Threads.cpp revision 58545be2ce4e701c8c37401edcc126a8b683890d
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_OUT_BUS, "BUS"}, 472 {AUDIO_DEVICE_NONE, "NONE"}, // must be last 473 }, mappingsIn[] = { 474 {AUDIO_DEVICE_IN_COMMUNICATION, "COMMUNICATION"}, 475 {AUDIO_DEVICE_IN_AMBIENT, "AMBIENT"}, 476 {AUDIO_DEVICE_IN_BUILTIN_MIC, "BUILTIN_MIC"}, 477 {AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET, "BLUETOOTH_SCO_HEADSET"}, 478 {AUDIO_DEVICE_IN_WIRED_HEADSET, "WIRED_HEADSET"}, 479 {AUDIO_DEVICE_IN_AUX_DIGITAL, "AUX_DIGITAL"}, 480 {AUDIO_DEVICE_IN_VOICE_CALL, "VOICE_CALL"}, 481 {AUDIO_DEVICE_IN_TELEPHONY_RX, "TELEPHONY_RX"}, 482 {AUDIO_DEVICE_IN_BACK_MIC, "BACK_MIC"}, 483 {AUDIO_DEVICE_IN_REMOTE_SUBMIX, "REMOTE_SUBMIX"}, 484 {AUDIO_DEVICE_IN_ANLG_DOCK_HEADSET, "ANLG_DOCK_HEADSET"}, 485 {AUDIO_DEVICE_IN_DGTL_DOCK_HEADSET, "DGTL_DOCK_HEADSET"}, 486 {AUDIO_DEVICE_IN_USB_ACCESSORY, "USB_ACCESSORY"}, 487 {AUDIO_DEVICE_IN_USB_DEVICE, "USB_DEVICE"}, 488 {AUDIO_DEVICE_IN_FM_TUNER, "FM_TUNER"}, 489 {AUDIO_DEVICE_IN_TV_TUNER, "TV_TUNER"}, 490 {AUDIO_DEVICE_IN_LINE, "LINE"}, 491 {AUDIO_DEVICE_IN_SPDIF, "SPDIF"}, 492 {AUDIO_DEVICE_IN_BLUETOOTH_A2DP, "BLUETOOTH_A2DP"}, 493 {AUDIO_DEVICE_IN_LOOPBACK, "LOOPBACK"}, 494 {AUDIO_DEVICE_IN_IP, "IP"}, 495 {AUDIO_DEVICE_IN_BUS, "BUS"}, 496 {AUDIO_DEVICE_NONE, "NONE"}, // must be last 497 }; 498 String8 result; 499 audio_devices_t allDevices = AUDIO_DEVICE_NONE; 500 const mapping *entry; 501 if (devices & AUDIO_DEVICE_BIT_IN) { 502 devices &= ~AUDIO_DEVICE_BIT_IN; 503 entry = mappingsIn; 504 } else { 505 entry = mappingsOut; 506 } 507 for ( ; entry->mDevices != AUDIO_DEVICE_NONE; entry++) { 508 allDevices = (audio_devices_t) (allDevices | entry->mDevices); 509 if (devices & entry->mDevices) { 510 if (!result.isEmpty()) { 511 result.append("|"); 512 } 513 result.append(entry->mString); 514 } 515 } 516 if (devices & ~allDevices) { 517 if (!result.isEmpty()) { 518 result.append("|"); 519 } 520 result.appendFormat("0x%X", devices & ~allDevices); 521 } 522 if (result.isEmpty()) { 523 result.append(entry->mString); 524 } 525 return result; 526} 527 528String8 inputFlagsToString(audio_input_flags_t flags) 529{ 530 static const struct mapping { 531 audio_input_flags_t mFlag; 532 const char * mString; 533 } mappings[] = { 534 {AUDIO_INPUT_FLAG_FAST, "FAST"}, 535 {AUDIO_INPUT_FLAG_HW_HOTWORD, "HW_HOTWORD"}, 536 {AUDIO_INPUT_FLAG_RAW, "RAW"}, 537 {AUDIO_INPUT_FLAG_SYNC, "SYNC"}, 538 {AUDIO_INPUT_FLAG_NONE, "NONE"}, // must be last 539 }; 540 String8 result; 541 audio_input_flags_t allFlags = AUDIO_INPUT_FLAG_NONE; 542 const mapping *entry; 543 for (entry = mappings; entry->mFlag != AUDIO_INPUT_FLAG_NONE; entry++) { 544 allFlags = (audio_input_flags_t) (allFlags | entry->mFlag); 545 if (flags & entry->mFlag) { 546 if (!result.isEmpty()) { 547 result.append("|"); 548 } 549 result.append(entry->mString); 550 } 551 } 552 if (flags & ~allFlags) { 553 if (!result.isEmpty()) { 554 result.append("|"); 555 } 556 result.appendFormat("0x%X", flags & ~allFlags); 557 } 558 if (result.isEmpty()) { 559 result.append(entry->mString); 560 } 561 return result; 562} 563 564String8 outputFlagsToString(audio_output_flags_t flags) 565{ 566 static const struct mapping { 567 audio_output_flags_t mFlag; 568 const char * mString; 569 } mappings[] = { 570 {AUDIO_OUTPUT_FLAG_DIRECT, "DIRECT"}, 571 {AUDIO_OUTPUT_FLAG_PRIMARY, "PRIMARY"}, 572 {AUDIO_OUTPUT_FLAG_FAST, "FAST"}, 573 {AUDIO_OUTPUT_FLAG_DEEP_BUFFER, "DEEP_BUFFER"}, 574 {AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD,"COMPRESS_OFFLOAD"}, 575 {AUDIO_OUTPUT_FLAG_NON_BLOCKING, "NON_BLOCKING"}, 576 {AUDIO_OUTPUT_FLAG_HW_AV_SYNC, "HW_AV_SYNC"}, 577 {AUDIO_OUTPUT_FLAG_RAW, "RAW"}, 578 {AUDIO_OUTPUT_FLAG_SYNC, "SYNC"}, 579 {AUDIO_OUTPUT_FLAG_IEC958_NONAUDIO, "IEC958_NONAUDIO"}, 580 {AUDIO_OUTPUT_FLAG_NONE, "NONE"}, // must be last 581 }; 582 String8 result; 583 audio_output_flags_t allFlags = AUDIO_OUTPUT_FLAG_NONE; 584 const mapping *entry; 585 for (entry = mappings; entry->mFlag != AUDIO_OUTPUT_FLAG_NONE; entry++) { 586 allFlags = (audio_output_flags_t) (allFlags | entry->mFlag); 587 if (flags & entry->mFlag) { 588 if (!result.isEmpty()) { 589 result.append("|"); 590 } 591 result.append(entry->mString); 592 } 593 } 594 if (flags & ~allFlags) { 595 if (!result.isEmpty()) { 596 result.append("|"); 597 } 598 result.appendFormat("0x%X", flags & ~allFlags); 599 } 600 if (result.isEmpty()) { 601 result.append(entry->mString); 602 } 603 return result; 604} 605 606const char *sourceToString(audio_source_t source) 607{ 608 switch (source) { 609 case AUDIO_SOURCE_DEFAULT: return "default"; 610 case AUDIO_SOURCE_MIC: return "mic"; 611 case AUDIO_SOURCE_VOICE_UPLINK: return "voice uplink"; 612 case AUDIO_SOURCE_VOICE_DOWNLINK: return "voice downlink"; 613 case AUDIO_SOURCE_VOICE_CALL: return "voice call"; 614 case AUDIO_SOURCE_CAMCORDER: return "camcorder"; 615 case AUDIO_SOURCE_VOICE_RECOGNITION: return "voice recognition"; 616 case AUDIO_SOURCE_VOICE_COMMUNICATION: return "voice communication"; 617 case AUDIO_SOURCE_REMOTE_SUBMIX: return "remote submix"; 618 case AUDIO_SOURCE_UNPROCESSED: return "unprocessed"; 619 case AUDIO_SOURCE_FM_TUNER: return "FM tuner"; 620 case AUDIO_SOURCE_HOTWORD: return "hotword"; 621 default: return "unknown"; 622 } 623} 624 625AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, 626 audio_devices_t outDevice, audio_devices_t inDevice, type_t type, bool systemReady) 627 : Thread(false /*canCallJava*/), 628 mType(type), 629 mAudioFlinger(audioFlinger), 630 // mSampleRate, mFrameCount, mChannelMask, mChannelCount, mFrameSize, mFormat, mBufferSize 631 // are set by PlaybackThread::readOutputParameters_l() or 632 // RecordThread::readInputParameters_l() 633 //FIXME: mStandby should be true here. Is this some kind of hack? 634 mStandby(false), mOutDevice(outDevice), mInDevice(inDevice), 635 mPrevOutDevice(AUDIO_DEVICE_NONE), mPrevInDevice(AUDIO_DEVICE_NONE), 636 mAudioSource(AUDIO_SOURCE_DEFAULT), mId(id), 637 // mName will be set by concrete (non-virtual) subclass 638 mDeathRecipient(new PMDeathRecipient(this)), 639 mSystemReady(systemReady), 640 mNotifiedBatteryStart(false) 641{ 642 memset(&mPatch, 0, sizeof(struct audio_patch)); 643} 644 645AudioFlinger::ThreadBase::~ThreadBase() 646{ 647 // mConfigEvents should be empty, but just in case it isn't, free the memory it owns 648 mConfigEvents.clear(); 649 650 // do not lock the mutex in destructor 651 releaseWakeLock_l(); 652 if (mPowerManager != 0) { 653 sp<IBinder> binder = IInterface::asBinder(mPowerManager); 654 binder->unlinkToDeath(mDeathRecipient); 655 } 656} 657 658status_t AudioFlinger::ThreadBase::readyToRun() 659{ 660 status_t status = initCheck(); 661 if (status == NO_ERROR) { 662 ALOGI("AudioFlinger's thread %p ready to run", this); 663 } else { 664 ALOGE("No working audio driver found."); 665 } 666 return status; 667} 668 669void AudioFlinger::ThreadBase::exit() 670{ 671 ALOGV("ThreadBase::exit"); 672 // do any cleanup required for exit to succeed 673 preExit(); 674 { 675 // This lock prevents the following race in thread (uniprocessor for illustration): 676 // if (!exitPending()) { 677 // // context switch from here to exit() 678 // // exit() calls requestExit(), what exitPending() observes 679 // // exit() calls signal(), which is dropped since no waiters 680 // // context switch back from exit() to here 681 // mWaitWorkCV.wait(...); 682 // // now thread is hung 683 // } 684 AutoMutex lock(mLock); 685 requestExit(); 686 mWaitWorkCV.broadcast(); 687 } 688 // When Thread::requestExitAndWait is made virtual and this method is renamed to 689 // "virtual status_t requestExitAndWait()", replace by "return Thread::requestExitAndWait();" 690 requestExitAndWait(); 691} 692 693status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs) 694{ 695 status_t status; 696 697 ALOGV("ThreadBase::setParameters() %s", keyValuePairs.string()); 698 Mutex::Autolock _l(mLock); 699 700 return sendSetParameterConfigEvent_l(keyValuePairs); 701} 702 703// sendConfigEvent_l() must be called with ThreadBase::mLock held 704// Can temporarily release the lock if waiting for a reply from processConfigEvents_l(). 705status_t AudioFlinger::ThreadBase::sendConfigEvent_l(sp<ConfigEvent>& event) 706{ 707 status_t status = NO_ERROR; 708 709 if (event->mRequiresSystemReady && !mSystemReady) { 710 event->mWaitStatus = false; 711 mPendingConfigEvents.add(event); 712 return status; 713 } 714 mConfigEvents.add(event); 715 ALOGV("sendConfigEvent_l() num events %d event %d", mConfigEvents.size(), event->mType); 716 mWaitWorkCV.signal(); 717 mLock.unlock(); 718 { 719 Mutex::Autolock _l(event->mLock); 720 while (event->mWaitStatus) { 721 if (event->mCond.waitRelative(event->mLock, kConfigEventTimeoutNs) != NO_ERROR) { 722 event->mStatus = TIMED_OUT; 723 event->mWaitStatus = false; 724 } 725 } 726 status = event->mStatus; 727 } 728 mLock.lock(); 729 return status; 730} 731 732void AudioFlinger::ThreadBase::sendIoConfigEvent(audio_io_config_event event, pid_t pid) 733{ 734 Mutex::Autolock _l(mLock); 735 sendIoConfigEvent_l(event, pid); 736} 737 738// sendIoConfigEvent_l() must be called with ThreadBase::mLock held 739void AudioFlinger::ThreadBase::sendIoConfigEvent_l(audio_io_config_event event, pid_t pid) 740{ 741 sp<ConfigEvent> configEvent = (ConfigEvent *)new IoConfigEvent(event, pid); 742 sendConfigEvent_l(configEvent); 743} 744 745void AudioFlinger::ThreadBase::sendPrioConfigEvent(pid_t pid, pid_t tid, int32_t prio) 746{ 747 Mutex::Autolock _l(mLock); 748 sendPrioConfigEvent_l(pid, tid, prio); 749} 750 751// sendPrioConfigEvent_l() must be called with ThreadBase::mLock held 752void AudioFlinger::ThreadBase::sendPrioConfigEvent_l(pid_t pid, pid_t tid, int32_t prio) 753{ 754 sp<ConfigEvent> configEvent = (ConfigEvent *)new PrioConfigEvent(pid, tid, prio); 755 sendConfigEvent_l(configEvent); 756} 757 758// sendSetParameterConfigEvent_l() must be called with ThreadBase::mLock held 759status_t AudioFlinger::ThreadBase::sendSetParameterConfigEvent_l(const String8& keyValuePair) 760{ 761 sp<ConfigEvent> configEvent; 762 AudioParameter param(keyValuePair); 763 int value; 764 if (param.getInt(String8(AUDIO_PARAMETER_MONO_OUTPUT), value) == NO_ERROR) { 765 setMasterMono_l(value != 0); 766 if (param.size() == 1) { 767 return NO_ERROR; // should be a solo parameter - we don't pass down 768 } 769 param.remove(String8(AUDIO_PARAMETER_MONO_OUTPUT)); 770 configEvent = new SetParameterConfigEvent(param.toString()); 771 } else { 772 configEvent = new SetParameterConfigEvent(keyValuePair); 773 } 774 return sendConfigEvent_l(configEvent); 775} 776 777status_t AudioFlinger::ThreadBase::sendCreateAudioPatchConfigEvent( 778 const struct audio_patch *patch, 779 audio_patch_handle_t *handle) 780{ 781 Mutex::Autolock _l(mLock); 782 sp<ConfigEvent> configEvent = (ConfigEvent *)new CreateAudioPatchConfigEvent(*patch, *handle); 783 status_t status = sendConfigEvent_l(configEvent); 784 if (status == NO_ERROR) { 785 CreateAudioPatchConfigEventData *data = 786 (CreateAudioPatchConfigEventData *)configEvent->mData.get(); 787 *handle = data->mHandle; 788 } 789 return status; 790} 791 792status_t AudioFlinger::ThreadBase::sendReleaseAudioPatchConfigEvent( 793 const audio_patch_handle_t handle) 794{ 795 Mutex::Autolock _l(mLock); 796 sp<ConfigEvent> configEvent = (ConfigEvent *)new ReleaseAudioPatchConfigEvent(handle); 797 return sendConfigEvent_l(configEvent); 798} 799 800 801// post condition: mConfigEvents.isEmpty() 802void AudioFlinger::ThreadBase::processConfigEvents_l() 803{ 804 bool configChanged = false; 805 806 while (!mConfigEvents.isEmpty()) { 807 ALOGV("processConfigEvents_l() remaining events %d", mConfigEvents.size()); 808 sp<ConfigEvent> event = mConfigEvents[0]; 809 mConfigEvents.removeAt(0); 810 switch (event->mType) { 811 case CFG_EVENT_PRIO: { 812 PrioConfigEventData *data = (PrioConfigEventData *)event->mData.get(); 813 // FIXME Need to understand why this has to be done asynchronously 814 int err = requestPriority(data->mPid, data->mTid, data->mPrio, 815 true /*asynchronous*/); 816 if (err != 0) { 817 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 818 data->mPrio, data->mPid, data->mTid, err); 819 } 820 } break; 821 case CFG_EVENT_IO: { 822 IoConfigEventData *data = (IoConfigEventData *)event->mData.get(); 823 ioConfigChanged(data->mEvent, data->mPid); 824 } break; 825 case CFG_EVENT_SET_PARAMETER: { 826 SetParameterConfigEventData *data = (SetParameterConfigEventData *)event->mData.get(); 827 if (checkForNewParameter_l(data->mKeyValuePairs, event->mStatus)) { 828 configChanged = true; 829 } 830 } break; 831 case CFG_EVENT_CREATE_AUDIO_PATCH: { 832 CreateAudioPatchConfigEventData *data = 833 (CreateAudioPatchConfigEventData *)event->mData.get(); 834 event->mStatus = createAudioPatch_l(&data->mPatch, &data->mHandle); 835 } break; 836 case CFG_EVENT_RELEASE_AUDIO_PATCH: { 837 ReleaseAudioPatchConfigEventData *data = 838 (ReleaseAudioPatchConfigEventData *)event->mData.get(); 839 event->mStatus = releaseAudioPatch_l(data->mHandle); 840 } break; 841 default: 842 ALOG_ASSERT(false, "processConfigEvents_l() unknown event type %d", event->mType); 843 break; 844 } 845 { 846 Mutex::Autolock _l(event->mLock); 847 if (event->mWaitStatus) { 848 event->mWaitStatus = false; 849 event->mCond.signal(); 850 } 851 } 852 ALOGV_IF(mConfigEvents.isEmpty(), "processConfigEvents_l() DONE thread %p", this); 853 } 854 855 if (configChanged) { 856 cacheParameters_l(); 857 } 858} 859 860String8 channelMaskToString(audio_channel_mask_t mask, bool output) { 861 String8 s; 862 const audio_channel_representation_t representation = 863 audio_channel_mask_get_representation(mask); 864 865 switch (representation) { 866 case AUDIO_CHANNEL_REPRESENTATION_POSITION: { 867 if (output) { 868 if (mask & AUDIO_CHANNEL_OUT_FRONT_LEFT) s.append("front-left, "); 869 if (mask & AUDIO_CHANNEL_OUT_FRONT_RIGHT) s.append("front-right, "); 870 if (mask & AUDIO_CHANNEL_OUT_FRONT_CENTER) s.append("front-center, "); 871 if (mask & AUDIO_CHANNEL_OUT_LOW_FREQUENCY) s.append("low freq, "); 872 if (mask & AUDIO_CHANNEL_OUT_BACK_LEFT) s.append("back-left, "); 873 if (mask & AUDIO_CHANNEL_OUT_BACK_RIGHT) s.append("back-right, "); 874 if (mask & AUDIO_CHANNEL_OUT_FRONT_LEFT_OF_CENTER) s.append("front-left-of-center, "); 875 if (mask & AUDIO_CHANNEL_OUT_FRONT_RIGHT_OF_CENTER) s.append("front-right-of-center, "); 876 if (mask & AUDIO_CHANNEL_OUT_BACK_CENTER) s.append("back-center, "); 877 if (mask & AUDIO_CHANNEL_OUT_SIDE_LEFT) s.append("side-left, "); 878 if (mask & AUDIO_CHANNEL_OUT_SIDE_RIGHT) s.append("side-right, "); 879 if (mask & AUDIO_CHANNEL_OUT_TOP_CENTER) s.append("top-center ,"); 880 if (mask & AUDIO_CHANNEL_OUT_TOP_FRONT_LEFT) s.append("top-front-left, "); 881 if (mask & AUDIO_CHANNEL_OUT_TOP_FRONT_CENTER) s.append("top-front-center, "); 882 if (mask & AUDIO_CHANNEL_OUT_TOP_FRONT_RIGHT) s.append("top-front-right, "); 883 if (mask & AUDIO_CHANNEL_OUT_TOP_BACK_LEFT) s.append("top-back-left, "); 884 if (mask & AUDIO_CHANNEL_OUT_TOP_BACK_CENTER) s.append("top-back-center, " ); 885 if (mask & AUDIO_CHANNEL_OUT_TOP_BACK_RIGHT) s.append("top-back-right, " ); 886 if (mask & ~AUDIO_CHANNEL_OUT_ALL) s.append("unknown, "); 887 } else { 888 if (mask & AUDIO_CHANNEL_IN_LEFT) s.append("left, "); 889 if (mask & AUDIO_CHANNEL_IN_RIGHT) s.append("right, "); 890 if (mask & AUDIO_CHANNEL_IN_FRONT) s.append("front, "); 891 if (mask & AUDIO_CHANNEL_IN_BACK) s.append("back, "); 892 if (mask & AUDIO_CHANNEL_IN_LEFT_PROCESSED) s.append("left-processed, "); 893 if (mask & AUDIO_CHANNEL_IN_RIGHT_PROCESSED) s.append("right-processed, "); 894 if (mask & AUDIO_CHANNEL_IN_FRONT_PROCESSED) s.append("front-processed, "); 895 if (mask & AUDIO_CHANNEL_IN_BACK_PROCESSED) s.append("back-processed, "); 896 if (mask & AUDIO_CHANNEL_IN_PRESSURE) s.append("pressure, "); 897 if (mask & AUDIO_CHANNEL_IN_X_AXIS) s.append("X, "); 898 if (mask & AUDIO_CHANNEL_IN_Y_AXIS) s.append("Y, "); 899 if (mask & AUDIO_CHANNEL_IN_Z_AXIS) s.append("Z, "); 900 if (mask & AUDIO_CHANNEL_IN_VOICE_UPLINK) s.append("voice-uplink, "); 901 if (mask & AUDIO_CHANNEL_IN_VOICE_DNLINK) s.append("voice-dnlink, "); 902 if (mask & ~AUDIO_CHANNEL_IN_ALL) s.append("unknown, "); 903 } 904 const int len = s.length(); 905 if (len > 2) { 906 char *str = s.lockBuffer(len); // needed? 907 s.unlockBuffer(len - 2); // remove trailing ", " 908 } 909 return s; 910 } 911 case AUDIO_CHANNEL_REPRESENTATION_INDEX: 912 s.appendFormat("index mask, bits:%#x", audio_channel_mask_get_bits(mask)); 913 return s; 914 default: 915 s.appendFormat("unknown mask, representation:%d bits:%#x", 916 representation, audio_channel_mask_get_bits(mask)); 917 return s; 918 } 919} 920 921void AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args __unused) 922{ 923 const size_t SIZE = 256; 924 char buffer[SIZE]; 925 String8 result; 926 927 bool locked = AudioFlinger::dumpTryLock(mLock); 928 if (!locked) { 929 dprintf(fd, "thread %p may be deadlocked\n", this); 930 } 931 932 dprintf(fd, " Thread name: %s\n", mThreadName); 933 dprintf(fd, " I/O handle: %d\n", mId); 934 dprintf(fd, " TID: %d\n", getTid()); 935 dprintf(fd, " Standby: %s\n", mStandby ? "yes" : "no"); 936 dprintf(fd, " Sample rate: %u Hz\n", mSampleRate); 937 dprintf(fd, " HAL frame count: %zu\n", mFrameCount); 938 dprintf(fd, " HAL format: 0x%x (%s)\n", mHALFormat, formatToString(mHALFormat)); 939 dprintf(fd, " HAL buffer size: %u bytes\n", mBufferSize); 940 dprintf(fd, " Channel count: %u\n", mChannelCount); 941 dprintf(fd, " Channel mask: 0x%08x (%s)\n", mChannelMask, 942 channelMaskToString(mChannelMask, mType != RECORD).string()); 943 dprintf(fd, " Processing format: 0x%x (%s)\n", mFormat, formatToString(mFormat)); 944 dprintf(fd, " Processing frame size: %zu bytes\n", mFrameSize); 945 dprintf(fd, " Pending config events:"); 946 size_t numConfig = mConfigEvents.size(); 947 if (numConfig) { 948 for (size_t i = 0; i < numConfig; i++) { 949 mConfigEvents[i]->dump(buffer, SIZE); 950 dprintf(fd, "\n %s", buffer); 951 } 952 dprintf(fd, "\n"); 953 } else { 954 dprintf(fd, " none\n"); 955 } 956 dprintf(fd, " Output device: %#x (%s)\n", mOutDevice, devicesToString(mOutDevice).string()); 957 dprintf(fd, " Input device: %#x (%s)\n", mInDevice, devicesToString(mInDevice).string()); 958 dprintf(fd, " Audio source: %d (%s)\n", mAudioSource, sourceToString(mAudioSource)); 959 960 if (locked) { 961 mLock.unlock(); 962 } 963} 964 965void AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args) 966{ 967 const size_t SIZE = 256; 968 char buffer[SIZE]; 969 String8 result; 970 971 size_t numEffectChains = mEffectChains.size(); 972 snprintf(buffer, SIZE, " %zu Effect Chains\n", numEffectChains); 973 write(fd, buffer, strlen(buffer)); 974 975 for (size_t i = 0; i < numEffectChains; ++i) { 976 sp<EffectChain> chain = mEffectChains[i]; 977 if (chain != 0) { 978 chain->dump(fd, args); 979 } 980 } 981} 982 983void AudioFlinger::ThreadBase::acquireWakeLock(int uid) 984{ 985 Mutex::Autolock _l(mLock); 986 acquireWakeLock_l(uid); 987} 988 989String16 AudioFlinger::ThreadBase::getWakeLockTag() 990{ 991 switch (mType) { 992 case MIXER: 993 return String16("AudioMix"); 994 case DIRECT: 995 return String16("AudioDirectOut"); 996 case DUPLICATING: 997 return String16("AudioDup"); 998 case RECORD: 999 return String16("AudioIn"); 1000 case OFFLOAD: 1001 return String16("AudioOffload"); 1002 default: 1003 ALOG_ASSERT(false); 1004 return String16("AudioUnknown"); 1005 } 1006} 1007 1008void AudioFlinger::ThreadBase::acquireWakeLock_l(int uid) 1009{ 1010 getPowerManager_l(); 1011 if (mPowerManager != 0) { 1012 sp<IBinder> binder = new BBinder(); 1013 status_t status; 1014 if (uid >= 0) { 1015 status = mPowerManager->acquireWakeLockWithUid(POWERMANAGER_PARTIAL_WAKE_LOCK, 1016 binder, 1017 getWakeLockTag(), 1018 String16("audioserver"), 1019 uid, 1020 true /* FIXME force oneway contrary to .aidl */); 1021 } else { 1022 status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK, 1023 binder, 1024 getWakeLockTag(), 1025 String16("audioserver"), 1026 true /* FIXME force oneway contrary to .aidl */); 1027 } 1028 if (status == NO_ERROR) { 1029 mWakeLockToken = binder; 1030 } 1031 ALOGV("acquireWakeLock_l() %s status %d", mThreadName, status); 1032 } 1033 1034 if (!mNotifiedBatteryStart) { 1035 BatteryNotifier::getInstance().noteStartAudio(); 1036 mNotifiedBatteryStart = true; 1037 } 1038 gBoottime.acquire(mWakeLockToken); 1039} 1040 1041void AudioFlinger::ThreadBase::releaseWakeLock() 1042{ 1043 Mutex::Autolock _l(mLock); 1044 releaseWakeLock_l(); 1045} 1046 1047void AudioFlinger::ThreadBase::releaseWakeLock_l() 1048{ 1049 gBoottime.release(mWakeLockToken); 1050 if (mWakeLockToken != 0) { 1051 ALOGV("releaseWakeLock_l() %s", mThreadName); 1052 if (mPowerManager != 0) { 1053 mPowerManager->releaseWakeLock(mWakeLockToken, 0, 1054 true /* FIXME force oneway contrary to .aidl */); 1055 } 1056 mWakeLockToken.clear(); 1057 } 1058 1059 if (mNotifiedBatteryStart) { 1060 BatteryNotifier::getInstance().noteStopAudio(); 1061 mNotifiedBatteryStart = false; 1062 } 1063} 1064 1065void AudioFlinger::ThreadBase::updateWakeLockUids(const SortedVector<int> &uids) { 1066 Mutex::Autolock _l(mLock); 1067 updateWakeLockUids_l(uids); 1068} 1069 1070void AudioFlinger::ThreadBase::getPowerManager_l() { 1071 if (mSystemReady && mPowerManager == 0) { 1072 // use checkService() to avoid blocking if power service is not up yet 1073 sp<IBinder> binder = 1074 defaultServiceManager()->checkService(String16("power")); 1075 if (binder == 0) { 1076 ALOGW("Thread %s cannot connect to the power manager service", mThreadName); 1077 } else { 1078 mPowerManager = interface_cast<IPowerManager>(binder); 1079 binder->linkToDeath(mDeathRecipient); 1080 } 1081 } 1082} 1083 1084void AudioFlinger::ThreadBase::updateWakeLockUids_l(const SortedVector<int> &uids) { 1085 getPowerManager_l(); 1086 if (mWakeLockToken == NULL) { // token may be NULL if AudioFlinger::systemReady() not called. 1087 if (mSystemReady) { 1088 ALOGE("no wake lock to update, but system ready!"); 1089 } else { 1090 ALOGW("no wake lock to update, system not ready yet"); 1091 } 1092 return; 1093 } 1094 if (mPowerManager != 0) { 1095 sp<IBinder> binder = new BBinder(); 1096 status_t status; 1097 status = mPowerManager->updateWakeLockUids(mWakeLockToken, uids.size(), uids.array(), 1098 true /* FIXME force oneway contrary to .aidl */); 1099 ALOGV("acquireWakeLock_l() %s status %d", mThreadName, status); 1100 } 1101} 1102 1103void AudioFlinger::ThreadBase::clearPowerManager() 1104{ 1105 Mutex::Autolock _l(mLock); 1106 releaseWakeLock_l(); 1107 mPowerManager.clear(); 1108} 1109 1110void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who __unused) 1111{ 1112 sp<ThreadBase> thread = mThread.promote(); 1113 if (thread != 0) { 1114 thread->clearPowerManager(); 1115 } 1116 ALOGW("power manager service died !!!"); 1117} 1118 1119void AudioFlinger::ThreadBase::setEffectSuspended( 1120 const effect_uuid_t *type, bool suspend, int sessionId) 1121{ 1122 Mutex::Autolock _l(mLock); 1123 setEffectSuspended_l(type, suspend, sessionId); 1124} 1125 1126void AudioFlinger::ThreadBase::setEffectSuspended_l( 1127 const effect_uuid_t *type, bool suspend, int sessionId) 1128{ 1129 sp<EffectChain> chain = getEffectChain_l(sessionId); 1130 if (chain != 0) { 1131 if (type != NULL) { 1132 chain->setEffectSuspended_l(type, suspend); 1133 } else { 1134 chain->setEffectSuspendedAll_l(suspend); 1135 } 1136 } 1137 1138 updateSuspendedSessions_l(type, suspend, sessionId); 1139} 1140 1141void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain) 1142{ 1143 ssize_t index = mSuspendedSessions.indexOfKey(chain->sessionId()); 1144 if (index < 0) { 1145 return; 1146 } 1147 1148 const KeyedVector <int, sp<SuspendedSessionDesc> >& sessionEffects = 1149 mSuspendedSessions.valueAt(index); 1150 1151 for (size_t i = 0; i < sessionEffects.size(); i++) { 1152 sp<SuspendedSessionDesc> desc = sessionEffects.valueAt(i); 1153 for (int j = 0; j < desc->mRefCount; j++) { 1154 if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) { 1155 chain->setEffectSuspendedAll_l(true); 1156 } else { 1157 ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x", 1158 desc->mType.timeLow); 1159 chain->setEffectSuspended_l(&desc->mType, true); 1160 } 1161 } 1162 } 1163} 1164 1165void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type, 1166 bool suspend, 1167 int sessionId) 1168{ 1169 ssize_t index = mSuspendedSessions.indexOfKey(sessionId); 1170 1171 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects; 1172 1173 if (suspend) { 1174 if (index >= 0) { 1175 sessionEffects = mSuspendedSessions.valueAt(index); 1176 } else { 1177 mSuspendedSessions.add(sessionId, sessionEffects); 1178 } 1179 } else { 1180 if (index < 0) { 1181 return; 1182 } 1183 sessionEffects = mSuspendedSessions.valueAt(index); 1184 } 1185 1186 1187 int key = EffectChain::kKeyForSuspendAll; 1188 if (type != NULL) { 1189 key = type->timeLow; 1190 } 1191 index = sessionEffects.indexOfKey(key); 1192 1193 sp<SuspendedSessionDesc> desc; 1194 if (suspend) { 1195 if (index >= 0) { 1196 desc = sessionEffects.valueAt(index); 1197 } else { 1198 desc = new SuspendedSessionDesc(); 1199 if (type != NULL) { 1200 desc->mType = *type; 1201 } 1202 sessionEffects.add(key, desc); 1203 ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key); 1204 } 1205 desc->mRefCount++; 1206 } else { 1207 if (index < 0) { 1208 return; 1209 } 1210 desc = sessionEffects.valueAt(index); 1211 if (--desc->mRefCount == 0) { 1212 ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key); 1213 sessionEffects.removeItemsAt(index); 1214 if (sessionEffects.isEmpty()) { 1215 ALOGV("updateSuspendedSessions_l() restore removing session %d", 1216 sessionId); 1217 mSuspendedSessions.removeItem(sessionId); 1218 } 1219 } 1220 } 1221 if (!sessionEffects.isEmpty()) { 1222 mSuspendedSessions.replaceValueFor(sessionId, sessionEffects); 1223 } 1224} 1225 1226void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 1227 bool enabled, 1228 int sessionId) 1229{ 1230 Mutex::Autolock _l(mLock); 1231 checkSuspendOnEffectEnabled_l(effect, enabled, sessionId); 1232} 1233 1234void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 1235 bool enabled, 1236 int sessionId) 1237{ 1238 if (mType != RECORD) { 1239 // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on 1240 // another session. This gives the priority to well behaved effect control panels 1241 // and applications not using global effects. 