Threads.cpp revision 3f0c902beb53a245c9db35e871607dba05b8d391
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// ---------------------------------------------------------------------------- 75 76// Note: the following macro is used for extremely verbose logging message. In 77// order to run with ALOG_ASSERT turned on, we need to have LOG_NDEBUG set to 78// 0; but one side effect of this is to turn all LOGV's as well. Some messages 79// are so verbose that we want to suppress them even when we have ALOG_ASSERT 80// turned on. Do not uncomment the #def below unless you really know what you 81// are doing and want to see all of the extremely verbose messages. 82//#define VERY_VERY_VERBOSE_LOGGING 83#ifdef VERY_VERY_VERBOSE_LOGGING 84#define ALOGVV ALOGV 85#else 86#define ALOGVV(a...) do { } while(0) 87#endif 88 89// TODO: Move these macro/inlines to a header file. 90#define max(a, b) ((a) > (b) ? (a) : (b)) 91template <typename T> 92static inline T min(const T& a, const T& b) 93{ 94 return a < b ? a : b; 95} 96 97#ifndef ARRAY_SIZE 98#define ARRAY_SIZE(a) (sizeof(a) / sizeof(a[0])) 99#endif 100 101namespace android { 102 103// retry counts for buffer fill timeout 104// 50 * ~20msecs = 1 second 105static const int8_t kMaxTrackRetries = 50; 106static const int8_t kMaxTrackStartupRetries = 50; 107// allow less retry attempts on direct output thread. 108// direct outputs can be a scarce resource in audio hardware and should 109// be released as quickly as possible. 110static const int8_t kMaxTrackRetriesDirect = 2; 111 112// don't warn about blocked writes or record buffer overflows more often than this 113static const nsecs_t kWarningThrottleNs = seconds(5); 114 115// RecordThread loop sleep time upon application overrun or audio HAL read error 116static const int kRecordThreadSleepUs = 5000; 117 118// maximum time to wait in sendConfigEvent_l() for a status to be received 119static const nsecs_t kConfigEventTimeoutNs = seconds(2); 120 121// minimum sleep time for the mixer thread loop when tracks are active but in underrun 122static const uint32_t kMinThreadSleepTimeUs = 5000; 123// maximum divider applied to the active sleep time in the mixer thread loop 124static const uint32_t kMaxThreadSleepTimeShift = 2; 125 126// minimum normal sink buffer size, expressed in milliseconds rather than frames 127// FIXME This should be based on experimentally observed scheduling jitter 128static const uint32_t kMinNormalSinkBufferSizeMs = 20; 129// maximum normal sink buffer size 130static const uint32_t kMaxNormalSinkBufferSizeMs = 24; 131 132// minimum capture buffer size in milliseconds to _not_ need a fast capture thread 133// FIXME This should be based on experimentally observed scheduling jitter 134static const uint32_t kMinNormalCaptureBufferSizeMs = 12; 135 136// Offloaded output thread standby delay: allows track transition without going to standby 137static const nsecs_t kOffloadStandbyDelayNs = seconds(1); 138 139// Whether to use fast mixer 140static const enum { 141 FastMixer_Never, // never initialize or use: for debugging only 142 FastMixer_Always, // always initialize and use, even if not needed: for debugging only 143 // normal mixer multiplier is 1 144 FastMixer_Static, // initialize if needed, then use all the time if initialized, 145 // multiplier is calculated based on min & max normal mixer buffer size 146 FastMixer_Dynamic, // initialize if needed, then use dynamically depending on track load, 147 // multiplier is calculated based on min & max normal mixer buffer size 148 // FIXME for FastMixer_Dynamic: 149 // Supporting this option will require fixing HALs that can't handle large writes. 150 // For example, one HAL implementation returns an error from a large write, 151 // and another HAL implementation corrupts memory, possibly in the sample rate converter. 152 // We could either fix the HAL implementations, or provide a wrapper that breaks 153 // up large writes into smaller ones, and the wrapper would need to deal with scheduler. 154} kUseFastMixer = FastMixer_Static; 155 156// Whether to use fast capture 157static const enum { 158 FastCapture_Never, // never initialize or use: for debugging only 159 FastCapture_Always, // always initialize and use, even if not needed: for debugging only 160 FastCapture_Static, // initialize if needed, then use all the time if initialized 161} kUseFastCapture = FastCapture_Static; 162 163// Priorities for requestPriority 164static const int kPriorityAudioApp = 2; 165static const int kPriorityFastMixer = 3; 166static const int kPriorityFastCapture = 3; 167 168// IAudioFlinger::createTrack() reports back to client the total size of shared memory area 169// for the track. The client then sub-divides this into smaller buffers for its use. 170// Currently the client uses N-buffering by default, but doesn't tell us about the value of N. 171// So for now we just assume that client is double-buffered for fast tracks. 172// FIXME It would be better for client to tell AudioFlinger the value of N, 173// so AudioFlinger could allocate the right amount of memory. 174// See the client's minBufCount and mNotificationFramesAct calculations for details. 175 176// This is the default value, if not specified by property. 177static const int kFastTrackMultiplier = 2; 178 179// The minimum and maximum allowed values 180static const int kFastTrackMultiplierMin = 1; 181static const int kFastTrackMultiplierMax = 2; 182 183// The actual value to use, which can be specified per-device via property af.fast_track_multiplier. 184static int sFastTrackMultiplier = kFastTrackMultiplier; 185 186// See Thread::readOnlyHeap(). 187// Initially this heap is used to allocate client buffers for "fast" AudioRecord. 188// Eventually it will be the single buffer that FastCapture writes into via HAL read(), 189// and that all "fast" AudioRecord clients read from. In either case, the size can be small. 190static const size_t kRecordThreadReadOnlyHeapSize = 0x2000; 191 192// ---------------------------------------------------------------------------- 193 194static pthread_once_t sFastTrackMultiplierOnce = PTHREAD_ONCE_INIT; 195 196static void sFastTrackMultiplierInit() 197{ 198 char value[PROPERTY_VALUE_MAX]; 199 if (property_get("af.fast_track_multiplier", value, NULL) > 0) { 200 char *endptr; 201 unsigned long ul = strtoul(value, &endptr, 0); 202 if (*endptr == '\0' && kFastTrackMultiplierMin <= ul && ul <= kFastTrackMultiplierMax) { 203 sFastTrackMultiplier = (int) ul; 204 } 205 } 206} 207 208// ---------------------------------------------------------------------------- 209 210#ifdef ADD_BATTERY_DATA 211// To collect the amplifier usage 212static void addBatteryData(uint32_t params) { 213 sp<IMediaPlayerService> service = IMediaDeathNotifier::getMediaPlayerService(); 214 if (service == NULL) { 215 // it already logged 216 return; 217 } 218 219 service->addBatteryData(params); 220} 221#endif 222 223// Track the CLOCK_BOOTTIME versus CLOCK_MONOTONIC timebase offset 224struct { 225 // call when you acquire a partial wakelock 226 void acquire(const sp<IBinder> &wakeLockToken) { 227 pthread_mutex_lock(&mLock); 228 if (wakeLockToken.get() == nullptr) { 229 adjustTimebaseOffset(&mBoottimeOffset, ExtendedTimestamp::TIMEBASE_BOOTTIME); 230 } else { 231 if (mCount == 0) { 232 adjustTimebaseOffset(&mBoottimeOffset, ExtendedTimestamp::TIMEBASE_BOOTTIME); 233 } 234 ++mCount; 235 } 236 pthread_mutex_unlock(&mLock); 237 } 238 239 // call when you release a partial wakelock. 240 void release(const sp<IBinder> &wakeLockToken) { 241 if (wakeLockToken.get() == nullptr) { 242 return; 243 } 244 pthread_mutex_lock(&mLock); 245 if (--mCount < 0) { 246 ALOGE("negative wakelock count"); 247 mCount = 0; 248 } 249 pthread_mutex_unlock(&mLock); 250 } 251 252 // retrieves the boottime timebase offset from monotonic. 253 int64_t getBoottimeOffset() { 254 pthread_mutex_lock(&mLock); 255 int64_t boottimeOffset = mBoottimeOffset; 256 pthread_mutex_unlock(&mLock); 257 return boottimeOffset; 258 } 259 260 // Adjusts the timebase offset between TIMEBASE_MONOTONIC 261 // and the selected timebase. 262 // Currently only TIMEBASE_BOOTTIME is allowed. 263 // 264 // This only needs to be called upon acquiring the first partial wakelock 265 // after all other partial wakelocks are released. 266 // 267 // We do an empirical measurement of the offset rather than parsing 268 // /proc/timer_list since the latter is not a formal kernel ABI. 269 static void adjustTimebaseOffset(int64_t *offset, ExtendedTimestamp::Timebase timebase) { 270 int clockbase; 271 switch (timebase) { 272 case ExtendedTimestamp::TIMEBASE_BOOTTIME: 273 clockbase = SYSTEM_TIME_BOOTTIME; 274 break; 275 default: 276 LOG_ALWAYS_FATAL("invalid timebase %d", timebase); 277 break; 278 } 279 // try three times to get the clock offset, choose the one 280 // with the minimum gap in measurements. 281 const int tries = 3; 282 nsecs_t bestGap, measured; 283 for (int i = 0; i < tries; ++i) { 284 const nsecs_t tmono = systemTime(SYSTEM_TIME_MONOTONIC); 285 const nsecs_t tbase = systemTime(clockbase); 286 const nsecs_t tmono2 = systemTime(SYSTEM_TIME_MONOTONIC); 287 const nsecs_t gap = tmono2 - tmono; 288 if (i == 0 || gap < bestGap) { 289 bestGap = gap; 290 measured = tbase - ((tmono + tmono2) >> 1); 291 } 292 } 293 294 // to avoid micro-adjusting, we don't change the timebase 295 // unless it is significantly different. 296 // 297 // Assumption: It probably takes more than toleranceNs to 298 // suspend and resume the device. 299 static int64_t toleranceNs = 10000; // 10 us 300 if (llabs(*offset - measured) > toleranceNs) { 301 ALOGV("Adjusting timebase offset old: %lld new: %lld", 302 (long long)*offset, (long long)measured); 303 *offset = measured; 304 } 305 } 306 307 pthread_mutex_t mLock; 308 int32_t mCount; 309 int64_t mBoottimeOffset; 310} gBoottime = { PTHREAD_MUTEX_INITIALIZER, 0, 0 }; // static, so use POD initialization 311 312// ---------------------------------------------------------------------------- 313// CPU Stats 314// ---------------------------------------------------------------------------- 315 316class CpuStats { 317public: 318 CpuStats(); 319 void sample(const String8 &title); 320#ifdef DEBUG_CPU_USAGE 321private: 322 ThreadCpuUsage mCpuUsage; // instantaneous thread CPU usage in wall clock ns 323 CentralTendencyStatistics mWcStats; // statistics on thread CPU usage in wall clock ns 324 325 CentralTendencyStatistics mHzStats; // statistics on thread CPU usage in cycles 326 327 int mCpuNum; // thread's current CPU number 328 int mCpukHz; // frequency of thread's current CPU in kHz 329#endif 330}; 331 332CpuStats::CpuStats() 333#ifdef DEBUG_CPU_USAGE 334 : mCpuNum(-1), mCpukHz(-1) 335#endif 336{ 337} 338 339void CpuStats::sample(const String8 &title 340#ifndef DEBUG_CPU_USAGE 341 __unused 342#endif 343 ) { 344#ifdef DEBUG_CPU_USAGE 345 // get current thread's delta CPU time in wall clock ns 346 double wcNs; 347 bool valid = mCpuUsage.sampleAndEnable(wcNs); 348 349 // record sample for wall clock statistics 350 if (valid) { 351 mWcStats.sample(wcNs); 352 } 353 354 // get the current CPU number 355 int cpuNum = sched_getcpu(); 356 357 // get the current CPU frequency in kHz 358 int cpukHz = mCpuUsage.getCpukHz(cpuNum); 359 360 // check if either CPU number or frequency changed 361 if (cpuNum != mCpuNum || cpukHz != mCpukHz) { 362 mCpuNum = cpuNum; 363 mCpukHz = cpukHz; 364 // ignore sample for purposes of cycles 365 valid = false; 366 } 367 368 // if no change in CPU number or frequency, then record sample for cycle statistics 369 if (valid && mCpukHz > 0) { 370 double cycles = wcNs * cpukHz * 0.000001; 371 mHzStats.sample(cycles); 372 } 373 374 unsigned n = mWcStats.n(); 375 // mCpuUsage.elapsed() is expensive, so don't call it every loop 376 if ((n & 127) == 1) { 377 long long elapsed = mCpuUsage.elapsed(); 378 if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) { 379 double perLoop = elapsed / (double) n; 380 double perLoop100 = perLoop * 0.01; 381 double perLoop1k = perLoop * 0.001; 382 double mean = mWcStats.mean(); 383 double stddev = mWcStats.stddev(); 384 double minimum = mWcStats.minimum(); 385 double maximum = mWcStats.maximum(); 386 double meanCycles = mHzStats.mean(); 387 double stddevCycles = mHzStats.stddev(); 388 double minCycles = mHzStats.minimum(); 389 double maxCycles = mHzStats.maximum(); 390 mCpuUsage.resetElapsed(); 391 mWcStats.reset(); 392 mHzStats.reset(); 393 ALOGD("CPU usage for %s over past %.1f secs\n" 394 " (%u mixer loops at %.1f mean ms per loop):\n" 395 " us per mix loop: mean=%.0f stddev=%.0f min=%.0f max=%.0f\n" 396 " %% of wall: mean=%.1f stddev=%.1f min=%.1f max=%.1f\n" 397 " MHz: mean=%.1f, stddev=%.1f, min=%.1f max=%.1f", 398 title.string(), 399 elapsed * .000000001, n, perLoop * .000001, 400 mean * .001, 401 stddev * .001, 402 minimum * .001, 403 maximum * .001, 404 mean / perLoop100, 405 stddev / perLoop100, 406 minimum / perLoop100, 407 maximum / perLoop100, 408 meanCycles / perLoop1k, 409 stddevCycles / perLoop1k, 410 minCycles / perLoop1k, 411 maxCycles / perLoop1k); 412 413 } 414 } 415#endif 416}; 417 418// ---------------------------------------------------------------------------- 419// ThreadBase 420// ---------------------------------------------------------------------------- 421 422// static 423const char *AudioFlinger::ThreadBase::threadTypeToString(AudioFlinger::ThreadBase::type_t type) 424{ 425 switch (type) { 426 case MIXER: 427 return "MIXER"; 428 case DIRECT: 429 return "DIRECT"; 430 case DUPLICATING: 431 return "DUPLICATING"; 432 case RECORD: 433 return "RECORD"; 434 case OFFLOAD: 435 return "OFFLOAD"; 436 default: 437 return "unknown"; 438 } 439} 440 441String8 devicesToString(audio_devices_t devices) 442{ 443 static const struct mapping { 444 audio_devices_t mDevices; 445 const char * mString; 446 } mappingsOut[] = { 447 {AUDIO_DEVICE_OUT_EARPIECE, "EARPIECE"}, 448 {AUDIO_DEVICE_OUT_SPEAKER, "SPEAKER"}, 449 {AUDIO_DEVICE_OUT_WIRED_HEADSET, "WIRED_HEADSET"}, 450 {AUDIO_DEVICE_OUT_WIRED_HEADPHONE, "WIRED_HEADPHONE"}, 451 {AUDIO_DEVICE_OUT_BLUETOOTH_SCO, "BLUETOOTH_SCO"}, 452 {AUDIO_DEVICE_OUT_BLUETOOTH_SCO_HEADSET, "BLUETOOTH_SCO_HEADSET"}, 453 {AUDIO_DEVICE_OUT_BLUETOOTH_SCO_CARKIT, "BLUETOOTH_SCO_CARKIT"}, 454 {AUDIO_DEVICE_OUT_BLUETOOTH_A2DP, "BLUETOOTH_A2DP"}, 455 {AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES,"BLUETOOTH_A2DP_HEADPHONES"}, 456 {AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER, "BLUETOOTH_A2DP_SPEAKER"}, 457 {AUDIO_DEVICE_OUT_AUX_DIGITAL, "AUX_DIGITAL"}, 458 {AUDIO_DEVICE_OUT_HDMI, "HDMI"}, 459 {AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET,"ANLG_DOCK_HEADSET"}, 460 {AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET,"DGTL_DOCK_HEADSET"}, 461 {AUDIO_DEVICE_OUT_USB_ACCESSORY, "USB_ACCESSORY"}, 462 {AUDIO_DEVICE_OUT_USB_DEVICE, "USB_DEVICE"}, 463 {AUDIO_DEVICE_OUT_TELEPHONY_TX, "TELEPHONY_TX"}, 464 {AUDIO_DEVICE_OUT_LINE, "LINE"}, 465 {AUDIO_DEVICE_OUT_HDMI_ARC, "HDMI_ARC"}, 466 {AUDIO_DEVICE_OUT_SPDIF, "SPDIF"}, 467 {AUDIO_DEVICE_OUT_FM, "FM"}, 468 {AUDIO_DEVICE_OUT_AUX_LINE, "AUX_LINE"}, 469 {AUDIO_DEVICE_OUT_SPEAKER_SAFE, "SPEAKER_SAFE"}, 470 {AUDIO_DEVICE_OUT_IP, "IP"}, 471 {AUDIO_DEVICE_NONE, "NONE"}, // must be last 472 }, mappingsIn[] = { 473 {AUDIO_DEVICE_IN_COMMUNICATION, "COMMUNICATION"}, 474 {AUDIO_DEVICE_IN_AMBIENT, "AMBIENT"}, 475 {AUDIO_DEVICE_IN_BUILTIN_MIC, "BUILTIN_MIC"}, 476 {AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET, "BLUETOOTH_SCO_HEADSET"}, 477 {AUDIO_DEVICE_IN_WIRED_HEADSET, "WIRED_HEADSET"}, 478 {AUDIO_DEVICE_IN_AUX_DIGITAL, "AUX_DIGITAL"}, 479 {AUDIO_DEVICE_IN_VOICE_CALL, "VOICE_CALL"}, 480 {AUDIO_DEVICE_IN_TELEPHONY_RX, "TELEPHONY_RX"}, 481 {AUDIO_DEVICE_IN_BACK_MIC, "BACK_MIC"}, 482 {AUDIO_DEVICE_IN_REMOTE_SUBMIX, "REMOTE_SUBMIX"}, 483 {AUDIO_DEVICE_IN_ANLG_DOCK_HEADSET, "ANLG_DOCK_HEADSET"}, 484 {AUDIO_DEVICE_IN_DGTL_DOCK_HEADSET, "DGTL_DOCK_HEADSET"}, 485 {AUDIO_DEVICE_IN_USB_ACCESSORY, "USB_ACCESSORY"}, 486 {AUDIO_DEVICE_IN_USB_DEVICE, "USB_DEVICE"}, 487 {AUDIO_DEVICE_IN_FM_TUNER, "FM_TUNER"}, 488 {AUDIO_DEVICE_IN_TV_TUNER, "TV_TUNER"}, 489 {AUDIO_DEVICE_IN_LINE, "LINE"}, 490 {AUDIO_DEVICE_IN_SPDIF, "SPDIF"}, 491 {AUDIO_DEVICE_IN_BLUETOOTH_A2DP, "BLUETOOTH_A2DP"}, 492 {AUDIO_DEVICE_IN_LOOPBACK, "LOOPBACK"}, 493 {AUDIO_DEVICE_IN_IP, "IP"}, 494 {AUDIO_DEVICE_NONE, "NONE"}, // must be last 495 }; 496 String8 result; 497 audio_devices_t allDevices = AUDIO_DEVICE_NONE; 498 const mapping *entry; 499 if (devices & AUDIO_DEVICE_BIT_IN) { 500 devices &= ~AUDIO_DEVICE_BIT_IN; 501 entry = mappingsIn; 502 } else { 503 entry = mappingsOut; 504 } 505 for ( ; entry->mDevices != AUDIO_DEVICE_NONE; entry++) { 506 allDevices = (audio_devices_t) (allDevices | entry->mDevices); 507 if (devices & entry->mDevices) { 508 if (!result.isEmpty()) { 509 result.append("|"); 510 } 511 result.append(entry->mString); 512 } 513 } 514 if (devices & ~allDevices) { 515 if (!result.isEmpty()) { 516 result.append("|"); 517 } 518 result.appendFormat("0x%X", devices & ~allDevices); 519 } 520 if (result.isEmpty()) { 521 result.append(entry->mString); 522 } 523 return result; 524} 525 526String8 inputFlagsToString(audio_input_flags_t flags) 527{ 528 static const struct mapping { 529 audio_input_flags_t mFlag; 530 const char * mString; 531 } mappings[] = { 532 {AUDIO_INPUT_FLAG_FAST, "FAST"}, 533 {AUDIO_INPUT_FLAG_HW_HOTWORD, "HW_HOTWORD"}, 534 {AUDIO_INPUT_FLAG_RAW, "RAW"}, 535 {AUDIO_INPUT_FLAG_SYNC, "SYNC"}, 536 {AUDIO_INPUT_FLAG_NONE, "NONE"}, // must be last 537 }; 538 String8 result; 539 audio_input_flags_t allFlags = AUDIO_INPUT_FLAG_NONE; 540 const mapping *entry; 541 for (entry = mappings; entry->mFlag != AUDIO_INPUT_FLAG_NONE; entry++) { 542 allFlags = (audio_input_flags_t) (allFlags | entry->mFlag); 543 if (flags & entry->mFlag) { 544 if (!result.isEmpty()) { 545 result.append("|"); 546 } 547 result.append(entry->mString); 548 } 549 } 550 if (flags & ~allFlags) { 551 if (!result.isEmpty()) { 552 result.append("|"); 553 } 554 result.appendFormat("0x%X", flags & ~allFlags); 555 } 556 if (result.isEmpty()) { 557 result.append(entry->mString); 558 } 559 return result; 560} 561 562String8 outputFlagsToString(audio_output_flags_t flags) 563{ 564 static const struct mapping { 565 audio_output_flags_t mFlag; 566 const char * mString; 567 } mappings[] = { 568 {AUDIO_OUTPUT_FLAG_DIRECT, "DIRECT"}, 569 {AUDIO_OUTPUT_FLAG_PRIMARY, "PRIMARY"}, 570 {AUDIO_OUTPUT_FLAG_FAST, "FAST"}, 571 {AUDIO_OUTPUT_FLAG_DEEP_BUFFER, "DEEP_BUFFER"}, 572 {AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD,"COMPRESS_OFFLOAD"}, 573 {AUDIO_OUTPUT_FLAG_NON_BLOCKING, "NON_BLOCKING"}, 574 {AUDIO_OUTPUT_FLAG_HW_AV_SYNC, "HW_AV_SYNC"}, 575 {AUDIO_OUTPUT_FLAG_RAW, "RAW"}, 576 {AUDIO_OUTPUT_FLAG_SYNC, "SYNC"}, 577 {AUDIO_OUTPUT_FLAG_IEC958_NONAUDIO, "IEC958_NONAUDIO"}, 578 {AUDIO_OUTPUT_FLAG_NONE, "NONE"}, // must be last 579 }; 580 String8 result; 581 audio_output_flags_t allFlags = AUDIO_OUTPUT_FLAG_NONE; 582 const mapping *entry; 583 for (entry = mappings; entry->mFlag != AUDIO_OUTPUT_FLAG_NONE; entry++) { 584 allFlags = (audio_output_flags_t) (allFlags | entry->mFlag); 585 if (flags & entry->mFlag) { 586 if (!result.isEmpty()) { 587 result.append("|"); 588 } 589 result.append(entry->mString); 590 } 591 } 592 if (flags & ~allFlags) { 593 if (!result.isEmpty()) { 594 result.append("|"); 595 } 596 result.appendFormat("0x%X", flags & ~allFlags); 597 } 598 if (result.isEmpty()) { 599 result.append(entry->mString); 600 } 601 return result; 602} 603 604const char *sourceToString(audio_source_t source) 605{ 606 switch (source) { 607 case AUDIO_SOURCE_DEFAULT: return "default"; 608 case AUDIO_SOURCE_MIC: return "mic"; 609 case AUDIO_SOURCE_VOICE_UPLINK: return "voice uplink"; 610 case AUDIO_SOURCE_VOICE_DOWNLINK: return "voice downlink"; 611 case AUDIO_SOURCE_VOICE_CALL: return "voice call"; 612 case AUDIO_SOURCE_CAMCORDER: return "camcorder"; 613 case AUDIO_SOURCE_VOICE_RECOGNITION: return "voice recognition"; 614 case AUDIO_SOURCE_VOICE_COMMUNICATION: return "voice communication"; 615 case AUDIO_SOURCE_REMOTE_SUBMIX: return "remote submix"; 616 case AUDIO_SOURCE_UNPROCESSED: return "unprocessed"; 617 case AUDIO_SOURCE_FM_TUNER: return "FM tuner"; 618 case AUDIO_SOURCE_HOTWORD: return "hotword"; 619 default: return "unknown"; 620 } 621} 622 623AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, 624 audio_devices_t outDevice, audio_devices_t inDevice, type_t type, bool systemReady) 625 : Thread(false /*canCallJava*/), 626 mType(type), 627 mAudioFlinger(audioFlinger), 628 // mSampleRate, mFrameCount, mChannelMask, mChannelCount, mFrameSize, mFormat, mBufferSize 629 // are set by PlaybackThread::readOutputParameters_l() or 630 // RecordThread::readInputParameters_l() 631 //FIXME: mStandby should be true here. Is this some kind of hack? 632 mStandby(false), mOutDevice(outDevice), mInDevice(inDevice), 633 mPrevOutDevice(AUDIO_DEVICE_NONE), mPrevInDevice(AUDIO_DEVICE_NONE), 634 mAudioSource(AUDIO_SOURCE_DEFAULT), mId(id), 635 // mName will be set by concrete (non-virtual) subclass 636 mDeathRecipient(new PMDeathRecipient(this)), 637 mSystemReady(systemReady), 638 mNotifiedBatteryStart(false) 639{ 640 memset(&mPatch, 0, sizeof(struct audio_patch)); 641} 642 643AudioFlinger::ThreadBase::~ThreadBase() 644{ 645 // mConfigEvents should be empty, but just in case it isn't, free the memory it owns 646 mConfigEvents.clear(); 647 648 // do not lock the mutex in destructor 649 releaseWakeLock_l(); 650 if (mPowerManager != 0) { 651 sp<IBinder> binder = IInterface::asBinder(mPowerManager); 652 binder->unlinkToDeath(mDeathRecipient); 653 } 654} 655 656status_t AudioFlinger::ThreadBase::readyToRun() 657{ 658 status_t status = initCheck(); 659 if (status == NO_ERROR) { 660 ALOGI("AudioFlinger's thread %p ready to run", this); 661 } else { 662 ALOGE("No working audio driver found."); 663 } 664 return status; 665} 666 667void AudioFlinger::ThreadBase::exit() 668{ 669 ALOGV("ThreadBase::exit"); 670 // do any cleanup required for exit to succeed 671 preExit(); 672 { 673 // This lock prevents the following race in thread (uniprocessor for illustration): 674 // if (!exitPending()) { 675 // // context switch from here to exit() 676 // // exit() calls requestExit(), what exitPending() observes 677 // // exit() calls signal(), which is dropped since no waiters 678 // // context switch back from exit() to here 679 // mWaitWorkCV.wait(...); 680 // // now thread is hung 681 // } 682 AutoMutex lock(mLock); 683 requestExit(); 684 mWaitWorkCV.broadcast(); 685 } 686 // When Thread::requestExitAndWait is made virtual and this method is renamed to 687 // "virtual status_t requestExitAndWait()", replace by "return Thread::requestExitAndWait();" 688 requestExitAndWait(); 689} 690 691status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs) 692{ 693 status_t status; 694 695 ALOGV("ThreadBase::setParameters() %s", keyValuePairs.string()); 696 Mutex::Autolock _l(mLock); 697 698 return sendSetParameterConfigEvent_l(keyValuePairs); 699} 700 701// sendConfigEvent_l() must be called with ThreadBase::mLock held 702// Can temporarily release the lock if waiting for a reply from processConfigEvents_l(). 703status_t AudioFlinger::ThreadBase::sendConfigEvent_l(sp<ConfigEvent>& event) 704{ 705 status_t status = NO_ERROR; 706 707 if (event->mRequiresSystemReady && !mSystemReady) { 708 event->mWaitStatus = false; 709 mPendingConfigEvents.add(event); 710 return status; 711 } 712 mConfigEvents.add(event); 713 ALOGV("sendConfigEvent_l() num events %d event %d", mConfigEvents.size(), event->mType); 714 mWaitWorkCV.signal(); 715 mLock.unlock(); 716 { 717 Mutex::Autolock _l(event->mLock); 718 while (event->mWaitStatus) { 719 if (event->mCond.waitRelative(event->mLock, kConfigEventTimeoutNs) != NO_ERROR) { 720 event->mStatus = TIMED_OUT; 721 event->mWaitStatus = false; 722 } 723 } 724 status = event->mStatus; 725 } 726 mLock.lock(); 727 return status; 728} 729 730void AudioFlinger::ThreadBase::sendIoConfigEvent(audio_io_config_event event, pid_t pid) 731{ 732 Mutex::Autolock _l(mLock); 733 sendIoConfigEvent_l(event, pid); 734} 735 736// sendIoConfigEvent_l() must be called with ThreadBase::mLock held 737void AudioFlinger::ThreadBase::sendIoConfigEvent_l(audio_io_config_event event, pid_t pid) 738{ 739 sp<ConfigEvent> configEvent = (ConfigEvent *)new IoConfigEvent(event, pid); 740 sendConfigEvent_l(configEvent); 741} 742 743void AudioFlinger::ThreadBase::sendPrioConfigEvent(pid_t pid, pid_t tid, int32_t prio) 744{ 745 Mutex::Autolock _l(mLock); 746 sendPrioConfigEvent_l(pid, tid, prio); 747} 748 749// sendPrioConfigEvent_l() must be called with ThreadBase::mLock held 750void AudioFlinger::ThreadBase::sendPrioConfigEvent_l(pid_t pid, pid_t tid, int32_t prio) 751{ 752 sp<ConfigEvent> configEvent = (ConfigEvent *)new PrioConfigEvent(pid, tid, prio); 753 sendConfigEvent_l(configEvent); 754} 755 756// sendSetParameterConfigEvent_l() must be called with ThreadBase::mLock held 757status_t AudioFlinger::ThreadBase::sendSetParameterConfigEvent_l(const String8& keyValuePair) 758{ 759 sp<ConfigEvent> configEvent; 760 AudioParameter param(keyValuePair); 761 int value; 762 if (param.getInt(String8(AUDIO_PARAMETER_MONO_OUTPUT), value) == NO_ERROR) { 763 setMasterMono_l(value != 0); 764 if (param.size() == 1) { 765 return NO_ERROR; // should be a solo parameter - we don't pass down 766 } 767 param.remove(String8(AUDIO_PARAMETER_MONO_OUTPUT)); 768 configEvent = new SetParameterConfigEvent(param.toString()); 769 } else { 770 configEvent = new SetParameterConfigEvent(keyValuePair); 771 } 772 return sendConfigEvent_l(configEvent); 773} 774 775status_t AudioFlinger::ThreadBase::sendCreateAudioPatchConfigEvent( 776 const struct audio_patch *patch, 777 audio_patch_handle_t *handle) 778{ 779 Mutex::Autolock _l(mLock); 780 sp<ConfigEvent> configEvent = (ConfigEvent *)new CreateAudioPatchConfigEvent(*patch, *handle); 781 status_t status = sendConfigEvent_l(configEvent); 782 if (status == NO_ERROR) { 783 CreateAudioPatchConfigEventData *data = 784 (CreateAudioPatchConfigEventData *)configEvent->mData.get(); 785 *handle = data->mHandle; 786 } 787 return status; 788} 789 790status_t AudioFlinger::ThreadBase::sendReleaseAudioPatchConfigEvent( 791 const audio_patch_handle_t handle) 792{ 793 Mutex::Autolock _l(mLock); 794 sp<ConfigEvent> configEvent = (ConfigEvent *)new ReleaseAudioPatchConfigEvent(handle); 795 return sendConfigEvent_l(configEvent); 796} 797 798 799// post condition: mConfigEvents.isEmpty() 800void AudioFlinger::ThreadBase::processConfigEvents_l() 801{ 802 bool configChanged = false; 803 804 while (!mConfigEvents.isEmpty()) { 805 ALOGV("processConfigEvents_l() remaining events %d", mConfigEvents.size()); 806 sp<ConfigEvent> event = mConfigEvents[0]; 807 mConfigEvents.removeAt(0); 808 switch (event->mType) { 809 case CFG_EVENT_PRIO: { 810 PrioConfigEventData *data = (PrioConfigEventData *)event->mData.get(); 811 // FIXME Need to understand why this has to be done asynchronously 812 int err = requestPriority(data->mPid, data->mTid, data->mPrio, 813 true /*asynchronous*/); 814 if (err != 0) { 815 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 816 data->mPrio, data->mPid, data->mTid, err); 817 } 818 } break; 819 case CFG_EVENT_IO: { 820 IoConfigEventData *data = (IoConfigEventData *)event->mData.get(); 821 ioConfigChanged(data->mEvent, data->mPid); 822 } break; 823 case CFG_EVENT_SET_PARAMETER: { 824 SetParameterConfigEventData *data = (SetParameterConfigEventData *)event->mData.get(); 825 if (checkForNewParameter_l(data->mKeyValuePairs, event->mStatus)) { 826 configChanged = true; 827 } 828 } break; 829 case CFG_EVENT_CREATE_AUDIO_PATCH: { 830 CreateAudioPatchConfigEventData *data = 831 (CreateAudioPatchConfigEventData *)event->mData.get(); 832 event->mStatus = createAudioPatch_l(&data->mPatch, &data->mHandle); 833 } break; 834 case CFG_EVENT_RELEASE_AUDIO_PATCH: { 835 ReleaseAudioPatchConfigEventData *data = 836 (ReleaseAudioPatchConfigEventData *)event->mData.get(); 837 event->mStatus = releaseAudioPatch_l(data->mHandle); 838 } break; 839 default: 840 ALOG_ASSERT(false, "processConfigEvents_l() unknown event type %d", event->mType); 841 break; 842 } 843 { 844 Mutex::Autolock _l(event->mLock); 845 if (event->mWaitStatus) { 846 event->mWaitStatus = false; 847 event->mCond.signal(); 848 } 849 } 850 ALOGV_IF(mConfigEvents.isEmpty(), "processConfigEvents_l() DONE thread %p", this); 851 } 852 853 if (configChanged) { 854 cacheParameters_l(); 855 } 856} 857 858String8 channelMaskToString(audio_channel_mask_t mask, bool output) { 859 String8 s; 860 const audio_channel_representation_t representation = 861 audio_channel_mask_get_representation(mask); 862 863 switch (representation) { 864 case AUDIO_CHANNEL_REPRESENTATION_POSITION: { 865 if (output) { 866 if (mask & AUDIO_CHANNEL_OUT_FRONT_LEFT) s.append("front-left, "); 867 if (mask & AUDIO_CHANNEL_OUT_FRONT_RIGHT) s.append("front-right, "); 868 if (mask & AUDIO_CHANNEL_OUT_FRONT_CENTER) s.append("front-center, "); 869 if (mask & AUDIO_CHANNEL_OUT_LOW_FREQUENCY) s.append("low freq, "); 870 if (mask & AUDIO_CHANNEL_OUT_BACK_LEFT) s.append("back-left, "); 871 if (mask & AUDIO_CHANNEL_OUT_BACK_RIGHT) s.append("back-right, "); 872 if (mask & AUDIO_CHANNEL_OUT_FRONT_LEFT_OF_CENTER) s.append("front-left-of-center, "); 873 if (mask & AUDIO_CHANNEL_OUT_FRONT_RIGHT_OF_CENTER) s.append("front-right-of-center, "); 874 if (mask & AUDIO_CHANNEL_OUT_BACK_CENTER) s.append("back-center, "); 875 if (mask & AUDIO_CHANNEL_OUT_SIDE_LEFT) s.append("side-left, "); 876 if (mask & AUDIO_CHANNEL_OUT_SIDE_RIGHT) s.append("side-right, "); 877 if (mask & AUDIO_CHANNEL_OUT_TOP_CENTER) s.append("top-center ,"); 878 if (mask & AUDIO_CHANNEL_OUT_TOP_FRONT_LEFT) s.append("top-front-left, "); 879 if (mask & AUDIO_CHANNEL_OUT_TOP_FRONT_CENTER) s.append("top-front-center, "); 880 if (mask & AUDIO_CHANNEL_OUT_TOP_FRONT_RIGHT) s.append("top-front-right, "); 881 if (mask & AUDIO_CHANNEL_OUT_TOP_BACK_LEFT) s.append("top-back-left, "); 882 if (mask & AUDIO_CHANNEL_OUT_TOP_BACK_CENTER) s.append("top-back-center, " ); 883 if (mask & AUDIO_CHANNEL_OUT_TOP_BACK_RIGHT) s.append("top-back-right, " ); 884 if (mask & ~AUDIO_CHANNEL_OUT_ALL) s.append("unknown, "); 885 } else { 886 if (mask & AUDIO_CHANNEL_IN_LEFT) s.append("left, "); 887 if (mask & AUDIO_CHANNEL_IN_RIGHT) s.append("right, "); 888 if (mask & AUDIO_CHANNEL_IN_FRONT) s.append("front, "); 889 if (mask & AUDIO_CHANNEL_IN_BACK) s.append("back, "); 890 if (mask & AUDIO_CHANNEL_IN_LEFT_PROCESSED) s.append("left-processed, "); 891 if (mask & AUDIO_CHANNEL_IN_RIGHT_PROCESSED) s.append("right-processed, "); 892 if (mask & AUDIO_CHANNEL_IN_FRONT_PROCESSED) s.append("front-processed, "); 893 if (mask & AUDIO_CHANNEL_IN_BACK_PROCESSED) s.append("back-processed, "); 894 if (mask & AUDIO_CHANNEL_IN_PRESSURE) s.append("pressure, "); 895 if (mask & AUDIO_CHANNEL_IN_X_AXIS) s.append("X, "); 896 if (mask & AUDIO_CHANNEL_IN_Y_AXIS) s.append("Y, "); 897 if (mask & AUDIO_CHANNEL_IN_Z_AXIS) s.append("Z, "); 898 if (mask & AUDIO_CHANNEL_IN_VOICE_UPLINK) s.append("voice-uplink, "); 899 if (mask & AUDIO_CHANNEL_IN_VOICE_DNLINK) s.append("voice-dnlink, "); 900 if (mask & ~AUDIO_CHANNEL_IN_ALL) s.append("unknown, "); 901 } 902 const int len = s.length(); 903 if (len > 2) { 904 char *str = s.lockBuffer(len); // needed? 