Threads.cpp revision e10393e72454bfd8298017dc193faf424f4e9a8f
1/* 2** 3** Copyright 2012, The Android Open Source Project 4** 5** Licensed under the Apache License, Version 2.0 (the "License"); 6** you may not use this file except in compliance with the License. 7** You may obtain a copy of the License at 8** 9** http://www.apache.org/licenses/LICENSE-2.0 10** 11** Unless required by applicable law or agreed to in writing, software 12** distributed under the License is distributed on an "AS IS" BASIS, 13** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. 14** See the License for the specific language governing permissions and 15** limitations under the License. 16*/ 17 18 19#define LOG_TAG "AudioFlinger" 20//#define LOG_NDEBUG 0 21#define ATRACE_TAG ATRACE_TAG_AUDIO 22 23#include "Configuration.h" 24#include <math.h> 25#include <fcntl.h> 26#include <linux/futex.h> 27#include <sys/stat.h> 28#include <sys/syscall.h> 29#include <cutils/properties.h> 30#include <media/AudioParameter.h> 31#include <media/AudioResamplerPublic.h> 32#include <utils/Log.h> 33#include <utils/Trace.h> 34 35#include <private/media/AudioTrackShared.h> 36#include <hardware/audio.h> 37#include <audio_effects/effect_ns.h> 38#include <audio_effects/effect_aec.h> 39#include <audio_utils/conversion.h> 40#include <audio_utils/primitives.h> 41#include <audio_utils/format.h> 42#include <audio_utils/minifloat.h> 43 44// NBAIO implementations 45#include <media/nbaio/AudioStreamInSource.h> 46#include <media/nbaio/AudioStreamOutSink.h> 47#include <media/nbaio/MonoPipe.h> 48#include <media/nbaio/MonoPipeReader.h> 49#include <media/nbaio/Pipe.h> 50#include <media/nbaio/PipeReader.h> 51#include <media/nbaio/SourceAudioBufferProvider.h> 52#include <mediautils/BatteryNotifier.h> 53 54#include <powermanager/PowerManager.h> 55 56#include "AudioFlinger.h" 57#include "AudioMixer.h" 58#include "BufferProviders.h" 59#include "FastMixer.h" 60#include "FastCapture.h" 61#include "ServiceUtilities.h" 62#include "mediautils/SchedulingPolicyService.h" 63 64#ifdef ADD_BATTERY_DATA 65#include <media/IMediaPlayerService.h> 66#include <media/IMediaDeathNotifier.h> 67#endif 68 69#ifdef DEBUG_CPU_USAGE 70#include <cpustats/CentralTendencyStatistics.h> 71#include <cpustats/ThreadCpuUsage.h> 72#endif 73 74// ---------------------------------------------------------------------------- 75 76// Note: the following macro is used for extremely verbose logging message. In 77// order to run with ALOG_ASSERT turned on, we need to have LOG_NDEBUG set to 78// 0; but one side effect of this is to turn all LOGV's as well. Some messages 79// are so verbose that we want to suppress them even when we have ALOG_ASSERT 80// turned on. Do not uncomment the #def below unless you really know what you 81// are doing and want to see all of the extremely verbose messages. 82//#define VERY_VERY_VERBOSE_LOGGING 83#ifdef VERY_VERY_VERBOSE_LOGGING 84#define ALOGVV ALOGV 85#else 86#define ALOGVV(a...) do { } while(0) 87#endif 88 89// TODO: Move these macro/inlines to a header file. 90#define max(a, b) ((a) > (b) ? (a) : (b)) 91template <typename T> 92static inline T min(const T& a, const T& b) 93{ 94 return a < b ? a : b; 95} 96 97#ifndef ARRAY_SIZE 98#define ARRAY_SIZE(a) (sizeof(a) / sizeof(a[0])) 99#endif 100 101namespace android { 102 103// retry counts for buffer fill timeout 104// 50 * ~20msecs = 1 second 105static const int8_t kMaxTrackRetries = 50; 106static const int8_t kMaxTrackStartupRetries = 50; 107// allow less retry attempts on direct output thread. 108// direct outputs can be a scarce resource in audio hardware and should 109// be released as quickly as possible. 110static const int8_t kMaxTrackRetriesDirect = 2; 111 112// don't warn about blocked writes or record buffer overflows more often than this 113static const nsecs_t kWarningThrottleNs = seconds(5); 114 115// RecordThread loop sleep time upon application overrun or audio HAL read error 116static const int kRecordThreadSleepUs = 5000; 117 118// maximum time to wait in sendConfigEvent_l() for a status to be received 119static const nsecs_t kConfigEventTimeoutNs = seconds(2); 120 121// minimum sleep time for the mixer thread loop when tracks are active but in underrun 122static const uint32_t kMinThreadSleepTimeUs = 5000; 123// maximum divider applied to the active sleep time in the mixer thread loop 124static const uint32_t kMaxThreadSleepTimeShift = 2; 125 126// minimum normal sink buffer size, expressed in milliseconds rather than frames 127// FIXME This should be based on experimentally observed scheduling jitter 128static const uint32_t kMinNormalSinkBufferSizeMs = 20; 129// maximum normal sink buffer size 130static const uint32_t kMaxNormalSinkBufferSizeMs = 24; 131 132// minimum capture buffer size in milliseconds to _not_ need a fast capture thread 133// FIXME This should be based on experimentally observed scheduling jitter 134static const uint32_t kMinNormalCaptureBufferSizeMs = 12; 135 136// Offloaded output thread standby delay: allows track transition without going to standby 137static const nsecs_t kOffloadStandbyDelayNs = seconds(1); 138 139// Whether to use fast mixer 140static const enum { 141 FastMixer_Never, // never initialize or use: for debugging only 142 FastMixer_Always, // always initialize and use, even if not needed: for debugging only 143 // normal mixer multiplier is 1 144 FastMixer_Static, // initialize if needed, then use all the time if initialized, 145 // multiplier is calculated based on min & max normal mixer buffer size 146 FastMixer_Dynamic, // initialize if needed, then use dynamically depending on track load, 147 // multiplier is calculated based on min & max normal mixer buffer size 148 // FIXME for FastMixer_Dynamic: 149 // Supporting this option will require fixing HALs that can't handle large writes. 150 // For example, one HAL implementation returns an error from a large write, 151 // and another HAL implementation corrupts memory, possibly in the sample rate converter. 152 // We could either fix the HAL implementations, or provide a wrapper that breaks 153 // up large writes into smaller ones, and the wrapper would need to deal with scheduler. 154} kUseFastMixer = FastMixer_Static; 155 156// Whether to use fast capture 157static const enum { 158 FastCapture_Never, // never initialize or use: for debugging only 159 FastCapture_Always, // always initialize and use, even if not needed: for debugging only 160 FastCapture_Static, // initialize if needed, then use all the time if initialized 161} kUseFastCapture = FastCapture_Static; 162 163// Priorities for requestPriority 164static const int kPriorityAudioApp = 2; 165static const int kPriorityFastMixer = 3; 166static const int kPriorityFastCapture = 3; 167 168// IAudioFlinger::createTrack() reports back to client the total size of shared memory area 169// for the track. The client then sub-divides this into smaller buffers for its use. 170// Currently the client uses N-buffering by default, but doesn't tell us about the value of N. 171// So for now we just assume that client is double-buffered for fast tracks. 172// FIXME It would be better for client to tell AudioFlinger the value of N, 173// so AudioFlinger could allocate the right amount of memory. 174// See the client's minBufCount and mNotificationFramesAct calculations for details. 175 176// This is the default value, if not specified by property. 177static const int kFastTrackMultiplier = 2; 178 179// The minimum and maximum allowed values 180static const int kFastTrackMultiplierMin = 1; 181static const int kFastTrackMultiplierMax = 2; 182 183// The actual value to use, which can be specified per-device via property af.fast_track_multiplier. 184static int sFastTrackMultiplier = kFastTrackMultiplier; 185 186// See Thread::readOnlyHeap(). 187// Initially this heap is used to allocate client buffers for "fast" AudioRecord. 188// Eventually it will be the single buffer that FastCapture writes into via HAL read(), 189// and that all "fast" AudioRecord clients read from. In either case, the size can be small. 190static const size_t kRecordThreadReadOnlyHeapSize = 0x2000; 191 192// ---------------------------------------------------------------------------- 193 194static pthread_once_t sFastTrackMultiplierOnce = PTHREAD_ONCE_INIT; 195 196static void sFastTrackMultiplierInit() 197{ 198 char value[PROPERTY_VALUE_MAX]; 199 if (property_get("af.fast_track_multiplier", value, NULL) > 0) { 200 char *endptr; 201 unsigned long ul = strtoul(value, &endptr, 0); 202 if (*endptr == '\0' && kFastTrackMultiplierMin <= ul && ul <= kFastTrackMultiplierMax) { 203 sFastTrackMultiplier = (int) ul; 204 } 205 } 206} 207 208// ---------------------------------------------------------------------------- 209 210#ifdef ADD_BATTERY_DATA 211// To collect the amplifier usage 212static void addBatteryData(uint32_t params) { 213 sp<IMediaPlayerService> service = IMediaDeathNotifier::getMediaPlayerService(); 214 if (service == NULL) { 215 // it already logged 216 return; 217 } 218 219 service->addBatteryData(params); 220} 221#endif 222 223// Track the CLOCK_BOOTTIME versus CLOCK_MONOTONIC timebase offset 224struct { 225 // call when you acquire a partial wakelock 226 void acquire(const sp<IBinder> &wakeLockToken) { 227 pthread_mutex_lock(&mLock); 228 if (wakeLockToken.get() == nullptr) { 229 adjustTimebaseOffset(&mBoottimeOffset, ExtendedTimestamp::TIMEBASE_BOOTTIME); 230 } else { 231 if (mCount == 0) { 232 adjustTimebaseOffset(&mBoottimeOffset, ExtendedTimestamp::TIMEBASE_BOOTTIME); 233 } 234 ++mCount; 235 } 236 pthread_mutex_unlock(&mLock); 237 } 238 239 // call when you release a partial wakelock. 240 void release(const sp<IBinder> &wakeLockToken) { 241 if (wakeLockToken.get() == nullptr) { 242 return; 243 } 244 pthread_mutex_lock(&mLock); 245 if (--mCount < 0) { 246 ALOGE("negative wakelock count"); 247 mCount = 0; 248 } 249 pthread_mutex_unlock(&mLock); 250 } 251 252 // retrieves the boottime timebase offset from monotonic. 253 int64_t getBoottimeOffset() { 254 pthread_mutex_lock(&mLock); 255 int64_t boottimeOffset = mBoottimeOffset; 256 pthread_mutex_unlock(&mLock); 257 return boottimeOffset; 258 } 259 260 // Adjusts the timebase offset between TIMEBASE_MONOTONIC 261 // and the selected timebase. 262 // Currently only TIMEBASE_BOOTTIME is allowed. 263 // 264 // This only needs to be called upon acquiring the first partial wakelock 265 // after all other partial wakelocks are released. 266 // 267 // We do an empirical measurement of the offset rather than parsing 268 // /proc/timer_list since the latter is not a formal kernel ABI. 269 static void adjustTimebaseOffset(int64_t *offset, ExtendedTimestamp::Timebase timebase) { 270 int clockbase; 271 switch (timebase) { 272 case ExtendedTimestamp::TIMEBASE_BOOTTIME: 273 clockbase = SYSTEM_TIME_BOOTTIME; 274 break; 275 default: 276 LOG_ALWAYS_FATAL("invalid timebase %d", timebase); 277 break; 278 } 279 // try three times to get the clock offset, choose the one 280 // with the minimum gap in measurements. 281 const int tries = 3; 282 nsecs_t bestGap, measured; 283 for (int i = 0; i < tries; ++i) { 284 const nsecs_t tmono = systemTime(SYSTEM_TIME_MONOTONIC); 285 const nsecs_t tbase = systemTime(clockbase); 286 const nsecs_t tmono2 = systemTime(SYSTEM_TIME_MONOTONIC); 287 const nsecs_t gap = tmono2 - tmono; 288 if (i == 0 || gap < bestGap) { 289 bestGap = gap; 290 measured = tbase - ((tmono + tmono2) >> 1); 291 } 292 } 293 294 // to avoid micro-adjusting, we don't change the timebase 295 // unless it is significantly different. 296 // 297 // Assumption: It probably takes more than toleranceNs to 298 // suspend and resume the device. 299 static int64_t toleranceNs = 10000; // 10 us 300 if (llabs(*offset - measured) > toleranceNs) { 301 ALOGV("Adjusting timebase offset old: %lld new: %lld", 302 (long long)*offset, (long long)measured); 303 *offset = measured; 304 } 305 } 306 307 pthread_mutex_t mLock; 308 int32_t mCount; 309 int64_t mBoottimeOffset; 310} gBoottime = { PTHREAD_MUTEX_INITIALIZER, 0, 0 }; // static, so use POD initialization 311 312// ---------------------------------------------------------------------------- 313// CPU Stats 314// ---------------------------------------------------------------------------- 315 316class CpuStats { 317public: 318 CpuStats(); 319 void sample(const String8 &title); 320#ifdef DEBUG_CPU_USAGE 321private: 322 ThreadCpuUsage mCpuUsage; // instantaneous thread CPU usage in wall clock ns 323 CentralTendencyStatistics mWcStats; // statistics on thread CPU usage in wall clock ns 324 325 CentralTendencyStatistics mHzStats; // statistics on thread CPU usage in cycles 326 327 int mCpuNum; // thread's current CPU number 328 int mCpukHz; // frequency of thread's current CPU in kHz 329#endif 330}; 331 332CpuStats::CpuStats() 333#ifdef DEBUG_CPU_USAGE 334 : mCpuNum(-1), mCpukHz(-1) 335#endif 336{ 337} 338 339void CpuStats::sample(const String8 &title 340#ifndef DEBUG_CPU_USAGE 341 __unused 342#endif 343 ) { 344#ifdef DEBUG_CPU_USAGE 345 // get current thread's delta CPU time in wall clock ns 346 double wcNs; 347 bool valid = mCpuUsage.sampleAndEnable(wcNs); 348 349 // record sample for wall clock statistics 350 if (valid) { 351 mWcStats.sample(wcNs); 352 } 353 354 // get the current CPU number 355 int cpuNum = sched_getcpu(); 356 357 // get the current CPU frequency in kHz 358 int cpukHz = mCpuUsage.getCpukHz(cpuNum); 359 360 // check if either CPU number or frequency changed 361 if (cpuNum != mCpuNum || cpukHz != mCpukHz) { 362 mCpuNum = cpuNum; 363 mCpukHz = cpukHz; 364 // ignore sample for purposes of cycles 365 valid = false; 366 } 367 368 // if no change in CPU number or frequency, then record sample for cycle statistics 369 if (valid && mCpukHz > 0) { 370 double cycles = wcNs * cpukHz * 0.000001; 371 mHzStats.sample(cycles); 372 } 373 374 unsigned n = mWcStats.n(); 375 // mCpuUsage.elapsed() is expensive, so don't call it every loop 376 if ((n & 127) == 1) { 377 long long elapsed = mCpuUsage.elapsed(); 378 if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) { 379 double perLoop = elapsed / (double) n; 380 double perLoop100 = perLoop * 0.01; 381 double perLoop1k = perLoop * 0.001; 382 double mean = mWcStats.mean(); 383 double stddev = mWcStats.stddev(); 384 double minimum = mWcStats.minimum(); 385 double maximum = mWcStats.maximum(); 386 double meanCycles = mHzStats.mean(); 387 double stddevCycles = mHzStats.stddev(); 388 double minCycles = mHzStats.minimum(); 389 double maxCycles = mHzStats.maximum(); 390 mCpuUsage.resetElapsed(); 391 mWcStats.reset(); 392 mHzStats.reset(); 393 ALOGD("CPU usage for %s over past %.1f secs\n" 394 " (%u mixer loops at %.1f mean ms per loop):\n" 395 " us per mix loop: mean=%.0f stddev=%.0f min=%.0f max=%.0f\n" 396 " %% of wall: mean=%.1f stddev=%.1f min=%.1f max=%.1f\n" 397 " MHz: mean=%.1f, stddev=%.1f, min=%.1f max=%.1f", 398 title.string(), 399 elapsed * .000000001, n, perLoop * .000001, 400 mean * .001, 401 stddev * .001, 402 minimum * .001, 403 maximum * .001, 404 mean / perLoop100, 405 stddev / perLoop100, 406 minimum / perLoop100, 407 maximum / perLoop100, 408 meanCycles / perLoop1k, 409 stddevCycles / perLoop1k, 410 minCycles / perLoop1k, 411 maxCycles / perLoop1k); 412 413 } 414 } 415#endif 416}; 417 418// ---------------------------------------------------------------------------- 419// ThreadBase 420// ---------------------------------------------------------------------------- 421 422// static 423const char *AudioFlinger::ThreadBase::threadTypeToString(AudioFlinger::ThreadBase::type_t type) 424{ 425 switch (type) { 426 case MIXER: 427 return "MIXER"; 428 case DIRECT: 429 return "DIRECT"; 430 case DUPLICATING: 431 return "DUPLICATING"; 432 case RECORD: 433 return "RECORD"; 434 case OFFLOAD: 435 return "OFFLOAD"; 436 default: 437 return "unknown"; 438 } 439} 440 441String8 devicesToString(audio_devices_t devices) 442{ 443 static const struct mapping { 444 audio_devices_t mDevices; 445 const char * mString; 446 } mappingsOut[] = { 447 {AUDIO_DEVICE_OUT_EARPIECE, "EARPIECE"}, 448 {AUDIO_DEVICE_OUT_SPEAKER, "SPEAKER"}, 449 {AUDIO_DEVICE_OUT_WIRED_HEADSET, "WIRED_HEADSET"}, 450 {AUDIO_DEVICE_OUT_WIRED_HEADPHONE, "WIRED_HEADPHONE"}, 451 {AUDIO_DEVICE_OUT_BLUETOOTH_SCO, "BLUETOOTH_SCO"}, 452 {AUDIO_DEVICE_OUT_BLUETOOTH_SCO_HEADSET, "BLUETOOTH_SCO_HEADSET"}, 453 {AUDIO_DEVICE_OUT_BLUETOOTH_SCO_CARKIT, "BLUETOOTH_SCO_CARKIT"}, 454 {AUDIO_DEVICE_OUT_BLUETOOTH_A2DP, "BLUETOOTH_A2DP"}, 455 {AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES,"BLUETOOTH_A2DP_HEADPHONES"}, 456 {AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER, "BLUETOOTH_A2DP_SPEAKER"}, 457 {AUDIO_DEVICE_OUT_AUX_DIGITAL, "AUX_DIGITAL"}, 458 {AUDIO_DEVICE_OUT_HDMI, "HDMI"}, 459 {AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET,"ANLG_DOCK_HEADSET"}, 460 {AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET,"DGTL_DOCK_HEADSET"}, 461 {AUDIO_DEVICE_OUT_USB_ACCESSORY, "USB_ACCESSORY"}, 462 {AUDIO_DEVICE_OUT_USB_DEVICE, "USB_DEVICE"}, 463 {AUDIO_DEVICE_OUT_TELEPHONY_TX, "TELEPHONY_TX"}, 464 {AUDIO_DEVICE_OUT_LINE, "LINE"}, 465 {AUDIO_DEVICE_OUT_HDMI_ARC, "HDMI_ARC"}, 466 {AUDIO_DEVICE_OUT_SPDIF, "SPDIF"}, 467 {AUDIO_DEVICE_OUT_FM, "FM"}, 468 {AUDIO_DEVICE_OUT_AUX_LINE, "AUX_LINE"}, 469 {AUDIO_DEVICE_OUT_SPEAKER_SAFE, "SPEAKER_SAFE"}, 470 {AUDIO_DEVICE_OUT_IP, "IP"}, 471 {AUDIO_DEVICE_NONE, "NONE"}, // must be last 472 }, mappingsIn[] = { 473 {AUDIO_DEVICE_IN_COMMUNICATION, "COMMUNICATION"}, 474 {AUDIO_DEVICE_IN_AMBIENT, "AMBIENT"}, 475 {AUDIO_DEVICE_IN_BUILTIN_MIC, "BUILTIN_MIC"}, 476 {AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET, "BLUETOOTH_SCO_HEADSET"}, 477 {AUDIO_DEVICE_IN_WIRED_HEADSET, "WIRED_HEADSET"}, 478 {AUDIO_DEVICE_IN_AUX_DIGITAL, "AUX_DIGITAL"}, 479 {AUDIO_DEVICE_IN_VOICE_CALL, "VOICE_CALL"}, 480 {AUDIO_DEVICE_IN_TELEPHONY_RX, "TELEPHONY_RX"}, 481 {AUDIO_DEVICE_IN_BACK_MIC, "BACK_MIC"}, 482 {AUDIO_DEVICE_IN_REMOTE_SUBMIX, "REMOTE_SUBMIX"}, 483 {AUDIO_DEVICE_IN_ANLG_DOCK_HEADSET, "ANLG_DOCK_HEADSET"}, 484 {AUDIO_DEVICE_IN_DGTL_DOCK_HEADSET, "DGTL_DOCK_HEADSET"}, 485 {AUDIO_DEVICE_IN_USB_ACCESSORY, "USB_ACCESSORY"}, 486 {AUDIO_DEVICE_IN_USB_DEVICE, "USB_DEVICE"}, 487 {AUDIO_DEVICE_IN_FM_TUNER, "FM_TUNER"}, 488 {AUDIO_DEVICE_IN_TV_TUNER, "TV_TUNER"}, 489 {AUDIO_DEVICE_IN_LINE, "LINE"}, 490 {AUDIO_DEVICE_IN_SPDIF, "SPDIF"}, 491 {AUDIO_DEVICE_IN_BLUETOOTH_A2DP, "BLUETOOTH_A2DP"}, 492 {AUDIO_DEVICE_IN_LOOPBACK, "LOOPBACK"}, 493 {AUDIO_DEVICE_IN_IP, "IP"}, 494 {AUDIO_DEVICE_NONE, "NONE"}, // must be last 495 }; 496 String8 result; 497 audio_devices_t allDevices = AUDIO_DEVICE_NONE; 498 const mapping *entry; 499 if (devices & AUDIO_DEVICE_BIT_IN) { 500 devices &= ~AUDIO_DEVICE_BIT_IN; 501 entry = mappingsIn; 502 } else { 503 entry = mappingsOut; 504 } 505 for ( ; entry->mDevices != AUDIO_DEVICE_NONE; entry++) { 506 allDevices = (audio_devices_t) (allDevices | entry->mDevices); 507 if (devices & entry->mDevices) { 508 if (!result.isEmpty()) { 509 result.append("|"); 510 } 511 result.append(entry->mString); 512 } 513 } 514 if (devices & ~allDevices) { 515 if (!result.isEmpty()) { 516 result.append("|"); 517 } 518 result.appendFormat("0x%X", devices & ~allDevices); 519 } 520 if (result.isEmpty()) { 521 result.append(entry->mString); 522 } 523 return result; 524} 525 526String8 inputFlagsToString(audio_input_flags_t flags) 527{ 528 static const struct mapping { 529 audio_input_flags_t mFlag; 530 const char * mString; 531 } mappings[] = { 532 {AUDIO_INPUT_FLAG_FAST, "FAST"}, 533 {AUDIO_INPUT_FLAG_HW_HOTWORD, "HW_HOTWORD"}, 534 {AUDIO_INPUT_FLAG_RAW, "RAW"}, 535 {AUDIO_INPUT_FLAG_SYNC, "SYNC"}, 536 {AUDIO_INPUT_FLAG_NONE, "NONE"}, // must be last 537 }; 538 String8 result; 539 audio_input_flags_t allFlags = AUDIO_INPUT_FLAG_NONE; 540 const mapping *entry; 541 for (entry = mappings; entry->mFlag != AUDIO_INPUT_FLAG_NONE; entry++) { 542 allFlags = (audio_input_flags_t) (allFlags | entry->mFlag); 543 if (flags & entry->mFlag) { 544 if (!result.isEmpty()) { 545 result.append("|"); 546 } 547 result.append(entry->mString); 548 } 549 } 550 if (flags & ~allFlags) { 551 if (!result.isEmpty()) { 552 result.append("|"); 553 } 554 result.appendFormat("0x%X", flags & ~allFlags); 555 } 556 if (result.isEmpty()) { 557 result.append(entry->mString); 558 } 559 return result; 560} 561 562String8 outputFlagsToString(audio_output_flags_t flags) 563{ 564 static const struct mapping { 565 audio_output_flags_t mFlag; 566 const char * mString; 567 } mappings[] = { 568 {AUDIO_OUTPUT_FLAG_DIRECT, "DIRECT"}, 569 {AUDIO_OUTPUT_FLAG_PRIMARY, "PRIMARY"}, 570 {AUDIO_OUTPUT_FLAG_FAST, "FAST"}, 571 {AUDIO_OUTPUT_FLAG_DEEP_BUFFER, "DEEP_BUFFER"}, 572 {AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD,"COMPRESS_OFFLOAD"}, 573 {AUDIO_OUTPUT_FLAG_NON_BLOCKING, "NON_BLOCKING"}, 574 {AUDIO_OUTPUT_FLAG_HW_AV_SYNC, "HW_AV_SYNC"}, 575 {AUDIO_OUTPUT_FLAG_RAW, "RAW"}, 576 {AUDIO_OUTPUT_FLAG_SYNC, "SYNC"}, 577 {AUDIO_OUTPUT_FLAG_IEC958_NONAUDIO, "IEC958_NONAUDIO"}, 578 {AUDIO_OUTPUT_FLAG_NONE, "NONE"}, // must be last 579 }; 580 String8 result; 581 audio_output_flags_t allFlags = AUDIO_OUTPUT_FLAG_NONE; 582 const mapping *entry; 583 for (entry = mappings; entry->mFlag != AUDIO_OUTPUT_FLAG_NONE; entry++) { 584 allFlags = (audio_output_flags_t) (allFlags | entry->mFlag); 585 if (flags & entry->mFlag) { 586 if (!result.isEmpty()) { 587 result.append("|"); 588 } 589 result.append(entry->mString); 590 } 591 } 592 if (flags & ~allFlags) { 593 if (!result.isEmpty()) { 594 result.append("|"); 595 } 596 result.appendFormat("0x%X", flags & ~allFlags); 597 } 598 if (result.isEmpty()) { 599 result.append(entry->mString); 600 } 601 return result; 602} 603 604const char *sourceToString(audio_source_t source) 605{ 606 switch (source) { 607 case AUDIO_SOURCE_DEFAULT: return "default"; 608 case AUDIO_SOURCE_MIC: return "mic"; 609 case AUDIO_SOURCE_VOICE_UPLINK: return "voice uplink"; 610 case AUDIO_SOURCE_VOICE_DOWNLINK: return "voice downlink"; 611 case AUDIO_SOURCE_VOICE_CALL: return "voice call"; 612 case AUDIO_SOURCE_CAMCORDER: return "camcorder"; 613 case AUDIO_SOURCE_VOICE_RECOGNITION: return "voice recognition"; 614 case AUDIO_SOURCE_VOICE_COMMUNICATION: return "voice communication"; 615 case AUDIO_SOURCE_REMOTE_SUBMIX: return "remote submix"; 616 case AUDIO_SOURCE_UNPROCESSED: return "unprocessed"; 617 case AUDIO_SOURCE_FM_TUNER: return "FM tuner"; 618 case AUDIO_SOURCE_HOTWORD: return "hotword"; 619 default: return "unknown"; 620 } 621} 622 623AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, 624 audio_devices_t outDevice, audio_devices_t inDevice, type_t type, bool systemReady) 625 : Thread(false /*canCallJava*/), 626 mType(type), 627 mAudioFlinger(audioFlinger), 628 // mSampleRate, mFrameCount, mChannelMask, mChannelCount, mFrameSize, mFormat, mBufferSize 629 // are set by PlaybackThread::readOutputParameters_l() or 630 // RecordThread::readInputParameters_l() 631 //FIXME: mStandby should be true here. Is this some kind of hack? 632 mStandby(false), mOutDevice(outDevice), mInDevice(inDevice), 633 mPrevOutDevice(AUDIO_DEVICE_NONE), mPrevInDevice(AUDIO_DEVICE_NONE), 634 mAudioSource(AUDIO_SOURCE_DEFAULT), mId(id), 635 // mName will be set by concrete (non-virtual) subclass 636 mDeathRecipient(new PMDeathRecipient(this)), 637 mSystemReady(systemReady), 638 mNotifiedBatteryStart(false) 639{ 640 memset(&mPatch, 0, sizeof(struct audio_patch)); 641} 642 643AudioFlinger::ThreadBase::~ThreadBase() 644{ 645 // mConfigEvents should be empty, but just in case it isn't, free the memory it owns 646 mConfigEvents.clear(); 647 648 // do not lock the mutex in destructor 649 releaseWakeLock_l(); 650 if (mPowerManager != 0) { 651 sp<IBinder> binder = IInterface::asBinder(mPowerManager); 652 binder->unlinkToDeath(mDeathRecipient); 653 } 654} 655 656status_t AudioFlinger::ThreadBase::readyToRun() 657{ 658 status_t status = initCheck(); 659 if (status == NO_ERROR) { 660 ALOGI("AudioFlinger's thread %p ready to run", this); 661 } else { 662 ALOGE("No working audio driver found."); 663 } 664 return status; 665} 666 667void AudioFlinger::ThreadBase::exit() 668{ 669 ALOGV("ThreadBase::exit"); 670 // do any cleanup required for exit to succeed 671 preExit(); 672 { 673 // This lock prevents the following race in thread (uniprocessor for illustration): 674 // if (!exitPending()) { 675 // // context switch from here to exit() 676 // // exit() calls requestExit(), what exitPending() observes 677 // // exit() calls signal(), which is dropped since no waiters 678 // // context switch back from exit() to here 679 // mWaitWorkCV.wait(...); 680 // // now thread is hung 681 // } 682 AutoMutex lock(mLock); 683 requestExit(); 684 mWaitWorkCV.broadcast(); 685 } 686 // When Thread::requestExitAndWait is made virtual and this method is renamed to 687 // "virtual status_t requestExitAndWait()", replace by "return Thread::requestExitAndWait();" 688 requestExitAndWait(); 689} 690 691status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs) 692{ 693 status_t status; 694 695 ALOGV("ThreadBase::setParameters() %s", keyValuePairs.string()); 696 Mutex::Autolock _l(mLock); 697 698 return sendSetParameterConfigEvent_l(keyValuePairs); 699} 700 701// sendConfigEvent_l() must be called with ThreadBase::mLock held 702// Can temporarily release the lock if waiting for a reply from processConfigEvents_l(). 703status_t AudioFlinger::ThreadBase::sendConfigEvent_l(sp<ConfigEvent>& event) 704{ 705 status_t status = NO_ERROR; 706 707 if (event->mRequiresSystemReady && !mSystemReady) { 708 event->mWaitStatus = false; 709 mPendingConfigEvents.add(event); 710 return status; 711 } 712 mConfigEvents.add(event); 713 ALOGV("sendConfigEvent_l() num events %d event %d", mConfigEvents.size(), event->mType); 714 mWaitWorkCV.signal(); 715 mLock.unlock(); 716 { 717 Mutex::Autolock _l(event->mLock); 718 while (event->mWaitStatus) { 719 if (event->mCond.waitRelative(event->mLock, kConfigEventTimeoutNs) != NO_ERROR) { 720 event->mStatus = TIMED_OUT; 721 event->mWaitStatus = false; 722 } 723 } 724 status = event->mStatus; 725 } 726 mLock.lock(); 727 return status; 728} 729 730void AudioFlinger::ThreadBase::sendIoConfigEvent(audio_io_config_event event, pid_t pid) 731{ 732 Mutex::Autolock _l(mLock); 733 sendIoConfigEvent_l(event, pid); 734} 735 736// sendIoConfigEvent_l() must be called with ThreadBase::mLock held 737void AudioFlinger::ThreadBase::sendIoConfigEvent_l(audio_io_config_event event, pid_t pid) 738{ 739 sp<ConfigEvent> configEvent = (ConfigEvent *)new IoConfigEvent(event, pid); 740 sendConfigEvent_l(configEvent); 741} 742 743void AudioFlinger::ThreadBase::sendPrioConfigEvent(pid_t pid, pid_t tid, int32_t prio) 744{ 745 Mutex::Autolock _l(mLock); 746 sendPrioConfigEvent_l(pid, tid, prio); 747} 748 749// sendPrioConfigEvent_l() must be called with ThreadBase::mLock held 750void AudioFlinger::ThreadBase::sendPrioConfigEvent_l(pid_t pid, pid_t tid, int32_t prio) 751{ 752 sp<ConfigEvent> configEvent = (ConfigEvent *)new PrioConfigEvent(pid, tid, prio); 753 sendConfigEvent_l(configEvent); 754} 755 756// sendSetParameterConfigEvent_l() must be called with ThreadBase::mLock held 757status_t AudioFlinger::ThreadBase::sendSetParameterConfigEvent_l(const String8& keyValuePair) 758{ 759 sp<ConfigEvent> configEvent; 760 AudioParameter param(keyValuePair); 761 int value; 762 if (param.getInt(String8(AUDIO_PARAMETER_MONO_OUTPUT), value) == NO_ERROR) { 763 setMasterMono_l(value != 0); 764 if (param.size() == 1) { 765 return NO_ERROR; // should be a solo parameter - we don't pass down 766 } 767 param.remove(String8(AUDIO_PARAMETER_MONO_OUTPUT)); 768 configEvent = new SetParameterConfigEvent(param.toString()); 769 } else { 770 configEvent = new SetParameterConfigEvent(keyValuePair); 771 } 772 return sendConfigEvent_l(configEvent); 773} 774 775status_t AudioFlinger::ThreadBase::sendCreateAudioPatchConfigEvent( 776 const struct audio_patch *patch, 777 audio_patch_handle_t *handle) 778{ 779 Mutex::Autolock _l(mLock); 780 sp<ConfigEvent> configEvent = (ConfigEvent *)new CreateAudioPatchConfigEvent(*patch, *handle); 781 status_t status = sendConfigEvent_l(configEvent); 782 if (status == NO_ERROR) { 783 CreateAudioPatchConfigEventData *data = 784 (CreateAudioPatchConfigEventData *)configEvent->mData.get(); 785 *handle = data->mHandle; 786 } 787 return status; 788} 789 790status_t AudioFlinger::ThreadBase::sendReleaseAudioPatchConfigEvent( 791 const audio_patch_handle_t handle) 792{ 793 Mutex::Autolock _l(mLock); 794 sp<ConfigEvent> configEvent = (ConfigEvent *)new ReleaseAudioPatchConfigEvent(handle); 795 return sendConfigEvent_l(configEvent); 796} 797 798 799// post condition: mConfigEvents.isEmpty() 800void AudioFlinger::ThreadBase::processConfigEvents_l() 801{ 802 bool configChanged = false; 803 804 while (!mConfigEvents.isEmpty()) { 805 ALOGV("processConfigEvents_l() remaining events %d", mConfigEvents.size()); 806 sp<ConfigEvent> event = mConfigEvents[0]; 807 mConfigEvents.removeAt(0); 808 switch (event->mType) { 809 case CFG_EVENT_PRIO: { 810 PrioConfigEventData *data = (PrioConfigEventData *)event->mData.get(); 811 // FIXME Need to understand why this has to be done asynchronously 812 int err = requestPriority(data->mPid, data->mTid, data->mPrio, 813 true /*asynchronous*/); 814 if (err != 0) { 815 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 816 data->mPrio, data->mPid, data->mTid, err); 817 } 818 } break; 819 case CFG_EVENT_IO: { 820 IoConfigEventData *data = (IoConfigEventData *)event->mData.get(); 821 ioConfigChanged(data->mEvent, data->mPid); 822 } break; 823 case CFG_EVENT_SET_PARAMETER: { 824 SetParameterConfigEventData *data = (SetParameterConfigEventData *)event->mData.get(); 825 if (checkForNewParameter_l(data->mKeyValuePairs, event->mStatus)) { 826 configChanged = true; 827 } 828 } break; 829 case CFG_EVENT_CREATE_AUDIO_PATCH: { 830 CreateAudioPatchConfigEventData *data = 831 (CreateAudioPatchConfigEventData *)event->mData.get(); 832 event->mStatus = createAudioPatch_l(&data->mPatch, &data->mHandle); 833 } break; 834 case CFG_EVENT_RELEASE_AUDIO_PATCH: { 835 ReleaseAudioPatchConfigEventData *data = 836 (ReleaseAudioPatchConfigEventData *)event->mData.get(); 837 event->mStatus = releaseAudioPatch_l(data->mHandle); 838 } break; 839 default: 840 ALOG_ASSERT(false, "processConfigEvents_l() unknown event type %d", event->mType); 841 break; 842 } 843 { 844 Mutex::Autolock _l(event->mLock); 845 if (event->mWaitStatus) { 846 event->mWaitStatus = false; 847 event->mCond.signal(); 848 } 849 } 850 ALOGV_IF(mConfigEvents.isEmpty(), "processConfigEvents_l() DONE thread %p", this); 851 } 852 853 if (configChanged) { 854 cacheParameters_l(); 855 } 856} 857 858String8 channelMaskToString(audio_channel_mask_t mask, bool output) { 859 String8 s; 860 const audio_channel_representation_t representation = 861 audio_channel_mask_get_representation(mask); 862 863 switch (representation) { 864 case AUDIO_CHANNEL_REPRESENTATION_POSITION: { 865 if (output) { 866 if (mask & AUDIO_CHANNEL_OUT_FRONT_LEFT) s.append("front-left, "); 867 if (mask & AUDIO_CHANNEL_OUT_FRONT_RIGHT) s.append("front-right, "); 868 if (mask & AUDIO_CHANNEL_OUT_FRONT_CENTER) s.append("front-center, "); 869 if (mask & AUDIO_CHANNEL_OUT_LOW_FREQUENCY) s.append("low freq, "); 870 if (mask & AUDIO_CHANNEL_OUT_BACK_LEFT) s.append("back-left, "); 871 if (mask & AUDIO_CHANNEL_OUT_BACK_RIGHT) s.append("back-right, "); 872 if (mask & AUDIO_CHANNEL_OUT_FRONT_LEFT_OF_CENTER) s.append("front-left-of-center, "); 873 if (mask & AUDIO_CHANNEL_OUT_FRONT_RIGHT_OF_CENTER) s.append("front-right-of-center, "); 874 if (mask & AUDIO_CHANNEL_OUT_BACK_CENTER) s.append("back-center, "); 875 if (mask & AUDIO_CHANNEL_OUT_SIDE_LEFT) s.append("side-left, "); 876 if (mask & AUDIO_CHANNEL_OUT_SIDE_RIGHT) s.append("side-right, "); 877 if (mask & AUDIO_CHANNEL_OUT_TOP_CENTER) s.append("top-center ,"); 878 if (mask & AUDIO_CHANNEL_OUT_TOP_FRONT_LEFT) s.append("top-front-left, "); 879 if (mask & AUDIO_CHANNEL_OUT_TOP_FRONT_CENTER) s.append("top-front-center, "); 880 if (mask & AUDIO_CHANNEL_OUT_TOP_FRONT_RIGHT) s.append("top-front-right, "); 881 if (mask & AUDIO_CHANNEL_OUT_TOP_BACK_LEFT) s.append("top-back-left, "); 882 if (mask & AUDIO_CHANNEL_OUT_TOP_BACK_CENTER) s.append("top-back-center, " ); 883 if (mask & AUDIO_CHANNEL_OUT_TOP_BACK_RIGHT) s.append("top-back-right, " ); 884 if (mask & ~AUDIO_CHANNEL_OUT_ALL) s.append("unknown, "); 885 } else { 886 if (mask & AUDIO_CHANNEL_IN_LEFT) s.append("left, "); 887 if (mask & AUDIO_CHANNEL_IN_RIGHT) s.append("right, "); 888 if (mask & AUDIO_CHANNEL_IN_FRONT) s.append("front, "); 889 if (mask & AUDIO_CHANNEL_IN_BACK) s.append("back, "); 890 if (mask & AUDIO_CHANNEL_IN_LEFT_PROCESSED) s.append("left-processed, "); 891 if (mask & AUDIO_CHANNEL_IN_RIGHT_PROCESSED) s.append("right-processed, "); 892 if (mask & AUDIO_CHANNEL_IN_FRONT_PROCESSED) s.append("front-processed, "); 893 if (mask & AUDIO_CHANNEL_IN_BACK_PROCESSED) s.append("back-processed, "); 894 if (mask & AUDIO_CHANNEL_IN_PRESSURE) s.append("pressure, "); 895 if (mask & AUDIO_CHANNEL_IN_X_AXIS) s.append("X, "); 896 if (mask & AUDIO_CHANNEL_IN_Y_AXIS) s.append("Y, "); 897 if (mask & AUDIO_CHANNEL_IN_Z_AXIS) s.append("Z, "); 898 if (mask & AUDIO_CHANNEL_IN_VOICE_UPLINK) s.append("voice-uplink, "); 899 if (mask & AUDIO_CHANNEL_IN_VOICE_DNLINK) s.append("voice-dnlink, "); 900 if (mask & ~AUDIO_CHANNEL_IN_ALL) s.append("unknown, "); 901 } 902 const int len = s.length(); 903 if (len > 2) { 904 char *str = s.lockBuffer(len); // needed? 