1242 // Enabling post processing in AUDIO_SESSION_OUTPUT_STAGE session does not affect 1243 // global effects 1244 if ((sessionId != AUDIO_SESSION_OUTPUT_MIX) && (sessionId != AUDIO_SESSION_OUTPUT_STAGE)) { 1245 setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX); 1246 } 1247 } 1248 1249 sp<EffectChain> chain = getEffectChain_l(sessionId); 1250 if (chain != 0) { 1251 chain->checkSuspendOnEffectEnabled(effect, enabled); 1252 } 1253} 1254 1255// ThreadBase::createEffect_l() must be called with AudioFlinger::mLock held 1256sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 1257 const sp<AudioFlinger::Client>& client, 1258 const sp<IEffectClient>& effectClient, 1259 int32_t priority, 1260 int sessionId, 1261 effect_descriptor_t *desc, 1262 int *enabled, 1263 status_t *status) 1264{ 1265 sp<EffectModule> effect; 1266 sp<EffectHandle> handle; 1267 status_t lStatus; 1268 sp<EffectChain> chain; 1269 bool chainCreated = false; 1270 bool effectCreated = false; 1271 bool effectRegistered = false; 1272 1273 lStatus = initCheck(); 1274 if (lStatus != NO_ERROR) { 1275 ALOGW("createEffect_l() Audio driver not initialized."); 1276 goto Exit; 1277 } 1278 1279 // Reject any effect on Direct output threads for now, since the format of 1280 // mSinkBuffer is not guaranteed to be compatible with effect processing (PCM 16 stereo). 1281 if (mType == DIRECT) { 1282 ALOGW("createEffect_l() Cannot add effect %s on Direct output type thread %s", 1283 desc->name, mThreadName); 1284 lStatus = BAD_VALUE; 1285 goto Exit; 1286 } 1287 1288 // Reject any effect on mixer or duplicating multichannel sinks. 1289 // TODO: fix both format and multichannel issues with effects. 1290 if ((mType == MIXER || mType == DUPLICATING) && mChannelCount != FCC_2) { 1291 ALOGW("createEffect_l() Cannot add effect %s for multichannel(%d) %s threads", 1292 desc->name, mChannelCount, mType == MIXER ? "MIXER" : "DUPLICATING"); 1293 lStatus = BAD_VALUE; 1294 goto Exit; 1295 } 1296 1297 // Allow global effects only on offloaded and mixer threads 1298 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 1299 switch (mType) { 1300 case MIXER: 1301 case OFFLOAD: 1302 break; 1303 case DIRECT: 1304 case DUPLICATING: 1305 case RECORD: 1306 default: 1307 ALOGW("createEffect_l() Cannot add global effect %s on thread %s", 1308 desc->name, mThreadName); 1309 lStatus = BAD_VALUE; 1310 goto Exit; 1311 } 1312 } 1313 1314 // Only Pre processor effects are allowed on input threads and only on input threads 1315 if ((mType == RECORD) != ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { 1316 ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d", 1317 desc->name, desc->flags, mType); 1318 lStatus = BAD_VALUE; 1319 goto Exit; 1320 } 1321 1322 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 1323 1324 { // scope for mLock 1325 Mutex::Autolock _l(mLock); 1326 1327 // check for existing effect chain with the requested audio session 1328 chain = getEffectChain_l(sessionId); 1329 if (chain == 0) { 1330 // create a new chain for this session 1331 ALOGV("createEffect_l() new effect chain for session %d", sessionId); 1332 chain = new EffectChain(this, sessionId); 1333 addEffectChain_l(chain); 1334 chain->setStrategy(getStrategyForSession_l(sessionId)); 1335 chainCreated = true; 1336 } else { 1337 effect = chain->getEffectFromDesc_l(desc); 1338 } 1339 1340 ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get()); 1341 1342 if (effect == 0) { 1343 int id = mAudioFlinger->nextUniqueId(); 1344 // Check CPU and memory usage 1345 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 1346 if (lStatus != NO_ERROR) { 1347 goto Exit; 1348 } 1349 effectRegistered = true; 1350 // create a new effect module if none present in the chain 1351 effect = new EffectModule(this, chain, desc, id, sessionId); 1352 lStatus = effect->status(); 1353 if (lStatus != NO_ERROR) { 1354 goto Exit; 1355 } 1356 effect->setOffloaded(mType == OFFLOAD, mId); 1357 1358 lStatus = chain->addEffect_l(effect); 1359 if (lStatus != NO_ERROR) { 1360 goto Exit; 1361 } 1362 effectCreated = true; 1363 1364 effect->setDevice(mOutDevice); 1365 effect->setDevice(mInDevice); 1366 effect->setMode(mAudioFlinger->getMode()); 1367 effect->setAudioSource(mAudioSource); 1368 } 1369 // create effect handle and connect it to effect module 1370 handle = new EffectHandle(effect, client, effectClient, priority); 1371 lStatus = handle->initCheck(); 1372 if (lStatus == OK) { 1373 lStatus = effect->addHandle(handle.get()); 1374 } 1375 if (enabled != NULL) { 1376 *enabled = (int)effect->isEnabled(); 1377 } 1378 } 1379 1380Exit: 1381 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 1382 Mutex::Autolock _l(mLock); 1383 if (effectCreated) { 1384 chain->removeEffect_l(effect); 1385 } 1386 if (effectRegistered) { 1387 AudioSystem::unregisterEffect(effect->id()); 1388 } 1389 if (chainCreated) { 1390 removeEffectChain_l(chain); 1391 } 1392 handle.clear(); 1393 } 1394 1395 *status = lStatus; 1396 return handle; 1397} 1398 1399sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect(int sessionId, int effectId) 1400{ 1401 Mutex::Autolock _l(mLock); 1402 return getEffect_l(sessionId, effectId); 1403} 1404 1405sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId) 1406{ 1407 sp<EffectChain> chain = getEffectChain_l(sessionId); 1408 return chain != 0 ? chain->getEffectFromId_l(effectId) : 0; 1409} 1410 1411// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 1412// PlaybackThread::mLock held 1413status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 1414{ 1415 // check for existing effect chain with the requested audio session 1416 int sessionId = effect->sessionId(); 1417 sp<EffectChain> chain = getEffectChain_l(sessionId); 1418 bool chainCreated = false; 1419 1420 ALOGD_IF((mType == OFFLOAD) && !effect->isOffloadable(), 1421 "addEffect_l() on offloaded thread %p: effect %s does not support offload flags %x", 1422 this, effect->desc().name, effect->desc().flags); 1423 1424 if (chain == 0) { 1425 // create a new chain for this session 1426 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 1427 chain = new EffectChain(this, sessionId); 1428 addEffectChain_l(chain); 1429 chain->setStrategy(getStrategyForSession_l(sessionId)); 1430 chainCreated = true; 1431 } 1432 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 1433 1434 if (chain->getEffectFromId_l(effect->id()) != 0) { 1435 ALOGW("addEffect_l() %p effect %s already present in chain %p", 1436 this, effect->desc().name, chain.get()); 1437 return BAD_VALUE; 1438 } 1439 1440 effect->setOffloaded(mType == OFFLOAD, mId); 1441 1442 status_t status = chain->addEffect_l(effect); 1443 if (status != NO_ERROR) { 1444 if (chainCreated) { 1445 removeEffectChain_l(chain); 1446 } 1447 return status; 1448 } 1449 1450 effect->setDevice(mOutDevice); 1451 effect->setDevice(mInDevice); 1452 effect->setMode(mAudioFlinger->getMode()); 1453 effect->setAudioSource(mAudioSource); 1454 return NO_ERROR; 1455} 1456 1457void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 1458 1459 ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 1460 effect_descriptor_t desc = effect->desc(); 1461 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 1462 detachAuxEffect_l(effect->id()); 1463 } 1464 1465 sp<EffectChain> chain = effect->chain().promote(); 1466 if (chain != 0) { 1467 // remove effect chain if removing last effect 1468 if (chain->removeEffect_l(effect) == 0) { 1469 removeEffectChain_l(chain); 1470 } 1471 } else { 1472 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 1473 } 1474} 1475 1476void AudioFlinger::ThreadBase::lockEffectChains_l( 1477 Vector< sp<AudioFlinger::EffectChain> >& effectChains) 1478{ 1479 effectChains = mEffectChains; 1480 for (size_t i = 0; i < mEffectChains.size(); i++) { 1481 mEffectChains[i]->lock(); 1482 } 1483} 1484 1485void AudioFlinger::ThreadBase::unlockEffectChains( 1486 const Vector< sp<AudioFlinger::EffectChain> >& effectChains) 1487{ 1488 for (size_t i = 0; i < effectChains.size(); i++) { 1489 effectChains[i]->unlock(); 1490 } 1491} 1492 1493sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId) 1494{ 1495 Mutex::Autolock _l(mLock); 1496 return getEffectChain_l(sessionId); 1497} 1498 1499sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) const 1500{ 1501 size_t size = mEffectChains.size(); 1502 for (size_t i = 0; i < size; i++) { 1503 if (mEffectChains[i]->sessionId() == sessionId) { 1504 return mEffectChains[i]; 1505 } 1506 } 1507 return 0; 1508} 1509 1510void AudioFlinger::ThreadBase::setMode(audio_mode_t mode) 1511{ 1512 Mutex::Autolock _l(mLock); 1513 size_t size = mEffectChains.size(); 1514 for (size_t i = 0; i < size; i++) { 1515 mEffectChains[i]->setMode_l(mode); 1516 } 1517} 1518 1519void AudioFlinger::ThreadBase::getAudioPortConfig(struct audio_port_config *config) 1520{ 1521 config->type = AUDIO_PORT_TYPE_MIX; 1522 config->ext.mix.handle = mId; 1523 config->sample_rate = mSampleRate; 1524 config->format = mFormat; 1525 config->channel_mask = mChannelMask; 1526 config->config_mask = AUDIO_PORT_CONFIG_SAMPLE_RATE|AUDIO_PORT_CONFIG_CHANNEL_MASK| 1527 AUDIO_PORT_CONFIG_FORMAT; 1528} 1529 1530void AudioFlinger::ThreadBase::systemReady() 1531{ 1532 Mutex::Autolock _l(mLock); 1533 if (mSystemReady) { 1534 return; 1535 } 1536 mSystemReady = true; 1537 1538 for (size_t i = 0; i < mPendingConfigEvents.size(); i++) { 1539 sendConfigEvent_l(mPendingConfigEvents.editItemAt(i)); 1540 } 1541 mPendingConfigEvents.clear(); 1542} 1543 1544 1545// ---------------------------------------------------------------------------- 1546// Playback 1547// ---------------------------------------------------------------------------- 1548 1549AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, 1550 AudioStreamOut* output, 1551 audio_io_handle_t id, 1552 audio_devices_t device, 1553 type_t type, 1554 bool systemReady) 1555 : ThreadBase(audioFlinger, id, device, AUDIO_DEVICE_NONE, type, systemReady), 1556 mNormalFrameCount(0), mSinkBuffer(NULL), 1557 mMixerBufferEnabled(AudioFlinger::kEnableExtendedPrecision), 1558 mMixerBuffer(NULL), 1559 mMixerBufferSize(0), 1560 mMixerBufferFormat(AUDIO_FORMAT_INVALID), 1561 mMixerBufferValid(false), 1562 mEffectBufferEnabled(AudioFlinger::kEnableExtendedPrecision), 1563 mEffectBuffer(NULL), 1564 mEffectBufferSize(0), 1565 mEffectBufferFormat(AUDIO_FORMAT_INVALID), 1566 mEffectBufferValid(false), 1567 mSuspended(0), mBytesWritten(0), 1568 mActiveTracksGeneration(0), 1569 // mStreamTypes[] initialized in constructor body 1570 mOutput(output), 1571 mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false), 1572 mMixerStatus(MIXER_IDLE), 1573 mMixerStatusIgnoringFastTracks(MIXER_IDLE), 1574 mStandbyDelayNs(AudioFlinger::mStandbyTimeInNsecs), 1575 mBytesRemaining(0), 1576 mCurrentWriteLength(0), 1577 mUseAsyncWrite(false), 1578 mWriteAckSequence(0), 1579 mDrainSequence(0), 1580 mSignalPending(false), 1581 mScreenState(AudioFlinger::mScreenState), 1582 // index 0 is reserved for normal mixer's submix 1583 mFastTrackAvailMask(((1 << FastMixerState::kMaxFastTracks) - 1) & ~1), 1584 mHwSupportsPause(false), mHwPaused(false), mFlushPending(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_has_proportional_frames(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)) { // TODO maybe use audio_has_proportional_frames()? 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 // otherwise use the HAL / AudioStreamOut directly 2558 } else { 2559 // Direct output and offload threads 2560 2561 if (mUseAsyncWrite) { 2562 ALOGW_IF(mWriteAckSequence & 1, "threadLoop_write(): out of sequence write request"); 2563 mWriteAckSequence += 2; 2564 mWriteAckSequence |= 1; 2565 ALOG_ASSERT(mCallbackThread != 0); 2566 mCallbackThread->setWriteBlocked(mWriteAckSequence); 2567 } 2568 // FIXME We should have an implementation of timestamps for direct output threads. 2569 // They are used e.g for multichannel PCM playback over HDMI. 2570 bytesWritten = mOutput->write((char *)mSinkBuffer + offset, mBytesRemaining); 2571 if (mUseAsyncWrite && 2572 ((bytesWritten < 0) || (bytesWritten == (ssize_t)mBytesRemaining))) { 2573 // do not wait for async callback in case of error of full write 2574 mWriteAckSequence &= ~1; 2575 ALOG_ASSERT(mCallbackThread != 0); 2576 mCallbackThread->setWriteBlocked(mWriteAckSequence); 2577 } 2578 } 2579 2580 mNumWrites++; 2581 mInWrite = false; 2582 mStandby = false; 2583 return bytesWritten; 2584} 2585 2586void AudioFlinger::PlaybackThread::threadLoop_drain() 2587{ 2588 if (mOutput->stream->drain) { 2589 ALOGV("draining %s", (mMixerStatus == MIXER_DRAIN_TRACK) ? "early" : "full"); 2590 if (mUseAsyncWrite) { 2591 ALOGW_IF(mDrainSequence & 1, "threadLoop_drain(): out of sequence drain request"); 2592 mDrainSequence |= 1; 2593 ALOG_ASSERT(mCallbackThread != 0); 2594 mCallbackThread->setDraining(mDrainSequence); 2595 } 2596 mOutput->stream->drain(mOutput->stream, 2597 (mMixerStatus == MIXER_DRAIN_TRACK) ? AUDIO_DRAIN_EARLY_NOTIFY 2598 : AUDIO_DRAIN_ALL); 2599 } 2600} 2601 2602void AudioFlinger::PlaybackThread::threadLoop_exit() 2603{ 2604 { 2605 Mutex::Autolock _l(mLock); 2606 for (size_t i = 0; i < mTracks.size(); i++) { 2607 sp<Track> track = mTracks[i]; 2608 track->invalidate(); 2609 } 2610 } 2611} 2612 2613/* 2614The derived values that are cached: 2615 - mSinkBufferSize from frame count * frame size 2616 - mActiveSleepTimeUs from activeSleepTimeUs() 2617 - mIdleSleepTimeUs from idleSleepTimeUs() 2618 - mStandbyDelayNs from mActiveSleepTimeUs (DIRECT only) or forced to at least 2619 kDefaultStandbyTimeInNsecs when connected to an A2DP device. 2620 - maxPeriod from frame count and sample rate (MIXER only) 2621 2622The parameters that affect these derived values are: 2623 - frame count 2624 - frame size 2625 - sample rate 2626 - device type: A2DP or not 2627 - device latency 2628 - format: PCM or not 2629 - active sleep time 2630 - idle sleep time 2631*/ 2632 2633void AudioFlinger::PlaybackThread::cacheParameters_l() 2634{ 2635 mSinkBufferSize = mNormalFrameCount * mFrameSize; 2636 mActiveSleepTimeUs = activeSleepTimeUs(); 2637 mIdleSleepTimeUs = idleSleepTimeUs(); 2638 2639 // make sure standby delay is not too short when connected to an A2DP sink to avoid 2640 // truncating audio when going to standby. 2641 mStandbyDelayNs = AudioFlinger::mStandbyTimeInNsecs; 2642 if ((mOutDevice & AUDIO_DEVICE_OUT_ALL_A2DP) != 0) { 2643 if (mStandbyDelayNs < kDefaultStandbyTimeInNsecs) { 2644 mStandbyDelayNs = kDefaultStandbyTimeInNsecs; 2645 } 2646 } 2647} 2648 2649void AudioFlinger::PlaybackThread::invalidateTracks(audio_stream_type_t streamType) 2650{ 2651 ALOGV("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 2652 this, streamType, mTracks.size()); 2653 Mutex::Autolock _l(mLock); 2654 2655 size_t size = mTracks.size(); 2656 for (size_t i = 0; i < size; i++) { 2657 sp<Track> t = mTracks[i]; 2658 if (t->streamType() == streamType && t->isExternalTrack()) { 2659 t->invalidate(); 2660 } 2661 } 2662} 2663 2664status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 2665{ 2666 int session = chain->sessionId(); 2667 int16_t* buffer = reinterpret_cast<int16_t*>(mEffectBufferEnabled 2668 ? mEffectBuffer : mSinkBuffer); 2669 bool ownsBuffer = false; 2670 2671 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 2672 if (session > 0) { 2673 // Only one effect chain can be present in direct output thread and it uses 2674 // the sink buffer as input 2675 if (mType != DIRECT) { 2676 size_t numSamples = mNormalFrameCount * mChannelCount; 2677 buffer = new int16_t[numSamples]; 2678 memset(buffer, 0, numSamples * sizeof(int16_t)); 2679 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 2680 ownsBuffer = true; 2681 } 2682 2683 // Attach all tracks with same session ID to this chain. 2684 for (size_t i = 0; i < mTracks.size(); ++i) { 2685 sp<Track> track = mTracks[i]; 2686 if (session == track->sessionId()) { 2687 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), 2688 buffer); 2689 track->setMainBuffer(buffer); 2690 chain->incTrackCnt(); 2691 } 2692 } 2693 2694 // indicate all active tracks in the chain 2695 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 2696 sp<Track> track = mActiveTracks[i].promote(); 2697 if (track == 0) { 2698 continue; 2699 } 2700 if (session == track->sessionId()) { 2701 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 2702 chain->incActiveTrackCnt(); 2703 } 2704 } 2705 } 2706 chain->setThread(this); 2707 chain->setInBuffer(buffer, ownsBuffer); 2708 chain->setOutBuffer(reinterpret_cast<int16_t*>(mEffectBufferEnabled 2709 ? mEffectBuffer : mSinkBuffer)); 2710 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 2711 // chains list in order to be processed last as it contains output stage effects 2712 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 2713 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 2714 // after track specific effects and before output stage 2715 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 2716 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 2717 // Effect chain for other sessions are inserted at beginning of effect 2718 // chains list to be processed before output mix effects. Relative order between other 2719 // sessions is not important 2720 size_t size = mEffectChains.size(); 2721 size_t i = 0; 2722 for (i = 0; i < size; i++) { 2723 if (mEffectChains[i]->sessionId() < session) { 2724 break; 2725 } 2726 } 2727 mEffectChains.insertAt(chain, i); 2728 checkSuspendOnAddEffectChain_l(chain); 2729 2730 return NO_ERROR; 2731} 2732 2733size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 2734{ 2735 int session = chain->sessionId(); 2736 2737 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 2738 2739 for (size_t i = 0; i < mEffectChains.size(); i++) { 2740 if (chain == mEffectChains[i]) { 2741 mEffectChains.removeAt(i); 2742 // detach all active tracks from the chain 2743 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 2744 sp<Track> track = mActiveTracks[i].promote(); 2745 if (track == 0) { 2746 continue; 2747 } 2748 if (session == track->sessionId()) { 2749 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 2750 chain.get(), session); 2751 chain->decActiveTrackCnt(); 2752 } 2753 } 2754 2755 // detach all tracks with same session ID from this chain 2756 for (size_t i = 0; i < mTracks.size(); ++i) { 2757 sp<Track> track = mTracks[i]; 2758 if (session == track->sessionId()) { 2759 track->setMainBuffer(reinterpret_cast<int16_t*>(mSinkBuffer)); 2760 chain->decTrackCnt(); 2761 } 2762 } 2763 break; 2764 } 2765 } 2766 return mEffectChains.size(); 2767} 2768 2769status_t AudioFlinger::PlaybackThread::attachAuxEffect( 2770 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2771{ 2772 Mutex::Autolock _l(mLock); 2773 return attachAuxEffect_l(track, EffectId); 2774} 2775 2776status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 2777 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2778{ 2779 status_t status = NO_ERROR; 2780 2781 if (EffectId == 0) { 2782 track->setAuxBuffer(0, NULL); 2783 } else { 2784 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 2785 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 2786 if (effect != 0) { 2787 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 2788 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 2789 } else { 2790 status = INVALID_OPERATION; 2791 } 2792 } else { 2793 status = BAD_VALUE; 2794 } 2795 } 2796 return status; 2797} 2798 2799void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 2800{ 2801 for (size_t i = 0; i < mTracks.size(); ++i) { 2802 sp<Track> track = mTracks[i]; 2803 if (track->auxEffectId() == effectId) { 2804 attachAuxEffect_l(track, 0); 2805 } 2806 } 2807} 2808 2809bool AudioFlinger::PlaybackThread::threadLoop() 2810{ 2811 Vector< sp<Track> > tracksToRemove; 2812 2813 mStandbyTimeNs = systemTime(); 2814 2815 // MIXER 2816 nsecs_t lastWarning = 0; 2817 2818 // DUPLICATING 2819 // FIXME could this be made local to while loop? 2820 writeFrames = 0; 2821 2822 int lastGeneration = 0; 2823 2824 cacheParameters_l(); 2825 mSleepTimeUs = mIdleSleepTimeUs; 2826 2827 if (mType == MIXER) { 2828 sleepTimeShift = 0; 2829 } 2830 2831 CpuStats cpuStats; 2832 const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid())); 2833 2834 acquireWakeLock(); 2835 2836 // mNBLogWriter->log can only be called while thread mutex mLock is held. 2837 // So if you need to log when mutex is unlocked, set logString to a non-NULL string, 2838 // and then that string will be logged at the next convenient opportunity. 2839 const char *logString = NULL; 2840 2841 checkSilentMode_l(); 2842 2843 while (!exitPending()) 2844 { 2845 cpuStats.sample(myName); 2846 2847 Vector< sp<EffectChain> > effectChains; 2848 2849 { // scope for mLock 2850 2851 Mutex::Autolock _l(mLock); 2852 2853 processConfigEvents_l(); 2854 2855 if (logString != NULL) { 2856 mNBLogWriter->logTimestamp(); 2857 mNBLogWriter->log(logString); 2858 logString = NULL; 2859 } 2860 2861 // Gather the framesReleased counters for all active tracks, 2862 // and associate with the sink frames written out. We need 2863 // this to convert the sink timestamp to the track timestamp. 2864 if (mNormalSink != 0) { 2865 bool updateTracks = true; 2866 bool cacheTimestamp = false; 2867 AudioTimestamp timeStamp; 2868 // FIXME: Use a 64 bit mNormalSink->framesWritten() counter. 2869 // At this time, we must always use cached timestamps even when 2870 // going through mPipeSink (which is non-blocking). The reason is that 2871 // the track may be removed from the active list for many hours and 2872 // the mNormalSink->framesWritten() will wrap making the linear 2873 // mapping fail. 2874 // 2875 // (Also mAudioTrackServerProxy->framesReleased() needs to be 2876 // updated to 64 bits for 64 bit frame position.) 2877 // 2878 if (true /* see comment above, should be: mNormalSink == mOutputSink */) { 2879 // If we use a hardware device, we must cache the sink timestamp now. 2880 // hardware devices can block timestamp access during data writes. 2881 if (mNormalSink->getTimestamp(timeStamp) == NO_ERROR) { 2882 cacheTimestamp = true; 2883 } else { 2884 updateTracks = false; 2885 } 2886 } 2887 if (updateTracks) { 2888 // sinkFramesWritten for non-offloaded tracks are contiguous 2889 // even after standby() is called. This is useful for the track frame 2890 // to sink frame mapping. 2891 const uint32_t sinkFramesWritten = mNormalSink->framesWritten(); 2892 const size_t size = mActiveTracks.size(); 2893 for (size_t i = 0; i < size; ++i) { 2894 sp<Track> t = mActiveTracks[i].promote(); 2895 if (t != 0 && !t->isFastTrack()) { 2896 t->updateTrackFrameInfo( 2897 t->mAudioTrackServerProxy->framesReleased(), 2898 sinkFramesWritten, 2899 cacheTimestamp ? &timeStamp : NULL); 2900 } 2901 } 2902 } 2903 } 2904 2905 saveOutputTracks(); 2906 if (mSignalPending) { 2907 // A signal was raised while we were unlocked 2908 mSignalPending = false; 2909 } else if (waitingAsyncCallback_l()) { 2910 if (exitPending()) { 2911 break; 2912 } 2913 bool released = false; 2914 // The following works around a bug in the offload driver. Ideally we would release 2915 // the wake lock every time, but that causes the last offload buffer(s) to be 2916 // dropped while the device is on battery, so we need to hold a wake lock during 2917 // the drain phase. 2918 if (mBytesRemaining && !