905 s.unlockBuffer(len - 2); // remove trailing ", " 906 } 907 return s; 908 } 909 case AUDIO_CHANNEL_REPRESENTATION_INDEX: 910 s.appendFormat("index mask, bits:%#x", audio_channel_mask_get_bits(mask)); 911 return s; 912 default: 913 s.appendFormat("unknown mask, representation:%d bits:%#x", 914 representation, audio_channel_mask_get_bits(mask)); 915 return s; 916 } 917} 918 919void AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args __unused) 920{ 921 const size_t SIZE = 256; 922 char buffer[SIZE]; 923 String8 result; 924 925 bool locked = AudioFlinger::dumpTryLock(mLock); 926 if (!locked) { 927 dprintf(fd, "thread %p may be deadlocked\n", this); 928 } 929 930 dprintf(fd, " Thread name: %s\n", mThreadName); 931 dprintf(fd, " I/O handle: %d\n", mId); 932 dprintf(fd, " TID: %d\n", getTid()); 933 dprintf(fd, " Standby: %s\n", mStandby ? "yes" : "no"); 934 dprintf(fd, " Sample rate: %u Hz\n", mSampleRate); 935 dprintf(fd, " HAL frame count: %zu\n", mFrameCount); 936 dprintf(fd, " HAL format: 0x%x (%s)\n", mHALFormat, formatToString(mHALFormat)); 937 dprintf(fd, " HAL buffer size: %u bytes\n", mBufferSize); 938 dprintf(fd, " Channel count: %u\n", mChannelCount); 939 dprintf(fd, " Channel mask: 0x%08x (%s)\n", mChannelMask, 940 channelMaskToString(mChannelMask, mType != RECORD).string()); 941 dprintf(fd, " Processing format: 0x%x (%s)\n", mFormat, formatToString(mFormat)); 942 dprintf(fd, " Processing frame size: %zu bytes\n", mFrameSize); 943 dprintf(fd, " Pending config events:"); 944 size_t numConfig = mConfigEvents.size(); 945 if (numConfig) { 946 for (size_t i = 0; i < numConfig; i++) { 947 mConfigEvents[i]->dump(buffer, SIZE); 948 dprintf(fd, "\n %s", buffer); 949 } 950 dprintf(fd, "\n"); 951 } else { 952 dprintf(fd, " none\n"); 953 } 954 dprintf(fd, " Output device: %#x (%s)\n", mOutDevice, devicesToString(mOutDevice).string()); 955 dprintf(fd, " Input device: %#x (%s)\n", mInDevice, devicesToString(mInDevice).string()); 956 dprintf(fd, " Audio source: %d (%s)\n", mAudioSource, sourceToString(mAudioSource)); 957 958 if (locked) { 959 mLock.unlock(); 960 } 961} 962 963void AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args) 964{ 965 const size_t SIZE = 256; 966 char buffer[SIZE]; 967 String8 result; 968 969 size_t numEffectChains = mEffectChains.size(); 970 snprintf(buffer, SIZE, " %zu Effect Chains\n", numEffectChains); 971 write(fd, buffer, strlen(buffer)); 972 973 for (size_t i = 0; i < numEffectChains; ++i) { 974 sp<EffectChain> chain = mEffectChains[i]; 975 if (chain != 0) { 976 chain->dump(fd, args); 977 } 978 } 979} 980 981void AudioFlinger::ThreadBase::acquireWakeLock(int uid) 982{ 983 Mutex::Autolock _l(mLock); 984 acquireWakeLock_l(uid); 985} 986 987String16 AudioFlinger::ThreadBase::getWakeLockTag() 988{ 989 switch (mType) { 990 case MIXER: 991 return String16("AudioMix"); 992 case DIRECT: 993 return String16("AudioDirectOut"); 994 case DUPLICATING: 995 return String16("AudioDup"); 996 case RECORD: 997 return String16("AudioIn"); 998 case OFFLOAD: 999 return String16("AudioOffload"); 1000 default: 1001 ALOG_ASSERT(false); 1002 return String16("AudioUnknown"); 1003 } 1004} 1005 1006void AudioFlinger::ThreadBase::acquireWakeLock_l(int uid) 1007{ 1008 getPowerManager_l(); 1009 if (mPowerManager != 0) { 1010 sp<IBinder> binder = new BBinder(); 1011 status_t status; 1012 if (uid >= 0) { 1013 status = mPowerManager->acquireWakeLockWithUid(POWERMANAGER_PARTIAL_WAKE_LOCK, 1014 binder, 1015 getWakeLockTag(), 1016 String16("audioserver"), 1017 uid, 1018 true /* FIXME force oneway contrary to .aidl */); 1019 } else { 1020 status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK, 1021 binder, 1022 getWakeLockTag(), 1023 String16("audioserver"), 1024 true /* FIXME force oneway contrary to .aidl */); 1025 } 1026 if (status == NO_ERROR) { 1027 mWakeLockToken = binder; 1028 } 1029 ALOGV("acquireWakeLock_l() %s status %d", mThreadName, status); 1030 } 1031 1032 if (!mNotifiedBatteryStart) { 1033 BatteryNotifier::getInstance().noteStartAudio(); 1034 mNotifiedBatteryStart = true; 1035 } 1036 gBoottime.acquire(mWakeLockToken); 1037} 1038 1039void AudioFlinger::ThreadBase::releaseWakeLock() 1040{ 1041 Mutex::Autolock _l(mLock); 1042 releaseWakeLock_l(); 1043} 1044 1045void AudioFlinger::ThreadBase::releaseWakeLock_l() 1046{ 1047 gBoottime.release(mWakeLockToken); 1048 if (mWakeLockToken != 0) { 1049 ALOGV("releaseWakeLock_l() %s", mThreadName); 1050 if (mPowerManager != 0) { 1051 mPowerManager->releaseWakeLock(mWakeLockToken, 0, 1052 true /* FIXME force oneway contrary to .aidl */); 1053 } 1054 mWakeLockToken.clear(); 1055 } 1056 1057 if (mNotifiedBatteryStart) { 1058 BatteryNotifier::getInstance().noteStopAudio(); 1059 mNotifiedBatteryStart = false; 1060 } 1061} 1062 1063void AudioFlinger::ThreadBase::updateWakeLockUids(const SortedVector<int> &uids) { 1064 Mutex::Autolock _l(mLock); 1065 updateWakeLockUids_l(uids); 1066} 1067 1068void AudioFlinger::ThreadBase::getPowerManager_l() { 1069 if (mSystemReady && mPowerManager == 0) { 1070 // use checkService() to avoid blocking if power service is not up yet 1071 sp<IBinder> binder = 1072 defaultServiceManager()->checkService(String16("power")); 1073 if (binder == 0) { 1074 ALOGW("Thread %s cannot connect to the power manager service", mThreadName); 1075 } else { 1076 mPowerManager = interface_cast<IPowerManager>(binder); 1077 binder->linkToDeath(mDeathRecipient); 1078 } 1079 } 1080} 1081 1082void AudioFlinger::ThreadBase::updateWakeLockUids_l(const SortedVector<int> &uids) { 1083 getPowerManager_l(); 1084 if (mWakeLockToken == NULL) { // token may be NULL if AudioFlinger::systemReady() not called. 1085 if (mSystemReady) { 1086 ALOGE("no wake lock to update, but system ready!"); 1087 } else { 1088 ALOGW("no wake lock to update, system not ready yet"); 1089 } 1090 return; 1091 } 1092 if (mPowerManager != 0) { 1093 sp<IBinder> binder = new BBinder(); 1094 status_t status; 1095 status = mPowerManager->updateWakeLockUids(mWakeLockToken, uids.size(), uids.array(), 1096 true /* FIXME force oneway contrary to .aidl */); 1097 ALOGV("acquireWakeLock_l() %s status %d", mThreadName, status); 1098 } 1099} 1100 1101void AudioFlinger::ThreadBase::clearPowerManager() 1102{ 1103 Mutex::Autolock _l(mLock); 1104 releaseWakeLock_l(); 1105 mPowerManager.clear(); 1106} 1107 1108void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who __unused) 1109{ 1110 sp<ThreadBase> thread = mThread.promote(); 1111 if (thread != 0) { 1112 thread->clearPowerManager(); 1113 } 1114 ALOGW("power manager service died !!!"); 1115} 1116 1117void AudioFlinger::ThreadBase::setEffectSuspended( 1118 const effect_uuid_t *type, bool suspend, int sessionId) 1119{ 1120 Mutex::Autolock _l(mLock); 1121 setEffectSuspended_l(type, suspend, sessionId); 1122} 1123 1124void AudioFlinger::ThreadBase::setEffectSuspended_l( 1125 const effect_uuid_t *type, bool suspend, int sessionId) 1126{ 1127 sp<EffectChain> chain = getEffectChain_l(sessionId); 1128 if (chain != 0) { 1129 if (type != NULL) { 1130 chain->setEffectSuspended_l(type, suspend); 1131 } else { 1132 chain->setEffectSuspendedAll_l(suspend); 1133 } 1134 } 1135 1136 updateSuspendedSessions_l(type, suspend, sessionId); 1137} 1138 1139void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain) 1140{ 1141 ssize_t index = mSuspendedSessions.indexOfKey(chain->sessionId()); 1142 if (index < 0) { 1143 return; 1144 } 1145 1146 const KeyedVector <int, sp<SuspendedSessionDesc> >& sessionEffects = 1147 mSuspendedSessions.valueAt(index); 1148 1149 for (size_t i = 0; i < sessionEffects.size(); i++) { 1150 sp<SuspendedSessionDesc> desc = sessionEffects.valueAt(i); 1151 for (int j = 0; j < desc->mRefCount; j++) { 1152 if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) { 1153 chain->setEffectSuspendedAll_l(true); 1154 } else { 1155 ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x", 1156 desc->mType.timeLow); 1157 chain->setEffectSuspended_l(&desc->mType, true); 1158 } 1159 } 1160 } 1161} 1162 1163void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type, 1164 bool suspend, 1165 int sessionId) 1166{ 1167 ssize_t index = mSuspendedSessions.indexOfKey(sessionId); 1168 1169 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects; 1170 1171 if (suspend) { 1172 if (index >= 0) { 1173 sessionEffects = mSuspendedSessions.valueAt(index); 1174 } else { 1175 mSuspendedSessions.add(sessionId, sessionEffects); 1176 } 1177 } else { 1178 if (index < 0) { 1179 return; 1180 } 1181 sessionEffects = mSuspendedSessions.valueAt(index); 1182 } 1183 1184 1185 int key = EffectChain::kKeyForSuspendAll; 1186 if (type != NULL) { 1187 key = type->timeLow; 1188 } 1189 index = sessionEffects.indexOfKey(key); 1190 1191 sp<SuspendedSessionDesc> desc; 1192 if (suspend) { 1193 if (index >= 0) { 1194 desc = sessionEffects.valueAt(index); 1195 } else { 1196 desc = new SuspendedSessionDesc(); 1197 if (type != NULL) { 1198 desc->mType = *type; 1199 } 1200 sessionEffects.add(key, desc); 1201 ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key); 1202 } 1203 desc->mRefCount++; 1204 } else { 1205 if (index < 0) { 1206 return; 1207 } 1208 desc = sessionEffects.valueAt(index); 1209 if (--desc->mRefCount == 0) { 1210 ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key); 1211 sessionEffects.removeItemsAt(index); 1212 if (sessionEffects.isEmpty()) { 1213 ALOGV("updateSuspendedSessions_l() restore removing session %d", 1214 sessionId); 1215 mSuspendedSessions.removeItem(sessionId); 1216 } 1217 } 1218 } 1219 if (!sessionEffects.isEmpty()) { 1220 mSuspendedSessions.replaceValueFor(sessionId, sessionEffects); 1221 } 1222} 1223 1224void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 1225 bool enabled, 1226 int sessionId) 1227{ 1228 Mutex::Autolock _l(mLock); 1229 checkSuspendOnEffectEnabled_l(effect, enabled, sessionId); 1230} 1231 1232void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 1233 bool enabled, 1234 int sessionId) 1235{ 1236 if (mType != RECORD) { 1237 // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on 1238 // another session. This gives the priority to well behaved effect control panels 1239 // and applications not using global effects. 1240 // Enabling post processing in AUDIO_SESSION_OUTPUT_STAGE session does not affect 1241 // global effects 1242 if ((sessionId != AUDIO_SESSION_OUTPUT_MIX) && (sessionId != AUDIO_SESSION_OUTPUT_STAGE)) { 1243 setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX); 1244 } 1245 } 1246 1247 sp<EffectChain> chain = getEffectChain_l(sessionId); 1248 if (chain != 0) { 1249 chain->checkSuspendOnEffectEnabled(effect, enabled); 1250 } 1251} 1252 1253// ThreadBase::createEffect_l() must be called with AudioFlinger::mLock held 1254sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 1255 const sp<AudioFlinger::Client>& client, 1256 const sp<IEffectClient>& effectClient, 1257 int32_t priority, 1258 int sessionId, 1259 effect_descriptor_t *desc, 1260 int *enabled, 1261 status_t *status) 1262{ 1263 sp<EffectModule> effect; 1264 sp<EffectHandle> handle; 1265 status_t lStatus; 1266 sp<EffectChain> chain; 1267 bool chainCreated = false; 1268 bool effectCreated = false; 1269 bool effectRegistered = false; 1270 1271 lStatus = initCheck(); 1272 if (lStatus != NO_ERROR) { 1273 ALOGW("createEffect_l() Audio driver not initialized."); 1274 goto Exit; 1275 } 1276 1277 // Reject any effect on Direct output threads for now, since the format of 1278 // mSinkBuffer is not guaranteed to be compatible with effect processing (PCM 16 stereo). 1279 if (mType == DIRECT) { 1280 ALOGW("createEffect_l() Cannot add effect %s on Direct output type thread %s", 1281 desc->name, mThreadName); 1282 lStatus = BAD_VALUE; 1283 goto Exit; 1284 } 1285 1286 // Reject any effect on mixer or duplicating multichannel sinks. 1287 // TODO: fix both format and multichannel issues with effects. 1288 if ((mType == MIXER || mType == DUPLICATING) && mChannelCount != FCC_2) { 1289 ALOGW("createEffect_l() Cannot add effect %s for multichannel(%d) %s threads", 1290 desc->name, mChannelCount, mType == MIXER ? "MIXER" : "DUPLICATING"); 1291 lStatus = BAD_VALUE; 1292 goto Exit; 1293 } 1294 1295 // Allow global effects only on offloaded and mixer threads 1296 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 1297 switch (mType) { 1298 case MIXER: 1299 case OFFLOAD: 1300 break; 1301 case DIRECT: 1302 case DUPLICATING: 1303 case RECORD: 1304 default: 1305 ALOGW("createEffect_l() Cannot add global effect %s on thread %s", 1306 desc->name, mThreadName); 1307 lStatus = BAD_VALUE; 1308 goto Exit; 1309 } 1310 } 1311 1312 // Only Pre processor effects are allowed on input threads and only on input threads 1313 if ((mType == RECORD) != ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { 1314 ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d", 1315 desc->name, desc->flags, mType); 1316 lStatus = BAD_VALUE; 1317 goto Exit; 1318 } 1319 1320 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 1321 1322 { // scope for mLock 1323 Mutex::Autolock _l(mLock); 1324 1325 // check for existing effect chain with the requested audio session 1326 chain = getEffectChain_l(sessionId); 1327 if (chain == 0) { 1328 // create a new chain for this session 1329 ALOGV("createEffect_l() new effect chain for session %d", sessionId); 1330 chain = new EffectChain(this, sessionId); 1331 addEffectChain_l(chain); 1332 chain->setStrategy(getStrategyForSession_l(sessionId)); 1333 chainCreated = true; 1334 } else { 1335 effect = chain->getEffectFromDesc_l(desc); 1336 } 1337 1338 ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get()); 1339 1340 if (effect == 0) { 1341 int id = mAudioFlinger->nextUniqueId(); 1342 // Check CPU and memory usage 1343 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 1344 if (lStatus != NO_ERROR) { 1345 goto Exit; 1346 } 1347 effectRegistered = true; 1348 // create a new effect module if none present in the chain 1349 effect = new EffectModule(this, chain, desc, id, sessionId); 1350 lStatus = effect->status(); 1351 if (lStatus != NO_ERROR) { 1352 goto Exit; 1353 } 1354 effect->setOffloaded(mType == OFFLOAD, mId); 1355 1356 lStatus = chain->addEffect_l(effect); 1357 if (lStatus != NO_ERROR) { 1358 goto Exit; 1359 } 1360 effectCreated = true; 1361 1362 effect->setDevice(mOutDevice); 1363 effect->setDevice(mInDevice); 1364 effect->setMode(mAudioFlinger->getMode()); 1365 effect->setAudioSource(mAudioSource); 1366 } 1367 // create effect handle and connect it to effect module 1368 handle = new EffectHandle(effect, client, effectClient, priority); 1369 lStatus = handle->initCheck(); 1370 if (lStatus == OK) { 1371 lStatus = effect->addHandle(handle.get()); 1372 } 1373 if (enabled != NULL) { 1374 *enabled = (int)effect->isEnabled(); 1375 } 1376 } 1377 1378Exit: 1379 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 1380 Mutex::Autolock _l(mLock); 1381 if (effectCreated) { 1382 chain->removeEffect_l(effect); 1383 } 1384 if (effectRegistered) { 1385 AudioSystem::unregisterEffect(effect->id()); 1386 } 1387 if (chainCreated) { 1388 removeEffectChain_l(chain); 1389 } 1390 handle.clear(); 1391 } 1392 1393 *status = lStatus; 1394 return handle; 1395} 1396 1397sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect(int sessionId, int effectId) 1398{ 1399 Mutex::Autolock _l(mLock); 1400 return getEffect_l(sessionId, effectId); 1401} 1402 1403sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId) 1404{ 1405 sp<EffectChain> chain = getEffectChain_l(sessionId); 1406 return chain != 0 ? chain->getEffectFromId_l(effectId) : 0; 1407} 1408 1409// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 1410// PlaybackThread::mLock held 1411status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 1412{ 1413 // check for existing effect chain with the requested audio session 1414 int sessionId = effect->sessionId(); 1415 sp<EffectChain> chain = getEffectChain_l(sessionId); 1416 bool chainCreated = false; 1417 1418 ALOGD_IF((mType == OFFLOAD) && !effect->isOffloadable(), 1419 "addEffect_l() on offloaded thread %p: effect %s does not support offload flags %x", 1420 this, effect->desc().name, effect->desc().flags); 1421 1422 if (chain == 0) { 1423 // create a new chain for this session 1424 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 1425 chain = new EffectChain(this, sessionId); 1426 addEffectChain_l(chain); 1427 chain->setStrategy(getStrategyForSession_l(sessionId)); 1428 chainCreated = true; 1429 } 1430 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 1431 1432 if (chain->getEffectFromId_l(effect->id()) != 0) { 1433 ALOGW("addEffect_l() %p effect %s already present in chain %p", 1434 this, effect->desc().name, chain.get()); 1435 return BAD_VALUE; 1436 } 1437 1438 effect->setOffloaded(mType == OFFLOAD, mId); 1439 1440 status_t status = chain->addEffect_l(effect); 1441 if (status != NO_ERROR) { 1442 if (chainCreated) { 1443 removeEffectChain_l(chain); 1444 } 1445 return status; 1446 } 1447 1448 effect->setDevice(mOutDevice); 1449 effect->setDevice(mInDevice); 1450 effect->setMode(mAudioFlinger->getMode()); 1451 effect->setAudioSource(mAudioSource); 1452 return NO_ERROR; 1453} 1454 1455void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 1456 1457 ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 1458 effect_descriptor_t desc = effect->desc(); 1459 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 1460 detachAuxEffect_l(effect->id()); 1461 } 1462 1463 sp<EffectChain> chain = effect->chain().promote(); 1464 if (chain != 0) { 1465 // remove effect chain if removing last effect 1466 if (chain->removeEffect_l(effect) == 0) { 1467 removeEffectChain_l(chain); 1468 } 1469 } else { 1470 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 1471 } 1472} 1473 1474void AudioFlinger::ThreadBase::lockEffectChains_l( 1475 Vector< sp<AudioFlinger::EffectChain> >& effectChains) 1476{ 1477 effectChains = mEffectChains; 1478 for (size_t i = 0; i < mEffectChains.size(); i++) { 1479 mEffectChains[i]->lock(); 1480 } 1481} 1482 1483void AudioFlinger::ThreadBase::unlockEffectChains( 1484 const Vector< sp<AudioFlinger::EffectChain> >& effectChains) 1485{ 1486 for (size_t i = 0; i < effectChains.size(); i++) { 1487 effectChains[i]->unlock(); 1488 } 1489} 1490 1491sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId) 1492{ 1493 Mutex::Autolock _l(mLock); 1494 return getEffectChain_l(sessionId); 1495} 1496 1497sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) const 1498{ 1499 size_t size = mEffectChains.size(); 1500 for (size_t i = 0; i < size; i++) { 1501 if (mEffectChains[i]->sessionId() == sessionId) { 1502 return mEffectChains[i]; 1503 } 1504 } 1505 return 0; 1506} 1507 1508void AudioFlinger::ThreadBase::setMode(audio_mode_t mode) 1509{ 1510 Mutex::Autolock _l(mLock); 1511 size_t size = mEffectChains.size(); 1512 for (size_t i = 0; i < size; i++) { 1513 mEffectChains[i]->setMode_l(mode); 1514 } 1515} 1516 1517void AudioFlinger::ThreadBase::getAudioPortConfig(struct audio_port_config *config) 1518{ 1519 config->type = AUDIO_PORT_TYPE_MIX; 1520 config->ext.mix.handle = mId; 1521 config->sample_rate = mSampleRate; 1522 config->format = mFormat; 1523 config->channel_mask = mChannelMask; 1524 config->config_mask = AUDIO_PORT_CONFIG_SAMPLE_RATE|AUDIO_PORT_CONFIG_CHANNEL_MASK| 1525 AUDIO_PORT_CONFIG_FORMAT; 1526} 1527 1528void AudioFlinger::ThreadBase::systemReady() 1529{ 1530 Mutex::Autolock _l(mLock); 1531 if (mSystemReady) { 1532 return; 1533 } 1534 mSystemReady = true; 1535 1536 for (size_t i = 0; i < mPendingConfigEvents.size(); i++) { 1537 sendConfigEvent_l(mPendingConfigEvents.editItemAt(i)); 1538 } 1539 mPendingConfigEvents.clear(); 1540} 1541 1542 1543// ---------------------------------------------------------------------------- 1544// Playback 1545// ---------------------------------------------------------------------------- 1546 1547AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, 1548 AudioStreamOut* output, 1549 audio_io_handle_t id, 1550 audio_devices_t device, 1551 type_t type, 1552 bool systemReady) 1553 : ThreadBase(audioFlinger, id, device, AUDIO_DEVICE_NONE, type, systemReady), 1554 mNormalFrameCount(0), mSinkBuffer(NULL), 1555 mMixerBufferEnabled(AudioFlinger::kEnableExtendedPrecision), 1556 mMixerBuffer(NULL), 1557 mMixerBufferSize(0), 1558 mMixerBufferFormat(AUDIO_FORMAT_INVALID), 1559 mMixerBufferValid(false), 1560 mEffectBufferEnabled(AudioFlinger::kEnableExtendedPrecision), 1561 mEffectBuffer(NULL), 1562 mEffectBufferSize(0), 1563 mEffectBufferFormat(AUDIO_FORMAT_INVALID), 1564 mEffectBufferValid(false), 1565 mSuspended(0), mBytesWritten(0), 1566 mActiveTracksGeneration(0), 1567 // mStreamTypes[] initialized in constructor body 1568 mOutput(output), 1569 mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false), 1570 mMixerStatus(MIXER_IDLE), 1571 mMixerStatusIgnoringFastTracks(MIXER_IDLE), 1572 mStandbyDelayNs(AudioFlinger::mStandbyTimeInNsecs), 1573 mBytesRemaining(0), 1574 mCurrentWriteLength(0), 1575 mUseAsyncWrite(false), 1576 mWriteAckSequence(0), 1577 mDrainSequence(0), 1578 mSignalPending(false), 1579 mScreenState(AudioFlinger::mScreenState), 1580 // index 0 is reserved for normal mixer's submix 1581 mFastTrackAvailMask(((1 << FastMixerState::kMaxFastTracks) - 1) & ~1), 1582 mHwSupportsPause(false), mHwPaused(false), mFlushPending(false), 1583 // mLatchD, mLatchQ, 1584 mLatchDValid(false), mLatchQValid(false) 1585{ 1586 snprintf(mThreadName, kThreadNameLength, "AudioOut_%X", id); 1587 mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mThreadName); 1588 1589 // Assumes constructor is called by AudioFlinger with it's mLock held, but 1590 // it would be safer to explicitly pass initial masterVolume/masterMute as 1591 // parameter. 1592 // 1593 // If the HAL we are using has support for master volume or master mute, 1594 // then do not attenuate or mute during mixing (just leave the volume at 1.0 1595 // and the mute set to false). 1596 mMasterVolume = audioFlinger->masterVolume_l(); 1597 mMasterMute = audioFlinger->masterMute_l(); 1598 if (mOutput && mOutput->audioHwDev) { 1599 if (mOutput->audioHwDev->canSetMasterVolume()) { 1600 mMasterVolume = 1.0; 1601 } 1602 1603 if (mOutput->audioHwDev->canSetMasterMute()) { 1604 mMasterMute = false; 1605 } 1606 } 1607 1608 readOutputParameters_l(); 1609 1610 // ++ operator does not compile 1611 for (audio_stream_type_t stream = AUDIO_STREAM_MIN; stream < AUDIO_STREAM_CNT; 1612 stream = (audio_stream_type_t) (stream + 1)) { 1613 mStreamTypes[stream].volume = mAudioFlinger->streamVolume_l(stream); 1614 mStreamTypes[stream].mute = mAudioFlinger->streamMute_l(stream); 1615 } 1616} 1617 1618AudioFlinger::PlaybackThread::~PlaybackThread() 1619{ 1620 mAudioFlinger->unregisterWriter(mNBLogWriter); 1621 free(mSinkBuffer); 1622 free(mMixerBuffer); 1623 free(mEffectBuffer); 1624} 1625 1626void AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) 1627{ 1628 dumpInternals(fd, args); 1629 dumpTracks(fd, args); 1630 dumpEffectChains(fd, args); 1631} 1632 1633void AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args __unused) 1634{ 1635 const size_t SIZE = 256; 1636 char buffer[SIZE]; 1637 String8 result; 1638 1639 result.appendFormat(" Stream volumes in dB: "); 1640 for (int i = 0; i < AUDIO_STREAM_CNT; ++i) { 1641 const stream_type_t *st = &mStreamTypes[i]; 1642 if (i > 0) { 1643 result.appendFormat(", "); 1644 } 1645 result.appendFormat("%d:%.2g", i, 20.0 * log10(st->volume)); 1646 if (st->mute) { 1647 result.append("M"); 1648 } 1649 } 1650 result.append("\n"); 1651 write(fd, result.string(), result.length()); 1652 result.clear(); 1653 1654 // These values are "raw"; they will wrap around. See prepareTracks_l() for a better way. 1655 FastTrackUnderruns underruns = getFastTrackUnderruns(0); 1656 dprintf(fd, " Normal mixer raw underrun counters: partial=%u empty=%u\n", 1657 underruns.mBitFields.mPartial, underruns.mBitFields.mEmpty); 1658 1659 size_t numtracks = mTracks.size(); 1660 size_t numactive = mActiveTracks.size(); 1661 dprintf(fd, " %d Tracks", numtracks); 1662 size_t numactiveseen = 0; 1663 if (numtracks) { 1664 dprintf(fd, " of which %d are active\n", numactive); 1665 Track::appendDumpHeader(result); 1666 for (size_t i = 0; i < numtracks; ++i) { 1667 sp<Track> track = mTracks[i]; 1668 if (track != 0) { 1669 bool active = mActiveTracks.indexOf(track) >= 0; 1670 if (active) { 1671 numactiveseen++; 1672 } 1673 track->dump(buffer, SIZE, active); 1674 result.append(buffer); 1675 } 1676 } 1677 } else { 1678 result.append("\n"); 1679 } 1680 if (numactiveseen != numactive) { 1681 // some tracks in the active list were not in the tracks list 1682 snprintf(buffer, SIZE, " The following tracks are in the active list but" 1683 " not in the track list\n"); 1684 result.append(buffer); 1685 Track::appendDumpHeader(result); 1686 for (size_t i = 0; i < numactive; ++i) { 1687 sp<Track> track = mActiveTracks[i].promote(); 1688 if (track != 0 && mTracks.indexOf(track) < 0) { 1689 track->dump(buffer, SIZE, true); 1690 result.append(buffer); 1691 } 1692 } 1693 } 1694 1695 write(fd, result.string(), result.size()); 1696} 1697 1698void AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 1699{ 1700 dprintf(fd, "\nOutput thread %p type %d (%s):\n", this, type(), threadTypeToString(type())); 1701 1702 dumpBase(fd, args); 1703 1704 dprintf(fd, " Normal frame count: %zu\n", mNormalFrameCount); 1705 dprintf(fd, " Last write occurred (msecs): %llu\n", ns2ms(systemTime() - mLastWriteTime)); 1706 dprintf(fd, " Total writes: %d\n", mNumWrites); 1707 dprintf(fd, " Delayed writes: %d\n", mNumDelayedWrites); 1708 dprintf(fd, " Blocked in write: %s\n", mInWrite ? "yes" : "no"); 1709 dprintf(fd, " Suspend count: %d\n", mSuspended); 1710 dprintf(fd, " Sink buffer : %p\n", mSinkBuffer); 1711 dprintf(fd, " Mixer buffer: %p\n", mMixerBuffer); 1712 dprintf(fd, " Effect buffer: %p\n", mEffectBuffer); 1713 dprintf(fd, " Fast track availMask=%#x\n", mFastTrackAvailMask); 1714 dprintf(fd, " Standby delay ns=%lld\n", (long long)mStandbyDelayNs); 1715 AudioStreamOut *output = mOutput; 1716 audio_output_flags_t flags = output != NULL ? output->flags : AUDIO_OUTPUT_FLAG_NONE; 1717 String8 flagsAsString = outputFlagsToString(flags); 1718 dprintf(fd, " AudioStreamOut: %p flags %#x (%s)\n", output, flags, flagsAsString.string()); 1719} 1720 1721// Thread virtuals 1722 1723void AudioFlinger::PlaybackThread::onFirstRef() 1724{ 1725 run(mThreadName, ANDROID_PRIORITY_URGENT_AUDIO); 1726} 1727 1728// ThreadBase virtuals 1729void AudioFlinger::PlaybackThread::preExit() 1730{ 1731 ALOGV(" preExit()"); 1732 // FIXME this is using hard-coded strings but in the future, this functionality will be 1733 // converted to use audio HAL extensions required to support tunneling 1734 mOutput->stream->common.set_parameters(&mOutput->stream->common, "exiting=1"); 1735} 1736 1737// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 1738sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 1739 const sp<AudioFlinger::Client>& client, 1740 audio_stream_type_t streamType, 1741 uint32_t sampleRate, 1742 audio_format_t format, 1743 audio_channel_mask_t channelMask, 1744 size_t *pFrameCount, 1745 const sp<IMemory>& sharedBuffer, 1746 int sessionId, 1747 IAudioFlinger::track_flags_t *flags, 1748 pid_t tid, 1749 int uid, 1750 status_t *status) 1751{ 1752 size_t frameCount = *pFrameCount; 1753 sp<Track> track; 1754 status_t lStatus; 1755 1756 // client expresses a preference for FAST, but we get the final say 1757 if (*flags & IAudioFlinger::TRACK_FAST) { 1758 if ( 1759 // either of these use cases: 1760 ( 1761 // use case 1: shared buffer with any frame count 1762 ( 1763 (sharedBuffer != 0) 1764 ) || 1765 // use case 2: frame count is default or at least as large as HAL 1766 ( 1767 // we formerly checked for a callback handler (non-0 tid), 1768 // but that is no longer required for TRANSFER_OBTAIN mode 1769 ((frameCount == 0) || 1770 (frameCount >= mFrameCount)) 1771 ) 1772 ) && 1773 // PCM data 1774 audio_is_linear_pcm(format) && 1775 // TODO: extract as a data library function that checks that a computationally 1776 // expensive downmixer is not required: isFastOutputChannelConversion() 1777 (channelMask == mChannelMask || 1778 mChannelMask != AUDIO_CHANNEL_OUT_STEREO || 1779 (channelMask == AUDIO_CHANNEL_OUT_MONO 1780 /* && mChannelMask == AUDIO_CHANNEL_OUT_STEREO */)) && 1781 // hardware sample rate 1782 (sampleRate == mSampleRate) && 1783 // normal mixer has an associated fast mixer 1784 hasFastMixer() && 1785 // there are sufficient fast track slots available 1786 (mFastTrackAvailMask != 0) 1787 // FIXME test that MixerThread for this fast track has a capable output HAL 1788 // FIXME add a permission test also? 1789 ) { 1790 // if frameCount not specified, then it defaults to fast mixer (HAL) frame count 1791 if (frameCount == 0) { 1792 // read the fast track multiplier property the first time it is needed 1793 int ok = pthread_once(&sFastTrackMultiplierOnce, sFastTrackMultiplierInit); 1794 if (ok != 0) { 1795 ALOGE("%s pthread_once failed: %d", __func__, ok); 1796 } 1797 frameCount = mFrameCount * sFastTrackMultiplier; 1798 } 1799 ALOGV("AUDIO_OUTPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d", 1800 frameCount, mFrameCount); 1801 } else { 1802 ALOGV("AUDIO_OUTPUT_FLAG_FAST denied: sharedBuffer=%p frameCount=%d " 1803 "mFrameCount=%d format=%#x mFormat=%#x isLinear=%d channelMask=%#x " 1804 "sampleRate=%u mSampleRate=%u " 1805 "hasFastMixer=%d tid=%d fastTrackAvailMask=%#x", 1806 sharedBuffer.get(), frameCount, mFrameCount, format, mFormat, 1807 audio_is_linear_pcm(format), 1808 channelMask, sampleRate, mSampleRate, hasFastMixer(), tid, mFastTrackAvailMask); 1809 *flags &= ~IAudioFlinger::TRACK_FAST; 1810 } 1811 } 1812 // For normal PCM streaming tracks, update minimum frame count. 1813 // For compatibility with AudioTrack calculation, buffer depth is forced 1814 // to be at least 2 x the normal mixer frame count and cover audio hardware latency. 1815 // This is probably too conservative, but legacy application code may depend on it. 1816 // If you change this calculation, also review the start threshold which is related. 1817 if (!(*flags & IAudioFlinger::TRACK_FAST) 1818 && audio_is_linear_pcm(format) && sharedBuffer == 0) { 1819 // this must match AudioTrack.cpp calculateMinFrameCount(). 1820 // TODO: Move to a common library 1821 uint32_t latencyMs = mOutput->stream->get_latency(mOutput->stream); 1822 uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate); 1823 if (minBufCount < 2) { 1824 minBufCount = 2; 1825 } 1826 // For normal mixing tracks, if speed is > 1.0f (normal), AudioTrack 1827 // or the client should compute and pass in a larger buffer request. 