905 s.unlockBuffer(len - 2); // remove trailing ", " 906 } 907 return s; 908 } 909 case AUDIO_CHANNEL_REPRESENTATION_INDEX: 910 s.appendFormat("index mask, bits:%#x", audio_channel_mask_get_bits(mask)); 911 return s; 912 default: 913 s.appendFormat("unknown mask, representation:%d bits:%#x", 914 representation, audio_channel_mask_get_bits(mask)); 915 return s; 916 } 917} 918 919void AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args __unused) 920{ 921 const size_t SIZE = 256; 922 char buffer[SIZE]; 923 String8 result; 924 925 bool locked = AudioFlinger::dumpTryLock(mLock); 926 if (!locked) { 927 dprintf(fd, "thread %p may be deadlocked\n", this); 928 } 929 930 dprintf(fd, " Thread name: %s\n", mThreadName); 931 dprintf(fd, " I/O handle: %d\n", mId); 932 dprintf(fd, " TID: %d\n", getTid()); 933 dprintf(fd, " Standby: %s\n", mStandby ? "yes" : "no"); 934 dprintf(fd, " Sample rate: %u Hz\n", mSampleRate); 935 dprintf(fd, " HAL frame count: %zu\n", mFrameCount); 936 dprintf(fd, " HAL format: 0x%x (%s)\n", mHALFormat, formatToString(mHALFormat)); 937 dprintf(fd, " HAL buffer size: %u bytes\n", mBufferSize); 938 dprintf(fd, " Channel count: %u\n", mChannelCount); 939 dprintf(fd, " Channel mask: 0x%08x (%s)\n", mChannelMask, 940 channelMaskToString(mChannelMask, mType != RECORD).string()); 941 dprintf(fd, " Processing format: 0x%x (%s)\n", mFormat, formatToString(mFormat)); 942 dprintf(fd, " Processing frame size: %zu bytes\n", mFrameSize); 943 dprintf(fd, " Pending config events:"); 944 size_t numConfig = mConfigEvents.size(); 945 if (numConfig) { 946 for (size_t i = 0; i < numConfig; i++) { 947 mConfigEvents[i]->dump(buffer, SIZE); 948 dprintf(fd, "\n %s", buffer); 949 } 950 dprintf(fd, "\n"); 951 } else { 952 dprintf(fd, " none\n"); 953 } 954 dprintf(fd, " Output device: %#x (%s)\n", mOutDevice, devicesToString(mOutDevice).string()); 955 dprintf(fd, " Input device: %#x (%s)\n", mInDevice, devicesToString(mInDevice).string()); 956 dprintf(fd, " Audio source: %d (%s)\n", mAudioSource, sourceToString(mAudioSource)); 957 958 if (locked) { 959 mLock.unlock(); 960 } 961} 962 963void AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args) 964{ 965 const size_t SIZE = 256; 966 char buffer[SIZE]; 967 String8 result; 968 969 size_t numEffectChains = mEffectChains.size(); 970 snprintf(buffer, SIZE, " %zu Effect Chains\n", numEffectChains); 971 write(fd, buffer, strlen(buffer)); 972 973 for (size_t i = 0; i < numEffectChains; ++i) { 974 sp<EffectChain> chain = mEffectChains[i]; 975 if (chain != 0) { 976 chain->dump(fd, args); 977 } 978 } 979} 980 981void AudioFlinger::ThreadBase::acquireWakeLock(int uid) 982{ 983 Mutex::Autolock _l(mLock); 984 acquireWakeLock_l(uid); 985} 986 987String16 AudioFlinger::ThreadBase::getWakeLockTag() 988{ 989 switch (mType) { 990 case MIXER: 991 return String16("AudioMix"); 992 case DIRECT: 993 return String16("AudioDirectOut"); 994 case DUPLICATING: 995 return String16("AudioDup"); 996 case RECORD: 997 return String16("AudioIn"); 998 case OFFLOAD: 999 return String16("AudioOffload"); 1000 default: 1001 ALOG_ASSERT(false); 1002 return String16("AudioUnknown"); 1003 } 1004} 1005 1006void AudioFlinger::ThreadBase::acquireWakeLock_l(int uid) 1007{ 1008 getPowerManager_l(); 1009 if (mPowerManager != 0) { 1010 sp<IBinder> binder = new BBinder(); 1011 status_t status; 1012 if (uid >= 0) { 1013 status = mPowerManager->acquireWakeLockWithUid(POWERMANAGER_PARTIAL_WAKE_LOCK, 1014 binder, 1015 getWakeLockTag(), 1016 String16("audioserver"), 1017 uid, 1018 true /* FIXME force oneway contrary to .aidl */); 1019 } else { 1020 status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK, 1021 binder, 1022 getWakeLockTag(), 1023 String16("audioserver"), 1024 true /* FIXME force oneway contrary to .aidl */); 1025 } 1026 if (status == NO_ERROR) { 1027 mWakeLockToken = binder; 1028 } 1029 ALOGV("acquireWakeLock_l() %s status %d", mThreadName, status); 1030 } 1031 1032 if (!mNotifiedBatteryStart) { 1033 BatteryNotifier::getInstance().noteStartAudio(); 1034 mNotifiedBatteryStart = true; 1035 } 1036 gBoottime.acquire(mWakeLockToken); 1037} 1038 1039void AudioFlinger::ThreadBase::releaseWakeLock() 1040{ 1041 Mutex::Autolock _l(mLock); 1042 releaseWakeLock_l(); 1043} 1044 1045void AudioFlinger::ThreadBase::releaseWakeLock_l() 1046{ 1047 gBoottime.release(mWakeLockToken); 1048 if (mWakeLockToken != 0) { 1049 ALOGV("releaseWakeLock_l() %s", mThreadName); 1050 if (mPowerManager != 0) { 1051 mPowerManager->releaseWakeLock(mWakeLockToken, 0, 1052 true /* FIXME force oneway contrary to .aidl */); 1053 } 1054 mWakeLockToken.clear(); 1055 } 1056 1057 if (mNotifiedBatteryStart) { 1058 BatteryNotifier::getInstance().noteStopAudio(); 1059 mNotifiedBatteryStart = false; 1060 } 1061} 1062 1063void AudioFlinger::ThreadBase::updateWakeLockUids(const SortedVector<int> &uids) { 1064 Mutex::Autolock _l(mLock); 1065 updateWakeLockUids_l(uids); 1066} 1067 1068void AudioFlinger::ThreadBase::getPowerManager_l() { 1069 if (mSystemReady && mPowerManager == 0) { 1070 // use checkService() to avoid blocking if power service is not up yet 1071 sp<IBinder> binder = 1072 defaultServiceManager()->checkService(String16("power")); 1073 if (binder == 0) { 1074 ALOGW("Thread %s cannot connect to the power manager service", mThreadName); 1075 } else { 1076 mPowerManager = interface_cast<IPowerManager>(binder); 1077 binder->linkToDeath(mDeathRecipient); 1078 } 1079 } 1080} 1081 1082void AudioFlinger::ThreadBase::updateWakeLockUids_l(const SortedVector<int> &uids) { 1083 getPowerManager_l(); 1084 if (mWakeLockToken == NULL) { // token may be NULL if AudioFlinger::systemReady() not called. 1085 if (mSystemReady) { 1086 ALOGE("no wake lock to update, but system ready!"); 1087 } else { 1088 ALOGW("no wake lock to update, system not ready yet"); 1089 } 1090 return; 1091 } 1092 if (mPowerManager != 0) { 1093 sp<IBinder> binder = new BBinder(); 1094 status_t status; 1095 status = mPowerManager->updateWakeLockUids(mWakeLockToken, uids.size(), uids.array(), 1096 true /* FIXME force oneway contrary to .aidl */); 1097 ALOGV("acquireWakeLock_l() %s status %d", mThreadName, status); 1098 } 1099} 1100 1101void AudioFlinger::ThreadBase::clearPowerManager() 1102{ 1103 Mutex::Autolock _l(mLock); 1104 releaseWakeLock_l(); 1105 mPowerManager.clear(); 1106} 1107 1108void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who __unused) 1109{ 1110 sp<ThreadBase> thread = mThread.promote(); 1111 if (thread != 0) { 1112 thread->clearPowerManager(); 1113 } 1114 ALOGW("power manager service died !!!"); 1115} 1116 1117void AudioFlinger::ThreadBase::setEffectSuspended( 1118 const effect_uuid_t *type, bool suspend, int sessionId) 1119{ 1120 Mutex::Autolock _l(mLock); 1121 setEffectSuspended_l(type, suspend, sessionId); 1122} 1123 1124void AudioFlinger::ThreadBase::setEffectSuspended_l( 1125 const effect_uuid_t *type, bool suspend, int sessionId) 1126{ 1127 sp<EffectChain> chain = getEffectChain_l(sessionId); 1128 if (chain != 0) { 1129 if (type != NULL) { 1130 chain->setEffectSuspended_l(type, suspend); 1131 } else { 1132 chain->setEffectSuspendedAll_l(suspend); 1133 } 1134 } 1135 1136 updateSuspendedSessions_l(type, suspend, sessionId); 1137} 1138 1139void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain) 1140{ 1141 ssize_t index = mSuspendedSessions.indexOfKey(chain->sessionId()); 1142 if (index < 0) { 1143 return; 1144 } 1145 1146 const KeyedVector <int, sp<SuspendedSessionDesc> >& sessionEffects = 1147 mSuspendedSessions.valueAt(index); 1148 1149 for (size_t i = 0; i < sessionEffects.size(); i++) { 1150 sp<SuspendedSessionDesc> desc = sessionEffects.valueAt(i); 1151 for (int j = 0; j < desc->mRefCount; j++) { 1152 if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) { 1153 chain->setEffectSuspendedAll_l(true); 1154 } else { 1155 ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x", 1156 desc->mType.timeLow); 1157 chain->setEffectSuspended_l(&desc->mType, true); 1158 } 1159 } 1160 } 1161} 1162 1163void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type, 1164 bool suspend, 1165 int sessionId) 1166{ 1167 ssize_t index = mSuspendedSessions.indexOfKey(sessionId); 1168 1169 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects; 1170 1171 if (suspend) { 1172 if (index >= 0) { 1173 sessionEffects = mSuspendedSessions.valueAt(index); 1174 } else { 1175 mSuspendedSessions.add(sessionId, sessionEffects); 1176 } 1177 } else { 1178 if (index < 0) { 1179 return; 1180 } 1181 sessionEffects = mSuspendedSessions.valueAt(index); 1182 } 1183 1184 1185 int key = EffectChain::kKeyForSuspendAll; 1186 if (type != NULL) { 1187 key = type->timeLow; 1188 } 1189 index = sessionEffects.indexOfKey(key); 1190 1191 sp<SuspendedSessionDesc> desc; 1192 if (suspend) { 1193 if (index >= 0) { 1194 desc = sessionEffects.valueAt(index); 1195 } else { 1196 desc = new SuspendedSessionDesc(); 1197 if (type != NULL) { 1198 desc->mType = *type; 1199 } 1200 sessionEffects.add(key, desc); 1201 ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key); 1202 } 1203 desc->mRefCount++; 1204 } else { 1205 if (index < 0) { 1206 return; 1207 } 1208 desc = sessionEffects.valueAt(index); 1209 if (--desc->mRefCount == 0) { 1210 ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key); 1211 sessionEffects.removeItemsAt(index); 1212 if (sessionEffects.isEmpty()) { 1213 ALOGV("updateSuspendedSessions_l() restore removing session %d", 1214 sessionId); 1215 mSuspendedSessions.removeItem(sessionId); 1216 } 1217 } 1218 } 1219 if (!sessionEffects.isEmpty()) { 1220 mSuspendedSessions.replaceValueFor(sessionId, sessionEffects); 1221 } 1222} 1223 1224void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 1225 bool enabled, 1226 int sessionId) 1227{ 1228 Mutex::Autolock _l(mLock); 1229 checkSuspendOnEffectEnabled_l(effect, enabled, sessionId); 1230} 1231 1232void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 1233 bool enabled, 1234 int sessionId) 1235{ 1236 if (mType != RECORD) { 1237 // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on 1238 // another session. This gives the priority to well behaved effect control panels 1239 // and applications not using global effects. 1240 // Enabling post processing in AUDIO_SESSION_OUTPUT_STAGE session does not affect 1241 // global effects 1242 if ((sessionId != AUDIO_SESSION_OUTPUT_MIX) && (sessionId != AUDIO_SESSION_OUTPUT_STAGE)) { 1243 setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX); 1244 } 1245 } 1246 1247 sp<EffectChain> chain = getEffectChain_l(sessionId); 1248 if (chain != 0) { 1249 chain->checkSuspendOnEffectEnabled(effect, enabled); 1250 } 1251} 1252 1253// ThreadBase::createEffect_l() must be called with AudioFlinger::mLock held 1254sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 1255 const sp<AudioFlinger::Client>& client, 1256 const sp<IEffectClient>& effectClient, 1257 int32_t priority, 1258 int sessionId, 1259 effect_descriptor_t *desc, 1260 int *enabled, 1261 status_t *status) 1262{ 1263 sp<EffectModule> effect; 1264 sp<EffectHandle> handle; 1265 status_t lStatus; 1266 sp<EffectChain> chain; 1267 bool chainCreated = false; 1268 bool effectCreated = false; 1269 bool effectRegistered = false; 1270 1271 lStatus = initCheck(); 1272 if (lStatus != NO_ERROR) { 1273 ALOGW("createEffect_l() Audio driver not initialized."); 1274 goto Exit; 1275 } 1276 1277 // Reject any effect on Direct output threads for now, since the format of 1278 // mSinkBuffer is not guaranteed to be compatible with effect processing (PCM 16 stereo). 1279 if (mType == DIRECT) { 1280 ALOGW("createEffect_l() Cannot add effect %s on Direct output type thread %s", 1281 desc->name, mThreadName); 1282 lStatus = BAD_VALUE; 1283 goto Exit; 1284 } 1285 1286 // Reject any effect on mixer or duplicating multichannel sinks. 1287 // TODO: fix both format and multichannel issues with effects. 1288 if ((mType == MIXER || mType == DUPLICATING) && mChannelCount != FCC_2) { 1289 ALOGW("createEffect_l() Cannot add effect %s for multichannel(%d) %s threads", 1290 desc->name, mChannelCount, mType == MIXER ? "MIXER" : "DUPLICATING"); 1291 lStatus = BAD_VALUE; 1292 goto Exit; 1293 } 1294 1295 // Allow global effects only on offloaded and mixer threads 1296 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 1297 switch (mType) { 1298 case MIXER: 1299 case OFFLOAD: 1300 break; 1301 case DIRECT: 1302 case DUPLICATING: 1303 case RECORD: 1304 default: 1305 ALOGW("createEffect_l() Cannot add global effect %s on thread %s", 1306 desc->name, mThreadName); 1307 lStatus = BAD_VALUE; 1308 goto Exit; 1309 } 1310 } 1311 1312 // Only Pre processor effects are allowed on input threads and only on input threads 1313 if ((mType == RECORD) != ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { 1314 ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d", 1315 desc->name, desc->flags, mType); 1316 lStatus = BAD_VALUE; 1317 goto Exit; 1318 } 1319 1320 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 1321 1322 { // scope for mLock 1323 Mutex::Autolock _l(mLock); 1324 1325 // check for existing effect chain with the requested audio session 1326 chain = getEffectChain_l(sessionId); 1327 if (chain == 0) { 1328 // create a new chain for this session 1329 ALOGV("createEffect_l() new effect chain for session %d", sessionId); 1330 chain = new EffectChain(this, sessionId); 1331 addEffectChain_l(chain); 1332 chain->setStrategy(getStrategyForSession_l(sessionId)); 1333 chainCreated = true; 1334 } else { 1335 effect = chain->getEffectFromDesc_l(desc); 1336 } 1337 1338 ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get()); 1339 1340 if (effect == 0) { 1341 int id = mAudioFlinger->nextUniqueId(); 1342 // Check CPU and memory usage 1343 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 1344 if (lStatus != NO_ERROR) { 1345 goto Exit; 1346 } 1347 effectRegistered = true; 1348 // create a new effect module if none present in the chain 1349 effect = new EffectModule(this, chain, desc, id, sessionId); 1350 lStatus = effect->status(); 1351 if (lStatus != NO_ERROR) { 1352 goto Exit; 1353 } 1354 effect->setOffloaded(mType == OFFLOAD, mId); 1355 1356 lStatus = chain->addEffect_l(effect); 1357 if (lStatus != NO_ERROR) { 1358 goto Exit; 1359 } 1360 effectCreated = true; 1361 1362 effect->setDevice(mOutDevice); 1363 effect->setDevice(mInDevice); 1364 effect->setMode(mAudioFlinger->getMode()); 1365 effect->setAudioSource(mAudioSource); 1366 } 1367 // create effect handle and connect it to effect module 1368 handle = new EffectHandle(effect, client, effectClient, priority); 1369 lStatus = handle->initCheck(); 1370 if (lStatus == OK) { 1371 lStatus = effect->addHandle(handle.get()); 1372 } 1373 if (enabled != NULL) { 1374 *enabled = (int)effect->isEnabled(); 1375 } 1376 } 1377 1378Exit: 1379 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 1380 Mutex::Autolock _l(mLock); 1381 if (effectCreated) { 1382 chain->removeEffect_l(effect); 1383 } 1384 if (effectRegistered) { 1385 AudioSystem::unregisterEffect(effect->id()); 1386 } 1387 if (chainCreated) { 1388 removeEffectChain_l(chain); 1389 } 1390 handle.clear(); 1391 } 1392 1393 *status = lStatus; 1394 return handle; 1395} 1396 1397sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect(int sessionId, int effectId) 1398{ 1399 Mutex::Autolock _l(mLock); 1400 return getEffect_l(sessionId, effectId); 1401} 1402 1403sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId) 1404{ 1405 sp<EffectChain> chain = getEffectChain_l(sessionId); 1406 return chain != 0 ? chain->getEffectFromId_l(effectId) : 0; 1407} 1408 1409// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 1410// PlaybackThread::mLock held 1411status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 1412{ 1413 // check for existing effect chain with the requested audio session 1414 int sessionId = effect->sessionId(); 1415 sp<EffectChain> chain = getEffectChain_l(sessionId); 1416 bool chainCreated = false; 1417 1418 ALOGD_IF((mType == OFFLOAD) && !effect->isOffloadable(), 1419 "addEffect_l() on offloaded thread %p: effect %s does not support offload flags %x", 1420 this, effect->desc().name, effect->desc().flags); 1421 1422 if (chain == 0) { 1423 // create a new chain for this session 1424 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 1425 chain = new EffectChain(this, sessionId); 1426 addEffectChain_l(chain); 1427 chain->setStrategy(getStrategyForSession_l(sessionId)); 1428 chainCreated = true; 1429 } 1430 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 1431 1432 if (chain->getEffectFromId_l(effect->id()) != 0) { 1433 ALOGW("addEffect_l() %p effect %s already present in chain %p", 1434 this, effect->desc().name, chain.get()); 1435 return BAD_VALUE; 1436 } 1437 1438 effect->setOffloaded(mType == OFFLOAD, mId); 1439 1440 status_t status = chain->addEffect_l(effect); 1441 if (status != NO_ERROR) { 1442 if (chainCreated) { 1443 removeEffectChain_l(chain); 1444 } 1445 return status; 1446 } 1447 1448 effect->setDevice(mOutDevice); 1449 effect->setDevice(mInDevice); 1450 effect->setMode(mAudioFlinger->getMode()); 1451 effect->setAudioSource(mAudioSource); 1452 return NO_ERROR; 1453} 1454 1455void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 1456 1457 ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 1458 effect_descriptor_t desc = effect->desc(); 1459 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 1460 detachAuxEffect_l(effect->id()); 1461 } 1462 1463 sp<EffectChain> chain = effect->chain().promote(); 1464 if (chain != 0) { 1465 // remove effect chain if removing last effect 1466 if (chain->removeEffect_l(effect) == 0) { 1467 removeEffectChain_l(chain); 1468 } 1469 } else { 1470 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 1471 } 1472} 1473 1474void AudioFlinger::ThreadBase::lockEffectChains_l( 1475 Vector< sp<AudioFlinger::EffectChain> >& effectChains) 1476{ 1477 effectChains = mEffectChains; 1478 for (size_t i = 0; i < mEffectChains.size(); i++) { 1479 mEffectChains[i]->lock(); 1480 } 1481} 1482 1483void AudioFlinger::ThreadBase::unlockEffectChains( 1484 const Vector< sp<AudioFlinger::EffectChain> >& effectChains) 1485{ 1486 for (size_t i = 0; i < effectChains.size(); i++) { 1487 effectChains[i]->unlock(); 1488 } 1489} 1490 1491sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId) 1492{ 1493 Mutex::Autolock _l(mLock); 1494 return getEffectChain_l(sessionId); 1495} 1496 1497sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) const 1498{ 1499 size_t size = mEffectChains.size(); 1500 for (size_t i = 0; i < size; i++) { 1501 if (mEffectChains[i]->sessionId() == sessionId) { 1502 return mEffectChains[i]; 1503 } 1504 } 1505 return 0; 1506} 1507 1508void AudioFlinger::ThreadBase::setMode(audio_mode_t mode) 1509{ 1510 Mutex::Autolock _l(mLock); 1511 size_t size = mEffectChains.size(); 1512 for (size_t i = 0; i < size; i++) { 1513 mEffectChains[i]->setMode_l(mode); 1514 } 1515} 1516 1517void AudioFlinger::ThreadBase::getAudioPortConfig(struct audio_port_config *config) 1518{ 1519 config->type = AUDIO_PORT_TYPE_MIX; 1520 config->ext.mix.handle = mId; 1521 config->sample_rate = mSampleRate; 1522 config->format = mFormat; 1523 config->channel_mask = mChannelMask; 1524 config->config_mask = AUDIO_PORT_CONFIG_SAMPLE_RATE|AUDIO_PORT_CONFIG_CHANNEL_MASK| 1525 AUDIO_PORT_CONFIG_FORMAT; 1526} 1527 1528void AudioFlinger::ThreadBase::systemReady() 1529{ 1530 Mutex::Autolock _l(mLock); 1531 if (mSystemReady) { 1532 return; 1533 } 1534 mSystemReady = true; 1535 1536 for (size_t i = 0; i < mPendingConfigEvents.size(); i++) { 1537 sendConfigEvent_l(mPendingConfigEvents.editItemAt(i)); 1538 } 1539 mPendingConfigEvents.clear(); 1540} 1541 1542 1543// ---------------------------------------------------------------------------- 1544// Playback 1545// ---------------------------------------------------------------------------- 1546 1547AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, 1548 AudioStreamOut* output, 1549 audio_io_handle_t id, 1550 audio_devices_t device, 1551 type_t type, 1552 bool systemReady) 1553 : ThreadBase(audioFlinger, id, device, AUDIO_DEVICE_NONE, type, systemReady), 1554 mNormalFrameCount(0), mSinkBuffer(NULL), 1555 mMixerBufferEnabled(AudioFlinger::kEnableExtendedPrecision), 1556 mMixerBuffer(NULL), 1557 mMixerBufferSize(0), 1558 mMixerBufferFormat(AUDIO_FORMAT_INVALID), 1559 mMixerBufferValid(false), 1560 mEffectBufferEnabled(AudioFlinger::kEnableExtendedPrecision), 1561 mEffectBuffer(NULL), 1562 mEffectBufferSize(0), 1563 mEffectBufferFormat(AUDIO_FORMAT_INVALID), 1564 mEffectBufferValid(false), 1565 mSuspended(0), mBytesWritten(0), 1566 mActiveTracksGeneration(0), 1567 // mStreamTypes[] initialized in constructor body 1568 mOutput(output), 1569 mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false), 1570 mMixerStatus(MIXER_IDLE), 1571 mMixerStatusIgnoringFastTracks(MIXER_IDLE), 1572 mStandbyDelayNs(AudioFlinger::mStandbyTimeInNsecs), 1573 mBytesRemaining(0), 1574 mCurrentWriteLength(0), 1575 mUseAsyncWrite(false), 1576 mWriteAckSequence(0), 1577 mDrainSequence(0), 1578 mSignalPending(false), 1579 mScreenState(AudioFlinger::mScreenState), 1580 // index 0 is reserved for normal mixer's submix 1581 mFastTrackAvailMask(((1 << FastMixerState::kMaxFastTracks) - 1) & ~1), 1582 mHwSupportsPause(false), mHwPaused(false), mFlushPending(false) 1583{ 1584 snprintf(mThreadName, kThreadNameLength, "AudioOut_%X", id); 1585 mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mThreadName); 1586 1587 // Assumes constructor is called by AudioFlinger with it's mLock held, but 1588 // it would be safer to explicitly pass initial masterVolume/masterMute as 1589 // parameter. 1590 // 1591 // If the HAL we are using has support for master volume or master mute, 1592 // then do not attenuate or mute during mixing (just leave the volume at 1.0 1593 // and the mute set to false). 1594 mMasterVolume = audioFlinger->masterVolume_l(); 1595 mMasterMute = audioFlinger->masterMute_l(); 1596 if (mOutput && mOutput->audioHwDev) { 1597 if (mOutput->audioHwDev->canSetMasterVolume()) { 1598 mMasterVolume = 1.0; 1599 } 1600 1601 if (mOutput->audioHwDev->canSetMasterMute()) { 1602 mMasterMute = false; 1603 } 1604 } 1605 1606 readOutputParameters_l(); 1607 1608 // ++ operator does not compile 1609 for (audio_stream_type_t stream = AUDIO_STREAM_MIN; stream < AUDIO_STREAM_CNT; 1610 stream = (audio_stream_type_t) (stream + 1)) { 1611 mStreamTypes[stream].volume = mAudioFlinger->streamVolume_l(stream); 1612 mStreamTypes[stream].mute = mAudioFlinger->streamMute_l(stream); 1613 } 1614} 1615 1616AudioFlinger::PlaybackThread::~PlaybackThread() 1617{ 1618 mAudioFlinger->unregisterWriter(mNBLogWriter); 1619 free(mSinkBuffer); 1620 free(mMixerBuffer); 1621 free(mEffectBuffer); 1622} 1623 1624void AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) 1625{ 1626 dumpInternals(fd, args); 1627 dumpTracks(fd, args); 1628 dumpEffectChains(fd, args); 1629} 1630 1631void AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args __unused) 1632{ 1633 const size_t SIZE = 256; 1634 char buffer[SIZE]; 1635 String8 result; 1636 1637 result.appendFormat(" Stream volumes in dB: "); 1638 for (int i = 0; i < AUDIO_STREAM_CNT; ++i) { 1639 const stream_type_t *st = &mStreamTypes[i]; 1640 if (i > 0) { 1641 result.appendFormat(", "); 1642 } 1643 result.appendFormat("%d:%.2g", i, 20.0 * log10(st->volume)); 1644 if (st->mute) { 1645 result.append("M"); 1646 } 1647 } 1648 result.append("\n"); 1649 write(fd, result.string(), result.length()); 1650 result.clear(); 1651 1652 // These values are "raw"; they will wrap around. See prepareTracks_l() for a better way. 1653 FastTrackUnderruns underruns = getFastTrackUnderruns(0); 1654 dprintf(fd, " Normal mixer raw underrun counters: partial=%u empty=%u\n", 1655 underruns.mBitFields.mPartial, underruns.mBitFields.mEmpty); 1656 1657 size_t numtracks = mTracks.size(); 1658 size_t numactive = mActiveTracks.size(); 1659 dprintf(fd, " %d Tracks", numtracks); 1660 size_t numactiveseen = 0; 1661 if (numtracks) { 1662 dprintf(fd, " of which %d are active\n", numactive); 1663 Track::appendDumpHeader(result); 1664 for (size_t i = 0; i < numtracks; ++i) { 1665 sp<Track> track = mTracks[i]; 1666 if (track != 0) { 1667 bool active = mActiveTracks.indexOf(track) >= 0; 1668 if (active) { 1669 numactiveseen++; 1670 } 1671 track->dump(buffer, SIZE, active); 1672 result.append(buffer); 1673 } 1674 } 1675 } else { 1676 result.append("\n"); 1677 } 1678 if (numactiveseen != numactive) { 1679 // some tracks in the active list were not in the tracks list 1680 snprintf(buffer, SIZE, " The following tracks are in the active list but" 1681 " not in the track list\n"); 1682 result.append(buffer); 1683 Track::appendDumpHeader(result); 1684 for (size_t i = 0; i < numactive; ++i) { 1685 sp<Track> track = mActiveTracks[i].promote(); 1686 if (track != 0 && mTracks.indexOf(track) < 0) { 1687 track->dump(buffer, SIZE, true); 1688 result.append(buffer); 1689 } 1690 } 1691 } 1692 1693 write(fd, result.string(), result.size()); 1694} 1695 1696void AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 1697{ 1698 dprintf(fd, "\nOutput thread %p type %d (%s):\n", this, type(), threadTypeToString(type())); 1699 1700 dumpBase(fd, args); 1701 1702 dprintf(fd, " Normal frame count: %zu\n", mNormalFrameCount); 1703 dprintf(fd, " Last write occurred (msecs): %llu\n", ns2ms(systemTime() - mLastWriteTime)); 1704 dprintf(fd, " Total writes: %d\n", mNumWrites); 1705 dprintf(fd, " Delayed writes: %d\n", mNumDelayedWrites); 1706 dprintf(fd, " Blocked in write: %s\n", mInWrite ? "yes" : "no"); 1707 dprintf(fd, " Suspend count: %d\n", mSuspended); 1708 dprintf(fd, " Sink buffer : %p\n", mSinkBuffer); 1709 dprintf(fd, " Mixer buffer: %p\n", mMixerBuffer); 1710 dprintf(fd, " Effect buffer: %p\n", mEffectBuffer); 1711 dprintf(fd, " Fast track availMask=%#x\n", mFastTrackAvailMask); 1712 dprintf(fd, " Standby delay ns=%lld\n", (long long)mStandbyDelayNs); 1713 AudioStreamOut *output = mOutput; 1714 audio_output_flags_t flags = output != NULL ? output->flags : AUDIO_OUTPUT_FLAG_NONE; 1715 String8 flagsAsString = outputFlagsToString(flags); 1716 dprintf(fd, " AudioStreamOut: %p flags %#x (%s)\n", output, flags, flagsAsString.string()); 1717} 1718 1719// Thread virtuals 1720 1721void AudioFlinger::PlaybackThread::onFirstRef() 1722{ 1723 run(mThreadName, ANDROID_PRIORITY_URGENT_AUDIO); 1724} 1725 1726// ThreadBase virtuals 1727void AudioFlinger::PlaybackThread::preExit() 1728{ 1729 ALOGV(" preExit()"); 1730 // FIXME this is using hard-coded strings but in the future, this functionality will be 1731 // converted to use audio HAL extensions required to support tunneling 1732 mOutput->stream->common.set_parameters(&mOutput->stream->common, "exiting=1"); 1733} 1734 1735// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 1736sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 1737 const sp<AudioFlinger::Client>& client, 1738 audio_stream_type_t streamType, 1739 uint32_t sampleRate, 1740 audio_format_t format, 1741 audio_channel_mask_t channelMask, 1742 size_t *pFrameCount, 1743 const sp<IMemory>& sharedBuffer, 1744 int sessionId, 1745 IAudioFlinger::track_flags_t *flags, 1746 pid_t tid, 1747 int uid, 1748 status_t *status) 1749{ 1750 size_t frameCount = *pFrameCount; 1751 sp<Track> track; 1752 status_t lStatus; 1753 1754 // client expresses a preference for FAST, but we get the final say 1755 if (*flags & IAudioFlinger::TRACK_FAST) { 1756 if ( 1757 // either of these use cases: 1758 ( 1759 // use case 1: shared buffer with any frame count 1760 ( 1761 (sharedBuffer != 0) 1762 ) || 1763 // use case 2: frame count is default or at least as large as HAL 1764 ( 1765 // we formerly checked for a callback handler (non-0 tid), 1766 // but that is no longer required for TRANSFER_OBTAIN mode 1767 ((frameCount == 0) || 1768 (frameCount >= mFrameCount)) 1769 ) 1770 ) && 1771 // PCM data 1772 audio_is_linear_pcm(format) && 1773 // TODO: extract as a data library function that checks that a computationally 1774 // expensive downmixer is not required: isFastOutputChannelConversion() 1775 (channelMask == mChannelMask || 1776 mChannelMask != AUDIO_CHANNEL_OUT_STEREO || 1777 (channelMask == AUDIO_CHANNEL_OUT_MONO 1778 /* && mChannelMask == AUDIO_CHANNEL_OUT_STEREO */)) && 1779 // hardware sample rate 1780 (sampleRate == mSampleRate) && 1781 // normal mixer has an associated fast mixer 1782 hasFastMixer() && 1783 // there are sufficient fast track slots available 1784 (mFastTrackAvailMask != 0) 1785 // FIXME test that MixerThread for this fast track has a capable output HAL 1786 // FIXME add a permission test also? 1787 ) { 1788 // if frameCount not specified, then it defaults to fast mixer (HAL) frame count 1789 if (frameCount == 0) { 1790 // read the fast track multiplier property the first time it is needed 1791 int ok = pthread_once(&sFastTrackMultiplierOnce, sFastTrackMultiplierInit); 1792 if (ok != 0) { 1793 ALOGE("%s pthread_once failed: %d", __func__, ok); 1794 } 1795 frameCount = mFrameCount * sFastTrackMultiplier; 1796 } 1797 ALOGV("AUDIO_OUTPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d", 1798 frameCount, mFrameCount); 1799 } else { 1800 ALOGV("AUDIO_OUTPUT_FLAG_FAST denied: sharedBuffer=%p frameCount=%d " 1801 "mFrameCount=%d format=%#x mFormat=%#x isLinear=%d channelMask=%#x " 1802 "sampleRate=%u mSampleRate=%u " 1803 "hasFastMixer=%d tid=%d fastTrackAvailMask=%#x", 1804 sharedBuffer.get(), frameCount, mFrameCount, format, mFormat, 1805 audio_is_linear_pcm(format), 1806 channelMask, sampleRate, mSampleRate, hasFastMixer(), tid, mFastTrackAvailMask); 1807 *flags &= ~IAudioFlinger::TRACK_FAST; 1808 } 1809 } 1810 // For normal PCM streaming tracks, update minimum frame count. 1811 // For compatibility with AudioTrack calculation, buffer depth is forced 1812 // to be at least 2 x the normal mixer frame count and cover audio hardware latency. 1813 // This is probably too conservative, but legacy application code may depend on it. 1814 // If you change this calculation, also review the start threshold which is related. 1815 if (!(*flags & IAudioFlinger::TRACK_FAST) 1816 && audio_is_linear_pcm(format) && sharedBuffer == 0) { 1817 // this must match AudioTrack.cpp calculateMinFrameCount(). 