain_l(chain); 7548 7549 // make sure enabled pre processing effects state is communicated to the HAL as we 7550 // just moved them to a new input stream. 7551 chain->syncHalEffectsState(); 7552 7553 mEffectChains.add(chain); 7554 7555 return NO_ERROR; 7556 } 7557 7558 size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 7559 { 7560 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 7561 ALOGW_IF(mEffectChains.size() != 1, 7562 "removeEffectChain_l() %p invalid chain size %zu on thread %p", 7563 chain.get(), mEffectChains.size(), this); 7564 if (mEffectChains.size() == 1) { 7565 mEffectChains.removeAt(0); 7566 } 7567 return 0; 7568 } 7569 7570 status_t AudioFlinger::RecordThread::createAudioPatch_l(const struct audio_patch *patch, 7571 audio_patch_handle_t *handle) 7572 { 7573 status_t status = NO_ERROR; 7574 7575 // store new device and send to effects 7576 mInDevice = patch->sources[0].ext.device.type; 7577 mPatch = *patch; 7578 for (size_t i = 0; i < mEffectChains.size(); i++) { 7579 mEffectChains[i]->setDevice_l(mInDevice); 7580 } 7581 7582 // disable AEC and NS if the device is a BT SCO headset supporting those 7583 // pre processings 7584 if (mTracks.size() > 0) { 7585 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 7586 mAudioFlinger->btNrecIsOff(); 7587 for (size_t i = 0; i < mTracks.size(); i++) { 7588 sp<RecordTrack> track = mTracks[i]; 7589 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId()); 7590 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId()); 7591 } 7592 } 7593 7594 // store new source and send to effects 7595 if (mAudioSource != patch->sinks[0].ext.mix.usecase.source) { 7596 mAudioSource = patch->sinks[0].ext.mix.usecase.source; 7597 for (size_t i = 0; i < mEffectChains.size(); i++) { 7598 mEffectChains[i]->setAudioSource_l(mAudioSource); 7599 } 7600 } 7601 7602 if (mInput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 7603 audio_hw_device_t *hwDevice = mInput->audioHwDev->hwDevice(); 7604 status = hwDevice->create_audio_patch(hwDevice, 7605 patch->num_sources, 7606 patch->sources, 7607 patch->num_sinks, 7608 patch->sinks, 7609 handle); 7610 } else { 7611 char *address; 7612 if (strcmp(patch->sources[0].ext.device.address, "") != 0) { 7613 address = audio_device_address_to_parameter( 7614 patch->sources[0].ext.device.type, 7615 patch->sources[0].ext.device.address); 7616 } else { 7617 address = (char *)calloc(1, 1); 7618 } 7619 AudioParameter param = AudioParameter(String8(address)); 7620 free(address); 7621 param.addInt(String8(AUDIO_PARAMETER_STREAM_ROUTING), 7622 (int)patch->sources[0].ext.device.type); 7623 param.addInt(String8(AUDIO_PARAMETER_STREAM_INPUT_SOURCE), 7624 (int)patch->sinks[0].ext.mix.usecase.source); 7625 status = mInput->stream->common.set_parameters(&mInput->stream->common, 7626 param.toString().string()); 7627 *handle = AUDIO_PATCH_HANDLE_NONE; 7628 } 7629 7630 if (mInDevice != mPrevInDevice) { 7631 sendIoConfigEvent_l(AUDIO_INPUT_CONFIG_CHANGED); 7632 mPrevInDevice = mInDevice; 7633 } 7634 7635 return status; 7636 } 7637 7638 status_t AudioFlinger::RecordThread::releaseAudioPatch_l(const audio_patch_handle_t handle) 7639 { 7640 status_t status = NO_ERROR; 7641 7642 mInDevice = AUDIO_DEVICE_NONE; 7643 7644 if (mInput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 7645 audio_hw_device_t *hwDevice = mInput->audioHwDev->hwDevice(); 7646 status = hwDevice->release_audio_patch(hwDevice, handle); 7647 } else { 7648 AudioParameter param; 7649 param.addInt(String8(AUDIO_PARAMETER_STREAM_ROUTING), 0); 7650 status = mInput->stream->common.set_parameters(&mInput->stream->common, 7651 param.toString().string()); 7652 } 7653 return status; 7654 } 7655 7656 void AudioFlinger::RecordThread::addPatchRecord(const sp<PatchRecord>& record) 7657 { 7658 Mutex::Autolock _l(mLock); 7659 mTracks.add(record); 7660 } 7661 7662 void AudioFlinger::RecordThread::deletePatchRecord(const sp<PatchRecord>& record) 7663 { 7664 Mutex::Autolock _l(mLock); 7665 destroyTrack_l(record); 7666 } 7667 7668 void AudioFlinger::RecordThread::getAudioPortConfig(struct audio_port_config *config) 7669 { 7670 ThreadBase::getAudioPortConfig(config); 7671 config->role = AUDIO_PORT_ROLE_SINK; 7672 config->ext.mix.hw_module = mInput->audioHwDev->handle(); 7673 config->ext.mix.usecase.source = mAudioSource; 7674 } 7675 7676 } // namespace android 7677