1828 size_t minFrameCount = 1829 minBufCount * sourceFramesNeededWithTimestretch( 1830 sampleRate, mNormalFrameCount, 1831 mSampleRate, AUDIO_TIMESTRETCH_SPEED_NORMAL /*speed*/); 1832 if (frameCount < minFrameCount) { // including frameCount == 0 1833 frameCount = minFrameCount; 1834 } 1835 } 1836 *pFrameCount = frameCount; 1837 1838 switch (mType) { 1839 1840 case DIRECT: 1841 if (audio_is_linear_pcm(format)) { 1842 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1843 ALOGE("createTrack_l() Bad parameter: sampleRate %u format %#x, channelMask 0x%08x " 1844 "for output %p with format %#x", 1845 sampleRate, format, channelMask, mOutput, mFormat); 1846 lStatus = BAD_VALUE; 1847 goto Exit; 1848 } 1849 } 1850 break; 1851 1852 case OFFLOAD: 1853 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1854 ALOGE("createTrack_l() Bad parameter: sampleRate %d format %#x, channelMask 0x%08x \"" 1855 "for output %p with format %#x", 1856 sampleRate, format, channelMask, mOutput, mFormat); 1857 lStatus = BAD_VALUE; 1858 goto Exit; 1859 } 1860 break; 1861 1862 default: 1863 if (!audio_is_linear_pcm(format)) { 1864 ALOGE("createTrack_l() Bad parameter: format %#x \"" 1865 "for output %p with format %#x", 1866 format, mOutput, mFormat); 1867 lStatus = BAD_VALUE; 1868 goto Exit; 1869 } 1870 if (sampleRate > mSampleRate * AUDIO_RESAMPLER_DOWN_RATIO_MAX) { 1871 ALOGE("Sample rate out of range: %u mSampleRate %u", sampleRate, mSampleRate); 1872 lStatus = BAD_VALUE; 1873 goto Exit; 1874 } 1875 break; 1876 1877 } 1878 1879 lStatus = initCheck(); 1880 if (lStatus != NO_ERROR) { 1881 ALOGE("createTrack_l() audio driver not initialized"); 1882 goto Exit; 1883 } 1884 1885 { // scope for mLock 1886 Mutex::Autolock _l(mLock); 1887 1888 // all tracks in same audio session must share the same routing strategy otherwise 1889 // conflicts will happen when tracks are moved from one output to another by audio policy 1890 // manager 1891 uint32_t strategy = AudioSystem::getStrategyForStream(streamType); 1892 for (size_t i = 0; i < mTracks.size(); ++i) { 1893 sp<Track> t = mTracks[i]; 1894 if (t != 0 && t->isExternalTrack()) { 1895 uint32_t actual = AudioSystem::getStrategyForStream(t->streamType()); 1896 if (sessionId == t->sessionId() && strategy != actual) { 1897 ALOGE("createTrack_l() mismatched strategy; expected %u but found %u", 1898 strategy, actual); 1899 lStatus = BAD_VALUE; 1900 goto Exit; 1901 } 1902 } 1903 } 1904 1905 track = new Track(this, client, streamType, sampleRate, format, 1906 channelMask, frameCount, NULL, sharedBuffer, 1907 sessionId, uid, *flags, TrackBase::TYPE_DEFAULT); 1908 1909 lStatus = track != 0 ? track->initCheck() : (status_t) NO_MEMORY; 1910 if (lStatus != NO_ERROR) { 1911 ALOGE("createTrack_l() initCheck failed %d; no control block?", lStatus); 1912 // track must be cleared from the caller as the caller has the AF lock 1913 goto Exit; 1914 } 1915 mTracks.add(track); 1916 1917 sp<EffectChain> chain = getEffectChain_l(sessionId); 1918 if (chain != 0) { 1919 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1920 track->setMainBuffer(chain->inBuffer()); 1921 chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType())); 1922 chain->incTrackCnt(); 1923 } 1924 1925 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 1926 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 1927 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 1928 // so ask activity manager to do this on our behalf 1929 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 1930 } 1931 } 1932 1933 lStatus = NO_ERROR; 1934 1935Exit: 1936 *status = lStatus; 1937 return track; 1938} 1939 1940uint32_t AudioFlinger::PlaybackThread::correctLatency_l(uint32_t latency) const 1941{ 1942 return latency; 1943} 1944 1945uint32_t AudioFlinger::PlaybackThread::latency() const 1946{ 1947 Mutex::Autolock _l(mLock); 1948 return latency_l(); 1949} 1950uint32_t AudioFlinger::PlaybackThread::latency_l() const 1951{ 1952 if (initCheck() == NO_ERROR) { 1953 return correctLatency_l(mOutput->stream->get_latency(mOutput->stream)); 1954 } else { 1955 return 0; 1956 } 1957} 1958 1959void AudioFlinger::PlaybackThread::setMasterVolume(float value) 1960{ 1961 Mutex::Autolock _l(mLock); 1962 // Don't apply master volume in SW if our HAL can do it for us. 1963 if (mOutput && mOutput->audioHwDev && 1964 mOutput->audioHwDev->canSetMasterVolume()) { 1965 mMasterVolume = 1.0; 1966 } else { 1967 mMasterVolume = value; 1968 } 1969} 1970 1971void AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1972{ 1973 Mutex::Autolock _l(mLock); 1974 // Don't apply master mute in SW if our HAL can do it for us. 1975 if (mOutput && mOutput->audioHwDev && 1976 mOutput->audioHwDev->canSetMasterMute()) { 1977 mMasterMute = false; 1978 } else { 1979 mMasterMute = muted; 1980 } 1981} 1982 1983void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) 1984{ 1985 Mutex::Autolock _l(mLock); 1986 mStreamTypes[stream].volume = value; 1987 broadcast_l(); 1988} 1989 1990void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) 1991{ 1992 Mutex::Autolock _l(mLock); 1993 mStreamTypes[stream].mute = muted; 1994 broadcast_l(); 1995} 1996 1997float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const 1998{ 1999 Mutex::Autolock _l(mLock); 2000 return mStreamTypes[stream].volume; 2001} 2002 2003// addTrack_l() must be called with ThreadBase::mLock held 2004status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 2005{ 2006 status_t status = ALREADY_EXISTS; 2007 2008 // set retry count for buffer fill 2009 track->mRetryCount = kMaxTrackStartupRetries; 2010 if (mActiveTracks.indexOf(track) < 0) { 2011 // the track is newly added, make sure it fills up all its 2012 // buffers before playing. This is to ensure the client will 2013 // effectively get the latency it requested. 2014 if (track->isExternalTrack()) { 2015 TrackBase::track_state state = track->mState; 2016 mLock.unlock(); 2017 status = AudioSystem::startOutput(mId, track->streamType(), 2018 (audio_session_t)track->sessionId()); 2019 mLock.lock(); 2020 // abort track was stopped/paused while we released the lock 2021 if (state != track->mState) { 2022 if (status == NO_ERROR) { 2023 mLock.unlock(); 2024 AudioSystem::stopOutput(mId, track->streamType(), 2025 (audio_session_t)track->sessionId()); 2026 mLock.lock(); 2027 } 2028 return INVALID_OPERATION; 2029 } 2030 // abort if start is rejected by audio policy manager 2031 if (status != NO_ERROR) { 2032 return PERMISSION_DENIED; 2033 } 2034#ifdef ADD_BATTERY_DATA 2035 // to track the speaker usage 2036 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 2037#endif 2038 } 2039 2040 track->mFillingUpStatus = track->sharedBuffer() != 0 ? Track::FS_FILLED : Track::FS_FILLING; 2041 track->mResetDone = false; 2042 track->mPresentationCompleteFrames = 0; 2043 mActiveTracks.add(track); 2044 mWakeLockUids.add(track->uid()); 2045 mActiveTracksGeneration++; 2046 mLatestActiveTrack = track; 2047 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 2048 if (chain != 0) { 2049 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), 2050 track->sessionId()); 2051 chain->incActiveTrackCnt(); 2052 } 2053 2054 status = NO_ERROR; 2055 } 2056 2057 onAddNewTrack_l(); 2058 return status; 2059} 2060 2061bool AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 2062{ 2063 track->terminate(); 2064 // active tracks are removed by threadLoop() 2065 bool trackActive = (mActiveTracks.indexOf(track) >= 0); 2066 track->mState = TrackBase::STOPPED; 2067 if (!trackActive) { 2068 removeTrack_l(track); 2069 } else if (track->isFastTrack() || track->isOffloaded() || track->isDirect()) { 2070 track->mState = TrackBase::STOPPING_1; 2071 } 2072 2073 return trackActive; 2074} 2075 2076void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 2077{ 2078 track->triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 2079 mTracks.remove(track); 2080 deleteTrackName_l(track->name()); 2081 // redundant as track is about to be destroyed, for dumpsys only 2082 track->mName = -1; 2083 if (track->isFastTrack()) { 2084 int index = track->mFastIndex; 2085 ALOG_ASSERT(0 < index && index < (int)FastMixerState::kMaxFastTracks); 2086 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << index))); 2087 mFastTrackAvailMask |= 1 << index; 2088 // redundant as track is about to be destroyed, for dumpsys only 2089 track->mFastIndex = -1; 2090 } 2091 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 2092 if (chain != 0) { 2093 chain->decTrackCnt(); 2094 } 2095} 2096 2097void AudioFlinger::PlaybackThread::broadcast_l() 2098{ 2099 // Thread could be blocked waiting for async 2100 // so signal it to handle state changes immediately 2101 // If threadLoop is currently unlocked a signal of mWaitWorkCV will 2102 // be lost so we also flag to prevent it blocking on mWaitWorkCV 2103 mSignalPending = true; 2104 mWaitWorkCV.broadcast(); 2105} 2106 2107String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 2108{ 2109 Mutex::Autolock _l(mLock); 2110 if (initCheck() != NO_ERROR) { 2111 return String8(); 2112 } 2113 2114 char *s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 2115 const String8 out_s8(s); 2116 free(s); 2117 return out_s8; 2118} 2119 2120void AudioFlinger::PlaybackThread::ioConfigChanged(audio_io_config_event event, pid_t pid) { 2121 sp<AudioIoDescriptor> desc = new AudioIoDescriptor(); 2122 ALOGV("PlaybackThread::ioConfigChanged, thread %p, event %d", this, event); 2123 2124 desc->mIoHandle = mId; 2125 2126 switch (event) { 2127 case AUDIO_OUTPUT_OPENED: 2128 case AUDIO_OUTPUT_CONFIG_CHANGED: 2129 desc->mPatch = mPatch; 2130 desc->mChannelMask = mChannelMask; 2131 desc->mSamplingRate = mSampleRate; 2132 desc->mFormat = mFormat; 2133 desc->mFrameCount = mNormalFrameCount; // FIXME see 2134 // AudioFlinger::frameCount(audio_io_handle_t) 2135 desc->mLatency = latency_l(); 2136 break; 2137 2138 case AUDIO_OUTPUT_CLOSED: 2139 default: 2140 break; 2141 } 2142 mAudioFlinger->ioConfigChanged(event, desc, pid); 2143} 2144 2145void AudioFlinger::PlaybackThread::writeCallback() 2146{ 2147 ALOG_ASSERT(mCallbackThread != 0); 2148 mCallbackThread->resetWriteBlocked(); 2149} 2150 2151void AudioFlinger::PlaybackThread::drainCallback() 2152{ 2153 ALOG_ASSERT(mCallbackThread != 0); 2154 mCallbackThread->resetDraining(); 2155} 2156 2157void AudioFlinger::PlaybackThread::resetWriteBlocked(uint32_t sequence) 2158{ 2159 Mutex::Autolock _l(mLock); 2160 // reject out of sequence requests 2161 if ((mWriteAckSequence & 1) && (sequence == mWriteAckSequence)) { 2162 mWriteAckSequence &= ~1; 2163 mWaitWorkCV.signal(); 2164 } 2165} 2166 2167void AudioFlinger::PlaybackThread::resetDraining(uint32_t sequence) 2168{ 2169 Mutex::Autolock _l(mLock); 2170 // reject out of sequence requests 2171 if ((mDrainSequence & 1) && (sequence == mDrainSequence)) { 2172 mDrainSequence &= ~1; 2173 mWaitWorkCV.signal(); 2174 } 2175} 2176 2177// static 2178int AudioFlinger::PlaybackThread::asyncCallback(stream_callback_event_t event, 2179 void *param __unused, 2180 void *cookie) 2181{ 2182 AudioFlinger::PlaybackThread *me = (AudioFlinger::PlaybackThread *)cookie; 2183 ALOGV("asyncCallback() event %d", event); 2184 switch (event) { 2185 case STREAM_CBK_EVENT_WRITE_READY: 2186 me->writeCallback(); 2187 break; 2188 case STREAM_CBK_EVENT_DRAIN_READY: 2189 me->drainCallback(); 2190 break; 2191 default: 2192 ALOGW("asyncCallback() unknown event %d", event); 2193 break; 2194 } 2195 return 0; 2196} 2197 2198void AudioFlinger::PlaybackThread::readOutputParameters_l() 2199{ 2200 // unfortunately we have no way of recovering from errors here, hence the LOG_ALWAYS_FATAL 2201 mSampleRate = mOutput->getSampleRate(); 2202 mChannelMask = mOutput->getChannelMask(); 2203 if (!audio_is_output_channel(mChannelMask)) { 2204 LOG_ALWAYS_FATAL("HAL channel mask %#x not valid for output", mChannelMask); 2205 } 2206 if ((mType == MIXER || mType == DUPLICATING) 2207 && !isValidPcmSinkChannelMask(mChannelMask)) { 2208 LOG_ALWAYS_FATAL("HAL channel mask %#x not supported for mixed output", 2209 mChannelMask); 2210 } 2211 mChannelCount = audio_channel_count_from_out_mask(mChannelMask); 2212 2213 // Get actual HAL format. 2214 mHALFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 2215 // Get format from the shim, which will be different than the HAL format 2216 // if playing compressed audio over HDMI passthrough. 2217 mFormat = mOutput->getFormat(); 2218 if (!audio_is_valid_format(mFormat)) { 2219 LOG_ALWAYS_FATAL("HAL format %#x not valid for output", mFormat); 2220 } 2221 if ((mType == MIXER || mType == DUPLICATING) 2222 && !isValidPcmSinkFormat(mFormat)) { 2223 LOG_FATAL("HAL format %#x not supported for mixed output", 2224 mFormat); 2225 } 2226 mFrameSize = mOutput->getFrameSize(); 2227 mBufferSize = mOutput->stream->common.get_buffer_size(&mOutput->stream->common); 2228 mFrameCount = mBufferSize / mFrameSize; 2229 if (mFrameCount & 15) { 2230 ALOGW("HAL output buffer size is %u frames but AudioMixer requires multiples of 16 frames", 2231 mFrameCount); 2232 } 2233 2234 if ((mOutput->flags & AUDIO_OUTPUT_FLAG_NON_BLOCKING) && 2235 (mOutput->stream->set_callback != NULL)) { 2236 if (mOutput->stream->set_callback(mOutput->stream, 2237 AudioFlinger::PlaybackThread::asyncCallback, this) == 0) { 2238 mUseAsyncWrite = true; 2239 mCallbackThread = new AudioFlinger::AsyncCallbackThread(this); 2240 } 2241 } 2242 2243 mHwSupportsPause = false; 2244 if (mOutput->flags & AUDIO_OUTPUT_FLAG_DIRECT) { 2245 if (mOutput->stream->pause != NULL) { 2246 if (mOutput->stream->resume != NULL) { 2247 mHwSupportsPause = true; 2248 } else { 2249 ALOGW("direct output implements pause but not resume"); 2250 } 2251 } else if (mOutput->stream->resume != NULL) { 2252 ALOGW("direct output implements resume but not pause"); 2253 } 2254 } 2255 if (!mHwSupportsPause && mOutput->flags & AUDIO_OUTPUT_FLAG_HW_AV_SYNC) { 2256 LOG_ALWAYS_FATAL("HW_AV_SYNC requested but HAL does not implement pause and resume"); 2257 } 2258 2259 if (mType == DUPLICATING && mMixerBufferEnabled && mEffectBufferEnabled) { 2260 // For best precision, we use float instead of the associated output 2261 // device format (typically PCM 16 bit). 2262 2263 mFormat = AUDIO_FORMAT_PCM_FLOAT; 2264 mFrameSize = mChannelCount * audio_bytes_per_sample(mFormat); 2265 mBufferSize = mFrameSize * mFrameCount; 2266 2267 // TODO: We currently use the associated output device channel mask and sample rate. 2268 // (1) Perhaps use the ORed channel mask of all downstream MixerThreads 2269 // (if a valid mask) to avoid premature downmix. 2270 // (2) Perhaps use the maximum sample rate of all downstream MixerThreads 2271 // instead of the output device sample rate to avoid loss of high frequency information. 2272 // This may need to be updated as MixerThread/OutputTracks are added and not here. 2273 } 2274 2275 // Calculate size of normal sink buffer relative to the HAL output buffer size 2276 double multiplier = 1.0; 2277 if (mType == MIXER && (kUseFastMixer == FastMixer_Static || 2278 kUseFastMixer == FastMixer_Dynamic)) { 2279 size_t minNormalFrameCount = (kMinNormalSinkBufferSizeMs * mSampleRate) / 1000; 2280 size_t maxNormalFrameCount = (kMaxNormalSinkBufferSizeMs * mSampleRate) / 1000; 2281 // round up minimum and round down maximum to nearest 16 frames to satisfy AudioMixer 2282 minNormalFrameCount = (minNormalFrameCount + 15) & ~15; 2283 maxNormalFrameCount = maxNormalFrameCount & ~15; 2284 if (maxNormalFrameCount < minNormalFrameCount) { 2285 maxNormalFrameCount = minNormalFrameCount; 2286 } 2287 multiplier = (double) minNormalFrameCount / (double) mFrameCount; 2288 if (multiplier <= 1.0) { 2289 multiplier = 1.0; 2290 } else if (multiplier <= 2.0) { 2291 if (2 * mFrameCount <= maxNormalFrameCount) { 2292 multiplier = 2.0; 2293 } else { 2294 multiplier = (double) maxNormalFrameCount / (double) mFrameCount; 2295 } 2296 } else { 2297 // prefer an even multiplier, for compatibility with doubling of fast tracks due to HAL 2298 // SRC (it would be unusual for the normal sink buffer size to not be a multiple of fast 2299 // track, but we sometimes have to do this to satisfy the maximum frame count 2300 // constraint) 2301 // FIXME this rounding up should not be done if no HAL SRC 2302 uint32_t truncMult = (uint32_t) multiplier; 2303 if ((truncMult & 1)) { 2304 if ((truncMult + 1) * mFrameCount <= maxNormalFrameCount) { 2305 ++truncMult; 2306 } 2307 } 2308 multiplier = (double) truncMult; 2309 } 2310 } 2311 mNormalFrameCount = multiplier * mFrameCount; 2312 // round up to nearest 16 frames to satisfy AudioMixer 2313 if (mType == MIXER || mType == DUPLICATING) { 2314 mNormalFrameCount = (mNormalFrameCount + 15) & ~15; 2315 } 2316 ALOGI("HAL output buffer size %u frames, normal sink buffer size %u frames", mFrameCount, 2317 mNormalFrameCount); 2318 2319 // Check if we want to throttle the processing to no more than 2x normal rate 2320 mThreadThrottle = property_get_bool("af.thread.throttle", true /* default_value */); 2321 mThreadThrottleTimeMs = 0; 2322 mThreadThrottleEndMs = 0; 2323 mHalfBufferMs = mNormalFrameCount * 1000 / (2 * mSampleRate); 2324 2325 // mSinkBuffer is the sink buffer. Size is always multiple-of-16 frames. 2326 // Originally this was int16_t[] array, need to remove legacy implications. 2327 free(mSinkBuffer); 2328 mSinkBuffer = NULL; 2329 // For sink buffer size, we use the frame size from the downstream sink to avoid problems 2330 // with non PCM formats for compressed music, e.g. AAC, and Offload threads. 2331 const size_t sinkBufferSize = mNormalFrameCount * mFrameSize; 2332 (void)posix_memalign(&mSinkBuffer, 32, sinkBufferSize); 2333 2334 // We resize the mMixerBuffer according to the requirements of the sink buffer which 2335 // drives the output. 2336 free(mMixerBuffer); 2337 mMixerBuffer = NULL; 2338 if (mMixerBufferEnabled) { 2339 mMixerBufferFormat = AUDIO_FORMAT_PCM_FLOAT; // also valid: AUDIO_FORMAT_PCM_16_BIT. 2340 mMixerBufferSize = mNormalFrameCount * mChannelCount 2341 * audio_bytes_per_sample(mMixerBufferFormat); 2342 (void)posix_memalign(&mMixerBuffer, 32, mMixerBufferSize); 2343 } 2344 free(mEffectBuffer); 2345 mEffectBuffer = NULL; 2346 if (mEffectBufferEnabled) { 2347 mEffectBufferFormat = AUDIO_FORMAT_PCM_16_BIT; // Note: Effects support 16b only 2348 mEffectBufferSize = mNormalFrameCount * mChannelCount 2349 * audio_bytes_per_sample(mEffectBufferFormat); 2350 (void)posix_memalign(&mEffectBuffer, 32, mEffectBufferSize); 2351 } 2352 2353 // force reconfiguration of effect chains and engines to take new buffer size and audio 2354 // parameters into account 2355 // Note that mLock is not held when readOutputParameters_l() is called from the constructor 2356 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 2357 // matter. 2358 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 2359 Vector< sp<EffectChain> > effectChains = mEffectChains; 2360 for (size_t i = 0; i < effectChains.size(); i ++) { 2361 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 2362 } 2363} 2364 2365 2366status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames) 2367{ 2368 if (halFrames == NULL || dspFrames == NULL) { 2369 return BAD_VALUE; 2370 } 2371 Mutex::Autolock _l(mLock); 2372 if (initCheck() != NO_ERROR) { 2373 return INVALID_OPERATION; 2374 } 2375 size_t framesWritten = mBytesWritten / mFrameSize; 2376 *halFrames = framesWritten; 2377 2378 if (isSuspended()) { 2379 // return an estimation of rendered frames when the output is suspended 2380 size_t latencyFrames = (latency_l() * mSampleRate) / 1000; 2381 *dspFrames = framesWritten >= latencyFrames ? framesWritten - latencyFrames : 0; 2382 return NO_ERROR; 2383 } else { 2384 status_t status; 2385 uint32_t frames; 2386 status = mOutput->getRenderPosition(&frames); 2387 *dspFrames = (size_t)frames; 2388 return status; 2389 } 2390} 2391 2392uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) const 2393{ 2394 Mutex::Autolock _l(mLock); 2395 uint32_t result = 0; 2396 if (getEffectChain_l(sessionId) != 0) { 2397 result = EFFECT_SESSION; 2398 } 2399 2400 for (size_t i = 0; i < mTracks.size(); ++i) { 2401 sp<Track> track = mTracks[i]; 2402 if (sessionId == track->sessionId() && !track->isInvalid()) { 2403 result |= TRACK_SESSION; 2404 break; 2405 } 2406 } 2407 2408 return result; 2409} 2410 2411uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 2412{ 2413 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 2414 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 2415 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 2416 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 2417 } 2418 for (size_t i = 0; i < mTracks.size(); i++) { 2419 sp<Track> track = mTracks[i]; 2420 if (sessionId == track->sessionId() && !track->isInvalid()) { 2421 return AudioSystem::getStrategyForStream(track->streamType()); 2422 } 2423 } 2424 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 2425} 2426 2427 2428AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 2429{ 2430 Mutex::Autolock _l(mLock); 2431 return mOutput; 2432} 2433 2434AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 2435{ 2436 Mutex::Autolock _l(mLock); 2437 AudioStreamOut *output = mOutput; 2438 mOutput = NULL; 2439 // FIXME FastMixer might also have a raw ptr to mOutputSink; 2440 // must push a NULL and wait for ack 2441 mOutputSink.clear(); 2442 mPipeSink.clear(); 2443 mNormalSink.clear(); 2444 return output; 2445} 2446 2447// this method must always be called either with ThreadBase mLock held or inside the thread loop 2448audio_stream_t* AudioFlinger::PlaybackThread::stream() const 2449{ 2450 if (mOutput == NULL) { 2451 return NULL; 2452 } 2453 return &mOutput->stream->common; 2454} 2455 2456uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() const 2457{ 2458 return (uint32_t)((uint32_t)((mNormalFrameCount * 1000) / mSampleRate) * 1000); 2459} 2460 2461status_t AudioFlinger::PlaybackThread::setSyncEvent(const sp<SyncEvent>& event) 2462{ 2463 if (!isValidSyncEvent(event)) { 2464 return BAD_VALUE; 2465 } 2466 2467 Mutex::Autolock _l(mLock); 2468 2469 for (size_t i = 0; i < mTracks.size(); ++i) { 2470 sp<Track> track = mTracks[i]; 2471 if (event->triggerSession() == track->sessionId()) { 2472 (void) track->setSyncEvent(event); 2473 return NO_ERROR; 2474 } 2475 } 2476 2477 return NAME_NOT_FOUND; 2478} 2479 2480bool AudioFlinger::PlaybackThread::isValidSyncEvent(const sp<SyncEvent>& event) const 2481{ 2482 return event->type() == AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE; 2483} 2484 2485void AudioFlinger::PlaybackThread::threadLoop_removeTracks( 2486 const Vector< sp<Track> >& tracksToRemove) 2487{ 2488 size_t count = tracksToRemove.size(); 2489 if (count > 0) { 2490 for (size_t i = 0 ; i < count ; i++) { 2491 const sp<Track>& track = tracksToRemove.itemAt(i); 2492 if (track->isExternalTrack()) { 2493 AudioSystem::stopOutput(mId, track->streamType(), 2494 (audio_session_t)track->sessionId()); 2495#ifdef ADD_BATTERY_DATA 2496 // to track the speaker usage 2497 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 2498#endif 2499 if (track->isTerminated()) { 2500 AudioSystem::releaseOutput(mId, track->streamType(), 2501 (audio_session_t)track->sessionId()); 2502 } 2503 } 2504 } 2505 } 2506} 2507 2508void AudioFlinger::PlaybackThread::checkSilentMode_l() 2509{ 2510 if (!mMasterMute) { 2511 char value[PROPERTY_VALUE_MAX]; 2512 if (property_get("ro.audio.silent", value, "0") > 0) { 2513 char *endptr; 2514 unsigned long ul = strtoul(value, &endptr, 0); 2515 if (*endptr == '\0' && ul != 0) { 2516 ALOGD("Silence is golden"); 2517 // The setprop command will not allow a property to be changed after 2518 // the first time it is set, so we don't have to worry about un-muting. 2519 setMasterMute_l(true); 2520 } 2521 } 2522 } 2523} 2524 2525// shared by MIXER and DIRECT, overridden by DUPLICATING 2526ssize_t AudioFlinger::PlaybackThread::threadLoop_write() 2527{ 2528 // FIXME rewrite to reduce number of system calls 2529 mLastWriteTime = systemTime(); 2530 mInWrite = true; 2531 ssize_t bytesWritten; 2532 const size_t offset = mCurrentWriteLength - mBytesRemaining; 2533 2534 // If an NBAIO sink is present, use it to write the normal mixer's submix 2535 if (mNormalSink != 0) { 2536 2537 const size_t count = mBytesRemaining / mFrameSize; 2538 2539 ATRACE_BEGIN("write"); 2540 // update the setpoint when AudioFlinger::mScreenState changes 2541 uint32_t screenState = AudioFlinger::mScreenState; 2542 if (screenState != mScreenState) { 2543 mScreenState = screenState; 2544 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 2545 if (pipe != NULL) { 2546 pipe->setAvgFrames((mScreenState & 1) ? 2547 (pipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2548 } 2549 } 2550 ssize_t framesWritten = mNormalSink->write((char *)mSinkBuffer + offset, count); 2551 ATRACE_END(); 2552 if (framesWritten > 0) { 2553 bytesWritten = framesWritten * mFrameSize; 2554 } else { 2555 bytesWritten = framesWritten; 2556 } 2557 mLatchDValid = false; 2558 status_t status = mNormalSink->getTimestamp(mLatchD.mTimestamp); 2559 if (status == NO_ERROR) { 2560 size_t totalFramesWritten = mNormalSink->framesWritten(); 2561 if (totalFramesWritten >= mLatchD.mTimestamp.mPosition) { 2562 mLatchD.mUnpresentedFrames = totalFramesWritten - mLatchD.mTimestamp.mPosition; 2563 // mLatchD.mFramesReleased is set immediately before D is clocked into Q 2564 mLatchDValid = true; 2565 } 2566 } 2567 // otherwise use the HAL / AudioStreamOut directly 2568 } else { 2569 // Direct output and offload threads 2570 2571 if (mUseAsyncWrite) { 2572 ALOGW_IF(mWriteAckSequence & 1, "threadLoop_write(): out of sequence write request"); 2573 mWriteAckSequence += 2; 2574 mWriteAckSequence |= 1; 2575 ALOG_ASSERT(mCallbackThread != 0); 2576 mCallbackThread->setWriteBlocked(mWriteAckSequence); 2577 } 2578 // FIXME We should have an implementation of timestamps for direct output threads. 2579 // They are used e.g for multichannel PCM playback over HDMI. 2580 bytesWritten = mOutput->write((char *)mSinkBuffer + offset, mBytesRemaining); 2581 if (mUseAsyncWrite && 2582 ((bytesWritten < 0) || (bytesWritten == (ssize_t)mBytesRemaining))) { 2583 // do not wait for async callback in case of error of full write 2584 mWriteAckSequence &= ~1; 2585 ALOG_ASSERT(mCallbackThread != 0); 2586 mCallbackThread->setWriteBlocked(mWriteAckSequence); 2587 } 2588 } 2589 2590 mNumWrites++; 2591 mInWrite = false; 2592 mStandby = false; 2593 return bytesWritten; 2594} 2595 2596void AudioFlinger::PlaybackThread::threadLoop_drain() 2597{ 2598 if (mOutput->stream->drain) { 2599 ALOGV("draining %s", (mMixerStatus == MIXER_DRAIN_TRACK) ? "early" : "full"); 2600 if (mUseAsyncWrite) { 2601 ALOGW_IF(mDrainSequence & 1, "threadLoop_drain(): out of sequence drain request"); 2602 mDrainSequence |= 1; 2603 ALOG_ASSERT(mCallbackThread != 0); 2604 mCallbackThread->setDraining(mDrainSequence); 2605 } 2606 mOutput->stream->drain(mOutput->stream, 2607 (mMixerStatus == MIXER_DRAIN_TRACK) ? AUDIO_DRAIN_EARLY_NOTIFY 2608 : AUDIO_DRAIN_ALL); 2609 } 2610} 2611 2612void AudioFlinger::PlaybackThread::threadLoop_exit() 2613{ 2614 { 2615 Mutex::Autolock _l(mLock); 2616 for (size_t i = 0; i < mTracks.size(); i++) { 2617 sp<Track> track = mTracks[i]; 2618 track->invalidate(); 2619 } 2620 } 2621} 2622 2623/* 2624The derived values that are cached: 2625 - mSinkBufferSize from frame count * frame size 2626 - mActiveSleepTimeUs from activeSleepTimeUs() 2627 - mIdleSleepTimeUs from idleSleepTimeUs() 2628 - mStandbyDelayNs from mActiveSleepTimeUs (DIRECT only) or forced to at least 2629 kDefaultStandbyTimeInNsecs when connected to an A2DP device. 2630 - maxPeriod from frame count and sample rate (MIXER only) 2631 2632The parameters that affect these derived values are: 2633 - frame count 2634 - frame size 2635 - sample rate 2636 - device type: A2DP or not 2637 - device latency 2638 - format: PCM or not 2639 - active sleep time 2640 - idle sleep time 2641*/ 2642 2643void AudioFlinger::PlaybackThread::cacheParameters_l() 2644{ 2645 mSinkBufferSize = mNormalFrameCount * mFrameSize; 2646 mActiveSleepTimeUs = activeSleepTimeUs(); 2647 mIdleSleepTimeUs = idleSleepTimeUs(); 2648 2649 // make sure standby delay is not too short when connected to an A2DP sink to avoid 2650 // truncating audio when going to standby. 2651 mStandbyDelayNs = AudioFlinger::mStandbyTimeInNsecs; 2652 if ((mOutDevice & AUDIO_DEVICE_OUT_ALL_A2DP) != 0) { 2653 if (mStandbyDelayNs < kDefaultStandbyTimeInNsecs) { 2654 mStandbyDelayNs = kDefaultStandbyTimeInNsecs; 2655 } 2656 } 2657} 2658 2659void AudioFlinger::PlaybackThread::invalidateTracks(audio_stream_type_t streamType) 2660{ 2661 ALOGV("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 2662 this, streamType, mTracks.size()); 2663 Mutex::Autolock _l(mLock); 2664 2665 size_t size = mTracks.size(); 2666 for (size_t i = 0; i < size; i++) { 2667 sp<Track> t = mTracks[i]; 2668 if (t->streamType() == streamType && t->isExternalTrack()) { 2669 t->invalidate(); 2670 } 2671 } 2672} 2673 2674status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 2675{ 2676 int session = chain->sessionId(); 2677 int16_t* buffer = reinterpret_cast<int16_t*>(mEffectBufferEnabled 2678 ? mEffectBuffer : mSinkBuffer); 2679 bool ownsBuffer = false; 2680 2681 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 2682 if (session > 0) { 2683 // Only one effect chain can be present in direct output thread and it uses 2684 // the sink buffer as input 2685 if (mType != DIRECT) { 2686 size_t numSamples = mNormalFrameCount * mChannelCount; 2687 buffer = new int16_t[numSamples]; 2688 memset(buffer, 0, numSamples * sizeof(int16_t)); 2689 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 2690 ownsBuffer = true; 2691 } 2692 2693 // Attach all tracks with same session ID to this chain. 2694 for (size_t i = 0; i < mTracks.size(); ++i) { 2695 sp<Track> track = mTracks[i]; 2696 if (session == track->sessionId()) { 2697 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), 2698 buffer); 2699 track->setMainBuffer(buffer); 2700 chain->incTrackCnt(); 2701 } 2702 } 2703 2704 // indicate all active tracks in the chain 2705 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 2706 sp<Track> track = mActiveTracks[i].promote(); 2707 if (track == 0) { 2708 continue; 2709 } 2710 if (session == track->sessionId()) { 2711 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 2712 chain->incActiveTrackCnt(); 2713 } 2714 } 2715 } 2716 chain->setThread(this); 2717 chain->setInBuffer(buffer, ownsBuffer); 2718 chain->setOutBuffer(reinterpret_cast<int16_t*>(mEffectBufferEnabled 2719 ? mEffectBuffer : mSinkBuffer)); 2720 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 2721 // chains list in order to be processed last as it contains output stage effects 2722 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 2723 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 2724 // after track specific effects and before output stage 2725 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 2726 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 2727 // Effect chain for other sessions are inserted at beginning of effect 2728 // chains list to be processed before output mix effects. Relative order between other 2729 // sessions is not important 2730 size_t size = mEffectChains.size(); 2731 size_t i = 0; 2732 for (i = 0; i < size; i++) { 2733 if (mEffectChains[i]->sessionId() < session) { 2734 break; 2735 } 2736 } 2737 mEffectChains.insertAt(chain, i); 2738 checkSuspendOnAddEffectChain_l(chain); 2739 2740 return NO_ERROR; 2741} 2742 2743size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 2744{ 2745 int session = chain->sessionId(); 2746 2747 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 2748 2749 for (size_t i = 0; i < mEffectChains.size(); i++) { 2750 if (chain == mEffectChains[i]) { 2751 mEffectChains.removeAt(i); 2752 // detach all active tracks from the chain 2753 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 2754 sp<Track> track = mActiveTracks[i].promote(); 2755 if (track == 0) { 2756 continue; 2757 } 2758 if (session == track->sessionId()) { 2759 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 2760 chain.get(), session); 2761 chain->decActiveTrackCnt(); 2762 } 2763 } 2764 2765 // detach all tracks with same session ID from this chain 2766 for (size_t i = 0; i < mTracks.size(); ++i) { 2767 sp<Track> track = mTracks[i]; 2768 if (session == track->sessionId()) { 2769 track->setMainBuffer(reinterpret_cast<int16_t*>(mSinkBuffer)); 2770 chain->decTrackCnt(); 2771 } 2772 } 2773 break; 2774 } 2775 } 2776 return mEffectChains.size(); 2777} 2778 2779status_t AudioFlinger::PlaybackThread::attachAuxEffect( 2780 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2781{ 2782 Mutex::Autolock _l(mLock); 2783 return attachAuxEffect_l(track, EffectId); 2784} 2785 2786status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 2787 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2788{ 2789 status_t status = NO_ERROR; 2790 2791 if (EffectId == 0) { 2792 track->setAuxBuffer(0, NULL); 2793 } else { 2794 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 2795 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 2796 if (effect != 0) { 2797 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 2798 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 2799 } else { 2800 status = INVALID_OPERATION; 2801 } 2802 } else { 2803 status = BAD_VALUE; 2804 } 2805 } 2806 return status; 2807} 2808 2809void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 2810{ 2811 for (size_t i = 0; i < mTracks.size(); ++i) { 2812 sp<Track> track = mTracks[i]; 2813 if (track->auxEffectId() == effectId) { 2814 attachAuxEffect_l(track, 0); 2815 } 2816 } 2817} 2818 2819bool AudioFlinger::PlaybackThread::threadLoop() 2820{ 2821 Vector< sp<Track> > tracksToRemove; 2822 2823 mStandbyTimeNs = systemTime(); 2824 2825 // MIXER 2826 nsecs_t lastWarning = 0; 2827 2828 // DUPLICATING 2829 // FIXME could this be made local to while loop? 2830 writeFrames = 0; 2831 2832 int lastGeneration = 0; 2833 2834 cacheParameters_l(); 2835 mSleepTimeUs = mIdleSleepTimeUs; 2836 2837 if (mType == MIXER) { 2838 sleepTimeShift = 0; 2839 } 2840 2841 CpuStats cpuStats; 2842 const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid())); 2843 2844 acquireWakeLock(); 2845 2846 // mNBLogWriter->log can only be called while thread mutex mLock is held. 2847 // So if you need to log when mutex is unlocked, set logString to a non-NULL string, 2848 // and then that string will be logged at the next convenient opportunity. 