1818 // TODO: Move to a common library 1819 uint32_t latencyMs = mOutput->stream->get_latency(mOutput->stream); 1820 uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate); 1821 if (minBufCount < 2) { 1822 minBufCount = 2; 1823 } 1824 // For normal mixing tracks, if speed is > 1.0f (normal), AudioTrack 1825 // or the client should compute and pass in a larger buffer request. 1826 size_t minFrameCount = 1827 minBufCount * sourceFramesNeededWithTimestretch( 1828 sampleRate, mNormalFrameCount, 1829 mSampleRate, AUDIO_TIMESTRETCH_SPEED_NORMAL /*speed*/); 1830 if (frameCount < minFrameCount) { // including frameCount == 0 1831 frameCount = minFrameCount; 1832 } 1833 } 1834 *pFrameCount = frameCount; 1835 1836 switch (mType) { 1837 1838 case DIRECT: 1839 if (audio_is_linear_pcm(format)) { 1840 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1841 ALOGE("createTrack_l() Bad parameter: sampleRate %u format %#x, channelMask 0x%08x " 1842 "for output %p with format %#x", 1843 sampleRate, format, channelMask, mOutput, mFormat); 1844 lStatus = BAD_VALUE; 1845 goto Exit; 1846 } 1847 } 1848 break; 1849 1850 case OFFLOAD: 1851 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1852 ALOGE("createTrack_l() Bad parameter: sampleRate %d format %#x, channelMask 0x%08x \"" 1853 "for output %p with format %#x", 1854 sampleRate, format, channelMask, mOutput, mFormat); 1855 lStatus = BAD_VALUE; 1856 goto Exit; 1857 } 1858 break; 1859 1860 default: 1861 if (!audio_is_linear_pcm(format)) { 1862 ALOGE("createTrack_l() Bad parameter: format %#x \"" 1863 "for output %p with format %#x", 1864 format, mOutput, mFormat); 1865 lStatus = BAD_VALUE; 1866 goto Exit; 1867 } 1868 if (sampleRate > mSampleRate * AUDIO_RESAMPLER_DOWN_RATIO_MAX) { 1869 ALOGE("Sample rate out of range: %u mSampleRate %u", sampleRate, mSampleRate); 1870 lStatus = BAD_VALUE; 1871 goto Exit; 1872 } 1873 break; 1874 1875 } 1876 1877 lStatus = initCheck(); 1878 if (lStatus != NO_ERROR) { 1879 ALOGE("createTrack_l() audio driver not initialized"); 1880 goto Exit; 1881 } 1882 1883 { // scope for mLock 1884 Mutex::Autolock _l(mLock); 1885 1886 // all tracks in same audio session must share the same routing strategy otherwise 1887 // conflicts will happen when tracks are moved from one output to another by audio policy 1888 // manager 1889 uint32_t strategy = AudioSystem::getStrategyForStream(streamType); 1890 for (size_t i = 0; i < mTracks.size(); ++i) { 1891 sp<Track> t = mTracks[i]; 1892 if (t != 0 && t->isExternalTrack()) { 1893 uint32_t actual = AudioSystem::getStrategyForStream(t->streamType()); 1894 if (sessionId == t->sessionId() && strategy != actual) { 1895 ALOGE("createTrack_l() mismatched strategy; expected %u but found %u", 1896 strategy, actual); 1897 lStatus = BAD_VALUE; 1898 goto Exit; 1899 } 1900 } 1901 } 1902 1903 track = new Track(this, client, streamType, sampleRate, format, 1904 channelMask, frameCount, NULL, sharedBuffer, 1905 sessionId, uid, *flags, TrackBase::TYPE_DEFAULT); 1906 1907 lStatus = track != 0 ? track->initCheck() : (status_t) NO_MEMORY; 1908 if (lStatus != NO_ERROR) { 1909 ALOGE("createTrack_l() initCheck failed %d; no control block?", lStatus); 1910 // track must be cleared from the caller as the caller has the AF lock 1911 goto Exit; 1912 } 1913 mTracks.add(track); 1914 1915 sp<EffectChain> chain = getEffectChain_l(sessionId); 1916 if (chain != 0) { 1917 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1918 track->setMainBuffer(chain->inBuffer()); 1919 chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType())); 1920 chain->incTrackCnt(); 1921 } 1922 1923 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 1924 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 1925 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 1926 // so ask activity manager to do this on our behalf 1927 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 1928 } 1929 } 1930 1931 lStatus = NO_ERROR; 1932 1933Exit: 1934 *status = lStatus; 1935 return track; 1936} 1937 1938uint32_t AudioFlinger::PlaybackThread::correctLatency_l(uint32_t latency) const 1939{ 1940 return latency; 1941} 1942 1943uint32_t AudioFlinger::PlaybackThread::latency() const 1944{ 1945 Mutex::Autolock _l(mLock); 1946 return latency_l(); 1947} 1948uint32_t AudioFlinger::PlaybackThread::latency_l() const 1949{ 1950 if (initCheck() == NO_ERROR) { 1951 return correctLatency_l(mOutput->stream->get_latency(mOutput->stream)); 1952 } else { 1953 return 0; 1954 } 1955} 1956 1957void AudioFlinger::PlaybackThread::setMasterVolume(float value) 1958{ 1959 Mutex::Autolock _l(mLock); 1960 // Don't apply master volume in SW if our HAL can do it for us. 1961 if (mOutput && mOutput->audioHwDev && 1962 mOutput->audioHwDev->canSetMasterVolume()) { 1963 mMasterVolume = 1.0; 1964 } else { 1965 mMasterVolume = value; 1966 } 1967} 1968 1969void AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1970{ 1971 Mutex::Autolock _l(mLock); 1972 // Don't apply master mute in SW if our HAL can do it for us. 1973 if (mOutput && mOutput->audioHwDev && 1974 mOutput->audioHwDev->canSetMasterMute()) { 1975 mMasterMute = false; 1976 } else { 1977 mMasterMute = muted; 1978 } 1979} 1980 1981void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) 1982{ 1983 Mutex::Autolock _l(mLock); 1984 mStreamTypes[stream].volume = value; 1985 broadcast_l(); 1986} 1987 1988void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) 1989{ 1990 Mutex::Autolock _l(mLock); 1991 mStreamTypes[stream].mute = muted; 1992 broadcast_l(); 1993} 1994 1995float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const 1996{ 1997 Mutex::Autolock _l(mLock); 1998 return mStreamTypes[stream].volume; 1999} 2000 2001// addTrack_l() must be called with ThreadBase::mLock held 2002status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 2003{ 2004 status_t status = ALREADY_EXISTS; 2005 2006 // set retry count for buffer fill 2007 track->mRetryCount = kMaxTrackStartupRetries; 2008 if (mActiveTracks.indexOf(track) < 0) { 2009 // the track is newly added, make sure it fills up all its 2010 // buffers before playing. This is to ensure the client will 2011 // effectively get the latency it requested. 2012 if (track->isExternalTrack()) { 2013 TrackBase::track_state state = track->mState; 2014 mLock.unlock(); 2015 status = AudioSystem::startOutput(mId, track->streamType(), 2016 (audio_session_t)track->sessionId()); 2017 mLock.lock(); 2018 // abort track was stopped/paused while we released the lock 2019 if (state != track->mState) { 2020 if (status == NO_ERROR) { 2021 mLock.unlock(); 2022 AudioSystem::stopOutput(mId, track->streamType(), 2023 (audio_session_t)track->sessionId()); 2024 mLock.lock(); 2025 } 2026 return INVALID_OPERATION; 2027 } 2028 // abort if start is rejected by audio policy manager 2029 if (status != NO_ERROR) { 2030 return PERMISSION_DENIED; 2031 } 2032#ifdef ADD_BATTERY_DATA 2033 // to track the speaker usage 2034 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 2035#endif 2036 } 2037 2038 track->mFillingUpStatus = track->sharedBuffer() != 0 ? Track::FS_FILLED : Track::FS_FILLING; 2039 track->mResetDone = false; 2040 track->mPresentationCompleteFrames = 0; 2041 mActiveTracks.add(track); 2042 mWakeLockUids.add(track->uid()); 2043 mActiveTracksGeneration++; 2044 mLatestActiveTrack = track; 2045 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 2046 if (chain != 0) { 2047 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), 2048 track->sessionId()); 2049 chain->incActiveTrackCnt(); 2050 } 2051 2052 status = NO_ERROR; 2053 } 2054 2055 onAddNewTrack_l(); 2056 return status; 2057} 2058 2059bool AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 2060{ 2061 track->terminate(); 2062 // active tracks are removed by threadLoop() 2063 bool trackActive = (mActiveTracks.indexOf(track) >= 0); 2064 track->mState = TrackBase::STOPPED; 2065 if (!trackActive) { 2066 removeTrack_l(track); 2067 } else if (track->isFastTrack() || track->isOffloaded() || track->isDirect()) { 2068 track->mState = TrackBase::STOPPING_1; 2069 } 2070 2071 return trackActive; 2072} 2073 2074void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 2075{ 2076 track->triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 2077 mTracks.remove(track); 2078 deleteTrackName_l(track->name()); 2079 // redundant as track is about to be destroyed, for dumpsys only 2080 track->mName = -1; 2081 if (track->isFastTrack()) { 2082 int index = track->mFastIndex; 2083 ALOG_ASSERT(0 < index && index < (int)FastMixerState::kMaxFastTracks); 2084 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << index))); 2085 mFastTrackAvailMask |= 1 << index; 2086 // redundant as track is about to be destroyed, for dumpsys only 2087 track->mFastIndex = -1; 2088 } 2089 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 2090 if (chain != 0) { 2091 chain->decTrackCnt(); 2092 } 2093} 2094 2095void AudioFlinger::PlaybackThread::broadcast_l() 2096{ 2097 // Thread could be blocked waiting for async 2098 // so signal it to handle state changes immediately 2099 // If threadLoop is currently unlocked a signal of mWaitWorkCV will 2100 // be lost so we also flag to prevent it blocking on mWaitWorkCV 2101 mSignalPending = true; 2102 mWaitWorkCV.broadcast(); 2103} 2104 2105String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 2106{ 2107 Mutex::Autolock _l(mLock); 2108 if (initCheck() != NO_ERROR) { 2109 return String8(); 2110 } 2111 2112 char *s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 2113 const String8 out_s8(s); 2114 free(s); 2115 return out_s8; 2116} 2117 2118void AudioFlinger::PlaybackThread::ioConfigChanged(audio_io_config_event event, pid_t pid) { 2119 sp<AudioIoDescriptor> desc = new AudioIoDescriptor(); 2120 ALOGV("PlaybackThread::ioConfigChanged, thread %p, event %d", this, event); 2121 2122 desc->mIoHandle = mId; 2123 2124 switch (event) { 2125 case AUDIO_OUTPUT_OPENED: 2126 case AUDIO_OUTPUT_CONFIG_CHANGED: 2127 desc->mPatch = mPatch; 2128 desc->mChannelMask = mChannelMask; 2129 desc->mSamplingRate = mSampleRate; 2130 desc->mFormat = mFormat; 2131 desc->mFrameCount = mNormalFrameCount; // FIXME see 2132 // AudioFlinger::frameCount(audio_io_handle_t) 2133 desc->mLatency = latency_l(); 2134 break; 2135 2136 case AUDIO_OUTPUT_CLOSED: 2137 default: 2138 break; 2139 } 2140 mAudioFlinger->ioConfigChanged(event, desc, pid); 2141} 2142 2143void AudioFlinger::PlaybackThread::writeCallback() 2144{ 2145 ALOG_ASSERT(mCallbackThread != 0); 2146 mCallbackThread->resetWriteBlocked(); 2147} 2148 2149void AudioFlinger::PlaybackThread::drainCallback() 2150{ 2151 ALOG_ASSERT(mCallbackThread != 0); 2152 mCallbackThread->resetDraining(); 2153} 2154 2155void AudioFlinger::PlaybackThread::resetWriteBlocked(uint32_t sequence) 2156{ 2157 Mutex::Autolock _l(mLock); 2158 // reject out of sequence requests 2159 if ((mWriteAckSequence & 1) && (sequence == mWriteAckSequence)) { 2160 mWriteAckSequence &= ~1; 2161 mWaitWorkCV.signal(); 2162 } 2163} 2164 2165void AudioFlinger::PlaybackThread::resetDraining(uint32_t sequence) 2166{ 2167 Mutex::Autolock _l(mLock); 2168 // reject out of sequence requests 2169 if ((mDrainSequence & 1) && (sequence == mDrainSequence)) { 2170 mDrainSequence &= ~1; 2171 mWaitWorkCV.signal(); 2172 } 2173} 2174 2175// static 2176int AudioFlinger::PlaybackThread::asyncCallback(stream_callback_event_t event, 2177 void *param __unused, 2178 void *cookie) 2179{ 2180 AudioFlinger::PlaybackThread *me = (AudioFlinger::PlaybackThread *)cookie; 2181 ALOGV("asyncCallback() event %d", event); 2182 switch (event) { 2183 case STREAM_CBK_EVENT_WRITE_READY: 2184 me->writeCallback(); 2185 break; 2186 case STREAM_CBK_EVENT_DRAIN_READY: 2187 me->drainCallback(); 2188 break; 2189 default: 2190 ALOGW("asyncCallback() unknown event %d", event); 2191 break; 2192 } 2193 return 0; 2194} 2195 2196void AudioFlinger::PlaybackThread::readOutputParameters_l() 2197{ 2198 // unfortunately we have no way of recovering from errors here, hence the LOG_ALWAYS_FATAL 2199 mSampleRate = mOutput->getSampleRate(); 2200 mChannelMask = mOutput->getChannelMask(); 2201 if (!audio_is_output_channel(mChannelMask)) { 2202 LOG_ALWAYS_FATAL("HAL channel mask %#x not valid for output", mChannelMask); 2203 } 2204 if ((mType == MIXER || mType == DUPLICATING) 2205 && !isValidPcmSinkChannelMask(mChannelMask)) { 2206 LOG_ALWAYS_FATAL("HAL channel mask %#x not supported for mixed output", 2207 mChannelMask); 2208 } 2209 mChannelCount = audio_channel_count_from_out_mask(mChannelMask); 2210 2211 // Get actual HAL format. 2212 mHALFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 2213 // Get format from the shim, which will be different than the HAL format 2214 // if playing compressed audio over HDMI passthrough. 2215 mFormat = mOutput->getFormat(); 2216 if (!audio_is_valid_format(mFormat)) { 2217 LOG_ALWAYS_FATAL("HAL format %#x not valid for output", mFormat); 2218 } 2219 if ((mType == MIXER || mType == DUPLICATING) 2220 && !isValidPcmSinkFormat(mFormat)) { 2221 LOG_FATAL("HAL format %#x not supported for mixed output", 2222 mFormat); 2223 } 2224 mFrameSize = mOutput->getFrameSize(); 2225 mBufferSize = mOutput->stream->common.get_buffer_size(&mOutput->stream->common); 2226 mFrameCount = mBufferSize / mFrameSize; 2227 if (mFrameCount & 15) { 2228 ALOGW("HAL output buffer size is %u frames but AudioMixer requires multiples of 16 frames", 2229 mFrameCount); 2230 } 2231 2232 if ((mOutput->flags & AUDIO_OUTPUT_FLAG_NON_BLOCKING) && 2233 (mOutput->stream->set_callback != NULL)) { 2234 if (mOutput->stream->set_callback(mOutput->stream, 2235 AudioFlinger::PlaybackThread::asyncCallback, this) == 0) { 2236 mUseAsyncWrite = true; 2237 mCallbackThread = new AudioFlinger::AsyncCallbackThread(this); 2238 } 2239 } 2240 2241 mHwSupportsPause = false; 2242 if (mOutput->flags & AUDIO_OUTPUT_FLAG_DIRECT) { 2243 if (mOutput->stream->pause != NULL) { 2244 if (mOutput->stream->resume != NULL) { 2245 mHwSupportsPause = true; 2246 } else { 2247 ALOGW("direct output implements pause but not resume"); 2248 } 2249 } else if (mOutput->stream->resume != NULL) { 2250 ALOGW("direct output implements resume but not pause"); 2251 } 2252 } 2253 if (!mHwSupportsPause && mOutput->flags & AUDIO_OUTPUT_FLAG_HW_AV_SYNC) { 2254 LOG_ALWAYS_FATAL("HW_AV_SYNC requested but HAL does not implement pause and resume"); 2255 } 2256 2257 if (mType == DUPLICATING && mMixerBufferEnabled && mEffectBufferEnabled) { 2258 // For best precision, we use float instead of the associated output 2259 // device format (typically PCM 16 bit). 2260 2261 mFormat = AUDIO_FORMAT_PCM_FLOAT; 2262 mFrameSize = mChannelCount * audio_bytes_per_sample(mFormat); 2263 mBufferSize = mFrameSize * mFrameCount; 2264 2265 // TODO: We currently use the associated output device channel mask and sample rate. 2266 // (1) Perhaps use the ORed channel mask of all downstream MixerThreads 2267 // (if a valid mask) to avoid premature downmix. 2268 // (2) Perhaps use the maximum sample rate of all downstream MixerThreads 2269 // instead of the output device sample rate to avoid loss of high frequency information. 2270 // This may need to be updated as MixerThread/OutputTracks are added and not here. 2271 } 2272 2273 // Calculate size of normal sink buffer relative to the HAL output buffer size 2274 double multiplier = 1.0; 2275 if (mType == MIXER && (kUseFastMixer == FastMixer_Static || 2276 kUseFastMixer == FastMixer_Dynamic)) { 2277 size_t minNormalFrameCount = (kMinNormalSinkBufferSizeMs * mSampleRate) / 1000; 2278 size_t maxNormalFrameCount = (kMaxNormalSinkBufferSizeMs * mSampleRate) / 1000; 2279 // round up minimum and round down maximum to nearest 16 frames to satisfy AudioMixer 2280 minNormalFrameCount = (minNormalFrameCount + 15) & ~15; 2281 maxNormalFrameCount = maxNormalFrameCount & ~15; 2282 if (maxNormalFrameCount < minNormalFrameCount) { 2283 maxNormalFrameCount = minNormalFrameCount; 2284 } 2285 multiplier = (double) minNormalFrameCount / (double) mFrameCount; 2286 if (multiplier <= 1.0) { 2287 multiplier = 1.0; 2288 } else if (multiplier <= 2.0) { 2289 if (2 * mFrameCount <= maxNormalFrameCount) { 2290 multiplier = 2.0; 2291 } else { 2292 multiplier = (double) maxNormalFrameCount / (double) mFrameCount; 2293 } 2294 } else { 2295 // prefer an even multiplier, for compatibility with doubling of fast tracks due to HAL 2296 // SRC (it would be unusual for the normal sink buffer size to not be a multiple of fast 2297 // track, but we sometimes have to do this to satisfy the maximum frame count 2298 // constraint) 2299 // FIXME this rounding up should not be done if no HAL SRC 2300 uint32_t truncMult = (uint32_t) multiplier; 2301 if ((truncMult & 1)) { 2302 if ((truncMult + 1) * mFrameCount <= maxNormalFrameCount) { 2303 ++truncMult; 2304 } 2305 } 2306 multiplier = (double) truncMult; 2307 } 2308 } 2309 mNormalFrameCount = multiplier * mFrameCount; 2310 // round up to nearest 16 frames to satisfy AudioMixer 2311 if (mType == MIXER || mType == DUPLICATING) { 2312 mNormalFrameCount = (mNormalFrameCount + 15) & ~15; 2313 } 2314 ALOGI("HAL output buffer size %u frames, normal sink buffer size %u frames", mFrameCount, 2315 mNormalFrameCount); 2316 2317 // Check if we want to throttle the processing to no more than 2x normal rate 2318 mThreadThrottle = property_get_bool("af.thread.throttle", true /* default_value */); 2319 mThreadThrottleTimeMs = 0; 2320 mThreadThrottleEndMs = 0; 2321 mHalfBufferMs = mNormalFrameCount * 1000 / (2 * mSampleRate); 2322 2323 // mSinkBuffer is the sink buffer. Size is always multiple-of-16 frames. 2324 // Originally this was int16_t[] array, need to remove legacy implications. 2325 free(mSinkBuffer); 2326 mSinkBuffer = NULL; 2327 // For sink buffer size, we use the frame size from the downstream sink to avoid problems 2328 // with non PCM formats for compressed music, e.g. AAC, and Offload threads. 2329 const size_t sinkBufferSize = mNormalFrameCount * mFrameSize; 2330 (void)posix_memalign(&mSinkBuffer, 32, sinkBufferSize); 2331 2332 // We resize the mMixerBuffer according to the requirements of the sink buffer which 2333 // drives the output. 2334 free(mMixerBuffer); 2335 mMixerBuffer = NULL; 2336 if (mMixerBufferEnabled) { 2337 mMixerBufferFormat = AUDIO_FORMAT_PCM_FLOAT; // also valid: AUDIO_FORMAT_PCM_16_BIT. 2338 mMixerBufferSize = mNormalFrameCount * mChannelCount 2339 * audio_bytes_per_sample(mMixerBufferFormat); 2340 (void)posix_memalign(&mMixerBuffer, 32, mMixerBufferSize); 2341 } 2342 free(mEffectBuffer); 2343 mEffectBuffer = NULL; 2344 if (mEffectBufferEnabled) { 2345 mEffectBufferFormat = AUDIO_FORMAT_PCM_16_BIT; // Note: Effects support 16b only 2346 mEffectBufferSize = mNormalFrameCount * mChannelCount 2347 * audio_bytes_per_sample(mEffectBufferFormat); 2348 (void)posix_memalign(&mEffectBuffer, 32, mEffectBufferSize); 2349 } 2350 2351 // force reconfiguration of effect chains and engines to take new buffer size and audio 2352 // parameters into account 2353 // Note that mLock is not held when readOutputParameters_l() is called from the constructor 2354 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 2355 // matter. 2356 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 2357 Vector< sp<EffectChain> > effectChains = mEffectChains; 2358 for (size_t i = 0; i < effectChains.size(); i ++) { 2359 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 2360 } 2361} 2362 2363 2364status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames) 2365{ 2366 if (halFrames == NULL || dspFrames == NULL) { 2367 return BAD_VALUE; 2368 } 2369 Mutex::Autolock _l(mLock); 2370 if (initCheck() != NO_ERROR) { 2371 return INVALID_OPERATION; 2372 } 2373 size_t framesWritten = mBytesWritten / mFrameSize; 2374 *halFrames = framesWritten; 2375 2376 if (isSuspended()) { 2377 // return an estimation of rendered frames when the output is suspended 2378 size_t latencyFrames = (latency_l() * mSampleRate) / 1000; 2379 *dspFrames = framesWritten >= latencyFrames ? framesWritten - latencyFrames : 0; 2380 return NO_ERROR; 2381 } else { 2382 status_t status; 2383 uint32_t frames; 2384 status = mOutput->getRenderPosition(&frames); 2385 *dspFrames = (size_t)frames; 2386 return status; 2387 } 2388} 2389 2390uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) const 2391{ 2392 Mutex::Autolock _l(mLock); 2393 uint32_t result = 0; 2394 if (getEffectChain_l(sessionId) != 0) { 2395 result = EFFECT_SESSION; 2396 } 2397 2398 for (size_t i = 0; i < mTracks.size(); ++i) { 2399 sp<Track> track = mTracks[i]; 2400 if (sessionId == track->sessionId() && !track->isInvalid()) { 2401 result |= TRACK_SESSION; 2402 break; 2403 } 2404 } 2405 2406 return result; 2407} 2408 2409uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 2410{ 2411 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 2412 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 2413 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 2414 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 2415 } 2416 for (size_t i = 0; i < mTracks.size(); i++) { 2417 sp<Track> track = mTracks[i]; 2418 if (sessionId == track->sessionId() && !track->isInvalid()) { 2419 return AudioSystem::getStrategyForStream(track->streamType()); 2420 } 2421 } 2422 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 2423} 2424 2425 2426AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 2427{ 2428 Mutex::Autolock _l(mLock); 2429 return mOutput; 2430} 2431 2432AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 2433{ 2434 Mutex::Autolock _l(mLock); 2435 AudioStreamOut *output = mOutput; 2436 mOutput = NULL; 2437 // FIXME FastMixer might also have a raw ptr to mOutputSink; 2438 // must push a NULL and wait for ack 2439 mOutputSink.clear(); 2440 mPipeSink.clear(); 2441 mNormalSink.clear(); 2442 return output; 2443} 2444 2445// this method must always be called either with ThreadBase mLock held or inside the thread loop 2446audio_stream_t* AudioFlinger::PlaybackThread::stream() const 2447{ 2448 if (mOutput == NULL) { 2449 return NULL; 2450 } 2451 return &mOutput->stream->common; 2452} 2453 2454uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() const 2455{ 2456 return (uint32_t)((uint32_t)((mNormalFrameCount * 1000) / mSampleRate) * 1000); 2457} 2458 2459status_t AudioFlinger::PlaybackThread::setSyncEvent(const sp<SyncEvent>& event) 2460{ 2461 if (!isValidSyncEvent(event)) { 2462 return BAD_VALUE; 2463 } 2464 2465 Mutex::Autolock _l(mLock); 2466 2467 for (size_t i = 0; i < mTracks.size(); ++i) { 2468 sp<Track> track = mTracks[i]; 2469 if (event->triggerSession() == track->sessionId()) { 2470 (void) track->setSyncEvent(event); 2471 return NO_ERROR; 2472 } 2473 } 2474 2475 return NAME_NOT_FOUND; 2476} 2477 2478bool AudioFlinger::PlaybackThread::isValidSyncEvent(const sp<SyncEvent>& event) const 2479{ 2480 return event->type() == AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE; 2481} 2482 2483void AudioFlinger::PlaybackThread::threadLoop_removeTracks( 2484 const Vector< sp<Track> >& tracksToRemove) 2485{ 2486 size_t count = tracksToRemove.size(); 2487 if (count > 0) { 2488 for (size_t i = 0 ; i < count ; i++) { 2489 const sp<Track>& track = tracksToRemove.itemAt(i); 2490 if (track->isExternalTrack()) { 2491 AudioSystem::stopOutput(mId, track->streamType(), 2492 (audio_session_t)track->sessionId()); 2493#ifdef ADD_BATTERY_DATA 2494 // to track the speaker usage 2495 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 2496#endif 2497 if (track->isTerminated()) { 2498 AudioSystem::releaseOutput(mId, track->streamType(), 2499 (audio_session_t)track->sessionId()); 2500 } 2501 } 2502 } 2503 } 2504} 2505 2506void AudioFlinger::PlaybackThread::checkSilentMode_l() 2507{ 2508 if (!mMasterMute) { 2509 char value[PROPERTY_VALUE_MAX]; 2510 if (property_get("ro.audio.silent", value, "0") > 0) { 2511 char *endptr; 2512 unsigned long ul = strtoul(value, &endptr, 0); 2513 if (*endptr == '\0' && ul != 0) { 2514 ALOGD("Silence is golden"); 2515 // The setprop command will not allow a property to be changed after 2516 // the first time it is set, so we don't have to worry about un-muting. 2517 setMasterMute_l(true); 2518 } 2519 } 2520 } 2521} 2522 2523// shared by MIXER and DIRECT, overridden by DUPLICATING 2524ssize_t AudioFlinger::PlaybackThread::threadLoop_write() 2525{ 2526 // FIXME rewrite to reduce number of system calls 2527 mLastWriteTime = systemTime(); 2528 mInWrite = true; 2529 ssize_t bytesWritten; 2530 const size_t offset = mCurrentWriteLength - mBytesRemaining; 2531 2532 // If an NBAIO sink is present, use it to write the normal mixer's submix 2533 if (mNormalSink != 0) { 2534 2535 const size_t count = mBytesRemaining / mFrameSize; 2536 2537 ATRACE_BEGIN("write"); 2538 // update the setpoint when AudioFlinger::mScreenState changes 2539 uint32_t screenState = AudioFlinger::mScreenState; 2540 if (screenState != mScreenState) { 2541 mScreenState = screenState; 2542 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 2543 if (pipe != NULL) { 2544 pipe->setAvgFrames((mScreenState & 1) ? 2545 (pipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2546 } 2547 } 2548 ssize_t framesWritten = mNormalSink->write((char *)mSinkBuffer + offset, count); 2549 ATRACE_END(); 2550 if (framesWritten > 0) { 2551 bytesWritten = framesWritten * mFrameSize; 2552 } else { 2553 bytesWritten = framesWritten; 2554 } 2555 // otherwise use the HAL / AudioStreamOut directly 2556 } else { 2557 // Direct output and offload threads 2558 2559 if (mUseAsyncWrite) { 2560 ALOGW_IF(mWriteAckSequence & 1, "threadLoop_write(): out of sequence write request"); 2561 mWriteAckSequence += 2; 2562 mWriteAckSequence |= 1; 2563 ALOG_ASSERT(mCallbackThread != 0); 2564 mCallbackThread->setWriteBlocked(mWriteAckSequence); 2565 } 2566 // FIXME We should have an implementation of timestamps for direct output threads. 2567 // They are used e.g for multichannel PCM playback over HDMI. 2568 bytesWritten = mOutput->write((char *)mSinkBuffer + offset, mBytesRemaining); 2569 if (mUseAsyncWrite && 2570 ((bytesWritten < 0) || (bytesWritten == (ssize_t)mBytesRemaining))) { 2571 // do not wait for async callback in case of error of full write 2572 mWriteAckSequence &= ~1; 2573 ALOG_ASSERT(mCallbackThread != 0); 2574 mCallbackThread->setWriteBlocked(mWriteAckSequence); 2575 } 2576 } 2577 2578 mNumWrites++; 2579 mInWrite = false; 2580 mStandby = false; 2581 return bytesWritten; 2582} 2583 2584void AudioFlinger::PlaybackThread::threadLoop_drain() 2585{ 2586 if (mOutput->stream->drain) { 2587 ALOGV("draining %s", (mMixerStatus == MIXER_DRAIN_TRACK) ? "early" : "full"); 2588 if (mUseAsyncWrite) { 2589 ALOGW_IF(mDrainSequence & 1, "threadLoop_drain(): out of sequence drain request"); 2590 mDrainSequence |= 1; 2591 ALOG_ASSERT(mCallbackThread != 0); 2592 mCallbackThread->setDraining(mDrainSequence); 2593 } 2594 mOutput->stream->drain(mOutput->stream, 2595 (mMixerStatus == MIXER_DRAIN_TRACK) ? AUDIO_DRAIN_EARLY_NOTIFY 2596 : AUDIO_DRAIN_ALL); 2597 } 2598} 2599 2600void AudioFlinger::PlaybackThread::threadLoop_exit() 2601{ 2602 { 2603 Mutex::Autolock _l(mLock); 2604 for (size_t i = 0; i < mTracks.size(); i++) { 2605 sp<Track> track = mTracks[i]; 2606 track->invalidate(); 2607 } 2608 } 2609} 2610 2611/* 2612The derived values that are cached: 2613 - mSinkBufferSize from frame count * frame size 2614 - mActiveSleepTimeUs from activeSleepTimeUs() 2615 - mIdleSleepTimeUs from idleSleepTimeUs() 2616 - mStandbyDelayNs from mActiveSleepTimeUs (DIRECT only) or forced to at least 2617 kDefaultStandbyTimeInNsecs when connected to an A2DP device. 2618 - maxPeriod from frame count and sample rate (MIXER only) 2619 2620The parameters that affect these derived values are: 2621 - frame count 2622 - frame size 2623 - sample rate 2624 - device type: A2DP or not 2625 - device latency 2626 - format: PCM or not 2627 - active sleep time 2628 - idle sleep time 2629*/ 2630 2631void AudioFlinger::PlaybackThread::cacheParameters_l() 2632{ 2633 mSinkBufferSize = mNormalFrameCount * mFrameSize; 2634 mActiveSleepTimeUs = activeSleepTimeUs(); 2635 mIdleSleepTimeUs = idleSleepTimeUs(); 2636 2637 // make sure standby delay is not too short when connected to an A2DP sink to avoid 2638 // truncating audio when going to standby. 2639 mStandbyDelayNs = AudioFlinger::mStandbyTimeInNsecs; 2640 if ((mOutDevice & AUDIO_DEVICE_OUT_ALL_A2DP) != 0) { 2641 if (mStandbyDelayNs < kDefaultStandbyTimeInNsecs) { 2642 mStandbyDelayNs = kDefaultStandbyTimeInNsecs; 2643 } 2644 } 2645} 2646 2647void AudioFlinger::PlaybackThread::invalidateTracks(audio_stream_type_t streamType) 2648{ 2649 ALOGV("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 2650 this, streamType, mTracks.size()); 2651 Mutex::Autolock _l(mLock); 2652 2653 size_t size = mTracks.size(); 2654 for (size_t i = 0; i < size; i++) { 2655 sp<Track> t = mTracks[i]; 2656 if (t->streamType() == streamType && t->isExternalTrack()) { 2657 t->invalidate(); 2658 } 2659 } 2660} 2661 2662status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 2663{ 2664 int session = chain->sessionId(); 2665 int16_t* buffer = reinterpret_cast<int16_t*>(mEffectBufferEnabled 2666 ? mEffectBuffer : mSinkBuffer); 2667 bool ownsBuffer = false; 2668 2669 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 2670 if (session > 0) { 2671 // Only one effect chain can be present in direct output thread and it uses 2672 // the sink buffer as input 2673 if (mType != DIRECT) { 2674 size_t numSamples = mNormalFrameCount * mChannelCount; 2675 buffer = new int16_t[numSamples]; 2676 memset(buffer, 0, numSamples * sizeof(int16_t)); 2677 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 2678 ownsBuffer = true; 2679 } 2680 2681 // Attach all tracks with same session ID to this chain. 2682 for (size_t i = 0; i < mTracks.size(); ++i) { 2683 sp<Track> track = mTracks[i]; 2684 if (session == track->sessionId()) { 2685 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), 2686 buffer); 2687 track->setMainBuffer(buffer); 2688 chain->incTrackCnt(); 2689 } 2690 } 2691 2692 // indicate all active tracks in the chain 2693 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 2694 sp<Track> track = mActiveTracks[i].promote(); 2695 if (track == 0) { 2696 continue; 2697 } 2698 if (session == track->sessionId()) { 2699 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 2700 chain->incActiveTrackCnt(); 2701 } 2702 } 2703 } 2704 chain->setThread(this); 2705 chain->setInBuffer(buffer, ownsBuffer); 2706 chain->setOutBuffer(reinterpret_cast<int16_t*>(mEffectBufferEnabled 2707 ? mEffectBuffer : mSinkBuffer)); 2708 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 2709 // chains list in order to be processed last as it contains output stage effects 2710 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 2711 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 2712 // after track specific effects and before output stage 2713 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 2714 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 2715 // Effect chain for other sessions are inserted at beginning of effect 2716 // chains list to be processed before output mix effects. Relative order between other 2717 // sessions is not important 2718 size_t size = mEffectChains.size(); 2719 size_t i = 0; 2720 for (i = 0; i < size; i++) { 2721 if (mEffectChains[i]->sessionId() < session) { 2722 break; 2723 } 2724 } 2725 mEffectChains.insertAt(chain, i); 2726 checkSuspendOnAddEffectChain_l(chain); 2727 2728 return NO_ERROR; 2729} 2730 2731size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 2732{ 2733 int session = chain->sessionId(); 2734 2735 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 2736 2737 for (size_t i = 0; i < mEffectChains.size(); i++) { 2738 if (chain == mEffectChains[i]) { 2739 mEffectChains.removeAt(i); 2740 // detach all active tracks from the chain 2741 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 2742 sp<Track> track = mActiveTracks[i].promote(); 2743 if (track == 0) { 2744 continue; 2745 } 2746 if (session == track->sessionId()) { 2747 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 2748 chain.get(), session); 2749 chain->decActiveTrackCnt(); 2750 } 2751 } 2752 2753 // detach all tracks with same session ID from this chain 2754 for (size_t i = 0; i < mTracks.size(); ++i) { 2755 sp<Track> track = mTracks[i]; 2756 if (session == track->sessionId()) { 2757 track->setMainBuffer(reinterpret_cast<int16_t*>(mSinkBuffer)); 2758 chain->decTrackCnt(); 2759 } 2760 } 2761 break; 2762 } 2763 } 2764 return mEffectChains.size(); 2765} 2766 2767status_t AudioFlinger::PlaybackThread::attachAuxEffect( 2768 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2769{ 2770 Mutex::Autolock _l(mLock); 2771 return attachAuxEffect_l(track, EffectId); 2772} 2773 2774status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 2775 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2776{ 2777 status_t status = NO_ERROR; 2778 2779 if (EffectId == 0) { 2780 track->setAuxBuffer(0, NULL); 2781 } else { 2782 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 2783 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 2784 if (effect != 0) { 2785 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 2786 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 2787 } else { 2788 status = INVALID_OPERATION; 2789 } 2790 } else { 2791 status = BAD_VALUE; 2792 } 2793 } 2794 return status; 2795} 2796 2797void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 2798{ 2799 for (size_t i = 0; i < mTracks.size(); ++i) { 2800 sp<Track> track = mTracks[i]; 2801 if (track->auxEffectId() == effectId) { 2802 attachAuxEffect_l(track, 0); 2803 } 2804 } 2805} 2806 2807bool AudioFlinger::PlaybackThread::threadLoop() 2808{ 2809 Vector< sp<Track> > tracksToRemove; 2810 2811 mStandbyTimeNs = systemTime(); 2812 2813 // MIXER 2814 nsecs_t lastWarning = 0; 2815 2816 // DUPLICATING 2817 // FIXME could this be made local to while loop? 2818 writeFrames = 0; 2819 2820 int lastGeneration = 0; 2821 2822 cacheParameters_l(); 2823 mSleepTimeUs = mIdleSleepTimeUs; 2824 2825 if (mType == MIXER) { 2826 sleepTimeShift = 0; 2827 } 2828 2829 CpuStats cpuStats; 2830 const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid())); 2831 2832 acquireWakeLock(); 2833 2834 // mNBLogWriter->log can only be called while thread mutex mLock is held. 2835 // So if you need to log when mutex is unlocked, set logString to a non-NULL string, 2836 // and then that string will be logged at the next convenient opportunity. 