(mDrainSequence & 1)) { 2919 releaseWakeLock_l(); 2920 released = true; 2921 } 2922 mWakeLockUids.clear(); 2923 mActiveTracksGeneration++; 2924 ALOGV("wait async completion"); 2925 mWaitWorkCV.wait(mLock); 2926 ALOGV("async completion/wake"); 2927 if (released) { 2928 acquireWakeLock_l(); 2929 } 2930 mStandbyTimeNs = systemTime() + mStandbyDelayNs; 2931 mSleepTimeUs = 0; 2932 2933 continue; 2934 } 2935 if ((!mActiveTracks.size() && systemTime() > mStandbyTimeNs) || 2936 isSuspended()) { 2937 // put audio hardware into standby after short delay 2938 if (shouldStandby_l()) { 2939 2940 threadLoop_standby(); 2941 2942 mStandby = true; 2943 } 2944 2945 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2946 // we're about to wait, flush the binder command buffer 2947 IPCThreadState::self()->flushCommands(); 2948 2949 clearOutputTracks(); 2950 2951 if (exitPending()) { 2952 break; 2953 } 2954 2955 releaseWakeLock_l(); 2956 mWakeLockUids.clear(); 2957 mActiveTracksGeneration++; 2958 // wait until we have something to do... 2959 ALOGV("%s going to sleep", myName.string()); 2960 mWaitWorkCV.wait(mLock); 2961 ALOGV("%s waking up", myName.string()); 2962 acquireWakeLock_l(); 2963 2964 mMixerStatus = MIXER_IDLE; 2965 mMixerStatusIgnoringFastTracks = MIXER_IDLE; 2966 mBytesWritten = 0; 2967 mBytesRemaining = 0; 2968 checkSilentMode_l(); 2969 2970 mStandbyTimeNs = systemTime() + mStandbyDelayNs; 2971 mSleepTimeUs = mIdleSleepTimeUs; 2972 if (mType == MIXER) { 2973 sleepTimeShift = 0; 2974 } 2975 2976 continue; 2977 } 2978 } 2979 // mMixerStatusIgnoringFastTracks is also updated internally 2980 mMixerStatus = prepareTracks_l(&tracksToRemove); 2981 2982 // compare with previously applied list 2983 if (lastGeneration != mActiveTracksGeneration) { 2984 // update wakelock 2985 updateWakeLockUids_l(mWakeLockUids); 2986 lastGeneration = mActiveTracksGeneration; 2987 } 2988 2989 // prevent any changes in effect chain list and in each effect chain 2990 // during mixing and effect process as the audio buffers could be deleted 2991 // or modified if an effect is created or deleted 2992 lockEffectChains_l(effectChains); 2993 } // mLock scope ends 2994 2995 if (mBytesRemaining == 0) { 2996 mCurrentWriteLength = 0; 2997 if (mMixerStatus == MIXER_TRACKS_READY) { 2998 // threadLoop_mix() sets mCurrentWriteLength 2999 threadLoop_mix(); 3000 } else if ((mMixerStatus != MIXER_DRAIN_TRACK) 3001 && (mMixerStatus != MIXER_DRAIN_ALL)) { 3002 // threadLoop_sleepTime sets mSleepTimeUs to 0 if data 3003 // must be written to HAL 3004 threadLoop_sleepTime(); 3005 if (mSleepTimeUs == 0) { 3006 mCurrentWriteLength = mSinkBufferSize; 3007 } 3008 } 3009 // Either threadLoop_mix() or threadLoop_sleepTime() should have set 3010 // mMixerBuffer with data if mMixerBufferValid is true and mSleepTimeUs == 0. 3011 // Merge mMixerBuffer data into mEffectBuffer (if any effects are valid) 3012 // or mSinkBuffer (if there are no effects). 3013 // 3014 // This is done pre-effects computation; if effects change to 3015 // support higher precision, this needs to move. 3016 // 3017 // mMixerBufferValid is only set true by MixerThread::prepareTracks_l(). 3018 // TODO use mSleepTimeUs == 0 as an additional condition. 3019 if (mMixerBufferValid) { 3020 void *buffer = mEffectBufferValid ? mEffectBuffer : mSinkBuffer; 3021 audio_format_t format = mEffectBufferValid ? mEffectBufferFormat : mFormat; 3022 3023 // mono blend occurs for mixer threads only (not direct or offloaded) 3024 // and is handled here if we're going directly to the sink. 3025 if (requireMonoBlend() && !mEffectBufferValid) { 3026 mono_blend(mMixerBuffer, mMixerBufferFormat, mChannelCount, mNormalFrameCount, 3027 true /*limit*/); 3028 } 3029 3030 memcpy_by_audio_format(buffer, format, mMixerBuffer, mMixerBufferFormat, 3031 mNormalFrameCount * mChannelCount); 3032 } 3033 3034 mBytesRemaining = mCurrentWriteLength; 3035 if (isSuspended()) { 3036 mSleepTimeUs = suspendSleepTimeUs(); 3037 // simulate write to HAL when suspended 3038 mBytesWritten += mSinkBufferSize; 3039 mBytesRemaining = 0; 3040 } 3041 3042 // only process effects if we're going to write 3043 if (mSleepTimeUs == 0 && mType != OFFLOAD) { 3044 for (size_t i = 0; i < effectChains.size(); i ++) { 3045 effectChains[i]->process_l(); 3046 } 3047 } 3048 } 3049 // Process effect chains for offloaded thread even if no audio 3050 // was read from audio track: process only updates effect state 3051 // and thus does have to be synchronized with audio writes but may have 3052 // to be called while waiting for async write callback 3053 if (mType == OFFLOAD) { 3054 for (size_t i = 0; i < effectChains.size(); i ++) { 3055 effectChains[i]->process_l(); 3056 } 3057 } 3058 3059 // Only if the Effects buffer is enabled and there is data in the 3060 // Effects buffer (buffer valid), we need to 3061 // copy into the sink buffer. 3062 // TODO use mSleepTimeUs == 0 as an additional condition. 3063 if (mEffectBufferValid) { 3064 //ALOGV("writing effect buffer to sink buffer format %#x", mFormat); 3065 3066 if (requireMonoBlend()) { 3067 mono_blend(mEffectBuffer, mEffectBufferFormat, mChannelCount, mNormalFrameCount, 3068 true /*limit*/); 3069 } 3070 3071 memcpy_by_audio_format(mSinkBuffer, mFormat, mEffectBuffer, mEffectBufferFormat, 3072 mNormalFrameCount * mChannelCount); 3073 } 3074 3075 // enable changes in effect chain 3076 unlockEffectChains(effectChains); 3077 3078 if (!waitingAsyncCallback()) { 3079 // mSleepTimeUs == 0 means we must write to audio hardware 3080 if (mSleepTimeUs == 0) { 3081 ssize_t ret = 0; 3082 if (mBytesRemaining) { 3083 ret = threadLoop_write(); 3084 if (ret < 0) { 3085 mBytesRemaining = 0; 3086 } else { 3087 mBytesWritten += ret; 3088 mBytesRemaining -= ret; 3089 } 3090 } else if ((mMixerStatus == MIXER_DRAIN_TRACK) || 3091 (mMixerStatus == MIXER_DRAIN_ALL)) { 3092 threadLoop_drain(); 3093 } 3094 if (mType == MIXER && !mStandby) { 3095 // write blocked detection 3096 nsecs_t now = systemTime(); 3097 nsecs_t delta = now - mLastWriteTime; 3098 if (delta > maxPeriod) { 3099 mNumDelayedWrites++; 3100 if ((now - lastWarning) > kWarningThrottleNs) { 3101 ATRACE_NAME("underrun"); 3102 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 3103 ns2ms(delta), mNumDelayedWrites, this); 3104 lastWarning = now; 3105 } 3106 } 3107 3108 if (mThreadThrottle 3109 && mMixerStatus == MIXER_TRACKS_READY // we are mixing (active tracks) 3110 && ret > 0) { // we wrote something 3111 // Limit MixerThread data processing to no more than twice the 3112 // expected processing rate. 3113 // 3114 // This helps prevent underruns with NuPlayer and other applications 3115 // which may set up buffers that are close to the minimum size, or use 3116 // deep buffers, and rely on a double-buffering sleep strategy to fill. 3117 // 3118 // The throttle smooths out sudden large data drains from the device, 3119 // e.g. when it comes out of standby, which often causes problems with 3120 // (1) mixer threads without a fast mixer (which has its own warm-up) 3121 // (2) minimum buffer sized tracks (even if the track is full, 3122 // the app won't fill fast enough to handle the sudden draw). 3123 3124 const int32_t deltaMs = delta / 1000000; 3125 const int32_t throttleMs = mHalfBufferMs - deltaMs; 3126 if ((signed)mHalfBufferMs >= throttleMs && throttleMs > 0) { 3127 usleep(throttleMs * 1000); 3128 // notify of throttle start on verbose log 3129 ALOGV_IF(mThreadThrottleEndMs == mThreadThrottleTimeMs, 3130 "mixer(%p) throttle begin:" 3131 " ret(%zd) deltaMs(%d) requires sleep %d ms", 3132 this, ret, deltaMs, throttleMs); 3133 mThreadThrottleTimeMs += throttleMs; 3134 } else { 3135 uint32_t diff = mThreadThrottleTimeMs - mThreadThrottleEndMs; 3136 if (diff > 0) { 3137 // notify of throttle end on debug log 3138 ALOGD("mixer(%p) throttle end: throttle time(%u)", this, diff); 3139 mThreadThrottleEndMs = mThreadThrottleTimeMs; 3140 } 3141 } 3142 } 3143 } 3144 3145 } else { 3146 ATRACE_BEGIN("sleep"); 3147 usleep(mSleepTimeUs); 3148 ATRACE_END(); 3149 } 3150 } 3151 3152 // Finally let go of removed track(s), without the lock held 3153 // since we can't guarantee the destructors won't acquire that 3154 // same lock. This will also mutate and push a new fast mixer state. 3155 threadLoop_removeTracks(tracksToRemove); 3156 tracksToRemove.clear(); 3157 3158 // FIXME I don't understand the need for this here; 3159 // it was in the original code but maybe the 3160 // assignment in saveOutputTracks() makes this unnecessary? 3161 clearOutputTracks(); 3162 3163 // Effect chains will be actually deleted here if they were removed from 3164 // mEffectChains list during mixing or effects processing 3165 effectChains.clear(); 3166 3167 // FIXME Note that the above .clear() is no longer necessary since effectChains 3168 // is now local to this block, but will keep it for now (at least until merge done). 3169 } 3170 3171 threadLoop_exit(); 3172 3173 if (!mStandby) { 3174 threadLoop_standby(); 3175 mStandby = true; 3176 } 3177 3178 releaseWakeLock(); 3179 mWakeLockUids.clear(); 3180 mActiveTracksGeneration++; 3181 3182 ALOGV("Thread %p type %d exiting", this, mType); 3183 return false; 3184} 3185 3186// removeTracks_l() must be called with ThreadBase::mLock held 3187void AudioFlinger::PlaybackThread::removeTracks_l(const Vector< sp<Track> >& tracksToRemove) 3188{ 3189 size_t count = tracksToRemove.size(); 3190 if (count > 0) { 3191 for (size_t i=0 ; i<count ; i++) { 3192 const sp<Track>& track = tracksToRemove.itemAt(i); 3193 mActiveTracks.remove(track); 3194 mWakeLockUids.remove(track->uid()); 3195 mActiveTracksGeneration++; 3196 ALOGV("removeTracks_l removing track on session %d", track->sessionId()); 3197 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 3198 if (chain != 0) { 3199 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), 3200 track->sessionId()); 3201 chain->decActiveTrackCnt(); 3202 } 3203 if (track->isTerminated()) { 3204 removeTrack_l(track); 3205 } 3206 } 3207 } 3208 3209} 3210 3211status_t AudioFlinger::PlaybackThread::getTimestamp_l(AudioTimestamp& timestamp) 3212{ 3213 if (mNormalSink != 0) { 3214 return mNormalSink->getTimestamp(timestamp); 3215 } 3216 if ((mType == OFFLOAD || mType == DIRECT) 3217 && mOutput != NULL && mOutput->stream->get_presentation_position) { 3218 uint64_t position64; 3219 int ret = mOutput->getPresentationPosition(&position64, ×tamp.mTime); 3220 if (ret == 0) { 3221 timestamp.mPosition = (uint32_t)position64; 3222 return NO_ERROR; 3223 } 3224 } 3225 return INVALID_OPERATION; 3226} 3227 3228status_t AudioFlinger::MixerThread::createAudioPatch_l(const struct audio_patch *patch, 3229 audio_patch_handle_t *handle) 3230{ 3231 // if !&IDLE, holds the FastMixer state to restore after new parameters processed 3232 FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE; 3233 if (mFastMixer != 0) { 3234 FastMixerStateQueue *sq = mFastMixer->sq(); 3235 FastMixerState *state = sq->begin(); 3236 if (!(state->mCommand & FastMixerState::IDLE)) { 3237 previousCommand = state->mCommand; 3238 state->mCommand = FastMixerState::HOT_IDLE; 3239 sq->end(); 3240 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3241 } else { 3242 sq->end(false /*didModify*/); 3243 } 3244 } 3245 status_t status = PlaybackThread::createAudioPatch_l(patch, handle); 3246 3247 if (!(previousCommand & FastMixerState::IDLE)) { 3248 ALOG_ASSERT(mFastMixer != 0); 3249 FastMixerStateQueue *sq = mFastMixer->sq(); 3250 FastMixerState *state = sq->begin(); 3251 ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE); 3252 state->mCommand = previousCommand; 3253 sq->end(); 3254 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3255 } 3256 3257 return status; 3258} 3259 3260status_t AudioFlinger::PlaybackThread::createAudioPatch_l(const struct audio_patch *patch, 3261 audio_patch_handle_t *handle) 3262{ 3263 status_t status = NO_ERROR; 3264 3265 // store new device and send to effects 3266 audio_devices_t type = AUDIO_DEVICE_NONE; 3267 for (unsigned int i = 0; i < patch->num_sinks; i++) { 3268 type |= patch->sinks[i].ext.device.type; 3269 } 3270 3271#ifdef ADD_BATTERY_DATA 3272 // when changing the audio output device, call addBatteryData to notify 3273 // the change 3274 if (mOutDevice != type) { 3275 uint32_t params = 0; 3276 // check whether speaker is on 3277 if (type & AUDIO_DEVICE_OUT_SPEAKER) { 3278 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 3279 } 3280 3281 audio_devices_t deviceWithoutSpeaker 3282 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 3283 // check if any other device (except speaker) is on 3284 if (type & deviceWithoutSpeaker) { 3285 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 3286 } 3287 3288 if (params != 0) { 3289 addBatteryData(params); 3290 } 3291 } 3292#endif 3293 3294 for (size_t i = 0; i < mEffectChains.size(); i++) { 3295 mEffectChains[i]->setDevice_l(type); 3296 } 3297 3298 // mPrevOutDevice is the latest device set by createAudioPatch_l(). It is not set when 3299 // the thread is created so that the first patch creation triggers an ioConfigChanged callback 3300 bool configChanged = mPrevOutDevice != type; 3301 mOutDevice = type; 3302 mPatch = *patch; 3303 3304 if (mOutput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 3305 audio_hw_device_t *hwDevice = mOutput->audioHwDev->hwDevice(); 3306 status = hwDevice->create_audio_patch(hwDevice, 3307 patch->num_sources, 3308 patch->sources, 3309 patch->num_sinks, 3310 patch->sinks, 3311 handle); 3312 } else { 3313 char *address; 3314 if (strcmp(patch->sinks[0].ext.device.address, "") != 0) { 3315 //FIXME: we only support address on first sink with HAL version < 3.0 3316 address = audio_device_address_to_parameter( 3317 patch->sinks[0].ext.device.type, 3318 patch->sinks[0].ext.device.address); 3319 } else { 3320 address = (char *)calloc(1, 1); 3321 } 3322 AudioParameter param = AudioParameter(String8(address)); 3323 free(address); 3324 param.addInt(String8(AUDIO_PARAMETER_STREAM_ROUTING), (int)type); 3325 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3326 param.toString().string()); 3327 *handle = AUDIO_PATCH_HANDLE_NONE; 3328 } 3329 if (configChanged) { 3330 mPrevOutDevice = type; 3331 sendIoConfigEvent_l(AUDIO_OUTPUT_CONFIG_CHANGED); 3332 } 3333 return status; 3334} 3335 3336status_t AudioFlinger::MixerThread::releaseAudioPatch_l(const audio_patch_handle_t handle) 3337{ 3338 // if !&IDLE, holds the FastMixer state to restore after new parameters processed 3339 FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE; 3340 if (mFastMixer != 0) { 3341 FastMixerStateQueue *sq = mFastMixer->sq(); 3342 FastMixerState *state = sq->begin(); 3343 if (!(state->mCommand & FastMixerState::IDLE)) { 3344 previousCommand = state->mCommand; 3345 state->mCommand = FastMixerState::HOT_IDLE; 3346 sq->end(); 3347 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3348 } else { 3349 sq->end(false /*didModify*/); 3350 } 3351 } 3352 3353 status_t status = PlaybackThread::releaseAudioPatch_l(handle); 3354 3355 if (!(previousCommand & FastMixerState::IDLE)) { 3356 ALOG_ASSERT(mFastMixer != 0); 3357 FastMixerStateQueue *sq = mFastMixer->sq(); 3358 FastMixerState *state = sq->begin(); 3359 ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE); 3360 state->mCommand = previousCommand; 3361 sq->end(); 3362 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3363 } 3364 3365 return status; 3366} 3367 3368status_t AudioFlinger::PlaybackThread::releaseAudioPatch_l(const audio_patch_handle_t handle) 3369{ 3370 status_t status = NO_ERROR; 3371 3372 mOutDevice = AUDIO_DEVICE_NONE; 3373 3374 if (mOutput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 3375 audio_hw_device_t *hwDevice = mOutput->audioHwDev->hwDevice(); 3376 status = hwDevice->release_audio_patch(hwDevice, handle); 3377 } else { 3378 AudioParameter param; 3379 param.addInt(String8(AUDIO_PARAMETER_STREAM_ROUTING), 0); 3380 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3381 param.toString().string()); 3382 } 3383 return status; 3384} 3385 3386void AudioFlinger::PlaybackThread::addPatchTrack(const sp<PatchTrack>& track) 3387{ 3388 Mutex::Autolock _l(mLock); 3389 mTracks.add(track); 3390} 3391 3392void AudioFlinger::PlaybackThread::deletePatchTrack(const sp<PatchTrack>& track) 3393{ 3394 Mutex::Autolock _l(mLock); 3395 destroyTrack_l(track); 3396} 3397 3398void AudioFlinger::PlaybackThread::getAudioPortConfig(struct audio_port_config *config) 3399{ 3400 ThreadBase::getAudioPortConfig(config); 3401 config->role = AUDIO_PORT_ROLE_SOURCE; 3402 config->ext.mix.hw_module = mOutput->audioHwDev->handle(); 3403 config->ext.mix.usecase.stream = AUDIO_STREAM_DEFAULT; 3404} 3405 3406// ---------------------------------------------------------------------------- 3407 3408AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 3409 audio_io_handle_t id, audio_devices_t device, bool systemReady, type_t type) 3410 : PlaybackThread(audioFlinger, output, id, device, type, systemReady), 3411 // mAudioMixer below 3412 // mFastMixer below 3413 mFastMixerFutex(0), 3414 mMasterMono(false) 3415 // mOutputSink below 3416 // mPipeSink below 3417 // mNormalSink below 3418{ 3419 ALOGV("MixerThread() id=%d device=%#x type=%d", id, device, type); 3420 ALOGV("mSampleRate=%u, mChannelMask=%#x, mChannelCount=%u, mFormat=%d, mFrameSize=%u, " 3421 "mFrameCount=%d, mNormalFrameCount=%d", 3422 mSampleRate, mChannelMask, mChannelCount, mFormat, mFrameSize, mFrameCount, 3423 mNormalFrameCount); 3424 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 3425 3426 if (type == DUPLICATING) { 3427 // The Duplicating thread uses the AudioMixer and delivers data to OutputTracks 3428 // (downstream MixerThreads) in DuplicatingThread::threadLoop_write(). 3429 // Do not create or use mFastMixer, mOutputSink, mPipeSink, or mNormalSink. 3430 return; 3431 } 3432 // create an NBAIO sink for the HAL output stream, and negotiate 3433 mOutputSink = new AudioStreamOutSink(output->stream); 3434 size_t numCounterOffers = 0; 3435 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount, mFormat)}; 3436 ssize_t index = mOutputSink->negotiate(offers, 1, NULL, numCounterOffers); 3437 ALOG_ASSERT(index == 0); 3438 3439 // initialize fast mixer depending on configuration 3440 bool initFastMixer; 3441 switch (kUseFastMixer) { 3442 case FastMixer_Never: 3443 initFastMixer = false; 3444 break; 3445 case FastMixer_Always: 3446 initFastMixer = true; 3447 break; 3448 case FastMixer_Static: 3449 case FastMixer_Dynamic: 3450 initFastMixer = mFrameCount < mNormalFrameCount; 3451 break; 3452 } 3453 if (initFastMixer) { 3454 audio_format_t fastMixerFormat; 3455 if (mMixerBufferEnabled && mEffectBufferEnabled) { 3456 fastMixerFormat = AUDIO_FORMAT_PCM_FLOAT; 3457 } else { 3458 fastMixerFormat = AUDIO_FORMAT_PCM_16_BIT; 3459 } 3460 if (mFormat != fastMixerFormat) { 3461 // change our Sink format to accept our intermediate precision 3462 mFormat = fastMixerFormat; 3463 free(mSinkBuffer); 3464 mFrameSize = mChannelCount * audio_bytes_per_sample(mFormat); 3465 const size_t sinkBufferSize = mNormalFrameCount * mFrameSize; 3466 (void)posix_memalign(&mSinkBuffer, 32, sinkBufferSize); 3467 } 3468 3469 // create a MonoPipe to connect our submix to FastMixer 3470 NBAIO_Format format = mOutputSink->format(); 3471 NBAIO_Format origformat = format; 3472 // adjust format to match that of the Fast Mixer 3473 ALOGV("format changed from %d to %d", format.mFormat, fastMixerFormat); 3474 format.mFormat = fastMixerFormat; 3475 format.mFrameSize = audio_bytes_per_sample(format.mFormat) * format.mChannelCount; 3476 3477 // This pipe depth compensates for scheduling latency of the normal mixer thread. 3478 // When it wakes up after a maximum latency, it runs a few cycles quickly before 3479 // finally blocking. Note the pipe implementation rounds up the request to a power of 2. 3480 MonoPipe *monoPipe = new MonoPipe(mNormalFrameCount * 4, format, true /*writeCanBlock*/); 3481 const NBAIO_Format offers[1] = {format}; 3482 size_t numCounterOffers = 0; 3483 ssize_t index = monoPipe->negotiate(offers, 1, NULL, numCounterOffers); 3484 ALOG_ASSERT(index == 0); 3485 monoPipe->setAvgFrames((mScreenState & 1) ? 3486 (monoPipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 3487 mPipeSink = monoPipe; 3488 3489#ifdef TEE_SINK 3490 if (mTeeSinkOutputEnabled) { 3491 // create a Pipe to archive a copy of FastMixer's output for dumpsys 3492 Pipe *teeSink = new Pipe(mTeeSinkOutputFrames, origformat); 3493 const NBAIO_Format offers2[1] = {origformat}; 3494 numCounterOffers = 0; 3495 index = teeSink->negotiate(offers2, 1, NULL, numCounterOffers); 3496 ALOG_ASSERT(index == 0); 3497 mTeeSink = teeSink; 3498 PipeReader *teeSource = new PipeReader(*teeSink); 3499 numCounterOffers = 0; 3500 index = teeSource->negotiate(offers2, 1, NULL, numCounterOffers); 3501 ALOG_ASSERT(index == 0); 3502 mTeeSource = teeSource; 3503 } 3504#endif 3505 3506 // create fast mixer and configure it initially with just one fast track for our submix 3507 mFastMixer = new FastMixer(); 3508 FastMixerStateQueue *sq = mFastMixer->sq(); 3509#ifdef STATE_QUEUE_DUMP 3510 sq->setObserverDump(&mStateQueueObserverDump); 3511 sq->setMutatorDump(&mStateQueueMutatorDump); 3512#endif 3513 FastMixerState *state = sq->begin(); 3514 FastTrack *fastTrack = &state->mFastTracks[0]; 3515 // wrap the source side of the MonoPipe to make it an AudioBufferProvider 3516 fastTrack->mBufferProvider = new SourceAudioBufferProvider(new MonoPipeReader(monoPipe)); 3517 fastTrack->mVolumeProvider = NULL; 3518 fastTrack->mChannelMask = mChannelMask; // mPipeSink channel mask for audio to FastMixer 3519 fastTrack->mFormat = mFormat; // mPipeSink format for audio to FastMixer 3520 fastTrack->mGeneration++; 3521 state->mFastTracksGen++; 3522 state->mTrackMask = 1; 3523 // fast mixer will use the HAL output sink 3524 state->mOutputSink = mOutputSink.get(); 3525 state->mOutputSinkGen++; 3526 state->mFrameCount = mFrameCount; 3527 state->mCommand = FastMixerState::COLD_IDLE; 3528 // already done in constructor initialization list 3529 //mFastMixerFutex = 0; 3530 state->mColdFutexAddr = &mFastMixerFutex; 3531 state->mColdGen++; 3532 state->mDumpState = &mFastMixerDumpState; 3533#ifdef TEE_SINK 3534 state->mTeeSink = mTeeSink.get(); 3535#endif 3536 mFastMixerNBLogWriter = audioFlinger->newWriter_l(kFastMixerLogSize, "FastMixer"); 3537 state->mNBLogWriter = mFastMixerNBLogWriter.get(); 3538 sq->end(); 3539 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3540 3541 // start the fast mixer 3542 mFastMixer->run("FastMixer", PRIORITY_URGENT_AUDIO); 3543 pid_t tid = mFastMixer->getTid(); 3544 sendPrioConfigEvent(getpid_cached, tid, kPriorityFastMixer); 3545 3546#ifdef AUDIO_WATCHDOG 3547 // create and start the watchdog 3548 mAudioWatchdog = new AudioWatchdog(); 3549 mAudioWatchdog->setDump(&mAudioWatchdogDump); 3550 mAudioWatchdog->run("AudioWatchdog", PRIORITY_URGENT_AUDIO); 3551 tid = mAudioWatchdog->getTid(); 3552 sendPrioConfigEvent(getpid_cached, tid, kPriorityFastMixer); 3553#endif 3554 3555 } 3556 3557 switch (kUseFastMixer) { 3558 case FastMixer_Never: 3559 case FastMixer_Dynamic: 3560 mNormalSink = mOutputSink; 3561 break; 3562 case FastMixer_Always: 3563 mNormalSink = mPipeSink; 3564 break; 3565 case FastMixer_Static: 3566 mNormalSink = initFastMixer ? mPipeSink : mOutputSink; 3567 break; 3568 } 3569} 3570 3571AudioFlinger::MixerThread::~MixerThread() 3572{ 3573 if (mFastMixer != 0) { 3574 FastMixerStateQueue *sq = mFastMixer->sq(); 3575 FastMixerState *state = sq->begin(); 3576 if (state->mCommand == FastMixerState::COLD_IDLE) { 3577 int32_t old = android_atomic_inc(&mFastMixerFutex); 3578 if (old == -1) { 3579 (void) syscall(__NR_futex, &mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 3580 } 3581 } 3582 state->mCommand = FastMixerState::EXIT; 3583 sq->end(); 3584 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3585 mFastMixer->join(); 3586 // Though the fast mixer thread has exited, it's state queue is still valid. 3587 // We'll use that extract the final state which contains one remaining fast track 3588 // corresponding to our sub-mix. 3589 state = sq->begin(); 3590 ALOG_ASSERT(state->mTrackMask == 1); 3591 FastTrack *fastTrack = &state->mFastTracks[0]; 3592 ALOG_ASSERT(fastTrack->mBufferProvider != NULL); 3593 delete fastTrack->mBufferProvider; 3594 sq->end(false /*didModify*/); 3595 mFastMixer.clear(); 3596#ifdef AUDIO_WATCHDOG 3597 if (mAudioWatchdog != 0) { 3598 mAudioWatchdog->requestExit(); 3599 mAudioWatchdog->requestExitAndWait(); 3600 mAudioWatchdog.clear(); 3601 } 3602#endif 3603 } 3604 mAudioFlinger->unregisterWriter(mFastMixerNBLogWriter); 3605 delete mAudioMixer; 3606} 3607 3608 3609uint32_t AudioFlinger::MixerThread::correctLatency_l(uint32_t latency) const 3610{ 3611 if (mFastMixer != 0) { 3612 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 3613 latency += (pipe->getAvgFrames() * 1000) / mSampleRate; 3614 } 3615 return latency; 3616} 3617 3618 3619void AudioFlinger::MixerThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) 3620{ 3621 PlaybackThread::threadLoop_removeTracks(tracksToRemove); 3622} 3623 3624ssize_t AudioFlinger::MixerThread::threadLoop_write() 3625{ 3626 // FIXME we should only do one push per cycle; confirm this is true 3627 // Start the fast mixer if it's not already running 3628 if (mFastMixer != 0) { 3629 FastMixerStateQueue *sq = mFastMixer->sq(); 3630 FastMixerState *state = sq->begin(); 3631 if (state->mCommand != FastMixerState::MIX_WRITE && 3632 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)) { 3633 if (state->mCommand == FastMixerState::COLD_IDLE) { 3634 3635 // FIXME workaround for first HAL write being CPU bound on some devices 3636 ATRACE_BEGIN("write"); 3637 mOutput->write((char *)mSinkBuffer, 0); 3638 ATRACE_END(); 3639 3640 int32_t old = android_atomic_inc(&mFastMixerFutex); 3641 if (old == -1) { 3642 (void) syscall(__NR_futex, &mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 3643 } 3644#ifdef AUDIO_WATCHDOG 3645 if (mAudioWatchdog != 0) { 3646 mAudioWatchdog->resume(); 3647 } 3648#endif 3649 } 3650 state->mCommand = FastMixerState::MIX_WRITE; 3651#ifdef FAST_THREAD_STATISTICS 3652 mFastMixerDumpState.increaseSamplingN(mAudioFlinger->isLowRamDevice() ? 3653 FastThreadDumpState::kSamplingNforLowRamDevice : FastThreadDumpState::kSamplingN); 3654#endif 3655 sq->end(); 3656 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3657 if (kUseFastMixer == FastMixer_Dynamic) { 3658 mNormalSink = mPipeSink; 3659 } 3660 } else { 3661 sq->end(false /*didModify*/); 3662 } 3663 } 3664 return PlaybackThread::threadLoop_write(); 3665} 3666 3667void AudioFlinger::MixerThread::threadLoop_standby() 3668{ 3669 // Idle the fast mixer if it's currently running 3670 if (mFastMixer != 0) { 3671 FastMixerStateQueue *sq = mFastMixer->sq(); 3672 FastMixerState *state = sq->begin(); 3673 if (!