2849 const char *logString = NULL; 2850 2851 checkSilentMode_l(); 2852 2853 while (!exitPending()) 2854 { 2855 cpuStats.sample(myName); 2856 2857 Vector< sp<EffectChain> > effectChains; 2858 2859 { // scope for mLock 2860 2861 Mutex::Autolock _l(mLock); 2862 2863 processConfigEvents_l(); 2864 2865 if (logString != NULL) { 2866 mNBLogWriter->logTimestamp(); 2867 mNBLogWriter->log(logString); 2868 logString = NULL; 2869 } 2870 2871 // Gather the framesReleased counters for all active tracks, 2872 // and latch them atomically with the timestamp. 2873 // FIXME We're using raw pointers as indices. A unique track ID would be a better index. 2874 mLatchD.mFramesReleased.clear(); 2875 size_t size = mActiveTracks.size(); 2876 for (size_t i = 0; i < size; i++) { 2877 sp<Track> t = mActiveTracks[i].promote(); 2878 if (t != 0) { 2879 mLatchD.mFramesReleased.add(t.get(), 2880 t->mAudioTrackServerProxy->framesReleased()); 2881 } 2882 } 2883 if (mLatchDValid) { 2884 mLatchQ = mLatchD; 2885 mLatchDValid = false; 2886 mLatchQValid = true; 2887 } 2888 2889 saveOutputTracks(); 2890 if (mSignalPending) { 2891 // A signal was raised while we were unlocked 2892 mSignalPending = false; 2893 } else if (waitingAsyncCallback_l()) { 2894 if (exitPending()) { 2895 break; 2896 } 2897 bool released = false; 2898 // The following works around a bug in the offload driver. Ideally we would release 2899 // the wake lock every time, but that causes the last offload buffer(s) to be 2900 // dropped while the device is on battery, so we need to hold a wake lock during 2901 // the drain phase. 2902 if (mBytesRemaining && !(mDrainSequence & 1)) { 2903 releaseWakeLock_l(); 2904 released = true; 2905 } 2906 mWakeLockUids.clear(); 2907 mActiveTracksGeneration++; 2908 ALOGV("wait async completion"); 2909 mWaitWorkCV.wait(mLock); 2910 ALOGV("async completion/wake"); 2911 if (released) { 2912 acquireWakeLock_l(); 2913 } 2914 mStandbyTimeNs = systemTime() + mStandbyDelayNs; 2915 mSleepTimeUs = 0; 2916 2917 continue; 2918 } 2919 if ((!mActiveTracks.size() && systemTime() > mStandbyTimeNs) || 2920 isSuspended()) { 2921 // put audio hardware into standby after short delay 2922 if (shouldStandby_l()) { 2923 2924 threadLoop_standby(); 2925 2926 mStandby = true; 2927 } 2928 2929 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2930 // we're about to wait, flush the binder command buffer 2931 IPCThreadState::self()->flushCommands(); 2932 2933 clearOutputTracks(); 2934 2935 if (exitPending()) { 2936 break; 2937 } 2938 2939 releaseWakeLock_l(); 2940 mWakeLockUids.clear(); 2941 mActiveTracksGeneration++; 2942 // wait until we have something to do... 2943 ALOGV("%s going to sleep", myName.string()); 2944 mWaitWorkCV.wait(mLock); 2945 ALOGV("%s waking up", myName.string()); 2946 acquireWakeLock_l(); 2947 2948 mMixerStatus = MIXER_IDLE; 2949 mMixerStatusIgnoringFastTracks = MIXER_IDLE; 2950 mBytesWritten = 0; 2951 mBytesRemaining = 0; 2952 checkSilentMode_l(); 2953 2954 mStandbyTimeNs = systemTime() + mStandbyDelayNs; 2955 mSleepTimeUs = mIdleSleepTimeUs; 2956 if (mType == MIXER) { 2957 sleepTimeShift = 0; 2958 } 2959 2960 continue; 2961 } 2962 } 2963 // mMixerStatusIgnoringFastTracks is also updated internally 2964 mMixerStatus = prepareTracks_l(&tracksToRemove); 2965 2966 // compare with previously applied list 2967 if (lastGeneration != mActiveTracksGeneration) { 2968 // update wakelock 2969 updateWakeLockUids_l(mWakeLockUids); 2970 lastGeneration = mActiveTracksGeneration; 2971 } 2972 2973 // prevent any changes in effect chain list and in each effect chain 2974 // during mixing and effect process as the audio buffers could be deleted 2975 // or modified if an effect is created or deleted 2976 lockEffectChains_l(effectChains); 2977 } // mLock scope ends 2978 2979 if (mBytesRemaining == 0) { 2980 mCurrentWriteLength = 0; 2981 if (mMixerStatus == MIXER_TRACKS_READY) { 2982 // threadLoop_mix() sets mCurrentWriteLength 2983 threadLoop_mix(); 2984 } else if ((mMixerStatus != MIXER_DRAIN_TRACK) 2985 && (mMixerStatus != MIXER_DRAIN_ALL)) { 2986 // threadLoop_sleepTime sets mSleepTimeUs to 0 if data 2987 // must be written to HAL 2988 threadLoop_sleepTime(); 2989 if (mSleepTimeUs == 0) { 2990 mCurrentWriteLength = mSinkBufferSize; 2991 } 2992 } 2993 // Either threadLoop_mix() or threadLoop_sleepTime() should have set 2994 // mMixerBuffer with data if mMixerBufferValid is true and mSleepTimeUs == 0. 2995 // Merge mMixerBuffer data into mEffectBuffer (if any effects are valid) 2996 // or mSinkBuffer (if there are no effects). 2997 // 2998 // This is done pre-effects computation; if effects change to 2999 // support higher precision, this needs to move. 3000 // 3001 // mMixerBufferValid is only set true by MixerThread::prepareTracks_l(). 3002 // TODO use mSleepTimeUs == 0 as an additional condition. 3003 if (mMixerBufferValid) { 3004 void *buffer = mEffectBufferValid ? mEffectBuffer : mSinkBuffer; 3005 audio_format_t format = mEffectBufferValid ? mEffectBufferFormat : mFormat; 3006 3007 // mono blend occurs for mixer threads only (not direct or offloaded) 3008 // and is handled here if we're going directly to the sink. 3009 if (requireMonoBlend() && !mEffectBufferValid) { 3010 mono_blend(mMixerBuffer, mMixerBufferFormat, mChannelCount, mNormalFrameCount, 3011 true /*limit*/); 3012 } 3013 3014 memcpy_by_audio_format(buffer, format, mMixerBuffer, mMixerBufferFormat, 3015 mNormalFrameCount * mChannelCount); 3016 } 3017 3018 mBytesRemaining = mCurrentWriteLength; 3019 if (isSuspended()) { 3020 mSleepTimeUs = suspendSleepTimeUs(); 3021 // simulate write to HAL when suspended 3022 mBytesWritten += mSinkBufferSize; 3023 mBytesRemaining = 0; 3024 } 3025 3026 // only process effects if we're going to write 3027 if (mSleepTimeUs == 0 && mType != OFFLOAD) { 3028 for (size_t i = 0; i < effectChains.size(); i ++) { 3029 effectChains[i]->process_l(); 3030 } 3031 } 3032 } 3033 // Process effect chains for offloaded thread even if no audio 3034 // was read from audio track: process only updates effect state 3035 // and thus does have to be synchronized with audio writes but may have 3036 // to be called while waiting for async write callback 3037 if (mType == OFFLOAD) { 3038 for (size_t i = 0; i < effectChains.size(); i ++) { 3039 effectChains[i]->process_l(); 3040 } 3041 } 3042 3043 // Only if the Effects buffer is enabled and there is data in the 3044 // Effects buffer (buffer valid), we need to 3045 // copy into the sink buffer. 3046 // TODO use mSleepTimeUs == 0 as an additional condition. 3047 if (mEffectBufferValid) { 3048 //ALOGV("writing effect buffer to sink buffer format %#x", mFormat); 3049 3050 if (requireMonoBlend()) { 3051 mono_blend(mEffectBuffer, mEffectBufferFormat, mChannelCount, mNormalFrameCount, 3052 true /*limit*/); 3053 } 3054 3055 memcpy_by_audio_format(mSinkBuffer, mFormat, mEffectBuffer, mEffectBufferFormat, 3056 mNormalFrameCount * mChannelCount); 3057 } 3058 3059 // enable changes in effect chain 3060 unlockEffectChains(effectChains); 3061 3062 if (!waitingAsyncCallback()) { 3063 // mSleepTimeUs == 0 means we must write to audio hardware 3064 if (mSleepTimeUs == 0) { 3065 ssize_t ret = 0; 3066 if (mBytesRemaining) { 3067 ret = threadLoop_write(); 3068 if (ret < 0) { 3069 mBytesRemaining = 0; 3070 } else { 3071 mBytesWritten += ret; 3072 mBytesRemaining -= ret; 3073 } 3074 } else if ((mMixerStatus == MIXER_DRAIN_TRACK) || 3075 (mMixerStatus == MIXER_DRAIN_ALL)) { 3076 threadLoop_drain(); 3077 } 3078 if (mType == MIXER && !mStandby) { 3079 // write blocked detection 3080 nsecs_t now = systemTime(); 3081 nsecs_t delta = now - mLastWriteTime; 3082 if (delta > maxPeriod) { 3083 mNumDelayedWrites++; 3084 if ((now - lastWarning) > kWarningThrottleNs) { 3085 ATRACE_NAME("underrun"); 3086 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 3087 ns2ms(delta), mNumDelayedWrites, this); 3088 lastWarning = now; 3089 } 3090 } 3091 3092 if (mThreadThrottle 3093 && mMixerStatus == MIXER_TRACKS_READY // we are mixing (active tracks) 3094 && ret > 0) { // we wrote something 3095 // Limit MixerThread data processing to no more than twice the 3096 // expected processing rate. 3097 // 3098 // This helps prevent underruns with NuPlayer and other applications 3099 // which may set up buffers that are close to the minimum size, or use 3100 // deep buffers, and rely on a double-buffering sleep strategy to fill. 3101 // 3102 // The throttle smooths out sudden large data drains from the device, 3103 // e.g. when it comes out of standby, which often causes problems with 3104 // (1) mixer threads without a fast mixer (which has its own warm-up) 3105 // (2) minimum buffer sized tracks (even if the track is full, 3106 // the app won't fill fast enough to handle the sudden draw). 3107 3108 const int32_t deltaMs = delta / 1000000; 3109 const int32_t throttleMs = mHalfBufferMs - deltaMs; 3110 if ((signed)mHalfBufferMs >= throttleMs && throttleMs > 0) { 3111 usleep(throttleMs * 1000); 3112 // notify of throttle start on verbose log 3113 ALOGV_IF(mThreadThrottleEndMs == mThreadThrottleTimeMs, 3114 "mixer(%p) throttle begin:" 3115 " ret(%zd) deltaMs(%d) requires sleep %d ms", 3116 this, ret, deltaMs, throttleMs); 3117 mThreadThrottleTimeMs += throttleMs; 3118 } else { 3119 uint32_t diff = mThreadThrottleTimeMs - mThreadThrottleEndMs; 3120 if (diff > 0) { 3121 // notify of throttle end on debug log 3122 ALOGD("mixer(%p) throttle end: throttle time(%u)", this, diff); 3123 mThreadThrottleEndMs = mThreadThrottleTimeMs; 3124 } 3125 } 3126 } 3127 } 3128 3129 } else { 3130 ATRACE_BEGIN("sleep"); 3131 usleep(mSleepTimeUs); 3132 ATRACE_END(); 3133 } 3134 } 3135 3136 // Finally let go of removed track(s), without the lock held 3137 // since we can't guarantee the destructors won't acquire that 3138 // same lock. This will also mutate and push a new fast mixer state. 3139 threadLoop_removeTracks(tracksToRemove); 3140 tracksToRemove.clear(); 3141 3142 // FIXME I don't understand the need for this here; 3143 // it was in the original code but maybe the 3144 // assignment in saveOutputTracks() makes this unnecessary? 3145 clearOutputTracks(); 3146 3147 // Effect chains will be actually deleted here if they were removed from 3148 // mEffectChains list during mixing or effects processing 3149 effectChains.clear(); 3150 3151 // FIXME Note that the above .clear() is no longer necessary since effectChains 3152 // is now local to this block, but will keep it for now (at least until merge done). 3153 } 3154 3155 threadLoop_exit(); 3156 3157 if (!mStandby) { 3158 threadLoop_standby(); 3159 mStandby = true; 3160 } 3161 3162 releaseWakeLock(); 3163 mWakeLockUids.clear(); 3164 mActiveTracksGeneration++; 3165 3166 ALOGV("Thread %p type %d exiting", this, mType); 3167 return false; 3168} 3169 3170// removeTracks_l() must be called with ThreadBase::mLock held 3171void AudioFlinger::PlaybackThread::removeTracks_l(const Vector< sp<Track> >& tracksToRemove) 3172{ 3173 size_t count = tracksToRemove.size(); 3174 if (count > 0) { 3175 for (size_t i=0 ; i<count ; i++) { 3176 const sp<Track>& track = tracksToRemove.itemAt(i); 3177 mActiveTracks.remove(track); 3178 mWakeLockUids.remove(track->uid()); 3179 mActiveTracksGeneration++; 3180 ALOGV("removeTracks_l removing track on session %d", track->sessionId()); 3181 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 3182 if (chain != 0) { 3183 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), 3184 track->sessionId()); 3185 chain->decActiveTrackCnt(); 3186 } 3187 if (track->isTerminated()) { 3188 removeTrack_l(track); 3189 } 3190 } 3191 } 3192 3193} 3194 3195status_t AudioFlinger::PlaybackThread::getTimestamp_l(AudioTimestamp& timestamp) 3196{ 3197 if (mNormalSink != 0) { 3198 return mNormalSink->getTimestamp(timestamp); 3199 } 3200 if ((mType == OFFLOAD || mType == DIRECT) 3201 && mOutput != NULL && mOutput->stream->get_presentation_position) { 3202 uint64_t position64; 3203 int ret = mOutput->getPresentationPosition(&position64, ×tamp.mTime); 3204 if (ret == 0) { 3205 timestamp.mPosition = (uint32_t)position64; 3206 return NO_ERROR; 3207 } 3208 } 3209 return INVALID_OPERATION; 3210} 3211 3212status_t AudioFlinger::MixerThread::createAudioPatch_l(const struct audio_patch *patch, 3213 audio_patch_handle_t *handle) 3214{ 3215 // if !&IDLE, holds the FastMixer state to restore after new parameters processed 3216 FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE; 3217 if (mFastMixer != 0) { 3218 FastMixerStateQueue *sq = mFastMixer->sq(); 3219 FastMixerState *state = sq->begin(); 3220 if (!(state->mCommand & FastMixerState::IDLE)) { 3221 previousCommand = state->mCommand; 3222 state->mCommand = FastMixerState::HOT_IDLE; 3223 sq->end(); 3224 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3225 } else { 3226 sq->end(false /*didModify*/); 3227 } 3228 } 3229 status_t status = PlaybackThread::createAudioPatch_l(patch, handle); 3230 3231 if (!(previousCommand & FastMixerState::IDLE)) { 3232 ALOG_ASSERT(mFastMixer != 0); 3233 FastMixerStateQueue *sq = mFastMixer->sq(); 3234 FastMixerState *state = sq->begin(); 3235 ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE); 3236 state->mCommand = previousCommand; 3237 sq->end(); 3238 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3239 } 3240 3241 return status; 3242} 3243 3244status_t AudioFlinger::PlaybackThread::createAudioPatch_l(const struct audio_patch *patch, 3245 audio_patch_handle_t *handle) 3246{ 3247 status_t status = NO_ERROR; 3248 3249 // store new device and send to effects 3250 audio_devices_t type = AUDIO_DEVICE_NONE; 3251 for (unsigned int i = 0; i < patch->num_sinks; i++) { 3252 type |= patch->sinks[i].ext.device.type; 3253 } 3254 3255#ifdef ADD_BATTERY_DATA 3256 // when changing the audio output device, call addBatteryData to notify 3257 // the change 3258 if (mOutDevice != type) { 3259 uint32_t params = 0; 3260 // check whether speaker is on 3261 if (type & AUDIO_DEVICE_OUT_SPEAKER) { 3262 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 3263 } 3264 3265 audio_devices_t deviceWithoutSpeaker 3266 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 3267 // check if any other device (except speaker) is on 3268 if (type & deviceWithoutSpeaker) { 3269 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 3270 } 3271 3272 if (params != 0) { 3273 addBatteryData(params); 3274 } 3275 } 3276#endif 3277 3278 for (size_t i = 0; i < mEffectChains.size(); i++) { 3279 mEffectChains[i]->setDevice_l(type); 3280 } 3281 3282 // mPrevOutDevice is the latest device set by createAudioPatch_l(). It is not set when 3283 // the thread is created so that the first patch creation triggers an ioConfigChanged callback 3284 bool configChanged = mPrevOutDevice != type; 3285 mOutDevice = type; 3286 mPatch = *patch; 3287 3288 if (mOutput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 3289 audio_hw_device_t *hwDevice = mOutput->audioHwDev->hwDevice(); 3290 status = hwDevice->create_audio_patch(hwDevice, 3291 patch->num_sources, 3292 patch->sources, 3293 patch->num_sinks, 3294 patch->sinks, 3295 handle); 3296 } else { 3297 char *address; 3298 if (strcmp(patch->sinks[0].ext.device.address, "") != 0) { 3299 //FIXME: we only support address on first sink with HAL version < 3.0 3300 address = audio_device_address_to_parameter( 3301 patch->sinks[0].ext.device.type, 3302 patch->sinks[0].ext.device.address); 3303 } else { 3304 address = (char *)calloc(1, 1); 3305 } 3306 AudioParameter param = AudioParameter(String8(address)); 3307 free(address); 3308 param.addInt(String8(AUDIO_PARAMETER_STREAM_ROUTING), (int)type); 3309 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3310 param.toString().string()); 3311 *handle = AUDIO_PATCH_HANDLE_NONE; 3312 } 3313 if (configChanged) { 3314 mPrevOutDevice = type; 3315 sendIoConfigEvent_l(AUDIO_OUTPUT_CONFIG_CHANGED); 3316 } 3317 return status; 3318} 3319 3320status_t AudioFlinger::MixerThread::releaseAudioPatch_l(const audio_patch_handle_t handle) 3321{ 3322 // if !&IDLE, holds the FastMixer state to restore after new parameters processed 3323 FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE; 3324 if (mFastMixer != 0) { 3325 FastMixerStateQueue *sq = mFastMixer->sq(); 3326 FastMixerState *state = sq->begin(); 3327 if (!(state->mCommand & FastMixerState::IDLE)) { 3328 previousCommand = state->mCommand; 3329 state->mCommand = FastMixerState::HOT_IDLE; 3330 sq->end(); 3331 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3332 } else { 3333 sq->end(false /*didModify*/); 3334 } 3335 } 3336 3337 status_t status = PlaybackThread::releaseAudioPatch_l(handle); 3338 3339 if (!(previousCommand & FastMixerState::IDLE)) { 3340 ALOG_ASSERT(mFastMixer != 0); 3341 FastMixerStateQueue *sq = mFastMixer->sq(); 3342 FastMixerState *state = sq->begin(); 3343 ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE); 3344 state->mCommand = previousCommand; 3345 sq->end(); 3346 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3347 } 3348 3349 return status; 3350} 3351 3352status_t AudioFlinger::PlaybackThread::releaseAudioPatch_l(const audio_patch_handle_t handle) 3353{ 3354 status_t status = NO_ERROR; 3355 3356 mOutDevice = AUDIO_DEVICE_NONE; 3357 3358 if (mOutput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 3359 audio_hw_device_t *hwDevice = mOutput->audioHwDev->hwDevice(); 3360 status = hwDevice->release_audio_patch(hwDevice, handle); 3361 } else { 3362 AudioParameter param; 3363 param.addInt(String8(AUDIO_PARAMETER_STREAM_ROUTING), 0); 3364 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3365 param.toString().string()); 3366 } 3367 return status; 3368} 3369 3370void AudioFlinger::PlaybackThread::addPatchTrack(const sp<PatchTrack>& track) 3371{ 3372 Mutex::Autolock _l(mLock); 3373 mTracks.add(track); 3374} 3375 3376void AudioFlinger::PlaybackThread::deletePatchTrack(const sp<PatchTrack>& track) 3377{ 3378 Mutex::Autolock _l(mLock); 3379 destroyTrack_l(track); 3380} 3381 3382void AudioFlinger::PlaybackThread::getAudioPortConfig(struct audio_port_config *config) 3383{ 3384 ThreadBase::getAudioPortConfig(config); 3385 config->role = AUDIO_PORT_ROLE_SOURCE; 3386 config->ext.mix.hw_module = mOutput->audioHwDev->handle(); 3387 config->ext.mix.usecase.stream = AUDIO_STREAM_DEFAULT; 3388} 3389 3390// ---------------------------------------------------------------------------- 3391 3392AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 3393 audio_io_handle_t id, audio_devices_t device, bool systemReady, type_t type) 3394 : PlaybackThread(audioFlinger, output, id, device, type, systemReady), 3395 // mAudioMixer below 3396 // mFastMixer below 3397 mFastMixerFutex(0), 3398 mMasterMono(false) 3399 // mOutputSink below 3400 // mPipeSink below 3401 // mNormalSink below 3402{ 3403 ALOGV("MixerThread() id=%d device=%#x type=%d", id, device, type); 3404 ALOGV("mSampleRate=%u, mChannelMask=%#x, mChannelCount=%u, mFormat=%d, mFrameSize=%u, " 3405 "mFrameCount=%d, mNormalFrameCount=%d", 3406 mSampleRate, mChannelMask, mChannelCount, mFormat, mFrameSize, mFrameCount, 3407 mNormalFrameCount); 3408 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 3409 3410 if (type == DUPLICATING) { 3411 // The Duplicating thread uses the AudioMixer and delivers data to OutputTracks 3412 // (downstream MixerThreads) in DuplicatingThread::threadLoop_write(). 3413 // Do not create or use mFastMixer, mOutputSink, mPipeSink, or mNormalSink. 3414 return; 3415 } 3416 // create an NBAIO sink for the HAL output stream, and negotiate 3417 mOutputSink = new AudioStreamOutSink(output->stream); 3418 size_t numCounterOffers = 0; 3419 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount, mFormat)}; 3420 ssize_t index = mOutputSink->negotiate(offers, 1, NULL, numCounterOffers); 3421 ALOG_ASSERT(index == 0); 3422 3423 // initialize fast mixer depending on configuration 3424 bool initFastMixer; 3425 switch (kUseFastMixer) { 3426 case FastMixer_Never: 3427 initFastMixer = false; 3428 break; 3429 case FastMixer_Always: 3430 initFastMixer = true; 3431 break; 3432 case FastMixer_Static: 3433 case FastMixer_Dynamic: 3434 initFastMixer = mFrameCount < mNormalFrameCount; 3435 break; 3436 } 3437 if (initFastMixer) { 3438 audio_format_t fastMixerFormat; 3439 if (mMixerBufferEnabled && mEffectBufferEnabled) { 3440 fastMixerFormat = AUDIO_FORMAT_PCM_FLOAT; 3441 } else { 3442 fastMixerFormat = AUDIO_FORMAT_PCM_16_BIT; 3443 } 3444 if (mFormat != fastMixerFormat) { 3445 // change our Sink format to accept our intermediate precision 3446 mFormat = fastMixerFormat; 3447 free(mSinkBuffer); 3448 mFrameSize = mChannelCount * audio_bytes_per_sample(mFormat); 3449 const size_t sinkBufferSize = mNormalFrameCount * mFrameSize; 3450 (void)posix_memalign(&mSinkBuffer, 32, sinkBufferSize); 3451 } 3452 3453 // create a MonoPipe to connect our submix to FastMixer 3454 NBAIO_Format format = mOutputSink->format(); 3455 NBAIO_Format origformat = format; 3456 // adjust format to match that of the Fast Mixer 3457 ALOGV("format changed from %d to %d", format.mFormat, fastMixerFormat); 3458 format.mFormat = fastMixerFormat; 3459 format.mFrameSize = audio_bytes_per_sample(format.mFormat) * format.mChannelCount; 3460 3461 // This pipe depth compensates for scheduling latency of the normal mixer thread. 3462 // When it wakes up after a maximum latency, it runs a few cycles quickly before 3463 // finally blocking. Note the pipe implementation rounds up the request to a power of 2. 3464 MonoPipe *monoPipe = new MonoPipe(mNormalFrameCount * 4, format, true /*writeCanBlock*/); 3465 const NBAIO_Format offers[1] = {format}; 3466 size_t numCounterOffers = 0; 3467 ssize_t index = monoPipe->negotiate(offers, 1, NULL, numCounterOffers); 3468 ALOG_ASSERT(index == 0); 3469 monoPipe->setAvgFrames((mScreenState & 1) ? 3470 (monoPipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 3471 mPipeSink = monoPipe; 3472 3473#ifdef TEE_SINK 3474 if (mTeeSinkOutputEnabled) { 3475 // create a Pipe to archive a copy of FastMixer's output for dumpsys 3476 Pipe *teeSink = new Pipe(mTeeSinkOutputFrames, origformat); 3477 const NBAIO_Format offers2[1] = {origformat}; 3478 numCounterOffers = 0; 3479 index = teeSink->negotiate(offers2, 1, NULL, numCounterOffers); 3480 ALOG_ASSERT(index == 0); 3481 mTeeSink = teeSink; 3482 PipeReader *teeSource = new PipeReader(*teeSink); 3483 numCounterOffers = 0; 3484 index = teeSource->negotiate(offers2, 1, NULL, numCounterOffers); 3485 ALOG_ASSERT(index == 0); 3486 mTeeSource = teeSource; 3487 } 3488#endif 3489 3490 // create fast mixer and configure it initially with just one fast track for our submix 3491 mFastMixer = new FastMixer(); 3492 FastMixerStateQueue *sq = mFastMixer->sq(); 3493#ifdef STATE_QUEUE_DUMP 3494 sq->setObserverDump(&mStateQueueObserverDump); 3495 sq->setMutatorDump(&mStateQueueMutatorDump); 3496#endif 3497 FastMixerState *state = sq->begin(); 3498 FastTrack *fastTrack = &state->mFastTracks[0]; 3499 // wrap the source side of the MonoPipe to make it an AudioBufferProvider 3500 fastTrack->mBufferProvider = new SourceAudioBufferProvider(new MonoPipeReader(monoPipe)); 3501 fastTrack->mVolumeProvider = NULL; 3502 fastTrack->mChannelMask = mChannelMask; // mPipeSink channel mask for audio to FastMixer 3503 fastTrack->mFormat = mFormat; // mPipeSink format for audio to FastMixer 3504 fastTrack->mGeneration++; 3505 state->mFastTracksGen++; 3506 state->mTrackMask = 1; 3507 // fast mixer will use the HAL output sink 3508 state->mOutputSink = mOutputSink.get(); 3509 state->mOutputSinkGen++; 3510 state->mFrameCount = mFrameCount; 3511 state->mCommand = FastMixerState::COLD_IDLE; 3512 // already done in constructor initialization list 3513 //mFastMixerFutex = 0; 3514 state->mColdFutexAddr = &mFastMixerFutex; 3515 state->mColdGen++; 3516 state->mDumpState = &mFastMixerDumpState; 3517#ifdef TEE_SINK 3518 state->mTeeSink = mTeeSink.get(); 3519#endif 3520 mFastMixerNBLogWriter = audioFlinger->newWriter_l(kFastMixerLogSize, "FastMixer"); 3521 state->mNBLogWriter = mFastMixerNBLogWriter.get(); 3522 sq->end(); 3523 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3524 3525 // start the fast mixer 3526 mFastMixer->run("FastMixer", PRIORITY_URGENT_AUDIO); 3527 pid_t tid = mFastMixer->getTid(); 3528 sendPrioConfigEvent(getpid_cached, tid, kPriorityFastMixer); 3529 3530#ifdef AUDIO_WATCHDOG 3531 // create and start the watchdog 3532 mAudioWatchdog = new AudioWatchdog(); 3533 mAudioWatchdog->setDump(&mAudioWatchdogDump); 3534 mAudioWatchdog->run("AudioWatchdog", PRIORITY_URGENT_AUDIO); 3535 tid = mAudioWatchdog->getTid(); 3536 sendPrioConfigEvent(getpid_cached, tid, kPriorityFastMixer); 3537#endif 3538 3539 } 3540 3541 switch (kUseFastMixer) { 3542 case FastMixer_Never: 3543 case FastMixer_Dynamic: 3544 mNormalSink = mOutputSink; 3545 break; 3546 case FastMixer_Always: 3547 mNormalSink = mPipeSink; 3548 break; 3549 case FastMixer_Static: 3550 mNormalSink = initFastMixer ? mPipeSink : mOutputSink; 3551 break; 3552 } 3553} 3554 3555AudioFlinger::MixerThread::~MixerThread() 3556{ 3557 if (mFastMixer != 0) { 3558 FastMixerStateQueue *sq = mFastMixer->sq(); 3559 FastMixerState *state = sq->begin(); 3560 if (state->mCommand == FastMixerState::COLD_IDLE) { 3561 int32_t old = android_atomic_inc(&mFastMixerFutex); 3562 if (old == -1) { 3563 (void) syscall(__NR_futex, &mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 3564 } 3565 } 3566 state->mCommand = FastMixerState::EXIT; 3567 sq->end(); 3568 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3569 mFastMixer->join(); 3570 // Though the fast mixer thread has exited, it's state queue is still valid. 3571 // We'll use that extract the final state which contains one remaining fast track 3572 // corresponding to our sub-mix. 3573 state = sq->begin(); 3574 ALOG_ASSERT(state->mTrackMask == 1); 3575 FastTrack *fastTrack = &state->mFastTracks[0]; 3576 ALOG_ASSERT(fastTrack->mBufferProvider != NULL); 3577 delete fastTrack->mBufferProvider; 3578 sq->end(false /*didModify*/); 3579 mFastMixer.clear(); 3580#ifdef AUDIO_WATCHDOG 3581 if (mAudioWatchdog != 0) { 3582 mAudioWatchdog->requestExit(); 3583 mAudioWatchdog->requestExitAndWait(); 3584 mAudioWatchdog.clear(); 3585 } 3586#endif 3587 } 3588 mAudioFlinger->unregisterWriter(mFastMixerNBLogWriter); 3589 delete mAudioMixer; 3590} 3591 3592 3593uint32_t AudioFlinger::MixerThread::correctLatency_l(uint32_t latency) const 3594{ 3595 if (mFastMixer != 0) { 3596 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 3597 latency += (pipe->getAvgFrames() * 1000) / mSampleRate; 3598 } 3599 return latency; 3600} 3601 3602 3603void AudioFlinger::MixerThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) 3604{ 3605 PlaybackThread::threadLoop_removeTracks(tracksToRemove); 3606} 3607 3608ssize_t AudioFlinger::MixerThread::threadLoop_write() 3609{ 3610 // FIXME we should only do one push per cycle; confirm this is true 3611 // Start the fast mixer if it's not already running 3612 if (mFastMixer != 0) { 3613 FastMixerStateQueue *sq = mFastMixer->sq(); 3614 FastMixerState *state = sq->begin(); 3615 if (state->mCommand != FastMixerState::MIX_WRITE && 3616 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)) { 3617 if (state->mCommand == FastMixerState::COLD_IDLE) { 3618 3619 // FIXME workaround for first HAL write being CPU bound on some devices 3620 ATRACE_BEGIN("write"); 3621 mOutput->write((char *)mSinkBuffer, 0); 3622 ATRACE_END(); 3623 3624 int32_t old = android_atomic_inc(&mFastMixerFutex); 3625 if (old == -1) { 3626 (void) syscall(__NR_futex, &mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 3627 } 3628#ifdef AUDIO_WATCHDOG 3629 if (mAudioWatchdog != 0) { 3630 mAudioWatchdog->resume(); 3631 } 3632#endif 3633 } 3634 state->mCommand = FastMixerState::MIX_WRITE; 3635#ifdef FAST_THREAD_STATISTICS 3636 mFastMixerDumpState.increaseSamplingN(mAudioFlinger->isLowRamDevice() ? 3637 FastThreadDumpState::kSamplingNforLowRamDevice : FastThreadDumpState::kSamplingN); 3638#endif 3639 sq->end(); 3640 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3641 if (kUseFastMixer == FastMixer_Dynamic) { 3642 mNormalSink = mPipeSink; 3643 } 3644 } else { 3645 sq->end(false /*didModify*/); 3646 } 3647 } 3648 return PlaybackThread::threadLoop_write(); 3649} 3650 3651void AudioFlinger::MixerThread::threadLoop_standby() 3652{ 3653 // Idle the fast mixer if it's currently running 3654 if (mFastMixer != 0) { 3655 FastMixerStateQueue *sq = mFastMixer->sq(); 3656 FastMixerState *state = sq->begin(); 3657 if (!(state->mCommand & FastMixerState::IDLE)) { 3658 state->mCommand = FastMixerState::COLD_IDLE; 3659 state->mColdFutexAddr = &mFastMixerFutex; 3660 state->mColdGen++; 3661 mFastMixerFutex = 0; 3662 sq->end(); 3663 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 3664 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3665 if (kUseFastMixer == FastMixer_Dynamic) { 3666 mNormalSink = mOutputSink; 3667 } 3668#ifdef AUDIO_WATCHDOG 3669 if (mAudioWatchdog != 0) { 3670 mAudioWatchdog->pause(); 3671 } 3672#endif 3673 } else { 3674 sq->end(false /*didModify*/); 3675 } 3676 } 3677 PlaybackThread::threadLoop_standby(); 3678} 3679 3680bool AudioFlinger::PlaybackThread::waitingAsyncCallback_l() 3681{ 3682 return false; 3683} 3684 3685bool AudioFlinger::PlaybackThread::shouldStandby_l() 3686{ 3687 return !mStandby; 3688} 3689 3690bool AudioFlinger::PlaybackThread::waitingAsyncCallback() 3691{ 3692 Mutex::Autolock _l(mLock); 3693 return waitingAsyncCallback_l(); 3694} 3695 3696// shared by MIXER and DIRECT, overridden by DUPLICATING 3697void AudioFlinger::PlaybackThread::threadLoop_standby() 3698{ 3699 ALOGV("Audio hardware entering standby, mixer %p, suspend count %d", this, mSuspended); 3700 mOutput->standby(); 3701 if (mUseAsyncWrite != 0) { 3702 // discard any pending drain or write ack by incrementing sequence 3703 mWriteAckSequence = (mWriteAckSequence + 2) & ~1; 3704 mDrainSequence = (mDrainSequence + 2) & ~1; 3705 ALOG_ASSERT(mCallbackThread != 0); 3706 mCallbackThread->setWriteBlocked(mWriteAckSequence); 3707 mCallbackThread->setDraining(mDrainSequence); 3708 } 3709 mHwPaused = false; 3710} 3711 3712void AudioFlinger::PlaybackThread::onAddNewTrack_l() 3713{ 3714 ALOGV("signal playback thread"); 3715 broadcast_l(); 3716} 3717 3718void AudioFlinger::MixerThread::threadLoop_mix() 3719{ 3720 // mix buffers... 