2837 const char *logString = NULL; 2838 2839 checkSilentMode_l(); 2840 2841 while (!exitPending()) 2842 { 2843 cpuStats.sample(myName); 2844 2845 Vector< sp<EffectChain> > effectChains; 2846 2847 { // scope for mLock 2848 2849 Mutex::Autolock _l(mLock); 2850 2851 processConfigEvents_l(); 2852 2853 if (logString != NULL) { 2854 mNBLogWriter->logTimestamp(); 2855 mNBLogWriter->log(logString); 2856 logString = NULL; 2857 } 2858 2859 // Gather the framesReleased counters for all active tracks, 2860 // and associate with the sink frames written out. We need 2861 // this to convert the sink timestamp to the track timestamp. 2862 if (mNormalSink != 0) { 2863 bool updateTracks = true; 2864 bool cacheTimestamp = false; 2865 AudioTimestamp timeStamp; 2866 // FIXME: Use a 64 bit mNormalSink->framesWritten() counter. 2867 // At this time, we must always use cached timestamps even when 2868 // going through mPipeSink (which is non-blocking). The reason is that 2869 // the track may be removed from the active list for many hours and 2870 // the mNormalSink->framesWritten() will wrap making the linear 2871 // mapping fail. 2872 // 2873 // (Also mAudioTrackServerProxy->framesReleased() needs to be 2874 // updated to 64 bits for 64 bit frame position.) 2875 // 2876 if (true /* see comment above, should be: mNormalSink == mOutputSink */) { 2877 // If we use a hardware device, we must cache the sink timestamp now. 2878 // hardware devices can block timestamp access during data writes. 2879 if (mNormalSink->getTimestamp(timeStamp) == NO_ERROR) { 2880 cacheTimestamp = true; 2881 } else { 2882 updateTracks = false; 2883 } 2884 } 2885 if (updateTracks) { 2886 // sinkFramesWritten for non-offloaded tracks are contiguous 2887 // even after standby() is called. This is useful for the track frame 2888 // to sink frame mapping. 2889 const uint32_t sinkFramesWritten = mNormalSink->framesWritten(); 2890 const size_t size = mActiveTracks.size(); 2891 for (size_t i = 0; i < size; ++i) { 2892 sp<Track> t = mActiveTracks[i].promote(); 2893 if (t != 0 && !t->isFastTrack()) { 2894 t->updateTrackFrameInfo( 2895 t->mAudioTrackServerProxy->framesReleased(), 2896 sinkFramesWritten, 2897 cacheTimestamp ? &timeStamp : NULL); 2898 } 2899 } 2900 } 2901 } 2902 2903 saveOutputTracks(); 2904 if (mSignalPending) { 2905 // A signal was raised while we were unlocked 2906 mSignalPending = false; 2907 } else if (waitingAsyncCallback_l()) { 2908 if (exitPending()) { 2909 break; 2910 } 2911 bool released = false; 2912 // The following works around a bug in the offload driver. Ideally we would release 2913 // the wake lock every time, but that causes the last offload buffer(s) to be 2914 // dropped while the device is on battery, so we need to hold a wake lock during 2915 // the drain phase. 2916 if (mBytesRemaining && !(mDrainSequence & 1)) { 2917 releaseWakeLock_l(); 2918 released = true; 2919 } 2920 mWakeLockUids.clear(); 2921 mActiveTracksGeneration++; 2922 ALOGV("wait async completion"); 2923 mWaitWorkCV.wait(mLock); 2924 ALOGV("async completion/wake"); 2925 if (released) { 2926 acquireWakeLock_l(); 2927 } 2928 mStandbyTimeNs = systemTime() + mStandbyDelayNs; 2929 mSleepTimeUs = 0; 2930 2931 continue; 2932 } 2933 if ((!mActiveTracks.size() && systemTime() > mStandbyTimeNs) || 2934 isSuspended()) { 2935 // put audio hardware into standby after short delay 2936 if (shouldStandby_l()) { 2937 2938 threadLoop_standby(); 2939 2940 mStandby = true; 2941 } 2942 2943 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2944 // we're about to wait, flush the binder command buffer 2945 IPCThreadState::self()->flushCommands(); 2946 2947 clearOutputTracks(); 2948 2949 if (exitPending()) { 2950 break; 2951 } 2952 2953 releaseWakeLock_l(); 2954 mWakeLockUids.clear(); 2955 mActiveTracksGeneration++; 2956 // wait until we have something to do... 2957 ALOGV("%s going to sleep", myName.string()); 2958 mWaitWorkCV.wait(mLock); 2959 ALOGV("%s waking up", myName.string()); 2960 acquireWakeLock_l(); 2961 2962 mMixerStatus = MIXER_IDLE; 2963 mMixerStatusIgnoringFastTracks = MIXER_IDLE; 2964 mBytesWritten = 0; 2965 mBytesRemaining = 0; 2966 checkSilentMode_l(); 2967 2968 mStandbyTimeNs = systemTime() + mStandbyDelayNs; 2969 mSleepTimeUs = mIdleSleepTimeUs; 2970 if (mType == MIXER) { 2971 sleepTimeShift = 0; 2972 } 2973 2974 continue; 2975 } 2976 } 2977 // mMixerStatusIgnoringFastTracks is also updated internally 2978 mMixerStatus = prepareTracks_l(&tracksToRemove); 2979 2980 // compare with previously applied list 2981 if (lastGeneration != mActiveTracksGeneration) { 2982 // update wakelock 2983 updateWakeLockUids_l(mWakeLockUids); 2984 lastGeneration = mActiveTracksGeneration; 2985 } 2986 2987 // prevent any changes in effect chain list and in each effect chain 2988 // during mixing and effect process as the audio buffers could be deleted 2989 // or modified if an effect is created or deleted 2990 lockEffectChains_l(effectChains); 2991 } // mLock scope ends 2992 2993 if (mBytesRemaining == 0) { 2994 mCurrentWriteLength = 0; 2995 if (mMixerStatus == MIXER_TRACKS_READY) { 2996 // threadLoop_mix() sets mCurrentWriteLength 2997 threadLoop_mix(); 2998 } else if ((mMixerStatus != MIXER_DRAIN_TRACK) 2999 && (mMixerStatus != MIXER_DRAIN_ALL)) { 3000 // threadLoop_sleepTime sets mSleepTimeUs to 0 if data 3001 // must be written to HAL 3002 threadLoop_sleepTime(); 3003 if (mSleepTimeUs == 0) { 3004 mCurrentWriteLength = mSinkBufferSize; 3005 } 3006 } 3007 // Either threadLoop_mix() or threadLoop_sleepTime() should have set 3008 // mMixerBuffer with data if mMixerBufferValid is true and mSleepTimeUs == 0. 3009 // Merge mMixerBuffer data into mEffectBuffer (if any effects are valid) 3010 // or mSinkBuffer (if there are no effects). 3011 // 3012 // This is done pre-effects computation; if effects change to 3013 // support higher precision, this needs to move. 3014 // 3015 // mMixerBufferValid is only set true by MixerThread::prepareTracks_l(). 3016 // TODO use mSleepTimeUs == 0 as an additional condition. 3017 if (mMixerBufferValid) { 3018 void *buffer = mEffectBufferValid ? mEffectBuffer : mSinkBuffer; 3019 audio_format_t format = mEffectBufferValid ? mEffectBufferFormat : mFormat; 3020 3021 // mono blend occurs for mixer threads only (not direct or offloaded) 3022 // and is handled here if we're going directly to the sink. 3023 if (requireMonoBlend() && !mEffectBufferValid) { 3024 mono_blend(mMixerBuffer, mMixerBufferFormat, mChannelCount, mNormalFrameCount, 3025 true /*limit*/); 3026 } 3027 3028 memcpy_by_audio_format(buffer, format, mMixerBuffer, mMixerBufferFormat, 3029 mNormalFrameCount * mChannelCount); 3030 } 3031 3032 mBytesRemaining = mCurrentWriteLength; 3033 if (isSuspended()) { 3034 mSleepTimeUs = suspendSleepTimeUs(); 3035 // simulate write to HAL when suspended 3036 mBytesWritten += mSinkBufferSize; 3037 mBytesRemaining = 0; 3038 } 3039 3040 // only process effects if we're going to write 3041 if (mSleepTimeUs == 0 && mType != OFFLOAD) { 3042 for (size_t i = 0; i < effectChains.size(); i ++) { 3043 effectChains[i]->process_l(); 3044 } 3045 } 3046 } 3047 // Process effect chains for offloaded thread even if no audio 3048 // was read from audio track: process only updates effect state 3049 // and thus does have to be synchronized with audio writes but may have 3050 // to be called while waiting for async write callback 3051 if (mType == OFFLOAD) { 3052 for (size_t i = 0; i < effectChains.size(); i ++) { 3053 effectChains[i]->process_l(); 3054 } 3055 } 3056 3057 // Only if the Effects buffer is enabled and there is data in the 3058 // Effects buffer (buffer valid), we need to 3059 // copy into the sink buffer. 3060 // TODO use mSleepTimeUs == 0 as an additional condition. 3061 if (mEffectBufferValid) { 3062 //ALOGV("writing effect buffer to sink buffer format %#x", mFormat); 3063 3064 if (requireMonoBlend()) { 3065 mono_blend(mEffectBuffer, mEffectBufferFormat, mChannelCount, mNormalFrameCount, 3066 true /*limit*/); 3067 } 3068 3069 memcpy_by_audio_format(mSinkBuffer, mFormat, mEffectBuffer, mEffectBufferFormat, 3070 mNormalFrameCount * mChannelCount); 3071 } 3072 3073 // enable changes in effect chain 3074 unlockEffectChains(effectChains); 3075 3076 if (!waitingAsyncCallback()) { 3077 // mSleepTimeUs == 0 means we must write to audio hardware 3078 if (mSleepTimeUs == 0) { 3079 ssize_t ret = 0; 3080 if (mBytesRemaining) { 3081 ret = threadLoop_write(); 3082 if (ret < 0) { 3083 mBytesRemaining = 0; 3084 } else { 3085 mBytesWritten += ret; 3086 mBytesRemaining -= ret; 3087 } 3088 } else if ((mMixerStatus == MIXER_DRAIN_TRACK) || 3089 (mMixerStatus == MIXER_DRAIN_ALL)) { 3090 threadLoop_drain(); 3091 } 3092 if (mType == MIXER && !mStandby) { 3093 // write blocked detection 3094 nsecs_t now = systemTime(); 3095 nsecs_t delta = now - mLastWriteTime; 3096 if (delta > maxPeriod) { 3097 mNumDelayedWrites++; 3098 if ((now - lastWarning) > kWarningThrottleNs) { 3099 ATRACE_NAME("underrun"); 3100 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 3101 ns2ms(delta), mNumDelayedWrites, this); 3102 lastWarning = now; 3103 } 3104 } 3105 3106 if (mThreadThrottle 3107 && mMixerStatus == MIXER_TRACKS_READY // we are mixing (active tracks) 3108 && ret > 0) { // we wrote something 3109 // Limit MixerThread data processing to no more than twice the 3110 // expected processing rate. 3111 // 3112 // This helps prevent underruns with NuPlayer and other applications 3113 // which may set up buffers that are close to the minimum size, or use 3114 // deep buffers, and rely on a double-buffering sleep strategy to fill. 3115 // 3116 // The throttle smooths out sudden large data drains from the device, 3117 // e.g. when it comes out of standby, which often causes problems with 3118 // (1) mixer threads without a fast mixer (which has its own warm-up) 3119 // (2) minimum buffer sized tracks (even if the track is full, 3120 // the app won't fill fast enough to handle the sudden draw). 3121 3122 const int32_t deltaMs = delta / 1000000; 3123 const int32_t throttleMs = mHalfBufferMs - deltaMs; 3124 if ((signed)mHalfBufferMs >= throttleMs && throttleMs > 0) { 3125 usleep(throttleMs * 1000); 3126 // notify of throttle start on verbose log 3127 ALOGV_IF(mThreadThrottleEndMs == mThreadThrottleTimeMs, 3128 "mixer(%p) throttle begin:" 3129 " ret(%zd) deltaMs(%d) requires sleep %d ms", 3130 this, ret, deltaMs, throttleMs); 3131 mThreadThrottleTimeMs += throttleMs; 3132 } else { 3133 uint32_t diff = mThreadThrottleTimeMs - mThreadThrottleEndMs; 3134 if (diff > 0) { 3135 // notify of throttle end on debug log 3136 ALOGD("mixer(%p) throttle end: throttle time(%u)", this, diff); 3137 mThreadThrottleEndMs = mThreadThrottleTimeMs; 3138 } 3139 } 3140 } 3141 } 3142 3143 } else { 3144 ATRACE_BEGIN("sleep"); 3145 usleep(mSleepTimeUs); 3146 ATRACE_END(); 3147 } 3148 } 3149 3150 // Finally let go of removed track(s), without the lock held 3151 // since we can't guarantee the destructors won't acquire that 3152 // same lock. This will also mutate and push a new fast mixer state. 3153 threadLoop_removeTracks(tracksToRemove); 3154 tracksToRemove.clear(); 3155 3156 // FIXME I don't understand the need for this here; 3157 // it was in the original code but maybe the 3158 // assignment in saveOutputTracks() makes this unnecessary? 3159 clearOutputTracks(); 3160 3161 // Effect chains will be actually deleted here if they were removed from 3162 // mEffectChains list during mixing or effects processing 3163 effectChains.clear(); 3164 3165 // FIXME Note that the above .clear() is no longer necessary since effectChains 3166 // is now local to this block, but will keep it for now (at least until merge done). 3167 } 3168 3169 threadLoop_exit(); 3170 3171 if (!mStandby) { 3172 threadLoop_standby(); 3173 mStandby = true; 3174 } 3175 3176 releaseWakeLock(); 3177 mWakeLockUids.clear(); 3178 mActiveTracksGeneration++; 3179 3180 ALOGV("Thread %p type %d exiting", this, mType); 3181 return false; 3182} 3183 3184// removeTracks_l() must be called with ThreadBase::mLock held 3185void AudioFlinger::PlaybackThread::removeTracks_l(const Vector< sp<Track> >& tracksToRemove) 3186{ 3187 size_t count = tracksToRemove.size(); 3188 if (count > 0) { 3189 for (size_t i=0 ; i<count ; i++) { 3190 const sp<Track>& track = tracksToRemove.itemAt(i); 3191 mActiveTracks.remove(track); 3192 mWakeLockUids.remove(track->uid()); 3193 mActiveTracksGeneration++; 3194 ALOGV("removeTracks_l removing track on session %d", track->sessionId()); 3195 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 3196 if (chain != 0) { 3197 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), 3198 track->sessionId()); 3199 chain->decActiveTrackCnt(); 3200 } 3201 if (track->isTerminated()) { 3202 removeTrack_l(track); 3203 } 3204 } 3205 } 3206 3207} 3208 3209status_t AudioFlinger::PlaybackThread::getTimestamp_l(AudioTimestamp& timestamp) 3210{ 3211 if (mNormalSink != 0) { 3212 return mNormalSink->getTimestamp(timestamp); 3213 } 3214 if ((mType == OFFLOAD || mType == DIRECT) 3215 && mOutput != NULL && mOutput->stream->get_presentation_position) { 3216 uint64_t position64; 3217 int ret = mOutput->getPresentationPosition(&position64, ×tamp.mTime); 3218 if (ret == 0) { 3219 timestamp.mPosition = (uint32_t)position64; 3220 return NO_ERROR; 3221 } 3222 } 3223 return INVALID_OPERATION; 3224} 3225 3226status_t AudioFlinger::MixerThread::createAudioPatch_l(const struct audio_patch *patch, 3227 audio_patch_handle_t *handle) 3228{ 3229 // if !&IDLE, holds the FastMixer state to restore after new parameters processed 3230 FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE; 3231 if (mFastMixer != 0) { 3232 FastMixerStateQueue *sq = mFastMixer->sq(); 3233 FastMixerState *state = sq->begin(); 3234 if (!(state->mCommand & FastMixerState::IDLE)) { 3235 previousCommand = state->mCommand; 3236 state->mCommand = FastMixerState::HOT_IDLE; 3237 sq->end(); 3238 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3239 } else { 3240 sq->end(false /*didModify*/); 3241 } 3242 } 3243 status_t status = PlaybackThread::createAudioPatch_l(patch, handle); 3244 3245 if (!(previousCommand & FastMixerState::IDLE)) { 3246 ALOG_ASSERT(mFastMixer != 0); 3247 FastMixerStateQueue *sq = mFastMixer->sq(); 3248 FastMixerState *state = sq->begin(); 3249 ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE); 3250 state->mCommand = previousCommand; 3251 sq->end(); 3252 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3253 } 3254 3255 return status; 3256} 3257 3258status_t AudioFlinger::PlaybackThread::createAudioPatch_l(const struct audio_patch *patch, 3259 audio_patch_handle_t *handle) 3260{ 3261 status_t status = NO_ERROR; 3262 3263 // store new device and send to effects 3264 audio_devices_t type = AUDIO_DEVICE_NONE; 3265 for (unsigned int i = 0; i < patch->num_sinks; i++) { 3266 type |= patch->sinks[i].ext.device.type; 3267 } 3268 3269#ifdef ADD_BATTERY_DATA 3270 // when changing the audio output device, call addBatteryData to notify 3271 // the change 3272 if (mOutDevice != type) { 3273 uint32_t params = 0; 3274 // check whether speaker is on 3275 if (type & AUDIO_DEVICE_OUT_SPEAKER) { 3276 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 3277 } 3278 3279 audio_devices_t deviceWithoutSpeaker 3280 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 3281 // check if any other device (except speaker) is on 3282 if (type & deviceWithoutSpeaker) { 3283 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 3284 } 3285 3286 if (params != 0) { 3287 addBatteryData(params); 3288 } 3289 } 3290#endif 3291 3292 for (size_t i = 0; i < mEffectChains.size(); i++) { 3293 mEffectChains[i]->setDevice_l(type); 3294 } 3295 3296 // mPrevOutDevice is the latest device set by createAudioPatch_l(). It is not set when 3297 // the thread is created so that the first patch creation triggers an ioConfigChanged callback 3298 bool configChanged = mPrevOutDevice != type; 3299 mOutDevice = type; 3300 mPatch = *patch; 3301 3302 if (mOutput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 3303 audio_hw_device_t *hwDevice = mOutput->audioHwDev->hwDevice(); 3304 status = hwDevice->create_audio_patch(hwDevice, 3305 patch->num_sources, 3306 patch->sources, 3307 patch->num_sinks, 3308 patch->sinks, 3309 handle); 3310 } else { 3311 char *address; 3312 if (strcmp(patch->sinks[0].ext.device.address, "") != 0) { 3313 //FIXME: we only support address on first sink with HAL version < 3.0 3314 address = audio_device_address_to_parameter( 3315 patch->sinks[0].ext.device.type, 3316 patch->sinks[0].ext.device.address); 3317 } else { 3318 address = (char *)calloc(1, 1); 3319 } 3320 AudioParameter param = AudioParameter(String8(address)); 3321 free(address); 3322 param.addInt(String8(AUDIO_PARAMETER_STREAM_ROUTING), (int)type); 3323 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3324 param.toString().string()); 3325 *handle = AUDIO_PATCH_HANDLE_NONE; 3326 } 3327 if (configChanged) { 3328 mPrevOutDevice = type; 3329 sendIoConfigEvent_l(AUDIO_OUTPUT_CONFIG_CHANGED); 3330 } 3331 return status; 3332} 3333 3334status_t AudioFlinger::MixerThread::releaseAudioPatch_l(const audio_patch_handle_t handle) 3335{ 3336 // if !&IDLE, holds the FastMixer state to restore after new parameters processed 3337 FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE; 3338 if (mFastMixer != 0) { 3339 FastMixerStateQueue *sq = mFastMixer->sq(); 3340 FastMixerState *state = sq->begin(); 3341 if (!(state->mCommand & FastMixerState::IDLE)) { 3342 previousCommand = state->mCommand; 3343 state->mCommand = FastMixerState::HOT_IDLE; 3344 sq->end(); 3345 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3346 } else { 3347 sq->end(false /*didModify*/); 3348 } 3349 } 3350 3351 status_t status = PlaybackThread::releaseAudioPatch_l(handle); 3352 3353 if (!(previousCommand & FastMixerState::IDLE)) { 3354 ALOG_ASSERT(mFastMixer != 0); 3355 FastMixerStateQueue *sq = mFastMixer->sq(); 3356 FastMixerState *state = sq->begin(); 3357 ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE); 3358 state->mCommand = previousCommand; 3359 sq->end(); 3360 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3361 } 3362 3363 return status; 3364} 3365 3366status_t AudioFlinger::PlaybackThread::releaseAudioPatch_l(const audio_patch_handle_t handle) 3367{ 3368 status_t status = NO_ERROR; 3369 3370 mOutDevice = AUDIO_DEVICE_NONE; 3371 3372 if (mOutput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 3373 audio_hw_device_t *hwDevice = mOutput->audioHwDev->hwDevice(); 3374 status = hwDevice->release_audio_patch(hwDevice, handle); 3375 } else { 3376 AudioParameter param; 3377 param.addInt(String8(AUDIO_PARAMETER_STREAM_ROUTING), 0); 3378 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3379 param.toString().string()); 3380 } 3381 return status; 3382} 3383 3384void AudioFlinger::PlaybackThread::addPatchTrack(const sp<PatchTrack>& track) 3385{ 3386 Mutex::Autolock _l(mLock); 3387 mTracks.add(track); 3388} 3389 3390void AudioFlinger::PlaybackThread::deletePatchTrack(const sp<PatchTrack>& track) 3391{ 3392 Mutex::Autolock _l(mLock); 3393 destroyTrack_l(track); 3394} 3395 3396void AudioFlinger::PlaybackThread::getAudioPortConfig(struct audio_port_config *config) 3397{ 3398 ThreadBase::getAudioPortConfig(config); 3399 config->role = AUDIO_PORT_ROLE_SOURCE; 3400 config->ext.mix.hw_module = mOutput->audioHwDev->handle(); 3401 config->ext.mix.usecase.stream = AUDIO_STREAM_DEFAULT; 3402} 3403 3404// ---------------------------------------------------------------------------- 3405 3406AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 3407 audio_io_handle_t id, audio_devices_t device, bool systemReady, type_t type) 3408 : PlaybackThread(audioFlinger, output, id, device, type, systemReady), 3409 // mAudioMixer below 3410 // mFastMixer below 3411 mFastMixerFutex(0), 3412 mMasterMono(false) 3413 // mOutputSink below 3414 // mPipeSink below 3415 // mNormalSink below 3416{ 3417 ALOGV("MixerThread() id=%d device=%#x type=%d", id, device, type); 3418 ALOGV("mSampleRate=%u, mChannelMask=%#x, mChannelCount=%u, mFormat=%d, mFrameSize=%u, " 3419 "mFrameCount=%d, mNormalFrameCount=%d", 3420 mSampleRate, mChannelMask, mChannelCount, mFormat, mFrameSize, mFrameCount, 3421 mNormalFrameCount); 3422 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 3423 3424 if (type == DUPLICATING) { 3425 // The Duplicating thread uses the AudioMixer and delivers data to OutputTracks 3426 // (downstream MixerThreads) in DuplicatingThread::threadLoop_write(). 3427 // Do not create or use mFastMixer, mOutputSink, mPipeSink, or mNormalSink. 3428 return; 3429 } 3430 // create an NBAIO sink for the HAL output stream, and negotiate 3431 mOutputSink = new AudioStreamOutSink(output->stream); 3432 size_t numCounterOffers = 0; 3433 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount, mFormat)}; 3434 ssize_t index = mOutputSink->negotiate(offers, 1, NULL, numCounterOffers); 3435 ALOG_ASSERT(index == 0); 3436 3437 // initialize fast mixer depending on configuration 3438 bool initFastMixer; 3439 switch (kUseFastMixer) { 3440 case FastMixer_Never: 3441 initFastMixer = false; 3442 break; 3443 case FastMixer_Always: 3444 initFastMixer = true; 3445 break; 3446 case FastMixer_Static: 3447 case FastMixer_Dynamic: 3448 initFastMixer = mFrameCount < mNormalFrameCount; 3449 break; 3450 } 3451 if (initFastMixer) { 3452 audio_format_t fastMixerFormat; 3453 if (mMixerBufferEnabled && mEffectBufferEnabled) { 3454 fastMixerFormat = AUDIO_FORMAT_PCM_FLOAT; 3455 } else { 3456 fastMixerFormat = AUDIO_FORMAT_PCM_16_BIT; 3457 } 3458 if (mFormat != fastMixerFormat) { 3459 // change our Sink format to accept our intermediate precision 3460 mFormat = fastMixerFormat; 3461 free(mSinkBuffer); 3462 mFrameSize = mChannelCount * audio_bytes_per_sample(mFormat); 3463 const size_t sinkBufferSize = mNormalFrameCount * mFrameSize; 3464 (void)posix_memalign(&mSinkBuffer, 32, sinkBufferSize); 3465 } 3466 3467 // create a MonoPipe to connect our submix to FastMixer 3468 NBAIO_Format format = mOutputSink->format(); 3469 NBAIO_Format origformat = format; 3470 // adjust format to match that of the Fast Mixer 3471 ALOGV("format changed from %d to %d", format.mFormat, fastMixerFormat); 3472 format.mFormat = fastMixerFormat; 3473 format.mFrameSize = audio_bytes_per_sample(format.mFormat) * format.mChannelCount; 3474 3475 // This pipe depth compensates for scheduling latency of the normal mixer thread. 3476 // When it wakes up after a maximum latency, it runs a few cycles quickly before 3477 // finally blocking. Note the pipe implementation rounds up the request to a power of 2. 3478 MonoPipe *monoPipe = new MonoPipe(mNormalFrameCount * 4, format, true /*writeCanBlock*/); 3479 const NBAIO_Format offers[1] = {format}; 3480 size_t numCounterOffers = 0; 3481 ssize_t index = monoPipe->negotiate(offers, 1, NULL, numCounterOffers); 3482 ALOG_ASSERT(index == 0); 3483 monoPipe->setAvgFrames((mScreenState & 1) ? 3484 (monoPipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 3485 mPipeSink = monoPipe; 3486 3487#ifdef TEE_SINK 3488 if (mTeeSinkOutputEnabled) { 3489 // create a Pipe to archive a copy of FastMixer's output for dumpsys 3490 Pipe *teeSink = new Pipe(mTeeSinkOutputFrames, origformat); 3491 const NBAIO_Format offers2[1] = {origformat}; 3492 numCounterOffers = 0; 3493 index = teeSink->negotiate(offers2, 1, NULL, numCounterOffers); 3494 ALOG_ASSERT(index == 0); 3495 mTeeSink = teeSink; 3496 PipeReader *teeSource = new PipeReader(*teeSink); 3497 numCounterOffers = 0; 3498 index = teeSource->negotiate(offers2, 1, NULL, numCounterOffers); 3499 ALOG_ASSERT(index == 0); 3500 mTeeSource = teeSource; 3501 } 3502#endif 3503 3504 // create fast mixer and configure it initially with just one fast track for our submix 3505 mFastMixer = new FastMixer(); 3506 FastMixerStateQueue *sq = mFastMixer->sq(); 3507#ifdef STATE_QUEUE_DUMP 3508 sq->setObserverDump(&mStateQueueObserverDump); 3509 sq->setMutatorDump(&mStateQueueMutatorDump); 3510#endif 3511 FastMixerState *state = sq->begin(); 3512 FastTrack *fastTrack = &state->mFastTracks[0]; 3513 // wrap the source side of the MonoPipe to make it an AudioBufferProvider 3514 fastTrack->mBufferProvider = new SourceAudioBufferProvider(new MonoPipeReader(monoPipe)); 3515 fastTrack->mVolumeProvider = NULL; 3516 fastTrack->mChannelMask = mChannelMask; // mPipeSink channel mask for audio to FastMixer 3517 fastTrack->mFormat = mFormat; // mPipeSink format for audio to FastMixer 3518 fastTrack->mGeneration++; 3519 state->mFastTracksGen++; 3520 state->mTrackMask = 1; 3521 // fast mixer will use the HAL output sink 3522 state->mOutputSink = mOutputSink.get(); 3523 state->mOutputSinkGen++; 3524 state->mFrameCount = mFrameCount; 3525 state->mCommand = FastMixerState::COLD_IDLE; 3526 // already done in constructor initialization list 3527 //mFastMixerFutex = 0; 3528 state->mColdFutexAddr = &mFastMixerFutex; 3529 state->mColdGen++; 3530 state->mDumpState = &mFastMixerDumpState; 3531#ifdef TEE_SINK 3532 state->mTeeSink = mTeeSink.get(); 3533#endif 3534 mFastMixerNBLogWriter = audioFlinger->newWriter_l(kFastMixerLogSize, "FastMixer"); 3535 state->mNBLogWriter = mFastMixerNBLogWriter.get(); 3536 sq->end(); 3537 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3538 3539 // start the fast mixer 3540 mFastMixer->run("FastMixer", PRIORITY_URGENT_AUDIO); 3541 pid_t tid = mFastMixer->getTid(); 3542 sendPrioConfigEvent(getpid_cached, tid, kPriorityFastMixer); 3543 3544#ifdef AUDIO_WATCHDOG 3545 // create and start the watchdog 3546 mAudioWatchdog = new AudioWatchdog(); 3547 mAudioWatchdog->setDump(&mAudioWatchdogDump); 3548 mAudioWatchdog->run("AudioWatchdog", PRIORITY_URGENT_AUDIO); 3549 tid = mAudioWatchdog->getTid(); 3550 sendPrioConfigEvent(getpid_cached, tid, kPriorityFastMixer); 3551#endif 3552 3553 } 3554 3555 switch (kUseFastMixer) { 3556 case FastMixer_Never: 3557 case FastMixer_Dynamic: 3558 mNormalSink = mOutputSink; 3559 break; 3560 case FastMixer_Always: 3561 mNormalSink = mPipeSink; 3562 break; 3563 case FastMixer_Static: 3564 mNormalSink = initFastMixer ? mPipeSink : mOutputSink; 3565 break; 3566 } 3567} 3568 3569AudioFlinger::MixerThread::~MixerThread() 3570{ 3571 if (mFastMixer != 0) { 3572 FastMixerStateQueue *sq = mFastMixer->sq(); 3573 FastMixerState *state = sq->begin(); 3574 if (state->mCommand == FastMixerState::COLD_IDLE) { 3575 int32_t old = android_atomic_inc(&mFastMixerFutex); 3576 if (old == -1) { 3577 (void) syscall(__NR_futex, &mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 3578 } 3579 } 3580 state->mCommand = FastMixerState::EXIT; 3581 sq->end(); 3582 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3583 mFastMixer->join(); 3584 // Though the fast mixer thread has exited, it's state queue is still valid. 3585 // We'll use that extract the final state which contains one remaining fast track 3586 // corresponding to our sub-mix. 3587 state = sq->begin(); 3588 ALOG_ASSERT(state->mTrackMask == 1); 3589 FastTrack *fastTrack = &state->mFastTracks[0]; 3590 ALOG_ASSERT(fastTrack->mBufferProvider != NULL); 3591 delete fastTrack->mBufferProvider; 3592 sq->end(false /*didModify*/); 3593 mFastMixer.clear(); 3594#ifdef AUDIO_WATCHDOG 3595 if (mAudioWatchdog != 0) { 3596 mAudioWatchdog->requestExit(); 3597 mAudioWatchdog->requestExitAndWait(); 3598 mAudioWatchdog.clear(); 3599 } 3600#endif 3601 } 3602 mAudioFlinger->unregisterWriter(mFastMixerNBLogWriter); 3603 delete mAudioMixer; 3604} 3605 3606 3607uint32_t AudioFlinger::MixerThread::correctLatency_l(uint32_t latency) const 3608{ 3609 if (mFastMixer != 0) { 3610 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 3611 latency += (pipe->getAvgFrames() * 1000) / mSampleRate; 3612 } 3613 return latency; 3614} 3615 3616 3617void AudioFlinger::MixerThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) 3618{ 3619 PlaybackThread::threadLoop_removeTracks(tracksToRemove); 3620} 3621 3622ssize_t AudioFlinger::MixerThread::threadLoop_write() 3623{ 3624 // FIXME we should only do one push per cycle; confirm this is true 3625 // Start the fast mixer if it's not already running 3626 if (mFastMixer != 0) { 3627 FastMixerStateQueue *sq = mFastMixer->sq(); 3628 FastMixerState *state = sq->begin(); 3629 if (state->mCommand != FastMixerState::MIX_WRITE && 3630 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)) { 3631 if (state->mCommand == FastMixerState::COLD_IDLE) { 3632 3633 // FIXME workaround for first HAL write being CPU bound on some devices 3634 ATRACE_BEGIN("write"); 3635 mOutput->write((char *)mSinkBuffer, 0); 3636 ATRACE_END(); 3637 3638 int32_t old = android_atomic_inc(&mFastMixerFutex); 3639 if (old == -1) { 3640 (void) syscall(__NR_futex, &mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 3641 } 3642#ifdef AUDIO_WATCHDOG 3643 if (mAudioWatchdog != 0) { 3644 mAudioWatchdog->resume(); 3645 } 3646#endif 3647 } 3648 state->mCommand = FastMixerState::MIX_WRITE; 3649#ifdef FAST_THREAD_STATISTICS 3650 mFastMixerDumpState.increaseSamplingN(mAudioFlinger->isLowRamDevice() ? 3651 FastThreadDumpState::kSamplingNforLowRamDevice : FastThreadDumpState::kSamplingN); 3652#endif 3653 sq->end(); 3654 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3655 if (kUseFastMixer == FastMixer_Dynamic) { 3656 mNormalSink = mPipeSink; 3657 } 3658 } else { 3659 sq->end(false /*didModify*/); 3660 } 3661 } 3662 return PlaybackThread::threadLoop_write(); 3663} 3664 3665void AudioFlinger::MixerThread::threadLoop_standby() 3666{ 3667 // Idle the fast mixer if it's currently running 3668 if (mFastMixer != 0) { 3669 FastMixerStateQueue *sq = mFastMixer->sq(); 3670 FastMixerState *state = sq->begin(); 3671 if (!