AudioFlinger.cpp revision 17a58b2560c38a8e31a38186f9ab6eb98a38e229
1/* 2** 3** Copyright 2007, 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 22#include "Configuration.h" 23#include <dirent.h> 24#include <math.h> 25#include <signal.h> 26#include <sys/time.h> 27#include <sys/resource.h> 28 29#include <binder/IPCThreadState.h> 30#include <binder/IServiceManager.h> 31#include <utils/Log.h> 32#include <utils/Trace.h> 33#include <binder/Parcel.h> 34#include <memunreachable/memunreachable.h> 35#include <utils/String16.h> 36#include <utils/threads.h> 37#include <utils/Atomic.h> 38 39#include <cutils/bitops.h> 40#include <cutils/properties.h> 41 42#include <system/audio.h> 43#include <hardware/audio.h> 44 45#include "AudioMixer.h" 46#include "AudioFlinger.h" 47#include "ServiceUtilities.h" 48 49#include <media/AudioResamplerPublic.h> 50 51#include <media/EffectsFactoryApi.h> 52#include <audio_effects/effect_visualizer.h> 53#include <audio_effects/effect_ns.h> 54#include <audio_effects/effect_aec.h> 55 56#include <audio_utils/primitives.h> 57 58#include <powermanager/PowerManager.h> 59 60#include <media/IMediaLogService.h> 61#include <media/MemoryLeakTrackUtil.h> 62#include <media/nbaio/Pipe.h> 63#include <media/nbaio/PipeReader.h> 64#include <media/AudioParameter.h> 65#include <mediautils/BatteryNotifier.h> 66#include <private/android_filesystem_config.h> 67 68// ---------------------------------------------------------------------------- 69 70// Note: the following macro is used for extremely verbose logging message. In 71// order to run with ALOG_ASSERT turned on, we need to have LOG_NDEBUG set to 72// 0; but one side effect of this is to turn all LOGV's as well. Some messages 73// are so verbose that we want to suppress them even when we have ALOG_ASSERT 74// turned on. Do not uncomment the #def below unless you really know what you 75// are doing and want to see all of the extremely verbose messages. 76//#define VERY_VERY_VERBOSE_LOGGING 77#ifdef VERY_VERY_VERBOSE_LOGGING 78#define ALOGVV ALOGV 79#else 80#define ALOGVV(a...) do { } while(0) 81#endif 82 83namespace android { 84 85static const char kDeadlockedString[] = "AudioFlinger may be deadlocked\n"; 86static const char kHardwareLockedString[] = "Hardware lock is taken\n"; 87static const char kClientLockedString[] = "Client lock is taken\n"; 88 89 90nsecs_t AudioFlinger::mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs; 91 92uint32_t AudioFlinger::mScreenState; 93 94#ifdef TEE_SINK 95bool AudioFlinger::mTeeSinkInputEnabled = false; 96bool AudioFlinger::mTeeSinkOutputEnabled = false; 97bool AudioFlinger::mTeeSinkTrackEnabled = false; 98 99size_t AudioFlinger::mTeeSinkInputFrames = kTeeSinkInputFramesDefault; 100size_t AudioFlinger::mTeeSinkOutputFrames = kTeeSinkOutputFramesDefault; 101size_t AudioFlinger::mTeeSinkTrackFrames = kTeeSinkTrackFramesDefault; 102#endif 103 104// In order to avoid invalidating offloaded tracks each time a Visualizer is turned on and off 105// we define a minimum time during which a global effect is considered enabled. 