3721 mAudioMixer->process(); 3722 mCurrentWriteLength = mSinkBufferSize; 3723 // increase sleep time progressively when application underrun condition clears. 3724 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 3725 // that a steady state of alternating ready/not ready conditions keeps the sleep time 3726 // such that we would underrun the audio HAL. 3727 if ((mSleepTimeUs == 0) && (sleepTimeShift > 0)) { 3728 sleepTimeShift--; 3729 } 3730 mSleepTimeUs = 0; 3731 mStandbyTimeNs = systemTime() + mStandbyDelayNs; 3732 //TODO: delay standby when effects have a tail 3733 3734} 3735 3736void AudioFlinger::MixerThread::threadLoop_sleepTime() 3737{ 3738 // If no tracks are ready, sleep once for the duration of an output 3739 // buffer size, then write 0s to the output 3740 if (mSleepTimeUs == 0) { 3741 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3742 mSleepTimeUs = mActiveSleepTimeUs >> sleepTimeShift; 3743 if (mSleepTimeUs < kMinThreadSleepTimeUs) { 3744 mSleepTimeUs = kMinThreadSleepTimeUs; 3745 } 3746 // reduce sleep time in case of consecutive application underruns to avoid 3747 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 3748 // duration we would end up writing less data than needed by the audio HAL if 3749 // the condition persists. 3750 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 3751 sleepTimeShift++; 3752 } 3753 } else { 3754 mSleepTimeUs = mIdleSleepTimeUs; 3755 } 3756 } else if (mBytesWritten != 0 || (mMixerStatus == MIXER_TRACKS_ENABLED)) { 3757 // clear out mMixerBuffer or mSinkBuffer, to ensure buffers are cleared 3758 // before effects processing or output. 3759 if (mMixerBufferValid) { 3760 memset(mMixerBuffer, 0, mMixerBufferSize); 3761 } else { 3762 memset(mSinkBuffer, 0, mSinkBufferSize); 3763 } 3764 mSleepTimeUs = 0; 3765 ALOGV_IF(mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED), 3766 "anticipated start"); 3767 } 3768 // TODO add standby time extension fct of effect tail 3769} 3770 3771// prepareTracks_l() must be called with ThreadBase::mLock held 3772AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 3773 Vector< sp<Track> > *tracksToRemove) 3774{ 3775 3776 mixer_state mixerStatus = MIXER_IDLE; 3777 // find out which tracks need to be processed 3778 size_t count = mActiveTracks.size(); 3779 size_t mixedTracks = 0; 3780 size_t tracksWithEffect = 0; 3781 // counts only _active_ fast tracks 3782 size_t fastTracks = 0; 3783 uint32_t resetMask = 0; // bit mask of fast tracks that need to be reset 3784 3785 float masterVolume = mMasterVolume; 3786 bool masterMute = mMasterMute; 3787 3788 if (masterMute) { 3789 masterVolume = 0; 3790 } 3791 // Delegate master volume control to effect in output mix effect chain if needed 3792 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 3793 if (chain != 0) { 3794 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 3795 chain->setVolume_l(&v, &v); 3796 masterVolume = (float)((v + (1 << 23)) >> 24); 3797 chain.clear(); 3798 } 3799 3800 // prepare a new state to push 3801 FastMixerStateQueue *sq = NULL; 3802 FastMixerState *state = NULL; 3803 bool didModify = false; 3804 FastMixerStateQueue::block_t block = FastMixerStateQueue::BLOCK_UNTIL_PUSHED; 3805 if (mFastMixer != 0) { 3806 sq = mFastMixer->sq(); 3807 state = sq->begin(); 3808 } 3809 3810 mMixerBufferValid = false; // mMixerBuffer has no valid data until appropriate tracks found. 3811 mEffectBufferValid = false; // mEffectBuffer has no valid data until tracks found. 3812 3813 for (size_t i=0 ; i<count ; i++) { 3814 const sp<Track> t = mActiveTracks[i].promote(); 3815 if (t == 0) { 3816 continue; 3817 } 3818 3819 // this const just means the local variable doesn't change 3820 Track* const track = t.get(); 3821 3822 // process fast tracks 3823 if (track->isFastTrack()) { 3824 3825 // It's theoretically possible (though unlikely) for a fast track to be created 3826 // and then removed within the same normal mix cycle. This is not a problem, as 3827 // the track never becomes active so it's fast mixer slot is never touched. 3828 // The converse, of removing an (active) track and then creating a new track 3829 // at the identical fast mixer slot within the same normal mix cycle, 3830 // is impossible because the slot isn't marked available until the end of each cycle. 3831 int j = track->mFastIndex; 3832 ALOG_ASSERT(0 < j && j < (int)FastMixerState::kMaxFastTracks); 3833 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << j))); 3834 FastTrack *fastTrack = &state->mFastTracks[j]; 3835 3836 // Determine whether the track is currently in underrun condition, 3837 // and whether it had a recent underrun. 3838 FastTrackDump *ftDump = &mFastMixerDumpState.mTracks[j]; 3839 FastTrackUnderruns underruns = ftDump->mUnderruns; 3840 uint32_t recentFull = (underruns.mBitFields.mFull - 3841 track->mObservedUnderruns.mBitFields.mFull) & UNDERRUN_MASK; 3842 uint32_t recentPartial = (underruns.mBitFields.mPartial - 3843 track->mObservedUnderruns.mBitFields.mPartial) & UNDERRUN_MASK; 3844 uint32_t recentEmpty = (underruns.mBitFields.mEmpty - 3845 track->mObservedUnderruns.mBitFields.mEmpty) & UNDERRUN_MASK; 3846 uint32_t recentUnderruns = recentPartial + recentEmpty; 3847 track->mObservedUnderruns = underruns; 3848 // don't count underruns that occur while stopping or pausing 3849 // or stopped which can occur when flush() is called while active 3850 if (!(track->isStopping() || track->isPausing() || track->isStopped()) && 3851 recentUnderruns > 0) { 3852 // FIXME fast mixer will pull & mix partial buffers, but we count as a full underrun 3853 track->mAudioTrackServerProxy->tallyUnderrunFrames(recentUnderruns * mFrameCount); 3854 } else { 3855 track->mAudioTrackServerProxy->tallyUnderrunFrames(0); 3856 } 3857 3858 // This is similar to the state machine for normal tracks, 3859 // with a few modifications for fast tracks. 3860 bool isActive = true; 3861 switch (track->mState) { 3862 case TrackBase::STOPPING_1: 3863 // track stays active in STOPPING_1 state until first underrun 3864 if (recentUnderruns > 0 || track->isTerminated()) { 3865 track->mState = TrackBase::STOPPING_2; 3866 } 3867 break; 3868 case TrackBase::PAUSING: 3869 // ramp down is not yet implemented 3870 track->setPaused(); 3871 break; 3872 case TrackBase::RESUMING: 3873 // ramp up is not yet implemented 3874 track->mState = TrackBase::ACTIVE; 3875 break; 3876 case TrackBase::ACTIVE: 3877 if (recentFull > 0 || recentPartial > 0) { 3878 // track has provided at least some frames recently: reset retry count 3879 track->mRetryCount = kMaxTrackRetries; 3880 } 3881 if (recentUnderruns == 0) { 3882 // no recent underruns: stay active 3883 break; 3884 } 3885 // there has recently been an underrun of some kind 3886 if (track->sharedBuffer() == 0) { 3887 // were any of the recent underruns "empty" (no frames available)? 3888 if (recentEmpty == 0) { 3889 // no, then ignore the partial underruns as they are allowed indefinitely 3890 break; 3891 } 3892 // there has recently been an "empty" underrun: decrement the retry counter 3893 if (--(track->mRetryCount) > 0) { 3894 break; 3895 } 3896 // indicate to client process that the track was disabled because of underrun; 3897 // it will then automatically call start() when data is available 3898 android_atomic_or(CBLK_DISABLED, &track->mCblk->mFlags); 3899 // remove from active list, but state remains ACTIVE [confusing but true] 3900 isActive = false; 3901 break; 3902 } 3903 // fall through 3904 case TrackBase::STOPPING_2: 3905 case TrackBase::PAUSED: 3906 case TrackBase::STOPPED: 3907 case TrackBase::FLUSHED: // flush() while active 3908 // Check for presentation complete if track is inactive 3909 // We have consumed all the buffers of this track. 3910 // This would be incomplete if we auto-paused on underrun 3911 { 3912 size_t audioHALFrames = 3913 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 3914 size_t framesWritten = mBytesWritten / mFrameSize; 3915 if (!(mStandby || track->presentationComplete(framesWritten, audioHALFrames))) { 3916 // track stays in active list until presentation is complete 3917 break; 3918 } 3919 } 3920 if (track->isStopping_2()) { 3921 track->mState = TrackBase::STOPPED; 3922 } 3923 if (track->isStopped()) { 3924 // Can't reset directly, as fast mixer is still polling this track 3925 // track->reset(); 3926 // So instead mark this track as needing to be reset after push with ack 3927 resetMask |= 1 << i; 3928 } 3929 isActive = false; 3930 break; 3931 case TrackBase::IDLE: 3932 default: 3933 LOG_ALWAYS_FATAL("unexpected track state %d", track->mState); 3934 } 3935 3936 if (isActive) { 3937 // was it previously inactive? 3938 if (!(state->mTrackMask & (1 << j))) { 3939 ExtendedAudioBufferProvider *eabp = track; 3940 VolumeProvider *vp = track; 3941 fastTrack->mBufferProvider = eabp; 3942 fastTrack->mVolumeProvider = vp; 3943 fastTrack->mChannelMask = track->mChannelMask; 3944 fastTrack->mFormat = track->mFormat; 3945 fastTrack->mGeneration++; 3946 state->mTrackMask |= 1 << j; 3947 didModify = true; 3948 // no acknowledgement required for newly active tracks 3949 } 3950 // cache the combined master volume and stream type volume for fast mixer; this 3951 // lacks any synchronization or barrier so VolumeProvider may read a stale value 3952 track->mCachedVolume = masterVolume * mStreamTypes[track->streamType()].volume; 3953 ++fastTracks; 3954 } else { 3955 // was it previously active? 3956 if (state->mTrackMask & (1 << j)) { 3957 fastTrack->mBufferProvider = NULL; 3958 fastTrack->mGeneration++; 3959 state->mTrackMask &= ~(1 << j); 3960 didModify = true; 3961 // If any fast tracks were removed, we must wait for acknowledgement 3962 // because we're about to decrement the last sp<> on those tracks. 3963 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3964 } else { 3965 LOG_ALWAYS_FATAL("fast track %d should have been active; " 3966 "mState=%d, mTrackMask=%#x, recentUnderruns=%u, isShared=%d", 3967 j, track->mState, state->mTrackMask, recentUnderruns, 3968 track->sharedBuffer() != 0); 3969 } 3970 tracksToRemove->add(track); 3971 // Avoids a misleading display in dumpsys 3972 track->mObservedUnderruns.mBitFields.mMostRecent = UNDERRUN_FULL; 3973 } 3974 continue; 3975 } 3976 3977 { // local variable scope to avoid goto warning 3978 3979 audio_track_cblk_t* cblk = track->cblk(); 3980 3981 // The first time a track is added we wait 3982 // for all its buffers to be filled before processing it 3983 int name = track->name(); 3984 // make sure that we have enough frames to mix one full buffer. 3985 // enforce this condition only once to enable draining the buffer in case the client 3986 // app does not call stop() and relies on underrun to stop: 3987 // hence the test on (mMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 3988 // during last round 3989 size_t desiredFrames; 3990 const uint32_t sampleRate = track->mAudioTrackServerProxy->getSampleRate(); 3991 AudioPlaybackRate playbackRate = track->mAudioTrackServerProxy->getPlaybackRate(); 3992 3993 desiredFrames = sourceFramesNeededWithTimestretch( 3994 sampleRate, mNormalFrameCount, mSampleRate, playbackRate.mSpeed); 3995 // TODO: ONLY USED FOR LEGACY RESAMPLERS, remove when they are removed. 3996 // add frames already consumed but not yet released by the resampler 3997 // because mAudioTrackServerProxy->framesReady() will include these frames 3998 desiredFrames += mAudioMixer->getUnreleasedFrames(track->name()); 3999 4000 uint32_t minFrames = 1; 4001 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() && 4002 (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) { 4003 minFrames = desiredFrames; 4004 } 4005 4006 size_t framesReady = track->framesReady(); 4007 if (ATRACE_ENABLED()) { 4008 // I wish we had formatted trace names 4009 char traceName[16]; 4010 strcpy(traceName, "nRdy"); 4011 int name = track->name(); 4012 if (AudioMixer::TRACK0 <= name && 4013 name < (int) (AudioMixer::TRACK0 + AudioMixer::MAX_NUM_TRACKS)) { 4014 name -= AudioMixer::TRACK0; 4015 traceName[4] = (name / 10) + '0'; 4016 traceName[5] = (name % 10) + '0'; 4017 } else { 4018 traceName[4] = '?'; 4019 traceName[5] = '?'; 4020 } 4021 traceName[6] = '\0'; 4022 ATRACE_INT(traceName, framesReady); 4023 } 4024 if ((framesReady >= minFrames) && track->isReady() && 4025 !track->isPaused() && !track->isTerminated()) 4026 { 4027 ALOGVV("track %d s=%08x [OK] on thread %p", name, cblk->mServer, this); 4028 4029 mixedTracks++; 4030 4031 // track->mainBuffer() != mSinkBuffer or mMixerBuffer means 4032 // there is an effect chain connected to the track 4033 chain.clear(); 4034 if (track->mainBuffer() != mSinkBuffer && 4035 track->mainBuffer() != mMixerBuffer) { 4036 if (mEffectBufferEnabled) { 4037 mEffectBufferValid = true; // Later can set directly. 4038 } 4039 chain = getEffectChain_l(track->sessionId()); 4040 // Delegate volume control to effect in track effect chain if needed 4041 if (chain != 0) { 4042 tracksWithEffect++; 4043 } else { 4044 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on " 4045 "session %d", 4046 name, track->sessionId()); 4047 } 4048 } 4049 4050 4051 int param = AudioMixer::VOLUME; 4052 if (track->mFillingUpStatus == Track::FS_FILLED) { 4053 // no ramp for the first volume setting 4054 track->mFillingUpStatus = Track::FS_ACTIVE; 4055 if (track->mState == TrackBase::RESUMING) { 4056 track->mState = TrackBase::ACTIVE; 4057 param = AudioMixer::RAMP_VOLUME; 4058 } 4059 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 4060 // FIXME should not make a decision based on mServer 4061 } else if (cblk->mServer != 0) { 4062 // If the track is stopped before the first frame was mixed, 4063 // do not apply ramp 4064 param = AudioMixer::RAMP_VOLUME; 4065 } 4066 4067 // compute volume for this track 4068 uint32_t vl, vr; // in U8.24 integer format 4069 float vlf, vrf, vaf; // in [0.0, 1.0] float format 4070 if (track->isPausing() || mStreamTypes[track->streamType()].mute) { 4071 vl = vr = 0; 4072 vlf = vrf = vaf = 0.; 4073 if (track->isPausing()) { 4074 track->setPaused(); 4075 } 4076 } else { 4077 4078 // read original volumes with volume control 4079 float typeVolume = mStreamTypes[track->streamType()].volume; 4080 float v = masterVolume * typeVolume; 4081 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 4082 gain_minifloat_packed_t vlr = proxy->getVolumeLR(); 4083 vlf = float_from_gain(gain_minifloat_unpack_left(vlr)); 4084 vrf = float_from_gain(gain_minifloat_unpack_right(vlr)); 4085 // track volumes come from shared memory, so can't be trusted and must be clamped 4086 if (vlf > GAIN_FLOAT_UNITY) { 4087 ALOGV("Track left volume out of range: %.3g", vlf); 4088 vlf = GAIN_FLOAT_UNITY; 4089 } 4090 if (vrf > GAIN_FLOAT_UNITY) { 4091 ALOGV("Track right volume out of range: %.3g", vrf); 4092 vrf = GAIN_FLOAT_UNITY; 4093 } 4094 // now apply the master volume and stream type volume 4095 vlf *= v; 4096 vrf *= v; 4097 // assuming master volume and stream type volume each go up to 1.0, 4098 // then derive vl and vr as U8.24 versions for the effect chain 4099 const float scaleto8_24 = MAX_GAIN_INT * MAX_GAIN_INT; 4100 vl = (uint32_t) (scaleto8_24 * vlf); 4101 vr = (uint32_t) (scaleto8_24 * vrf); 4102 // vl and vr are now in U8.24 format 4103 uint16_t sendLevel = proxy->getSendLevel_U4_12(); 4104 // send level comes from shared memory and so may be corrupt 4105 if (sendLevel > MAX_GAIN_INT) { 4106 ALOGV("Track send level out of range: %04X", sendLevel); 4107 sendLevel = MAX_GAIN_INT; 4108 } 4109 // vaf is represented as [0.0, 1.0] float by rescaling sendLevel 4110 vaf = v * sendLevel * (1. / MAX_GAIN_INT); 4111 } 4112 4113 // Delegate volume control to effect in track effect chain if needed 4114 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 4115 // Do not ramp volume if volume is controlled by effect 4116 param = AudioMixer::VOLUME; 4117 // Update remaining floating point volume levels 4118 vlf = (float)vl / (1 << 24); 4119 vrf = (float)vr / (1 << 24); 4120 track->mHasVolumeController = true; 4121 } else { 4122 // force no volume ramp when volume controller was just disabled or removed 4123 // from effect chain to avoid volume spike 4124 if (track->mHasVolumeController) { 4125 param = AudioMixer::VOLUME; 4126 } 4127 track->mHasVolumeController = false; 4128 } 4129 4130 // XXX: these things DON'T need to be done each time 4131 mAudioMixer->setBufferProvider(name, track); 4132 mAudioMixer->enable(name); 4133 4134 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, &vlf); 4135 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, &vrf); 4136 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, &vaf); 4137 mAudioMixer->setParameter( 4138 name, 4139 AudioMixer::TRACK, 4140 AudioMixer::FORMAT, (void *)track->format()); 4141 mAudioMixer->setParameter( 4142 name, 4143 AudioMixer::TRACK, 4144 AudioMixer::CHANNEL_MASK, (void *)(uintptr_t)track->channelMask()); 4145 mAudioMixer->setParameter( 4146 name, 4147 AudioMixer::TRACK, 4148 AudioMixer::MIXER_CHANNEL_MASK, (void *)(uintptr_t)mChannelMask); 4149 // limit track sample rate to 2 x output sample rate, which changes at re-configuration 4150 uint32_t maxSampleRate = mSampleRate * AUDIO_RESAMPLER_DOWN_RATIO_MAX; 4151 uint32_t reqSampleRate = track->mAudioTrackServerProxy->getSampleRate(); 4152 if (reqSampleRate == 0) { 4153 reqSampleRate = mSampleRate; 4154 } else if (reqSampleRate > maxSampleRate) { 4155 reqSampleRate = maxSampleRate; 4156 } 4157 mAudioMixer->setParameter( 4158 name, 4159 AudioMixer::RESAMPLE, 4160 AudioMixer::SAMPLE_RATE, 4161 (void *)(uintptr_t)reqSampleRate); 4162 4163 AudioPlaybackRate playbackRate = track->mAudioTrackServerProxy->getPlaybackRate(); 4164 mAudioMixer->setParameter( 4165 name, 4166 AudioMixer::TIMESTRETCH, 4167 AudioMixer::PLAYBACK_RATE, 4168 &playbackRate); 4169 4170 /* 4171 * Select the appropriate output buffer for the track. 4172 * 4173 * Tracks with effects go into their own effects chain buffer 4174 * and from there into either mEffectBuffer or mSinkBuffer. 4175 * 4176 * Other tracks can use mMixerBuffer for higher precision 4177 * channel accumulation. If this buffer is enabled 4178 * (mMixerBufferEnabled true), then selected tracks will accumulate 4179 * into it. 4180 * 4181 */ 4182 if (mMixerBufferEnabled 4183 && (track->mainBuffer() == mSinkBuffer 4184 || track->mainBuffer() == mMixerBuffer)) { 4185 mAudioMixer->setParameter( 4186 name, 4187 AudioMixer::TRACK, 4188 AudioMixer::MIXER_FORMAT, (void *)mMixerBufferFormat); 4189 mAudioMixer->setParameter( 4190 name, 4191 AudioMixer::TRACK, 4192 AudioMixer::MAIN_BUFFER, (void *)mMixerBuffer); 4193 // TODO: override track->mainBuffer()? 4194 mMixerBufferValid = true; 4195 } else { 4196 mAudioMixer->setParameter( 4197 name, 4198 AudioMixer::TRACK, 4199 AudioMixer::MIXER_FORMAT, (void *)AUDIO_FORMAT_PCM_16_BIT); 4200 mAudioMixer->setParameter( 4201 name, 4202 AudioMixer::TRACK, 4203 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 4204 } 4205 mAudioMixer->setParameter( 4206 name, 4207 AudioMixer::TRACK, 4208 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 4209 4210 // reset retry count 4211 track->mRetryCount = kMaxTrackRetries; 4212 4213 // If one track is ready, set the mixer ready if: 4214 // - the mixer was not ready during previous round OR 4215 // - no other track is not ready 4216 if (mMixerStatusIgnoringFastTracks != MIXER_TRACKS_READY || 4217 mixerStatus != MIXER_TRACKS_ENABLED) { 4218 mixerStatus = MIXER_TRACKS_READY; 4219 } 4220 } else { 4221 if (framesReady < desiredFrames && !track->isStopped() && !track->isPaused()) { 4222 ALOGV("track(%p) underrun, framesReady(%zu) < framesDesired(%zd)", 4223 track, framesReady, desiredFrames); 4224 track->mAudioTrackServerProxy->tallyUnderrunFrames(desiredFrames); 4225 } else { 4226 track->mAudioTrackServerProxy->tallyUnderrunFrames(0); 4227 } 4228 4229 // clear effect chain input buffer if an active track underruns to avoid sending 4230 // previous audio buffer again to effects 4231 chain = getEffectChain_l(track->sessionId()); 4232 if (chain != 0) { 4233 chain->clearInputBuffer(); 4234 } 4235 4236 ALOGVV("track %d s=%08x [NOT READY] on thread %p", name, cblk->mServer, this); 4237 if ((track->sharedBuffer() != 0) || track->isTerminated() || 4238 track->isStopped() || track->isPaused()) { 4239 // We have consumed all the buffers of this track. 4240 // Remove it from the list of active tracks. 4241 // TODO: use actual buffer filling status instead of latency when available from 4242 // audio HAL 4243 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 4244 size_t framesWritten = mBytesWritten / mFrameSize; 4245 if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) { 4246 if (track->isStopped()) { 4247 track->reset(); 4248 } 4249 tracksToRemove->add(track); 4250 } 4251 } else { 4252 // No buffers for this track. Give it a few chances to 4253 // fill a buffer, then remove it from active list. 4254 if (--(track->mRetryCount) <= 0) { 4255 ALOGI("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 4256 tracksToRemove->add(track); 4257 // indicate to client process that the track was disabled because of underrun; 4258 // it will then automatically call start() when data is available 4259 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 4260 // If one track is not ready, mark the mixer also not ready if: 4261 // - the mixer was ready during previous round OR 4262 // - no other track is ready 4263 } else if (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY || 4264 mixerStatus != MIXER_TRACKS_READY) { 4265 mixerStatus = MIXER_TRACKS_ENABLED; 4266 } 4267 } 4268 mAudioMixer->disable(name); 4269 } 4270 4271 } // local variable scope to avoid goto warning 4272track_is_ready: ; 4273 4274 } 4275 4276 // Push the new FastMixer state if necessary 4277 bool pauseAudioWatchdog = false; 4278 if (didModify) { 4279 state->mFastTracksGen++; 4280 // if the fast mixer was active, but now there are no fast tracks, then put it in cold idle 4281 if (kUseFastMixer == FastMixer_Dynamic && 4282 state->mCommand == FastMixerState::MIX_WRITE && state->mTrackMask <= 1) { 4283 state->mCommand = FastMixerState::COLD_IDLE; 4284 state->mColdFutexAddr = &mFastMixerFutex; 4285 state->mColdGen++; 4286 mFastMixerFutex = 0; 4287 if (kUseFastMixer == FastMixer_Dynamic) { 4288 mNormalSink = mOutputSink; 4289 } 4290 // If we go into cold idle, need to wait for acknowledgement 4291 // so that fast mixer stops doing I/O. 4292 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 4293 pauseAudioWatchdog = true; 4294 } 4295 } 4296 if (sq != NULL) { 4297 sq->end(didModify); 4298 sq->push(block); 4299 } 4300#ifdef AUDIO_WATCHDOG 4301 if (pauseAudioWatchdog && mAudioWatchdog != 0) { 4302 mAudioWatchdog->pause(); 4303 } 4304#endif 4305 4306 // Now perform the deferred reset on fast tracks that have stopped 4307 while (resetMask != 0) { 4308 size_t i = __builtin_ctz(resetMask); 4309 ALOG_ASSERT(i < count); 4310 resetMask &= ~(1 << i); 4311 sp<Track> t = mActiveTracks[i].promote(); 4312 if (t == 0) { 4313 continue; 4314 } 4315 Track* track = t.get(); 4316 ALOG_ASSERT(track->isFastTrack() && track->isStopped()); 4317 track->reset(); 4318 } 4319 4320 // remove all the tracks that need to be... 4321 removeTracks_l(*tracksToRemove); 4322 4323 if (getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX) != 0) { 4324 mEffectBufferValid = true; 4325 } 4326 4327 if (mEffectBufferValid) { 4328 // as long as there are effects we should clear the effects buffer, to avoid 4329 // passing a non-clean buffer to the effect chain 4330 memset(mEffectBuffer, 0, mEffectBufferSize); 4331 } 4332 // sink or mix buffer must be cleared if all tracks are connected to an 4333 // effect chain as in this case the mixer will not write to the sink or mix buffer 4334 // and track effects will accumulate into it 4335 if ((mBytesRemaining == 0) && ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 4336 (mixedTracks == 0 && fastTracks > 0))) { 4337 // FIXME as a performance optimization, should remember previous zero status 4338 if (mMixerBufferValid) { 4339 memset(mMixerBuffer, 0, mMixerBufferSize); 4340 // TODO: In testing, mSinkBuffer below need not be cleared because 4341 // the PlaybackThread::threadLoop() copies mMixerBuffer into mSinkBuffer 4342 // after mixing. 4343 // 4344 // To enforce this guarantee: 4345 // ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 4346 // (mixedTracks == 0 && fastTracks > 0)) 4347 // must imply MIXER_TRACKS_READY. 4348 // Later, we may clear buffers regardless, and skip much of this logic. 4349 } 4350 // FIXME as a performance optimization, should remember previous zero status 4351 memset(mSinkBuffer, 0, mNormalFrameCount * mFrameSize); 4352 } 4353 4354 // if any fast tracks, then status is ready 4355 mMixerStatusIgnoringFastTracks = mixerStatus; 4356 if (fastTracks > 0) { 4357 mixerStatus = MIXER_TRACKS_READY; 4358 } 4359 return mixerStatus; 4360} 4361 4362// getTrackName_l() must be called with ThreadBase::mLock held 4363int AudioFlinger::MixerThread::getTrackName_l(audio_channel_mask_t channelMask, 4364 audio_format_t format, int sessionId) 4365{ 4366 return mAudioMixer->getTrackName(channelMask, format, sessionId); 4367} 4368 4369// deleteTrackName_l() must be called with ThreadBase::mLock held 4370void AudioFlinger::MixerThread::deleteTrackName_l(int name) 4371{ 4372 ALOGV("remove track (%d) and delete from mixer", name); 4373 mAudioMixer->deleteTrackName(name); 4374} 4375 4376// checkForNewParameter_l() must be called with ThreadBase::mLock held 4377bool AudioFlinger::MixerThread::checkForNewParameter_l(const String8& keyValuePair, 4378 status_t& status) 4379{ 4380 bool reconfig = false; 4381 bool a2dpDeviceChanged = false; 4382 4383 status = NO_ERROR; 4384 4385 // if !&IDLE, holds the FastMixer state to restore after new parameters processed 4386 FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE; 4387 if (mFastMixer != 0) { 4388 FastMixerStateQueue *sq = mFastMixer->sq(); 4389 FastMixerState *state = sq->begin(); 4390 if (!(state->mCommand & FastMixerState::IDLE)) { 4391 previousCommand = state->mCommand; 4392 state->mCommand = FastMixerState::HOT_IDLE; 4393 sq->end(); 4394 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 4395 } else { 4396 sq->end(false /*didModify*/); 4397 } 4398 } 4399 4400 AudioParameter param = AudioParameter(keyValuePair); 4401 int value; 4402 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 4403 reconfig = true; 4404 } 4405 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 4406 if (!isValidPcmSinkFormat((audio_format_t) value)) { 4407 status = BAD_VALUE; 4408 } else { 4409 // no need to save value, since it's constant 4410 reconfig = true; 4411 } 4412 } 4413 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 4414 if (!isValidPcmSinkChannelMask((audio_channel_mask_t) value)) { 4415 status = BAD_VALUE; 4416 } else { 4417 // no need to save value, since it's constant 4418 reconfig = true; 4419 } 4420 } 4421 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 4422 // do not accept frame count changes if tracks are open as the track buffer 4423 // size depends on frame count and correct behavior would not be guaranteed 4424 // if frame count is changed after track creation 4425 if (!mTracks.isEmpty()) { 4426 status = INVALID_OPERATION; 4427 } else { 4428 reconfig = true; 4429 } 4430 } 4431 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 4432#ifdef ADD_BATTERY_DATA 4433 // when changing the audio output device, call addBatteryData to notify 4434 // the change 4435 if (mOutDevice != value) { 4436 uint32_t params = 0; 4437 // check whether speaker is on 4438 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 4439 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 4440 } 4441 4442 audio_devices_t deviceWithoutSpeaker 4443 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 4444 // check if any other device (except speaker) is on 4445 if (value & deviceWithoutSpeaker) { 4446 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 4447 } 4448 4449 if (params != 0) { 4450 addBatteryData(params); 4451 } 4452 } 4453#endif 4454 4455 // forward device change to effects that have requested to be 4456 // aware of attached audio device. 4457 if (value != AUDIO_DEVICE_NONE) { 4458 a2dpDeviceChanged = 4459 (mOutDevice & AUDIO_DEVICE_OUT_ALL_A2DP) != (value & AUDIO_DEVICE_OUT_ALL_A2DP); 4460 mOutDevice = value; 4461 for (size_t i = 0; i < mEffectChains.size(); i++) { 4462 mEffectChains[i]->setDevice_l(mOutDevice); 4463 } 4464 } 4465 } 4466 4467 if (status == NO_ERROR) { 4468 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 4469 keyValuePair.string()); 4470 if (!mStandby && status == INVALID_OPERATION) { 4471 mOutput->standby(); 4472 mStandby = true; 4473 mBytesWritten = 0; 4474 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 4475 keyValuePair.string()); 4476 } 4477 if (status == NO_ERROR && reconfig) { 4478 readOutputParameters_l(); 4479 delete mAudioMixer; 4480 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 4481 for (size_t i = 0; i < mTracks.size() ; i++) { 4482 int name = getTrackName_l(mTracks[i]->mChannelMask, 4483 mTracks[i]->mFormat, mTracks[i]->mSessionId); 4484 if (name < 0) { 4485 break; 4486 } 4487 mTracks[i]->mName = name; 4488 } 4489 sendIoConfigEvent_l(AUDIO_OUTPUT_CONFIG_CHANGED); 4490 } 4491 } 4492 4493 if (!(previousCommand & FastMixerState::IDLE)) { 4494 ALOG_ASSERT(mFastMixer != 0); 4495 FastMixerStateQueue *sq = mFastMixer->sq(); 4496 FastMixerState *state = sq->begin(); 4497 ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE); 4498 state->mCommand = previousCommand; 4499 sq->end(); 4500 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 4501 } 4502 4503 return reconfig || a2dpDeviceChanged; 4504} 4505 4506 4507void AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 4508{ 4509 const size_t SIZE = 256; 4510 char buffer[SIZE]; 4511 String8 result; 4512 4513 PlaybackThread::dumpInternals(fd, args); 4514 dprintf(fd, " Thread throttle time (msecs): %u\n", mThreadThrottleTimeMs); 4515 dprintf(fd, " AudioMixer tracks: 0x%08x\n", mAudioMixer->trackNames()); 4516 dprintf(fd, " Master mono: %s\n", mMasterMono ? "on" : "off"); 4517 4518 // Make a non-atomic copy of fast mixer dump state so it won't change underneath us 4519 // while we are dumping it. It may be inconsistent, but it won't mutate! 4520 // This is a large object so we place it on the heap. 4521 // FIXME 25972958: Need an intelligent copy constructor that does not touch unused pages. 