(state->mCommand & FastMixerState::IDLE)) { 3672 state->mCommand = FastMixerState::COLD_IDLE; 3673 state->mColdFutexAddr = &mFastMixerFutex; 3674 state->mColdGen++; 3675 mFastMixerFutex = 0; 3676 sq->end(); 3677 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 3678 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3679 if (kUseFastMixer == FastMixer_Dynamic) { 3680 mNormalSink = mOutputSink; 3681 } 3682#ifdef AUDIO_WATCHDOG 3683 if (mAudioWatchdog != 0) { 3684 mAudioWatchdog->pause(); 3685 } 3686#endif 3687 } else { 3688 sq->end(false /*didModify*/); 3689 } 3690 } 3691 PlaybackThread::threadLoop_standby(); 3692} 3693 3694bool AudioFlinger::PlaybackThread::waitingAsyncCallback_l() 3695{ 3696 return false; 3697} 3698 3699bool AudioFlinger::PlaybackThread::shouldStandby_l() 3700{ 3701 return !mStandby; 3702} 3703 3704bool AudioFlinger::PlaybackThread::waitingAsyncCallback() 3705{ 3706 Mutex::Autolock _l(mLock); 3707 return waitingAsyncCallback_l(); 3708} 3709 3710// shared by MIXER and DIRECT, overridden by DUPLICATING 3711void AudioFlinger::PlaybackThread::threadLoop_standby() 3712{ 3713 ALOGV("Audio hardware entering standby, mixer %p, suspend count %d", this, mSuspended); 3714 mOutput->standby(); 3715 if (mUseAsyncWrite != 0) { 3716 // discard any pending drain or write ack by incrementing sequence 3717 mWriteAckSequence = (mWriteAckSequence + 2) & ~1; 3718 mDrainSequence = (mDrainSequence + 2) & ~1; 3719 ALOG_ASSERT(mCallbackThread != 0); 3720 mCallbackThread->setWriteBlocked(mWriteAckSequence); 3721 mCallbackThread->setDraining(mDrainSequence); 3722 } 3723 mHwPaused = false; 3724} 3725 3726void AudioFlinger::PlaybackThread::onAddNewTrack_l() 3727{ 3728 ALOGV("signal playback thread"); 3729 broadcast_l(); 3730} 3731 3732void AudioFlinger::MixerThread::threadLoop_mix() 3733{ 3734 // mix buffers... 3735 mAudioMixer->process(); 3736 mCurrentWriteLength = mSinkBufferSize; 3737 // increase sleep time progressively when application underrun condition clears. 3738 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 3739 // that a steady state of alternating ready/not ready conditions keeps the sleep time 3740 // such that we would underrun the audio HAL. 3741 if ((mSleepTimeUs == 0) && (sleepTimeShift > 0)) { 3742 sleepTimeShift--; 3743 } 3744 mSleepTimeUs = 0; 3745 mStandbyTimeNs = systemTime() + mStandbyDelayNs; 3746 //TODO: delay standby when effects have a tail 3747 3748} 3749 3750void AudioFlinger::MixerThread::threadLoop_sleepTime() 3751{ 3752 // If no tracks are ready, sleep once for the duration of an output 3753 // buffer size, then write 0s to the output 3754 if (mSleepTimeUs == 0) { 3755 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3756 mSleepTimeUs = mActiveSleepTimeUs >> sleepTimeShift; 3757 if (mSleepTimeUs < kMinThreadSleepTimeUs) { 3758 mSleepTimeUs = kMinThreadSleepTimeUs; 3759 } 3760 // reduce sleep time in case of consecutive application underruns to avoid 3761 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 3762 // duration we would end up writing less data than needed by the audio HAL if 3763 // the condition persists. 3764 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 3765 sleepTimeShift++; 3766 } 3767 } else { 3768 mSleepTimeUs = mIdleSleepTimeUs; 3769 } 3770 } else if (mBytesWritten != 0 || (mMixerStatus == MIXER_TRACKS_ENABLED)) { 3771 // clear out mMixerBuffer or mSinkBuffer, to ensure buffers are cleared 3772 // before effects processing or output. 3773 if (mMixerBufferValid) { 3774 memset(mMixerBuffer, 0, mMixerBufferSize); 3775 } else { 3776 memset(mSinkBuffer, 0, mSinkBufferSize); 3777 } 3778 mSleepTimeUs = 0; 3779 ALOGV_IF(mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED), 3780 "anticipated start"); 3781 } 3782 // TODO add standby time extension fct of effect tail 3783} 3784 3785// prepareTracks_l() must be called with ThreadBase::mLock held 3786AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 3787 Vector< sp<Track> > *tracksToRemove) 3788{ 3789 3790 mixer_state mixerStatus = MIXER_IDLE; 3791 // find out which tracks need to be processed 3792 size_t count = mActiveTracks.size(); 3793 size_t mixedTracks = 0; 3794 size_t tracksWithEffect = 0; 3795 // counts only _active_ fast tracks 3796 size_t fastTracks = 0; 3797 uint32_t resetMask = 0; // bit mask of fast tracks that need to be reset 3798 3799 float masterVolume = mMasterVolume; 3800 bool masterMute = mMasterMute; 3801 3802 if (masterMute) { 3803 masterVolume = 0; 3804 } 3805 // Delegate master volume control to effect in output mix effect chain if needed 3806 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 3807 if (chain != 0) { 3808 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 3809 chain->setVolume_l(&v, &v); 3810 masterVolume = (float)((v + (1 << 23)) >> 24); 3811 chain.clear(); 3812 } 3813 3814 // prepare a new state to push 3815 FastMixerStateQueue *sq = NULL; 3816 FastMixerState *state = NULL; 3817 bool didModify = false; 3818 FastMixerStateQueue::block_t block = FastMixerStateQueue::BLOCK_UNTIL_PUSHED; 3819 if (mFastMixer != 0) { 3820 sq = mFastMixer->sq(); 3821 state = sq->begin(); 3822 } 3823 3824 mMixerBufferValid = false; // mMixerBuffer has no valid data until appropriate tracks found. 3825 mEffectBufferValid = false; // mEffectBuffer has no valid data until tracks found. 3826 3827 for (size_t i=0 ; i<count ; i++) { 3828 const sp<Track> t = mActiveTracks[i].promote(); 3829 if (t == 0) { 3830 continue; 3831 } 3832 3833 // this const just means the local variable doesn't change 3834 Track* const track = t.get(); 3835 3836 // process fast tracks 3837 if (track->isFastTrack()) { 3838 3839 // It's theoretically possible (though unlikely) for a fast track to be created 3840 // and then removed within the same normal mix cycle. This is not a problem, as 3841 // the track never becomes active so it's fast mixer slot is never touched. 3842 // The converse, of removing an (active) track and then creating a new track 3843 // at the identical fast mixer slot within the same normal mix cycle, 3844 // is impossible because the slot isn't marked available until the end of each cycle. 3845 int j = track->mFastIndex; 3846 ALOG_ASSERT(0 < j && j < (int)FastMixerState::kMaxFastTracks); 3847 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << j))); 3848 FastTrack *fastTrack = &state->mFastTracks[j]; 3849 3850 // Determine whether the track is currently in underrun condition, 3851 // and whether it had a recent underrun. 3852 FastTrackDump *ftDump = &mFastMixerDumpState.mTracks[j]; 3853 FastTrackUnderruns underruns = ftDump->mUnderruns; 3854 uint32_t recentFull = (underruns.mBitFields.mFull - 3855 track->mObservedUnderruns.mBitFields.mFull) & UNDERRUN_MASK; 3856 uint32_t recentPartial = (underruns.mBitFields.mPartial - 3857 track->mObservedUnderruns.mBitFields.mPartial) & UNDERRUN_MASK; 3858 uint32_t recentEmpty = (underruns.mBitFields.mEmpty - 3859 track->mObservedUnderruns.mBitFields.mEmpty) & UNDERRUN_MASK; 3860 uint32_t recentUnderruns = recentPartial + recentEmpty; 3861 track->mObservedUnderruns = underruns; 3862 // don't count underruns that occur while stopping or pausing 3863 // or stopped which can occur when flush() is called while active 3864 if (!(track->isStopping() || track->isPausing() || track->isStopped()) && 3865 recentUnderruns > 0) { 3866 // FIXME fast mixer will pull & mix partial buffers, but we count as a full underrun 3867 track->mAudioTrackServerProxy->tallyUnderrunFrames(recentUnderruns * mFrameCount); 3868 } else { 3869 track->mAudioTrackServerProxy->tallyUnderrunFrames(0); 3870 } 3871 3872 // This is similar to the state machine for normal tracks, 3873 // with a few modifications for fast tracks. 3874 bool isActive = true; 3875 switch (track->mState) { 3876 case TrackBase::STOPPING_1: 3877 // track stays active in STOPPING_1 state until first underrun 3878 if (recentUnderruns > 0 || track->isTerminated()) { 3879 track->mState = TrackBase::STOPPING_2; 3880 } 3881 break; 3882 case TrackBase::PAUSING: 3883 // ramp down is not yet implemented 3884 track->setPaused(); 3885 break; 3886 case TrackBase::RESUMING: 3887 // ramp up is not yet implemented 3888 track->mState = TrackBase::ACTIVE; 3889 break; 3890 case TrackBase::ACTIVE: 3891 if (recentFull > 0 || recentPartial > 0) { 3892 // track has provided at least some frames recently: reset retry count 3893 track->mRetryCount = kMaxTrackRetries; 3894 } 3895 if (recentUnderruns == 0) { 3896 // no recent underruns: stay active 3897 break; 3898 } 3899 // there has recently been an underrun of some kind 3900 if (track->sharedBuffer() == 0) { 3901 // were any of the recent underruns "empty" (no frames available)? 3902 if (recentEmpty == 0) { 3903 // no, then ignore the partial underruns as they are allowed indefinitely 3904 break; 3905 } 3906 // there has recently been an "empty" underrun: decrement the retry counter 3907 if (--(track->mRetryCount) > 0) { 3908 break; 3909 } 3910 // indicate to client process that the track was disabled because of underrun; 3911 // it will then automatically call start() when data is available 3912 android_atomic_or(CBLK_DISABLED, &track->mCblk->mFlags); 3913 // remove from active list, but state remains ACTIVE [confusing but true] 3914 isActive = false; 3915 break; 3916 } 3917 // fall through 3918 case TrackBase::STOPPING_2: 3919 case TrackBase::PAUSED: 3920 case TrackBase::STOPPED: 3921 case TrackBase::FLUSHED: // flush() while active 3922 // Check for presentation complete if track is inactive 3923 // We have consumed all the buffers of this track. 3924 // This would be incomplete if we auto-paused on underrun 3925 { 3926 size_t audioHALFrames = 3927 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 3928 size_t framesWritten = mBytesWritten / mFrameSize; 3929 if (!(mStandby || track->presentationComplete(framesWritten, audioHALFrames))) { 3930 // track stays in active list until presentation is complete 3931 break; 3932 } 3933 } 3934 if (track->isStopping_2()) { 3935 track->mState = TrackBase::STOPPED; 3936 } 3937 if (track->isStopped()) { 3938 // Can't reset directly, as fast mixer is still polling this track 3939 // track->reset(); 3940 // So instead mark this track as needing to be reset after push with ack 3941 resetMask |= 1 << i; 3942 } 3943 isActive = false; 3944 break; 3945 case TrackBase::IDLE: 3946 default: 3947 LOG_ALWAYS_FATAL("unexpected track state %d", track->mState); 3948 } 3949 3950 if (isActive) { 3951 // was it previously inactive? 3952 if (!(state->mTrackMask & (1 << j))) { 3953 ExtendedAudioBufferProvider *eabp = track; 3954 VolumeProvider *vp = track; 3955 fastTrack->mBufferProvider = eabp; 3956 fastTrack->mVolumeProvider = vp; 3957 fastTrack->mChannelMask = track->mChannelMask; 3958 fastTrack->mFormat = track->mFormat; 3959 fastTrack->mGeneration++; 3960 state->mTrackMask |= 1 << j; 3961 didModify = true; 3962 // no acknowledgement required for newly active tracks 3963 } 3964 // cache the combined master volume and stream type volume for fast mixer; this 3965 // lacks any synchronization or barrier so VolumeProvider may read a stale value 3966 track->mCachedVolume = masterVolume * mStreamTypes[track->streamType()].volume; 3967 ++fastTracks; 3968 } else { 3969 // was it previously active? 3970 if (state->mTrackMask & (1 << j)) { 3971 fastTrack->mBufferProvider = NULL; 3972 fastTrack->mGeneration++; 3973 state->mTrackMask &= ~(1 << j); 3974 didModify = true; 3975 // If any fast tracks were removed, we must wait for acknowledgement 3976 // because we're about to decrement the last sp<> on those tracks. 3977 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3978 } else { 3979 LOG_ALWAYS_FATAL("fast track %d should have been active; " 3980 "mState=%d, mTrackMask=%#x, recentUnderruns=%u, isShared=%d", 3981 j, track->mState, state->mTrackMask, recentUnderruns, 3982 track->sharedBuffer() != 0); 3983 } 3984 tracksToRemove->add(track); 3985 // Avoids a misleading display in dumpsys 3986 track->mObservedUnderruns.mBitFields.mMostRecent = UNDERRUN_FULL; 3987 } 3988 continue; 3989 } 3990 3991 { // local variable scope to avoid goto warning 3992 3993 audio_track_cblk_t* cblk = track->cblk(); 3994 3995 // The first time a track is added we wait 3996 // for all its buffers to be filled before processing it 3997 int name = track->name(); 3998 // make sure that we have enough frames to mix one full buffer. 3999 // enforce this condition only once to enable draining the buffer in case the client 4000 // app does not call stop() and relies on underrun to stop: 4001 // hence the test on (mMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 4002 // during last round 4003 size_t desiredFrames; 4004 const uint32_t sampleRate = track->mAudioTrackServerProxy->getSampleRate(); 4005 AudioPlaybackRate playbackRate = track->mAudioTrackServerProxy->getPlaybackRate(); 4006 4007 desiredFrames = sourceFramesNeededWithTimestretch( 4008 sampleRate, mNormalFrameCount, mSampleRate, playbackRate.mSpeed); 4009 // TODO: ONLY USED FOR LEGACY RESAMPLERS, remove when they are removed. 4010 // add frames already consumed but not yet released by the resampler 4011 // because mAudioTrackServerProxy->framesReady() will include these frames 4012 desiredFrames += mAudioMixer->getUnreleasedFrames(track->name()); 4013 4014 uint32_t minFrames = 1; 4015 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() && 4016 (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) { 4017 minFrames = desiredFrames; 4018 } 4019 4020 size_t framesReady = track->framesReady(); 4021 if (ATRACE_ENABLED()) { 4022 // I wish we had formatted trace names 4023 char traceName[16]; 4024 strcpy(traceName, "nRdy"); 4025 int name = track->name(); 4026 if (AudioMixer::TRACK0 <= name && 4027 name < (int) (AudioMixer::TRACK0 + AudioMixer::MAX_NUM_TRACKS)) { 4028 name -= AudioMixer::TRACK0; 4029 traceName[4] = (name / 10) + '0'; 4030 traceName[5] = (name % 10) + '0'; 4031 } else { 4032 traceName[4] = '?'; 4033 traceName[5] = '?'; 4034 } 4035 traceName[6] = '\0'; 4036 ATRACE_INT(traceName, framesReady); 4037 } 4038 if ((framesReady >= minFrames) && track->isReady() && 4039 !track->isPaused() && !track->isTerminated()) 4040 { 4041 ALOGVV("track %d s=%08x [OK] on thread %p", name, cblk->mServer, this); 4042 4043 mixedTracks++; 4044 4045 // track->mainBuffer() != mSinkBuffer or mMixerBuffer means 4046 // there is an effect chain connected to the track 4047 chain.clear(); 4048 if (track->mainBuffer() != mSinkBuffer && 4049 track->mainBuffer() != mMixerBuffer) { 4050 if (mEffectBufferEnabled) { 4051 mEffectBufferValid = true; // Later can set directly. 4052 } 4053 chain = getEffectChain_l(track->sessionId()); 4054 // Delegate volume control to effect in track effect chain if needed 4055 if (chain != 0) { 4056 tracksWithEffect++; 4057 } else { 4058 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on " 4059 "session %d", 4060 name, track->sessionId()); 4061 } 4062 } 4063 4064 4065 int param = AudioMixer::VOLUME; 4066 if (track->mFillingUpStatus == Track::FS_FILLED) { 4067 // no ramp for the first volume setting 4068 track->mFillingUpStatus = Track::FS_ACTIVE; 4069 if (track->mState == TrackBase::RESUMING) { 4070 track->mState = TrackBase::ACTIVE; 4071 param = AudioMixer::RAMP_VOLUME; 4072 } 4073 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 4074 // FIXME should not make a decision based on mServer 4075 } else if (cblk->mServer != 0) { 4076 // If the track is stopped before the first frame was mixed, 4077 // do not apply ramp 4078 param = AudioMixer::RAMP_VOLUME; 4079 } 4080 4081 // compute volume for this track 4082 uint32_t vl, vr; // in U8.24 integer format 4083 float vlf, vrf, vaf; // in [0.0, 1.0] float format 4084 if (track->isPausing() || mStreamTypes[track->streamType()].mute) { 4085 vl = vr = 0; 4086 vlf = vrf = vaf = 0.; 4087 if (track->isPausing()) { 4088 track->setPaused(); 4089 } 4090 } else { 4091 4092 // read original volumes with volume control 4093 float typeVolume = mStreamTypes[track->streamType()].volume; 4094 float v = masterVolume * typeVolume; 4095 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 4096 gain_minifloat_packed_t vlr = proxy->getVolumeLR(); 4097 vlf = float_from_gain(gain_minifloat_unpack_left(vlr)); 4098 vrf = float_from_gain(gain_minifloat_unpack_right(vlr)); 4099 // track volumes come from shared memory, so can't be trusted and must be clamped 4100 if (vlf > GAIN_FLOAT_UNITY) { 4101 ALOGV("Track left volume out of range: %.3g", vlf); 4102 vlf = GAIN_FLOAT_UNITY; 4103 } 4104 if (vrf > GAIN_FLOAT_UNITY) { 4105 ALOGV("Track right volume out of range: %.3g", vrf); 4106 vrf = GAIN_FLOAT_UNITY; 4107 } 4108 // now apply the master volume and stream type volume 4109 vlf *= v; 4110 vrf *= v; 4111 // assuming master volume and stream type volume each go up to 1.0, 4112 // then derive vl and vr as U8.24 versions for the effect chain 4113 const float scaleto8_24 = MAX_GAIN_INT * MAX_GAIN_INT; 4114 vl = (uint32_t) (scaleto8_24 * vlf); 4115 vr = (uint32_t) (scaleto8_24 * vrf); 4116 // vl and vr are now in U8.24 format 4117 uint16_t sendLevel = proxy->getSendLevel_U4_12(); 4118 // send level comes from shared memory and so may be corrupt 4119 if (sendLevel > MAX_GAIN_INT) { 4120 ALOGV("Track send level out of range: %04X", sendLevel); 4121 sendLevel = MAX_GAIN_INT; 4122 } 4123 // vaf is represented as [0.0, 1.0] float by rescaling sendLevel 4124 vaf = v * sendLevel * (1. / MAX_GAIN_INT); 4125 } 4126 4127 // Delegate volume control to effect in track effect chain if needed 4128 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 4129 // Do not ramp volume if volume is controlled by effect 4130 param = AudioMixer::VOLUME; 4131 // Update remaining floating point volume levels 4132 vlf = (float)vl / (1 << 24); 4133 vrf = (float)vr / (1 << 24); 4134 track->mHasVolumeController = true; 4135 } else { 4136 // force no volume ramp when volume controller was just disabled or removed 4137 // from effect chain to avoid volume spike 4138 if (track->mHasVolumeController) { 4139 param = AudioMixer::VOLUME; 4140 } 4141 track->mHasVolumeController = false; 4142 } 4143 4144 // XXX: these things DON'T need to be done each time 4145 mAudioMixer->setBufferProvider(name, track); 4146 mAudioMixer->enable(name); 4147 4148 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, &vlf); 4149 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, &vrf); 4150 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, &vaf); 4151 mAudioMixer->setParameter( 4152 name, 4153 AudioMixer::TRACK, 4154 AudioMixer::FORMAT, (void *)track->format()); 4155 mAudioMixer->setParameter( 4156 name, 4157 AudioMixer::TRACK, 4158 AudioMixer::CHANNEL_MASK, (void *)(uintptr_t)track->channelMask()); 4159 mAudioMixer->setParameter( 4160 name, 4161 AudioMixer::TRACK, 4162 AudioMixer::MIXER_CHANNEL_MASK, (void *)(uintptr_t)mChannelMask); 4163 // limit track sample rate to 2 x output sample rate, which changes at re-configuration 4164 uint32_t maxSampleRate = mSampleRate * AUDIO_RESAMPLER_DOWN_RATIO_MAX; 4165 uint32_t reqSampleRate = track->mAudioTrackServerProxy->getSampleRate(); 4166 if (reqSampleRate == 0) { 4167 reqSampleRate = mSampleRate; 4168 } else if (reqSampleRate > maxSampleRate) { 4169 reqSampleRate = maxSampleRate; 4170 } 4171 mAudioMixer->setParameter( 4172 name, 4173 AudioMixer::RESAMPLE, 4174 AudioMixer::SAMPLE_RATE, 4175 (void *)(uintptr_t)reqSampleRate); 4176 4177 AudioPlaybackRate playbackRate = track->mAudioTrackServerProxy->getPlaybackRate(); 4178 mAudioMixer->setParameter( 4179 name, 4180 AudioMixer::TIMESTRETCH, 4181 AudioMixer::PLAYBACK_RATE, 4182 &playbackRate); 4183 4184 /* 4185 * Select the appropriate output buffer for the track. 4186 * 4187 * Tracks with effects go into their own effects chain buffer 4188 * and from there into either mEffectBuffer or mSinkBuffer. 4189 * 4190 * Other tracks can use mMixerBuffer for higher precision 4191 * channel accumulation. If this buffer is enabled 4192 * (mMixerBufferEnabled true), then selected tracks will accumulate 4193 * into it. 4194 * 4195 */ 4196 if (mMixerBufferEnabled 4197 && (track->mainBuffer() == mSinkBuffer 4198 || track->mainBuffer() == mMixerBuffer)) { 4199 mAudioMixer->setParameter( 4200 name, 4201 AudioMixer::TRACK, 4202 AudioMixer::MIXER_FORMAT, (void *)mMixerBufferFormat); 4203 mAudioMixer->setParameter( 4204 name, 4205 AudioMixer::TRACK, 4206 AudioMixer::MAIN_BUFFER, (void *)mMixerBuffer); 4207 // TODO: override track->mainBuffer()? 4208 mMixerBufferValid = true; 4209 } else { 4210 mAudioMixer->setParameter( 4211 name, 4212 AudioMixer::TRACK, 4213 AudioMixer::MIXER_FORMAT, (void *)AUDIO_FORMAT_PCM_16_BIT); 4214 mAudioMixer->setParameter( 4215 name, 4216 AudioMixer::TRACK, 4217 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 4218 } 4219 mAudioMixer->setParameter( 4220 name, 4221 AudioMixer::TRACK, 4222 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 4223 4224 // reset retry count 4225 track->mRetryCount = kMaxTrackRetries; 4226 4227 // If one track is ready, set the mixer ready if: 4228 // - the mixer was not ready during previous round OR 4229 // - no other track is not ready 4230 if (mMixerStatusIgnoringFastTracks != MIXER_TRACKS_READY || 4231 mixerStatus != MIXER_TRACKS_ENABLED) { 4232 mixerStatus = MIXER_TRACKS_READY; 4233 } 4234 } else { 4235 if (framesReady < desiredFrames && !track->isStopped() && !track->isPaused()) { 4236 ALOGV("track(%p) underrun, framesReady(%zu) < framesDesired(%zd)", 4237 track, framesReady, desiredFrames); 4238 track->mAudioTrackServerProxy->tallyUnderrunFrames(desiredFrames); 4239 } else { 4240 track->mAudioTrackServerProxy->tallyUnderrunFrames(0); 4241 } 4242 4243 // clear effect chain input buffer if an active track underruns to avoid sending 4244 // previous audio buffer again to effects 4245 chain = getEffectChain_l(track->sessionId()); 4246 if (chain != 0) { 4247 chain->clearInputBuffer(); 4248 } 4249 4250 ALOGVV("track %d s=%08x [NOT READY] on thread %p", name, cblk->mServer, this); 4251 if ((track->sharedBuffer() != 0) || track->isTerminated() || 4252 track->isStopped() || track->isPaused()) { 4253 // We have consumed all the buffers of this track. 4254 // Remove it from the list of active tracks. 4255 // TODO: use actual buffer filling status instead of latency when available from 4256 // audio HAL 4257 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 4258 size_t framesWritten = mBytesWritten / mFrameSize; 4259 if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) { 4260 if (track->isStopped()) { 4261 track->reset(); 4262 } 4263 tracksToRemove->add(track); 4264 } 4265 } else { 4266 // No buffers for this track. Give it a few chances to 4267 // fill a buffer, then remove it from active list. 4268 if (--(track->mRetryCount) <= 0) { 4269 ALOGI("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 4270 tracksToRemove->add(track); 4271 // indicate to client process that the track was disabled because of underrun; 4272 // it will then automatically call start() when data is available 4273 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 4274 // If one track is not ready, mark the mixer also not ready if: 4275 // - the mixer was ready during previous round OR 4276 // - no other track is ready 4277 } else if (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY || 4278 mixerStatus != MIXER_TRACKS_READY) { 4279 mixerStatus = MIXER_TRACKS_ENABLED; 4280 } 4281 } 4282 mAudioMixer->disable(name); 4283 } 4284 4285 } // local variable scope to avoid goto warning 4286track_is_ready: ; 4287 4288 } 4289 4290 // Push the new FastMixer state if necessary 4291 bool pauseAudioWatchdog = false; 4292 if (didModify) { 4293 state->mFastTracksGen++; 4294 // if the fast mixer was active, but now there are no fast tracks, then put it in cold idle 4295 if (kUseFastMixer == FastMixer_Dynamic && 4296 state->mCommand == FastMixerState::MIX_WRITE && state->mTrackMask <= 1) { 4297 state->mCommand = FastMixerState::COLD_IDLE; 4298 state->mColdFutexAddr = &mFastMixerFutex; 4299 state->mColdGen++; 4300 mFastMixerFutex = 0; 4301 if (kUseFastMixer == FastMixer_Dynamic) { 4302 mNormalSink = mOutputSink; 4303 } 4304 // If we go into cold idle, need to wait for acknowledgement 4305 // so that fast mixer stops doing I/O. 4306 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 4307 pauseAudioWatchdog = true; 4308 } 4309 } 4310 if (sq != NULL) { 4311 sq->end(didModify); 4312 sq->push(block); 4313 } 4314#ifdef AUDIO_WATCHDOG 4315 if (pauseAudioWatchdog && mAudioWatchdog != 0) { 4316 mAudioWatchdog->pause(); 4317 } 4318#endif 4319 4320 // Now perform the deferred reset on fast tracks that have stopped 4321 while (resetMask != 0) { 4322 size_t i = __builtin_ctz(resetMask); 4323 ALOG_ASSERT(i < count); 4324 resetMask &= ~(1 << i); 4325 sp<Track> t = mActiveTracks[i].promote(); 4326 if (t == 0) { 4327 continue; 4328 } 4329 Track* track = t.get(); 4330 ALOG_ASSERT(track->isFastTrack() && track->isStopped()); 4331 track->reset(); 4332 } 4333 4334 // remove all the tracks that need to be... 4335 removeTracks_l(*tracksToRemove); 4336 4337 if (getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX) != 0) { 4338 mEffectBufferValid = true; 4339 } 4340 4341 if (mEffectBufferValid) { 4342 // as long as there are effects we should clear the effects buffer, to avoid 4343 // passing a non-clean buffer to the effect chain 4344 memset(mEffectBuffer, 0, mEffectBufferSize); 4345 } 4346 // sink or mix buffer must be cleared if all tracks are connected to an 4347 // effect chain as in this case the mixer will not write to the sink or mix buffer 4348 // and track effects will accumulate into it 4349 if ((mBytesRemaining == 0) && ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 4350 (mixedTracks == 0 && fastTracks > 0))) { 4351 // FIXME as a performance optimization, should remember previous zero status 4352 if (mMixerBufferValid) { 4353 memset(mMixerBuffer, 0, mMixerBufferSize); 4354 // TODO: In testing, mSinkBuffer below need not be cleared because 4355 // the PlaybackThread::threadLoop() copies mMixerBuffer into mSinkBuffer 4356 // after mixing. 4357 // 4358 // To enforce this guarantee: 4359 // ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 4360 // (mixedTracks == 0 && fastTracks > 0)) 4361 // must imply MIXER_TRACKS_READY. 4362 // Later, we may clear buffers regardless, and skip much of this logic. 4363 } 4364 // FIXME as a performance optimization, should remember previous zero status 4365 memset(mSinkBuffer, 0, mNormalFrameCount * mFrameSize); 4366 } 4367 4368 // if any fast tracks, then status is ready 4369 mMixerStatusIgnoringFastTracks = mixerStatus; 4370 if (fastTracks > 0) { 4371 mixerStatus = MIXER_TRACKS_READY; 4372 } 4373 return mixerStatus; 4374} 4375 4376// getTrackName_l() must be called with ThreadBase::mLock held 4377int AudioFlinger::MixerThread::getTrackName_l(audio_channel_mask_t channelMask, 4378 audio_format_t format, int sessionId) 4379{ 4380 return mAudioMixer->getTrackName(channelMask, format, sessionId); 4381} 4382 4383// deleteTrackName_l() must be called with ThreadBase::mLock held 4384void AudioFlinger::MixerThread::deleteTrackName_l(int name) 4385{ 4386 ALOGV("remove track (%d) and delete from mixer", name); 4387 mAudioMixer->deleteTrackName(name); 4388} 4389 4390// checkForNewParameter_l() must be called with ThreadBase::mLock held 4391bool AudioFlinger::MixerThread::checkForNewParameter_l(const String8& keyValuePair, 4392 status_t& status) 4393{ 4394 bool reconfig = false; 4395 bool a2dpDeviceChanged = false; 4396 4397 status = NO_ERROR; 4398 4399 // if !&IDLE, holds the FastMixer state to restore after new parameters processed 4400 FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE; 4401 if (mFastMixer != 0) { 4402 FastMixerStateQueue *sq = mFastMixer->sq(); 4403 FastMixerState *state = sq->begin(); 4404 if (!(state->mCommand & FastMixerState::IDLE)) { 4405 previousCommand = state->mCommand; 4406 state->mCommand = FastMixerState::HOT_IDLE; 4407 sq->end(); 4408 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 4409 } else { 4410 sq->end(false /*didModify*/); 4411 } 4412 } 4413 4414 AudioParameter param = AudioParameter(keyValuePair); 4415 int value; 4416 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 4417 reconfig = true; 4418 } 4419 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 4420 if (!isValidPcmSinkFormat((audio_format_t) value)) { 4421 status = BAD_VALUE; 4422 } else { 4423 // no need to save value, since it's constant 4424 reconfig = true; 4425 } 4426 } 4427 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 4428 if (!isValidPcmSinkChannelMask((audio_channel_mask_t) value)) { 4429 status = BAD_VALUE; 4430 } else { 4431 // no need to save value, since it's constant 4432 reconfig = true; 4433 } 4434 } 4435 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 4436 // do not accept frame count changes if tracks are open as the track buffer 4437 // size depends on frame count and correct behavior would not be guaranteed 4438 // if frame count is changed after track creation 4439 if (!mTracks.isEmpty()) { 4440 status = INVALID_OPERATION; 4441 } else { 4442 reconfig = true; 4443 } 4444 } 4445 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 4446#ifdef ADD_BATTERY_DATA 4447 // when changing the audio output device, call addBatteryData to notify 4448 // the change 4449 if (mOutDevice != value) { 4450 uint32_t params = 0; 4451 // check whether speaker is on 4452 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 4453 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 4454 } 4455 4456 audio_devices_t deviceWithoutSpeaker 4457 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 4458 // check if any other device (except speaker) is on 4459 if (value & deviceWithoutSpeaker) { 4460 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 4461 } 4462 4463 if (params != 0) { 4464 addBatteryData(params); 4465 } 4466 } 4467#endif 4468 4469 // forward device change to effects that have requested to be 4470 // aware of attached audio device. 4471 if (value != AUDIO_DEVICE_NONE) { 4472 a2dpDeviceChanged = 4473 (mOutDevice & AUDIO_DEVICE_OUT_ALL_A2DP) != (value & AUDIO_DEVICE_OUT_ALL_A2DP); 4474 mOutDevice = value; 4475 for (size_t i = 0; i < mEffectChains.size(); i++) { 4476 mEffectChains[i]->setDevice_l(mOutDevice); 4477 } 4478 } 4479 } 4480 4481 if (status == NO_ERROR) { 4482 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 4483 keyValuePair.string()); 4484 if (!mStandby && status == INVALID_OPERATION) { 4485 mOutput->standby(); 4486 mStandby = true; 4487 mBytesWritten = 0; 4488 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 4489 keyValuePair.string()); 4490 } 4491 if (status == NO_ERROR && reconfig) { 4492 readOutputParameters_l(); 4493 delete mAudioMixer; 4494 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 4495 for (size_t i = 0; i < mTracks.size() ; i++) { 4496 int name = getTrackName_l(mTracks[i]->mChannelMask, 4497 mTracks[i]->mFormat, mTracks[i]->mSessionId); 4498 if (name < 0) { 4499 break; 4500 } 4501 mTracks[i]->mName = name; 4502 } 4503 sendIoConfigEvent_l(AUDIO_OUTPUT_CONFIG_CHANGED); 4504 } 4505 } 4506 4507 if (!(previousCommand & FastMixerState::IDLE)) { 4508 ALOG_ASSERT(mFastMixer != 0); 4509 FastMixerStateQueue *sq = mFastMixer->sq(); 4510 FastMixerState *state = sq->begin(); 4511 ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE); 4512 state->mCommand = previousCommand; 4513 sq->end(); 4514 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 4515 } 4516 4517 return reconfig || a2dpDeviceChanged; 4518} 4519 4520 4521void AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 4522{ 4523 const size_t SIZE = 256; 4524 char buffer[SIZE]; 4525 String8 result; 4526 4527 PlaybackThread::dumpInternals(fd, args); 4528 dprintf(fd, " Thread throttle time (msecs): %u\n", mThreadThrottleTimeMs); 4529 dprintf(fd, " AudioMixer tracks: 0x%08x\n", mAudioMixer->trackNames()); 4530 dprintf(fd, " Master mono: %s\n", mMasterMono ? "on" : "off"); 4531 4532 // Make a non-atomic copy of fast mixer dump state so it won't change underneath us 4533 // while we are dumping it. It may be inconsistent, but it won't mutate! 