106static const nsecs_t kMinGlobalEffectEnabletimeNs = seconds(7200); 107 108// ---------------------------------------------------------------------------- 109 110const char *formatToString(audio_format_t format) { 111 switch (audio_get_main_format(format)) { 112 case AUDIO_FORMAT_PCM: 113 switch (format) { 114 case AUDIO_FORMAT_PCM_16_BIT: return "pcm16"; 115 case AUDIO_FORMAT_PCM_8_BIT: return "pcm8"; 116 case AUDIO_FORMAT_PCM_32_BIT: return "pcm32"; 117 case AUDIO_FORMAT_PCM_8_24_BIT: return "pcm8.24"; 118 case AUDIO_FORMAT_PCM_FLOAT: return "pcmfloat"; 119 case AUDIO_FORMAT_PCM_24_BIT_PACKED: return "pcm24"; 120 default: 121 break; 122 } 123 break; 124 case AUDIO_FORMAT_MP3: return "mp3"; 125 case AUDIO_FORMAT_AMR_NB: return "amr-nb"; 126 case AUDIO_FORMAT_AMR_WB: return "amr-wb"; 127 case AUDIO_FORMAT_AAC: return "aac"; 128 case AUDIO_FORMAT_HE_AAC_V1: return "he-aac-v1"; 129 case AUDIO_FORMAT_HE_AAC_V2: return "he-aac-v2"; 130 case AUDIO_FORMAT_VORBIS: return "vorbis"; 131 case AUDIO_FORMAT_OPUS: return "opus"; 132 case AUDIO_FORMAT_AC3: return "ac-3"; 133 case AUDIO_FORMAT_E_AC3: return "e-ac-3"; 134 case AUDIO_FORMAT_IEC61937: return "iec61937"; 135 case AUDIO_FORMAT_DTS: return "dts"; 136 case AUDIO_FORMAT_DTS_HD: return "dts-hd"; 137 case AUDIO_FORMAT_DOLBY_TRUEHD: return "dolby-truehd"; 138 default: 139 break; 140 } 141 return "unknown"; 142} 143 144static int load_audio_interface(const char *if_name, audio_hw_device_t **dev) 145{ 146 const hw_module_t *mod; 147 int rc; 148 149 rc = hw_get_module_by_class(AUDIO_HARDWARE_MODULE_ID, if_name, &mod); 150 ALOGE_IF(rc, "%s couldn't load audio hw module %s.%s (%s)", __func__, 151 AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc)); 152 if (rc) { 153 goto out; 154 } 155 rc = audio_hw_device_open(mod, dev); 156 ALOGE_IF(rc, "%s couldn't open audio hw device in %s.%s (%s)", __func__, 157 AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc)); 158 if (rc) { 159 goto out; 160 } 161 if ((*dev)->common.version < AUDIO_DEVICE_API_VERSION_MIN) { 162 ALOGE("%s wrong audio hw device version %04x", __func__, (*dev)->common.version); 163 rc = BAD_VALUE; 164 goto out; 165 } 166 return 0; 167 168out: 169 *dev = NULL; 170 return rc; 171} 172 173// ---------------------------------------------------------------------------- 174 175AudioFlinger::AudioFlinger() 176 : BnAudioFlinger(), 177 mPrimaryHardwareDev(NULL), 178 mAudioHwDevs(NULL), 179 mHardwareStatus(AUDIO_HW_IDLE), 180 mMasterVolume(1.0f), 181 mMasterMute(false), 182 // mNextUniqueId(AUDIO_UNIQUE_ID_USE_MAX), 183 mMode(AUDIO_MODE_INVALID), 184 mBtNrecIsOff(false), 185 mIsLowRamDevice(true), 186 mIsDeviceTypeKnown(false), 187 mGlobalEffectEnableTime(0), 188 mSystemReady(false) 189{ 190 // unsigned instead of audio_unique_id_use_t, because ++ operator is unavailable for enum 191 for (unsigned use = AUDIO_UNIQUE_ID_USE_UNSPECIFIED; use < AUDIO_UNIQUE_ID_USE_MAX; use++) { 192 // zero ID has a special meaning, so unavailable 193 mNextUniqueIds[use] = AUDIO_UNIQUE_ID_USE_MAX; 194 } 195 196 getpid_cached = getpid(); 197 const bool doLog = property_get_bool("ro.test_harness", false); 198 if (doLog) { 199 mLogMemoryDealer = new MemoryDealer(kLogMemorySize, "LogWriters", 200 MemoryHeapBase::READ_ONLY); 201 } 202 203 // reset battery stats. 204 // if the audio service has crashed, battery stats could be left 205 // in bad state, reset the state upon service start. 