4522 const FastMixerDumpState *copy = new FastMixerDumpState(mFastMixerDumpState); 4523 copy->dump(fd); 4524 delete copy; 4525 4526#ifdef STATE_QUEUE_DUMP 4527 // Similar for state queue 4528 StateQueueObserverDump observerCopy = mStateQueueObserverDump; 4529 observerCopy.dump(fd); 4530 StateQueueMutatorDump mutatorCopy = mStateQueueMutatorDump; 4531 mutatorCopy.dump(fd); 4532#endif 4533 4534#ifdef TEE_SINK 4535 // Write the tee output to a .wav file 4536 dumpTee(fd, mTeeSource, mId); 4537#endif 4538 4539#ifdef AUDIO_WATCHDOG 4540 if (mAudioWatchdog != 0) { 4541 // Make a non-atomic copy of audio watchdog dump so it won't change underneath us 4542 AudioWatchdogDump wdCopy = mAudioWatchdogDump; 4543 wdCopy.dump(fd); 4544 } 4545#endif 4546} 4547 4548uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const 4549{ 4550 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000) / 2; 4551} 4552 4553uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() const 4554{ 4555 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000); 4556} 4557 4558void AudioFlinger::MixerThread::cacheParameters_l() 4559{ 4560 PlaybackThread::cacheParameters_l(); 4561 4562 // FIXME: Relaxed timing because of a certain device that can't meet latency 4563 // Should be reduced to 2x after the vendor fixes the driver issue 4564 // increase threshold again due to low power audio mode. The way this warning 4565 // threshold is calculated and its usefulness should be reconsidered anyway. 4566 maxPeriod = seconds(mNormalFrameCount) / mSampleRate * 15; 4567} 4568 4569// ---------------------------------------------------------------------------- 4570 4571AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 4572 AudioStreamOut* output, audio_io_handle_t id, audio_devices_t device, bool systemReady) 4573 : PlaybackThread(audioFlinger, output, id, device, DIRECT, systemReady) 4574 // mLeftVolFloat, mRightVolFloat 4575{ 4576} 4577 4578AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 4579 AudioStreamOut* output, audio_io_handle_t id, uint32_t device, 4580 ThreadBase::type_t type, bool systemReady) 4581 : PlaybackThread(audioFlinger, output, id, device, type, systemReady) 4582 // mLeftVolFloat, mRightVolFloat 4583{ 4584} 4585 4586AudioFlinger::DirectOutputThread::~DirectOutputThread() 4587{ 4588} 4589 4590void AudioFlinger::DirectOutputThread::processVolume_l(Track *track, bool lastTrack) 4591{ 4592 audio_track_cblk_t* cblk = track->cblk(); 4593 float left, right; 4594 4595 if (mMasterMute || mStreamTypes[track->streamType()].mute) { 4596 left = right = 0; 4597 } else { 4598 float typeVolume = mStreamTypes[track->streamType()].volume; 4599 float v = mMasterVolume * typeVolume; 4600 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 4601 gain_minifloat_packed_t vlr = proxy->getVolumeLR(); 4602 left = float_from_gain(gain_minifloat_unpack_left(vlr)); 4603 if (left > GAIN_FLOAT_UNITY) { 4604 left = GAIN_FLOAT_UNITY; 4605 } 4606 left *= v; 4607 right = float_from_gain(gain_minifloat_unpack_right(vlr)); 4608 if (right > GAIN_FLOAT_UNITY) { 4609 right = GAIN_FLOAT_UNITY; 4610 } 4611 right *= v; 4612 } 4613 4614 if (lastTrack) { 4615 if (left != mLeftVolFloat || right != mRightVolFloat) { 4616 mLeftVolFloat = left; 4617 mRightVolFloat = right; 4618 4619 // Convert volumes from float to 8.24 4620 uint32_t vl = (uint32_t)(left * (1 << 24)); 4621 uint32_t vr = (uint32_t)(right * (1 << 24)); 4622 4623 // Delegate volume control to effect in track effect chain if needed 4624 // only one effect chain can be present on DirectOutputThread, so if 4625 // there is one, the track is connected to it 4626 if (!mEffectChains.isEmpty()) { 4627 mEffectChains[0]->setVolume_l(&vl, &vr); 4628 left = (float)vl / (1 << 24); 4629 right = (float)vr / (1 << 24); 4630 } 4631 if (mOutput->stream->set_volume) { 4632 mOutput->stream->set_volume(mOutput->stream, left, right); 4633 } 4634 } 4635 } 4636} 4637 4638void AudioFlinger::DirectOutputThread::onAddNewTrack_l() 4639{ 4640 sp<Track> previousTrack = mPreviousTrack.promote(); 4641 sp<Track> latestTrack = mLatestActiveTrack.promote(); 4642 4643 if (previousTrack != 0 && latestTrack != 0) { 4644 if (mType == DIRECT) { 4645 if (previousTrack.get() != latestTrack.get()) { 4646 mFlushPending = true; 4647 } 4648 } else /* mType == OFFLOAD */ { 4649 if (previousTrack->sessionId() != latestTrack->sessionId()) { 4650 mFlushPending = true; 4651 } 4652 } 4653 } 4654 PlaybackThread::onAddNewTrack_l(); 4655} 4656 4657AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l( 4658 Vector< sp<Track> > *tracksToRemove 4659) 4660{ 4661 size_t count = mActiveTracks.size(); 4662 mixer_state mixerStatus = MIXER_IDLE; 4663 bool doHwPause = false; 4664 bool doHwResume = false; 4665 4666 // find out which tracks need to be processed 4667 for (size_t i = 0; i < count; i++) { 4668 sp<Track> t = mActiveTracks[i].promote(); 4669 // The track died recently 4670 if (t == 0) { 4671 continue; 4672 } 4673 4674 if (t->isInvalid()) { 4675 ALOGW("An invalidated track shouldn't be in active list"); 4676 tracksToRemove->add(t); 4677 continue; 4678 } 4679 4680 Track* const track = t.get(); 4681 audio_track_cblk_t* cblk = track->cblk(); 4682 // Only consider last track started for volume and mixer state control. 4683 // In theory an older track could underrun and restart after the new one starts 4684 // but as we only care about the transition phase between two tracks on a 4685 // direct output, it is not a problem to ignore the underrun case. 4686 sp<Track> l = mLatestActiveTrack.promote(); 4687 bool last = l.get() == track; 4688 4689 if (track->isPausing()) { 4690 track->setPaused(); 4691 if (mHwSupportsPause && last && !mHwPaused) { 4692 doHwPause = true; 4693 mHwPaused = true; 4694 } 4695 tracksToRemove->add(track); 4696 } else if (track->isFlushPending()) { 4697 track->flushAck(); 4698 if (last) { 4699 mFlushPending = true; 4700 } 4701 } else if (track->isResumePending()) { 4702 track->resumeAck(); 4703 if (last && mHwPaused) { 4704 doHwResume = true; 4705 mHwPaused = false; 4706 } 4707 } 4708 4709 // The first time a track is added we wait 4710 // for all its buffers to be filled before processing it. 4711 // Allow draining the buffer in case the client 4712 // app does not call stop() and relies on underrun to stop: 4713 // hence the test on (track->mRetryCount > 1). 4714 // If retryCount<=1 then track is about to underrun and be removed. 4715 // Do not use a high threshold for compressed audio. 4716 uint32_t minFrames; 4717 if ((track->sharedBuffer() == 0) && !track->isStopping_1() && !track->isPausing() 4718 && (track->mRetryCount > 1) && audio_is_linear_pcm(mFormat)) { 4719 minFrames = mNormalFrameCount; 4720 } else { 4721 minFrames = 1; 4722 } 4723 4724 if ((track->framesReady() >= minFrames) && track->isReady() && !track->isPaused() && 4725 !track->isStopping_2() && !track->isStopped()) 4726 { 4727 ALOGVV("track %d s=%08x [OK]", track->name(), cblk->mServer); 4728 4729 if (track->mFillingUpStatus == Track::FS_FILLED) { 4730 track->mFillingUpStatus = Track::FS_ACTIVE; 4731 // make sure processVolume_l() will apply new volume even if 0 4732 mLeftVolFloat = mRightVolFloat = -1.0; 4733 if (!mHwSupportsPause) { 4734 track->resumeAck(); 4735 } 4736 } 4737 4738 // compute volume for this track 4739 processVolume_l(track, last); 4740 if (last) { 4741 sp<Track> previousTrack = mPreviousTrack.promote(); 4742 if (previousTrack != 0) { 4743 if (track != previousTrack.get()) { 4744 // Flush any data still being written from last track 4745 mBytesRemaining = 0; 4746 // Invalidate previous track to force a seek when resuming. 4747 previousTrack->invalidate(); 4748 } 4749 } 4750 mPreviousTrack = track; 4751 4752 // reset retry count 4753 track->mRetryCount = kMaxTrackRetriesDirect; 4754 mActiveTrack = t; 4755 mixerStatus = MIXER_TRACKS_READY; 4756 if (mHwPaused) { 4757 doHwResume = true; 4758 mHwPaused = false; 4759 } 4760 } 4761 } else { 4762 // clear effect chain input buffer if the last active track started underruns 4763 // to avoid sending previous audio buffer again to effects 4764 if (!mEffectChains.isEmpty() && last) { 4765 mEffectChains[0]->clearInputBuffer(); 4766 } 4767 if (track->isStopping_1()) { 4768 track->mState = TrackBase::STOPPING_2; 4769 if (last && mHwPaused) { 4770 doHwResume = true; 4771 mHwPaused = false; 4772 } 4773 } 4774 if ((track->sharedBuffer() != 0) || track->isStopped() || 4775 track->isStopping_2() || track->isPaused()) { 4776 // We have consumed all the buffers of this track. 4777 // Remove it from the list of active tracks. 4778 size_t audioHALFrames; 4779 if (audio_is_linear_pcm(mFormat)) { 4780 audioHALFrames = (latency_l() * mSampleRate) / 1000; 4781 } else { 4782 audioHALFrames = 0; 4783 } 4784 4785 size_t framesWritten = mBytesWritten / mFrameSize; 4786 if (mStandby || !last || 4787 track->presentationComplete(framesWritten, audioHALFrames)) { 4788 if (track->isStopping_2()) { 4789 track->mState = TrackBase::STOPPED; 4790 } 4791 if (track->isStopped()) { 4792 track->reset(); 4793 } 4794 tracksToRemove->add(track); 4795 } 4796 } else { 4797 // No buffers for this track. Give it a few chances to 4798 // fill a buffer, then remove it from active list. 4799 // Only consider last track started for mixer state control 4800 if (--(track->mRetryCount) <= 0) { 4801 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 4802 tracksToRemove->add(track); 4803 // indicate to client process that the track was disabled because of underrun; 4804 // it will then automatically call start() when data is available 4805 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 4806 } else if (last) { 4807 ALOGW("pause because of UNDERRUN, framesReady = %zu," 4808 "minFrames = %u, mFormat = %#x", 4809 track->framesReady(), minFrames, mFormat); 4810 mixerStatus = MIXER_TRACKS_ENABLED; 4811 if (mHwSupportsPause && !mHwPaused && !mStandby) { 4812 doHwPause = true; 4813 mHwPaused = true; 4814 } 4815 } 4816 } 4817 } 4818 } 4819 4820 // if an active track did not command a flush, check for pending flush on stopped tracks 4821 if (!mFlushPending) { 4822 for (size_t i = 0; i < mTracks.size(); i++) { 4823 if (mTracks[i]->isFlushPending()) { 4824 mTracks[i]->flushAck(); 4825 mFlushPending = true; 4826 } 4827 } 4828 } 4829 4830 // make sure the pause/flush/resume sequence is executed in the right order. 4831 // If a flush is pending and a track is active but the HW is not paused, force a HW pause 4832 // before flush and then resume HW. This can happen in case of pause/flush/resume 4833 // if resume is received before pause is executed. 4834 if (mHwSupportsPause && !mStandby && 4835 (doHwPause || (mFlushPending && !mHwPaused && (count != 0)))) { 4836 mOutput->stream->pause(mOutput->stream); 4837 } 4838 if (mFlushPending) { 4839 flushHw_l(); 4840 } 4841 if (mHwSupportsPause && !mStandby && doHwResume) { 4842 mOutput->stream->resume(mOutput->stream); 4843 } 4844 // remove all the tracks that need to be... 4845 removeTracks_l(*tracksToRemove); 4846 4847 return mixerStatus; 4848} 4849 4850void AudioFlinger::DirectOutputThread::threadLoop_mix() 4851{ 4852 size_t frameCount = mFrameCount; 4853 int8_t *curBuf = (int8_t *)mSinkBuffer; 4854 // output audio to hardware 4855 while (frameCount) { 4856 AudioBufferProvider::Buffer buffer; 4857 buffer.frameCount = frameCount; 4858 status_t status = mActiveTrack->getNextBuffer(&buffer); 4859 if (status != NO_ERROR || buffer.raw == NULL) { 4860 memset(curBuf, 0, frameCount * mFrameSize); 4861 break; 4862 } 4863 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 4864 frameCount -= buffer.frameCount; 4865 curBuf += buffer.frameCount * mFrameSize; 4866 mActiveTrack->releaseBuffer(&buffer); 4867 } 4868 mCurrentWriteLength = curBuf - (int8_t *)mSinkBuffer; 4869 mSleepTimeUs = 0; 4870 mStandbyTimeNs = systemTime() + mStandbyDelayNs; 4871 mActiveTrack.clear(); 4872} 4873 4874void AudioFlinger::DirectOutputThread::threadLoop_sleepTime() 4875{ 4876 // do not write to HAL when paused 4877 if (mHwPaused || (usesHwAvSync() && mStandby)) { 4878 mSleepTimeUs = mIdleSleepTimeUs; 4879 return; 4880 } 4881 if (mSleepTimeUs == 0) { 4882 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4883 mSleepTimeUs = mActiveSleepTimeUs; 4884 } else { 4885 mSleepTimeUs = mIdleSleepTimeUs; 4886 } 4887 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 4888 memset(mSinkBuffer, 0, mFrameCount * mFrameSize); 4889 mSleepTimeUs = 0; 4890 } 4891} 4892 4893void AudioFlinger::DirectOutputThread::threadLoop_exit() 4894{ 4895 { 4896 Mutex::Autolock _l(mLock); 4897 for (size_t i = 0; i < mTracks.size(); i++) { 4898 if (mTracks[i]->isFlushPending()) { 4899 mTracks[i]->flushAck(); 4900 mFlushPending = true; 4901 } 4902 } 4903 if (mFlushPending) { 4904 flushHw_l(); 4905 } 4906 } 4907 PlaybackThread::threadLoop_exit(); 4908} 4909 4910// must be called with thread mutex locked 4911bool AudioFlinger::DirectOutputThread::shouldStandby_l() 4912{ 4913 bool trackPaused = false; 4914 bool trackStopped = false; 4915 4916 // do not put the HAL in standby when paused. AwesomePlayer clear the offloaded AudioTrack 4917 // after a timeout and we will enter standby then. 4918 if (mTracks.size() > 0) { 4919 trackPaused = mTracks[mTracks.size() - 1]->isPaused(); 4920 trackStopped = mTracks[mTracks.size() - 1]->isStopped() || 4921 mTracks[mTracks.size() - 1]->mState == TrackBase::IDLE; 4922 } 4923 4924 return !mStandby && !(trackPaused || (mHwPaused && !trackStopped)); 4925} 4926 4927// getTrackName_l() must be called with ThreadBase::mLock held 4928int AudioFlinger::DirectOutputThread::getTrackName_l(audio_channel_mask_t channelMask __unused, 4929 audio_format_t format __unused, int sessionId __unused) 4930{ 4931 return 0; 4932} 4933 4934// deleteTrackName_l() must be called with ThreadBase::mLock held 4935void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name __unused) 4936{ 4937} 4938 4939// checkForNewParameter_l() must be called with ThreadBase::mLock held 4940bool AudioFlinger::DirectOutputThread::checkForNewParameter_l(const String8& keyValuePair, 4941 status_t& status) 4942{ 4943 bool reconfig = false; 4944 bool a2dpDeviceChanged = false; 4945 4946 status = NO_ERROR; 4947 4948 AudioParameter param = AudioParameter(keyValuePair); 4949 int value; 4950 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 4951 // forward device change to effects that have requested to be 4952 // aware of attached audio device. 4953 if (value != AUDIO_DEVICE_NONE) { 4954 a2dpDeviceChanged = 4955 (mOutDevice & AUDIO_DEVICE_OUT_ALL_A2DP) != (value & AUDIO_DEVICE_OUT_ALL_A2DP); 4956 mOutDevice = value; 4957 for (size_t i = 0; i < mEffectChains.size(); i++) { 4958 mEffectChains[i]->setDevice_l(mOutDevice); 4959 } 4960 } 4961 } 4962 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 4963 // do not accept frame count changes if tracks are open as the track buffer 4964 // size depends on frame count and correct behavior would not be garantied 4965 // if frame count is changed after track creation 4966 if (!mTracks.isEmpty()) { 4967 status = INVALID_OPERATION; 4968 } else { 4969 reconfig = true; 4970 } 4971 } 4972 if (status == NO_ERROR) { 4973 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 4974 keyValuePair.string()); 4975 if (!mStandby && status == INVALID_OPERATION) { 4976 mOutput->standby(); 4977 mStandby = true; 4978 mBytesWritten = 0; 4979 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 4980 keyValuePair.string()); 4981 } 4982 if (status == NO_ERROR && reconfig) { 4983 readOutputParameters_l(); 4984 sendIoConfigEvent_l(AUDIO_OUTPUT_CONFIG_CHANGED); 4985 } 4986 } 4987 4988 return reconfig || a2dpDeviceChanged; 4989} 4990 4991uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() const 4992{ 4993 uint32_t time; 4994 if (audio_is_linear_pcm(mFormat)) { 4995 time = PlaybackThread::activeSleepTimeUs(); 4996 } else { 4997 time = 10000; 4998 } 4999 return time; 5000} 5001 5002uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() const 5003{ 5004 uint32_t time; 5005 if (audio_is_linear_pcm(mFormat)) { 5006 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 5007 } else { 5008 time = 10000; 5009 } 5010 return time; 5011} 5012 5013uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() const 5014{ 5015 uint32_t time; 5016 if (audio_is_linear_pcm(mFormat)) { 5017 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 5018 } else { 5019 time = 10000; 5020 } 5021 return time; 5022} 5023 5024void AudioFlinger::DirectOutputThread::cacheParameters_l() 5025{ 5026 PlaybackThread::cacheParameters_l(); 5027 5028 // use shorter standby delay as on normal output to release 5029 // hardware resources as soon as possible 5030 // no delay on outputs with HW A/V sync 5031 if (usesHwAvSync()) { 5032 mStandbyDelayNs = 0; 5033 } else if ((mType == OFFLOAD) && !audio_is_linear_pcm(mFormat)) { 5034 mStandbyDelayNs = kOffloadStandbyDelayNs; 5035 } else { 5036 mStandbyDelayNs = microseconds(mActiveSleepTimeUs*2); 5037 } 5038} 5039 5040void AudioFlinger::DirectOutputThread::flushHw_l() 5041{ 5042 mOutput->flush(); 5043 mHwPaused = false; 5044 mFlushPending = false; 5045} 5046 5047// ---------------------------------------------------------------------------- 5048 5049AudioFlinger::AsyncCallbackThread::AsyncCallbackThread( 5050 const wp<AudioFlinger::PlaybackThread>& playbackThread) 5051 : Thread(false /*canCallJava*/), 5052 mPlaybackThread(playbackThread), 5053 mWriteAckSequence(0), 5054 mDrainSequence(0) 5055{ 5056} 5057 5058AudioFlinger::AsyncCallbackThread::~AsyncCallbackThread() 5059{ 5060} 5061 5062void AudioFlinger::AsyncCallbackThread::onFirstRef() 5063{ 5064 run("Offload Cbk", ANDROID_PRIORITY_URGENT_AUDIO); 5065} 5066 5067bool AudioFlinger::AsyncCallbackThread::threadLoop() 5068{ 5069 while (!exitPending()) { 5070 uint32_t writeAckSequence; 5071 uint32_t drainSequence; 5072 5073 { 5074 Mutex::Autolock _l(mLock); 5075 while (!((mWriteAckSequence & 1) || 5076 (mDrainSequence & 1) || 5077 exitPending())) { 5078 mWaitWorkCV.wait(mLock); 5079 } 5080 5081 if (exitPending()) { 5082 break; 5083 } 5084 ALOGV("AsyncCallbackThread mWriteAckSequence %d mDrainSequence %d", 5085 mWriteAckSequence, mDrainSequence); 5086 writeAckSequence = mWriteAckSequence; 5087 mWriteAckSequence &= ~1; 5088 drainSequence = mDrainSequence; 5089 mDrainSequence &= ~1; 5090 } 5091 { 5092 sp<AudioFlinger::PlaybackThread> playbackThread = mPlaybackThread.promote(); 5093 if (playbackThread != 0) { 5094 if (writeAckSequence & 1) { 5095 playbackThread->resetWriteBlocked(writeAckSequence >> 1); 5096 } 5097 if (drainSequence & 1) { 5098 playbackThread->resetDraining(drainSequence >> 1); 5099 } 5100 } 5101 } 5102 } 5103 return false; 5104} 5105 5106void AudioFlinger::AsyncCallbackThread::exit() 5107{ 5108 ALOGV("AsyncCallbackThread::exit"); 5109 Mutex::Autolock _l(mLock); 5110 requestExit(); 5111 mWaitWorkCV.broadcast(); 5112} 5113 5114void AudioFlinger::AsyncCallbackThread::setWriteBlocked(uint32_t sequence) 5115{ 5116 Mutex::Autolock _l(mLock); 5117 // bit 0 is cleared 5118 mWriteAckSequence = sequence << 1; 5119} 5120 5121void AudioFlinger::AsyncCallbackThread::resetWriteBlocked() 5122{ 5123 Mutex::Autolock _l(mLock); 5124 // ignore unexpected callbacks 5125 if (mWriteAckSequence & 2) { 5126 mWriteAckSequence |= 1; 5127 mWaitWorkCV.signal(); 5128 } 5129} 5130 5131void AudioFlinger::AsyncCallbackThread::setDraining(uint32_t sequence) 5132{ 5133 Mutex::Autolock _l(mLock); 5134 // bit 0 is cleared 5135 mDrainSequence = sequence << 1; 5136} 5137 5138void AudioFlinger::AsyncCallbackThread::resetDraining() 5139{ 5140 Mutex::Autolock _l(mLock); 5141 // ignore unexpected callbacks 5142 if (mDrainSequence & 2) { 5143 mDrainSequence |= 1; 5144 mWaitWorkCV.signal(); 5145 } 5146} 5147 5148 5149// ---------------------------------------------------------------------------- 5150AudioFlinger::OffloadThread::OffloadThread(const sp<AudioFlinger>& audioFlinger, 5151 AudioStreamOut* output, audio_io_handle_t id, uint32_t device, bool systemReady) 5152 : DirectOutputThread(audioFlinger, output, id, device, OFFLOAD, systemReady), 5153 mPausedBytesRemaining(0) 5154{ 5155 //FIXME: mStandby should be set to true by ThreadBase constructor 5156 mStandby = true; 5157} 5158 5159void AudioFlinger::OffloadThread::threadLoop_exit() 5160{ 5161 if (mFlushPending || mHwPaused) { 5162 // If a flush is pending or track was paused, just discard buffered data 5163 flushHw_l(); 5164 } else { 5165 mMixerStatus = MIXER_DRAIN_ALL; 5166 threadLoop_drain(); 5167 } 5168 if (mUseAsyncWrite) { 5169 ALOG_ASSERT(mCallbackThread != 0); 5170 mCallbackThread->exit(); 5171 } 5172 PlaybackThread::threadLoop_exit(); 5173} 5174 5175AudioFlinger::PlaybackThread::mixer_state AudioFlinger::OffloadThread::prepareTracks_l( 5176 Vector< sp<Track> > *tracksToRemove 5177) 5178{ 5179 size_t count = mActiveTracks.size(); 5180 5181 mixer_state mixerStatus = MIXER_IDLE; 5182 bool doHwPause = false; 5183 bool doHwResume = false; 5184 5185 ALOGV("OffloadThread::prepareTracks_l active tracks %d", count); 5186 5187 // find out which tracks need to be processed 5188 for (size_t i = 0; i < count; i++) { 5189 sp<Track> t = mActiveTracks[i].promote(); 5190 // The track died recently 5191 if (t == 0) { 5192 continue; 5193 } 5194 Track* const track = t.get(); 5195 audio_track_cblk_t* cblk = track->cblk(); 5196 // Only consider last track started for volume and mixer state control. 5197 // In theory an older track could underrun and restart after the new one starts 5198 // but as we only care about the transition phase between two tracks on a 5199 // direct output, it is not a problem to ignore the underrun case. 5200 sp<Track> l = mLatestActiveTrack.promote(); 5201 bool last = l.get() == track; 5202 5203 if (track->isInvalid()) { 5204 ALOGW("An invalidated track shouldn't be in active list"); 5205 tracksToRemove->add(track); 5206 continue; 5207 } 5208 5209 if (track->mState == TrackBase::IDLE) { 5210 ALOGW("An idle track shouldn't be in active list"); 5211 continue; 5212 } 5213 5214 if (track->isPausing()) { 5215 track->setPaused(); 5216 if (last) { 5217 if (mHwSupportsPause && !mHwPaused) { 5218 doHwPause = true; 5219 mHwPaused = true; 5220 } 5221 // If we were part way through writing the mixbuffer to 5222 // the HAL we must save this until we resume 5223 // BUG - this will be wrong if a different track is made active, 5224 // in that case we want to discard the pending data in the 5225 // mixbuffer and tell the client to present it again when the 5226 // track is resumed 5227 mPausedWriteLength = mCurrentWriteLength; 5228 mPausedBytesRemaining = mBytesRemaining; 5229 mBytesRemaining = 0; // stop writing 5230 } 5231 tracksToRemove->add(track); 5232 } else if (track->isFlushPending()) { 5233 track->flushAck(); 5234 if (last) { 5235 mFlushPending = true; 5236 } 5237 } else if (track->isResumePending()){ 5238 track->resumeAck(); 5239 if (last) { 5240 if (mPausedBytesRemaining) { 5241 // Need to continue write that was interrupted 5242 mCurrentWriteLength = mPausedWriteLength; 5243 mBytesRemaining = mPausedBytesRemaining; 5244 mPausedBytesRemaining = 0; 5245 } 5246 if (mHwPaused) { 5247 doHwResume = true; 5248 mHwPaused = false; 5249 // threadLoop_mix() will handle the case that we need to 5250 // resume an interrupted write 5251 } 5252 // enable write to audio HAL 5253 mSleepTimeUs = 0; 5254 5255 // Do not handle new data in this iteration even if track->framesReady() 5256 mixerStatus = MIXER_TRACKS_ENABLED; 5257 } 5258 } else if (track->framesReady() && track->isReady() && 5259 !track->isPaused() && !track->isTerminated() && !track->isStopping_2()) { 5260 ALOGVV("OffloadThread: track %d s=%08x [OK]", track->name(), cblk->mServer); 5261 if (track->mFillingUpStatus == Track::FS_FILLED) { 5262 track->mFillingUpStatus = Track::FS_ACTIVE; 5263 // make sure processVolume_l() will apply new volume even if 0 5264 mLeftVolFloat = mRightVolFloat = -1.0; 5265 } 5266 5267 if (last) { 5268 sp<Track> previousTrack = mPreviousTrack.promote(); 5269 if (previousTrack != 0) { 5270 if (track != previousTrack.get()) { 5271 // Flush any data still being written from last track 5272 mBytesRemaining = 0; 5273 if (mPausedBytesRemaining) { 5274 // Last track was paused so we also need to flush saved 5275 // mixbuffer state and invalidate track so that it will 5276 // re-submit that unwritten data when it is next resumed 5277 mPausedBytesRemaining = 0; 5278 // Invalidate is a bit drastic - would be more efficient 5279 // to have a flag to tell client that some of the 5280 // previously written data was lost 5281 previousTrack->invalidate(); 5282 } 5283 // flush data already sent to the DSP if changing audio session as audio 5284 // comes from a different source. Also invalidate previous track to force a 5285 // seek when resuming. 5286 if (previousTrack->sessionId() != track->sessionId()) { 5287 previousTrack->invalidate(); 5288 } 5289 } 5290 } 5291 mPreviousTrack = track; 5292 // reset retry count 5293 track->mRetryCount = kMaxTrackRetriesOffload; 5294 mActiveTrack = t; 5295 mixerStatus = MIXER_TRACKS_READY; 5296 } 5297 } else { 5298 ALOGVV("OffloadThread: track %d s=%08x [NOT READY]", track->name(), cblk->mServer); 5299 if (track->isStopping_1()) { 5300 // Hardware buffer can hold a large amount of audio so we must 5301 // wait for all current track's data to drain before we say 5302 // that the track is stopped. 5303 if (mBytesRemaining == 0) { 5304 // Only start draining when all data in mixbuffer 5305 // has been written 5306 ALOGV("OffloadThread: underrun and STOPPING_1 -> draining, STOPPING_2"); 5307 track->mState = TrackBase::STOPPING_2; // so presentation completes after drain 5308 // do not drain if no data was ever sent to HAL (mStandby == true) 5309 if (last && !mStandby) { 5310 // do not modify drain sequence if we are already draining. This happens 5311 // when resuming from pause after drain. 5312 if ((mDrainSequence & 1) == 0) { 5313 mSleepTimeUs = 0; 5314 mStandbyTimeNs = systemTime() + mStandbyDelayNs; 5315 mixerStatus = MIXER_DRAIN_TRACK; 5316 mDrainSequence += 2; 5317 } 5318 if (mHwPaused) { 5319 // It is possible to move from PAUSED to STOPPING_1 without 5320 // a resume so we must ensure hardware is running 5321 doHwResume = true; 5322 mHwPaused = false; 5323 } 5324 } 5325 } 5326 } else if (track->isStopping_2()) { 5327 // Drain has completed or we are in standby, signal presentation complete 5328 if (!(mDrainSequence & 1) || !last || mStandby) { 5329 track->mState = TrackBase::STOPPED; 5330 size_t audioHALFrames = 5331 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 5332 size_t framesWritten = 5333 mBytesWritten / mOutput->getFrameSize(); 5334 track->presentationComplete(framesWritten, audioHALFrames); 5335 track->reset(); 5336 tracksToRemove->add(track); 5337 } 5338 } else { 5339 // No buffers for this track. Give it a few chances to 5340 // fill a buffer, then remove it from active list. 5341 if (--(track->mRetryCount) <= 0) { 5342 ALOGV("OffloadThread: BUFFER TIMEOUT: remove(%d) from active list", 5343 track->name()); 5344 tracksToRemove->add(track); 5345 // indicate to client process that the track was disabled because of underrun; 5346 // it will then automatically call start() when data is available 5347 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 5348 } else if (last){ 5349 mixerStatus = MIXER_TRACKS_ENABLED; 5350 } 5351 } 5352 } 5353 // compute volume for this track 5354 processVolume_l(track, last); 5355 } 5356 5357 // make sure the pause/flush/resume sequence is executed in the right order. 5358 // If a flush is pending and a track is active but the HW is not paused, force a HW pause 5359 // before flush and then resume HW. This can happen in case of pause/flush/resume 5360 // if resume is received before pause is executed. 5361 if (!mStandby && (doHwPause || (mFlushPending && !mHwPaused && (count != 0)))) { 5362 mOutput->stream->pause(mOutput->stream); 5363 } 5364 if (mFlushPending) { 5365 flushHw_l(); 5366 } 5367 if (!mStandby && doHwResume) { 5368 mOutput->stream->resume(mOutput->stream); 5369 } 5370 5371 // remove all the tracks that need to be... 5372 removeTracks_l(*tracksToRemove); 5373 5374 return mixerStatus; 5375} 5376 5377// must be called with thread mutex locked 5378bool AudioFlinger::OffloadThread::waitingAsyncCallback_l() 5379{ 5380 ALOGVV("waitingAsyncCallback_l mWriteAckSequence %d mDrainSequence %d", 5381 mWriteAckSequence, mDrainSequence); 5382 if (mUseAsyncWrite && ((mWriteAckSequence & 1) || (mDrainSequence & 1))) { 5383 return true; 5384 } 5385 return false; 5386} 5387 5388bool AudioFlinger::OffloadThread::waitingAsyncCallback() 5389{ 5390 Mutex::Autolock _l(mLock); 5391 return waitingAsyncCallback_l(); 5392} 5393 5394void AudioFlinger::OffloadThread::flushHw_l() 5395{ 5396 DirectOutputThread::flushHw_l(); 5397 // Flush anything still waiting in the mixbuffer 5398 mCurrentWriteLength = 0; 5399 mBytesRemaining = 0; 5400 mPausedWriteLength = 0; 5401 mPausedBytesRemaining = 0; 5402 5403 if (mUseAsyncWrite) { 5404 // discard any pending drain or write ack by incrementing sequence 5405 mWriteAckSequence = (mWriteAckSequence + 2) & ~1; 5406 mDrainSequence = (mDrainSequence + 2) & ~1; 5407 ALOG_ASSERT(mCallbackThread != 0); 5408 mCallbackThread->setWriteBlocked(mWriteAckSequence); 5409 mCallbackThread->setDraining(mDrainSequence); 5410 } 5411} 5412 5413// ---------------------------------------------------------------------------- 5414 5415AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 5416 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id, bool systemReady) 5417 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->outDevice(), 5418 systemReady, DUPLICATING), 5419 mWaitTimeMs(UINT_MAX) 5420{ 5421 addOutputTrack(mainThread); 5422} 5423 5424AudioFlinger::DuplicatingThread::~DuplicatingThread() 5425{ 5426 for (size_t i = 0; i < mOutputTracks.size(); i++) { 5427 mOutputTracks[i]->destroy(); 5428 } 5429} 5430 5431void AudioFlinger::DuplicatingThread::threadLoop_mix() 5432{ 5433 // mix buffers... 5434 if (outputsReady(outputTracks)) { 5435 mAudioMixer->process(); 5436 } else { 5437 if (mMixerBufferValid) { 5438 memset(mMixerBuffer, 0, mMixerBufferSize); 5439 } else { 5440 memset(mSinkBuffer, 0, mSinkBufferSize); 5441 } 5442 } 5443 mSleepTimeUs = 0; 5444 writeFrames = mNormalFrameCount; 5445 mCurrentWriteLength = mSinkBufferSize; 5446 mStandbyTimeNs = systemTime() + mStandbyDelayNs; 5447} 5448 5449void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() 5450{ 5451 if (mSleepTimeUs == 0) { 5452 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 5453 mSleepTimeUs = mActiveSleepTimeUs; 5454 } else { 5455 mSleepTimeUs = mIdleSleepTimeUs; 5456 } 5457 } else if (mBytesWritten != 0) { 5458 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 5459 writeFrames = mNormalFrameCount; 5460 memset(mSinkBuffer, 0, mSinkBufferSize); 5461 } else { 5462 // flush remaining overflow buffers in output tracks 5463 writeFrames = 0; 5464 } 5465 mSleepTimeUs = 0; 5466 } 5467} 5468 5469ssize_t AudioFlinger::DuplicatingThread::threadLoop_write() 5470{ 5471 for (size_t i = 0; i < outputTracks.size(); i++) { 5472 outputTracks[i]->write(mSinkBuffer, writeFrames); 5473 } 5474 mStandby = false; 5475 return (ssize_t)mSinkBufferSize; 5476} 5477 5478void AudioFlinger::DuplicatingThread::threadLoop_standby() 5479{ 5480 // DuplicatingThread implements standby by stopping all tracks 5481 for (size_t i = 0; i < outputTracks.size(); i++) { 5482 outputTracks[i]->stop(); 5483 } 5484} 5485 5486void AudioFlinger::DuplicatingThread::saveOutputTracks() 5487{ 5488 outputTracks = mOutputTracks; 5489} 5490 5491void AudioFlinger::DuplicatingThread::clearOutputTracks() 5492{ 5493 outputTracks.clear(); 5494} 5495 5496void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 5497{ 5498 Mutex::Autolock _l(mLock); 5499 // The downstream MixerThread consumes thread->frameCount() amount of frames per mix pass. 5500 // Adjust for thread->sampleRate() to determine minimum buffer frame count. 5501 // Then triple buffer because Threads do not run synchronously and may not be clock locked. 5502 const size_t frameCount = 5503 3 * sourceFramesNeeded(mSampleRate, thread->frameCount(), thread->sampleRate()); 5504 // TODO: Consider asynchronous sample rate conversion to handle clock disparity 5505 // from different OutputTracks and their associated MixerThreads (e.g. one may 5506 // nearly empty and the other may be dropping data). 