4534 // This is a large object so we place it on the heap. 4535 // FIXME 25972958: Need an intelligent copy constructor that does not touch unused pages. 4536 const FastMixerDumpState *copy = new FastMixerDumpState(mFastMixerDumpState); 4537 copy->dump(fd); 4538 delete copy; 4539 4540#ifdef STATE_QUEUE_DUMP 4541 // Similar for state queue 4542 StateQueueObserverDump observerCopy = mStateQueueObserverDump; 4543 observerCopy.dump(fd); 4544 StateQueueMutatorDump mutatorCopy = mStateQueueMutatorDump; 4545 mutatorCopy.dump(fd); 4546#endif 4547 4548#ifdef TEE_SINK 4549 // Write the tee output to a .wav file 4550 dumpTee(fd, mTeeSource, mId); 4551#endif 4552 4553#ifdef AUDIO_WATCHDOG 4554 if (mAudioWatchdog != 0) { 4555 // Make a non-atomic copy of audio watchdog dump so it won't change underneath us 4556 AudioWatchdogDump wdCopy = mAudioWatchdogDump; 4557 wdCopy.dump(fd); 4558 } 4559#endif 4560} 4561 4562uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const 4563{ 4564 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000) / 2; 4565} 4566 4567uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() const 4568{ 4569 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000); 4570} 4571 4572void AudioFlinger::MixerThread::cacheParameters_l() 4573{ 4574 PlaybackThread::cacheParameters_l(); 4575 4576 // FIXME: Relaxed timing because of a certain device that can't meet latency 4577 // Should be reduced to 2x after the vendor fixes the driver issue 4578 // increase threshold again due to low power audio mode. The way this warning 4579 // threshold is calculated and its usefulness should be reconsidered anyway. 4580 maxPeriod = seconds(mNormalFrameCount) / mSampleRate * 15; 4581} 4582 4583// ---------------------------------------------------------------------------- 4584 4585AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 4586 AudioStreamOut* output, audio_io_handle_t id, audio_devices_t device, bool systemReady) 4587 : PlaybackThread(audioFlinger, output, id, device, DIRECT, systemReady) 4588 // mLeftVolFloat, mRightVolFloat 4589{ 4590} 4591 4592AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 4593 AudioStreamOut* output, audio_io_handle_t id, uint32_t device, 4594 ThreadBase::type_t type, bool systemReady) 4595 : PlaybackThread(audioFlinger, output, id, device, type, systemReady) 4596 // mLeftVolFloat, mRightVolFloat 4597{ 4598} 4599 4600AudioFlinger::DirectOutputThread::~DirectOutputThread() 4601{ 4602} 4603 4604void AudioFlinger::DirectOutputThread::processVolume_l(Track *track, bool lastTrack) 4605{ 4606 audio_track_cblk_t* cblk = track->cblk(); 4607 float left, right; 4608 4609 if (mMasterMute || mStreamTypes[track->streamType()].mute) { 4610 left = right = 0; 4611 } else { 4612 float typeVolume = mStreamTypes[track->streamType()].volume; 4613 float v = mMasterVolume * typeVolume; 4614 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 4615 gain_minifloat_packed_t vlr = proxy->getVolumeLR(); 4616 left = float_from_gain(gain_minifloat_unpack_left(vlr)); 4617 if (left > GAIN_FLOAT_UNITY) { 4618 left = GAIN_FLOAT_UNITY; 4619 } 4620 left *= v; 4621 right = float_from_gain(gain_minifloat_unpack_right(vlr)); 4622 if (right > GAIN_FLOAT_UNITY) { 4623 right = GAIN_FLOAT_UNITY; 4624 } 4625 right *= v; 4626 } 4627 4628 if (lastTrack) { 4629 if (left != mLeftVolFloat || right != mRightVolFloat) { 4630 mLeftVolFloat = left; 4631 mRightVolFloat = right; 4632 4633 // Convert volumes from float to 8.24 4634 uint32_t vl = (uint32_t)(left * (1 << 24)); 4635 uint32_t vr = (uint32_t)(right * (1 << 24)); 4636 4637 // Delegate volume control to effect in track effect chain if needed 4638 // only one effect chain can be present on DirectOutputThread, so if 4639 // there is one, the track is connected to it 4640 if (!mEffectChains.isEmpty()) { 4641 mEffectChains[0]->setVolume_l(&vl, &vr); 4642 left = (float)vl / (1 << 24); 4643 right = (float)vr / (1 << 24); 4644 } 4645 if (mOutput->stream->set_volume) { 4646 mOutput->stream->set_volume(mOutput->stream, left, right); 4647 } 4648 } 4649 } 4650} 4651 4652void AudioFlinger::DirectOutputThread::onAddNewTrack_l() 4653{ 4654 sp<Track> previousTrack = mPreviousTrack.promote(); 4655 sp<Track> latestTrack = mLatestActiveTrack.promote(); 4656 4657 if (previousTrack != 0 && latestTrack != 0) { 4658 if (mType == DIRECT) { 4659 if (previousTrack.get() != latestTrack.get()) { 4660 mFlushPending = true; 4661 } 4662 } else /* mType == OFFLOAD */ { 4663 if (previousTrack->sessionId() != latestTrack->sessionId()) { 4664 mFlushPending = true; 4665 } 4666 } 4667 } 4668 PlaybackThread::onAddNewTrack_l(); 4669} 4670 4671AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l( 4672 Vector< sp<Track> > *tracksToRemove 4673) 4674{ 4675 size_t count = mActiveTracks.size(); 4676 mixer_state mixerStatus = MIXER_IDLE; 4677 bool doHwPause = false; 4678 bool doHwResume = false; 4679 4680 // find out which tracks need to be processed 4681 for (size_t i = 0; i < count; i++) { 4682 sp<Track> t = mActiveTracks[i].promote(); 4683 // The track died recently 4684 if (t == 0) { 4685 continue; 4686 } 4687 4688 if (t->isInvalid()) { 4689 ALOGW("An invalidated track shouldn't be in active list"); 4690 tracksToRemove->add(t); 4691 continue; 4692 } 4693 4694 Track* const track = t.get(); 4695 audio_track_cblk_t* cblk = track->cblk(); 4696 // Only consider last track started for volume and mixer state control. 4697 // In theory an older track could underrun and restart after the new one starts 4698 // but as we only care about the transition phase between two tracks on a 4699 // direct output, it is not a problem to ignore the underrun case. 4700 sp<Track> l = mLatestActiveTrack.promote(); 4701 bool last = l.get() == track; 4702 4703 if (track->isPausing()) { 4704 track->setPaused(); 4705 if (mHwSupportsPause && last && !mHwPaused) { 4706 doHwPause = true; 4707 mHwPaused = true; 4708 } 4709 tracksToRemove->add(track); 4710 } else if (track->isFlushPending()) { 4711 track->flushAck(); 4712 if (last) { 4713 mFlushPending = true; 4714 } 4715 } else if (track->isResumePending()) { 4716 track->resumeAck(); 4717 if (last && mHwPaused) { 4718 doHwResume = true; 4719 mHwPaused = false; 4720 } 4721 } 4722 4723 // The first time a track is added we wait 4724 // for all its buffers to be filled before processing it. 4725 // Allow draining the buffer in case the client 4726 // app does not call stop() and relies on underrun to stop: 4727 // hence the test on (track->mRetryCount > 1). 4728 // If retryCount<=1 then track is about to underrun and be removed. 4729 // Do not use a high threshold for compressed audio. 4730 uint32_t minFrames; 4731 if ((track->sharedBuffer() == 0) && !track->isStopping_1() && !track->isPausing() 4732 && (track->mRetryCount > 1) && audio_is_linear_pcm(mFormat)) { 4733 minFrames = mNormalFrameCount; 4734 } else { 4735 minFrames = 1; 4736 } 4737 4738 if ((track->framesReady() >= minFrames) && track->isReady() && !track->isPaused() && 4739 !track->isStopping_2() && !track->isStopped()) 4740 { 4741 ALOGVV("track %d s=%08x [OK]", track->name(), cblk->mServer); 4742 4743 if (track->mFillingUpStatus == Track::FS_FILLED) { 4744 track->mFillingUpStatus = Track::FS_ACTIVE; 4745 // make sure processVolume_l() will apply new volume even if 0 4746 mLeftVolFloat = mRightVolFloat = -1.0; 4747 if (!mHwSupportsPause) { 4748 track->resumeAck(); 4749 } 4750 } 4751 4752 // compute volume for this track 4753 processVolume_l(track, last); 4754 if (last) { 4755 sp<Track> previousTrack = mPreviousTrack.promote(); 4756 if (previousTrack != 0) { 4757 if (track != previousTrack.get()) { 4758 // Flush any data still being written from last track 4759 mBytesRemaining = 0; 4760 // Invalidate previous track to force a seek when resuming. 4761 previousTrack->invalidate(); 4762 } 4763 } 4764 mPreviousTrack = track; 4765 4766 // reset retry count 4767 track->mRetryCount = kMaxTrackRetriesDirect; 4768 mActiveTrack = t; 4769 mixerStatus = MIXER_TRACKS_READY; 4770 if (mHwPaused) { 4771 doHwResume = true; 4772 mHwPaused = false; 4773 } 4774 } 4775 } else { 4776 // clear effect chain input buffer if the last active track started underruns 4777 // to avoid sending previous audio buffer again to effects 4778 if (!mEffectChains.isEmpty() && last) { 4779 mEffectChains[0]->clearInputBuffer(); 4780 } 4781 if (track->isStopping_1()) { 4782 track->mState = TrackBase::STOPPING_2; 4783 if (last && mHwPaused) { 4784 doHwResume = true; 4785 mHwPaused = false; 4786 } 4787 } 4788 if ((track->sharedBuffer() != 0) || track->isStopped() || 4789 track->isStopping_2() || track->isPaused()) { 4790 // We have consumed all the buffers of this track. 4791 // Remove it from the list of active tracks. 4792 size_t audioHALFrames; 4793 if (audio_is_linear_pcm(mFormat)) { 4794 audioHALFrames = (latency_l() * mSampleRate) / 1000; 4795 } else { 4796 audioHALFrames = 0; 4797 } 4798 4799 size_t framesWritten = mBytesWritten / mFrameSize; 4800 if (mStandby || !last || 4801 track->presentationComplete(framesWritten, audioHALFrames)) { 4802 if (track->isStopping_2()) { 4803 track->mState = TrackBase::STOPPED; 4804 } 4805 if (track->isStopped()) { 4806 track->reset(); 4807 } 4808 tracksToRemove->add(track); 4809 } 4810 } else { 4811 // No buffers for this track. Give it a few chances to 4812 // fill a buffer, then remove it from active list. 4813 // Only consider last track started for mixer state control 4814 if (--(track->mRetryCount) <= 0) { 4815 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 4816 tracksToRemove->add(track); 4817 // indicate to client process that the track was disabled because of underrun; 4818 // it will then automatically call start() when data is available 4819 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 4820 } else if (last) { 4821 ALOGW("pause because of UNDERRUN, framesReady = %zu," 4822 "minFrames = %u, mFormat = %#x", 4823 track->framesReady(), minFrames, mFormat); 4824 mixerStatus = MIXER_TRACKS_ENABLED; 4825 if (mHwSupportsPause && !mHwPaused && !mStandby) { 4826 doHwPause = true; 4827 mHwPaused = true; 4828 } 4829 } 4830 } 4831 } 4832 } 4833 4834 // if an active track did not command a flush, check for pending flush on stopped tracks 4835 if (!mFlushPending) { 4836 for (size_t i = 0; i < mTracks.size(); i++) { 4837 if (mTracks[i]->isFlushPending()) { 4838 mTracks[i]->flushAck(); 4839 mFlushPending = true; 4840 } 4841 } 4842 } 4843 4844 // make sure the pause/flush/resume sequence is executed in the right order. 4845 // If a flush is pending and a track is active but the HW is not paused, force a HW pause 4846 // before flush and then resume HW. This can happen in case of pause/flush/resume 4847 // if resume is received before pause is executed. 4848 if (mHwSupportsPause && !mStandby && 4849 (doHwPause || (mFlushPending && !mHwPaused && (count != 0)))) { 4850 mOutput->stream->pause(mOutput->stream); 4851 } 4852 if (mFlushPending) { 4853 flushHw_l(); 4854 } 4855 if (mHwSupportsPause && !mStandby && doHwResume) { 4856 mOutput->stream->resume(mOutput->stream); 4857 } 4858 // remove all the tracks that need to be... 4859 removeTracks_l(*tracksToRemove); 4860 4861 return mixerStatus; 4862} 4863 4864void AudioFlinger::DirectOutputThread::threadLoop_mix() 4865{ 4866 size_t frameCount = mFrameCount; 4867 int8_t *curBuf = (int8_t *)mSinkBuffer; 4868 // output audio to hardware 4869 while (frameCount) { 4870 AudioBufferProvider::Buffer buffer; 4871 buffer.frameCount = frameCount; 4872 status_t status = mActiveTrack->getNextBuffer(&buffer); 4873 if (status != NO_ERROR || buffer.raw == NULL) { 4874 memset(curBuf, 0, frameCount * mFrameSize); 4875 break; 4876 } 4877 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 4878 frameCount -= buffer.frameCount; 4879 curBuf += buffer.frameCount * mFrameSize; 4880 mActiveTrack->releaseBuffer(&buffer); 4881 } 4882 mCurrentWriteLength = curBuf - (int8_t *)mSinkBuffer; 4883 mSleepTimeUs = 0; 4884 mStandbyTimeNs = systemTime() + mStandbyDelayNs; 4885 mActiveTrack.clear(); 4886} 4887 4888void AudioFlinger::DirectOutputThread::threadLoop_sleepTime() 4889{ 4890 // do not write to HAL when paused 4891 if (mHwPaused || (usesHwAvSync() && mStandby)) { 4892 mSleepTimeUs = mIdleSleepTimeUs; 4893 return; 4894 } 4895 if (mSleepTimeUs == 0) { 4896 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4897 mSleepTimeUs = mActiveSleepTimeUs; 4898 } else { 4899 mSleepTimeUs = mIdleSleepTimeUs; 4900 } 4901 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 4902 memset(mSinkBuffer, 0, mFrameCount * mFrameSize); 4903 mSleepTimeUs = 0; 4904 } 4905} 4906 4907void AudioFlinger::DirectOutputThread::threadLoop_exit() 4908{ 4909 { 4910 Mutex::Autolock _l(mLock); 4911 for (size_t i = 0; i < mTracks.size(); i++) { 4912 if (mTracks[i]->isFlushPending()) { 4913 mTracks[i]->flushAck(); 4914 mFlushPending = true; 4915 } 4916 } 4917 if (mFlushPending) { 4918 flushHw_l(); 4919 } 4920 } 4921 PlaybackThread::threadLoop_exit(); 4922} 4923 4924// must be called with thread mutex locked 4925bool AudioFlinger::DirectOutputThread::shouldStandby_l() 4926{ 4927 bool trackPaused = false; 4928 bool trackStopped = false; 4929 4930 // do not put the HAL in standby when paused. AwesomePlayer clear the offloaded AudioTrack 4931 // after a timeout and we will enter standby then. 4932 if (mTracks.size() > 0) { 4933 trackPaused = mTracks[mTracks.size() - 1]->isPaused(); 4934 trackStopped = mTracks[mTracks.size() - 1]->isStopped() || 4935 mTracks[mTracks.size() - 1]->mState == TrackBase::IDLE; 4936 } 4937 4938 return !mStandby && !(trackPaused || (mHwPaused && !trackStopped)); 4939} 4940 4941// getTrackName_l() must be called with ThreadBase::mLock held 4942int AudioFlinger::DirectOutputThread::getTrackName_l(audio_channel_mask_t channelMask __unused, 4943 audio_format_t format __unused, int sessionId __unused) 4944{ 4945 return 0; 4946} 4947 4948// deleteTrackName_l() must be called with ThreadBase::mLock held 4949void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name __unused) 4950{ 4951} 4952 4953// checkForNewParameter_l() must be called with ThreadBase::mLock held 4954bool AudioFlinger::DirectOutputThread::checkForNewParameter_l(const String8& keyValuePair, 4955 status_t& status) 4956{ 4957 bool reconfig = false; 4958 bool a2dpDeviceChanged = false; 4959 4960 status = NO_ERROR; 4961 4962 AudioParameter param = AudioParameter(keyValuePair); 4963 int value; 4964 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 4965 // forward device change to effects that have requested to be 4966 // aware of attached audio device. 4967 if (value != AUDIO_DEVICE_NONE) { 4968 a2dpDeviceChanged = 4969 (mOutDevice & AUDIO_DEVICE_OUT_ALL_A2DP) != (value & AUDIO_DEVICE_OUT_ALL_A2DP); 4970 mOutDevice = value; 4971 for (size_t i = 0; i < mEffectChains.size(); i++) { 4972 mEffectChains[i]->setDevice_l(mOutDevice); 4973 } 4974 } 4975 } 4976 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 4977 // do not accept frame count changes if tracks are open as the track buffer 4978 // size depends on frame count and correct behavior would not be garantied 4979 // if frame count is changed after track creation 4980 if (!mTracks.isEmpty()) { 4981 status = INVALID_OPERATION; 4982 } else { 4983 reconfig = true; 4984 } 4985 } 4986 if (status == NO_ERROR) { 4987 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 4988 keyValuePair.string()); 4989 if (!mStandby && status == INVALID_OPERATION) { 4990 mOutput->standby(); 4991 mStandby = true; 4992 mBytesWritten = 0; 4993 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 4994 keyValuePair.string()); 4995 } 4996 if (status == NO_ERROR && reconfig) { 4997 readOutputParameters_l(); 4998 sendIoConfigEvent_l(AUDIO_OUTPUT_CONFIG_CHANGED); 4999 } 5000 } 5001 5002 return reconfig || a2dpDeviceChanged; 5003} 5004 5005uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() const 5006{ 5007 uint32_t time; 5008 if (audio_is_linear_pcm(mFormat)) { 5009 time = PlaybackThread::activeSleepTimeUs(); 5010 } else { 5011 time = 10000; 5012 } 5013 return time; 5014} 5015 5016uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() const 5017{ 5018 uint32_t time; 5019 if (audio_is_linear_pcm(mFormat)) { 5020 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 5021 } else { 5022 time = 10000; 5023 } 5024 return time; 5025} 5026 5027uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() const 5028{ 5029 uint32_t time; 5030 if (audio_is_linear_pcm(mFormat)) { 5031 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 5032 } else { 5033 time = 10000; 5034 } 5035 return time; 5036} 5037 5038void AudioFlinger::DirectOutputThread::cacheParameters_l() 5039{ 5040 PlaybackThread::cacheParameters_l(); 5041 5042 // use shorter standby delay as on normal output to release 5043 // hardware resources as soon as possible 5044 // no delay on outputs with HW A/V sync 5045 if (usesHwAvSync()) { 5046 mStandbyDelayNs = 0; 5047 } else if ((mType == OFFLOAD) && !audio_is_linear_pcm(mFormat)) { 5048 mStandbyDelayNs = kOffloadStandbyDelayNs; 5049 } else { 5050 mStandbyDelayNs = microseconds(mActiveSleepTimeUs*2); 5051 } 5052} 5053 5054void AudioFlinger::DirectOutputThread::flushHw_l() 5055{ 5056 mOutput->flush(); 5057 mHwPaused = false; 5058 mFlushPending = false; 5059} 5060 5061// ---------------------------------------------------------------------------- 5062 5063AudioFlinger::AsyncCallbackThread::AsyncCallbackThread( 5064 const wp<AudioFlinger::PlaybackThread>& playbackThread) 5065 : Thread(false /*canCallJava*/), 5066 mPlaybackThread(playbackThread), 5067 mWriteAckSequence(0), 5068 mDrainSequence(0) 5069{ 5070} 5071 5072AudioFlinger::AsyncCallbackThread::~AsyncCallbackThread() 5073{ 5074} 5075 5076void AudioFlinger::AsyncCallbackThread::onFirstRef() 5077{ 5078 run("Offload Cbk", ANDROID_PRIORITY_URGENT_AUDIO); 5079} 5080 5081bool AudioFlinger::AsyncCallbackThread::threadLoop() 5082{ 5083 while (!exitPending()) { 5084 uint32_t writeAckSequence; 5085 uint32_t drainSequence; 5086 5087 { 5088 Mutex::Autolock _l(mLock); 5089 while (!((mWriteAckSequence & 1) || 5090 (mDrainSequence & 1) || 5091 exitPending())) { 5092 mWaitWorkCV.wait(mLock); 5093 } 5094 5095 if (exitPending()) { 5096 break; 5097 } 5098 ALOGV("AsyncCallbackThread mWriteAckSequence %d mDrainSequence %d", 5099 mWriteAckSequence, mDrainSequence); 5100 writeAckSequence = mWriteAckSequence; 5101 mWriteAckSequence &= ~1; 5102 drainSequence = mDrainSequence; 5103 mDrainSequence &= ~1; 5104 } 5105 { 5106 sp<AudioFlinger::PlaybackThread> playbackThread = mPlaybackThread.promote(); 5107 if (playbackThread != 0) { 5108 if (writeAckSequence & 1) { 5109 playbackThread->resetWriteBlocked(writeAckSequence >> 1); 5110 } 5111 if (drainSequence & 1) { 5112 playbackThread->resetDraining(drainSequence >> 1); 5113 } 5114 } 5115 } 5116 } 5117 return false; 5118} 5119 5120void AudioFlinger::AsyncCallbackThread::exit() 5121{ 5122 ALOGV("AsyncCallbackThread::exit"); 5123 Mutex::Autolock _l(mLock); 5124 requestExit(); 5125 mWaitWorkCV.broadcast(); 5126} 5127 5128void AudioFlinger::AsyncCallbackThread::setWriteBlocked(uint32_t sequence) 5129{ 5130 Mutex::Autolock _l(mLock); 5131 // bit 0 is cleared 5132 mWriteAckSequence = sequence << 1; 5133} 5134 5135void AudioFlinger::AsyncCallbackThread::resetWriteBlocked() 5136{ 5137 Mutex::Autolock _l(mLock); 5138 // ignore unexpected callbacks 5139 if (mWriteAckSequence & 2) { 5140 mWriteAckSequence |= 1; 5141 mWaitWorkCV.signal(); 5142 } 5143} 5144 5145void AudioFlinger::AsyncCallbackThread::setDraining(uint32_t sequence) 5146{ 5147 Mutex::Autolock _l(mLock); 5148 // bit 0 is cleared 5149 mDrainSequence = sequence << 1; 5150} 5151 5152void AudioFlinger::AsyncCallbackThread::resetDraining() 5153{ 5154 Mutex::Autolock _l(mLock); 5155 // ignore unexpected callbacks 5156 if (mDrainSequence & 2) { 5157 mDrainSequence |= 1; 5158 mWaitWorkCV.signal(); 5159 } 5160} 5161 5162 5163// ---------------------------------------------------------------------------- 5164AudioFlinger::OffloadThread::OffloadThread(const sp<AudioFlinger>& audioFlinger, 5165 AudioStreamOut* output, audio_io_handle_t id, uint32_t device, bool systemReady) 5166 : DirectOutputThread(audioFlinger, output, id, device, OFFLOAD, systemReady), 5167 mPausedBytesRemaining(0) 5168{ 5169 //FIXME: mStandby should be set to true by ThreadBase constructor 5170 mStandby = true; 5171} 5172 5173void AudioFlinger::OffloadThread::threadLoop_exit() 5174{ 5175 if (mFlushPending || mHwPaused) { 5176 // If a flush is pending or track was paused, just discard buffered data 5177 flushHw_l(); 5178 } else { 5179 mMixerStatus = MIXER_DRAIN_ALL; 5180 threadLoop_drain(); 5181 } 5182 if (mUseAsyncWrite) { 5183 ALOG_ASSERT(mCallbackThread != 0); 5184 mCallbackThread->exit(); 5185 } 5186 PlaybackThread::threadLoop_exit(); 5187} 5188 5189AudioFlinger::PlaybackThread::mixer_state AudioFlinger::OffloadThread::prepareTracks_l( 5190 Vector< sp<Track> > *tracksToRemove 5191) 5192{ 5193 size_t count = mActiveTracks.size(); 5194 5195 mixer_state mixerStatus = MIXER_IDLE; 5196 bool doHwPause = false; 5197 bool doHwResume = false; 5198 5199 ALOGV("OffloadThread::prepareTracks_l active tracks %d", count); 5200 5201 // find out which tracks need to be processed 5202 for (size_t i = 0; i < count; i++) { 5203 sp<Track> t = mActiveTracks[i].promote(); 5204 // The track died recently 5205 if (t == 0) { 5206 continue; 5207 } 5208 Track* const track = t.get(); 5209 audio_track_cblk_t* cblk = track->cblk(); 5210 // Only consider last track started for volume and mixer state control. 5211 // In theory an older track could underrun and restart after the new one starts 5212 // but as we only care about the transition phase between two tracks on a 5213 // direct output, it is not a problem to ignore the underrun case. 5214 sp<Track> l = mLatestActiveTrack.promote(); 5215 bool last = l.get() == track; 5216 5217 if (track->isInvalid()) { 5218 ALOGW("An invalidated track shouldn't be in active list"); 5219 tracksToRemove->add(track); 5220 continue; 5221 } 5222 5223 if (track->mState == TrackBase::IDLE) { 5224 ALOGW("An idle track shouldn't be in active list"); 5225 continue; 5226 } 5227 5228 if (track->isPausing()) { 5229 track->setPaused(); 5230 if (last) { 5231 if (mHwSupportsPause && !mHwPaused) { 5232 doHwPause = true; 5233 mHwPaused = true; 5234 } 5235 // If we were part way through writing the mixbuffer to 5236 // the HAL we must save this until we resume 5237 // BUG - this will be wrong if a different track is made active, 5238 // in that case we want to discard the pending data in the 5239 // mixbuffer and tell the client to present it again when the 5240 // track is resumed 5241 mPausedWriteLength = mCurrentWriteLength; 5242 mPausedBytesRemaining = mBytesRemaining; 5243 mBytesRemaining = 0; // stop writing 5244 } 5245 tracksToRemove->add(track); 5246 } else if (track->isFlushPending()) { 5247 track->flushAck(); 5248 if (last) { 5249 mFlushPending = true; 5250 } 5251 } else if (track->isResumePending()){ 5252 track->resumeAck(); 5253 if (last) { 5254 if (mPausedBytesRemaining) { 5255 // Need to continue write that was interrupted 5256 mCurrentWriteLength = mPausedWriteLength; 5257 mBytesRemaining = mPausedBytesRemaining; 5258 mPausedBytesRemaining = 0; 5259 } 5260 if (mHwPaused) { 5261 doHwResume = true; 5262 mHwPaused = false; 5263 // threadLoop_mix() will handle the case that we need to 5264 // resume an interrupted write 5265 } 5266 // enable write to audio HAL 5267 mSleepTimeUs = 0; 5268 5269 // Do not handle new data in this iteration even if track->framesReady() 5270 mixerStatus = MIXER_TRACKS_ENABLED; 5271 } 5272 } else if (track->framesReady() && track->isReady() && 5273 !track->isPaused() && !track->isTerminated() && !track->isStopping_2()) { 5274 ALOGVV("OffloadThread: track %d s=%08x [OK]", track->name(), cblk->mServer); 5275 if (track->mFillingUpStatus == Track::FS_FILLED) { 5276 track->mFillingUpStatus = Track::FS_ACTIVE; 5277 // make sure processVolume_l() will apply new volume even if 0 5278 mLeftVolFloat = mRightVolFloat = -1.0; 5279 } 5280 5281 if (last) { 5282 sp<Track> previousTrack = mPreviousTrack.promote(); 5283 if (previousTrack != 0) { 5284 if (track != previousTrack.get()) { 5285 // Flush any data still being written from last track 5286 mBytesRemaining = 0; 5287 if (mPausedBytesRemaining) { 5288 // Last track was paused so we also need to flush saved 5289 // mixbuffer state and invalidate track so that it will 5290 // re-submit that unwritten data when it is next resumed 5291 mPausedBytesRemaining = 0; 5292 // Invalidate is a bit drastic - would be more efficient 5293 // to have a flag to tell client that some of the 5294 // previously written data was lost 5295 previousTrack->invalidate(); 5296 } 5297 // flush data already sent to the DSP if changing audio session as audio 5298 // comes from a different source. Also invalidate previous track to force a 5299 // seek when resuming. 5300 if (previousTrack->sessionId() != track->sessionId()) { 5301 previousTrack->invalidate(); 5302 } 5303 } 5304 } 5305 mPreviousTrack = track; 5306 // reset retry count 5307 track->mRetryCount = kMaxTrackRetriesOffload; 5308 mActiveTrack = t; 5309 mixerStatus = MIXER_TRACKS_READY; 5310 } 5311 } else { 5312 ALOGVV("OffloadThread: track %d s=%08x [NOT READY]", track->name(), cblk->mServer); 5313 if (track->isStopping_1()) { 5314 // Hardware buffer can hold a large amount of audio so we must 5315 // wait for all current track's data to drain before we say 5316 // that the track is stopped. 5317 if (mBytesRemaining == 0) { 5318 // Only start draining when all data in mixbuffer 5319 // has been written 5320 ALOGV("OffloadThread: underrun and STOPPING_1 -> draining, STOPPING_2"); 5321 track->mState = TrackBase::STOPPING_2; // so presentation completes after drain 5322 // do not drain if no data was ever sent to HAL (mStandby == true) 5323 if (last && !mStandby) { 5324 // do not modify drain sequence if we are already draining. This happens 5325 // when resuming from pause after drain. 5326 if ((mDrainSequence & 1) == 0) { 5327 mSleepTimeUs = 0; 5328 mStandbyTimeNs = systemTime() + mStandbyDelayNs; 5329 mixerStatus = MIXER_DRAIN_TRACK; 5330 mDrainSequence += 2; 5331 } 5332 if (mHwPaused) { 5333 // It is possible to move from PAUSED to STOPPING_1 without 5334 // a resume so we must ensure hardware is running 5335 doHwResume = true; 5336 mHwPaused = false; 5337 } 5338 } 5339 } 5340 } else if (track->isStopping_2()) { 5341 // Drain has completed or we are in standby, signal presentation complete 5342 if (!(mDrainSequence & 1) || !last || mStandby) { 5343 track->mState = TrackBase::STOPPED; 5344 size_t audioHALFrames = 5345 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 5346 size_t framesWritten = 5347 mBytesWritten / mOutput->getFrameSize(); 5348 track->presentationComplete(framesWritten, audioHALFrames); 5349 track->reset(); 5350 tracksToRemove->add(track); 5351 } 5352 } else { 5353 // No buffers for this track. Give it a few chances to 5354 // fill a buffer, then remove it from active list. 5355 if (--(track->mRetryCount) <= 0) { 5356 ALOGV("OffloadThread: BUFFER TIMEOUT: remove(%d) from active list", 5357 track->name()); 5358 tracksToRemove->add(track); 5359 // indicate to client process that the track was disabled because of underrun; 5360 // it will then automatically call start() when data is available 5361 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 5362 } else if (last){ 5363 mixerStatus = MIXER_TRACKS_ENABLED; 5364 } 5365 } 5366 } 5367 // compute volume for this track 5368 processVolume_l(track, last); 5369 } 5370 5371 // make sure the pause/flush/resume sequence is executed in the right order. 5372 // If a flush is pending and a track is active but the HW is not paused, force a HW pause 5373 // before flush and then resume HW. This can happen in case of pause/flush/resume 5374 // if resume is received before pause is executed. 5375 if (!mStandby && (doHwPause || (mFlushPending && !mHwPaused && (count != 0)))) { 5376 mOutput->stream->pause(mOutput->stream); 5377 } 5378 if (mFlushPending) { 5379 flushHw_l(); 5380 } 5381 if (!mStandby && doHwResume) { 5382 mOutput->stream->resume(mOutput->stream); 5383 } 5384 5385 // remove all the tracks that need to be... 5386 removeTracks_l(*tracksToRemove); 5387 5388 return mixerStatus; 5389} 5390 5391// must be called with thread mutex locked 5392bool AudioFlinger::OffloadThread::waitingAsyncCallback_l() 5393{ 5394 ALOGVV("waitingAsyncCallback_l mWriteAckSequence %d mDrainSequence %d", 5395 mWriteAckSequence, mDrainSequence); 5396 if (mUseAsyncWrite && ((mWriteAckSequence & 1) || (mDrainSequence & 1))) { 5397 return true; 5398 } 5399 return false; 5400} 5401 5402bool AudioFlinger::OffloadThread::waitingAsyncCallback() 5403{ 5404 Mutex::Autolock _l(mLock); 5405 return waitingAsyncCallback_l(); 5406} 5407 5408void AudioFlinger::OffloadThread::flushHw_l() 5409{ 5410 DirectOutputThread::flushHw_l(); 5411 // Flush anything still waiting in the mixbuffer 5412 mCurrentWriteLength = 0; 5413 mBytesRemaining = 0; 5414 mPausedWriteLength = 0; 5415 mPausedBytesRemaining = 0; 5416 5417 if (mUseAsyncWrite) { 5418 // discard any pending drain or write ack by incrementing sequence 5419 mWriteAckSequence = (mWriteAckSequence + 2) & ~1; 5420 mDrainSequence = (mDrainSequence + 2) & ~1; 5421 ALOG_ASSERT(mCallbackThread != 0); 5422 mCallbackThread->setWriteBlocked(mWriteAckSequence); 5423 mCallbackThread->setDraining(mDrainSequence); 5424 } 5425} 5426 5427// ---------------------------------------------------------------------------- 5428 5429AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 5430 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id, bool systemReady) 5431 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->outDevice(), 5432 systemReady, DUPLICATING), 5433 mWaitTimeMs(UINT_MAX) 5434{ 5435 addOutputTrack(mainThread); 5436} 5437 5438AudioFlinger::DuplicatingThread::~DuplicatingThread() 5439{ 5440 for (size_t i = 0; i < mOutputTracks.size(); i++) { 5441 mOutputTracks[i]->destroy(); 5442 } 5443} 5444 5445void AudioFlinger::DuplicatingThread::threadLoop_mix() 5446{ 5447 // mix buffers... 5448 if (outputsReady(outputTracks)) { 5449 mAudioMixer->process(); 5450 } else { 5451 if (mMixerBufferValid) { 5452 memset(mMixerBuffer, 0, mMixerBufferSize); 5453 } else { 5454 memset(mSinkBuffer, 0, mSinkBufferSize); 5455 } 5456 } 5457 mSleepTimeUs = 0; 5458 writeFrames = mNormalFrameCount; 5459 mCurrentWriteLength = mSinkBufferSize; 5460 mStandbyTimeNs = systemTime() + mStandbyDelayNs; 5461} 5462 5463void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() 5464{ 5465 if (mSleepTimeUs == 0) { 5466 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 5467 mSleepTimeUs = mActiveSleepTimeUs; 5468 } else { 5469 mSleepTimeUs = mIdleSleepTimeUs; 5470 } 5471 } else if (mBytesWritten != 0) { 5472 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 5473 writeFrames = mNormalFrameCount; 5474 memset(mSinkBuffer, 0, mSinkBufferSize); 5475 } else { 5476 // flush remaining overflow buffers in output tracks 5477 writeFrames = 0; 5478 } 5479 mSleepTimeUs = 0; 5480 } 5481} 5482 5483ssize_t AudioFlinger::DuplicatingThread::threadLoop_write() 5484{ 5485 for (size_t i = 0; i < outputTracks.size(); i++) { 5486 outputTracks[i]->write(mSinkBuffer, writeFrames); 5487 } 5488 mStandby = false; 5489 return (ssize_t)mSinkBufferSize; 5490} 5491 5492void AudioFlinger::DuplicatingThread::threadLoop_standby() 5493{ 5494 // DuplicatingThread implements standby by stopping all tracks 5495 for (size_t i = 0; i < outputTracks.size(); i++) { 5496 outputTracks[i]->stop(); 5497 } 5498} 5499 5500void AudioFlinger::DuplicatingThread::saveOutputTracks() 5501{ 5502 outputTracks = mOutputTracks; 5503} 5504 5505void AudioFlinger::DuplicatingThread::clearOutputTracks() 5506{ 5507 outputTracks.clear(); 5508} 5509 5510void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 5511{ 5512 Mutex::Autolock _l(mLock); 5513 // The downstream MixerThread consumes thread->frameCount() amount of frames per mix pass. 5514 // Adjust for thread->sampleRate() to determine minimum buffer frame count. 