(state->mCommand & FastMixerState::IDLE)) { 3674 state->mCommand = FastMixerState::COLD_IDLE; 3675 state->mColdFutexAddr = &mFastMixerFutex; 3676 state->mColdGen++; 3677 mFastMixerFutex = 0; 3678 sq->end(); 3679 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 3680 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3681 if (kUseFastMixer == FastMixer_Dynamic) { 3682 mNormalSink = mOutputSink; 3683 } 3684#ifdef AUDIO_WATCHDOG 3685 if (mAudioWatchdog != 0) { 3686 mAudioWatchdog->pause(); 3687 } 3688#endif 3689 } else { 3690 sq->end(false /*didModify*/); 3691 } 3692 } 3693 PlaybackThread::threadLoop_standby(); 3694} 3695 3696bool AudioFlinger::PlaybackThread::waitingAsyncCallback_l() 3697{ 3698 return false; 3699} 3700 3701bool AudioFlinger::PlaybackThread::shouldStandby_l() 3702{ 3703 return !mStandby; 3704} 3705 3706bool AudioFlinger::PlaybackThread::waitingAsyncCallback() 3707{ 3708 Mutex::Autolock _l(mLock); 3709 return waitingAsyncCallback_l(); 3710} 3711 3712// shared by MIXER and DIRECT, overridden by DUPLICATING 3713void AudioFlinger::PlaybackThread::threadLoop_standby() 3714{ 3715 ALOGV("Audio hardware entering standby, mixer %p, suspend count %d", this, mSuspended); 3716 mOutput->standby(); 3717 if (mUseAsyncWrite != 0) { 3718 // discard any pending drain or write ack by incrementing sequence 3719 mWriteAckSequence = (mWriteAckSequence + 2) & ~1; 3720 mDrainSequence = (mDrainSequence + 2) & ~1; 3721 ALOG_ASSERT(mCallbackThread != 0); 3722 mCallbackThread->setWriteBlocked(mWriteAckSequence); 3723 mCallbackThread->setDraining(mDrainSequence); 3724 } 3725 mHwPaused = false; 3726} 3727 3728void AudioFlinger::PlaybackThread::onAddNewTrack_l() 3729{ 3730 ALOGV("signal playback thread"); 3731 broadcast_l(); 3732} 3733 3734void AudioFlinger::MixerThread::threadLoop_mix() 3735{ 3736 // mix buffers... 3737 mAudioMixer->process(); 3738 mCurrentWriteLength = mSinkBufferSize; 3739 // increase sleep time progressively when application underrun condition clears. 3740 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 3741 // that a steady state of alternating ready/not ready conditions keeps the sleep time 3742 // such that we would underrun the audio HAL. 3743 if ((mSleepTimeUs == 0) && (sleepTimeShift > 0)) { 3744 sleepTimeShift--; 3745 } 3746 mSleepTimeUs = 0; 3747 mStandbyTimeNs = systemTime() + mStandbyDelayNs; 3748 //TODO: delay standby when effects have a tail 3749 3750} 3751 3752void AudioFlinger::MixerThread::threadLoop_sleepTime() 3753{ 3754 // If no tracks are ready, sleep once for the duration of an output 3755 // buffer size, then write 0s to the output 3756 if (mSleepTimeUs == 0) { 3757 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3758 mSleepTimeUs = mActiveSleepTimeUs >> sleepTimeShift; 3759 if (mSleepTimeUs < kMinThreadSleepTimeUs) { 3760 mSleepTimeUs = kMinThreadSleepTimeUs; 3761 } 3762 // reduce sleep time in case of consecutive application underruns to avoid 3763 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 3764 // duration we would end up writing less data than needed by the audio HAL if 3765 // the condition persists. 3766 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 3767 sleepTimeShift++; 3768 } 3769 } else { 3770 mSleepTimeUs = mIdleSleepTimeUs; 3771 } 3772 } else if (mBytesWritten != 0 || (mMixerStatus == MIXER_TRACKS_ENABLED)) { 3773 // clear out mMixerBuffer or mSinkBuffer, to ensure buffers are cleared 3774 // before effects processing or output. 3775 if (mMixerBufferValid) { 3776 memset(mMixerBuffer, 0, mMixerBufferSize); 3777 } else { 3778 memset(mSinkBuffer, 0, mSinkBufferSize); 3779 } 3780 mSleepTimeUs = 0; 3781 ALOGV_IF(mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED), 3782 "anticipated start"); 3783 } 3784 // TODO add standby time extension fct of effect tail 3785} 3786 3787// prepareTracks_l() must be called with ThreadBase::mLock held 3788AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 3789 Vector< sp<Track> > *tracksToRemove) 3790{ 3791 3792 mixer_state mixerStatus = MIXER_IDLE; 3793 // find out which tracks need to be processed 3794 size_t count = mActiveTracks.size(); 3795 size_t mixedTracks = 0; 3796 size_t tracksWithEffect = 0; 3797 // counts only _active_ fast tracks 3798 size_t fastTracks = 0; 3799 uint32_t resetMask = 0; // bit mask of fast tracks that need to be reset 3800 3801 float masterVolume = mMasterVolume; 3802 bool masterMute = mMasterMute; 3803 3804 if (masterMute) { 3805 masterVolume = 0; 3806 } 3807 // Delegate master volume control to effect in output mix effect chain if needed 3808 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 3809 if (chain != 0) { 3810 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 3811 chain->setVolume_l(&v, &v); 3812 masterVolume = (float)((v + (1 << 23)) >> 24); 3813 chain.clear(); 3814 } 3815 3816 // prepare a new state to push 3817 FastMixerStateQueue *sq = NULL; 3818 FastMixerState *state = NULL; 3819 bool didModify = false; 3820 FastMixerStateQueue::block_t block = FastMixerStateQueue::BLOCK_UNTIL_PUSHED; 3821 if (mFastMixer != 0) { 3822 sq = mFastMixer->sq(); 3823 state = sq->begin(); 3824 } 3825 3826 mMixerBufferValid = false; // mMixerBuffer has no valid data until appropriate tracks found. 3827 mEffectBufferValid = false; // mEffectBuffer has no valid data until tracks found. 3828 3829 for (size_t i=0 ; i<count ; i++) { 3830 const sp<Track> t = mActiveTracks[i].promote(); 3831 if (t == 0) { 3832 continue; 3833 } 3834 3835 // this const just means the local variable doesn't change 3836 Track* const track = t.get(); 3837 3838 // process fast tracks 3839 if (track->isFastTrack()) { 3840 3841 // It's theoretically possible (though unlikely) for a fast track to be created 3842 // and then removed within the same normal mix cycle. This is not a problem, as 3843 // the track never becomes active so it's fast mixer slot is never touched. 3844 // The converse, of removing an (active) track and then creating a new track 3845 // at the identical fast mixer slot within the same normal mix cycle, 3846 // is impossible because the slot isn't marked available until the end of each cycle. 3847 int j = track->mFastIndex; 3848 ALOG_ASSERT(0 < j && j < (int)FastMixerState::kMaxFastTracks); 3849 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << j))); 3850 FastTrack *fastTrack = &state->mFastTracks[j]; 3851 3852 // Determine whether the track is currently in underrun condition, 3853 // and whether it had a recent underrun. 3854 FastTrackDump *ftDump = &mFastMixerDumpState.mTracks[j]; 3855 FastTrackUnderruns underruns = ftDump->mUnderruns; 3856 uint32_t recentFull = (underruns.mBitFields.mFull - 3857 track->mObservedUnderruns.mBitFields.mFull) & UNDERRUN_MASK; 3858 uint32_t recentPartial = (underruns.mBitFields.mPartial - 3859 track->mObservedUnderruns.mBitFields.mPartial) & UNDERRUN_MASK; 3860 uint32_t recentEmpty = (underruns.mBitFields.mEmpty - 3861 track->mObservedUnderruns.mBitFields.mEmpty) & UNDERRUN_MASK; 3862 uint32_t recentUnderruns = recentPartial + recentEmpty; 3863 track->mObservedUnderruns = underruns; 3864 // don't count underruns that occur while stopping or pausing 3865 // or stopped which can occur when flush() is called while active 3866 if (!(track->isStopping() || track->isPausing() || track->isStopped()) && 3867 recentUnderruns > 0) { 3868 // FIXME fast mixer will pull & mix partial buffers, but we count as a full underrun 3869 track->mAudioTrackServerProxy->tallyUnderrunFrames(recentUnderruns * mFrameCount); 3870 } else { 3871 track->mAudioTrackServerProxy->tallyUnderrunFrames(0); 3872 } 3873 3874 // This is similar to the state machine for normal tracks, 3875 // with a few modifications for fast tracks. 3876 bool isActive = true; 3877 switch (track->mState) { 3878 case TrackBase::STOPPING_1: 3879 // track stays active in STOPPING_1 state until first underrun 3880 if (recentUnderruns > 0 || track->isTerminated()) { 3881 track->mState = TrackBase::STOPPING_2; 3882 } 3883 break; 3884 case TrackBase::PAUSING: 3885 // ramp down is not yet implemented 3886 track->setPaused(); 3887 break; 3888 case TrackBase::RESUMING: 3889 // ramp up is not yet implemented 3890 track->mState = TrackBase::ACTIVE; 3891 break; 3892 case TrackBase::ACTIVE: 3893 if (recentFull > 0 || recentPartial > 0) { 3894 // track has provided at least some frames recently: reset retry count 3895 track->mRetryCount = kMaxTrackRetries; 3896 } 3897 if (recentUnderruns == 0) { 3898 // no recent underruns: stay active 3899 break; 3900 } 3901 // there has recently been an underrun of some kind 3902 if (track->sharedBuffer() == 0) { 3903 // were any of the recent underruns "empty" (no frames available)? 3904 if (recentEmpty == 0) { 3905 // no, then ignore the partial underruns as they are allowed indefinitely 3906 break; 3907 } 3908 // there has recently been an "empty" underrun: decrement the retry counter 3909 if (--(track->mRetryCount) > 0) { 3910 break; 3911 } 3912 // indicate to client process that the track was disabled because of underrun; 3913 // it will then automatically call start() when data is available 3914 android_atomic_or(CBLK_DISABLED, &track->mCblk->mFlags); 3915 // remove from active list, but state remains ACTIVE [confusing but true] 3916 isActive = false; 3917 break; 3918 } 3919 // fall through 3920 case TrackBase::STOPPING_2: 3921 case TrackBase::PAUSED: 3922 case TrackBase::STOPPED: 3923 case TrackBase::FLUSHED: // flush() while active 3924 // Check for presentation complete if track is inactive 3925 // We have consumed all the buffers of this track. 3926 // This would be incomplete if we auto-paused on underrun 3927 { 3928 size_t audioHALFrames = 3929 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 3930 size_t framesWritten = mBytesWritten / mFrameSize; 3931 if (!(mStandby || track->presentationComplete(framesWritten, audioHALFrames))) { 3932 // track stays in active list until presentation is complete 3933 break; 3934 } 3935 } 3936 if (track->isStopping_2()) { 3937 track->mState = TrackBase::STOPPED; 3938 } 3939 if (track->isStopped()) { 3940 // Can't reset directly, as fast mixer is still polling this track 3941 // track->reset(); 3942 // So instead mark this track as needing to be reset after push with ack 3943 resetMask |= 1 << i; 3944 } 3945 isActive = false; 3946 break; 3947 case TrackBase::IDLE: 3948 default: 3949 LOG_ALWAYS_FATAL("unexpected track state %d", track->mState); 3950 } 3951 3952 if (isActive) { 3953 // was it previously inactive? 3954 if (!(state->mTrackMask & (1 << j))) { 3955 ExtendedAudioBufferProvider *eabp = track; 3956 VolumeProvider *vp = track; 3957 fastTrack->mBufferProvider = eabp; 3958 fastTrack->mVolumeProvider = vp; 3959 fastTrack->mChannelMask = track->mChannelMask; 3960 fastTrack->mFormat = track->mFormat; 3961 fastTrack->mGeneration++; 3962 state->mTrackMask |= 1 << j; 3963 didModify = true; 3964 // no acknowledgement required for newly active tracks 3965 } 3966 // cache the combined master volume and stream type volume for fast mixer; this 3967 // lacks any synchronization or barrier so VolumeProvider may read a stale value 3968 track->mCachedVolume = masterVolume * mStreamTypes[track->streamType()].volume; 3969 ++fastTracks; 3970 } else { 3971 // was it previously active? 3972 if (state->mTrackMask & (1 << j)) { 3973 fastTrack->mBufferProvider = NULL; 3974 fastTrack->mGeneration++; 3975 state->mTrackMask &= ~(1 << j); 3976 didModify = true; 3977 // If any fast tracks were removed, we must wait for acknowledgement 3978 // because we're about to decrement the last sp<> on those tracks. 3979 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3980 } else { 3981 LOG_ALWAYS_FATAL("fast track %d should have been active; " 3982 "mState=%d, mTrackMask=%#x, recentUnderruns=%u, isShared=%d", 3983 j, track->mState, state->mTrackMask, recentUnderruns, 3984 track->sharedBuffer() != 0); 3985 } 3986 tracksToRemove->add(track); 3987 // Avoids a misleading display in dumpsys 3988 track->mObservedUnderruns.mBitFields.mMostRecent = UNDERRUN_FULL; 3989 } 3990 continue; 3991 } 3992 3993 { // local variable scope to avoid goto warning 3994 3995 audio_track_cblk_t* cblk = track->cblk(); 3996 3997 // The first time a track is added we wait 3998 // for all its buffers to be filled before processing it 3999 int name = track->name(); 4000 // make sure that we have enough frames to mix one full buffer. 4001 // enforce this condition only once to enable draining the buffer in case the client 4002 // app does not call stop() and relies on underrun to stop: 4003 // hence the test on (mMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 4004 // during last round 4005 size_t desiredFrames; 4006 const uint32_t sampleRate = track->mAudioTrackServerProxy->getSampleRate(); 4007 AudioPlaybackRate playbackRate = track->mAudioTrackServerProxy->getPlaybackRate(); 4008 4009 desiredFrames = sourceFramesNeededWithTimestretch( 4010 sampleRate, mNormalFrameCount, mSampleRate, playbackRate.mSpeed); 4011 // TODO: ONLY USED FOR LEGACY RESAMPLERS, remove when they are removed. 4012 // add frames already consumed but not yet released by the resampler 4013 // because mAudioTrackServerProxy->framesReady() will include these frames 4014 desiredFrames += mAudioMixer->getUnreleasedFrames(track->name()); 4015 4016 uint32_t minFrames = 1; 4017 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() && 4018 (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) { 4019 minFrames = desiredFrames; 4020 } 4021 4022 size_t framesReady = track->framesReady(); 4023 if (ATRACE_ENABLED()) { 4024 // I wish we had formatted trace names 4025 char traceName[16]; 4026 strcpy(traceName, "nRdy"); 4027 int name = track->name(); 4028 if (AudioMixer::TRACK0 <= name && 4029 name < (int) (AudioMixer::TRACK0 + AudioMixer::MAX_NUM_TRACKS)) { 4030 name -= AudioMixer::TRACK0; 4031 traceName[4] = (name / 10) + '0'; 4032 traceName[5] = (name % 10) + '0'; 4033 } else { 4034 traceName[4] = '?'; 4035 traceName[5] = '?'; 4036 } 4037 traceName[6] = '\0'; 4038 ATRACE_INT(traceName, framesReady); 4039 } 4040 if ((framesReady >= minFrames) && track->isReady() && 4041 !track->isPaused() && !track->isTerminated()) 4042 { 4043 ALOGVV("track %d s=%08x [OK] on thread %p", name, cblk->mServer, this); 4044 4045 mixedTracks++; 4046 4047 // track->mainBuffer() != mSinkBuffer or mMixerBuffer means 4048 // there is an effect chain connected to the track 4049 chain.clear(); 4050 if (track->mainBuffer() != mSinkBuffer && 4051 track->mainBuffer() != mMixerBuffer) { 4052 if (mEffectBufferEnabled) { 4053 mEffectBufferValid = true; // Later can set directly. 4054 } 4055 chain = getEffectChain_l(track->sessionId()); 4056 // Delegate volume control to effect in track effect chain if needed 4057 if (chain != 0) { 4058 tracksWithEffect++; 4059 } else { 4060 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on " 4061 "session %d", 4062 name, track->sessionId()); 4063 } 4064 } 4065 4066 4067 int param = AudioMixer::VOLUME; 4068 if (track->mFillingUpStatus == Track::FS_FILLED) { 4069 // no ramp for the first volume setting 4070 track->mFillingUpStatus = Track::FS_ACTIVE; 4071 if (track->mState == TrackBase::RESUMING) { 4072 track->mState = TrackBase::ACTIVE; 4073 param = AudioMixer::RAMP_VOLUME; 4074 } 4075 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 4076 // FIXME should not make a decision based on mServer 4077 } else if (cblk->mServer != 0) { 4078 // If the track is stopped before the first frame was mixed, 4079 // do not apply ramp 4080 param = AudioMixer::RAMP_VOLUME; 4081 } 4082 4083 // compute volume for this track 4084 uint32_t vl, vr; // in U8.24 integer format 4085 float vlf, vrf, vaf; // in [0.0, 1.0] float format 4086 if (track->isPausing() || mStreamTypes[track->streamType()].mute) { 4087 vl = vr = 0; 4088 vlf = vrf = vaf = 0.; 4089 if (track->isPausing()) { 4090 track->setPaused(); 4091 } 4092 } else { 4093 4094 // read original volumes with volume control 4095 float typeVolume = mStreamTypes[track->streamType()].volume; 4096 float v = masterVolume * typeVolume; 4097 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 4098 gain_minifloat_packed_t vlr = proxy->getVolumeLR(); 4099 vlf = float_from_gain(gain_minifloat_unpack_left(vlr)); 4100 vrf = float_from_gain(gain_minifloat_unpack_right(vlr)); 4101 // track volumes come from shared memory, so can't be trusted and must be clamped 4102 if (vlf > GAIN_FLOAT_UNITY) { 4103 ALOGV("Track left volume out of range: %.3g", vlf); 4104 vlf = GAIN_FLOAT_UNITY; 4105 } 4106 if (vrf > GAIN_FLOAT_UNITY) { 4107 ALOGV("Track right volume out of range: %.3g", vrf); 4108 vrf = GAIN_FLOAT_UNITY; 4109 } 4110 // now apply the master volume and stream type volume 4111 vlf *= v; 4112 vrf *= v; 4113 // assuming master volume and stream type volume each go up to 1.0, 4114 // then derive vl and vr as U8.24 versions for the effect chain 4115 const float scaleto8_24 = MAX_GAIN_INT * MAX_GAIN_INT; 4116 vl = (uint32_t) (scaleto8_24 * vlf); 4117 vr = (uint32_t) (scaleto8_24 * vrf); 4118 // vl and vr are now in U8.24 format 4119 uint16_t sendLevel = proxy->getSendLevel_U4_12(); 4120 // send level comes from shared memory and so may be corrupt 4121 if (sendLevel > MAX_GAIN_INT) { 4122 ALOGV("Track send level out of range: %04X", sendLevel); 4123 sendLevel = MAX_GAIN_INT; 4124 } 4125 // vaf is represented as [0.0, 1.0] float by rescaling sendLevel 4126 vaf = v * sendLevel * (1. / MAX_GAIN_INT); 4127 } 4128 4129 // Delegate volume control to effect in track effect chain if needed 4130 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 4131 // Do not ramp volume if volume is controlled by effect 4132 param = AudioMixer::VOLUME; 4133 // Update remaining floating point volume levels 4134 vlf = (float)vl / (1 << 24); 4135 vrf = (float)vr / (1 << 24); 4136 track->mHasVolumeController = true; 4137 } else { 4138 // force no volume ramp when volume controller was just disabled or removed 4139 // from effect chain to avoid volume spike 4140 if (track->mHasVolumeController) { 4141 param = AudioMixer::VOLUME; 4142 } 4143 track->mHasVolumeController = false; 4144 } 4145 4146 // XXX: these things DON'T need to be done each time 4147 mAudioMixer->setBufferProvider(name, track); 4148 mAudioMixer->enable(name); 4149 4150 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, &vlf); 4151 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, &vrf); 4152 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, &vaf); 4153 mAudioMixer->setParameter( 4154 name, 4155 AudioMixer::TRACK, 4156 AudioMixer::FORMAT, (void *)track->format()); 4157 mAudioMixer->setParameter( 4158 name, 4159 AudioMixer::TRACK, 4160 AudioMixer::CHANNEL_MASK, (void *)(uintptr_t)track->channelMask()); 4161 mAudioMixer->setParameter( 4162 name, 4163 AudioMixer::TRACK, 4164 AudioMixer::MIXER_CHANNEL_MASK, (void *)(uintptr_t)mChannelMask); 4165 // limit track sample rate to 2 x output sample rate, which changes at re-configuration 4166 uint32_t maxSampleRate = mSampleRate * AUDIO_RESAMPLER_DOWN_RATIO_MAX; 4167 uint32_t reqSampleRate = track->mAudioTrackServerProxy->getSampleRate(); 4168 if (reqSampleRate == 0) { 4169 reqSampleRate = mSampleRate; 4170 } else if (reqSampleRate > maxSampleRate) { 4171 reqSampleRate = maxSampleRate; 4172 } 4173 mAudioMixer->setParameter( 4174 name, 4175 AudioMixer::RESAMPLE, 4176 AudioMixer::SAMPLE_RATE, 4177 (void *)(uintptr_t)reqSampleRate); 4178 4179 AudioPlaybackRate playbackRate = track->mAudioTrackServerProxy->getPlaybackRate(); 4180 mAudioMixer->setParameter( 4181 name, 4182 AudioMixer::TIMESTRETCH, 4183 AudioMixer::PLAYBACK_RATE, 4184 &playbackRate); 4185 4186 /* 4187 * Select the appropriate output buffer for the track. 4188 * 4189 * Tracks with effects go into their own effects chain buffer 4190 * and from there into either mEffectBuffer or mSinkBuffer. 4191 * 4192 * Other tracks can use mMixerBuffer for higher precision 4193 * channel accumulation. If this buffer is enabled 4194 * (mMixerBufferEnabled true), then selected tracks will accumulate 4195 * into it. 4196 * 4197 */ 4198 if (mMixerBufferEnabled 4199 && (track->mainBuffer() == mSinkBuffer 4200 || track->mainBuffer() == mMixerBuffer)) { 4201 mAudioMixer->setParameter( 4202 name, 4203 AudioMixer::TRACK, 4204 AudioMixer::MIXER_FORMAT, (void *)mMixerBufferFormat); 4205 mAudioMixer->setParameter( 4206 name, 4207 AudioMixer::TRACK, 4208 AudioMixer::MAIN_BUFFER, (void *)mMixerBuffer); 4209 // TODO: override track->mainBuffer()? 4210 mMixerBufferValid = true; 4211 } else { 4212 mAudioMixer->setParameter( 4213 name, 4214 AudioMixer::TRACK, 4215 AudioMixer::MIXER_FORMAT, (void *)AUDIO_FORMAT_PCM_16_BIT); 4216 mAudioMixer->setParameter( 4217 name, 4218 AudioMixer::TRACK, 4219 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 4220 } 4221 mAudioMixer->setParameter( 4222 name, 4223 AudioMixer::TRACK, 4224 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 4225 4226 // reset retry count 4227 track->mRetryCount = kMaxTrackRetries; 4228 4229 // If one track is ready, set the mixer ready if: 4230 // - the mixer was not ready during previous round OR 4231 // - no other track is not ready 4232 if (mMixerStatusIgnoringFastTracks != MIXER_TRACKS_READY || 4233 mixerStatus != MIXER_TRACKS_ENABLED) { 4234 mixerStatus = MIXER_TRACKS_READY; 4235 } 4236 } else { 4237 if (framesReady < desiredFrames && !track->isStopped() && !track->isPaused()) { 4238 ALOGV("track(%p) underrun, framesReady(%zu) < framesDesired(%zd)", 4239 track, framesReady, desiredFrames); 4240 track->mAudioTrackServerProxy->tallyUnderrunFrames(desiredFrames); 4241 } else { 4242 track->mAudioTrackServerProxy->tallyUnderrunFrames(0); 4243 } 4244 4245 // clear effect chain input buffer if an active track underruns to avoid sending 4246 // previous audio buffer again to effects 4247 chain = getEffectChain_l(track->sessionId()); 4248 if (chain != 0) { 4249 chain->clearInputBuffer(); 4250 } 4251 4252 ALOGVV("track %d s=%08x [NOT READY] on thread %p", name, cblk->mServer, this); 4253 if ((track->sharedBuffer() != 0) || track->isTerminated() || 4254 track->isStopped() || track->isPaused()) { 4255 // We have consumed all the buffers of this track. 4256 // Remove it from the list of active tracks. 4257 // TODO: use actual buffer filling status instead of latency when available from 4258 // audio HAL 4259 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 4260 size_t framesWritten = mBytesWritten / mFrameSize; 4261 if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) { 4262 if (track->isStopped()) { 4263 track->reset(); 4264 } 4265 tracksToRemove->add(track); 4266 } 4267 } else { 4268 // No buffers for this track. Give it a few chances to 4269 // fill a buffer, then remove it from active list. 4270 if (--(track->mRetryCount) <= 0) { 4271 ALOGI("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 4272 tracksToRemove->add(track); 4273 // indicate to client process that the track was disabled because of underrun; 4274 // it will then automatically call start() when data is available 4275 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 4276 // If one track is not ready, mark the mixer also not ready if: 4277 // - the mixer was ready during previous round OR 4278 // - no other track is ready 4279 } else if (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY || 4280 mixerStatus != MIXER_TRACKS_READY) { 4281 mixerStatus = MIXER_TRACKS_ENABLED; 4282 } 4283 } 4284 mAudioMixer->disable(name); 4285 } 4286 4287 } // local variable scope to avoid goto warning 4288track_is_ready: ; 4289 4290 } 4291 4292 // Push the new FastMixer state if necessary 4293 bool pauseAudioWatchdog = false; 4294 if (didModify) { 4295 state->mFastTracksGen++; 4296 // if the fast mixer was active, but now there are no fast tracks, then put it in cold idle 4297 if (kUseFastMixer == FastMixer_Dynamic && 4298 state->mCommand == FastMixerState::MIX_WRITE && state->mTrackMask <= 1) { 4299 state->mCommand = FastMixerState::COLD_IDLE; 4300 state->mColdFutexAddr = &mFastMixerFutex; 4301 state->mColdGen++; 4302 mFastMixerFutex = 0; 4303 if (kUseFastMixer == FastMixer_Dynamic) { 4304 mNormalSink = mOutputSink; 4305 } 4306 // If we go into cold idle, need to wait for acknowledgement 4307 // so that fast mixer stops doing I/O. 4308 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 4309 pauseAudioWatchdog = true; 4310 } 4311 } 4312 if (sq != NULL) { 4313 sq->end(didModify); 4314 sq->push(block); 4315 } 4316#ifdef AUDIO_WATCHDOG 4317 if (pauseAudioWatchdog && mAudioWatchdog != 0) { 4318 mAudioWatchdog->pause(); 4319 } 4320#endif 4321 4322 // Now perform the deferred reset on fast tracks that have stopped 4323 while (resetMask != 0) { 4324 size_t i = __builtin_ctz(resetMask); 4325 ALOG_ASSERT(i < count); 4326 resetMask &= ~(1 << i); 4327 sp<Track> t = mActiveTracks[i].promote(); 4328 if (t == 0) { 4329 continue; 4330 } 4331 Track* track = t.get(); 4332 ALOG_ASSERT(track->isFastTrack() && track->isStopped()); 4333 track->reset(); 4334 } 4335 4336 // remove all the tracks that need to be... 4337 removeTracks_l(*tracksToRemove); 4338 4339 if (getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX) != 0) { 4340 mEffectBufferValid = true; 4341 } 4342 4343 if (mEffectBufferValid) { 4344 // as long as there are effects we should clear the effects buffer, to avoid 4345 // passing a non-clean buffer to the effect chain 4346 memset(mEffectBuffer, 0, mEffectBufferSize); 4347 } 4348 // sink or mix buffer must be cleared if all tracks are connected to an 4349 // effect chain as in this case the mixer will not write to the sink or mix buffer 4350 // and track effects will accumulate into it 4351 if ((mBytesRemaining == 0) && ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 4352 (mixedTracks == 0 && fastTracks > 0))) { 4353 // FIXME as a performance optimization, should remember previous zero status 4354 if (mMixerBufferValid) { 4355 memset(mMixerBuffer, 0, mMixerBufferSize); 4356 // TODO: In testing, mSinkBuffer below need not be cleared because 4357 // the PlaybackThread::threadLoop() copies mMixerBuffer into mSinkBuffer 4358 // after mixing. 