5507 5508 sp<OutputTrack> outputTrack = new OutputTrack(thread, 5509 this, 5510 mSampleRate, 5511 mFormat, 5512 mChannelMask, 5513 frameCount, 5514 IPCThreadState::self()->getCallingUid()); 5515 if (outputTrack->cblk() != NULL) { 5516 thread->setStreamVolume(AUDIO_STREAM_PATCH, 1.0f); 5517 mOutputTracks.add(outputTrack); 5518 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack.get(), thread); 5519 updateWaitTime_l(); 5520 } 5521} 5522 5523void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 5524{ 5525 Mutex::Autolock _l(mLock); 5526 for (size_t i = 0; i < mOutputTracks.size(); i++) { 5527 if (mOutputTracks[i]->thread() == thread) { 5528 mOutputTracks[i]->destroy(); 5529 mOutputTracks.removeAt(i); 5530 updateWaitTime_l(); 5531 if (thread->getOutput() == mOutput) { 5532 mOutput = NULL; 5533 } 5534 return; 5535 } 5536 } 5537 ALOGV("removeOutputTrack(): unknown thread: %p", thread); 5538} 5539 5540// caller must hold mLock 5541void AudioFlinger::DuplicatingThread::updateWaitTime_l() 5542{ 5543 mWaitTimeMs = UINT_MAX; 5544 for (size_t i = 0; i < mOutputTracks.size(); i++) { 5545 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 5546 if (strong != 0) { 5547 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 5548 if (waitTimeMs < mWaitTimeMs) { 5549 mWaitTimeMs = waitTimeMs; 5550 } 5551 } 5552 } 5553} 5554 5555 5556bool AudioFlinger::DuplicatingThread::outputsReady( 5557 const SortedVector< sp<OutputTrack> > &outputTracks) 5558{ 5559 for (size_t i = 0; i < outputTracks.size(); i++) { 5560 sp<ThreadBase> thread = outputTracks[i]->thread().promote(); 5561 if (thread == 0) { 5562 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", 5563 outputTracks[i].get()); 5564 return false; 5565 } 5566 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 5567 // see note at standby() declaration 5568 if (playbackThread->standby() && !playbackThread->isSuspended()) { 5569 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), 5570 thread.get()); 5571 return false; 5572 } 5573 } 5574 return true; 5575} 5576 5577uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() const 5578{ 5579 return (mWaitTimeMs * 1000) / 2; 5580} 5581 5582void AudioFlinger::DuplicatingThread::cacheParameters_l() 5583{ 5584 // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first 5585 updateWaitTime_l(); 5586 5587 MixerThread::cacheParameters_l(); 5588} 5589 5590// ---------------------------------------------------------------------------- 5591// Record 5592// ---------------------------------------------------------------------------- 5593 5594AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 5595 AudioStreamIn *input, 5596 audio_io_handle_t id, 5597 audio_devices_t outDevice, 5598 audio_devices_t inDevice, 5599 bool systemReady 5600#ifdef TEE_SINK 5601 , const sp<NBAIO_Sink>& teeSink 5602#endif 5603 ) : 5604 ThreadBase(audioFlinger, id, outDevice, inDevice, RECORD, systemReady), 5605 mInput(input), mActiveTracksGen(0), mRsmpInBuffer(NULL), 5606 // mRsmpInFrames and mRsmpInFramesP2 are set by readInputParameters_l() 5607 mRsmpInRear(0) 5608#ifdef TEE_SINK 5609 , mTeeSink(teeSink) 5610#endif 5611 , mReadOnlyHeap(new MemoryDealer(kRecordThreadReadOnlyHeapSize, 5612 "RecordThreadRO", MemoryHeapBase::READ_ONLY)) 5613 // mFastCapture below 5614 , mFastCaptureFutex(0) 5615 // mInputSource 5616 // mPipeSink 5617 // mPipeSource 5618 , mPipeFramesP2(0) 5619 // mPipeMemory 5620 // mFastCaptureNBLogWriter 5621 , mFastTrackAvail(false) 5622{ 5623 snprintf(mThreadName, kThreadNameLength, "AudioIn_%X", id); 5624 mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mThreadName); 5625 5626 readInputParameters_l(); 5627 5628 // create an NBAIO source for the HAL input stream, and negotiate 5629 mInputSource = new AudioStreamInSource(input->stream); 5630 size_t numCounterOffers = 0; 5631 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount, mFormat)}; 5632 ssize_t index = mInputSource->negotiate(offers, 1, NULL, numCounterOffers); 5633 ALOG_ASSERT(index == 0); 5634 5635 // initialize fast capture depending on configuration 5636 bool initFastCapture; 5637 switch (kUseFastCapture) { 5638 case FastCapture_Never: 5639 initFastCapture = false; 5640 break; 5641 case FastCapture_Always: 5642 initFastCapture = true; 5643 break; 5644 case FastCapture_Static: 5645 initFastCapture = (mFrameCount * 1000) / mSampleRate < kMinNormalCaptureBufferSizeMs; 5646 break; 5647 // case FastCapture_Dynamic: 5648 } 5649 5650 if (initFastCapture) { 5651 // create a Pipe for FastCapture to write to, and for us and fast tracks to read from 5652 NBAIO_Format format = mInputSource->format(); 5653 size_t pipeFramesP2 = roundup(mSampleRate / 25); // double-buffering of 20 ms each 5654 size_t pipeSize = pipeFramesP2 * Format_frameSize(format); 5655 void *pipeBuffer; 5656 const sp<MemoryDealer> roHeap(readOnlyHeap()); 5657 sp<IMemory> pipeMemory; 5658 if ((roHeap == 0) || 5659 (pipeMemory = roHeap->allocate(pipeSize)) == 0 || 5660 (pipeBuffer = pipeMemory->pointer()) == NULL) { 5661 ALOGE("not enough memory for pipe buffer size=%zu", pipeSize); 5662 goto failed; 5663 } 5664 // pipe will be shared directly with fast clients, so clear to avoid leaking old information 5665 memset(pipeBuffer, 0, pipeSize); 5666 Pipe *pipe = new Pipe(pipeFramesP2, format, pipeBuffer); 5667 const NBAIO_Format offers[1] = {format}; 5668 size_t numCounterOffers = 0; 5669 ssize_t index = pipe->negotiate(offers, 1, NULL, numCounterOffers); 5670 ALOG_ASSERT(index == 0); 5671 mPipeSink = pipe; 5672 PipeReader *pipeReader = new PipeReader(*pipe); 5673 numCounterOffers = 0; 5674 index = pipeReader->negotiate(offers, 1, NULL, numCounterOffers); 5675 ALOG_ASSERT(index == 0); 5676 mPipeSource = pipeReader; 5677 mPipeFramesP2 = pipeFramesP2; 5678 mPipeMemory = pipeMemory; 5679 5680 // create fast capture 5681 mFastCapture = new FastCapture(); 5682 FastCaptureStateQueue *sq = mFastCapture->sq(); 5683#ifdef STATE_QUEUE_DUMP 5684 // FIXME 5685#endif 5686 FastCaptureState *state = sq->begin(); 5687 state->mCblk = NULL; 5688 state->mInputSource = mInputSource.get(); 5689 state->mInputSourceGen++; 5690 state->mPipeSink = pipe; 5691 state->mPipeSinkGen++; 5692 state->mFrameCount = mFrameCount; 5693 state->mCommand = FastCaptureState::COLD_IDLE; 5694 // already done in constructor initialization list 5695 //mFastCaptureFutex = 0; 5696 state->mColdFutexAddr = &mFastCaptureFutex; 5697 state->mColdGen++; 5698 state->mDumpState = &mFastCaptureDumpState; 5699#ifdef TEE_SINK 5700 // FIXME 5701#endif 5702 mFastCaptureNBLogWriter = audioFlinger->newWriter_l(kFastCaptureLogSize, "FastCapture"); 5703 state->mNBLogWriter = mFastCaptureNBLogWriter.get(); 5704 sq->end(); 5705 sq->push(FastCaptureStateQueue::BLOCK_UNTIL_PUSHED); 5706 5707 // start the fast capture 5708 mFastCapture->run("FastCapture", ANDROID_PRIORITY_URGENT_AUDIO); 5709 pid_t tid = mFastCapture->getTid(); 5710 sendPrioConfigEvent(getpid_cached, tid, kPriorityFastMixer); 5711#ifdef AUDIO_WATCHDOG 5712 // FIXME 5713#endif 5714 5715 mFastTrackAvail = true; 5716 } 5717failed: ; 5718 5719 // FIXME mNormalSource 5720} 5721 5722AudioFlinger::RecordThread::~RecordThread() 5723{ 5724 if (mFastCapture != 0) { 5725 FastCaptureStateQueue *sq = mFastCapture->sq(); 5726 FastCaptureState *state = sq->begin(); 5727 if (state->mCommand == FastCaptureState::COLD_IDLE) { 5728 int32_t old = android_atomic_inc(&mFastCaptureFutex); 5729 if (old == -1) { 5730 (void) syscall(__NR_futex, &mFastCaptureFutex, FUTEX_WAKE_PRIVATE, 1); 5731 } 5732 } 5733 state->mCommand = FastCaptureState::EXIT; 5734 sq->end(); 5735 sq->push(FastCaptureStateQueue::BLOCK_UNTIL_PUSHED); 5736 mFastCapture->join(); 5737 mFastCapture.clear(); 5738 } 5739 mAudioFlinger->unregisterWriter(mFastCaptureNBLogWriter); 5740 mAudioFlinger->unregisterWriter(mNBLogWriter); 5741 free(mRsmpInBuffer); 5742} 5743 5744void AudioFlinger::RecordThread::onFirstRef() 5745{ 5746 run(mThreadName, PRIORITY_URGENT_AUDIO); 5747} 5748 5749bool AudioFlinger::RecordThread::threadLoop() 5750{ 5751 nsecs_t lastWarning = 0; 5752 5753 inputStandBy(); 5754 5755reacquire_wakelock: 5756 sp<RecordTrack> activeTrack; 5757 int activeTracksGen; 5758 { 5759 Mutex::Autolock _l(mLock); 5760 size_t size = mActiveTracks.size(); 5761 activeTracksGen = mActiveTracksGen; 5762 if (size > 0) { 5763 // FIXME an arbitrary choice 5764 activeTrack = mActiveTracks[0]; 5765 acquireWakeLock_l(activeTrack->uid()); 5766 if (size > 1) { 5767 SortedVector<int> tmp; 5768 for (size_t i = 0; i < size; i++) { 5769 tmp.add(mActiveTracks[i]->uid()); 5770 } 5771 updateWakeLockUids_l(tmp); 5772 } 5773 } else { 5774 acquireWakeLock_l(-1); 5775 } 5776 } 5777 5778 mTimestamp.mTimebaseOffset[ExtendedTimestamp::TIMEBASE_BOOTTIME] = 5779 gBoottime.getBoottimeOffset(); 5780 5781 // used to request a deferred sleep, to be executed later while mutex is unlocked 5782 uint32_t sleepUs = 0; 5783 5784 // loop while there is work to do 5785 for (;;) { 5786 Vector< sp<EffectChain> > effectChains; 5787 5788 // sleep with mutex unlocked 5789 if (sleepUs > 0) { 5790 ATRACE_BEGIN("sleep"); 5791 usleep(sleepUs); 5792 ATRACE_END(); 5793 sleepUs = 0; 5794 } 5795 5796 // activeTracks accumulates a copy of a subset of mActiveTracks 5797 Vector< sp<RecordTrack> > activeTracks; 5798 5799 // reference to the (first and only) active fast track 5800 sp<RecordTrack> fastTrack; 5801 5802 // reference to a fast track which is about to be removed 5803 sp<RecordTrack> fastTrackToRemove; 5804 5805 { // scope for mLock 5806 Mutex::Autolock _l(mLock); 5807 5808 processConfigEvents_l(); 5809 5810 // check exitPending here because checkForNewParameters_l() and 5811 // checkForNewParameters_l() can temporarily release mLock 5812 if (exitPending()) { 5813 break; 5814 } 5815 5816 // if no active track(s), then standby and release wakelock 5817 size_t size = mActiveTracks.size(); 5818 if (size == 0) { 5819 standbyIfNotAlreadyInStandby(); 5820 // exitPending() can't become true here 5821 releaseWakeLock_l(); 5822 ALOGV("RecordThread: loop stopping"); 5823 // go to sleep 5824 mWaitWorkCV.wait(mLock); 5825 ALOGV("RecordThread: loop starting"); 5826 goto reacquire_wakelock; 5827 } 5828 5829 if (mActiveTracksGen != activeTracksGen) { 5830 activeTracksGen = mActiveTracksGen; 5831 SortedVector<int> tmp; 5832 for (size_t i = 0; i < size; i++) { 5833 tmp.add(mActiveTracks[i]->uid()); 5834 } 5835 updateWakeLockUids_l(tmp); 5836 } 5837 5838 bool doBroadcast = false; 5839 for (size_t i = 0; i < size; ) { 5840 5841 activeTrack = mActiveTracks[i]; 5842 if (activeTrack->isTerminated()) { 5843 if (activeTrack->isFastTrack()) { 5844 ALOG_ASSERT(fastTrackToRemove == 0); 5845 fastTrackToRemove = activeTrack; 5846 } 5847 removeTrack_l(activeTrack); 5848 mActiveTracks.remove(activeTrack); 5849 mActiveTracksGen++; 5850 size--; 5851 continue; 5852 } 5853 5854 TrackBase::track_state activeTrackState = activeTrack->mState; 5855 switch (activeTrackState) { 5856 5857 case TrackBase::PAUSING: 5858 mActiveTracks.remove(activeTrack); 5859 mActiveTracksGen++; 5860 doBroadcast = true; 5861 size--; 5862 continue; 5863 5864 case TrackBase::STARTING_1: 5865 sleepUs = 10000; 5866 i++; 5867 continue; 5868 5869 case TrackBase::STARTING_2: 5870 doBroadcast = true; 5871 mStandby = false; 5872 activeTrack->mState = TrackBase::ACTIVE; 5873 break; 5874 5875 case TrackBase::ACTIVE: 5876 break; 5877 5878 case TrackBase::IDLE: 5879 i++; 5880 continue; 5881 5882 default: 5883 LOG_ALWAYS_FATAL("Unexpected activeTrackState %d", activeTrackState); 5884 } 5885 5886 activeTracks.add(activeTrack); 5887 i++; 5888 5889 if (activeTrack->isFastTrack()) { 5890 ALOG_ASSERT(!mFastTrackAvail); 5891 ALOG_ASSERT(fastTrack == 0); 5892 fastTrack = activeTrack; 5893 } 5894 } 5895 if (doBroadcast) { 5896 mStartStopCond.broadcast(); 5897 } 5898 5899 // sleep if there are no active tracks to process 5900 if (activeTracks.size() == 0) { 5901 if (sleepUs == 0) { 5902 sleepUs = kRecordThreadSleepUs; 5903 } 5904 continue; 5905 } 5906 sleepUs = 0; 5907 5908 lockEffectChains_l(effectChains); 5909 } 5910 5911 // thread mutex is now unlocked, mActiveTracks unknown, activeTracks.size() > 0 5912 5913 size_t size = effectChains.size(); 5914 for (size_t i = 0; i < size; i++) { 5915 // thread mutex is not locked, but effect chain is locked 5916 effectChains[i]->process_l(); 5917 } 5918 5919 // Push a new fast capture state if fast capture is not already running, or cblk change 5920 if (mFastCapture != 0) { 5921 FastCaptureStateQueue *sq = mFastCapture->sq(); 5922 FastCaptureState *state = sq->begin(); 5923 bool didModify = false; 5924 FastCaptureStateQueue::block_t block = FastCaptureStateQueue::BLOCK_UNTIL_PUSHED; 5925 if (state->mCommand != FastCaptureState::READ_WRITE /* FIXME && 5926 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)*/) { 5927 if (state->mCommand == FastCaptureState::COLD_IDLE) { 5928 int32_t old = android_atomic_inc(&mFastCaptureFutex); 5929 if (old == -1) { 5930 (void) syscall(__NR_futex, &mFastCaptureFutex, FUTEX_WAKE_PRIVATE, 1); 5931 } 5932 } 5933 state->mCommand = FastCaptureState::READ_WRITE; 5934#if 0 // FIXME 5935 mFastCaptureDumpState.increaseSamplingN(mAudioFlinger->isLowRamDevice() ? 5936 FastThreadDumpState::kSamplingNforLowRamDevice : 5937 FastThreadDumpState::kSamplingN); 5938#endif 5939 didModify = true; 5940 } 5941 audio_track_cblk_t *cblkOld = state->mCblk; 5942 audio_track_cblk_t *cblkNew = fastTrack != 0 ? fastTrack->cblk() : NULL; 5943 if (cblkNew != cblkOld) { 5944 state->mCblk = cblkNew; 5945 // block until acked if removing a fast track 5946 if (cblkOld != NULL) { 5947 block = FastCaptureStateQueue::BLOCK_UNTIL_ACKED; 5948 } 5949 didModify = true; 5950 } 5951 sq->end(didModify); 5952 if (didModify) { 5953 sq->push(block); 5954#if 0 5955 if (kUseFastCapture == FastCapture_Dynamic) { 5956 mNormalSource = mPipeSource; 5957 } 5958#endif 5959 } 5960 } 5961 5962 // now run the fast track destructor with thread mutex unlocked 5963 fastTrackToRemove.clear(); 5964 5965 // Read from HAL to keep up with fastest client if multiple active tracks, not slowest one. 5966 // Only the client(s) that are too slow will overrun. But if even the fastest client is too 5967 // slow, then this RecordThread will overrun by not calling HAL read often enough. 5968 // If destination is non-contiguous, first read past the nominal end of buffer, then 5969 // copy to the right place. Permitted because mRsmpInBuffer was over-allocated. 5970 5971 int32_t rear = mRsmpInRear & (mRsmpInFramesP2 - 1); 5972 ssize_t framesRead; 5973 5974 // If an NBAIO source is present, use it to read the normal capture's data 5975 if (mPipeSource != 0) { 5976 size_t framesToRead = mBufferSize / mFrameSize; 5977 framesRead = mPipeSource->read((uint8_t*)mRsmpInBuffer + rear * mFrameSize, 5978 framesToRead); 5979 if (framesRead == 0) { 5980 // since pipe is non-blocking, simulate blocking input 5981 sleepUs = (framesToRead * 1000000LL) / mSampleRate; 5982 } 5983 // otherwise use the HAL / AudioStreamIn directly 5984 } else { 5985 ssize_t bytesRead = mInput->stream->read(mInput->stream, 5986 (uint8_t*)mRsmpInBuffer + rear * mFrameSize, mBufferSize); 5987 if (bytesRead < 0) { 5988 framesRead = bytesRead; 5989 } else { 5990 framesRead = bytesRead / mFrameSize; 5991 } 5992 } 5993 5994 // Update server timestamp with server stats 5995 // systemTime() is optional if the hardware supports timestamps. 5996 mTimestamp.mPosition[ExtendedTimestamp::LOCATION_SERVER] += framesRead; 5997 mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_SERVER] = systemTime(); 5998 5999 // Update server timestamp with kernel stats 6000 if (mInput->stream->get_capture_position != nullptr) { 6001 int64_t position, time; 6002 int ret = mInput->stream->get_capture_position(mInput->stream, &position, &time); 6003 if (ret == NO_ERROR) { 6004 mTimestamp.mPosition[ExtendedTimestamp::LOCATION_KERNEL] = position; 6005 mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL] = time; 6006 // Note: In general record buffers should tend to be empty in 6007 // a properly running pipeline. 6008 // 6009 // Also, it is not advantageous to call get_presentation_position during the read 6010 // as the read obtains a lock, preventing the timestamp call from executing. 6011 } 6012 } 6013 // Use this to track timestamp information 6014 // ALOGD("%s", mTimestamp.toString().c_str()); 6015 6016 if (framesRead < 0 || (framesRead == 0 && mPipeSource == 0)) { 6017 ALOGE("read failed: framesRead=%d", framesRead); 6018 // Force input into standby so that it tries to recover at next read attempt 6019 inputStandBy(); 6020 sleepUs = kRecordThreadSleepUs; 6021 } 6022 if (framesRead <= 0) { 6023 goto unlock; 6024 } 6025 ALOG_ASSERT(framesRead > 0); 6026 6027 if (mTeeSink != 0) { 6028 (void) mTeeSink->write((uint8_t*)mRsmpInBuffer + rear * mFrameSize, framesRead); 6029 } 6030 // If destination is non-contiguous, we now correct for reading past end of buffer. 6031 { 6032 size_t part1 = mRsmpInFramesP2 - rear; 6033 if ((size_t) framesRead > part1) { 6034 memcpy(mRsmpInBuffer, (uint8_t*)mRsmpInBuffer + mRsmpInFramesP2 * mFrameSize, 6035 (framesRead - part1) * mFrameSize); 6036 } 6037 } 6038 rear = mRsmpInRear += framesRead; 6039 6040 size = activeTracks.size(); 6041 // loop over each active track 6042 for (size_t i = 0; i < size; i++) { 6043 activeTrack = activeTracks[i]; 6044 6045 // skip fast tracks, as those are handled directly by FastCapture 6046 if (activeTrack->isFastTrack()) { 6047 continue; 6048 } 6049 6050 // TODO: This code probably should be moved to RecordTrack. 6051 // TODO: Update the activeTrack buffer converter in case of reconfigure. 6052 6053 enum { 6054 OVERRUN_UNKNOWN, 6055 OVERRUN_TRUE, 6056 OVERRUN_FALSE 6057 } overrun = OVERRUN_UNKNOWN; 6058 6059 // loop over getNextBuffer to handle circular sink 6060 for (;;) { 6061 6062 activeTrack->mSink.frameCount = ~0; 6063 status_t status = activeTrack->getNextBuffer(&activeTrack->mSink); 6064 size_t framesOut = activeTrack->mSink.frameCount; 6065 LOG_ALWAYS_FATAL_IF((status == OK) != (framesOut > 0)); 6066 6067 // check available frames and handle overrun conditions 6068 // if the record track isn't draining fast enough. 6069 bool hasOverrun; 6070 size_t framesIn; 6071 activeTrack->mResamplerBufferProvider->sync(&framesIn, &hasOverrun); 6072 if (hasOverrun) { 6073 overrun = OVERRUN_TRUE; 6074 } 6075 if (framesOut == 0 || framesIn == 0) { 6076 break; 6077 } 6078 6079 // Don't allow framesOut to be larger than what is possible with resampling 6080 // from framesIn. 6081 // This isn't strictly necessary but helps limit buffer resizing in 6082 // RecordBufferConverter. TODO: remove when no longer needed. 6083 framesOut = min(framesOut, 6084 destinationFramesPossible( 6085 framesIn, mSampleRate, activeTrack->mSampleRate)); 6086 // process frames from the RecordThread buffer provider to the RecordTrack buffer 6087 framesOut = activeTrack->mRecordBufferConverter->convert( 6088 activeTrack->mSink.raw, activeTrack->mResamplerBufferProvider, framesOut); 6089 6090 if (framesOut > 0 && (overrun == OVERRUN_UNKNOWN)) { 6091 overrun = OVERRUN_FALSE; 6092 } 6093 6094 if (activeTrack->mFramesToDrop == 0) { 6095 if (framesOut > 0) { 6096 activeTrack->mSink.frameCount = framesOut; 6097 activeTrack->releaseBuffer(&activeTrack->mSink); 6098 } 6099 } else { 6100 // FIXME could do a partial drop of framesOut 6101 if (activeTrack->mFramesToDrop > 0) { 6102 activeTrack->mFramesToDrop -= framesOut; 6103 if (activeTrack->mFramesToDrop <= 0) { 6104 activeTrack->clearSyncStartEvent(); 6105 } 6106 } else { 6107 activeTrack->mFramesToDrop += framesOut; 6108 if (activeTrack->mFramesToDrop >= 0 || activeTrack->mSyncStartEvent == 0 || 6109 activeTrack->mSyncStartEvent->isCancelled()) { 6110 ALOGW("Synced record %s, session %d, trigger session %d", 6111 (activeTrack->mFramesToDrop >= 0) ? "timed out" : "cancelled", 6112 activeTrack->sessionId(), 6113 (activeTrack->mSyncStartEvent != 0) ? 6114 activeTrack->mSyncStartEvent->triggerSession() : 0); 6115 activeTrack->clearSyncStartEvent(); 6116 } 6117 } 6118 } 6119 6120 if (framesOut == 0) { 6121 break; 6122 } 6123 } 6124 6125 switch (overrun) { 6126 case OVERRUN_TRUE: 6127 // client isn't retrieving buffers fast enough 6128 if (!activeTrack->setOverflow()) { 6129 nsecs_t now = systemTime(); 6130 // FIXME should lastWarning per track? 6131 if ((now - lastWarning) > kWarningThrottleNs) { 6132 ALOGW("RecordThread: buffer overflow"); 6133 lastWarning = now; 6134 } 6135 } 6136 break; 6137 case OVERRUN_FALSE: 6138 activeTrack->clearOverflow(); 6139 break; 6140 case OVERRUN_UNKNOWN: 6141 break; 6142 } 6143 6144 // update frame information and push timestamp out 6145 activeTrack->updateTrackFrameInfo( 6146 activeTrack->mAudioRecordServerProxy->framesReleased(), 6147 mTimestamp.mPosition[ExtendedTimestamp::LOCATION_SERVER], 6148 mSampleRate, mTimestamp); 6149 } 6150 6151unlock: 6152 // enable changes in effect chain 6153 unlockEffectChains(effectChains); 6154 // effectChains doesn't need to be cleared, since it is cleared by destructor at scope end 6155 } 6156 6157 standbyIfNotAlreadyInStandby(); 6158 6159 { 6160 Mutex::Autolock _l(mLock); 6161 for (size_t i = 0; i < mTracks.size(); i++) { 6162 sp<RecordTrack> track = mTracks[i]; 6163 track->invalidate(); 6164 } 6165 mActiveTracks.clear(); 6166 mActiveTracksGen++; 6167 mStartStopCond.broadcast(); 6168 } 6169 6170 releaseWakeLock(); 6171 6172 ALOGV("RecordThread %p exiting", this); 6173 return false; 6174} 6175 6176void AudioFlinger::RecordThread::standbyIfNotAlreadyInStandby() 6177{ 6178 if (!mStandby) { 6179 inputStandBy(); 6180 mStandby = true; 6181 } 6182} 6183 6184void AudioFlinger::RecordThread::inputStandBy() 6185{ 6186 // Idle the fast capture if it's currently running 6187 if (mFastCapture != 0) { 6188 FastCaptureStateQueue *sq = mFastCapture->sq(); 6189 FastCaptureState *state = sq->begin(); 6190 if (!(state->mCommand & FastCaptureState::IDLE)) { 6191 state->mCommand = FastCaptureState::COLD_IDLE; 6192 state->mColdFutexAddr = &mFastCaptureFutex; 6193 state->mColdGen++; 6194 mFastCaptureFutex = 0; 6195 sq->end(); 6196 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 6197 sq->push(FastCaptureStateQueue::BLOCK_UNTIL_ACKED); 6198#if 0 6199 if (kUseFastCapture == FastCapture_Dynamic) { 6200 // FIXME 6201 } 6202#endif 6203#ifdef AUDIO_WATCHDOG 6204 // FIXME 6205#endif 6206 } else { 6207 sq->end(false /*didModify*/); 6208 } 6209 } 6210 mInput->stream->common.standby(&mInput->stream->common); 6211} 6212 6213// RecordThread::createRecordTrack_l() must be called with AudioFlinger::mLock held 6214sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 6215 const sp<AudioFlinger::Client>& client, 6216 uint32_t sampleRate, 6217 audio_format_t format, 6218 audio_channel_mask_t channelMask, 6219 size_t *pFrameCount, 6220 int sessionId, 6221 size_t *notificationFrames, 6222 int uid, 6223 IAudioFlinger::track_flags_t *flags, 6224 pid_t tid, 6225 status_t *status) 6226{ 6227 size_t frameCount = *pFrameCount; 6228 sp<RecordTrack> track; 6229 status_t lStatus; 6230 6231 // client expresses a preference for FAST, but we get the final say 6232 if (*flags & IAudioFlinger::TRACK_FAST) { 6233 if ( 6234 // we formerly checked for a callback handler (non-0 tid), 6235 // but that is no longer required for TRANSFER_OBTAIN mode 6236 // 6237 // frame count is not specified, or is exactly the pipe depth 6238 ((frameCount == 0) || (frameCount == mPipeFramesP2)) && 6239 // PCM data 6240 audio_is_linear_pcm(format) && 6241 // native format 6242 (format == mFormat) && 6243 // native channel mask 6244 (channelMask == mChannelMask) && 6245 // native hardware sample rate 6246 (sampleRate == mSampleRate) && 6247 // record thread has an associated fast capture 6248 hasFastCapture() && 6249 // there are sufficient fast track slots available 6250 mFastTrackAvail 6251 ) { 6252 ALOGV("AUDIO_INPUT_FLAG_FAST accepted: frameCount=%u mFrameCount=%u", 6253 frameCount, mFrameCount); 6254 } else { 6255 ALOGV("AUDIO_INPUT_FLAG_FAST denied: frameCount=%u mFrameCount=%u mPipeFramesP2=%u " 6256 "format=%#x isLinear=%d channelMask=%#x sampleRate=%u mSampleRate=%u " 6257 "hasFastCapture=%d tid=%d mFastTrackAvail=%d", 6258 frameCount, mFrameCount, mPipeFramesP2, 6259 format, audio_is_linear_pcm(format), channelMask, sampleRate, mSampleRate, 6260 hasFastCapture(), tid, mFastTrackAvail); 6261 *flags &= ~IAudioFlinger::TRACK_FAST; 6262 } 6263 } 6264 6265 // compute track buffer size in frames, and suggest the notification frame count 6266 if (*flags & IAudioFlinger::TRACK_FAST) { 6267 // fast track: frame count is exactly the pipe depth 6268 frameCount = mPipeFramesP2; 6269 // ignore requested notificationFrames, and always notify exactly once every HAL buffer 6270 *notificationFrames = mFrameCount; 6271 } else { 6272 // not fast track: max notification period is resampled equivalent of one HAL buffer time 6273 // or 20 ms if there is a fast capture 6274 // TODO This could be a roundupRatio inline, and const 6275 size_t maxNotificationFrames = ((int64_t) (hasFastCapture() ? mSampleRate/50 : mFrameCount) 6276 * sampleRate + mSampleRate - 1) / mSampleRate; 6277 // minimum number of notification periods is at least kMinNotifications, 6278 // and at least kMinMs rounded up to a whole notification period (minNotificationsByMs) 6279 static const size_t kMinNotifications = 3; 6280 static const uint32_t kMinMs = 30; 6281 // TODO This could be a roundupRatio inline 6282 const size_t minFramesByMs = (sampleRate * kMinMs + 1000 - 1) / 1000; 6283 // TODO This could be a roundupRatio inline 6284 const size_t minNotificationsByMs = (minFramesByMs + maxNotificationFrames - 1) / 6285 maxNotificationFrames; 6286 const size_t minFrameCount = maxNotificationFrames * 6287 max(kMinNotifications, minNotificationsByMs); 6288 frameCount = max(frameCount, minFrameCount); 6289 if (*notificationFrames == 0 || *notificationFrames > maxNotificationFrames) { 6290 *notificationFrames = maxNotificationFrames; 6291 } 6292 } 6293 *pFrameCount = frameCount; 6294 6295 lStatus = initCheck(); 6296 if (lStatus != NO_ERROR) { 6297 ALOGE("createRecordTrack_l() audio driver not initialized"); 6298 goto Exit; 6299 } 6300 6301 { // scope for mLock 6302 Mutex::Autolock _l(mLock); 6303 6304 track = new RecordTrack(this, client, sampleRate, 6305 format, channelMask, frameCount, NULL, sessionId, uid, 6306 *flags, TrackBase::TYPE_DEFAULT); 6307 6308 lStatus = track->initCheck(); 6309 if (lStatus != NO_ERROR) { 6310 ALOGE("createRecordTrack_l() initCheck failed %d; no control block?", lStatus); 6311 // track must be cleared from the caller as the caller has the AF lock 6312 goto Exit; 6313 } 6314 mTracks.add(track); 6315 6316 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 6317 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 6318 mAudioFlinger->btNrecIsOff(); 6319 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 6320 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 6321 6322 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 6323 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 6324 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 6325 // so ask activity manager to do this on our behalf 6326 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 6327 } 6328 } 6329 6330 lStatus = NO_ERROR; 6331 6332Exit: 6333 *status = lStatus; 6334 return track; 6335} 6336 6337status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, 6338 AudioSystem::sync_event_t event, 6339 int triggerSession) 6340{ 6341 ALOGV("RecordThread::start event %d, triggerSession %d", event, triggerSession); 6342 sp<ThreadBase> strongMe = this; 6343 status_t status = NO_ERROR; 6344 6345 if (event == AudioSystem::SYNC_EVENT_NONE) { 6346 recordTrack->clearSyncStartEvent(); 6347 } else if (event != AudioSystem::SYNC_EVENT_SAME) { 6348 recordTrack->mSyncStartEvent = mAudioFlinger->createSyncEvent(event, 6349 triggerSession, 6350 recordTrack->sessionId(), 6351 syncStartEventCallback, 6352 recordTrack); 6353 // Sync event can be cancelled by the trigger session if the track is not in a 6354 // compatible state in which case we start record immediately 6355 if (recordTrack->mSyncStartEvent->isCancelled()) { 6356 recordTrack->clearSyncStartEvent(); 6357 } else { 6358 // do not wait for the event for more than AudioSystem::kSyncRecordStartTimeOutMs 6359 recordTrack->mFramesToDrop = - 6360 ((AudioSystem::kSyncRecordStartTimeOutMs * recordTrack->mSampleRate) / 1000); 6361 } 6362 } 6363 6364 { 6365 // This section is a rendezvous between binder thread executing start() and RecordThread 6366 AutoMutex lock(mLock); 6367 if (mActiveTracks.indexOf(recordTrack) >= 0) { 6368 if (recordTrack->mState == TrackBase::PAUSING) { 6369 ALOGV("active record track PAUSING -> ACTIVE"); 6370 recordTrack->mState = TrackBase::ACTIVE; 6371 } else { 6372 ALOGV("active record track state %d", recordTrack->mState); 6373 } 6374 return status; 6375 } 6376 6377 // TODO consider other ways of handling this, such as changing the state to :STARTING and 6378 // adding the track to mActiveTracks after returning from AudioSystem::startInput(), 6379 // or using a separate command thread 6380 recordTrack->mState = TrackBase::STARTING_1; 6381 mActiveTracks.add(recordTrack); 6382 mActiveTracksGen++; 6383 status_t status = NO_ERROR; 6384 if (recordTrack->isExternalTrack()) { 6385 mLock.unlock(); 6386 status = AudioSystem::startInput(mId, (audio_session_t)recordTrack->sessionId()); 6387 mLock.lock(); 6388 // FIXME should verify that recordTrack is still in mActiveTracks 6389 if (status != NO_ERROR) { 6390 mActiveTracks.remove(recordTrack); 6391 mActiveTracksGen++; 6392 recordTrack->clearSyncStartEvent(); 6393 ALOGV("RecordThread::start error %d", status); 6394 return status; 6395 } 6396 } 6397 // Catch up with current buffer indices if thread is already running. 6398 // This is what makes a new client discard all buffered data. If the track's mRsmpInFront 6399 // was initialized to some value closer to the thread's mRsmpInFront, then the track could 6400 // see previously buffered data before it called start(), but with greater risk of overrun. 