5515 // Then triple buffer because Threads do not run synchronously and may not be clock locked. 5516 const size_t frameCount = 5517 3 * sourceFramesNeeded(mSampleRate, thread->frameCount(), thread->sampleRate()); 5518 // TODO: Consider asynchronous sample rate conversion to handle clock disparity 5519 // from different OutputTracks and their associated MixerThreads (e.g. one may 5520 // nearly empty and the other may be dropping data). 5521 5522 sp<OutputTrack> outputTrack = new OutputTrack(thread, 5523 this, 5524 mSampleRate, 5525 mFormat, 5526 mChannelMask, 5527 frameCount, 5528 IPCThreadState::self()->getCallingUid()); 5529 if (outputTrack->cblk() != NULL) { 5530 thread->setStreamVolume(AUDIO_STREAM_PATCH, 1.0f); 5531 mOutputTracks.add(outputTrack); 5532 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack.get(), thread); 5533 updateWaitTime_l(); 5534 } 5535} 5536 5537void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 5538{ 5539 Mutex::Autolock _l(mLock); 5540 for (size_t i = 0; i < mOutputTracks.size(); i++) { 5541 if (mOutputTracks[i]->thread() == thread) { 5542 mOutputTracks[i]->destroy(); 5543 mOutputTracks.removeAt(i); 5544 updateWaitTime_l(); 5545 if (thread->getOutput() == mOutput) { 5546 mOutput = NULL; 5547 } 5548 return; 5549 } 5550 } 5551 ALOGV("removeOutputTrack(): unknown thread: %p", thread); 5552} 5553 5554// caller must hold mLock 5555void AudioFlinger::DuplicatingThread::updateWaitTime_l() 5556{ 5557 mWaitTimeMs = UINT_MAX; 5558 for (size_t i = 0; i < mOutputTracks.size(); i++) { 5559 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 5560 if (strong != 0) { 5561 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 5562 if (waitTimeMs < mWaitTimeMs) { 5563 mWaitTimeMs = waitTimeMs; 5564 } 5565 } 5566 } 5567} 5568 5569 5570bool AudioFlinger::DuplicatingThread::outputsReady( 5571 const SortedVector< sp<OutputTrack> > &outputTracks) 5572{ 5573 for (size_t i = 0; i < outputTracks.size(); i++) { 5574 sp<ThreadBase> thread = outputTracks[i]->thread().promote(); 5575 if (thread == 0) { 5576 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", 5577 outputTracks[i].get()); 5578 return false; 5579 } 5580 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 5581 // see note at standby() declaration 5582 if (playbackThread->standby() && !playbackThread->isSuspended()) { 5583 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), 5584 thread.get()); 5585 return false; 5586 } 5587 } 5588 return true; 5589} 5590 5591uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() const 5592{ 5593 return (mWaitTimeMs * 1000) / 2; 5594} 5595 5596void AudioFlinger::DuplicatingThread::cacheParameters_l() 5597{ 5598 // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first 5599 updateWaitTime_l(); 5600 5601 MixerThread::cacheParameters_l(); 5602} 5603 5604// ---------------------------------------------------------------------------- 5605// Record 5606// ---------------------------------------------------------------------------- 5607 5608AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 5609 AudioStreamIn *input, 5610 audio_io_handle_t id, 5611 audio_devices_t outDevice, 5612 audio_devices_t inDevice, 5613 bool systemReady 5614#ifdef TEE_SINK 5615 , const sp<NBAIO_Sink>& teeSink 5616#endif 5617 ) : 5618 ThreadBase(audioFlinger, id, outDevice, inDevice, RECORD, systemReady), 5619 mInput(input), mActiveTracksGen(0), mRsmpInBuffer(NULL), 5620 // mRsmpInFrames and mRsmpInFramesP2 are set by readInputParameters_l() 5621 mRsmpInRear(0) 5622#ifdef TEE_SINK 5623 , mTeeSink(teeSink) 5624#endif 5625 , mReadOnlyHeap(new MemoryDealer(kRecordThreadReadOnlyHeapSize, 5626 "RecordThreadRO", MemoryHeapBase::READ_ONLY)) 5627 // mFastCapture below 5628 , mFastCaptureFutex(0) 5629 // mInputSource 5630 // mPipeSink 5631 // mPipeSource 5632 , mPipeFramesP2(0) 5633 // mPipeMemory 5634 // mFastCaptureNBLogWriter 5635 , mFastTrackAvail(false) 5636{ 5637 snprintf(mThreadName, kThreadNameLength, "AudioIn_%X", id); 5638 mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mThreadName); 5639 5640 readInputParameters_l(); 5641 5642 // create an NBAIO source for the HAL input stream, and negotiate 5643 mInputSource = new AudioStreamInSource(input->stream); 5644 size_t numCounterOffers = 0; 5645 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount, mFormat)}; 5646 ssize_t index = mInputSource->negotiate(offers, 1, NULL, numCounterOffers); 5647 ALOG_ASSERT(index == 0); 5648 5649 // initialize fast capture depending on configuration 5650 bool initFastCapture; 5651 switch (kUseFastCapture) { 5652 case FastCapture_Never: 5653 initFastCapture = false; 5654 break; 5655 case FastCapture_Always: 5656 initFastCapture = true; 5657 break; 5658 case FastCapture_Static: 5659 initFastCapture = (mFrameCount * 1000) / mSampleRate < kMinNormalCaptureBufferSizeMs; 5660 break; 5661 // case FastCapture_Dynamic: 5662 } 5663 5664 if (initFastCapture) { 5665 // create a Pipe for FastCapture to write to, and for us and fast tracks to read from 5666 NBAIO_Format format = mInputSource->format(); 5667 size_t pipeFramesP2 = roundup(mSampleRate / 25); // double-buffering of 20 ms each 5668 size_t pipeSize = pipeFramesP2 * Format_frameSize(format); 5669 void *pipeBuffer; 5670 const sp<MemoryDealer> roHeap(readOnlyHeap()); 5671 sp<IMemory> pipeMemory; 5672 if ((roHeap == 0) || 5673 (pipeMemory = roHeap->allocate(pipeSize)) == 0 || 5674 (pipeBuffer = pipeMemory->pointer()) == NULL) { 5675 ALOGE("not enough memory for pipe buffer size=%zu", pipeSize); 5676 goto failed; 5677 } 5678 // pipe will be shared directly with fast clients, so clear to avoid leaking old information 5679 memset(pipeBuffer, 0, pipeSize); 5680 Pipe *pipe = new Pipe(pipeFramesP2, format, pipeBuffer); 5681 const NBAIO_Format offers[1] = {format}; 5682 size_t numCounterOffers = 0; 5683 ssize_t index = pipe->negotiate(offers, 1, NULL, numCounterOffers); 5684 ALOG_ASSERT(index == 0); 5685 mPipeSink = pipe; 5686 PipeReader *pipeReader = new PipeReader(*pipe); 5687 numCounterOffers = 0; 5688 index = pipeReader->negotiate(offers, 1, NULL, numCounterOffers); 5689 ALOG_ASSERT(index == 0); 5690 mPipeSource = pipeReader; 5691 mPipeFramesP2 = pipeFramesP2; 5692 mPipeMemory = pipeMemory; 5693 5694 // create fast capture 5695 mFastCapture = new FastCapture(); 5696 FastCaptureStateQueue *sq = mFastCapture->sq(); 5697#ifdef STATE_QUEUE_DUMP 5698 // FIXME 5699#endif 5700 FastCaptureState *state = sq->begin(); 5701 state->mCblk = NULL; 5702 state->mInputSource = mInputSource.get(); 5703 state->mInputSourceGen++; 5704 state->mPipeSink = pipe; 5705 state->mPipeSinkGen++; 5706 state->mFrameCount = mFrameCount; 5707 state->mCommand = FastCaptureState::COLD_IDLE; 5708 // already done in constructor initialization list 5709 //mFastCaptureFutex = 0; 5710 state->mColdFutexAddr = &mFastCaptureFutex; 5711 state->mColdGen++; 5712 state->mDumpState = &mFastCaptureDumpState; 5713#ifdef TEE_SINK 5714 // FIXME 5715#endif 5716 mFastCaptureNBLogWriter = audioFlinger->newWriter_l(kFastCaptureLogSize, "FastCapture"); 5717 state->mNBLogWriter = mFastCaptureNBLogWriter.get(); 5718 sq->end(); 5719 sq->push(FastCaptureStateQueue::BLOCK_UNTIL_PUSHED); 5720 5721 // start the fast capture 5722 mFastCapture->run("FastCapture", ANDROID_PRIORITY_URGENT_AUDIO); 5723 pid_t tid = mFastCapture->getTid(); 5724 sendPrioConfigEvent(getpid_cached, tid, kPriorityFastMixer); 5725#ifdef AUDIO_WATCHDOG 5726 // FIXME 5727#endif 5728 5729 mFastTrackAvail = true; 5730 } 5731failed: ; 5732 5733 // FIXME mNormalSource 5734} 5735 5736AudioFlinger::RecordThread::~RecordThread() 5737{ 5738 if (mFastCapture != 0) { 5739 FastCaptureStateQueue *sq = mFastCapture->sq(); 5740 FastCaptureState *state = sq->begin(); 5741 if (state->mCommand == FastCaptureState::COLD_IDLE) { 5742 int32_t old = android_atomic_inc(&mFastCaptureFutex); 5743 if (old == -1) { 5744 (void) syscall(__NR_futex, &mFastCaptureFutex, FUTEX_WAKE_PRIVATE, 1); 5745 } 5746 } 5747 state->mCommand = FastCaptureState::EXIT; 5748 sq->end(); 5749 sq->push(FastCaptureStateQueue::BLOCK_UNTIL_PUSHED); 5750 mFastCapture->join(); 5751 mFastCapture.clear(); 5752 } 5753 mAudioFlinger->unregisterWriter(mFastCaptureNBLogWriter); 5754 mAudioFlinger->unregisterWriter(mNBLogWriter); 5755 free(mRsmpInBuffer); 5756} 5757 5758void AudioFlinger::RecordThread::onFirstRef() 5759{ 5760 run(mThreadName, PRIORITY_URGENT_AUDIO); 5761} 5762 5763bool AudioFlinger::RecordThread::threadLoop() 5764{ 5765 nsecs_t lastWarning = 0; 5766 5767 inputStandBy(); 5768 5769reacquire_wakelock: 5770 sp<RecordTrack> activeTrack; 5771 int activeTracksGen; 5772 { 5773 Mutex::Autolock _l(mLock); 5774 size_t size = mActiveTracks.size(); 5775 activeTracksGen = mActiveTracksGen; 5776 if (size > 0) { 5777 // FIXME an arbitrary choice 5778 activeTrack = mActiveTracks[0]; 5779 acquireWakeLock_l(activeTrack->uid()); 5780 if (size > 1) { 5781 SortedVector<int> tmp; 5782 for (size_t i = 0; i < size; i++) { 5783 tmp.add(mActiveTracks[i]->uid()); 5784 } 5785 updateWakeLockUids_l(tmp); 5786 } 5787 } else { 5788 acquireWakeLock_l(-1); 5789 } 5790 } 5791 5792 mTimestamp.mTimebaseOffset[ExtendedTimestamp::TIMEBASE_BOOTTIME] = 5793 gBoottime.getBoottimeOffset(); 5794 5795 // used to request a deferred sleep, to be executed later while mutex is unlocked 5796 uint32_t sleepUs = 0; 5797 5798 // loop while there is work to do 5799 for (;;) { 5800 Vector< sp<EffectChain> > effectChains; 5801 5802 // sleep with mutex unlocked 5803 if (sleepUs > 0) { 5804 ATRACE_BEGIN("sleep"); 5805 usleep(sleepUs); 5806 ATRACE_END(); 5807 sleepUs = 0; 5808 } 5809 5810 // activeTracks accumulates a copy of a subset of mActiveTracks 5811 Vector< sp<RecordTrack> > activeTracks; 5812 5813 // reference to the (first and only) active fast track 5814 sp<RecordTrack> fastTrack; 5815 5816 // reference to a fast track which is about to be removed 5817 sp<RecordTrack> fastTrackToRemove; 5818 5819 { // scope for mLock 5820 Mutex::Autolock _l(mLock); 5821 5822 processConfigEvents_l(); 5823 5824 // check exitPending here because checkForNewParameters_l() and 5825 // checkForNewParameters_l() can temporarily release mLock 5826 if (exitPending()) { 5827 break; 5828 } 5829 5830 // if no active track(s), then standby and release wakelock 5831 size_t size = mActiveTracks.size(); 5832 if (size == 0) { 5833 standbyIfNotAlreadyInStandby(); 5834 // exitPending() can't become true here 5835 releaseWakeLock_l(); 5836 ALOGV("RecordThread: loop stopping"); 5837 // go to sleep 5838 mWaitWorkCV.wait(mLock); 5839 ALOGV("RecordThread: loop starting"); 5840 goto reacquire_wakelock; 5841 } 5842 5843 if (mActiveTracksGen != activeTracksGen) { 5844 activeTracksGen = mActiveTracksGen; 5845 SortedVector<int> tmp; 5846 for (size_t i = 0; i < size; i++) { 5847 tmp.add(mActiveTracks[i]->uid()); 5848 } 5849 updateWakeLockUids_l(tmp); 5850 } 5851 5852 bool doBroadcast = false; 5853 for (size_t i = 0; i < size; ) { 5854 5855 activeTrack = mActiveTracks[i]; 5856 if (activeTrack->isTerminated()) { 5857 if (activeTrack->isFastTrack()) { 5858 ALOG_ASSERT(fastTrackToRemove == 0); 5859 fastTrackToRemove = activeTrack; 5860 } 5861 removeTrack_l(activeTrack); 5862 mActiveTracks.remove(activeTrack); 5863 mActiveTracksGen++; 5864 size--; 5865 continue; 5866 } 5867 5868 TrackBase::track_state activeTrackState = activeTrack->mState; 5869 switch (activeTrackState) { 5870 5871 case TrackBase::PAUSING: 5872 mActiveTracks.remove(activeTrack); 5873 mActiveTracksGen++; 5874 doBroadcast = true; 5875 size--; 5876 continue; 5877 5878 case TrackBase::STARTING_1: 5879 sleepUs = 10000; 5880 i++; 5881 continue; 5882 5883 case TrackBase::STARTING_2: 5884 doBroadcast = true; 5885 mStandby = false; 5886 activeTrack->mState = TrackBase::ACTIVE; 5887 break; 5888 5889 case TrackBase::ACTIVE: 5890 break; 5891 5892 case TrackBase::IDLE: 5893 i++; 5894 continue; 5895 5896 default: 5897 LOG_ALWAYS_FATAL("Unexpected activeTrackState %d", activeTrackState); 5898 } 5899 5900 activeTracks.add(activeTrack); 5901 i++; 5902 5903 if (activeTrack->isFastTrack()) { 5904 ALOG_ASSERT(!mFastTrackAvail); 5905 ALOG_ASSERT(fastTrack == 0); 5906 fastTrack = activeTrack; 5907 } 5908 } 5909 if (doBroadcast) { 5910 mStartStopCond.broadcast(); 5911 } 5912 5913 // sleep if there are no active tracks to process 5914 if (activeTracks.size() == 0) { 5915 if (sleepUs == 0) { 5916 sleepUs = kRecordThreadSleepUs; 5917 } 5918 continue; 5919 } 5920 sleepUs = 0; 5921 5922 lockEffectChains_l(effectChains); 5923 } 5924 5925 // thread mutex is now unlocked, mActiveTracks unknown, activeTracks.size() > 0 5926 5927 size_t size = effectChains.size(); 5928 for (size_t i = 0; i < size; i++) { 5929 // thread mutex is not locked, but effect chain is locked 5930 effectChains[i]->process_l(); 5931 } 5932 5933 // Push a new fast capture state if fast capture is not already running, or cblk change 5934 if (mFastCapture != 0) { 5935 FastCaptureStateQueue *sq = mFastCapture->sq(); 5936 FastCaptureState *state = sq->begin(); 5937 bool didModify = false; 5938 FastCaptureStateQueue::block_t block = FastCaptureStateQueue::BLOCK_UNTIL_PUSHED; 5939 if (state->mCommand != FastCaptureState::READ_WRITE /* FIXME && 5940 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)*/) { 5941 if (state->mCommand == FastCaptureState::COLD_IDLE) { 5942 int32_t old = android_atomic_inc(&mFastCaptureFutex); 5943 if (old == -1) { 5944 (void) syscall(__NR_futex, &mFastCaptureFutex, FUTEX_WAKE_PRIVATE, 1); 5945 } 5946 } 5947 state->mCommand = FastCaptureState::READ_WRITE; 5948#if 0 // FIXME 5949 mFastCaptureDumpState.increaseSamplingN(mAudioFlinger->isLowRamDevice() ? 5950 FastThreadDumpState::kSamplingNforLowRamDevice : 5951 FastThreadDumpState::kSamplingN); 5952#endif 5953 didModify = true; 5954 } 5955 audio_track_cblk_t *cblkOld = state->mCblk; 5956 audio_track_cblk_t *cblkNew = fastTrack != 0 ? fastTrack->cblk() : NULL; 5957 if (cblkNew != cblkOld) { 5958 state->mCblk = cblkNew; 5959 // block until acked if removing a fast track 5960 if (cblkOld != NULL) { 5961 block = FastCaptureStateQueue::BLOCK_UNTIL_ACKED; 5962 } 5963 didModify = true; 5964 } 5965 sq->end(didModify); 5966 if (didModify) { 5967 sq->push(block); 5968#if 0 5969 if (kUseFastCapture == FastCapture_Dynamic) { 5970 mNormalSource = mPipeSource; 5971 } 5972#endif 5973 } 5974 } 5975 5976 // now run the fast track destructor with thread mutex unlocked 5977 fastTrackToRemove.clear(); 5978 5979 // Read from HAL to keep up with fastest client if multiple active tracks, not slowest one. 5980 // Only the client(s) that are too slow will overrun. But if even the fastest client is too 5981 // slow, then this RecordThread will overrun by not calling HAL read often enough. 5982 // If destination is non-contiguous, first read past the nominal end of buffer, then 5983 // copy to the right place. Permitted because mRsmpInBuffer was over-allocated. 5984 5985 int32_t rear = mRsmpInRear & (mRsmpInFramesP2 - 1); 5986 ssize_t framesRead; 5987 5988 // If an NBAIO source is present, use it to read the normal capture's data 5989 if (mPipeSource != 0) { 5990 size_t framesToRead = mBufferSize / mFrameSize; 5991 framesRead = mPipeSource->read((uint8_t*)mRsmpInBuffer + rear * mFrameSize, 5992 framesToRead); 5993 if (framesRead == 0) { 5994 // since pipe is non-blocking, simulate blocking input 5995 sleepUs = (framesToRead * 1000000LL) / mSampleRate; 5996 } 5997 // otherwise use the HAL / AudioStreamIn directly 5998 } else { 5999 ssize_t bytesRead = mInput->stream->read(mInput->stream, 6000 (uint8_t*)mRsmpInBuffer + rear * mFrameSize, mBufferSize); 6001 if (bytesRead < 0) { 6002 framesRead = bytesRead; 6003 } else { 6004 framesRead = bytesRead / mFrameSize; 6005 } 6006 } 6007 6008 // Update server timestamp with server stats 6009 // systemTime() is optional if the hardware supports timestamps. 6010 mTimestamp.mPosition[ExtendedTimestamp::LOCATION_SERVER] += framesRead; 6011 mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_SERVER] = systemTime(); 6012 6013 // Update server timestamp with kernel stats 6014 if (mInput->stream->get_capture_position != nullptr) { 6015 int64_t position, time; 6016 int ret = mInput->stream->get_capture_position(mInput->stream, &position, &time); 6017 if (ret == NO_ERROR) { 6018 mTimestamp.mPosition[ExtendedTimestamp::LOCATION_KERNEL] = position; 6019 mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL] = time; 6020 // Note: In general record buffers should tend to be empty in 6021 // a properly running pipeline. 6022 // 6023 // Also, it is not advantageous to call get_presentation_position during the read 6024 // as the read obtains a lock, preventing the timestamp call from executing. 6025 } 6026 } 6027 // Use this to track timestamp information 6028 // ALOGD("%s", mTimestamp.toString().c_str()); 6029 6030 if (framesRead < 0 || (framesRead == 0 && mPipeSource == 0)) { 6031 ALOGE("read failed: framesRead=%d", framesRead); 6032 // Force input into standby so that it tries to recover at next read attempt 6033 inputStandBy(); 6034 sleepUs = kRecordThreadSleepUs; 6035 } 6036 if (framesRead <= 0) { 6037 goto unlock; 6038 } 6039 ALOG_ASSERT(framesRead > 0); 6040 6041 if (mTeeSink != 0) { 6042 (void) mTeeSink->write((uint8_t*)mRsmpInBuffer + rear * mFrameSize, framesRead); 6043 } 6044 // If destination is non-contiguous, we now correct for reading past end of buffer. 6045 { 6046 size_t part1 = mRsmpInFramesP2 - rear; 6047 if ((size_t) framesRead > part1) { 6048 memcpy(mRsmpInBuffer, (uint8_t*)mRsmpInBuffer + mRsmpInFramesP2 * mFrameSize, 6049 (framesRead - part1) * mFrameSize); 6050 } 6051 } 6052 rear = mRsmpInRear += framesRead; 6053 6054 size = activeTracks.size(); 6055 // loop over each active track 6056 for (size_t i = 0; i < size; i++) { 6057 activeTrack = activeTracks[i]; 6058 6059 // skip fast tracks, as those are handled directly by FastCapture 6060 if (activeTrack->isFastTrack()) { 6061 continue; 6062 } 6063 6064 // TODO: This code probably should be moved to RecordTrack. 6065 // TODO: Update the activeTrack buffer converter in case of reconfigure. 6066 6067 enum { 6068 OVERRUN_UNKNOWN, 6069 OVERRUN_TRUE, 6070 OVERRUN_FALSE 6071 } overrun = OVERRUN_UNKNOWN; 6072 6073 // loop over getNextBuffer to handle circular sink 6074 for (;;) { 6075 6076 activeTrack->mSink.frameCount = ~0; 6077 status_t status = activeTrack->getNextBuffer(&activeTrack->mSink); 6078 size_t framesOut = activeTrack->mSink.frameCount; 6079 LOG_ALWAYS_FATAL_IF((status == OK) != (framesOut > 0)); 6080 6081 // check available frames and handle overrun conditions 6082 // if the record track isn't draining fast enough. 6083 bool hasOverrun; 6084 size_t framesIn; 6085 activeTrack->mResamplerBufferProvider->sync(&framesIn, &hasOverrun); 6086 if (hasOverrun) { 6087 overrun = OVERRUN_TRUE; 6088 } 6089 if (framesOut == 0 || framesIn == 0) { 6090 break; 6091 } 6092 6093 // Don't allow framesOut to be larger than what is possible with resampling 6094 // from framesIn. 6095 // This isn't strictly necessary but helps limit buffer resizing in 6096 // RecordBufferConverter. TODO: remove when no longer needed. 6097 framesOut = min(framesOut, 6098 destinationFramesPossible( 6099 framesIn, mSampleRate, activeTrack->mSampleRate)); 6100 // process frames from the RecordThread buffer provider to the RecordTrack buffer 6101 framesOut = activeTrack->mRecordBufferConverter->convert( 6102 activeTrack->mSink.raw, activeTrack->mResamplerBufferProvider, framesOut); 6103 6104 if (framesOut > 0 && (overrun == OVERRUN_UNKNOWN)) { 6105 overrun = OVERRUN_FALSE; 6106 } 6107 6108 if (activeTrack->mFramesToDrop == 0) { 6109 if (framesOut > 0) { 6110 activeTrack->mSink.frameCount = framesOut; 6111 activeTrack->releaseBuffer(&activeTrack->mSink); 6112 } 6113 } else { 6114 // FIXME could do a partial drop of framesOut 6115 if (activeTrack->mFramesToDrop > 0) { 6116 activeTrack->mFramesToDrop -= framesOut; 6117 if (activeTrack->mFramesToDrop <= 0) { 6118 activeTrack->clearSyncStartEvent(); 6119 } 6120 } else { 6121 activeTrack->mFramesToDrop += framesOut; 6122 if (activeTrack->mFramesToDrop >= 0 || activeTrack->mSyncStartEvent == 0 || 6123 activeTrack->mSyncStartEvent->isCancelled()) { 6124 ALOGW("Synced record %s, session %d, trigger session %d", 6125 (activeTrack->mFramesToDrop >= 0) ? "timed out" : "cancelled", 6126 activeTrack->sessionId(), 6127 (activeTrack->mSyncStartEvent != 0) ? 6128 activeTrack->mSyncStartEvent->triggerSession() : 0); 6129 activeTrack->clearSyncStartEvent(); 6130 } 6131 } 6132 } 6133 6134 if (framesOut == 0) { 6135 break; 6136 } 6137 } 6138 6139 switch (overrun) { 6140 case OVERRUN_TRUE: 6141 // client isn't retrieving buffers fast enough 6142 if (!activeTrack->setOverflow()) { 6143 nsecs_t now = systemTime(); 6144 // FIXME should lastWarning per track? 6145 if ((now - lastWarning) > kWarningThrottleNs) { 6146 ALOGW("RecordThread: buffer overflow"); 6147 lastWarning = now; 6148 } 6149 } 6150 break; 6151 case OVERRUN_FALSE: 6152 activeTrack->clearOverflow(); 6153 break; 6154 case OVERRUN_UNKNOWN: 6155 break; 6156 } 6157 6158 // update frame information and push timestamp out 6159 activeTrack->updateTrackFrameInfo( 6160 activeTrack->mAudioRecordServerProxy->framesReleased(), 6161 mTimestamp.mPosition[ExtendedTimestamp::LOCATION_SERVER], 6162 mSampleRate, mTimestamp); 6163 } 6164 6165unlock: 6166 // enable changes in effect chain 6167 unlockEffectChains(effectChains); 6168 // effectChains doesn't need to be cleared, since it is cleared by destructor at scope end 6169 } 6170 6171 standbyIfNotAlreadyInStandby(); 6172 6173 { 6174 Mutex::Autolock _l(mLock); 6175 for (size_t i = 0; i < mTracks.size(); i++) { 6176 sp<RecordTrack> track = mTracks[i]; 6177 track->invalidate(); 6178 } 6179 mActiveTracks.clear(); 6180 mActiveTracksGen++; 6181 mStartStopCond.broadcast(); 6182 } 6183 6184 releaseWakeLock(); 6185 6186 ALOGV("RecordThread %p exiting", this); 6187 return false; 6188} 6189 6190void AudioFlinger::RecordThread::standbyIfNotAlreadyInStandby() 6191{ 6192 if (!mStandby) { 6193 inputStandBy(); 6194 mStandby = true; 6195 } 6196} 6197 6198void AudioFlinger::RecordThread::inputStandBy() 6199{ 6200 // Idle the fast capture if it's currently running 6201 if (mFastCapture != 0) { 6202 FastCaptureStateQueue *sq = mFastCapture->sq(); 6203 FastCaptureState *state = sq->begin(); 6204 if (!(state->mCommand & FastCaptureState::IDLE)) { 6205 state->mCommand = FastCaptureState::COLD_IDLE; 6206 state->mColdFutexAddr = &mFastCaptureFutex; 6207 state->mColdGen++; 6208 mFastCaptureFutex = 0; 6209 sq->end(); 6210 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 6211 sq->push(FastCaptureStateQueue::BLOCK_UNTIL_ACKED); 6212#if 0 6213 if (kUseFastCapture == FastCapture_Dynamic) { 6214 // FIXME 6215 } 6216#endif 6217#ifdef AUDIO_WATCHDOG 6218 // FIXME 6219#endif 6220 } else { 6221 sq->end(false /*didModify*/); 6222 } 6223 } 6224 mInput->stream->common.standby(&mInput->stream->common); 6225} 6226 6227// RecordThread::createRecordTrack_l() must be called with AudioFlinger::mLock held 6228sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 6229 const sp<AudioFlinger::Client>& client, 6230 uint32_t sampleRate, 6231 audio_format_t format, 6232 audio_channel_mask_t channelMask, 6233 size_t *pFrameCount, 6234 int sessionId, 6235 size_t *notificationFrames, 6236 int uid, 6237 IAudioFlinger::track_flags_t *flags, 6238 pid_t tid, 6239 status_t *status) 6240{ 6241 size_t frameCount = *pFrameCount; 6242 sp<RecordTrack> track; 6243 status_t lStatus; 6244 6245 // client expresses a preference for FAST, but we get the final say 6246 if (*flags & IAudioFlinger::TRACK_FAST) { 6247 if ( 6248 // we formerly checked for a callback handler (non-0 tid), 6249 // but that is no longer required for TRANSFER_OBTAIN mode 6250 // 6251 // frame count is not specified, or is exactly the pipe depth 6252 ((frameCount == 0) || (frameCount == mPipeFramesP2)) && 6253 // PCM data 6254 audio_is_linear_pcm(format) && 6255 // native format 6256 (format == mFormat) && 6257 // native channel mask 6258 (channelMask == mChannelMask) && 6259 // native hardware sample rate 6260 (sampleRate == mSampleRate) && 6261 // record thread has an associated fast capture 6262 hasFastCapture() && 6263 // there are sufficient fast track slots available 6264 mFastTrackAvail 6265 ) { 6266 ALOGV("AUDIO_INPUT_FLAG_FAST accepted: frameCount=%u mFrameCount=%u", 6267 frameCount, mFrameCount); 6268 } else { 6269 ALOGV("AUDIO_INPUT_FLAG_FAST denied: frameCount=%u mFrameCount=%u mPipeFramesP2=%u " 6270 "format=%#x isLinear=%d channelMask=%#x sampleRate=%u mSampleRate=%u " 6271 "hasFastCapture=%d tid=%d mFastTrackAvail=%d", 6272 frameCount, mFrameCount, mPipeFramesP2, 6273 format, audio_is_linear_pcm(format), channelMask, sampleRate, mSampleRate, 6274 hasFastCapture(), tid, mFastTrackAvail); 6275 *flags &= ~IAudioFlinger::TRACK_FAST; 6276 } 6277 } 6278 6279 // compute track buffer size in frames, and suggest the notification frame count 6280 if (*flags & IAudioFlinger::TRACK_FAST) { 6281 // fast track: frame count is exactly the pipe depth 6282 frameCount = mPipeFramesP2; 6283 // ignore requested notificationFrames, and always notify exactly once every HAL buffer 6284 *notificationFrames = mFrameCount; 6285 } else { 6286 // not fast track: max notification period is resampled equivalent of one HAL buffer time 6287 // or 20 ms if there is a fast capture 6288 // TODO This could be a roundupRatio inline, and const 6289 size_t maxNotificationFrames = ((int64_t) (hasFastCapture() ? mSampleRate/50 : mFrameCount) 6290 * sampleRate + mSampleRate - 1) / mSampleRate; 6291 // minimum number of notification periods is at least kMinNotifications, 6292 // and at least kMinMs rounded up to a whole notification period (minNotificationsByMs) 6293 static const size_t kMinNotifications = 3; 6294 static const uint32_t kMinMs = 30; 6295 // TODO This could be a roundupRatio inline 6296 const size_t minFramesByMs = (sampleRate * kMinMs + 1000 - 1) / 1000; 6297 // TODO This could be a roundupRatio inline 6298 const size_t minNotificationsByMs = (minFramesByMs + maxNotificationFrames - 1) / 6299 maxNotificationFrames; 6300 const size_t minFrameCount = maxNotificationFrames * 6301 max(kMinNotifications, minNotificationsByMs); 6302 frameCount = max(frameCount, minFrameCount); 6303 if (*notificationFrames == 0 || *notificationFrames > maxNotificationFrames) { 6304 *notificationFrames = maxNotificationFrames; 6305 } 6306 } 6307 *pFrameCount = frameCount; 6308 6309 lStatus = initCheck(); 6310 if (lStatus != NO_ERROR) { 6311 ALOGE("createRecordTrack_l() audio driver not initialized"); 6312 goto Exit; 6313 } 6314 6315 { // scope for mLock 6316 Mutex::Autolock _l(mLock); 6317 6318 track = new RecordTrack(this, client, sampleRate, 6319 format, channelMask, frameCount, NULL, sessionId, uid, 6320 *flags, TrackBase::TYPE_DEFAULT); 6321 6322 lStatus = track->initCheck(); 6323 if (lStatus != NO_ERROR) { 6324 ALOGE("createRecordTrack_l() initCheck failed %d; no control block?", lStatus); 6325 // track must be cleared from the caller as the caller has the AF lock 6326 goto Exit; 6327 } 6328 mTracks.add(track); 6329 6330 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 6331 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 6332 mAudioFlinger->btNrecIsOff(); 6333 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 6334 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 6335 6336 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 6337 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 6338 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 6339 // so ask activity manager to do this on our behalf 6340 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 6341 } 6342 } 6343 6344 lStatus = NO_ERROR; 6345 6346Exit: 6347 *status = lStatus; 6348 return track; 6349} 6350 6351status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, 6352 AudioSystem::sync_event_t event, 6353 int triggerSession) 6354{ 6355 ALOGV("RecordThread::start event %d, triggerSession %d", event, triggerSession); 6356 sp<ThreadBase> strongMe = this; 6357 status_t status = NO_ERROR; 6358 6359 if (event == AudioSystem::SYNC_EVENT_NONE) { 6360 recordTrack->clearSyncStartEvent(); 6361 } else if (event != AudioSystem::SYNC_EVENT_SAME) { 6362 recordTrack->mSyncStartEvent = mAudioFlinger->createSyncEvent(event, 6363 triggerSession, 6364 recordTrack->sessionId(), 6365 syncStartEventCallback, 6366 recordTrack); 6367 // Sync event can be cancelled by the trigger session if the track is not in a 6368 // compatible state in which case we start record immediately 6369 if (recordTrack->mSyncStartEvent->isCancelled()) { 6370 recordTrack->clearSyncStartEvent(); 6371 } else { 6372 // do not wait for the event for more than AudioSystem::kSyncRecordStartTimeOutMs 6373 recordTrack->mFramesToDrop = - 6374 ((AudioSystem::kSyncRecordStartTimeOutMs * recordTrack->mSampleRate) / 1000); 6375 } 6376 } 6377 6378 { 6379 // This section is a rendezvous between binder thread executing start() and RecordThread 6380 AutoMutex lock(mLock); 6381 if (mActiveTracks.indexOf(recordTrack) >= 0) { 6382 if (recordTrack->mState == TrackBase::PAUSING) { 6383 ALOGV("active record track PAUSING -> ACTIVE"); 6384 recordTrack->mState = TrackBase::ACTIVE; 6385 } else { 6386 ALOGV("active record track state %d", recordTrack->mState); 6387 } 6388 return status; 6389 } 6390 6391 // TODO consider other ways of handling this, such as changing the state to :STARTING and 6392 // adding the track to mActiveTracks after returning from AudioSystem::startInput(), 6393 // or using a separate command thread 6394 recordTrack->mState = TrackBase::STARTING_1; 6395 mActiveTracks.add(recordTrack); 6396 mActiveTracksGen++; 6397 status_t status = NO_ERROR; 6398 if (recordTrack->isExternalTrack()) { 6399 mLock.unlock(); 6400 status = AudioSystem::startInput(mId, (audio_session_t)recordTrack->sessionId()); 6401 mLock.lock(); 6402 // FIXME should verify that recordTrack is still in mActiveTracks 6403 if (status != NO_ERROR) { 6404 mActiveTracks.remove(recordTrack); 6405 mActiveTracksGen++; 6406 recordTrack->clearSyncStartEvent(); 6407 ALOGV("RecordThread::start error %d", status); 6408 return status; 6409 } 6410 } 6411 // Catch up with current buffer indices if thread is already running. 6412 // This is what makes a new client discard all buffered data. If the track's mRsmpInFront 6413 // was initialized to some value closer to the thread's mRsmpInFront, then the track could 6414 // see previously buffered data before it called start(), but with greater risk of overrun. 