4359 // 4360 // To enforce this guarantee: 4361 // ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 4362 // (mixedTracks == 0 && fastTracks > 0)) 4363 // must imply MIXER_TRACKS_READY. 4364 // Later, we may clear buffers regardless, and skip much of this logic. 4365 } 4366 // FIXME as a performance optimization, should remember previous zero status 4367 memset(mSinkBuffer, 0, mNormalFrameCount * mFrameSize); 4368 } 4369 4370 // if any fast tracks, then status is ready 4371 mMixerStatusIgnoringFastTracks = mixerStatus; 4372 if (fastTracks > 0) { 4373 mixerStatus = MIXER_TRACKS_READY; 4374 } 4375 return mixerStatus; 4376} 4377 4378// getTrackName_l() must be called with ThreadBase::mLock held 4379int AudioFlinger::MixerThread::getTrackName_l(audio_channel_mask_t channelMask, 4380 audio_format_t format, int sessionId) 4381{ 4382 return mAudioMixer->getTrackName(channelMask, format, sessionId); 4383} 4384 4385// deleteTrackName_l() must be called with ThreadBase::mLock held 4386void AudioFlinger::MixerThread::deleteTrackName_l(int name) 4387{ 4388 ALOGV("remove track (%d) and delete from mixer", name); 4389 mAudioMixer->deleteTrackName(name); 4390} 4391 4392// checkForNewParameter_l() must be called with ThreadBase::mLock held 4393bool AudioFlinger::MixerThread::checkForNewParameter_l(const String8& keyValuePair, 4394 status_t& status) 4395{ 4396 bool reconfig = false; 4397 bool a2dpDeviceChanged = false; 4398 4399 status = NO_ERROR; 4400 4401 // if !&IDLE, holds the FastMixer state to restore after new parameters processed 4402 FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE; 4403 if (mFastMixer != 0) { 4404 FastMixerStateQueue *sq = mFastMixer->sq(); 4405 FastMixerState *state = sq->begin(); 4406 if (!(state->mCommand & FastMixerState::IDLE)) { 4407 previousCommand = state->mCommand; 4408 state->mCommand = FastMixerState::HOT_IDLE; 4409 sq->end(); 4410 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 4411 } else { 4412 sq->end(false /*didModify*/); 4413 } 4414 } 4415 4416 AudioParameter param = AudioParameter(keyValuePair); 4417 int value; 4418 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 4419 reconfig = true; 4420 } 4421 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 4422 if (!isValidPcmSinkFormat((audio_format_t) value)) { 4423 status = BAD_VALUE; 4424 } else { 4425 // no need to save value, since it's constant 4426 reconfig = true; 4427 } 4428 } 4429 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 4430 if (!isValidPcmSinkChannelMask((audio_channel_mask_t) value)) { 4431 status = BAD_VALUE; 4432 } else { 4433 // no need to save value, since it's constant 4434 reconfig = true; 4435 } 4436 } 4437 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 4438 // do not accept frame count changes if tracks are open as the track buffer 4439 // size depends on frame count and correct behavior would not be guaranteed 4440 // if frame count is changed after track creation 4441 if (!mTracks.isEmpty()) { 4442 status = INVALID_OPERATION; 4443 } else { 4444 reconfig = true; 4445 } 4446 } 4447 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 4448#ifdef ADD_BATTERY_DATA 4449 // when changing the audio output device, call addBatteryData to notify 4450 // the change 4451 if (mOutDevice != value) { 4452 uint32_t params = 0; 4453 // check whether speaker is on 4454 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 4455 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 4456 } 4457 4458 audio_devices_t deviceWithoutSpeaker 4459 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 4460 // check if any other device (except speaker) is on 4461 if (value & deviceWithoutSpeaker) { 4462 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 4463 } 4464 4465 if (params != 0) { 4466 addBatteryData(params); 4467 } 4468 } 4469#endif 4470 4471 // forward device change to effects that have requested to be 4472 // aware of attached audio device. 4473 if (value != AUDIO_DEVICE_NONE) { 4474 a2dpDeviceChanged = 4475 (mOutDevice & AUDIO_DEVICE_OUT_ALL_A2DP) != (value & AUDIO_DEVICE_OUT_ALL_A2DP); 4476 mOutDevice = value; 4477 for (size_t i = 0; i < mEffectChains.size(); i++) { 4478 mEffectChains[i]->setDevice_l(mOutDevice); 4479 } 4480 } 4481 } 4482 4483 if (status == NO_ERROR) { 4484 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 4485 keyValuePair.string()); 4486 if (!mStandby && status == INVALID_OPERATION) { 4487 mOutput->standby(); 4488 mStandby = true; 4489 mBytesWritten = 0; 4490 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 4491 keyValuePair.string()); 4492 } 4493 if (status == NO_ERROR && reconfig) { 4494 readOutputParameters_l(); 4495 delete mAudioMixer; 4496 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 4497 for (size_t i = 0; i < mTracks.size() ; i++) { 4498 int name = getTrackName_l(mTracks[i]->mChannelMask, 4499 mTracks[i]->mFormat, mTracks[i]->mSessionId); 4500 if (name < 0) { 4501 break; 4502 } 4503 mTracks[i]->mName = name; 4504 } 4505 sendIoConfigEvent_l(AUDIO_OUTPUT_CONFIG_CHANGED); 4506 } 4507 } 4508 4509 if (!(previousCommand & FastMixerState::IDLE)) { 4510 ALOG_ASSERT(mFastMixer != 0); 4511 FastMixerStateQueue *sq = mFastMixer->sq(); 4512 FastMixerState *state = sq->begin(); 4513 ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE); 4514 state->mCommand = previousCommand; 4515 sq->end(); 4516 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 4517 } 4518 4519 return reconfig || a2dpDeviceChanged; 4520} 4521 4522 4523void AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 4524{ 4525 const size_t SIZE = 256; 4526 char buffer[SIZE]; 4527 String8 result; 4528 4529 PlaybackThread::dumpInternals(fd, args); 4530 dprintf(fd, " Thread throttle time (msecs): %u\n", mThreadThrottleTimeMs); 4531 dprintf(fd, " AudioMixer tracks: 0x%08x\n", mAudioMixer->trackNames()); 4532 dprintf(fd, " Master mono: %s\n", mMasterMono ? "on" : "off"); 4533 4534 // Make a non-atomic copy of fast mixer dump state so it won't change underneath us 4535 // while we are dumping it. It may be inconsistent, but it won't mutate! 4536 // This is a large object so we place it on the heap. 4537 // FIXME 25972958: Need an intelligent copy constructor that does not touch unused pages. 4538 const FastMixerDumpState *copy = new FastMixerDumpState(mFastMixerDumpState); 4539 copy->dump(fd); 4540 delete copy; 4541 4542#ifdef STATE_QUEUE_DUMP 4543 // Similar for state queue 4544 StateQueueObserverDump observerCopy = mStateQueueObserverDump; 4545 observerCopy.dump(fd); 4546 StateQueueMutatorDump mutatorCopy = mStateQueueMutatorDump; 4547 mutatorCopy.dump(fd); 4548#endif 4549 4550#ifdef TEE_SINK 4551 // Write the tee output to a .wav file 4552 dumpTee(fd, mTeeSource, mId); 4553#endif 4554 4555#ifdef AUDIO_WATCHDOG 4556 if (mAudioWatchdog != 0) { 4557 // Make a non-atomic copy of audio watchdog dump so it won't change underneath us 4558 AudioWatchdogDump wdCopy = mAudioWatchdogDump; 4559 wdCopy.dump(fd); 4560 } 4561#endif 4562} 4563 4564uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const 4565{ 4566 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000) / 2; 4567} 4568 4569uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() const 4570{ 4571 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000); 4572} 4573 4574void AudioFlinger::MixerThread::cacheParameters_l() 4575{ 4576 PlaybackThread::cacheParameters_l(); 4577 4578 // FIXME: Relaxed timing because of a certain device that can't meet latency 4579 // Should be reduced to 2x after the vendor fixes the driver issue 4580 // increase threshold again due to low power audio mode. The way this warning 4581 // threshold is calculated and its usefulness should be reconsidered anyway. 4582 maxPeriod = seconds(mNormalFrameCount) / mSampleRate * 15; 4583} 4584 4585// ---------------------------------------------------------------------------- 4586 4587AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 4588 AudioStreamOut* output, audio_io_handle_t id, audio_devices_t device, bool systemReady) 4589 : PlaybackThread(audioFlinger, output, id, device, DIRECT, systemReady) 4590 // mLeftVolFloat, mRightVolFloat 4591{ 4592} 4593 4594AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 4595 AudioStreamOut* output, audio_io_handle_t id, uint32_t device, 4596 ThreadBase::type_t type, bool systemReady) 4597 : PlaybackThread(audioFlinger, output, id, device, type, systemReady) 4598 // mLeftVolFloat, mRightVolFloat 4599{ 4600} 4601 4602AudioFlinger::DirectOutputThread::~DirectOutputThread() 4603{ 4604} 4605 4606void AudioFlinger::DirectOutputThread::processVolume_l(Track *track, bool lastTrack) 4607{ 4608 audio_track_cblk_t* cblk = track->cblk(); 4609 float left, right; 4610 4611 if (mMasterMute || mStreamTypes[track->streamType()].mute) { 4612 left = right = 0; 4613 } else { 4614 float typeVolume = mStreamTypes[track->streamType()].volume; 4615 float v = mMasterVolume * typeVolume; 4616 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 4617 gain_minifloat_packed_t vlr = proxy->getVolumeLR(); 4618 left = float_from_gain(gain_minifloat_unpack_left(vlr)); 4619 if (left > GAIN_FLOAT_UNITY) { 4620 left = GAIN_FLOAT_UNITY; 4621 } 4622 left *= v; 4623 right = float_from_gain(gain_minifloat_unpack_right(vlr)); 4624 if (right > GAIN_FLOAT_UNITY) { 4625 right = GAIN_FLOAT_UNITY; 4626 } 4627 right *= v; 4628 } 4629 4630 if (lastTrack) { 4631 if (left != mLeftVolFloat || right != mRightVolFloat) { 4632 mLeftVolFloat = left; 4633 mRightVolFloat = right; 4634 4635 // Convert volumes from float to 8.24 4636 uint32_t vl = (uint32_t)(left * (1 << 24)); 4637 uint32_t vr = (uint32_t)(right * (1 << 24)); 4638 4639 // Delegate volume control to effect in track effect chain if needed 4640 // only one effect chain can be present on DirectOutputThread, so if 4641 // there is one, the track is connected to it 4642 if (!mEffectChains.isEmpty()) { 4643 mEffectChains[0]->setVolume_l(&vl, &vr); 4644 left = (float)vl / (1 << 24); 4645 right = (float)vr / (1 << 24); 4646 } 4647 if (mOutput->stream->set_volume) { 4648 mOutput->stream->set_volume(mOutput->stream, left, right); 4649 } 4650 } 4651 } 4652} 4653 4654void AudioFlinger::DirectOutputThread::onAddNewTrack_l() 4655{ 4656 sp<Track> previousTrack = mPreviousTrack.promote(); 4657 sp<Track> latestTrack = mLatestActiveTrack.promote(); 4658 4659 if (previousTrack != 0 && latestTrack != 0) { 4660 if (mType == DIRECT) { 4661 if (previousTrack.get() != latestTrack.get()) { 4662 mFlushPending = true; 4663 } 4664 } else /* mType == OFFLOAD */ { 4665 if (previousTrack->sessionId() != latestTrack->sessionId()) { 4666 mFlushPending = true; 4667 } 4668 } 4669 } 4670 PlaybackThread::onAddNewTrack_l(); 4671} 4672 4673AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l( 4674 Vector< sp<Track> > *tracksToRemove 4675) 4676{ 4677 size_t count = mActiveTracks.size(); 4678 mixer_state mixerStatus = MIXER_IDLE; 4679 bool doHwPause = false; 4680 bool doHwResume = false; 4681 4682 // find out which tracks need to be processed 4683 for (size_t i = 0; i < count; i++) { 4684 sp<Track> t = mActiveTracks[i].promote(); 4685 // The track died recently 4686 if (t == 0) { 4687 continue; 4688 } 4689 4690 if (t->isInvalid()) { 4691 ALOGW("An invalidated track shouldn't be in active list"); 4692 tracksToRemove->add(t); 4693 continue; 4694 } 4695 4696 Track* const track = t.get(); 4697 audio_track_cblk_t* cblk = track->cblk(); 4698 // Only consider last track started for volume and mixer state control. 4699 // In theory an older track could underrun and restart after the new one starts 4700 // but as we only care about the transition phase between two tracks on a 4701 // direct output, it is not a problem to ignore the underrun case. 4702 sp<Track> l = mLatestActiveTrack.promote(); 4703 bool last = l.get() == track; 4704 4705 if (track->isPausing()) { 4706 track->setPaused(); 4707 if (mHwSupportsPause && last && !mHwPaused) { 4708 doHwPause = true; 4709 mHwPaused = true; 4710 } 4711 tracksToRemove->add(track); 4712 } else if (track->isFlushPending()) { 4713 track->flushAck(); 4714 if (last) { 4715 mFlushPending = true; 4716 } 4717 } else if (track->isResumePending()) { 4718 track->resumeAck(); 4719 if (last && mHwPaused) { 4720 doHwResume = true; 4721 mHwPaused = false; 4722 } 4723 } 4724 4725 // The first time a track is added we wait 4726 // for all its buffers to be filled before processing it. 4727 // Allow draining the buffer in case the client 4728 // app does not call stop() and relies on underrun to stop: 4729 // hence the test on (track->mRetryCount > 1). 4730 // If retryCount<=1 then track is about to underrun and be removed. 4731 // Do not use a high threshold for compressed audio. 4732 uint32_t minFrames; 4733 if ((track->sharedBuffer() == 0) && !track->isStopping_1() && !track->isPausing() 4734 && (track->mRetryCount > 1) && audio_has_proportional_frames(mFormat)) { 4735 minFrames = mNormalFrameCount; 4736 } else { 4737 minFrames = 1; 4738 } 4739 4740 if ((track->framesReady() >= minFrames) && track->isReady() && !track->isPaused() && 4741 !track->isStopping_2() && !track->isStopped()) 4742 { 4743 ALOGVV("track %d s=%08x [OK]", track->name(), cblk->mServer); 4744 4745 if (track->mFillingUpStatus == Track::FS_FILLED) { 4746 track->mFillingUpStatus = Track::FS_ACTIVE; 4747 // make sure processVolume_l() will apply new volume even if 0 4748 mLeftVolFloat = mRightVolFloat = -1.0; 4749 if (!mHwSupportsPause) { 4750 track->resumeAck(); 4751 } 4752 } 4753 4754 // compute volume for this track 4755 processVolume_l(track, last); 4756 if (last) { 4757 sp<Track> previousTrack = mPreviousTrack.promote(); 4758 if (previousTrack != 0) { 4759 if (track != previousTrack.get()) { 4760 // Flush any data still being written from last track 4761 mBytesRemaining = 0; 4762 // Invalidate previous track to force a seek when resuming. 4763 previousTrack->invalidate(); 4764 } 4765 } 4766 mPreviousTrack = track; 4767 4768 // reset retry count 4769 track->mRetryCount = kMaxTrackRetriesDirect; 4770 mActiveTrack = t; 4771 mixerStatus = MIXER_TRACKS_READY; 4772 if (mHwPaused) { 4773 doHwResume = true; 4774 mHwPaused = false; 4775 } 4776 } 4777 } else { 4778 // clear effect chain input buffer if the last active track started underruns 4779 // to avoid sending previous audio buffer again to effects 4780 if (!mEffectChains.isEmpty() && last) { 4781 mEffectChains[0]->clearInputBuffer(); 4782 } 4783 if (track->isStopping_1()) { 4784 track->mState = TrackBase::STOPPING_2; 4785 if (last && mHwPaused) { 4786 doHwResume = true; 4787 mHwPaused = false; 4788 } 4789 } 4790 if ((track->sharedBuffer() != 0) || track->isStopped() || 4791 track->isStopping_2() || track->isPaused()) { 4792 // We have consumed all the buffers of this track. 4793 // Remove it from the list of active tracks. 4794 size_t audioHALFrames; 4795 if (audio_has_proportional_frames(mFormat)) { 4796 audioHALFrames = (latency_l() * mSampleRate) / 1000; 4797 } else { 4798 audioHALFrames = 0; 4799 } 4800 4801 size_t framesWritten = mBytesWritten / mFrameSize; 4802 if (mStandby || !last || 4803 track->presentationComplete(framesWritten, audioHALFrames)) { 4804 if (track->isStopping_2()) { 4805 track->mState = TrackBase::STOPPED; 4806 } 4807 if (track->isStopped()) { 4808 track->reset(); 4809 } 4810 tracksToRemove->add(track); 4811 } 4812 } else { 4813 // No buffers for this track. Give it a few chances to 4814 // fill a buffer, then remove it from active list. 4815 // Only consider last track started for mixer state control 4816 if (--(track->mRetryCount) <= 0) { 4817 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 4818 tracksToRemove->add(track); 4819 // indicate to client process that the track was disabled because of underrun; 4820 // it will then automatically call start() when data is available 4821 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 4822 } else if (last) { 4823 ALOGW("pause because of UNDERRUN, framesReady = %zu," 4824 "minFrames = %u, mFormat = %#x", 4825 track->framesReady(), minFrames, mFormat); 4826 mixerStatus = MIXER_TRACKS_ENABLED; 4827 if (mHwSupportsPause && !mHwPaused && !mStandby) { 4828 doHwPause = true; 4829 mHwPaused = true; 4830 } 4831 } 4832 } 4833 } 4834 } 4835 4836 // if an active track did not command a flush, check for pending flush on stopped tracks 4837 if (!mFlushPending) { 4838 for (size_t i = 0; i < mTracks.size(); i++) { 4839 if (mTracks[i]->isFlushPending()) { 4840 mTracks[i]->flushAck(); 4841 mFlushPending = true; 4842 } 4843 } 4844 } 4845 4846 // make sure the pause/flush/resume sequence is executed in the right order. 4847 // If a flush is pending and a track is active but the HW is not paused, force a HW pause 4848 // before flush and then resume HW. This can happen in case of pause/flush/resume 4849 // if resume is received before pause is executed. 4850 if (mHwSupportsPause && !mStandby && 4851 (doHwPause || (mFlushPending && !mHwPaused && (count != 0)))) { 4852 mOutput->stream->pause(mOutput->stream); 4853 } 4854 if (mFlushPending) { 4855 flushHw_l(); 4856 } 4857 if (mHwSupportsPause && !mStandby && doHwResume) { 4858 mOutput->stream->resume(mOutput->stream); 4859 } 4860 // remove all the tracks that need to be... 4861 removeTracks_l(*tracksToRemove); 4862 4863 return mixerStatus; 4864} 4865 4866void AudioFlinger::DirectOutputThread::threadLoop_mix() 4867{ 4868 size_t frameCount = mFrameCount; 4869 int8_t *curBuf = (int8_t *)mSinkBuffer; 4870 // output audio to hardware 4871 while (frameCount) { 4872 AudioBufferProvider::Buffer buffer; 4873 buffer.frameCount = frameCount; 4874 status_t status = mActiveTrack->getNextBuffer(&buffer); 4875 if (status != NO_ERROR || buffer.raw == NULL) { 4876 memset(curBuf, 0, frameCount * mFrameSize); 4877 break; 4878 } 4879 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 4880 frameCount -= buffer.frameCount; 4881 curBuf += buffer.frameCount * mFrameSize; 4882 mActiveTrack->releaseBuffer(&buffer); 4883 } 4884 mCurrentWriteLength = curBuf - (int8_t *)mSinkBuffer; 4885 mSleepTimeUs = 0; 4886 mStandbyTimeNs = systemTime() + mStandbyDelayNs; 4887 mActiveTrack.clear(); 4888} 4889 4890void AudioFlinger::DirectOutputThread::threadLoop_sleepTime() 4891{ 4892 // do not write to HAL when paused 4893 if (mHwPaused || (usesHwAvSync() && mStandby)) { 4894 mSleepTimeUs = mIdleSleepTimeUs; 4895 return; 4896 } 4897 if (mSleepTimeUs == 0) { 4898 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4899 mSleepTimeUs = mActiveSleepTimeUs; 4900 } else { 4901 mSleepTimeUs = mIdleSleepTimeUs; 4902 } 4903 } else if (mBytesWritten != 0 && audio_has_proportional_frames(mFormat)) { 4904 memset(mSinkBuffer, 0, mFrameCount * mFrameSize); 4905 mSleepTimeUs = 0; 4906 } 4907} 4908 4909void AudioFlinger::DirectOutputThread::threadLoop_exit() 4910{ 4911 { 4912 Mutex::Autolock _l(mLock); 4913 for (size_t i = 0; i < mTracks.size(); i++) { 4914 if (mTracks[i]->isFlushPending()) { 4915 mTracks[i]->flushAck(); 4916 mFlushPending = true; 4917 } 4918 } 4919 if (mFlushPending) { 4920 flushHw_l(); 4921 } 4922 } 4923 PlaybackThread::threadLoop_exit(); 4924} 4925 4926// must be called with thread mutex locked 4927bool AudioFlinger::DirectOutputThread::shouldStandby_l() 4928{ 4929 bool trackPaused = false; 4930 bool trackStopped = false; 4931 4932 // do not put the HAL in standby when paused. AwesomePlayer clear the offloaded AudioTrack 4933 // after a timeout and we will enter standby then. 4934 if (mTracks.size() > 0) { 4935 trackPaused = mTracks[mTracks.size() - 1]->isPaused(); 4936 trackStopped = mTracks[mTracks.size() - 1]->isStopped() || 4937 mTracks[mTracks.size() - 1]->mState == TrackBase::IDLE; 4938 } 4939 4940 return !mStandby && !(trackPaused || (mHwPaused && !trackStopped)); 4941} 4942 4943// getTrackName_l() must be called with ThreadBase::mLock held 4944int AudioFlinger::DirectOutputThread::getTrackName_l(audio_channel_mask_t channelMask __unused, 4945 audio_format_t format __unused, int sessionId __unused) 4946{ 4947 return 0; 4948} 4949 4950// deleteTrackName_l() must be called with ThreadBase::mLock held 4951void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name __unused) 4952{ 4953} 4954 4955// checkForNewParameter_l() must be called with ThreadBase::mLock held 4956bool AudioFlinger::DirectOutputThread::checkForNewParameter_l(const String8& keyValuePair, 4957 status_t& status) 4958{ 4959 bool reconfig = false; 4960 bool a2dpDeviceChanged = false; 4961 4962 status = NO_ERROR; 4963 4964 AudioParameter param = AudioParameter(keyValuePair); 4965 int value; 4966 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 4967 // forward device change to effects that have requested to be 4968 // aware of attached audio device. 4969 if (value != AUDIO_DEVICE_NONE) { 4970 a2dpDeviceChanged = 4971 (mOutDevice & AUDIO_DEVICE_OUT_ALL_A2DP) != (value & AUDIO_DEVICE_OUT_ALL_A2DP); 4972 mOutDevice = value; 4973 for (size_t i = 0; i < mEffectChains.size(); i++) { 4974 mEffectChains[i]->setDevice_l(mOutDevice); 4975 } 4976 } 4977 } 4978 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 4979 // do not accept frame count changes if tracks are open as the track buffer 4980 // size depends on frame count and correct behavior would not be garantied 4981 // if frame count is changed after track creation 4982 if (!mTracks.isEmpty()) { 4983 status = INVALID_OPERATION; 4984 } else { 4985 reconfig = true; 4986 } 4987 } 4988 if (status == NO_ERROR) { 4989 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 4990 keyValuePair.string()); 4991 if (!mStandby && status == INVALID_OPERATION) { 4992 mOutput->standby(); 4993 mStandby = true; 4994 mBytesWritten = 0; 4995 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 4996 keyValuePair.string()); 4997 } 4998 if (status == NO_ERROR && reconfig) { 4999 readOutputParameters_l(); 5000 sendIoConfigEvent_l(AUDIO_OUTPUT_CONFIG_CHANGED); 5001 } 5002 } 5003 5004 return reconfig || a2dpDeviceChanged; 5005} 5006 5007uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() const 5008{ 5009 uint32_t time; 5010 if (audio_has_proportional_frames(mFormat)) { 5011 time = PlaybackThread::activeSleepTimeUs(); 5012 } else { 5013 time = 10000; 5014 } 5015 return time; 5016} 5017 5018uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() const 5019{ 5020 uint32_t time; 5021 if (audio_has_proportional_frames(mFormat)) { 5022 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 5023 } else { 5024 time = 10000; 5025 } 5026 return time; 5027} 5028 5029uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() const 5030{ 5031 uint32_t time; 5032 if (audio_has_proportional_frames(mFormat)) { 5033 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 5034 } else { 5035 time = 10000; 5036 } 5037 return time; 5038} 5039 5040void AudioFlinger::DirectOutputThread::cacheParameters_l() 5041{ 5042 PlaybackThread::cacheParameters_l(); 5043 5044 // use shorter standby delay as on normal output to release 5045 // hardware resources as soon as possible 5046 // no delay on outputs with HW A/V sync 5047 if (usesHwAvSync()) { 5048 mStandbyDelayNs = 0; 5049 } else if ((mType == OFFLOAD) && !audio_has_proportional_frames(mFormat)) { 5050 mStandbyDelayNs = kOffloadStandbyDelayNs; 5051 } else { 5052 mStandbyDelayNs = microseconds(mActiveSleepTimeUs*2); 5053 } 5054} 5055 5056void AudioFlinger::DirectOutputThread::flushHw_l() 5057{ 5058 mOutput->flush(); 5059 mHwPaused = false; 5060 mFlushPending = false; 5061} 5062 5063// ---------------------------------------------------------------------------- 5064 5065AudioFlinger::AsyncCallbackThread::AsyncCallbackThread( 5066 const wp<AudioFlinger::PlaybackThread>& playbackThread) 5067 : Thread(false /*canCallJava*/), 5068 mPlaybackThread(playbackThread), 5069 mWriteAckSequence(0), 5070 mDrainSequence(0) 5071{ 5072} 5073 5074AudioFlinger::AsyncCallbackThread::~AsyncCallbackThread() 5075{ 5076} 5077 5078void AudioFlinger::AsyncCallbackThread::onFirstRef() 5079{ 5080 run("Offload Cbk", ANDROID_PRIORITY_URGENT_AUDIO); 5081} 5082 5083bool AudioFlinger::AsyncCallbackThread::threadLoop() 5084{ 5085 while (!exitPending()) { 5086 uint32_t writeAckSequence; 5087 uint32_t drainSequence; 5088 5089 { 5090 Mutex::Autolock _l(mLock); 5091 while (!