6401 6402 recordTrack->mResamplerBufferProvider->reset(); 6403 // clear any converter state as new data will be discontinuous 6404 recordTrack->mRecordBufferConverter->reset(); 6405 recordTrack->mState = TrackBase::STARTING_2; 6406 // signal thread to start 6407 mWaitWorkCV.broadcast(); 6408 if (mActiveTracks.indexOf(recordTrack) < 0) { 6409 ALOGV("Record failed to start"); 6410 status = BAD_VALUE; 6411 goto startError; 6412 } 6413 return status; 6414 } 6415 6416startError: 6417 if (recordTrack->isExternalTrack()) { 6418 AudioSystem::stopInput(mId, (audio_session_t)recordTrack->sessionId()); 6419 } 6420 recordTrack->clearSyncStartEvent(); 6421 // FIXME I wonder why we do not reset the state here? 6422 return status; 6423} 6424 6425void AudioFlinger::RecordThread::syncStartEventCallback(const wp<SyncEvent>& event) 6426{ 6427 sp<SyncEvent> strongEvent = event.promote(); 6428 6429 if (strongEvent != 0) { 6430 sp<RefBase> ptr = strongEvent->cookie().promote(); 6431 if (ptr != 0) { 6432 RecordTrack *recordTrack = (RecordTrack *)ptr.get(); 6433 recordTrack->handleSyncStartEvent(strongEvent); 6434 } 6435 } 6436} 6437 6438bool AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 6439 ALOGV("RecordThread::stop"); 6440 AutoMutex _l(mLock); 6441 if (mActiveTracks.indexOf(recordTrack) != 0 || recordTrack->mState == TrackBase::PAUSING) { 6442 return false; 6443 } 6444 // note that threadLoop may still be processing the track at this point [without lock] 6445 recordTrack->mState = TrackBase::PAUSING; 6446 // do not wait for mStartStopCond if exiting 6447 if (exitPending()) { 6448 return true; 6449 } 6450 // FIXME incorrect usage of wait: no explicit predicate or loop 6451 mStartStopCond.wait(mLock); 6452 // if we have been restarted, recordTrack is in mActiveTracks here 6453 if (exitPending() || mActiveTracks.indexOf(recordTrack) != 0) { 6454 ALOGV("Record stopped OK"); 6455 return true; 6456 } 6457 return false; 6458} 6459 6460bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event __unused) const 6461{ 6462 return false; 6463} 6464 6465status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event __unused) 6466{ 6467#if 0 // This branch is currently dead code, but is preserved in case it will be needed in future 6468 if (!isValidSyncEvent(event)) { 6469 return BAD_VALUE; 6470 } 6471 6472 int eventSession = event->triggerSession(); 6473 status_t ret = NAME_NOT_FOUND; 6474 6475 Mutex::Autolock _l(mLock); 6476 6477 for (size_t i = 0; i < mTracks.size(); i++) { 6478 sp<RecordTrack> track = mTracks[i]; 6479 if (eventSession == track->sessionId()) { 6480 (void) track->setSyncEvent(event); 6481 ret = NO_ERROR; 6482 } 6483 } 6484 return ret; 6485#else 6486 return BAD_VALUE; 6487#endif 6488} 6489 6490// destroyTrack_l() must be called with ThreadBase::mLock held 6491void AudioFlinger::RecordThread::destroyTrack_l(const sp<RecordTrack>& track) 6492{ 6493 track->terminate(); 6494 track->mState = TrackBase::STOPPED; 6495 // active tracks are removed by threadLoop() 6496 if (mActiveTracks.indexOf(track) < 0) { 6497 removeTrack_l(track); 6498 } 6499} 6500 6501void AudioFlinger::RecordThread::removeTrack_l(const sp<RecordTrack>& track) 6502{ 6503 mTracks.remove(track); 6504 // need anything related to effects here? 6505 if (track->isFastTrack()) { 6506 ALOG_ASSERT(!mFastTrackAvail); 6507 mFastTrackAvail = true; 6508 } 6509} 6510 6511void AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 6512{ 6513 dumpInternals(fd, args); 6514 dumpTracks(fd, args); 6515 dumpEffectChains(fd, args); 6516} 6517 6518void AudioFlinger::RecordThread::dumpInternals(int fd, const Vector<String16>& args) 6519{ 6520 dprintf(fd, "\nInput thread %p:\n", this); 6521 6522 dumpBase(fd, args); 6523 6524 if (mActiveTracks.size() == 0) { 6525 dprintf(fd, " No active record clients\n"); 6526 } 6527 dprintf(fd, " Fast capture thread: %s\n", hasFastCapture() ? "yes" : "no"); 6528 dprintf(fd, " Fast track available: %s\n", mFastTrackAvail ? "yes" : "no"); 6529 6530 // Make a non-atomic copy of fast capture dump state so it won't change underneath us 6531 // while we are dumping it. It may be inconsistent, but it won't mutate! 6532 // This is a large object so we place it on the heap. 6533 // FIXME 25972958: Need an intelligent copy constructor that does not touch unused pages. 6534 const FastCaptureDumpState *copy = new FastCaptureDumpState(mFastCaptureDumpState); 6535 copy->dump(fd); 6536 delete copy; 6537} 6538 6539void AudioFlinger::RecordThread::dumpTracks(int fd, const Vector<String16>& args __unused) 6540{ 6541 const size_t SIZE = 256; 6542 char buffer[SIZE]; 6543 String8 result; 6544 6545 size_t numtracks = mTracks.size(); 6546 size_t numactive = mActiveTracks.size(); 6547 size_t numactiveseen = 0; 6548 dprintf(fd, " %d Tracks", numtracks); 6549 if (numtracks) { 6550 dprintf(fd, " of which %d are active\n", numactive); 6551 RecordTrack::appendDumpHeader(result); 6552 for (size_t i = 0; i < numtracks ; ++i) { 6553 sp<RecordTrack> track = mTracks[i]; 6554 if (track != 0) { 6555 bool active = mActiveTracks.indexOf(track) >= 0; 6556 if (active) { 6557 numactiveseen++; 6558 } 6559 track->dump(buffer, SIZE, active); 6560 result.append(buffer); 6561 } 6562 } 6563 } else { 6564 dprintf(fd, "\n"); 6565 } 6566 6567 if (numactiveseen != numactive) { 6568 snprintf(buffer, SIZE, " The following tracks are in the active list but" 6569 " not in the track list\n"); 6570 result.append(buffer); 6571 RecordTrack::appendDumpHeader(result); 6572 for (size_t i = 0; i < numactive; ++i) { 6573 sp<RecordTrack> track = mActiveTracks[i]; 6574 if (mTracks.indexOf(track) < 0) { 6575 track->dump(buffer, SIZE, true); 6576 result.append(buffer); 6577 } 6578 } 6579 6580 } 6581 write(fd, result.string(), result.size()); 6582} 6583 6584 6585void AudioFlinger::RecordThread::ResamplerBufferProvider::reset() 6586{ 6587 sp<ThreadBase> threadBase = mRecordTrack->mThread.promote(); 6588 RecordThread *recordThread = (RecordThread *) threadBase.get(); 6589 mRsmpInFront = recordThread->mRsmpInRear; 6590 mRsmpInUnrel = 0; 6591} 6592 6593void AudioFlinger::RecordThread::ResamplerBufferProvider::sync( 6594 size_t *framesAvailable, bool *hasOverrun) 6595{ 6596 sp<ThreadBase> threadBase = mRecordTrack->mThread.promote(); 6597 RecordThread *recordThread = (RecordThread *) threadBase.get(); 6598 const int32_t rear = recordThread->mRsmpInRear; 6599 const int32_t front = mRsmpInFront; 6600 const ssize_t filled = rear - front; 6601 6602 size_t framesIn; 6603 bool overrun = false; 6604 if (filled < 0) { 6605 // should not happen, but treat like a massive overrun and re-sync 6606 framesIn = 0; 6607 mRsmpInFront = rear; 6608 overrun = true; 6609 } else if ((size_t) filled <= recordThread->mRsmpInFrames) { 6610 framesIn = (size_t) filled; 6611 } else { 6612 // client is not keeping up with server, but give it latest data 6613 framesIn = recordThread->mRsmpInFrames; 6614 mRsmpInFront = /* front = */ rear - framesIn; 6615 overrun = true; 6616 } 6617 if (framesAvailable != NULL) { 6618 *framesAvailable = framesIn; 6619 } 6620 if (hasOverrun != NULL) { 6621 *hasOverrun = overrun; 6622 } 6623} 6624 6625// AudioBufferProvider interface 6626status_t AudioFlinger::RecordThread::ResamplerBufferProvider::getNextBuffer( 6627 AudioBufferProvider::Buffer* buffer) 6628{ 6629 sp<ThreadBase> threadBase = mRecordTrack->mThread.promote(); 6630 if (threadBase == 0) { 6631 buffer->frameCount = 0; 6632 buffer->raw = NULL; 6633 return NOT_ENOUGH_DATA; 6634 } 6635 RecordThread *recordThread = (RecordThread *) threadBase.get(); 6636 int32_t rear = recordThread->mRsmpInRear; 6637 int32_t front = mRsmpInFront; 6638 ssize_t filled = rear - front; 6639 // FIXME should not be P2 (don't want to increase latency) 6640 // FIXME if client not keeping up, discard 6641 LOG_ALWAYS_FATAL_IF(!(0 <= filled && (size_t) filled <= recordThread->mRsmpInFrames)); 6642 // 'filled' may be non-contiguous, so return only the first contiguous chunk 6643 front &= recordThread->mRsmpInFramesP2 - 1; 6644 size_t part1 = recordThread->mRsmpInFramesP2 - front; 6645 if (part1 > (size_t) filled) { 6646 part1 = filled; 6647 } 6648 size_t ask = buffer->frameCount; 6649 ALOG_ASSERT(ask > 0); 6650 if (part1 > ask) { 6651 part1 = ask; 6652 } 6653 if (part1 == 0) { 6654 // out of data is fine since the resampler will return a short-count. 6655 buffer->raw = NULL; 6656 buffer->frameCount = 0; 6657 mRsmpInUnrel = 0; 6658 return NOT_ENOUGH_DATA; 6659 } 6660 6661 buffer->raw = (uint8_t*)recordThread->mRsmpInBuffer + front * recordThread->mFrameSize; 6662 buffer->frameCount = part1; 6663 mRsmpInUnrel = part1; 6664 return NO_ERROR; 6665} 6666 6667// AudioBufferProvider interface 6668void AudioFlinger::RecordThread::ResamplerBufferProvider::releaseBuffer( 6669 AudioBufferProvider::Buffer* buffer) 6670{ 6671 size_t stepCount = buffer->frameCount; 6672 if (stepCount == 0) { 6673 return; 6674 } 6675 ALOG_ASSERT(stepCount <= mRsmpInUnrel); 6676 mRsmpInUnrel -= stepCount; 6677 mRsmpInFront += stepCount; 6678 buffer->raw = NULL; 6679 buffer->frameCount = 0; 6680} 6681 6682AudioFlinger::RecordThread::RecordBufferConverter::RecordBufferConverter( 6683 audio_channel_mask_t srcChannelMask, audio_format_t srcFormat, 6684 uint32_t srcSampleRate, 6685 audio_channel_mask_t dstChannelMask, audio_format_t dstFormat, 6686 uint32_t dstSampleRate) : 6687 mSrcChannelMask(AUDIO_CHANNEL_INVALID), // updateParameters will set following vars 6688 // mSrcFormat 6689 // mSrcSampleRate 6690 // mDstChannelMask 6691 // mDstFormat 6692 // mDstSampleRate 6693 // mSrcChannelCount 6694 // mDstChannelCount 6695 // mDstFrameSize 6696 mBuf(NULL), mBufFrames(0), mBufFrameSize(0), 6697 mResampler(NULL), 6698 mIsLegacyDownmix(false), 6699 mIsLegacyUpmix(false), 6700 mRequiresFloat(false), 6701 mInputConverterProvider(NULL) 6702{ 6703 (void)updateParameters(srcChannelMask, srcFormat, srcSampleRate, 6704 dstChannelMask, dstFormat, dstSampleRate); 6705} 6706 6707AudioFlinger::RecordThread::RecordBufferConverter::~RecordBufferConverter() { 6708 free(mBuf); 6709 delete mResampler; 6710 delete mInputConverterProvider; 6711} 6712 6713size_t AudioFlinger::RecordThread::RecordBufferConverter::convert(void *dst, 6714 AudioBufferProvider *provider, size_t frames) 6715{ 6716 if (mInputConverterProvider != NULL) { 6717 mInputConverterProvider->setBufferProvider(provider); 6718 provider = mInputConverterProvider; 6719 } 6720 6721 if (mResampler == NULL) { 6722 ALOGVV("NO RESAMPLING sampleRate:%u mSrcFormat:%#x mDstFormat:%#x", 6723 mSrcSampleRate, mSrcFormat, mDstFormat); 6724 6725 AudioBufferProvider::Buffer buffer; 6726 for (size_t i = frames; i > 0; ) { 6727 buffer.frameCount = i; 6728 status_t status = provider->getNextBuffer(&buffer); 6729 if (status != OK || buffer.frameCount == 0) { 6730 frames -= i; // cannot fill request. 6731 break; 6732 } 6733 // format convert to destination buffer 6734 convertNoResampler(dst, buffer.raw, buffer.frameCount); 6735 6736 dst = (int8_t*)dst + buffer.frameCount * mDstFrameSize; 6737 i -= buffer.frameCount; 6738 provider->releaseBuffer(&buffer); 6739 } 6740 } else { 6741 ALOGVV("RESAMPLING mSrcSampleRate:%u mDstSampleRate:%u mSrcFormat:%#x mDstFormat:%#x", 6742 mSrcSampleRate, mDstSampleRate, mSrcFormat, mDstFormat); 6743 6744 // reallocate buffer if needed 6745 if (mBufFrameSize != 0 && mBufFrames < frames) { 6746 free(mBuf); 6747 mBufFrames = frames; 6748 (void)posix_memalign(&mBuf, 32, mBufFrames * mBufFrameSize); 6749 } 6750 // resampler accumulates, but we only have one source track 6751 memset(mBuf, 0, frames * mBufFrameSize); 6752 frames = mResampler->resample((int32_t*)mBuf, frames, provider); 6753 // format convert to destination buffer 6754 convertResampler(dst, mBuf, frames); 6755 } 6756 return frames; 6757} 6758 6759status_t AudioFlinger::RecordThread::RecordBufferConverter::updateParameters( 6760 audio_channel_mask_t srcChannelMask, audio_format_t srcFormat, 6761 uint32_t srcSampleRate, 6762 audio_channel_mask_t dstChannelMask, audio_format_t dstFormat, 6763 uint32_t dstSampleRate) 6764{ 6765 // quick evaluation if there is any change. 6766 if (mSrcFormat == srcFormat 6767 && mSrcChannelMask == srcChannelMask 6768 && mSrcSampleRate == srcSampleRate 6769 && mDstFormat == dstFormat 6770 && mDstChannelMask == dstChannelMask 6771 && mDstSampleRate == dstSampleRate) { 6772 return NO_ERROR; 6773 } 6774 6775 ALOGV("RecordBufferConverter updateParameters srcMask:%#x dstMask:%#x" 6776 " srcFormat:%#x dstFormat:%#x srcRate:%u dstRate:%u", 6777 srcChannelMask, dstChannelMask, srcFormat, dstFormat, srcSampleRate, dstSampleRate); 6778 const bool valid = 6779 audio_is_input_channel(srcChannelMask) 6780 && audio_is_input_channel(dstChannelMask) 6781 && audio_is_valid_format(srcFormat) && audio_is_linear_pcm(srcFormat) 6782 && audio_is_valid_format(dstFormat) && audio_is_linear_pcm(dstFormat) 6783 && (srcSampleRate <= dstSampleRate * AUDIO_RESAMPLER_DOWN_RATIO_MAX) 6784 ; // no upsampling checks for now 6785 if (!valid) { 6786 return BAD_VALUE; 6787 } 6788 6789 mSrcFormat = srcFormat; 6790 mSrcChannelMask = srcChannelMask; 6791 mSrcSampleRate = srcSampleRate; 6792 mDstFormat = dstFormat; 6793 mDstChannelMask = dstChannelMask; 6794 mDstSampleRate = dstSampleRate; 6795 6796 // compute derived parameters 6797 mSrcChannelCount = audio_channel_count_from_in_mask(srcChannelMask); 6798 mDstChannelCount = audio_channel_count_from_in_mask(dstChannelMask); 6799 mDstFrameSize = mDstChannelCount * audio_bytes_per_sample(mDstFormat); 6800 6801 // do we need to resample? 6802 delete mResampler; 6803 mResampler = NULL; 6804 if (mSrcSampleRate != mDstSampleRate) { 6805 mResampler = AudioResampler::create(AUDIO_FORMAT_PCM_FLOAT, 6806 mSrcChannelCount, mDstSampleRate); 6807 mResampler->setSampleRate(mSrcSampleRate); 6808 mResampler->setVolume(AudioMixer::UNITY_GAIN_FLOAT, AudioMixer::UNITY_GAIN_FLOAT); 6809 } 6810 6811 // are we running legacy channel conversion modes? 6812 mIsLegacyDownmix = (mSrcChannelMask == AUDIO_CHANNEL_IN_STEREO 6813 || mSrcChannelMask == AUDIO_CHANNEL_IN_FRONT_BACK) 6814 && mDstChannelMask == AUDIO_CHANNEL_IN_MONO; 6815 mIsLegacyUpmix = mSrcChannelMask == AUDIO_CHANNEL_IN_MONO 6816 && (mDstChannelMask == AUDIO_CHANNEL_IN_STEREO 6817 || mDstChannelMask == AUDIO_CHANNEL_IN_FRONT_BACK); 6818 6819 // do we need to process in float? 6820 mRequiresFloat = mResampler != NULL || mIsLegacyDownmix || mIsLegacyUpmix; 6821 6822 // do we need a staging buffer to convert for destination (we can still optimize this)? 6823 // we use mBufFrameSize > 0 to indicate both frame size as well as buffer necessity 6824 if (mResampler != NULL) { 6825 mBufFrameSize = max(mSrcChannelCount, FCC_2) 6826 * audio_bytes_per_sample(AUDIO_FORMAT_PCM_FLOAT); 6827 } else if (mIsLegacyUpmix || mIsLegacyDownmix) { // legacy modes always float 6828 mBufFrameSize = mDstChannelCount * audio_bytes_per_sample(AUDIO_FORMAT_PCM_FLOAT); 6829 } else if (mSrcChannelMask != mDstChannelMask && mDstFormat != mSrcFormat) { 6830 mBufFrameSize = mDstChannelCount * audio_bytes_per_sample(mSrcFormat); 6831 } else { 6832 mBufFrameSize = 0; 6833 } 6834 mBufFrames = 0; // force the buffer to be resized. 6835 6836 // do we need an input converter buffer provider to give us float? 6837 delete mInputConverterProvider; 6838 mInputConverterProvider = NULL; 6839 if (mRequiresFloat && mSrcFormat != AUDIO_FORMAT_PCM_FLOAT) { 6840 mInputConverterProvider = new ReformatBufferProvider( 6841 audio_channel_count_from_in_mask(mSrcChannelMask), 6842 mSrcFormat, 6843 AUDIO_FORMAT_PCM_FLOAT, 6844 256 /* provider buffer frame count */); 6845 } 6846 6847 // do we need a remixer to do channel mask conversion 6848 if (!mIsLegacyDownmix && !mIsLegacyUpmix && mSrcChannelMask != mDstChannelMask) { 6849 (void) memcpy_by_index_array_initialization_from_channel_mask( 6850 mIdxAry, ARRAY_SIZE(mIdxAry), mDstChannelMask, mSrcChannelMask); 6851 } 6852 return NO_ERROR; 6853} 6854 6855void AudioFlinger::RecordThread::RecordBufferConverter::convertNoResampler( 6856 void *dst, const void *src, size_t frames) 6857{ 6858 // src is native type unless there is legacy upmix or downmix, whereupon it is float. 6859 if (mBufFrameSize != 0 && mBufFrames < frames) { 6860 free(mBuf); 6861 mBufFrames = frames; 6862 (void)posix_memalign(&mBuf, 32, mBufFrames * mBufFrameSize); 6863 } 6864 // do we need to do legacy upmix and downmix? 6865 if (mIsLegacyUpmix || mIsLegacyDownmix) { 6866 void *dstBuf = mBuf != NULL ? mBuf : dst; 6867 if (mIsLegacyUpmix) { 6868 upmix_to_stereo_float_from_mono_float((float *)dstBuf, 6869 (const float *)src, frames); 6870 } else /*mIsLegacyDownmix */ { 6871 downmix_to_mono_float_from_stereo_float((float *)dstBuf, 6872 (const float *)src, frames); 6873 } 6874 if (mBuf != NULL) { 6875 memcpy_by_audio_format(dst, mDstFormat, mBuf, AUDIO_FORMAT_PCM_FLOAT, 6876 frames * mDstChannelCount); 6877 } 6878 return; 6879 } 6880 // do we need to do channel mask conversion? 6881 if (mSrcChannelMask != mDstChannelMask) { 6882 void *dstBuf = mBuf != NULL ? mBuf : dst; 6883 memcpy_by_index_array(dstBuf, mDstChannelCount, 6884 src, mSrcChannelCount, mIdxAry, audio_bytes_per_sample(mSrcFormat), frames); 6885 if (dstBuf == dst) { 6886 return; // format is the same 6887 } 6888 } 6889 // convert to destination buffer 6890 const void *convertBuf = mBuf != NULL ? mBuf : src; 6891 memcpy_by_audio_format(dst, mDstFormat, convertBuf, mSrcFormat, 6892 frames * mDstChannelCount); 6893} 6894 6895void AudioFlinger::RecordThread::RecordBufferConverter::convertResampler( 6896 void *dst, /*not-a-const*/ void *src, size_t frames) 6897{ 6898 // src buffer format is ALWAYS float when entering this routine 6899 if (mIsLegacyUpmix) { 6900 ; // mono to stereo already handled by resampler 6901 } else if (mIsLegacyDownmix 6902 || (mSrcChannelMask == mDstChannelMask && mSrcChannelCount == 1)) { 6903 // the resampler outputs stereo for mono input channel (a feature?) 6904 // must convert to mono 6905 downmix_to_mono_float_from_stereo_float((float *)src, 6906 (const float *)src, frames); 6907 } else if (mSrcChannelMask != mDstChannelMask) { 6908 // convert to mono channel again for channel mask conversion (could be skipped 6909 // with further optimization). 6910 if (mSrcChannelCount == 1) { 6911 downmix_to_mono_float_from_stereo_float((float *)src, 6912 (const float *)src, frames); 6913 } 6914 // convert to destination format (in place, OK as float is larger than other types) 6915 if (mDstFormat != AUDIO_FORMAT_PCM_FLOAT) { 6916 memcpy_by_audio_format(src, mDstFormat, src, AUDIO_FORMAT_PCM_FLOAT, 6917 frames * mSrcChannelCount); 6918 } 6919 // channel convert and save to dst 6920 memcpy_by_index_array(dst, mDstChannelCount, 6921 src, mSrcChannelCount, mIdxAry, audio_bytes_per_sample(mDstFormat), frames); 6922 return; 6923 } 6924 // convert to destination format and save to dst 6925 memcpy_by_audio_format(dst, mDstFormat, src, AUDIO_FORMAT_PCM_FLOAT, 6926 frames * mDstChannelCount); 6927} 6928 6929bool AudioFlinger::RecordThread::checkForNewParameter_l(const String8& keyValuePair, 6930 status_t& status) 6931{ 6932 bool reconfig = false; 6933 6934 status = NO_ERROR; 6935 6936 audio_format_t reqFormat = mFormat; 6937 uint32_t samplingRate = mSampleRate; 6938 // TODO this may change if we want to support capture from HDMI PCM multi channel (e.g on TVs). 6939 audio_channel_mask_t channelMask = audio_channel_in_mask_from_count(mChannelCount); 6940 6941 AudioParameter param = AudioParameter(keyValuePair); 6942 int value; 6943 // TODO Investigate when this code runs. Check with audio policy when a sample rate and 6944 // channel count change can be requested. Do we mandate the first client defines the 6945 // HAL sampling rate and channel count or do we allow changes on the fly? 6946 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 6947 samplingRate = value; 6948 reconfig = true; 6949 } 6950 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 6951 if (!audio_is_linear_pcm((audio_format_t) value)) { 6952 status = BAD_VALUE; 6953 } else { 6954 reqFormat = (audio_format_t) value; 6955 reconfig = true; 6956 } 6957 } 6958 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 6959 audio_channel_mask_t mask = (audio_channel_mask_t) value; 6960 if (!audio_is_input_channel(mask) || 6961 audio_channel_count_from_in_mask(mask) > FCC_8) { 6962 status = BAD_VALUE; 6963 } else { 6964 channelMask = mask; 6965 reconfig = true; 6966 } 6967 } 6968 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 6969 // do not accept frame count changes if tracks are open as the track buffer 6970 // size depends on frame count and correct behavior would not be guaranteed 6971 // if frame count is changed after track creation 6972 if (mActiveTracks.size() > 0) { 6973 status = INVALID_OPERATION; 6974 } else { 6975 reconfig = true; 6976 } 6977 } 6978 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 6979 // forward device change to effects that have requested to be 6980 // aware of attached audio device. 6981 for (size_t i = 0; i < mEffectChains.size(); i++) { 6982 mEffectChains[i]->setDevice_l(value); 6983 } 6984 6985 // store input device and output device but do not forward output device to audio HAL. 6986 // Note that status is ignored by the caller for output device 6987 // (see AudioFlinger::setParameters() 6988 if (audio_is_output_devices(value)) { 6989 mOutDevice = value; 6990 status = BAD_VALUE; 6991 } else { 6992 mInDevice = value; 6993 if (value != AUDIO_DEVICE_NONE) { 6994 mPrevInDevice = value; 6995 } 6996 // disable AEC and NS if the device is a BT SCO headset supporting those 6997 // pre processings 6998 if (mTracks.size() > 0) { 6999 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 7000 mAudioFlinger->btNrecIsOff(); 7001 for (size_t i = 0; i < mTracks.size(); i++) { 7002 sp<RecordTrack> track = mTracks[i]; 7003 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId()); 7004 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId()); 7005 } 7006 } 7007 } 7008 } 7009 if (param.getInt(String8(AudioParameter::keyInputSource), value) == NO_ERROR && 7010 mAudioSource != (audio_source_t)value) { 7011 // forward device change to effects that have requested to be 7012 // aware of attached audio device. 7013 for (size_t i = 0; i < mEffectChains.size(); i++) { 7014 mEffectChains[i]->setAudioSource_l((audio_source_t)value); 7015 } 7016 mAudioSource = (audio_source_t)value; 7017 } 7018 7019 if (status == NO_ERROR) { 7020 status = mInput->stream->common.set_parameters(&mInput->stream->common, 7021 keyValuePair.string()); 7022 if (status == INVALID_OPERATION) { 7023 inputStandBy(); 7024 status = mInput->stream->common.set_parameters(&mInput->stream->common, 7025 keyValuePair.string()); 7026 } 7027 if (reconfig) { 7028 if (status == BAD_VALUE && 7029 audio_is_linear_pcm(mInput->stream->common.get_format(&mInput->stream->common)) && 7030 audio_is_linear_pcm(reqFormat) && 7031 (mInput->stream->common.get_sample_rate(&mInput->stream->common) 7032 <= (AUDIO_RESAMPLER_DOWN_RATIO_MAX * samplingRate)) && 7033 audio_channel_count_from_in_mask( 7034 mInput->stream->common.get_channels(&mInput->stream->common)) <= FCC_8) { 7035 status = NO_ERROR; 7036 } 7037 if (status == NO_ERROR) { 7038 readInputParameters_l(); 7039 sendIoConfigEvent_l(AUDIO_INPUT_CONFIG_CHANGED); 7040 } 7041 } 7042 } 7043 7044 return reconfig; 7045} 7046 7047String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 7048{ 7049 Mutex::Autolock _l(mLock); 7050 if (initCheck() != NO_ERROR) { 7051 return String8(); 7052 } 7053 7054 char *s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 7055 const String8 out_s8(s); 7056 free(s); 7057 return out_s8; 7058} 7059 7060void AudioFlinger::RecordThread::ioConfigChanged(audio_io_config_event event, pid_t pid) { 7061 sp<AudioIoDescriptor> desc = new AudioIoDescriptor(); 7062 7063 desc->mIoHandle = mId; 7064 7065 switch (event) { 7066 case AUDIO_INPUT_OPENED: 7067 case AUDIO_INPUT_CONFIG_CHANGED: 7068 desc->mPatch = mPatch; 7069 desc->mChannelMask = mChannelMask; 7070 desc->mSamplingRate = mSampleRate; 7071 desc->mFormat = mFormat; 7072 desc->mFrameCount = mFrameCount; 7073 desc->mLatency = 0; 7074 break; 7075 7076 case AUDIO_INPUT_CLOSED: 7077 default: 7078 break; 7079 } 7080 mAudioFlinger->ioConfigChanged(event, desc, pid); 7081} 7082 7083void AudioFlinger::RecordThread::readInputParameters_l() 7084{ 7085 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 7086 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 7087 mChannelCount = audio_channel_count_from_in_mask(mChannelMask); 7088 if (mChannelCount > FCC_8) { 7089 ALOGE("HAL channel count %d > %d", mChannelCount, FCC_8); 7090 } 7091 mHALFormat = mInput->stream->common.get_format(&mInput->stream->common); 7092 mFormat = mHALFormat; 7093 if (!audio_is_linear_pcm(mFormat)) { 7094 ALOGE("HAL format %#x is not linear pcm", mFormat); 7095 } 7096 mFrameSize = audio_stream_in_frame_size(mInput->stream); 7097 mBufferSize = mInput->stream->common.get_buffer_size(&mInput->stream->common); 7098 mFrameCount = mBufferSize / mFrameSize; 7099 // This is the formula for calculating the temporary buffer size. 7100 // With 7 HAL buffers, we can guarantee ability to down-sample the input by ratio of 6:1 to 7101 // 1 full output buffer, regardless of the alignment of the available input. 7102 // The value is somewhat arbitrary, and could probably be even larger. 7103 // A larger value should allow more old data to be read after a track calls start(), 7104 // without increasing latency. 7105 // 7106 // Note this is independent of the maximum downsampling ratio permitted for capture. 7107 mRsmpInFrames = mFrameCount * 7; 7108 mRsmpInFramesP2 = roundup(mRsmpInFrames); 7109 free(mRsmpInBuffer); 7110 mRsmpInBuffer = NULL; 7111 7112 // TODO optimize audio capture buffer sizes ... 7113 // Here we calculate the size of the sliding buffer used as a source 7114 // for resampling. mRsmpInFramesP2 is currently roundup(mFrameCount * 7). 7115 // For current HAL frame counts, this is usually 2048 = 40 ms. It would 7116 // be better to have it derived from the pipe depth in the long term. 7117 // The current value is higher than necessary. However it should not add to latency. 7118 7119 // Over-allocate beyond mRsmpInFramesP2 to permit a HAL read past end of buffer 7120 size_t bufferSize = (mRsmpInFramesP2 + mFrameCount - 1) * mFrameSize; 7121 (void)posix_memalign(&mRsmpInBuffer, 32, bufferSize); 7122 memset(mRsmpInBuffer, 0, bufferSize); // if posix_memalign fails, will segv here. 7123 7124 // AudioRecord mSampleRate and mChannelCount are constant due to AudioRecord API constraints. 7125 // But if thread's mSampleRate or mChannelCount changes, how will that affect active tracks? 7126} 7127 7128uint32_t AudioFlinger::RecordThread::getInputFramesLost() 7129{ 7130 Mutex::Autolock _l(mLock); 7131 if (initCheck() != NO_ERROR) { 7132 return 0; 7133 } 7134 7135 return mInput->stream->get_input_frames_lost(mInput->stream); 7136} 7137 7138uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) const 7139{ 7140 Mutex::Autolock _l(mLock); 7141 uint32_t result = 0; 7142 if (getEffectChain_l(sessionId) != 0) { 7143 result = EFFECT_SESSION; 7144 } 7145 7146 for (size_t i = 0; i < mTracks.size(); ++i) { 7147 if (sessionId == mTracks[i]->sessionId()) { 7148 result |= TRACK_SESSION; 7149 break; 7150 } 7151 } 7152 7153 return result; 7154} 7155 7156KeyedVector<int, bool> AudioFlinger::RecordThread::sessionIds() const 7157{ 7158 KeyedVector<int, bool> ids; 7159 Mutex::Autolock _l(mLock); 7160 for (size_t j = 0; j < mTracks.size(); ++j) { 7161 sp<RecordThread::RecordTrack> track = mTracks[j]; 7162 int sessionId = track->sessionId(); 7163 if (ids.indexOfKey(sessionId) < 0) { 7164 ids.add(sessionId, true); 7165 } 7166 } 7167 return ids; 7168} 7169 7170AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 7171{ 7172 Mutex::Autolock _l(mLock); 7173 AudioStreamIn *input = mInput; 7174 mInput = NULL; 7175 return input; 7176} 7177 7178// this method must always be called either with ThreadBase mLock held or inside the thread loop 7179audio_stream_t* AudioFlinger::RecordThread::stream() const 7180{ 7181 if (mInput == NULL) { 7182 return NULL; 7183 } 7184 return &mInput->stream->common; 7185} 7186 7187status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 7188{ 7189 // only one chain per input thread 7190 if (mEffectChains.size() != 0) { 7191 ALOGW("addEffectChain_l() already one chain %p on thread %p", chain.get(), this); 7192 return INVALID_OPERATION; 7193 } 7194 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 7195 chain->setThread(this); 7196 chain->setInBuffer(NULL); 7197 chain->setOutBuffer(NULL); 7198 7199 checkSuspendOnAddEffectChain_l(chain); 7200 7201 // make sure enabled pre processing effects state is communicated to the HAL as we 7202 // just moved them to a new input stream. 7203 chain->syncHalEffectsState(); 7204 7205 mEffectChains.add(chain); 7206 7207 return NO_ERROR; 7208} 7209 7210size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 7211{ 7212 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 7213 ALOGW_IF(mEffectChains.size() != 1, 7214 "removeEffectChain_l() %p invalid chain size %d on thread %p", 7215 chain.get(), mEffectChains.size(), this); 7216 if (mEffectChains.size() == 1) { 7217 mEffectChains.removeAt(0); 7218 } 7219 return 0; 7220} 7221 7222status_t AudioFlinger::RecordThread::createAudioPatch_l(const struct audio_patch *patch, 7223 audio_patch_handle_t *handle) 7224{ 7225 status_t status = NO_ERROR; 7226 7227 // store new device and send to effects 7228 mInDevice = patch->sources[0].ext.device.type; 7229 mPatch = *patch; 7230 for (size_t i = 0; i < mEffectChains.size(); i++) { 7231 mEffectChains[i]->setDevice_l(mInDevice); 7232 } 7233 7234 // disable AEC and NS if the device is a BT SCO headset supporting those 7235 // pre processings 7236 if (mTracks.size() > 0) { 7237 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 7238 mAudioFlinger->btNrecIsOff(); 7239 for (size_t i = 0; i < mTracks.size(); i++) { 7240 sp<RecordTrack> track = mTracks[i]; 7241 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId()); 7242 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId()); 7243 } 7244 } 7245 7246 // store new source and send to effects 7247 if (mAudioSource != patch->sinks[0].ext.mix.usecase.source) { 7248 mAudioSource = patch->sinks[0].ext.mix.usecase.source; 7249 for (size_t i = 0; i < mEffectChains.size(); i++) { 7250 mEffectChains[i]->setAudioSource_l(mAudioSource); 7251 } 7252 } 7253 7254 if (mInput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 7255 audio_hw_device_t *hwDevice = mInput->audioHwDev->hwDevice(); 7256 status = hwDevice->create_audio_patch(hwDevice, 7257 patch->num_sources, 7258 patch->sources, 7259 patch->num_sinks, 7260 patch->sinks, 7261 handle); 7262 } else { 7263 char *address; 7264 if (strcmp(patch->sources[0].ext.device.address, "") != 0) { 7265 address = audio_device_address_to_parameter( 7266 patch->sources[0].ext.device.type, 7267 patch->sources[0].ext.device.address); 7268 } else { 7269 address = (char *)calloc(1, 1); 7270 } 7271 AudioParameter param = AudioParameter(String8(address)); 7272 free(address); 7273 param.addInt(String8(AUDIO_PARAMETER_STREAM_ROUTING), 7274 (int)patch->sources[0].ext.device.type); 7275 param.addInt(String8(AUDIO_PARAMETER_STREAM_INPUT_SOURCE), 7276 (int)patch->sinks[0].ext.mix.usecase.source); 7277 status = mInput->stream->common.set_parameters(&mInput->stream->common, 7278 param.toString().string()); 7279 *handle = AUDIO_PATCH_HANDLE_NONE; 7280 } 7281 7282 if (mInDevice != mPrevInDevice) { 7283 sendIoConfigEvent_l(AUDIO_INPUT_CONFIG_CHANGED); 7284 mPrevInDevice = mInDevice; 7285 } 7286 7287 return status; 7288} 7289 7290status_t AudioFlinger::RecordThread::releaseAudioPatch_l(const audio_patch_handle_t handle) 7291{ 7292 status_t status = NO_ERROR; 7293 7294 mInDevice = AUDIO_DEVICE_NONE; 7295 7296 if (mInput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 7297 audio_hw_device_t *hwDevice = mInput->audioHwDev->hwDevice(); 7298 status = hwDevice->release_audio_patch(hwDevice, handle); 7299 } else { 7300 AudioParameter param; 7301 param.addInt(String8(AUDIO_PARAMETER_STREAM_ROUTING), 0); 7302 status = mInput->stream->common.set_parameters(&mInput->stream->common, 7303 param.toString().string()); 7304 } 7305 return status; 7306} 7307 7308void AudioFlinger::RecordThread::addPatchRecord(const sp<PatchRecord>& record) 7309{ 7310 Mutex::Autolock _l(mLock); 7311 mTracks.add(record); 7312} 7313 7314void AudioFlinger::RecordThread::deletePatchRecord(const sp<PatchRecord>& record) 7315{ 7316 Mutex::Autolock _l(mLock); 7317 destroyTrack_l(record); 7318} 7319 7320void AudioFlinger::RecordThread::getAudioPortConfig(struct audio_port_config *config) 7321{ 7322 ThreadBase::getAudioPortConfig(config); 7323 config->role = AUDIO_PORT_ROLE_SINK; 7324 config->ext.mix.hw_module = mInput->audioHwDev->handle(); 7325 config->ext.mix.usecase.source = mAudioSource; 7326} 7327 7328} // namespace android 7329