6415 6416 recordTrack->mResamplerBufferProvider->reset(); 6417 // clear any converter state as new data will be discontinuous 6418 recordTrack->mRecordBufferConverter->reset(); 6419 recordTrack->mState = TrackBase::STARTING_2; 6420 // signal thread to start 6421 mWaitWorkCV.broadcast(); 6422 if (mActiveTracks.indexOf(recordTrack) < 0) { 6423 ALOGV("Record failed to start"); 6424 status = BAD_VALUE; 6425 goto startError; 6426 } 6427 return status; 6428 } 6429 6430startError: 6431 if (recordTrack->isExternalTrack()) { 6432 AudioSystem::stopInput(mId, (audio_session_t)recordTrack->sessionId()); 6433 } 6434 recordTrack->clearSyncStartEvent(); 6435 // FIXME I wonder why we do not reset the state here? 6436 return status; 6437} 6438 6439void AudioFlinger::RecordThread::syncStartEventCallback(const wp<SyncEvent>& event) 6440{ 6441 sp<SyncEvent> strongEvent = event.promote(); 6442 6443 if (strongEvent != 0) { 6444 sp<RefBase> ptr = strongEvent->cookie().promote(); 6445 if (ptr != 0) { 6446 RecordTrack *recordTrack = (RecordTrack *)ptr.get(); 6447 recordTrack->handleSyncStartEvent(strongEvent); 6448 } 6449 } 6450} 6451 6452bool AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 6453 ALOGV("RecordThread::stop"); 6454 AutoMutex _l(mLock); 6455 if (mActiveTracks.indexOf(recordTrack) != 0 || recordTrack->mState == TrackBase::PAUSING) { 6456 return false; 6457 } 6458 // note that threadLoop may still be processing the track at this point [without lock] 6459 recordTrack->mState = TrackBase::PAUSING; 6460 // do not wait for mStartStopCond if exiting 6461 if (exitPending()) { 6462 return true; 6463 } 6464 // FIXME incorrect usage of wait: no explicit predicate or loop 6465 mStartStopCond.wait(mLock); 6466 // if we have been restarted, recordTrack is in mActiveTracks here 6467 if (exitPending() || mActiveTracks.indexOf(recordTrack) != 0) { 6468 ALOGV("Record stopped OK"); 6469 return true; 6470 } 6471 return false; 6472} 6473 6474bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event __unused) const 6475{ 6476 return false; 6477} 6478 6479status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event __unused) 6480{ 6481#if 0 // This branch is currently dead code, but is preserved in case it will be needed in future 6482 if (!isValidSyncEvent(event)) { 6483 return BAD_VALUE; 6484 } 6485 6486 int eventSession = event->triggerSession(); 6487 status_t ret = NAME_NOT_FOUND; 6488 6489 Mutex::Autolock _l(mLock); 6490 6491 for (size_t i = 0; i < mTracks.size(); i++) { 6492 sp<RecordTrack> track = mTracks[i]; 6493 if (eventSession == track->sessionId()) { 6494 (void) track->setSyncEvent(event); 6495 ret = NO_ERROR; 6496 } 6497 } 6498 return ret; 6499#else 6500 return BAD_VALUE; 6501#endif 6502} 6503 6504// destroyTrack_l() must be called with ThreadBase::mLock held 6505void AudioFlinger::RecordThread::destroyTrack_l(const sp<RecordTrack>& track) 6506{ 6507 track->terminate(); 6508 track->mState = TrackBase::STOPPED; 6509 // active tracks are removed by threadLoop() 6510 if (mActiveTracks.indexOf(track) < 0) { 6511 removeTrack_l(track); 6512 } 6513} 6514 6515void AudioFlinger::RecordThread::removeTrack_l(const sp<RecordTrack>& track) 6516{ 6517 mTracks.remove(track); 6518 // need anything related to effects here? 6519 if (track->isFastTrack()) { 6520 ALOG_ASSERT(!mFastTrackAvail); 6521 mFastTrackAvail = true; 6522 } 6523} 6524 6525void AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 6526{ 6527 dumpInternals(fd, args); 6528 dumpTracks(fd, args); 6529 dumpEffectChains(fd, args); 6530} 6531 6532void AudioFlinger::RecordThread::dumpInternals(int fd, const Vector<String16>& args) 6533{ 6534 dprintf(fd, "\nInput thread %p:\n", this); 6535 6536 dumpBase(fd, args); 6537 6538 if (mActiveTracks.size() == 0) { 6539 dprintf(fd, " No active record clients\n"); 6540 } 6541 dprintf(fd, " Fast capture thread: %s\n", hasFastCapture() ? "yes" : "no"); 6542 dprintf(fd, " Fast track available: %s\n", mFastTrackAvail ? "yes" : "no"); 6543 6544 // Make a non-atomic copy of fast capture dump state so it won't change underneath us 6545 // while we are dumping it. It may be inconsistent, but it won't mutate! 6546 // This is a large object so we place it on the heap. 6547 // FIXME 25972958: Need an intelligent copy constructor that does not touch unused pages. 6548 const FastCaptureDumpState *copy = new FastCaptureDumpState(mFastCaptureDumpState); 6549 copy->dump(fd); 6550 delete copy; 6551} 6552 6553void AudioFlinger::RecordThread::dumpTracks(int fd, const Vector<String16>& args __unused) 6554{ 6555 const size_t SIZE = 256; 6556 char buffer[SIZE]; 6557 String8 result; 6558 6559 size_t numtracks = mTracks.size(); 6560 size_t numactive = mActiveTracks.size(); 6561 size_t numactiveseen = 0; 6562 dprintf(fd, " %d Tracks", numtracks); 6563 if (numtracks) { 6564 dprintf(fd, " of which %d are active\n", numactive); 6565 RecordTrack::appendDumpHeader(result); 6566 for (size_t i = 0; i < numtracks ; ++i) { 6567 sp<RecordTrack> track = mTracks[i]; 6568 if (track != 0) { 6569 bool active = mActiveTracks.indexOf(track) >= 0; 6570 if (active) { 6571 numactiveseen++; 6572 } 6573 track->dump(buffer, SIZE, active); 6574 result.append(buffer); 6575 } 6576 } 6577 } else { 6578 dprintf(fd, "\n"); 6579 } 6580 6581 if (numactiveseen != numactive) { 6582 snprintf(buffer, SIZE, " The following tracks are in the active list but" 6583 " not in the track list\n"); 6584 result.append(buffer); 6585 RecordTrack::appendDumpHeader(result); 6586 for (size_t i = 0; i < numactive; ++i) { 6587 sp<RecordTrack> track = mActiveTracks[i]; 6588 if (mTracks.indexOf(track) < 0) { 6589 track->dump(buffer, SIZE, true); 6590 result.append(buffer); 6591 } 6592 } 6593 6594 } 6595 write(fd, result.string(), result.size()); 6596} 6597 6598 6599void AudioFlinger::RecordThread::ResamplerBufferProvider::reset() 6600{ 6601 sp<ThreadBase> threadBase = mRecordTrack->mThread.promote(); 6602 RecordThread *recordThread = (RecordThread *) threadBase.get(); 6603 mRsmpInFront = recordThread->mRsmpInRear; 6604 mRsmpInUnrel = 0; 6605} 6606 6607void AudioFlinger::RecordThread::ResamplerBufferProvider::sync( 6608 size_t *framesAvailable, bool *hasOverrun) 6609{ 6610 sp<ThreadBase> threadBase = mRecordTrack->mThread.promote(); 6611 RecordThread *recordThread = (RecordThread *) threadBase.get(); 6612 const int32_t rear = recordThread->mRsmpInRear; 6613 const int32_t front = mRsmpInFront; 6614 const ssize_t filled = rear - front; 6615 6616 size_t framesIn; 6617 bool overrun = false; 6618 if (filled < 0) { 6619 // should not happen, but treat like a massive overrun and re-sync 6620 framesIn = 0; 6621 mRsmpInFront = rear; 6622 overrun = true; 6623 } else if ((size_t) filled <= recordThread->mRsmpInFrames) { 6624 framesIn = (size_t) filled; 6625 } else { 6626 // client is not keeping up with server, but give it latest data 6627 framesIn = recordThread->mRsmpInFrames; 6628 mRsmpInFront = /* front = */ rear - framesIn; 6629 overrun = true; 6630 } 6631 if (framesAvailable != NULL) { 6632 *framesAvailable = framesIn; 6633 } 6634 if (hasOverrun != NULL) { 6635 *hasOverrun = overrun; 6636 } 6637} 6638 6639// AudioBufferProvider interface 6640status_t AudioFlinger::RecordThread::ResamplerBufferProvider::getNextBuffer( 6641 AudioBufferProvider::Buffer* buffer) 6642{ 6643 sp<ThreadBase> threadBase = mRecordTrack->mThread.promote(); 6644 if (threadBase == 0) { 6645 buffer->frameCount = 0; 6646 buffer->raw = NULL; 6647 return NOT_ENOUGH_DATA; 6648 } 6649 RecordThread *recordThread = (RecordThread *) threadBase.get(); 6650 int32_t rear = recordThread->mRsmpInRear; 6651 int32_t front = mRsmpInFront; 6652 ssize_t filled = rear - front; 6653 // FIXME should not be P2 (don't want to increase latency) 6654 // FIXME if client not keeping up, discard 6655 LOG_ALWAYS_FATAL_IF(!(0 <= filled && (size_t) filled <= recordThread->mRsmpInFrames)); 6656 // 'filled' may be non-contiguous, so return only the first contiguous chunk 6657 front &= recordThread->mRsmpInFramesP2 - 1; 6658 size_t part1 = recordThread->mRsmpInFramesP2 - front; 6659 if (part1 > (size_t) filled) { 6660 part1 = filled; 6661 } 6662 size_t ask = buffer->frameCount; 6663 ALOG_ASSERT(ask > 0); 6664 if (part1 > ask) { 6665 part1 = ask; 6666 } 6667 if (part1 == 0) { 6668 // out of data is fine since the resampler will return a short-count. 6669 buffer->raw = NULL; 6670 buffer->frameCount = 0; 6671 mRsmpInUnrel = 0; 6672 return NOT_ENOUGH_DATA; 6673 } 6674 6675 buffer->raw = (uint8_t*)recordThread->mRsmpInBuffer + front * recordThread->mFrameSize; 6676 buffer->frameCount = part1; 6677 mRsmpInUnrel = part1; 6678 return NO_ERROR; 6679} 6680 6681// AudioBufferProvider interface 6682void AudioFlinger::RecordThread::ResamplerBufferProvider::releaseBuffer( 6683 AudioBufferProvider::Buffer* buffer) 6684{ 6685 size_t stepCount = buffer->frameCount; 6686 if (stepCount == 0) { 6687 return; 6688 } 6689 ALOG_ASSERT(stepCount <= mRsmpInUnrel); 6690 mRsmpInUnrel -= stepCount; 6691 mRsmpInFront += stepCount; 6692 buffer->raw = NULL; 6693 buffer->frameCount = 0; 6694} 6695 6696AudioFlinger::RecordThread::RecordBufferConverter::RecordBufferConverter( 6697 audio_channel_mask_t srcChannelMask, audio_format_t srcFormat, 6698 uint32_t srcSampleRate, 6699 audio_channel_mask_t dstChannelMask, audio_format_t dstFormat, 6700 uint32_t dstSampleRate) : 6701 mSrcChannelMask(AUDIO_CHANNEL_INVALID), // updateParameters will set following vars 6702 // mSrcFormat 6703 // mSrcSampleRate 6704 // mDstChannelMask 6705 // mDstFormat 6706 // mDstSampleRate 6707 // mSrcChannelCount 6708 // mDstChannelCount 6709 // mDstFrameSize 6710 mBuf(NULL), mBufFrames(0), mBufFrameSize(0), 6711 mResampler(NULL), 6712 mIsLegacyDownmix(false), 6713 mIsLegacyUpmix(false), 6714 mRequiresFloat(false), 6715 mInputConverterProvider(NULL) 6716{ 6717 (void)updateParameters(srcChannelMask, srcFormat, srcSampleRate, 6718 dstChannelMask, dstFormat, dstSampleRate); 6719} 6720 6721AudioFlinger::RecordThread::RecordBufferConverter::~RecordBufferConverter() { 6722 free(mBuf); 6723 delete mResampler; 6724 delete mInputConverterProvider; 6725} 6726 6727size_t AudioFlinger::RecordThread::RecordBufferConverter::convert(void *dst, 6728 AudioBufferProvider *provider, size_t frames) 6729{ 6730 if (mInputConverterProvider != NULL) { 6731 mInputConverterProvider->setBufferProvider(provider); 6732 provider = mInputConverterProvider; 6733 } 6734 6735 if (mResampler == NULL) { 6736 ALOGVV("NO RESAMPLING sampleRate:%u mSrcFormat:%#x mDstFormat:%#x", 6737 mSrcSampleRate, mSrcFormat, mDstFormat); 6738 6739 AudioBufferProvider::Buffer buffer; 6740 for (size_t i = frames; i > 0; ) { 6741 buffer.frameCount = i; 6742 status_t status = provider->getNextBuffer(&buffer); 6743 if (status != OK || buffer.frameCount == 0) { 6744 frames -= i; // cannot fill request. 6745 break; 6746 } 6747 // format convert to destination buffer 6748 convertNoResampler(dst, buffer.raw, buffer.frameCount); 6749 6750 dst = (int8_t*)dst + buffer.frameCount * mDstFrameSize; 6751 i -= buffer.frameCount; 6752 provider->releaseBuffer(&buffer); 6753 } 6754 } else { 6755 ALOGVV("RESAMPLING mSrcSampleRate:%u mDstSampleRate:%u mSrcFormat:%#x mDstFormat:%#x", 6756 mSrcSampleRate, mDstSampleRate, mSrcFormat, mDstFormat); 6757 6758 // reallocate buffer if needed 6759 if (mBufFrameSize != 0 && mBufFrames < frames) { 6760 free(mBuf); 6761 mBufFrames = frames; 6762 (void)posix_memalign(&mBuf, 32, mBufFrames * mBufFrameSize); 6763 } 6764 // resampler accumulates, but we only have one source track 6765 memset(mBuf, 0, frames * mBufFrameSize); 6766 frames = mResampler->resample((int32_t*)mBuf, frames, provider); 6767 // format convert to destination buffer 6768 convertResampler(dst, mBuf, frames); 6769 } 6770 return frames; 6771} 6772 6773status_t AudioFlinger::RecordThread::RecordBufferConverter::updateParameters( 6774 audio_channel_mask_t srcChannelMask, audio_format_t srcFormat, 6775 uint32_t srcSampleRate, 6776 audio_channel_mask_t dstChannelMask, audio_format_t dstFormat, 6777 uint32_t dstSampleRate) 6778{ 6779 // quick evaluation if there is any change. 6780 if (mSrcFormat == srcFormat 6781 && mSrcChannelMask == srcChannelMask 6782 && mSrcSampleRate == srcSampleRate 6783 && mDstFormat == dstFormat 6784 && mDstChannelMask == dstChannelMask 6785 && mDstSampleRate == dstSampleRate) { 6786 return NO_ERROR; 6787 } 6788 6789 ALOGV("RecordBufferConverter updateParameters srcMask:%#x dstMask:%#x" 6790 " srcFormat:%#x dstFormat:%#x srcRate:%u dstRate:%u", 6791 srcChannelMask, dstChannelMask, srcFormat, dstFormat, srcSampleRate, dstSampleRate); 6792 const bool valid = 6793 audio_is_input_channel(srcChannelMask) 6794 && audio_is_input_channel(dstChannelMask) 6795 && audio_is_valid_format(srcFormat) && audio_is_linear_pcm(srcFormat) 6796 && audio_is_valid_format(dstFormat) && audio_is_linear_pcm(dstFormat) 6797 && (srcSampleRate <= dstSampleRate * AUDIO_RESAMPLER_DOWN_RATIO_MAX) 6798 ; // no upsampling checks for now 6799 if (!valid) { 6800 return BAD_VALUE; 6801 } 6802 6803 mSrcFormat = srcFormat; 6804 mSrcChannelMask = srcChannelMask; 6805 mSrcSampleRate = srcSampleRate; 6806 mDstFormat = dstFormat; 6807 mDstChannelMask = dstChannelMask; 6808 mDstSampleRate = dstSampleRate; 6809 6810 // compute derived parameters 6811 mSrcChannelCount = audio_channel_count_from_in_mask(srcChannelMask); 6812 mDstChannelCount = audio_channel_count_from_in_mask(dstChannelMask); 6813 mDstFrameSize = mDstChannelCount * audio_bytes_per_sample(mDstFormat); 6814 6815 // do we need to resample? 6816 delete mResampler; 6817 mResampler = NULL; 6818 if (mSrcSampleRate != mDstSampleRate) { 6819 mResampler = AudioResampler::create(AUDIO_FORMAT_PCM_FLOAT, 6820 mSrcChannelCount, mDstSampleRate); 6821 mResampler->setSampleRate(mSrcSampleRate); 6822 mResampler->setVolume(AudioMixer::UNITY_GAIN_FLOAT, AudioMixer::UNITY_GAIN_FLOAT); 6823 } 6824 6825 // are we running legacy channel conversion modes? 6826 mIsLegacyDownmix = (mSrcChannelMask == AUDIO_CHANNEL_IN_STEREO 6827 || mSrcChannelMask == AUDIO_CHANNEL_IN_FRONT_BACK) 6828 && mDstChannelMask == AUDIO_CHANNEL_IN_MONO; 6829 mIsLegacyUpmix = mSrcChannelMask == AUDIO_CHANNEL_IN_MONO 6830 && (mDstChannelMask == AUDIO_CHANNEL_IN_STEREO 6831 || mDstChannelMask == AUDIO_CHANNEL_IN_FRONT_BACK); 6832 6833 // do we need to process in float? 6834 mRequiresFloat = mResampler != NULL || mIsLegacyDownmix || mIsLegacyUpmix; 6835 6836 // do we need a staging buffer to convert for destination (we can still optimize this)? 6837 // we use mBufFrameSize > 0 to indicate both frame size as well as buffer necessity 6838 if (mResampler != NULL) { 6839 mBufFrameSize = max(mSrcChannelCount, FCC_2) 6840 * audio_bytes_per_sample(AUDIO_FORMAT_PCM_FLOAT); 6841 } else if (mIsLegacyUpmix || mIsLegacyDownmix) { // legacy modes always float 6842 mBufFrameSize = mDstChannelCount * audio_bytes_per_sample(AUDIO_FORMAT_PCM_FLOAT); 6843 } else if (mSrcChannelMask != mDstChannelMask && mDstFormat != mSrcFormat) { 6844 mBufFrameSize = mDstChannelCount * audio_bytes_per_sample(mSrcFormat); 6845 } else { 6846 mBufFrameSize = 0; 6847 } 6848 mBufFrames = 0; // force the buffer to be resized. 6849 6850 // do we need an input converter buffer provider to give us float? 6851 delete mInputConverterProvider; 6852 mInputConverterProvider = NULL; 6853 if (mRequiresFloat && mSrcFormat != AUDIO_FORMAT_PCM_FLOAT) { 6854 mInputConverterProvider = new ReformatBufferProvider( 6855 audio_channel_count_from_in_mask(mSrcChannelMask), 6856 mSrcFormat, 6857 AUDIO_FORMAT_PCM_FLOAT, 6858 256 /* provider buffer frame count */); 6859 } 6860 6861 // do we need a remixer to do channel mask conversion 6862 if (!mIsLegacyDownmix && !mIsLegacyUpmix && mSrcChannelMask != mDstChannelMask) { 6863 (void) memcpy_by_index_array_initialization_from_channel_mask( 6864 mIdxAry, ARRAY_SIZE(mIdxAry), mDstChannelMask, mSrcChannelMask); 6865 } 6866 return NO_ERROR; 6867} 6868 6869void AudioFlinger::RecordThread::RecordBufferConverter::convertNoResampler( 6870 void *dst, const void *src, size_t frames) 6871{ 6872 // src is native type unless there is legacy upmix or downmix, whereupon it is float. 6873 if (mBufFrameSize != 0 && mBufFrames < frames) { 6874 free(mBuf); 6875 mBufFrames = frames; 6876 (void)posix_memalign(&mBuf, 32, mBufFrames * mBufFrameSize); 6877 } 6878 // do we need to do legacy upmix and downmix? 6879 if (mIsLegacyUpmix || mIsLegacyDownmix) { 6880 void *dstBuf = mBuf != NULL ? mBuf : dst; 6881 if (mIsLegacyUpmix) { 6882 upmix_to_stereo_float_from_mono_float((float *)dstBuf, 6883 (const float *)src, frames); 6884 } else /*mIsLegacyDownmix */ { 6885 downmix_to_mono_float_from_stereo_float((float *)dstBuf, 6886 (const float *)src, frames); 6887 } 6888 if (mBuf != NULL) { 6889 memcpy_by_audio_format(dst, mDstFormat, mBuf, AUDIO_FORMAT_PCM_FLOAT, 6890 frames * mDstChannelCount); 6891 } 6892 return; 6893 } 6894 // do we need to do channel mask conversion? 6895 if (mSrcChannelMask != mDstChannelMask) { 6896 void *dstBuf = mBuf != NULL ? mBuf : dst; 6897 memcpy_by_index_array(dstBuf, mDstChannelCount, 6898 src, mSrcChannelCount, mIdxAry, audio_bytes_per_sample(mSrcFormat), frames); 6899 if (dstBuf == dst) { 6900 return; // format is the same 6901 } 6902 } 6903 // convert to destination buffer 6904 const void *convertBuf = mBuf != NULL ? mBuf : src; 6905 memcpy_by_audio_format(dst, mDstFormat, convertBuf, mSrcFormat, 6906 frames * mDstChannelCount); 6907} 6908 6909void AudioFlinger::RecordThread::RecordBufferConverter::convertResampler( 6910 void *dst, /*not-a-const*/ void *src, size_t frames) 6911{ 6912 // src buffer format is ALWAYS float when entering this routine 6913 if (mIsLegacyUpmix) { 6914 ; // mono to stereo already handled by resampler 6915 } else if (mIsLegacyDownmix 6916 || (mSrcChannelMask == mDstChannelMask && mSrcChannelCount == 1)) { 6917 // the resampler outputs stereo for mono input channel (a feature?) 6918 // must convert to mono 6919 downmix_to_mono_float_from_stereo_float((float *)src, 6920 (const float *)src, frames); 6921 } else if (mSrcChannelMask != mDstChannelMask) { 6922 // convert to mono channel again for channel mask conversion (could be skipped 6923 // with further optimization). 6924 if (mSrcChannelCount == 1) { 6925 downmix_to_mono_float_from_stereo_float((float *)src, 6926 (const float *)src, frames); 6927 } 6928 // convert to destination format (in place, OK as float is larger than other types) 6929 if (mDstFormat != AUDIO_FORMAT_PCM_FLOAT) { 6930 memcpy_by_audio_format(src, mDstFormat, src, AUDIO_FORMAT_PCM_FLOAT, 6931 frames * mSrcChannelCount); 6932 } 6933 // channel convert and save to dst 6934 memcpy_by_index_array(dst, mDstChannelCount, 6935 src, mSrcChannelCount, mIdxAry, audio_bytes_per_sample(mDstFormat), frames); 6936 return; 6937 } 6938 // convert to destination format and save to dst 6939 memcpy_by_audio_format(dst, mDstFormat, src, AUDIO_FORMAT_PCM_FLOAT, 6940 frames * mDstChannelCount); 6941} 6942 6943bool AudioFlinger::RecordThread::checkForNewParameter_l(const String8& keyValuePair, 6944 status_t& status) 6945{ 6946 bool reconfig = false; 6947 6948 status = NO_ERROR; 6949 6950 audio_format_t reqFormat = mFormat; 6951 uint32_t samplingRate = mSampleRate; 6952 // TODO this may change if we want to support capture from HDMI PCM multi channel (e.g on TVs). 6953 audio_channel_mask_t channelMask = audio_channel_in_mask_from_count(mChannelCount); 6954 6955 AudioParameter param = AudioParameter(keyValuePair); 6956 int value; 6957 // TODO Investigate when this code runs. Check with audio policy when a sample rate and 6958 // channel count change can be requested. Do we mandate the first client defines the 6959 // HAL sampling rate and channel count or do we allow changes on the fly? 6960 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 6961 samplingRate = value; 6962 reconfig = true; 6963 } 6964 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 6965 if (!audio_is_linear_pcm((audio_format_t) value)) { 6966 status = BAD_VALUE; 6967 } else { 6968 reqFormat = (audio_format_t) value; 6969 reconfig = true; 6970 } 6971 } 6972 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 6973 audio_channel_mask_t mask = (audio_channel_mask_t) value; 6974 if (!audio_is_input_channel(mask) || 6975 audio_channel_count_from_in_mask(mask) > FCC_8) { 6976 status = BAD_VALUE; 6977 } else { 6978 channelMask = mask; 6979 reconfig = true; 6980 } 6981 } 6982 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 6983 // do not accept frame count changes if tracks are open as the track buffer 6984 // size depends on frame count and correct behavior would not be guaranteed 6985 // if frame count is changed after track creation 6986 if (mActiveTracks.size() > 0) { 6987 status = INVALID_OPERATION; 6988 } else { 6989 reconfig = true; 6990 } 6991 } 6992 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 6993 // forward device change to effects that have requested to be 6994 // aware of attached audio device. 6995 for (size_t i = 0; i < mEffectChains.size(); i++) { 6996 mEffectChains[i]->setDevice_l(value); 6997 } 6998 6999 // store input device and output device but do not forward output device to audio HAL. 7000 // Note that status is ignored by the caller for output device 7001 // (see AudioFlinger::setParameters() 7002 if (audio_is_output_devices(value)) { 7003 mOutDevice = value; 7004 status = BAD_VALUE; 7005 } else { 7006 mInDevice = value; 7007 if (value != AUDIO_DEVICE_NONE) { 7008 mPrevInDevice = value; 7009 } 7010 // disable AEC and NS if the device is a BT SCO headset supporting those 7011 // pre processings 7012 if (mTracks.size() > 0) { 7013 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 7014 mAudioFlinger->btNrecIsOff(); 7015 for (size_t i = 0; i < mTracks.size(); i++) { 7016 sp<RecordTrack> track = mTracks[i]; 7017 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId()); 7018 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId()); 7019 } 7020 } 7021 } 7022 } 7023 if (param.getInt(String8(AudioParameter::keyInputSource), value) == NO_ERROR && 7024 mAudioSource != (audio_source_t)value) { 7025 // forward device change to effects that have requested to be 7026 // aware of attached audio device. 7027 for (size_t i = 0; i < mEffectChains.size(); i++) { 7028 mEffectChains[i]->setAudioSource_l((audio_source_t)value); 7029 } 7030 mAudioSource = (audio_source_t)value; 7031 } 7032 7033 if (status == NO_ERROR) { 7034 status = mInput->stream->common.set_parameters(&mInput->stream->common, 7035 keyValuePair.string()); 7036 if (status == INVALID_OPERATION) { 7037 inputStandBy(); 7038 status = mInput->stream->common.set_parameters(&mInput->stream->common, 7039 keyValuePair.string()); 7040 } 7041 if (reconfig) { 7042 if (status == BAD_VALUE && 7043 audio_is_linear_pcm(mInput->stream->common.get_format(&mInput->stream->common)) && 7044 audio_is_linear_pcm(reqFormat) && 7045 (mInput->stream->common.get_sample_rate(&mInput->stream->common) 7046 <= (AUDIO_RESAMPLER_DOWN_RATIO_MAX * samplingRate)) && 7047 audio_channel_count_from_in_mask( 7048 mInput->stream->common.get_channels(&mInput->stream->common)) <= FCC_8) { 7049 status = NO_ERROR; 7050 } 7051 if (status == NO_ERROR) { 7052 readInputParameters_l(); 7053 sendIoConfigEvent_l(AUDIO_INPUT_CONFIG_CHANGED); 7054 } 7055 } 7056 } 7057 7058 return reconfig; 7059} 7060 7061String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 7062{ 7063 Mutex::Autolock _l(mLock); 7064 if (initCheck() != NO_ERROR) { 7065 return String8(); 7066 } 7067 7068 char *s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 7069 const String8 out_s8(s); 7070 free(s); 7071 return out_s8; 7072} 7073 7074void AudioFlinger::RecordThread::ioConfigChanged(audio_io_config_event event, pid_t pid) { 7075 sp<AudioIoDescriptor> desc = new AudioIoDescriptor(); 7076 7077 desc->mIoHandle = mId; 7078 7079 switch (event) { 7080 case AUDIO_INPUT_OPENED: 7081 case AUDIO_INPUT_CONFIG_CHANGED: 7082 desc->mPatch = mPatch; 7083 desc->mChannelMask = mChannelMask; 7084 desc->mSamplingRate = mSampleRate; 7085 desc->mFormat = mFormat; 7086 desc->mFrameCount = mFrameCount; 7087 desc->mLatency = 0; 7088 break; 7089 7090 case AUDIO_INPUT_CLOSED: 7091 default: 7092 break; 7093 } 7094 mAudioFlinger->ioConfigChanged(event, desc, pid); 7095} 7096 7097void AudioFlinger::RecordThread::readInputParameters_l() 7098{ 7099 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 7100 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 7101 mChannelCount = audio_channel_count_from_in_mask(mChannelMask); 7102 if (mChannelCount > FCC_8) { 7103 ALOGE("HAL channel count %d > %d", mChannelCount, FCC_8); 7104 } 7105 mHALFormat = mInput->stream->common.get_format(&mInput->stream->common); 7106 mFormat = mHALFormat; 7107 if (!audio_is_linear_pcm(mFormat)) { 7108 ALOGE("HAL format %#x is not linear pcm", mFormat); 7109 } 7110 mFrameSize = audio_stream_in_frame_size(mInput->stream); 7111 mBufferSize = mInput->stream->common.get_buffer_size(&mInput->stream->common); 7112 mFrameCount = mBufferSize / mFrameSize; 7113 // This is the formula for calculating the temporary buffer size. 7114 // With 7 HAL buffers, we can guarantee ability to down-sample the input by ratio of 6:1 to 7115 // 1 full output buffer, regardless of the alignment of the available input. 7116 // The value is somewhat arbitrary, and could probably be even larger. 7117 // A larger value should allow more old data to be read after a track calls start(), 7118 // without increasing latency. 7119 // 7120 // Note this is independent of the maximum downsampling ratio permitted for capture. 7121 mRsmpInFrames = mFrameCount * 7; 7122 mRsmpInFramesP2 = roundup(mRsmpInFrames); 7123 free(mRsmpInBuffer); 7124 mRsmpInBuffer = NULL; 7125 7126 // TODO optimize audio capture buffer sizes ... 7127 // Here we calculate the size of the sliding buffer used as a source 7128 // for resampling. mRsmpInFramesP2 is currently roundup(mFrameCount * 7). 7129 // For current HAL frame counts, this is usually 2048 = 40 ms. It would 7130 // be better to have it derived from the pipe depth in the long term. 7131 // The current value is higher than necessary. However it should not add to latency. 7132 7133 // Over-allocate beyond mRsmpInFramesP2 to permit a HAL read past end of buffer 7134 size_t bufferSize = (mRsmpInFramesP2 + mFrameCount - 1) * mFrameSize; 7135 (void)posix_memalign(&mRsmpInBuffer, 32, bufferSize); 7136 memset(mRsmpInBuffer, 0, bufferSize); // if posix_memalign fails, will segv here. 7137 7138 // AudioRecord mSampleRate and mChannelCount are constant due to AudioRecord API constraints. 7139 // But if thread's mSampleRate or mChannelCount changes, how will that affect active tracks? 7140} 7141 7142uint32_t AudioFlinger::RecordThread::getInputFramesLost() 7143{ 7144 Mutex::Autolock _l(mLock); 7145 if (initCheck() != NO_ERROR) { 7146 return 0; 7147 } 7148 7149 return mInput->stream->get_input_frames_lost(mInput->stream); 7150} 7151 7152uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) const 7153{ 7154 Mutex::Autolock _l(mLock); 7155 uint32_t result = 0; 7156 if (getEffectChain_l(sessionId) != 0) { 7157 result = EFFECT_SESSION; 7158 } 7159 7160 for (size_t i = 0; i < mTracks.size(); ++i) { 7161 if (sessionId == mTracks[i]->sessionId()) { 7162 result |= TRACK_SESSION; 7163 break; 7164 } 7165 } 7166 7167 return result; 7168} 7169 7170KeyedVector<int, bool> AudioFlinger::RecordThread::sessionIds() const 7171{ 7172 KeyedVector<int, bool> ids; 7173 Mutex::Autolock _l(mLock); 7174 for (size_t j = 0; j < mTracks.size(); ++j) { 7175 sp<RecordThread::RecordTrack> track = mTracks[j]; 7176 int sessionId = track->sessionId(); 7177 if (ids.indexOfKey(sessionId) < 0) { 7178 ids.add(sessionId, true); 7179 } 7180 } 7181 return ids; 7182} 7183 7184AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 7185{ 7186 Mutex::Autolock _l(mLock); 7187 AudioStreamIn *input = mInput; 7188 mInput = NULL; 7189 return input; 7190} 7191 7192// this method must always be called either with ThreadBase mLock held or inside the thread loop 7193audio_stream_t* AudioFlinger::RecordThread::stream() const 7194{ 7195 if (mInput == NULL) { 7196 return NULL; 7197 } 7198 return &mInput->stream->common; 7199} 7200 7201status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 7202{ 7203 // only one chain per input thread 7204 if (mEffectChains.size() != 0) { 7205 ALOGW("addEffectChain_l() already one chain %p on thread %p", chain.get(), this); 7206 return INVALID_OPERATION; 7207 } 7208 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 7209 chain->setThread(this); 7210 chain->setInBuffer(NULL); 7211 chain->setOutBuffer(NULL); 7212 7213 checkSuspendOnAddEffectChain_l(chain); 7214 7215 // make sure enabled pre processing effects state is communicated to the HAL as we 7216 // just moved them to a new input stream. 7217 chain->syncHalEffectsState(); 7218 7219 mEffectChains.add(chain); 7220 7221 return NO_ERROR; 7222} 7223 7224size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 7225{ 7226 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 7227 ALOGW_IF(mEffectChains.size() != 1, 7228 "removeEffectChain_l() %p invalid chain size %d on thread %p", 7229 chain.get(), mEffectChains.size(), this); 7230 if (mEffectChains.size() == 1) { 7231 mEffectChains.removeAt(0); 7232 } 7233 return 0; 7234} 7235 7236status_t AudioFlinger::RecordThread::createAudioPatch_l(const struct audio_patch *patch, 7237 audio_patch_handle_t *handle) 7238{ 7239 status_t status = NO_ERROR; 7240 7241 // store new device and send to effects 7242 mInDevice = patch->sources[0].ext.device.type; 7243 mPatch = *patch; 7244 for (size_t i = 0; i < mEffectChains.size(); i++) { 7245 mEffectChains[i]->setDevice_l(mInDevice); 7246 } 7247 7248 // disable AEC and NS if the device is a BT SCO headset supporting those 7249 // pre processings 7250 if (mTracks.size() > 0) { 7251 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 7252 mAudioFlinger->btNrecIsOff(); 7253 for (size_t i = 0; i < mTracks.size(); i++) { 7254 sp<RecordTrack> track = mTracks[i]; 7255 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId()); 7256 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId()); 7257 } 7258 } 7259 7260 // store new source and send to effects 7261 if (mAudioSource != patch->sinks[0].ext.mix.usecase.source) { 7262 mAudioSource = patch->sinks[0].ext.mix.usecase.source; 7263 for (size_t i = 0; i < mEffectChains.size(); i++) { 7264 mEffectChains[i]->setAudioSource_l(mAudioSource); 7265 } 7266 } 7267 7268 if (mInput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 7269 audio_hw_device_t *hwDevice = mInput->audioHwDev->hwDevice(); 7270 status = hwDevice->create_audio_patch(hwDevice, 7271 patch->num_sources, 7272 patch->sources, 7273 patch->num_sinks, 7274 patch->sinks, 7275 handle); 7276 } else { 7277 char *address; 7278 if (strcmp(patch->sources[0].ext.device.address, "") != 0) { 7279 address = audio_device_address_to_parameter( 7280 patch->sources[0].ext.device.type, 7281 patch->sources[0].ext.device.address); 7282 } else { 7283 address = (char *)calloc(1, 1); 7284 } 7285 AudioParameter param = AudioParameter(String8(address)); 7286 free(address); 7287 param.addInt(String8(AUDIO_PARAMETER_STREAM_ROUTING), 7288 (int)patch->sources[0].ext.device.type); 7289 param.addInt(String8(AUDIO_PARAMETER_STREAM_INPUT_SOURCE), 7290 (int)patch->sinks[0].ext.mix.usecase.source); 7291 status = mInput->stream->common.set_parameters(&mInput->stream->common, 7292 param.toString().string()); 7293 *handle = AUDIO_PATCH_HANDLE_NONE; 7294 } 7295 7296 if (mInDevice != mPrevInDevice) { 7297 sendIoConfigEvent_l(AUDIO_INPUT_CONFIG_CHANGED); 7298 mPrevInDevice = mInDevice; 7299 } 7300 7301 return status; 7302} 7303 7304status_t AudioFlinger::RecordThread::releaseAudioPatch_l(const audio_patch_handle_t handle) 7305{ 7306 status_t status = NO_ERROR; 7307 7308 mInDevice = AUDIO_DEVICE_NONE; 7309 7310 if (mInput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 7311 audio_hw_device_t *hwDevice = mInput->audioHwDev->hwDevice(); 7312 status = hwDevice->release_audio_patch(hwDevice, handle); 7313 } else { 7314 AudioParameter param; 7315 param.addInt(String8(AUDIO_PARAMETER_STREAM_ROUTING), 0); 7316 status = mInput->stream->common.set_parameters(&mInput->stream->common, 7317 param.toString().string()); 7318 } 7319 return status; 7320} 7321 7322void AudioFlinger::RecordThread::addPatchRecord(const sp<PatchRecord>& record) 7323{ 7324 Mutex::Autolock _l(mLock); 7325 mTracks.add(record); 7326} 7327 7328void AudioFlinger::RecordThread::deletePatchRecord(const sp<PatchRecord>& record) 7329{ 7330 Mutex::Autolock _l(mLock); 7331 destroyTrack_l(record); 7332} 7333 7334void AudioFlinger::RecordThread::getAudioPortConfig(struct audio_port_config *config) 7335{ 7336 ThreadBase::getAudioPortConfig(config); 7337 config->role = AUDIO_PORT_ROLE_SINK; 7338 config->ext.mix.hw_module = mInput->audioHwDev->handle(); 7339 config->ext.mix.usecase.source = mAudioSource; 7340} 7341 7342} // namespace android 7343