((mWriteAckSequence & 1) || 5092 (mDrainSequence & 1) || 5093 exitPending())) { 5094 mWaitWorkCV.wait(mLock); 5095 } 5096 5097 if (exitPending()) { 5098 break; 5099 } 5100 ALOGV("AsyncCallbackThread mWriteAckSequence %d mDrainSequence %d", 5101 mWriteAckSequence, mDrainSequence); 5102 writeAckSequence = mWriteAckSequence; 5103 mWriteAckSequence &= ~1; 5104 drainSequence = mDrainSequence; 5105 mDrainSequence &= ~1; 5106 } 5107 { 5108 sp<AudioFlinger::PlaybackThread> playbackThread = mPlaybackThread.promote(); 5109 if (playbackThread != 0) { 5110 if (writeAckSequence & 1) { 5111 playbackThread->resetWriteBlocked(writeAckSequence >> 1); 5112 } 5113 if (drainSequence & 1) { 5114 playbackThread->resetDraining(drainSequence >> 1); 5115 } 5116 } 5117 } 5118 } 5119 return false; 5120} 5121 5122void AudioFlinger::AsyncCallbackThread::exit() 5123{ 5124 ALOGV("AsyncCallbackThread::exit"); 5125 Mutex::Autolock _l(mLock); 5126 requestExit(); 5127 mWaitWorkCV.broadcast(); 5128} 5129 5130void AudioFlinger::AsyncCallbackThread::setWriteBlocked(uint32_t sequence) 5131{ 5132 Mutex::Autolock _l(mLock); 5133 // bit 0 is cleared 5134 mWriteAckSequence = sequence << 1; 5135} 5136 5137void AudioFlinger::AsyncCallbackThread::resetWriteBlocked() 5138{ 5139 Mutex::Autolock _l(mLock); 5140 // ignore unexpected callbacks 5141 if (mWriteAckSequence & 2) { 5142 mWriteAckSequence |= 1; 5143 mWaitWorkCV.signal(); 5144 } 5145} 5146 5147void AudioFlinger::AsyncCallbackThread::setDraining(uint32_t sequence) 5148{ 5149 Mutex::Autolock _l(mLock); 5150 // bit 0 is cleared 5151 mDrainSequence = sequence << 1; 5152} 5153 5154void AudioFlinger::AsyncCallbackThread::resetDraining() 5155{ 5156 Mutex::Autolock _l(mLock); 5157 // ignore unexpected callbacks 5158 if (mDrainSequence & 2) { 5159 mDrainSequence |= 1; 5160 mWaitWorkCV.signal(); 5161 } 5162} 5163 5164 5165// ---------------------------------------------------------------------------- 5166AudioFlinger::OffloadThread::OffloadThread(const sp<AudioFlinger>& audioFlinger, 5167 AudioStreamOut* output, audio_io_handle_t id, uint32_t device, bool systemReady) 5168 : DirectOutputThread(audioFlinger, output, id, device, OFFLOAD, systemReady), 5169 mPausedBytesRemaining(0) 5170{ 5171 //FIXME: mStandby should be set to true by ThreadBase constructor 5172 mStandby = true; 5173} 5174 5175void AudioFlinger::OffloadThread::threadLoop_exit() 5176{ 5177 if (mFlushPending || mHwPaused) { 5178 // If a flush is pending or track was paused, just discard buffered data 5179 flushHw_l(); 5180 } else { 5181 mMixerStatus = MIXER_DRAIN_ALL; 5182 threadLoop_drain(); 5183 } 5184 if (mUseAsyncWrite) { 5185 ALOG_ASSERT(mCallbackThread != 0); 5186 mCallbackThread->exit(); 5187 } 5188 PlaybackThread::threadLoop_exit(); 5189} 5190 5191AudioFlinger::PlaybackThread::mixer_state AudioFlinger::OffloadThread::prepareTracks_l( 5192 Vector< sp<Track> > *tracksToRemove 5193) 5194{ 5195 size_t count = mActiveTracks.size(); 5196 5197 mixer_state mixerStatus = MIXER_IDLE; 5198 bool doHwPause = false; 5199 bool doHwResume = false; 5200 5201 ALOGV("OffloadThread::prepareTracks_l active tracks %d", count); 5202 5203 // find out which tracks need to be processed 5204 for (size_t i = 0; i < count; i++) { 5205 sp<Track> t = mActiveTracks[i].promote(); 5206 // The track died recently 5207 if (t == 0) { 5208 continue; 5209 } 5210 Track* const track = t.get(); 5211 audio_track_cblk_t* cblk = track->cblk(); 5212 // Only consider last track started for volume and mixer state control. 5213 // In theory an older track could underrun and restart after the new one starts 5214 // but as we only care about the transition phase between two tracks on a 5215 // direct output, it is not a problem to ignore the underrun case. 5216 sp<Track> l = mLatestActiveTrack.promote(); 5217 bool last = l.get() == track; 5218 5219 if (track->isInvalid()) { 5220 ALOGW("An invalidated track shouldn't be in active list"); 5221 tracksToRemove->add(track); 5222 continue; 5223 } 5224 5225 if (track->mState == TrackBase::IDLE) { 5226 ALOGW("An idle track shouldn't be in active list"); 5227 continue; 5228 } 5229 5230 if (track->isPausing()) { 5231 track->setPaused(); 5232 if (last) { 5233 if (mHwSupportsPause && !mHwPaused) { 5234 doHwPause = true; 5235 mHwPaused = true; 5236 } 5237 // If we were part way through writing the mixbuffer to 5238 // the HAL we must save this until we resume 5239 // BUG - this will be wrong if a different track is made active, 5240 // in that case we want to discard the pending data in the 5241 // mixbuffer and tell the client to present it again when the 5242 // track is resumed 5243 mPausedWriteLength = mCurrentWriteLength; 5244 mPausedBytesRemaining = mBytesRemaining; 5245 mBytesRemaining = 0; // stop writing 5246 } 5247 tracksToRemove->add(track); 5248 } else if (track->isFlushPending()) { 5249 track->flushAck(); 5250 if (last) { 5251 mFlushPending = true; 5252 } 5253 } else if (track->isResumePending()){ 5254 track->resumeAck(); 5255 if (last) { 5256 if (mPausedBytesRemaining) { 5257 // Need to continue write that was interrupted 5258 mCurrentWriteLength = mPausedWriteLength; 5259 mBytesRemaining = mPausedBytesRemaining; 5260 mPausedBytesRemaining = 0; 5261 } 5262 if (mHwPaused) { 5263 doHwResume = true; 5264 mHwPaused = false; 5265 // threadLoop_mix() will handle the case that we need to 5266 // resume an interrupted write 5267 } 5268 // enable write to audio HAL 5269 mSleepTimeUs = 0; 5270 5271 // Do not handle new data in this iteration even if track->framesReady() 5272 mixerStatus = MIXER_TRACKS_ENABLED; 5273 } 5274 } else if (track->framesReady() && track->isReady() && 5275 !track->isPaused() && !track->isTerminated() && !track->isStopping_2()) { 5276 ALOGVV("OffloadThread: track %d s=%08x [OK]", track->name(), cblk->mServer); 5277 if (track->mFillingUpStatus == Track::FS_FILLED) { 5278 track->mFillingUpStatus = Track::FS_ACTIVE; 5279 // make sure processVolume_l() will apply new volume even if 0 5280 mLeftVolFloat = mRightVolFloat = -1.0; 5281 } 5282 5283 if (last) { 5284 sp<Track> previousTrack = mPreviousTrack.promote(); 5285 if (previousTrack != 0) { 5286 if (track != previousTrack.get()) { 5287 // Flush any data still being written from last track 5288 mBytesRemaining = 0; 5289 if (mPausedBytesRemaining) { 5290 // Last track was paused so we also need to flush saved 5291 // mixbuffer state and invalidate track so that it will 5292 // re-submit that unwritten data when it is next resumed 5293 mPausedBytesRemaining = 0; 5294 // Invalidate is a bit drastic - would be more efficient 5295 // to have a flag to tell client that some of the 5296 // previously written data was lost 5297 previousTrack->invalidate(); 5298 } 5299 // flush data already sent to the DSP if changing audio session as audio 5300 // comes from a different source. Also invalidate previous track to force a 5301 // seek when resuming. 5302 if (previousTrack->sessionId() != track->sessionId()) { 5303 previousTrack->invalidate(); 5304 } 5305 } 5306 } 5307 mPreviousTrack = track; 5308 // reset retry count 5309 track->mRetryCount = kMaxTrackRetriesOffload; 5310 mActiveTrack = t; 5311 mixerStatus = MIXER_TRACKS_READY; 5312 } 5313 } else { 5314 ALOGVV("OffloadThread: track %d s=%08x [NOT READY]", track->name(), cblk->mServer); 5315 if (track->isStopping_1()) { 5316 // Hardware buffer can hold a large amount of audio so we must 5317 // wait for all current track's data to drain before we say 5318 // that the track is stopped. 5319 if (mBytesRemaining == 0) { 5320 // Only start draining when all data in mixbuffer 5321 // has been written 5322 ALOGV("OffloadThread: underrun and STOPPING_1 -> draining, STOPPING_2"); 5323 track->mState = TrackBase::STOPPING_2; // so presentation completes after drain 5324 // do not drain if no data was ever sent to HAL (mStandby == true) 5325 if (last && !mStandby) { 5326 // do not modify drain sequence if we are already draining. This happens 5327 // when resuming from pause after drain. 5328 if ((mDrainSequence & 1) == 0) { 5329 mSleepTimeUs = 0; 5330 mStandbyTimeNs = systemTime() + mStandbyDelayNs; 5331 mixerStatus = MIXER_DRAIN_TRACK; 5332 mDrainSequence += 2; 5333 } 5334 if (mHwPaused) { 5335 // It is possible to move from PAUSED to STOPPING_1 without 5336 // a resume so we must ensure hardware is running 5337 doHwResume = true; 5338 mHwPaused = false; 5339 } 5340 } 5341 } 5342 } else if (track->isStopping_2()) { 5343 // Drain has completed or we are in standby, signal presentation complete 5344 if (!(mDrainSequence & 1) || !last || mStandby) { 5345 track->mState = TrackBase::STOPPED; 5346 size_t audioHALFrames = 5347 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 5348 size_t framesWritten = 5349 mBytesWritten / mOutput->getFrameSize(); 5350 track->presentationComplete(framesWritten, audioHALFrames); 5351 track->reset(); 5352 tracksToRemove->add(track); 5353 } 5354 } else { 5355 // No buffers for this track. Give it a few chances to 5356 // fill a buffer, then remove it from active list. 5357 if (--(track->mRetryCount) <= 0) { 5358 ALOGV("OffloadThread: BUFFER TIMEOUT: remove(%d) from active list", 5359 track->name()); 5360 tracksToRemove->add(track); 5361 // indicate to client process that the track was disabled because of underrun; 5362 // it will then automatically call start() when data is available 5363 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 5364 } else if (last){ 5365 mixerStatus = MIXER_TRACKS_ENABLED; 5366 } 5367 } 5368 } 5369 // compute volume for this track 5370 processVolume_l(track, last); 5371 } 5372 5373 // make sure the pause/flush/resume sequence is executed in the right order. 5374 // If a flush is pending and a track is active but the HW is not paused, force a HW pause 5375 // before flush and then resume HW. This can happen in case of pause/flush/resume 5376 // if resume is received before pause is executed. 5377 if (!mStandby && (doHwPause || (mFlushPending && !mHwPaused && (count != 0)))) { 5378 mOutput->stream->pause(mOutput->stream); 5379 } 5380 if (mFlushPending) { 5381 flushHw_l(); 5382 } 5383 if (!mStandby && doHwResume) { 5384 mOutput->stream->resume(mOutput->stream); 5385 } 5386 5387 // remove all the tracks that need to be... 5388 removeTracks_l(*tracksToRemove); 5389 5390 return mixerStatus; 5391} 5392 5393// must be called with thread mutex locked 5394bool AudioFlinger::OffloadThread::waitingAsyncCallback_l() 5395{ 5396 ALOGVV("waitingAsyncCallback_l mWriteAckSequence %d mDrainSequence %d", 5397 mWriteAckSequence, mDrainSequence); 5398 if (mUseAsyncWrite && ((mWriteAckSequence & 1) || (mDrainSequence & 1))) { 5399 return true; 5400 } 5401 return false; 5402} 5403 5404bool AudioFlinger::OffloadThread::waitingAsyncCallback() 5405{ 5406 Mutex::Autolock _l(mLock); 5407 return waitingAsyncCallback_l(); 5408} 5409 5410void AudioFlinger::OffloadThread::flushHw_l() 5411{ 5412 DirectOutputThread::flushHw_l(); 5413 // Flush anything still waiting in the mixbuffer 5414 mCurrentWriteLength = 0; 5415 mBytesRemaining = 0; 5416 mPausedWriteLength = 0; 5417 mPausedBytesRemaining = 0; 5418 5419 if (mUseAsyncWrite) { 5420 // discard any pending drain or write ack by incrementing sequence 5421 mWriteAckSequence = (mWriteAckSequence + 2) & ~1; 5422 mDrainSequence = (mDrainSequence + 2) & ~1; 5423 ALOG_ASSERT(mCallbackThread != 0); 5424 mCallbackThread->setWriteBlocked(mWriteAckSequence); 5425 mCallbackThread->setDraining(mDrainSequence); 5426 } 5427} 5428 5429// ---------------------------------------------------------------------------- 5430 5431AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 5432 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id, bool systemReady) 5433 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->outDevice(), 5434 systemReady, DUPLICATING), 5435 mWaitTimeMs(UINT_MAX) 5436{ 5437 addOutputTrack(mainThread); 5438} 5439 5440AudioFlinger::DuplicatingThread::~DuplicatingThread() 5441{ 5442 for (size_t i = 0; i < mOutputTracks.size(); i++) { 5443 mOutputTracks[i]->destroy(); 5444 } 5445} 5446 5447void AudioFlinger::DuplicatingThread::threadLoop_mix() 5448{ 5449 // mix buffers... 5450 if (outputsReady(outputTracks)) { 5451 mAudioMixer->process(); 5452 } else { 5453 if (mMixerBufferValid) { 5454 memset(mMixerBuffer, 0, mMixerBufferSize); 5455 } else { 5456 memset(mSinkBuffer, 0, mSinkBufferSize); 5457 } 5458 } 5459 mSleepTimeUs = 0; 5460 writeFrames = mNormalFrameCount; 5461 mCurrentWriteLength = mSinkBufferSize; 5462 mStandbyTimeNs = systemTime() + mStandbyDelayNs; 5463} 5464 5465void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() 5466{ 5467 if (mSleepTimeUs == 0) { 5468 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 5469 mSleepTimeUs = mActiveSleepTimeUs; 5470 } else { 5471 mSleepTimeUs = mIdleSleepTimeUs; 5472 } 5473 } else if (mBytesWritten != 0) { 5474 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 5475 writeFrames = mNormalFrameCount; 5476 memset(mSinkBuffer, 0, mSinkBufferSize); 5477 } else { 5478 // flush remaining overflow buffers in output tracks 5479 writeFrames = 0; 5480 } 5481 mSleepTimeUs = 0; 5482 } 5483} 5484 5485ssize_t AudioFlinger::DuplicatingThread::threadLoop_write() 5486{ 5487 for (size_t i = 0; i < outputTracks.size(); i++) { 5488 outputTracks[i]->write(mSinkBuffer, writeFrames); 5489 } 5490 mStandby = false; 5491 return (ssize_t)mSinkBufferSize; 5492} 5493 5494void AudioFlinger::DuplicatingThread::threadLoop_standby() 5495{ 5496 // DuplicatingThread implements standby by stopping all tracks 5497 for (size_t i = 0; i < outputTracks.size(); i++) { 5498 outputTracks[i]->stop(); 5499 } 5500} 5501 5502void AudioFlinger::DuplicatingThread::saveOutputTracks() 5503{ 5504 outputTracks = mOutputTracks; 5505} 5506 5507void AudioFlinger::DuplicatingThread::clearOutputTracks() 5508{ 5509 outputTracks.clear(); 5510} 5511 5512void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 5513{ 5514 Mutex::Autolock _l(mLock); 5515 // The downstream MixerThread consumes thread->frameCount() amount of frames per mix pass. 5516 // Adjust for thread->sampleRate() to determine minimum buffer frame count. 5517 // Then triple buffer because Threads do not run synchronously and may not be clock locked. 5518 const size_t frameCount = 5519 3 * sourceFramesNeeded(mSampleRate, thread->frameCount(), thread->sampleRate()); 5520 // TODO: Consider asynchronous sample rate conversion to handle clock disparity 5521 // from different OutputTracks and their associated MixerThreads (e.g. one may 5522 // nearly empty and the other may be dropping data). 5523 5524 sp<OutputTrack> outputTrack = new OutputTrack(thread, 5525 this, 5526 mSampleRate, 5527 mFormat, 5528 mChannelMask, 5529 frameCount, 5530 IPCThreadState::self()->getCallingUid()); 5531 if (outputTrack->cblk() != NULL) { 5532 thread->setStreamVolume(AUDIO_STREAM_PATCH, 1.0f); 5533 mOutputTracks.add(outputTrack); 5534 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack.get(), thread); 5535 updateWaitTime_l(); 5536 } 5537} 5538 5539void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 5540{ 5541 Mutex::Autolock _l(mLock); 5542 for (size_t i = 0; i < mOutputTracks.size(); i++) { 5543 if (mOutputTracks[i]->thread() == thread) { 5544 mOutputTracks[i]->destroy(); 5545 mOutputTracks.removeAt(i); 5546 updateWaitTime_l(); 5547 if (thread->getOutput() == mOutput) { 5548 mOutput = NULL; 5549 } 5550 return; 5551 } 5552 } 5553 ALOGV("removeOutputTrack(): unknown thread: %p", thread); 5554} 5555 5556// caller must hold mLock 5557void AudioFlinger::DuplicatingThread::updateWaitTime_l() 5558{ 5559 mWaitTimeMs = UINT_MAX; 5560 for (size_t i = 0; i < mOutputTracks.size(); i++) { 5561 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 5562 if (strong != 0) { 5563 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 5564 if (waitTimeMs < mWaitTimeMs) { 5565 mWaitTimeMs = waitTimeMs; 5566 } 5567 } 5568 } 5569} 5570 5571 5572bool AudioFlinger::DuplicatingThread::outputsReady( 5573 const SortedVector< sp<OutputTrack> > &outputTracks) 5574{ 5575 for (size_t i = 0; i < outputTracks.size(); i++) { 5576 sp<ThreadBase> thread = outputTracks[i]->thread().promote(); 5577 if (thread == 0) { 5578 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", 5579 outputTracks[i].get()); 5580 return false; 5581 } 5582 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 5583 // see note at standby() declaration 5584 if (playbackThread->standby() && !playbackThread->isSuspended()) { 5585 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), 5586 thread.get()); 5587 return false; 5588 } 5589 } 5590 return true; 5591} 5592 5593uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() const 5594{ 5595 return (mWaitTimeMs * 1000) / 2; 5596} 5597 5598void AudioFlinger::DuplicatingThread::cacheParameters_l() 5599{ 5600 // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first 5601 updateWaitTime_l(); 5602 5603 MixerThread::cacheParameters_l(); 5604} 5605 5606// ---------------------------------------------------------------------------- 5607// Record 5608// ---------------------------------------------------------------------------- 5609 5610AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 5611 AudioStreamIn *input, 5612 audio_io_handle_t id, 5613 audio_devices_t outDevice, 5614 audio_devices_t inDevice, 5615 bool systemReady 5616#ifdef TEE_SINK 5617 , const sp<NBAIO_Sink>& teeSink 5618#endif 5619 ) : 5620 ThreadBase(audioFlinger, id, outDevice, inDevice, RECORD, systemReady), 5621 mInput(input), mActiveTracksGen(0), mRsmpInBuffer(NULL), 5622 // mRsmpInFrames and mRsmpInFramesP2 are set by readInputParameters_l() 5623 mRsmpInRear(0) 5624#ifdef TEE_SINK 5625 , mTeeSink(teeSink) 5626#endif 5627 , mReadOnlyHeap(new MemoryDealer(kRecordThreadReadOnlyHeapSize, 5628 "RecordThreadRO", MemoryHeapBase::READ_ONLY)) 5629 // mFastCapture below 5630 , mFastCaptureFutex(0) 5631 // mInputSource 5632 // mPipeSink 5633 // mPipeSource 5634 , mPipeFramesP2(0) 5635 // mPipeMemory 5636 // mFastCaptureNBLogWriter 5637 , mFastTrackAvail(false) 5638{ 5639 snprintf(mThreadName, kThreadNameLength, "AudioIn_%X", id); 5640 mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mThreadName); 5641 5642 readInputParameters_l(); 5643 5644 // create an NBAIO source for the HAL input stream, and negotiate 5645 mInputSource = new AudioStreamInSource(input->stream); 5646 size_t numCounterOffers = 0; 5647 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount, mFormat)}; 5648 ssize_t index = mInputSource->negotiate(offers, 1, NULL, numCounterOffers); 5649 ALOG_ASSERT(index == 0); 5650 5651 // initialize fast capture depending on configuration 5652 bool initFastCapture; 5653 switch (kUseFastCapture) { 5654 case FastCapture_Never: 5655 initFastCapture = false; 5656 break; 5657 case FastCapture_Always: 5658 initFastCapture = true; 5659 break; 5660 case FastCapture_Static: 5661 initFastCapture = (mFrameCount * 1000) / mSampleRate < kMinNormalCaptureBufferSizeMs; 5662 break; 5663 // case FastCapture_Dynamic: 5664 } 5665 5666 if (initFastCapture) { 5667 // create a Pipe for FastCapture to write to, and for us and fast tracks to read from 5668 NBAIO_Format format = mInputSource->format(); 5669 size_t pipeFramesP2 = roundup(mSampleRate / 25); // double-buffering of 20 ms each 5670 size_t pipeSize = pipeFramesP2 * Format_frameSize(format); 5671 void *pipeBuffer; 5672 const sp<MemoryDealer> roHeap(readOnlyHeap()); 5673 sp<IMemory> pipeMemory; 5674 if ((roHeap == 0) || 5675 (pipeMemory = roHeap->allocate(pipeSize)) == 0 || 5676 (pipeBuffer = pipeMemory->pointer()) == NULL) { 5677 ALOGE("not enough memory for pipe buffer size=%zu", pipeSize); 5678 goto failed; 5679 } 5680 // pipe will be shared directly with fast clients, so clear to avoid leaking old information 5681 memset(pipeBuffer, 0, pipeSize); 5682 Pipe *pipe = new Pipe(pipeFramesP2, format, pipeBuffer); 5683 const NBAIO_Format offers[1] = {format}; 5684 size_t numCounterOffers = 0; 5685 ssize_t index = pipe->negotiate(offers, 1, NULL, numCounterOffers); 5686 ALOG_ASSERT(index == 0); 5687 mPipeSink = pipe; 5688 PipeReader *pipeReader = new PipeReader(*pipe); 5689 numCounterOffers = 0; 5690 index = pipeReader->negotiate(offers, 1, NULL, numCounterOffers); 5691 ALOG_ASSERT(index == 0); 5692 mPipeSource = pipeReader; 5693 mPipeFramesP2 = pipeFramesP2; 5694 mPipeMemory = pipeMemory; 5695 5696 // create fast capture 5697 mFastCapture = new FastCapture(); 5698 FastCaptureStateQueue *sq = mFastCapture->sq(); 5699#ifdef STATE_QUEUE_DUMP 5700 // FIXME 5701#endif 5702 FastCaptureState *state = sq->begin(); 5703 state->mCblk = NULL; 5704 state->mInputSource = mInputSource.get(); 5705 state->mInputSourceGen++; 5706 state->mPipeSink = pipe; 5707 state->mPipeSinkGen++; 5708 state->mFrameCount = mFrameCount; 5709 state->mCommand = FastCaptureState::COLD_IDLE; 5710 // already done in constructor initialization list 5711 //mFastCaptureFutex = 0; 5712 state->mColdFutexAddr = &mFastCaptureFutex; 5713 state->mColdGen++; 5714 state->mDumpState = &mFastCaptureDumpState; 5715#ifdef TEE_SINK 5716 // FIXME 5717#endif 5718 mFastCaptureNBLogWriter = audioFlinger->newWriter_l(kFastCaptureLogSize, "FastCapture"); 5719 state->mNBLogWriter = mFastCaptureNBLogWriter.get(); 5720 sq->end(); 5721 sq->push(FastCaptureStateQueue::BLOCK_UNTIL_PUSHED); 5722 5723 // start the fast capture 5724 mFastCapture->run("FastCapture", ANDROID_PRIORITY_URGENT_AUDIO); 5725 pid_t tid = mFastCapture->getTid(); 5726 sendPrioConfigEvent(getpid_cached, tid, kPriorityFastMixer); 5727#ifdef AUDIO_WATCHDOG 5728 // FIXME 5729#endif 5730 5731 mFastTrackAvail = true; 5732 } 5733failed: ; 5734 5735 // FIXME mNormalSource 5736} 5737 5738AudioFlinger::RecordThread::~RecordThread() 5739{ 5740 if (mFastCapture != 0) { 5741 FastCaptureStateQueue *sq = mFastCapture->sq(); 5742 FastCaptureState *state = sq->begin(); 5743 if (state->mCommand == FastCaptureState::COLD_IDLE) { 5744 int32_t old = android_atomic_inc(&mFastCaptureFutex); 5745 if (old == -1) { 5746 (void) syscall(__NR_futex, &mFastCaptureFutex, FUTEX_WAKE_PRIVATE, 1); 5747 } 5748 } 5749 state->mCommand = FastCaptureState::EXIT; 5750 sq->end(); 5751 sq->push(FastCaptureStateQueue::BLOCK_UNTIL_PUSHED); 5752 mFastCapture->join(); 5753 mFastCapture.clear(); 5754 } 5755 mAudioFlinger->unregisterWriter(mFastCaptureNBLogWriter); 5756 mAudioFlinger->unregisterWriter(mNBLogWriter); 5757 free(mRsmpInBuffer); 5758} 5759 5760void AudioFlinger::RecordThread::onFirstRef() 5761{ 5762 run(mThreadName, PRIORITY_URGENT_AUDIO); 5763} 5764 5765bool AudioFlinger::RecordThread::threadLoop() 5766{ 5767 nsecs_t lastWarning = 0; 5768 5769 inputStandBy(); 5770 5771reacquire_wakelock: 5772 sp<RecordTrack> activeTrack; 5773 int activeTracksGen; 5774 { 5775 Mutex::Autolock _l(mLock); 5776 size_t size = mActiveTracks.size(); 5777 activeTracksGen = mActiveTracksGen; 5778 if (size > 0) { 5779 // FIXME an arbitrary choice 5780 activeTrack = mActiveTracks[0]; 5781 acquireWakeLock_l(activeTrack->uid()); 5782 if (size > 1) { 5783 SortedVector<int> tmp; 5784 for (size_t i = 0; i < size; i++) { 5785 tmp.add(mActiveTracks[i]->uid()); 5786 } 5787 updateWakeLockUids_l(tmp); 5788 } 5789 } else { 5790 acquireWakeLock_l(-1); 5791 } 5792 } 5793 5794 mTimestamp.mTimebaseOffset[ExtendedTimestamp::TIMEBASE_BOOTTIME] = 5795 gBoottime.getBoottimeOffset(); 5796 5797 // used to request a deferred sleep, to be executed later while mutex is unlocked 5798 uint32_t sleepUs = 0; 5799 5800 // loop while there is work to do 5801 for (;;) { 5802 Vector< sp<EffectChain> > effectChains; 5803 5804 // sleep with mutex unlocked 5805 if (sleepUs > 0) { 5806 ATRACE_BEGIN("sleep"); 5807 usleep(sleepUs); 5808 ATRACE_END(); 5809 sleepUs = 0; 5810 } 5811 5812 // activeTracks accumulates a copy of a subset of mActiveTracks 5813 Vector< sp<RecordTrack> > activeTracks; 5814 5815 // reference to the (first and only) active fast track 5816 sp<RecordTrack> fastTrack; 5817 5818 // reference to a fast track which is about to be removed 5819 sp<RecordTrack> fastTrackToRemove; 5820 5821 { // scope for mLock 5822 Mutex::Autolock _l(mLock); 5823 5824 processConfigEvents_l(); 5825 5826 // check exitPending here because checkForNewParameters_l() and 5827 // checkForNewParameters_l() can temporarily release mLock 5828 if (exitPending()) { 5829 break; 5830 } 5831 5832 // if no active track(s), then standby and release wakelock 5833 size_t size = mActiveTracks.size(); 5834 if (size == 0) { 5835 standbyIfNotAlreadyInStandby(); 5836 // exitPending() can't become true here 5837 releaseWakeLock_l(); 5838 ALOGV("RecordThread: loop stopping"); 5839 // go to sleep 5840 mWaitWorkCV.wait(mLock); 5841 ALOGV("RecordThread: loop starting"); 5842 goto reacquire_wakelock; 5843 } 5844 5845 if (mActiveTracksGen != activeTracksGen) { 5846 activeTracksGen = mActiveTracksGen; 5847 SortedVector<int> tmp; 5848 for (size_t i = 0; i < size; i++) { 5849 tmp.add(mActiveTracks[i]->uid()); 5850 } 5851 updateWakeLockUids_l(tmp); 5852 } 5853 5854 bool doBroadcast = false; 5855 for (size_t i = 0; i < size; ) { 5856 5857 activeTrack = mActiveTracks[i]; 5858 if (activeTrack->isTerminated()) { 5859 if (activeTrack->isFastTrack()) { 5860 ALOG_ASSERT(fastTrackToRemove == 0); 5861 fastTrackToRemove = activeTrack; 5862 } 5863 removeTrack_l(activeTrack); 5864 mActiveTracks.remove(activeTrack); 5865 mActiveTracksGen++; 5866 size--; 5867 continue; 5868 } 5869 5870 TrackBase::track_state activeTrackState = activeTrack->mState; 5871 switch (activeTrackState) { 5872 5873 case TrackBase::PAUSING: 5874 mActiveTracks.remove(activeTrack); 5875 mActiveTracksGen++; 5876 doBroadcast = true; 5877 size--; 5878 continue; 5879 5880 case TrackBase::STARTING_1: 5881 sleepUs = 10000; 5882 i++; 5883 continue; 5884 5885 case TrackBase::STARTING_2: 5886 doBroadcast = true; 5887 mStandby = false; 5888 activeTrack->mState = TrackBase::ACTIVE; 5889 break; 5890 5891 case TrackBase::ACTIVE: 5892 break; 5893 5894 case TrackBase::IDLE: 5895 i++; 5896 continue; 5897 5898 default: 5899 LOG_ALWAYS_FATAL("Unexpected activeTrackState %d", activeTrackState); 5900 } 5901 5902 activeTracks.add(activeTrack); 5903 i++; 5904 5905 if (activeTrack->isFastTrack()) { 5906 ALOG_ASSERT(!mFastTrackAvail); 5907 ALOG_ASSERT(fastTrack == 0); 5908 fastTrack = activeTrack; 5909 } 5910 } 5911 if (doBroadcast) { 5912 mStartStopCond.broadcast(); 5913 } 5914 5915 // sleep if there are no active tracks to process 5916 if (activeTracks.size() == 0) { 5917 if (sleepUs == 0) { 5918 sleepUs = kRecordThreadSleepUs; 5919 } 5920 continue; 5921 } 5922 sleepUs = 0; 5923 5924 lockEffectChains_l(effectChains); 5925 } 5926 5927 // thread mutex is now unlocked, mActiveTracks unknown, activeTracks.size() > 0 5928 5929 size_t size = effectChains.size(); 5930 for (size_t i = 0; i < size; i++) { 5931 // thread mutex is not locked, but effect chain is locked 5932 effectChains[i]->process_l(); 5933 } 5934 5935 // Push a new fast capture state if fast capture is not already running, or cblk change 5936 if (mFastCapture != 0) { 5937 FastCaptureStateQueue *sq = mFastCapture->sq(); 5938 FastCaptureState *state = sq->begin(); 5939 bool didModify = false; 5940 FastCaptureStateQueue::block_t block = FastCaptureStateQueue::BLOCK_UNTIL_PUSHED; 5941 if (state->mCommand != FastCaptureState::READ_WRITE /* FIXME && 5942 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)*/) { 5943 if (state->mCommand == FastCaptureState::COLD_IDLE) { 5944 int32_t old = android_atomic_inc(&mFastCaptureFutex); 5945 if (old == -1) { 5946 (void) syscall(__NR_futex, &mFastCaptureFutex, FUTEX_WAKE_PRIVATE, 1); 5947 } 5948 } 5949 state->mCommand = FastCaptureState::READ_WRITE; 5950#if 0 // FIXME 5951 mFastCaptureDumpState.increaseSamplingN(mAudioFlinger->isLowRamDevice() ? 5952 FastThreadDumpState::kSamplingNforLowRamDevice : 5953 FastThreadDumpState::kSamplingN); 5954#endif 5955 didModify = true; 5956 } 5957 audio_track_cblk_t *cblkOld = state->mCblk; 5958 audio_track_cblk_t *cblkNew = fastTrack != 0 ? fastTrack->cblk() : NULL; 5959 if (cblkNew != cblkOld) { 5960 state->mCblk = cblkNew; 5961 // block until acked if removing a fast track 5962 if (cblkOld != NULL) { 5963 block = FastCaptureStateQueue::BLOCK_UNTIL_ACKED; 5964 } 5965 didModify = true; 5966 } 5967 sq->end(didModify); 5968 if (didModify) { 5969 sq->push(block); 5970#if 0 5971 if (kUseFastCapture == FastCapture_Dynamic) { 5972 mNormalSource = mPipeSource; 5973 } 5974#endif 5975 } 5976 } 5977 5978 // now run the fast track destructor with thread mutex unlocked 5979 fastTrackToRemove.clear(); 5980 5981 // Read from HAL to keep up with fastest client if multiple active tracks, not slowest one. 5982 // Only the client(s) that are too slow will overrun. But if even the fastest client is too 5983 // slow, then this RecordThread will overrun by not calling HAL read often enough. 5984 // If destination is non-contiguous, first read past the nominal end of buffer, then 5985 // copy to the right place. Permitted because mRsmpInBuffer was over-allocated. 5986 5987 int32_t rear = mRsmpInRear & (mRsmpInFramesP2 - 1); 5988 ssize_t framesRead; 5989 5990 // If an NBAIO source is present, use it to read the normal capture's data 5991 if (mPipeSource != 0) { 5992 size_t framesToRead = mBufferSize / mFrameSize; 5993 framesRead = mPipeSource->read((uint8_t*)mRsmpInBuffer + rear * mFrameSize, 5994 framesToRead); 5995 if (framesRead == 0) { 5996 // since pipe is non-blocking, simulate blocking input 5997 sleepUs = (framesToRead * 1000000LL) / mSampleRate; 5998 } 5999 // otherwise use the HAL / AudioStreamIn directly 6000 } else { 6001 ssize_t bytesRead = mInput->stream->read(mInput->stream, 6002 (uint8_t*)mRsmpInBuffer + rear * mFrameSize, mBufferSize); 6003 if (bytesRead < 0) { 6004 framesRead = bytesRead; 6005 } else { 6006 framesRead = bytesRead / mFrameSize; 6007 } 6008 } 6009 6010 // Update server timestamp with server stats 6011 // systemTime() is optional if the hardware supports timestamps. 6012 mTimestamp.mPosition[ExtendedTimestamp::LOCATION_SERVER] += framesRead; 6013 mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_SERVER] = systemTime(); 6014 6015 // Update server timestamp with kernel stats 6016 if (mInput->stream->get_capture_position != nullptr) { 6017 int64_t position, time; 6018 int ret = mInput->stream->get_capture_position(mInput->stream, &position, &time); 6019 if (ret == NO_ERROR) { 6020 mTimestamp.mPosition[ExtendedTimestamp::LOCATION_KERNEL] = position; 6021 mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL] = time; 6022 // Note: In general record buffers should tend to be empty in 6023 // a properly running pipeline. 6024 // 6025 // Also, it is not advantageous to call get_presentation_position during the read 6026 // as the read obtains a lock, preventing the timestamp call from executing. 6027 } 6028 } 6029 // Use this to track timestamp information 6030 // ALOGD("%s", mTimestamp.toString().c_str()); 6031 6032 if (framesRead < 0 || (framesRead == 0 && mPipeSource == 0)) { 6033 ALOGE("read failed: framesRead=%d", framesRead); 6034 // Force input into standby so that it tries to recover at next read attempt 6035 inputStandBy(); 6036 sleepUs = kRecordThreadSleepUs; 6037 } 6038 if (framesRead <= 0) { 6039 goto unlock; 6040 } 6041 ALOG_ASSERT(framesRead > 0); 6042 6043 if (mTeeSink != 0) { 6044 (void) mTeeSink->write((uint8_t*)mRsmpInBuffer + rear * mFrameSize, framesRead); 6045 } 6046 // If destination is non-contiguous, we now correct for reading past end of buffer. 6047 { 6048 size_t part1 = mRsmpInFramesP2 - rear; 6049 if ((size_t) framesRead > part1) { 6050 memcpy(mRsmpInBuffer, (uint8_t*)mRsmpInBuffer + mRsmpInFramesP2 * mFrameSize, 6051 (framesRead - part1) * mFrameSize); 6052 } 6053 } 6054 rear = mRsmpInRear += framesRead; 6055 6056 size = activeTracks.size(); 6057 // loop over each active track 6058 for (size_t i = 0; i < size; i++) { 6059 activeTrack = activeTracks[i]; 6060 6061 // skip fast tracks, as those are handled directly by FastCapture 6062 if (activeTrack->isFastTrack()) { 6063 continue; 6064 } 6065 6066 // TODO: This code probably should be moved to RecordTrack. 6067 // TODO: Update the activeTrack buffer converter in case of reconfigure. 6068 6069 enum { 6070 OVERRUN_UNKNOWN, 6071 OVERRUN_TRUE, 6072 OVERRUN_FALSE 6073 } overrun = OVERRUN_UNKNOWN; 6074 6075 // loop over getNextBuffer to handle circular sink 6076 for (;;) { 6077 6078 activeTrack->mSink.frameCount = ~0; 6079 status_t status = activeTrack->getNextBuffer(&activeTrack->mSink); 6080 size_t framesOut = activeTrack->mSink.frameCount; 6081 LOG_ALWAYS_FATAL_IF((status == OK) != (framesOut > 0)); 6082 6083 // check available frames and handle overrun conditions 6084 // if the record track isn't draining fast enough. 6085 bool hasOverrun; 6086 size_t framesIn; 6087 activeTrack->mResamplerBufferProvider->sync(&framesIn, &hasOverrun); 6088 if (hasOverrun) { 6089 overrun = OVERRUN_TRUE; 6090 } 6091 if (framesOut == 0 || framesIn == 0) { 6092 break; 6093 } 6094 6095 // Don't allow framesOut to be larger than what is possible with resampling 6096 // from framesIn. 6097 // This isn't strictly necessary but helps limit buffer resizing in 6098 // RecordBufferConverter. TODO: remove when no longer needed. 6099 framesOut = min(framesOut, 6100 destinationFramesPossible( 6101 framesIn, mSampleRate, activeTrack->mSampleRate)); 6102 // process frames from the RecordThread buffer provider to the RecordTrack buffer 6103 framesOut = activeTrack->mRecordBufferConverter->convert( 6104 activeTrack->mSink.raw, activeTrack->mResamplerBufferProvider, framesOut); 6105 6106 if (framesOut > 0 && (overrun == OVERRUN_UNKNOWN)) { 6107 overrun = OVERRUN_FALSE; 6108 } 6109 6110 if (activeTrack->mFramesToDrop == 0) { 6111 if (framesOut > 0) { 6112 activeTrack->mSink.frameCount = framesOut; 6113 activeTrack->releaseBuffer(&activeTrack->mSink); 6114 } 6115 } else { 6116 // FIXME could do a partial drop of framesOut 6117 if (activeTrack->mFramesToDrop > 0) { 6118 activeTrack->mFramesToDrop -= framesOut; 6119 if (activeTrack->mFramesToDrop <= 0) { 6120 activeTrack->clearSyncStartEvent(); 6121 } 6122 } else { 6123 activeTrack->mFramesToDrop += framesOut; 6124 if (activeTrack->mFramesToDrop >= 0 || activeTrack->mSyncStartEvent == 0 || 6125 activeTrack->mSyncStartEvent->isCancelled()) { 6126 ALOGW("Synced record %s, session %d, trigger session %d", 6127 (activeTrack->mFramesToDrop >= 0) ? "timed out" : "cancelled", 6128 activeTrack->sessionId(), 6129 (activeTrack->mSyncStartEvent != 0) ? 6130 activeTrack->mSyncStartEvent->triggerSession() : 0); 6131 activeTrack->clearSyncStartEvent(); 6132 } 6133 } 6134 } 6135 6136 if (framesOut == 0) { 6137 break; 6138 } 6139 } 6140 6141 switch (overrun) { 6142 case OVERRUN_TRUE: 6143 // client isn't retrieving buffers fast enough 6144 if (!activeTrack->setOverflow()) { 6145 nsecs_t now = systemTime(); 6146 // FIXME should lastWarning per track? 6147 if ((now - lastWarning) > kWarningThrottleNs) { 6148 ALOGW("RecordThread: buffer overflow"); 6149 lastWarning = now; 6150 } 6151 } 6152 break; 6153 case OVERRUN_FALSE: 6154 activeTrack->clearOverflow(); 6155 break; 6156 case OVERRUN_UNKNOWN: 6157 break; 6158 } 6159 6160 // update frame information and push timestamp out 6161 activeTrack->updateTrackFrameInfo( 6162 activeTrack->mServerProxy->framesReleased(), 6163 mTimestamp.mPosition[ExtendedTimestamp::LOCATION_SERVER], 6164 mSampleRate, mTimestamp); 6165 } 6166 6167unlock: 6168 // enable changes in effect chain 6169 unlockEffectChains(effectChains); 6170 // effectChains doesn't need to be cleared, since it is cleared by destructor at scope end 6171 } 6172 6173 standbyIfNotAlreadyInStandby(); 6174 6175 { 6176 Mutex::Autolock _l(mLock); 6177 for (size_t i = 0; i < mTracks.size(); i++) { 6178 sp<RecordTrack> track = mTracks[i]; 6179 track->invalidate(); 6180 } 6181 mActiveTracks.clear(); 6182 mActiveTracksGen++; 6183 mStartStopCond.broadcast(); 6184 } 6185 6186 releaseWakeLock(); 6187 6188 ALOGV("RecordThread %p exiting", this); 6189 return false; 6190} 6191 6192void AudioFlinger::RecordThread::standbyIfNotAlreadyInStandby() 6193{ 6194 if (!mStandby) { 6195 inputStandBy(); 6196 mStandby = true; 6197 } 6198} 6199 6200void AudioFlinger::RecordThread::inputStandBy() 6201{ 6202 // Idle the fast capture if it's currently running 6203 if (mFastCapture != 0) { 6204 FastCaptureStateQueue *sq = mFastCapture->sq(); 6205 FastCaptureState *state = sq->begin(); 6206 if (!(state->mCommand & FastCaptureState::IDLE)) { 6207 state->mCommand = FastCaptureState::COLD_IDLE; 6208 state->mColdFutexAddr = &mFastCaptureFutex; 6209 state->mColdGen++; 6210 mFastCaptureFutex = 0; 6211 sq->end(); 6212 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 6213 sq->push(FastCaptureStateQueue::BLOCK_UNTIL_ACKED); 6214#if 0 6215 if (kUseFastCapture == FastCapture_Dynamic) { 6216 // FIXME 6217 } 6218#endif 6219#ifdef AUDIO_WATCHDOG 6220 // FIXME 6221#endif 6222 } else { 6223 sq->end(false /*didModify*/); 6224 } 6225 } 6226 mInput->stream->common.standby(&mInput->stream->common); 6227} 6228 6229// RecordThread::createRecordTrack_l() must be called with AudioFlinger::mLock held 6230sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 6231 const sp<AudioFlinger::Client>& client, 6232 uint32_t sampleRate, 6233 audio_format_t format, 6234 audio_channel_mask_t channelMask, 6235 size_t *pFrameCount, 6236 int sessionId, 6237 size_t *notificationFrames, 6238 int uid, 6239 IAudioFlinger::track_flags_t *flags, 6240 pid_t tid, 6241 status_t *status) 6242{ 6243 size_t frameCount = *pFrameCount; 6244 sp<RecordTrack> track; 6245 status_t lStatus; 6246 6247 // client expresses a preference for FAST, but we get the final say 6248 if (*flags & IAudioFlinger::TRACK_FAST) { 6249 if ( 6250 // we formerly checked for a callback handler (non-0 tid), 6251 // but that is no longer required for TRANSFER_OBTAIN mode 6252 // 6253 // frame count is not specified, or is exactly the pipe depth 6254 ((frameCount == 0) || (frameCount == mPipeFramesP2)) && 6255 // PCM data 6256 audio_is_linear_pcm(format) && 6257 // native format 6258 (format == mFormat) && 6259 // native channel mask 6260 (channelMask == mChannelMask) && 6261 // native hardware sample rate 6262 (sampleRate == mSampleRate) && 6263 // record thread has an associated fast capture 6264 hasFastCapture() && 6265 // there are sufficient fast track slots available 6266 mFastTrackAvail 6267 ) { 6268 ALOGV("AUDIO_INPUT_FLAG_FAST accepted: frameCount=%u mFrameCount=%u", 6269 frameCount, mFrameCount); 6270 } else { 6271 ALOGV("AUDIO_INPUT_FLAG_FAST denied: frameCount=%u mFrameCount=%u mPipeFramesP2=%u " 6272 "format=%#x isLinear=%d channelMask=%#x sampleRate=%u mSampleRate=%u " 6273 "hasFastCapture=%d tid=%d mFastTrackAvail=%d", 6274 frameCount, mFrameCount, mPipeFramesP2, 6275 format, audio_is_linear_pcm(format), channelMask, sampleRate, mSampleRate, 6276 hasFastCapture(), tid, mFastTrackAvail); 6277 *flags &= ~IAudioFlinger::TRACK_FAST; 6278 } 6279 } 6280 6281 // compute track buffer size in frames, and suggest the notification frame count 6282 if (*flags & IAudioFlinger::TRACK_FAST) { 6283 // fast track: frame count is exactly the pipe depth 6284 frameCount = mPipeFramesP2; 6285 // ignore requested notificationFrames, and always notify exactly once every HAL buffer 6286 *notificationFrames = mFrameCount; 6287 } else { 6288 // not fast track: max notification period is resampled equivalent of one HAL buffer time 6289 // or 20 ms if there is a fast capture 6290 // TODO This could be a roundupRatio inline, and const 6291 size_t maxNotificationFrames = ((int64_t) (hasFastCapture() ? mSampleRate/50 : mFrameCount) 6292 * sampleRate + mSampleRate - 1) / mSampleRate; 6293 // minimum number of notification periods is at least kMinNotifications, 6294 // and at least kMinMs rounded up to a whole notification period (minNotificationsByMs) 6295 static const size_t kMinNotifications = 3; 6296 static const uint32_t kMinMs = 30; 6297 // TODO This could be a roundupRatio inline 6298 const size_t minFramesByMs = (sampleRate * kMinMs + 1000 - 1) / 1000; 6299 // TODO This could be a roundupRatio inline 6300 const size_t minNotificationsByMs = (minFramesByMs + maxNotificationFrames - 1) / 6301 maxNotificationFrames; 6302 const size_t minFrameCount = maxNotificationFrames * 6303 max(kMinNotifications, minNotificationsByMs); 6304 frameCount = max(frameCount, minFrameCount); 6305 if (*notificationFrames == 0 || *notificationFrames > maxNotificationFrames) { 6306 *notificationFrames = maxNotificationFrames; 6307 } 6308 } 6309 *pFrameCount = frameCount; 6310 6311 lStatus = initCheck(); 6312 if (lStatus != NO_ERROR) { 6313 ALOGE("createRecordTrack_l() audio driver not initialized"); 6314 goto Exit; 6315 } 6316 6317 { // scope for mLock 6318 Mutex::Autolock _l(mLock); 6319 6320 track = new RecordTrack(this, client, sampleRate, 6321 format, channelMask, frameCount, NULL, sessionId, uid, 6322 *flags, TrackBase::TYPE_DEFAULT); 6323 6324 lStatus = track->initCheck(); 6325 if (lStatus != NO_ERROR) { 6326 ALOGE("createRecordTrack_l() initCheck failed %d; no control block?", lStatus); 6327 // track must be cleared from the caller as the caller has the AF lock 6328 goto Exit; 6329 } 6330 mTracks.add(track); 6331 6332 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 6333 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 6334 mAudioFlinger->btNrecIsOff(); 6335 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 6336 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 6337 6338 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 6339 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 6340 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 6341 // so ask activity manager to do this on our behalf 6342 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 6343 } 6344 } 6345 6346 lStatus = NO_ERROR; 6347 6348Exit: 6349 *status = lStatus; 6350 return track; 6351} 6352 6353status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, 6354 AudioSystem::sync_event_t event, 6355 int triggerSession) 6356{ 6357 ALOGV("RecordThread::start event %d, triggerSession %d", event, triggerSession); 6358 sp<ThreadBase> strongMe = this; 6359 status_t status = NO_ERROR; 6360 6361 if (event == AudioSystem::SYNC_EVENT_NONE) { 6362 recordTrack->clearSyncStartEvent(); 6363 } else if (event != AudioSystem::SYNC_EVENT_SAME) { 6364 recordTrack->mSyncStartEvent = mAudioFlinger->createSyncEvent(event, 6365 triggerSession, 6366 recordTrack->sessionId(), 6367 syncStartEventCallback, 6368 recordTrack); 6369 // Sync event can be cancelled by the trigger session if the track is not in a 6370 // compatible state in which case we start record immediately 6371 if (recordTrack->mSyncStartEvent->isCancelled()) { 6372 recordTrack->clearSyncStartEvent(); 6373 } else { 6374 // do not wait for the event for more than AudioSystem::kSyncRecordStartTimeOutMs 6375 recordTrack->mFramesToDrop = - 6376 ((AudioSystem::kSyncRecordStartTimeOutMs * recordTrack->mSampleRate) / 1000); 6377 } 6378 } 6379 6380 { 6381 // This section is a rendezvous between binder thread executing start() and RecordThread 6382 AutoMutex lock(mLock); 6383 if (mActiveTracks.indexOf(recordTrack) >= 0) { 6384 if (recordTrack->mState == TrackBase::PAUSING) { 6385 ALOGV("active record track PAUSING -> ACTIVE"); 6386 recordTrack->mState = TrackBase::ACTIVE; 6387 } else { 6388 ALOGV("active record track state %d", recordTrack->mState); 6389 } 6390 return status; 6391 } 6392 6393 // TODO consider other ways of handling this, such as changing the state to :STARTING and 6394 // adding the track to mActiveTracks after returning from AudioSystem::startInput(), 6395 // or using a separate command thread 6396 recordTrack->mState = TrackBase::STARTING_1; 6397 mActiveTracks.add(recordTrack); 6398 mActiveTracksGen++; 6399 status_t status = NO_ERROR; 6400 if (recordTrack->isExternalTrack()) { 6401 mLock.unlock(); 6402 status = AudioSystem::startInput(mId, (audio_session_t)recordTrack->sessionId()); 6403 mLock.lock(); 6404 // FIXME should verify that recordTrack is still in mActiveTracks 6405 if (status != NO_ERROR) { 6406 mActiveTracks.remove(recordTrack); 6407 mActiveTracksGen++; 6408 recordTrack->clearSyncStartEvent(); 6409 ALOGV("RecordThread::start error %d", status); 6410 return status; 6411 } 6412 } 6413 // Catch up with current buffer indices if thread is already running. 6414 // This is what makes a new client discard all buffered data. If the track's mRsmpInFront 6415 // was initialized to some value closer to the thread's mRsmpInFront, then the track could 6416 // see previously buffered data before it called start(), but with greater risk of overrun. 6417 6418 recordTrack->mResamplerBufferProvider->reset(); 6419 // clear any converter state as new data will be discontinuous 6420 recordTrack->mRecordBufferConverter->reset(); 6421 recordTrack->mState = TrackBase::STARTING_2; 6422 // signal thread to start 6423 mWaitWorkCV.broadcast(); 6424 if (mActiveTracks.indexOf(recordTrack) < 0) { 6425 ALOGV("Record failed to start"); 6426 status = BAD_VALUE; 6427 goto startError; 6428 } 6429 return status; 6430 } 6431 6432startError: 6433 if (recordTrack->isExternalTrack()) { 6434 AudioSystem::stopInput(mId, (audio_session_t)recordTrack->sessionId()); 6435 } 6436 recordTrack->clearSyncStartEvent(); 6437 // FIXME I wonder why we do not reset the state here? 6438 return status; 6439} 6440 6441void AudioFlinger::RecordThread::syncStartEventCallback(const wp<SyncEvent>& event) 6442{ 6443 sp<SyncEvent> strongEvent = event.promote(); 6444 6445 if (strongEvent != 0) { 6446 sp<RefBase> ptr = strongEvent->cookie().promote(); 6447 if (ptr != 0) { 6448 RecordTrack *recordTrack = (RecordTrack *)ptr.get(); 6449 recordTrack->handleSyncStartEvent(strongEvent); 6450 } 6451 } 6452} 6453 6454bool AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 6455 ALOGV("RecordThread::stop"); 6456 AutoMutex _l(mLock); 6457 if (mActiveTracks.indexOf(recordTrack) != 0 || recordTrack->mState == TrackBase::PAUSING) { 6458 return false; 6459 } 6460 // note that threadLoop may still be processing the track at this point [without lock] 6461 recordTrack->mState = TrackBase::PAUSING; 6462 // do not wait for mStartStopCond if exiting 6463 if (exitPending()) { 6464 return true; 6465 } 6466 // FIXME incorrect usage of wait: no explicit predicate or loop 6467 mStartStopCond.wait(mLock); 6468 // if we have been restarted, recordTrack is in mActiveTracks here 6469 if (exitPending() || mActiveTracks.indexOf(recordTrack) != 0) { 6470 ALOGV("Record stopped OK"); 6471 return true; 6472 } 6473 return false; 6474} 6475 6476bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event __unused) const 6477{ 6478 return false; 6479} 6480 6481status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event __unused) 6482{ 6483#if 0 // This branch is currently dead code, but is preserved in case it will be needed in future 6484 if (!isValidSyncEvent(event)) { 6485 return BAD_VALUE; 6486 } 6487 6488 int eventSession = event->triggerSession(); 6489 status_t ret = NAME_NOT_FOUND; 6490 6491 Mutex::Autolock _l(mLock); 6492 6493 for (size_t i = 0; i < mTracks.size(); i++) { 6494 sp<RecordTrack> track = mTracks[i]; 6495 if (eventSession == track->sessionId()) { 6496 (void) track->setSyncEvent(event); 6497 ret = NO_ERROR; 6498 } 6499 } 6500 return ret; 6501#else 6502 return BAD_VALUE; 6503#endif 6504} 6505 6506// destroyTrack_l() must be called with ThreadBase::mLock held 6507void AudioFlinger::RecordThread::destroyTrack_l(const sp<RecordTrack>& track) 6508{ 6509 track->terminate(); 6510 track->mState = TrackBase::STOPPED; 6511 // active tracks are removed by threadLoop() 6512 if (mActiveTracks.indexOf(track) < 0) { 6513 removeTrack_l(track); 6514 } 6515} 6516 6517void AudioFlinger::RecordThread::removeTrack_l(const sp<RecordTrack>& track) 6518{ 6519 mTracks.remove(track); 6520 // need anything related to effects here? 6521 if (track->isFastTrack()) { 6522 ALOG_ASSERT(!mFastTrackAvail); 6523 mFastTrackAvail = true; 6524 } 6525} 6526 6527void AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 6528{ 6529 dumpInternals(fd, args); 6530 dumpTracks(fd, args); 6531 dumpEffectChains(fd, args); 6532} 6533 6534void AudioFlinger::RecordThread::dumpInternals(int fd, const Vector<String16>& args) 6535{ 6536 dprintf(fd, "\nInput thread %p:\n", this); 6537 6538 dumpBase(fd, args); 6539 6540 if (mActiveTracks.size() == 0) { 6541 dprintf(fd, " No active record clients\n"); 6542 } 6543 dprintf(fd, " Fast capture thread: %s\n", hasFastCapture() ? "yes" : "no"); 6544 dprintf(fd, " Fast track available: %s\n", mFastTrackAvail ? "yes" : "no"); 6545 6546 // Make a non-atomic copy of fast capture dump state so it won't change underneath us 6547 // while we are dumping it. It may be inconsistent, but it won't mutate! 6548 // This is a large object so we place it on the heap. 6549 // FIXME 25972958: Need an intelligent copy constructor that does not touch unused pages. 6550 const FastCaptureDumpState *copy = new FastCaptureDumpState(mFastCaptureDumpState); 6551 copy->dump(fd); 6552 delete copy; 6553} 6554 6555void AudioFlinger::RecordThread::dumpTracks(int fd, const Vector<String16>& args __unused) 6556{ 6557 const size_t SIZE = 256; 6558 char buffer[SIZE]; 6559 String8 result; 6560 6561 size_t numtracks = mTracks.size(); 6562 size_t numactive = mActiveTracks.size(); 6563 size_t numactiveseen = 0; 6564 dprintf(fd, " %d Tracks", numtracks); 6565 if (numtracks) { 6566 dprintf(fd, " of which %d are active\n", numactive); 6567 RecordTrack::appendDumpHeader(result); 6568 for (size_t i = 0; i < numtracks ; ++i) { 6569 sp<RecordTrack> track = mTracks[i]; 6570 if (track != 0) { 6571 bool active = mActiveTracks.indexOf(track) >= 0; 6572 if (active) { 6573 numactiveseen++; 6574 } 6575 track->dump(buffer, SIZE, active); 6576 result.append(buffer); 6577 } 6578 } 6579 } else { 6580 dprintf(fd, "\n"); 6581 } 6582 6583 if (numactiveseen != numactive) { 6584 snprintf(buffer, SIZE, " The following tracks are in the active list but" 6585 " not in the track list\n"); 6586 result.append(buffer); 6587 RecordTrack::appendDumpHeader(result); 6588 for (size_t i = 0; i < numactive; ++i) { 6589 sp<RecordTrack> track = mActiveTracks[i]; 6590 if (mTracks.indexOf(track) < 0) { 6591 track->dump(buffer, SIZE, true); 6592 result.append(buffer); 6593 } 6594 } 6595 6596 } 6597 write(fd, result.string(), result.size()); 6598} 6599 6600 6601void AudioFlinger::RecordThread::ResamplerBufferProvider::reset() 6602{ 6603 sp<ThreadBase> threadBase = mRecordTrack->mThread.promote(); 6604 RecordThread *recordThread = (RecordThread *) threadBase.get(); 6605 mRsmpInFront = recordThread->mRsmpInRear; 6606 mRsmpInUnrel = 0; 6607} 6608 6609void AudioFlinger::RecordThread::ResamplerBufferProvider::sync( 6610 size_t *framesAvailable, bool *hasOverrun) 6611{ 6612 sp<ThreadBase> threadBase = mRecordTrack->mThread.promote(); 6613 RecordThread *recordThread = (RecordThread *) threadBase.get(); 6614 const int32_t rear = recordThread->mRsmpInRear; 6615 const int32_t front = mRsmpInFront; 6616 const ssize_t filled = rear - front; 6617 6618 size_t framesIn; 6619 bool overrun = false; 6620 if (filled < 0) { 6621 // should not happen, but treat like a massive overrun and re-sync 6622 framesIn = 0; 6623 mRsmpInFront = rear; 6624 overrun = true; 6625 } else if ((size_t) filled <= recordThread->mRsmpInFrames) { 6626 framesIn = (size_t) filled; 6627 } else { 6628 // client is not keeping up with server, but give it latest data 6629 framesIn = recordThread->mRsmpInFrames; 6630 mRsmpInFront = /* front = */ rear - framesIn; 6631 overrun = true; 6632 } 6633 if (framesAvailable != NULL) { 6634 *framesAvailable = framesIn; 6635 } 6636 if (hasOverrun != NULL) { 6637 *hasOverrun = overrun; 6638 } 6639} 6640 6641// AudioBufferProvider interface 6642status_t AudioFlinger::RecordThread::ResamplerBufferProvider::getNextBuffer( 6643 AudioBufferProvider::Buffer* buffer) 6644{ 6645 sp<ThreadBase> threadBase = mRecordTrack->mThread.promote(); 6646 if (threadBase == 0) { 6647 buffer->frameCount = 0; 6648 buffer->raw = NULL; 6649 return NOT_ENOUGH_DATA; 6650 } 6651 RecordThread *recordThread = (RecordThread *) threadBase.get(); 6652 int32_t rear = recordThread->mRsmpInRear; 6653 int32_t front = mRsmpInFront; 6654 ssize_t filled = rear - front; 6655 // FIXME should not be P2 (don't want to increase latency) 6656 // FIXME if client not keeping up, discard 6657 LOG_ALWAYS_FATAL_IF(!(0 <= filled && (size_t) filled <= recordThread->mRsmpInFrames)); 6658 // 'filled' may be non-contiguous, so return only the first contiguous chunk 6659 front &= recordThread->mRsmpInFramesP2 - 1; 6660 size_t part1 = recordThread->mRsmpInFramesP2 - front; 6661 if (part1 > (size_t) filled) { 6662 part1 = filled; 6663 } 6664 size_t ask = buffer->frameCount; 6665 ALOG_ASSERT(ask > 0); 6666 if (part1 > ask) { 6667 part1 = ask; 6668 } 6669 if (part1 == 0) { 6670 // out of data is fine since the resampler will return a short-count. 6671 buffer->raw = NULL; 6672 buffer->frameCount = 0; 6673 mRsmpInUnrel = 0; 6674 return NOT_ENOUGH_DATA; 6675 } 6676 6677 buffer->raw = (uint8_t*)recordThread->mRsmpInBuffer + front * recordThread->mFrameSize; 6678 buffer->frameCount = part1; 6679 mRsmpInUnrel = part1; 6680 return NO_ERROR; 6681} 6682 6683// AudioBufferProvider interface 6684void AudioFlinger::RecordThread::ResamplerBufferProvider::releaseBuffer( 6685 AudioBufferProvider::Buffer* buffer) 6686{ 6687 size_t stepCount = buffer->frameCount; 6688 if (stepCount == 0) { 6689 return; 6690 } 6691 ALOG_ASSERT(stepCount <= mRsmpInUnrel); 6692 mRsmpInUnrel -= stepCount; 6693 mRsmpInFront += stepCount; 6694 buffer->raw = NULL; 6695 buffer->frameCount = 0; 6696} 6697 6698AudioFlinger::RecordThread::RecordBufferConverter::RecordBufferConverter( 6699 audio_channel_mask_t srcChannelMask, audio_format_t srcFormat, 6700 uint32_t srcSampleRate, 6701 audio_channel_mask_t dstChannelMask, audio_format_t dstFormat, 6702 uint32_t dstSampleRate) : 6703 mSrcChannelMask(AUDIO_CHANNEL_INVALID), // updateParameters will set following vars 6704 // mSrcFormat 6705 // mSrcSampleRate 6706 // mDstChannelMask 6707 // mDstFormat 6708 // mDstSampleRate 6709 // mSrcChannelCount 6710 // mDstChannelCount 6711 // mDstFrameSize 6712 mBuf(NULL), mBufFrames(0), mBufFrameSize(0), 6713 mResampler(NULL), 6714 mIsLegacyDownmix(false), 6715 mIsLegacyUpmix(false), 6716 mRequiresFloat(false), 6717 mInputConverterProvider(NULL) 6718{ 6719 (void)updateParameters(srcChannelMask, srcFormat, srcSampleRate, 6720 dstChannelMask, dstFormat, dstSampleRate); 6721} 6722 6723AudioFlinger::RecordThread::RecordBufferConverter::~RecordBufferConverter() { 6724 free(mBuf); 6725 delete mResampler; 6726 delete mInputConverterProvider; 6727} 6728 6729size_t AudioFlinger::RecordThread::RecordBufferConverter::convert(void *dst, 6730 AudioBufferProvider *provider, size_t frames) 6731{ 6732 if (mInputConverterProvider != NULL) { 6733 mInputConverterProvider->setBufferProvider(provider); 6734 provider = mInputConverterProvider; 6735 } 6736 6737 if (mResampler == NULL) { 6738 ALOGVV("NO RESAMPLING sampleRate:%u mSrcFormat:%#x mDstFormat:%#x", 6739 mSrcSampleRate, mSrcFormat, mDstFormat); 6740 6741 AudioBufferProvider::Buffer buffer; 6742 for (size_t i = frames; i > 0; ) { 6743 buffer.frameCount = i; 6744 status_t status = provider->getNextBuffer(&buffer); 6745 if (status != OK || buffer.frameCount == 0) { 6746 frames -= i; // cannot fill request. 6747 break; 6748 } 6749 // format convert to destination buffer 6750 convertNoResampler(dst, buffer.raw, buffer.frameCount); 6751 6752 dst = (int8_t*)dst + buffer.frameCount * mDstFrameSize; 6753 i -= buffer.frameCount; 6754 provider->releaseBuffer(&buffer); 6755 } 6756 } else { 6757 ALOGVV("RESAMPLING mSrcSampleRate:%u mDstSampleRate:%u mSrcFormat:%#x mDstFormat:%#x", 6758 mSrcSampleRate, mDstSampleRate, mSrcFormat, mDstFormat); 6759 6760 // reallocate buffer if needed 6761 if (mBufFrameSize != 0 && mBufFrames < frames) { 6762 free(mBuf); 6763 mBufFrames = frames; 6764 (void)posix_memalign(&mBuf, 32, mBufFrames * mBufFrameSize); 6765 } 6766 // resampler accumulates, but we only have one source track 6767 memset(mBuf, 0, frames * mBufFrameSize); 6768 frames = mResampler->resample((int32_t*)mBuf, frames, provider); 6769 // format convert to destination buffer 6770 convertResampler(dst, mBuf, frames); 6771 } 6772 return frames; 6773} 6774 6775status_t AudioFlinger::RecordThread::RecordBufferConverter::updateParameters( 6776 audio_channel_mask_t srcChannelMask, audio_format_t srcFormat, 6777 uint32_t srcSampleRate, 6778 audio_channel_mask_t dstChannelMask, audio_format_t dstFormat, 6779 uint32_t dstSampleRate) 6780{ 6781 // quick evaluation if there is any change. 6782 if (mSrcFormat == srcFormat 6783 && mSrcChannelMask == srcChannelMask 6784 && mSrcSampleRate == srcSampleRate 6785 && mDstFormat == dstFormat 6786 && mDstChannelMask == dstChannelMask 6787 && mDstSampleRate == dstSampleRate) { 6788 return NO_ERROR; 6789 } 6790 6791 ALOGV("RecordBufferConverter updateParameters srcMask:%#x dstMask:%#x" 6792 " srcFormat:%#x dstFormat:%#x srcRate:%u dstRate:%u", 6793 srcChannelMask, dstChannelMask, srcFormat, dstFormat, srcSampleRate, dstSampleRate); 6794 const bool valid = 6795 audio_is_input_channel(srcChannelMask) 6796 && audio_is_input_channel(dstChannelMask) 6797 && audio_is_valid_format(srcFormat) && audio_is_linear_pcm(srcFormat) 6798 && audio_is_valid_format(dstFormat) && audio_is_linear_pcm(dstFormat) 6799 && (srcSampleRate <= dstSampleRate * AUDIO_RESAMPLER_DOWN_RATIO_MAX) 6800 ; // no upsampling checks for now 6801 if (!valid) { 6802 return BAD_VALUE; 6803 } 6804 6805 mSrcFormat = srcFormat; 6806 mSrcChannelMask = srcChannelMask; 6807 mSrcSampleRate = srcSampleRate; 6808 mDstFormat = dstFormat; 6809 mDstChannelMask = dstChannelMask; 6810 mDstSampleRate = dstSampleRate; 6811 6812 // compute derived parameters 6813 mSrcChannelCount = audio_channel_count_from_in_mask(srcChannelMask); 6814 mDstChannelCount = audio_channel_count_from_in_mask(dstChannelMask); 6815 mDstFrameSize = mDstChannelCount * audio_bytes_per_sample(mDstFormat); 6816 6817 // do we need to resample? 6818 delete mResampler; 6819 mResampler = NULL; 6820 if (mSrcSampleRate != mDstSampleRate) { 6821 mResampler = AudioResampler::create(AUDIO_FORMAT_PCM_FLOAT, 6822 mSrcChannelCount, mDstSampleRate); 6823 mResampler->setSampleRate(mSrcSampleRate); 6824 mResampler->setVolume(AudioMixer::UNITY_GAIN_FLOAT, AudioMixer::UNITY_GAIN_FLOAT); 6825 } 6826 6827 // are we running legacy channel conversion modes? 6828 mIsLegacyDownmix = (mSrcChannelMask == AUDIO_CHANNEL_IN_STEREO 6829 || mSrcChannelMask == AUDIO_CHANNEL_IN_FRONT_BACK) 6830 && mDstChannelMask == AUDIO_CHANNEL_IN_MONO; 6831 mIsLegacyUpmix = mSrcChannelMask == AUDIO_CHANNEL_IN_MONO 6832 && (mDstChannelMask == AUDIO_CHANNEL_IN_STEREO 6833 || mDstChannelMask == AUDIO_CHANNEL_IN_FRONT_BACK); 6834 6835 // do we need to process in float? 6836 mRequiresFloat = mResampler != NULL || mIsLegacyDownmix || mIsLegacyUpmix; 6837 6838 // do we need a staging buffer to convert for destination (we can still optimize this)? 6839 // we use mBufFrameSize > 0 to indicate both frame size as well as buffer necessity 6840 if (mResampler != NULL) { 6841 mBufFrameSize = max(mSrcChannelCount, FCC_2) 6842 * audio_bytes_per_sample(AUDIO_FORMAT_PCM_FLOAT); 6843 } else if (mIsLegacyUpmix || mIsLegacyDownmix) { // legacy modes always float 6844 mBufFrameSize = mDstChannelCount * audio_bytes_per_sample(AUDIO_FORMAT_PCM_FLOAT); 6845 } else if (mSrcChannelMask != mDstChannelMask && mDstFormat != mSrcFormat) { 6846 mBufFrameSize = mDstChannelCount * audio_bytes_per_sample(mSrcFormat); 6847 } else { 6848 mBufFrameSize = 0; 6849 } 6850 mBufFrames = 0; // force the buffer to be resized. 6851 6852 // do we need an input converter buffer provider to give us float? 6853 delete mInputConverterProvider; 6854 mInputConverterProvider = NULL; 6855 if (mRequiresFloat && mSrcFormat != AUDIO_FORMAT_PCM_FLOAT) { 6856 mInputConverterProvider = new ReformatBufferProvider( 6857 audio_channel_count_from_in_mask(mSrcChannelMask), 6858 mSrcFormat, 6859 AUDIO_FORMAT_PCM_FLOAT, 6860 256 /* provider buffer frame count */); 6861 } 6862 6863 // do we need a remixer to do channel mask conversion 6864 if (!mIsLegacyDownmix && !mIsLegacyUpmix && mSrcChannelMask != mDstChannelMask) { 6865 (void) memcpy_by_index_array_initialization_from_channel_mask( 6866 mIdxAry, ARRAY_SIZE(mIdxAry), mDstChannelMask, mSrcChannelMask); 6867 } 6868 return NO_ERROR; 6869} 6870 6871void AudioFlinger::RecordThread::RecordBufferConverter::convertNoResampler( 6872 void *dst, const void *src, size_t frames) 6873{ 6874 // src is native type unless there is legacy upmix or downmix, whereupon it is float. 6875 if (mBufFrameSize != 0 && mBufFrames < frames) { 6876 free(mBuf); 6877 mBufFrames = frames; 6878 (void)posix_memalign(&mBuf, 32, mBufFrames * mBufFrameSize); 6879 } 6880 // do we need to do legacy upmix and downmix? 6881 if (mIsLegacyUpmix || mIsLegacyDownmix) { 6882 void *dstBuf = mBuf != NULL ? mBuf : dst; 6883 if (mIsLegacyUpmix) { 6884 upmix_to_stereo_float_from_mono_float((float *)dstBuf, 6885 (const float *)src, frames); 6886 } else /*mIsLegacyDownmix */ { 6887 downmix_to_mono_float_from_stereo_float((float *)dstBuf, 6888 (const float *)src, frames); 6889 } 6890 if (mBuf != NULL) { 6891 memcpy_by_audio_format(dst, mDstFormat, mBuf, AUDIO_FORMAT_PCM_FLOAT, 6892 frames * mDstChannelCount); 6893 } 6894 return; 6895 } 6896 // do we need to do channel mask conversion? 6897 if (mSrcChannelMask != mDstChannelMask) { 6898 void *dstBuf = mBuf != NULL ? mBuf : dst; 6899 memcpy_by_index_array(dstBuf, mDstChannelCount, 6900 src, mSrcChannelCount, mIdxAry, audio_bytes_per_sample(mSrcFormat), frames); 6901 if (dstBuf == dst) { 6902 return; // format is the same 6903 } 6904 } 6905 // convert to destination buffer 6906 const void *convertBuf = mBuf != NULL ? mBuf : src; 6907 memcpy_by_audio_format(dst, mDstFormat, convertBuf, mSrcFormat, 6908 frames * mDstChannelCount); 6909} 6910 6911void AudioFlinger::RecordThread::RecordBufferConverter::convertResampler( 6912 void *dst, /*not-a-const*/ void *src, size_t frames) 6913{ 6914 // src buffer format is ALWAYS float when entering this routine 6915 if (mIsLegacyUpmix) { 6916 ; // mono to stereo already handled by resampler 6917 } else if (mIsLegacyDownmix 6918 || (mSrcChannelMask == mDstChannelMask && mSrcChannelCount == 1)) { 6919 // the resampler outputs stereo for mono input channel (a feature?) 6920 // must convert to mono 6921 downmix_to_mono_float_from_stereo_float((float *)src, 6922 (const float *)src, frames); 6923 } else if (mSrcChannelMask != mDstChannelMask) { 6924 // convert to mono channel again for channel mask conversion (could be skipped 6925 // with further optimization). 6926 if (mSrcChannelCount == 1) { 6927 downmix_to_mono_float_from_stereo_float((float *)src, 6928 (const float *)src, frames); 6929 } 6930 // convert to destination format (in place, OK as float is larger than other types) 6931 if (mDstFormat != AUDIO_FORMAT_PCM_FLOAT) { 6932 memcpy_by_audio_format(src, mDstFormat, src, AUDIO_FORMAT_PCM_FLOAT, 6933 frames * mSrcChannelCount); 6934 } 6935 // channel convert and save to dst 6936 memcpy_by_index_array(dst, mDstChannelCount, 6937 src, mSrcChannelCount, mIdxAry, audio_bytes_per_sample(mDstFormat), frames); 6938 return; 6939 } 6940 // convert to destination format and save to dst 6941 memcpy_by_audio_format(dst, mDstFormat, src, AUDIO_FORMAT_PCM_FLOAT, 6942 frames * mDstChannelCount); 6943} 6944 6945bool AudioFlinger::RecordThread::checkForNewParameter_l(const String8& keyValuePair, 6946 status_t& status) 6947{ 6948 bool reconfig = false; 6949 6950 status = NO_ERROR; 6951 6952 audio_format_t reqFormat = mFormat; 6953 uint32_t samplingRate = mSampleRate; 6954 // TODO this may change if we want to support capture from HDMI PCM multi channel (e.g on TVs). 6955 audio_channel_mask_t channelMask = audio_channel_in_mask_from_count(mChannelCount); 6956 6957 AudioParameter param = AudioParameter(keyValuePair); 6958 int value; 6959 // TODO Investigate when this code runs. Check with audio policy when a sample rate and 6960 // channel count change can be requested. Do we mandate the first client defines the 6961 // HAL sampling rate and channel count or do we allow changes on the fly? 6962 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 6963 samplingRate = value; 6964 reconfig = true; 6965 } 6966 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 6967 if (!audio_is_linear_pcm((audio_format_t) value)) { 6968 status = BAD_VALUE; 6969 } else { 6970 reqFormat = (audio_format_t) value; 6971 reconfig = true; 6972 } 6973 } 6974 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 6975 audio_channel_mask_t mask = (audio_channel_mask_t) value; 6976 if (!audio_is_input_channel(mask) || 6977 audio_channel_count_from_in_mask(mask) > FCC_8) { 6978 status = BAD_VALUE; 6979 } else { 6980 channelMask = mask; 6981 reconfig = true; 6982 } 6983 } 6984 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 6985 // do not accept frame count changes if tracks are open as the track buffer 6986 // size depends on frame count and correct behavior would not be guaranteed 6987 // if frame count is changed after track creation 6988 if (mActiveTracks.size() > 0) { 6989 status = INVALID_OPERATION; 6990 } else { 6991 reconfig = true; 6992 } 6993 } 6994 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 6995 // forward device change to effects that have requested to be 6996 // aware of attached audio device. 6997 for (size_t i = 0; i < mEffectChains.size(); i++) { 6998 mEffectChains[i]->setDevice_l(value); 6999 } 7000 7001 // store input device and output device but do not forward output device to audio HAL. 7002 // Note that status is ignored by the caller for output device 7003 // (see AudioFlinger::setParameters() 7004 if (audio_is_output_devices(value)) { 7005 mOutDevice = value; 7006 status = BAD_VALUE; 7007 } else { 7008 mInDevice = value; 7009 if (value != AUDIO_DEVICE_NONE) { 7010 mPrevInDevice = value; 7011 } 7012 // disable AEC and NS if the device is a BT SCO headset supporting those 7013 // pre processings 7014 if (mTracks.size() > 0) { 7015 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 7016 mAudioFlinger->btNrecIsOff(); 7017 for (size_t i = 0; i < mTracks.size(); i++) { 7018 sp<RecordTrack> track = mTracks[i]; 7019 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId()); 7020 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId()); 7021 } 7022 } 7023 } 7024 } 7025 if (param.getInt(String8(AudioParameter::keyInputSource), value) == NO_ERROR && 7026 mAudioSource != (audio_source_t)value) { 7027 // forward device change to effects that have requested to be 7028 // aware of attached audio device. 7029 for (size_t i = 0; i < mEffectChains.size(); i++) { 7030 mEffectChains[i]->setAudioSource_l((audio_source_t)value); 7031 } 7032 mAudioSource = (audio_source_t)value; 7033 } 7034 7035 if (status == NO_ERROR) { 7036 status = mInput->stream->common.set_parameters(&mInput->stream->common, 7037 keyValuePair.string()); 7038 if (status == INVALID_OPERATION) { 7039 inputStandBy(); 7040 status = mInput->stream->common.set_parameters(&mInput->stream->common, 7041 keyValuePair.string()); 7042 } 7043 if (reconfig) { 7044 if (status == BAD_VALUE && 7045 audio_is_linear_pcm(mInput->stream->common.get_format(&mInput->stream->common)) && 7046 audio_is_linear_pcm(reqFormat) && 7047 (mInput->stream->common.get_sample_rate(&mInput->stream->common) 7048 <= (AUDIO_RESAMPLER_DOWN_RATIO_MAX * samplingRate)) && 7049 audio_channel_count_from_in_mask( 7050 mInput->stream->common.get_channels(&mInput->stream->common)) <= FCC_8) { 7051 status = NO_ERROR; 7052 } 7053 if (status == NO_ERROR) { 7054 readInputParameters_l(); 7055 sendIoConfigEvent_l(AUDIO_INPUT_CONFIG_CHANGED); 7056 } 7057 } 7058 } 7059 7060 return reconfig; 7061} 7062 7063String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 7064{ 7065 Mutex::Autolock _l(mLock); 7066 if (initCheck() != NO_ERROR) { 7067 return String8(); 7068 } 7069 7070 char *s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 7071 const String8 out_s8(s); 7072 free(s); 7073 return out_s8; 7074} 7075 7076void AudioFlinger::RecordThread::ioConfigChanged(audio_io_config_event event, pid_t pid) { 7077 sp<AudioIoDescriptor> desc = new AudioIoDescriptor(); 7078 7079 desc->mIoHandle = mId; 7080 7081 switch (event) { 7082 case AUDIO_INPUT_OPENED: 7083 case AUDIO_INPUT_CONFIG_CHANGED: 7084 desc->mPatch = mPatch; 7085 desc->mChannelMask = mChannelMask; 7086 desc->mSamplingRate = mSampleRate; 7087 desc->mFormat = mFormat; 7088 desc->mFrameCount = mFrameCount; 7089 desc->mLatency = 0; 7090 break; 7091 7092 case AUDIO_INPUT_CLOSED: 7093 default: 7094 break; 7095 } 7096 mAudioFlinger->ioConfigChanged(event, desc, pid); 7097} 7098 7099void AudioFlinger::RecordThread::readInputParameters_l() 7100{ 7101 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 7102 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 7103 mChannelCount = audio_channel_count_from_in_mask(mChannelMask); 7104 if (mChannelCount > FCC_8) { 7105 ALOGE("HAL channel count %d > %d", mChannelCount, FCC_8); 7106 } 7107 mHALFormat = mInput->stream->common.get_format(&mInput->stream->common); 7108 mFormat = mHALFormat; 7109 if (!audio_is_linear_pcm(mFormat)) { 7110 ALOGE("HAL format %#x is not linear pcm", mFormat); 7111 } 7112 mFrameSize = audio_stream_in_frame_size(mInput->stream); 7113 mBufferSize = mInput->stream->common.get_buffer_size(&mInput->stream->common); 7114 mFrameCount = mBufferSize / mFrameSize; 7115 // This is the formula for calculating the temporary buffer size. 7116 // With 7 HAL buffers, we can guarantee ability to down-sample the input by ratio of 6:1 to 7117 // 1 full output buffer, regardless of the alignment of the available input. 7118 // The value is somewhat arbitrary, and could probably be even larger. 7119 // A larger value should allow more old data to be read after a track calls start(), 7120 // without increasing latency. 7121 // 7122 // Note this is independent of the maximum downsampling ratio permitted for capture. 7123 mRsmpInFrames = mFrameCount * 7; 7124 mRsmpInFramesP2 = roundup(mRsmpInFrames); 7125 free(mRsmpInBuffer); 7126 mRsmpInBuffer = NULL; 7127 7128 // TODO optimize audio capture buffer sizes ... 7129 // Here we calculate the size of the sliding buffer used as a source 7130 // for resampling. mRsmpInFramesP2 is currently roundup(mFrameCount * 7). 7131 // For current HAL frame counts, this is usually 2048 = 40 ms. It would 7132 // be better to have it derived from the pipe depth in the long term. 7133 // The current value is higher than necessary. However it should not add to latency. 7134 7135 // Over-allocate beyond mRsmpInFramesP2 to permit a HAL read past end of buffer 7136 size_t bufferSize = (mRsmpInFramesP2 + mFrameCount - 1) * mFrameSize; 7137 (void)posix_memalign(&mRsmpInBuffer, 32, bufferSize); 7138 memset(mRsmpInBuffer, 0, bufferSize); // if posix_memalign fails, will segv here. 7139 7140 // AudioRecord mSampleRate and mChannelCount are constant due to AudioRecord API constraints. 7141 // But if thread's mSampleRate or mChannelCount changes, how will that affect active tracks? 7142} 7143 7144uint32_t AudioFlinger::RecordThread::getInputFramesLost() 7145{ 7146 Mutex::Autolock _l(mLock); 7147 if (initCheck() != NO_ERROR) { 7148 return 0; 7149 } 7150 7151 return mInput->stream->get_input_frames_lost(mInput->stream); 7152} 7153 7154uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) const 7155{ 7156 Mutex::Autolock _l(mLock); 7157 uint32_t result = 0; 7158 if (getEffectChain_l(sessionId) != 0) { 7159 result = EFFECT_SESSION; 7160 } 7161 7162 for (size_t i = 0; i < mTracks.size(); ++i) { 7163 if (sessionId == mTracks[i]->sessionId()) { 7164 result |= TRACK_SESSION; 7165 break; 7166 } 7167 } 7168 7169 return result; 7170} 7171 7172KeyedVector<int, bool> AudioFlinger::RecordThread::sessionIds() const 7173{ 7174 KeyedVector<int, bool> ids; 7175 Mutex::Autolock _l(mLock); 7176 for (size_t j = 0; j < mTracks.size(); ++j) { 7177 sp<RecordThread::RecordTrack> track = mTracks[j]; 7178 int sessionId = track->sessionId(); 7179 if (ids.indexOfKey(sessionId) < 0) { 7180 ids.add(sessionId, true); 7181 } 7182 } 7183 return ids; 7184} 7185 7186AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 7187{ 7188 Mutex::Autolock _l(mLock); 7189 AudioStreamIn *input = mInput; 7190 mInput = NULL; 7191 return input; 7192} 7193 7194// this method must always be called either with ThreadBase mLock held or inside the thread loop 7195audio_stream_t* AudioFlinger::RecordThread::stream() const 7196{ 7197 if (mInput == NULL) { 7198 return NULL; 7199 } 7200 return &mInput->stream->common; 7201} 7202 7203status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 7204{ 7205 // only one chain per input thread 7206 if (mEffectChains.size() != 0) { 7207 ALOGW("addEffectChain_l() already one chain %p on thread %p", chain.get(), this); 7208 return INVALID_OPERATION; 7209 } 7210 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 7211 chain->setThread(this); 7212 chain->setInBuffer(NULL); 7213 chain->setOutBuffer(NULL); 7214 7215 checkSuspendOnAddEffectChain_l(chain); 7216 7217 // make sure enabled pre processing effects state is communicated to the HAL as we 7218 // just moved them to a new input stream. 7219 chain->syncHalEffectsState(); 7220 7221 mEffectChains.add(chain); 7222 7223 return NO_ERROR; 7224} 7225 7226size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 7227{ 7228 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 7229 ALOGW_IF(mEffectChains.size() != 1, 7230 "removeEffectChain_l() %p invalid chain size %d on thread %p", 7231 chain.get(), mEffectChains.size(), this); 7232 if (mEffectChains.size() == 1) { 7233 mEffectChains.removeAt(0); 7234 } 7235 return 0; 7236} 7237 7238status_t AudioFlinger::RecordThread::createAudioPatch_l(const struct audio_patch *patch, 7239 audio_patch_handle_t *handle) 7240{ 7241 status_t status = NO_ERROR; 7242 7243 // store new device and send to effects 7244 mInDevice = patch->sources[0].ext.device.type; 7245 mPatch = *patch; 7246 for (size_t i = 0; i < mEffectChains.size(); i++) { 7247 mEffectChains[i]->setDevice_l(mInDevice); 7248 } 7249 7250 // disable AEC and NS if the device is a BT SCO headset supporting those 7251 // pre processings 7252 if (mTracks.size() > 0) { 7253 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 7254 mAudioFlinger->btNrecIsOff(); 7255 for (size_t i = 0; i < mTracks.size(); i++) { 7256 sp<RecordTrack> track = mTracks[i]; 7257 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId()); 7258 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId()); 7259 } 7260 } 7261 7262 // store new source and send to effects 7263 if (mAudioSource != patch->sinks[0].ext.mix.usecase.source) { 7264 mAudioSource = patch->sinks[0].ext.mix.usecase.source; 7265 for (size_t i = 0; i < mEffectChains.size(); i++) { 7266 mEffectChains[i]->setAudioSource_l(mAudioSource); 7267 } 7268 } 7269 7270 if (mInput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 7271 audio_hw_device_t *hwDevice = mInput->audioHwDev->hwDevice(); 7272 status = hwDevice->create_audio_patch(hwDevice, 7273 patch->num_sources, 7274 patch->sources, 7275 patch->num_sinks, 7276 patch->sinks, 7277 handle); 7278 } else { 7279 char *address; 7280 if (strcmp(patch->sources[0].ext.device.address, "") != 0) { 7281 address = audio_device_address_to_parameter( 7282 patch->sources[0].ext.device.type, 7283 patch->sources[0].ext.device.address); 7284 } else { 7285 address = (char *)calloc(1, 1); 7286 } 7287 AudioParameter param = AudioParameter(String8(address)); 7288 free(address); 7289 param.addInt(String8(AUDIO_PARAMETER_STREAM_ROUTING), 7290 (int)patch->sources[0].ext.device.type); 7291 param.addInt(String8(AUDIO_PARAMETER_STREAM_INPUT_SOURCE), 7292 (int)patch->sinks[0].ext.mix.usecase.source); 7293 status = mInput->stream->common.set_parameters(&mInput->stream->common, 7294 param.toString().string()); 7295 *handle = AUDIO_PATCH_HANDLE_NONE; 7296 } 7297 7298 if (mInDevice != mPrevInDevice) { 7299 sendIoConfigEvent_l(AUDIO_INPUT_CONFIG_CHANGED); 7300 mPrevInDevice = mInDevice; 7301 } 7302 7303 return status; 7304} 7305 7306status_t AudioFlinger::RecordThread::releaseAudioPatch_l(const audio_patch_handle_t handle) 7307{ 7308 status_t status = NO_ERROR; 7309 7310 mInDevice = AUDIO_DEVICE_NONE; 7311 7312 if (mInput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 7313 audio_hw_device_t *hwDevice = mInput->audioHwDev->hwDevice(); 7314 status = hwDevice->release_audio_patch(hwDevice, handle); 7315 } else { 7316 AudioParameter param; 7317 param.addInt(String8(AUDIO_PARAMETER_STREAM_ROUTING), 0); 7318 status = mInput->stream->common.set_parameters(&mInput->stream->common, 7319 param.toString().string()); 7320 } 7321 return status; 7322} 7323 7324void AudioFlinger::RecordThread::addPatchRecord(const sp<PatchRecord>& record) 7325{ 7326 Mutex::Autolock _l(mLock); 7327 mTracks.add(record); 7328} 7329 7330void AudioFlinger::RecordThread::deletePatchRecord(const sp<PatchRecord>& record) 7331{ 7332 Mutex::Autolock _l(mLock); 7333 destroyTrack_l(record); 7334} 7335 7336void AudioFlinger::RecordThread::getAudioPortConfig(struct audio_port_config *config) 7337{ 7338 ThreadBase::getAudioPortConfig(config); 7339 config->role = AUDIO_PORT_ROLE_SINK; 7340 config->ext.mix.hw_module = mInput->audioHwDev->handle(); 7341 config->ext.mix.usecase.source = mAudioSource; 7342} 7343 7344} // namespace android 7345