206 BatteryNotifier::getInstance().noteResetAudio(); 207 208#ifdef TEE_SINK 209 char value[PROPERTY_VALUE_MAX]; 210 (void) property_get("ro.debuggable", value, "0"); 211 int debuggable = atoi(value); 212 int teeEnabled = 0; 213 if (debuggable) { 214 (void) property_get("af.tee", value, "0"); 215 teeEnabled = atoi(value); 216 } 217 // FIXME symbolic constants here 218 if (teeEnabled & 1) { 219 mTeeSinkInputEnabled = true; 220 } 221 if (teeEnabled & 2) { 222 mTeeSinkOutputEnabled = true; 223 } 224 if (teeEnabled & 4) { 225 mTeeSinkTrackEnabled = true; 226 } 227#endif 228} 229 230void AudioFlinger::onFirstRef() 231{ 232 Mutex::Autolock _l(mLock); 233 234 /* TODO: move all this work into an Init() function */ 235 char val_str[PROPERTY_VALUE_MAX] = { 0 }; 236 if (property_get("ro.audio.flinger_standbytime_ms", val_str, NULL) >= 0) { 237 uint32_t int_val; 238 if (1 == sscanf(val_str, "%u", &int_val)) { 239 mStandbyTimeInNsecs = milliseconds(int_val); 240 ALOGI("Using %u mSec as standby time.", int_val); 241 } else { 242 mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs; 243 ALOGI("Using default %u mSec as standby time.", 244 (uint32_t)(mStandbyTimeInNsecs / 1000000)); 245 } 246 } 247 248 mPatchPanel = new PatchPanel(this); 249 // FIXME: bug 30737845: trigger audioserver restart if main audioflinger lock 250 // is held continuously for more than 3 seconds 251 mLockWatch = new LockWatch(mLock, String8("AudioFlinger")); 252 mMode = AUDIO_MODE_NORMAL; 253} 254 255AudioFlinger::~AudioFlinger() 256{ 257 while (!mRecordThreads.isEmpty()) { 258 // closeInput_nonvirtual() will remove specified entry from mRecordThreads 259 closeInput_nonvirtual(mRecordThreads.keyAt(0)); 260 } 261 while (!mPlaybackThreads.isEmpty()) { 262 // closeOutput_nonvirtual() will remove specified entry from mPlaybackThreads 263 closeOutput_nonvirtual(mPlaybackThreads.keyAt(0)); 264 } 265 266 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 267 // no mHardwareLock needed, as there are no other references to this 268 audio_hw_device_close(mAudioHwDevs.valueAt(i)->hwDevice()); 269 delete mAudioHwDevs.valueAt(i); 270 } 271 272 // Tell media.log service about any old writers that still need to be unregistered 273 if (mLogMemoryDealer != 0) { 274 sp<IBinder> binder = defaultServiceManager()->getService(String16("media.log")); 275 if (binder != 0) { 276 sp<IMediaLogService> mediaLogService(interface_cast<IMediaLogService>(binder)); 277 for (size_t count = mUnregisteredWriters.size(); count > 0; count--) { 278 sp<IMemory> iMemory(mUnregisteredWriters.top()->getIMemory()); 279 mUnregisteredWriters.pop(); 280 mediaLogService->unregisterWriter(iMemory); 281 } 282 } 283 } 284 mLockWatch->requestExitAndWait(); 285} 286 287static const char * const audio_interfaces[] = { 288 AUDIO_HARDWARE_MODULE_ID_PRIMARY, 289 AUDIO_HARDWARE_MODULE_ID_A2DP, 290 AUDIO_HARDWARE_MODULE_ID_USB, 291}; 292#define ARRAY_SIZE(x) (sizeof((x))/sizeof(((x)[0]))) 293 294AudioHwDevice* AudioFlinger::findSuitableHwDev_l( 295 audio_module_handle_t module, 296 audio_devices_t devices) 297{ 298 // if module is 0, the request comes from an old policy manager and we should load 299 // well known modules 300 if (module == 0) { 301 ALOGW("findSuitableHwDev_l() loading well know audio hw modules"); 302 for (size_t i = 0; i < ARRAY_SIZE(audio_interfaces); i++) { 303 loadHwModule_l(audio_interfaces[i]); 304 } 305 // then try to find a module supporting the requested device. 306 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 307 AudioHwDevice *audioHwDevice = mAudioHwDevs.valueAt(i); 308 audio_hw_device_t *dev = audioHwDevice->hwDevice(); 309 if ((dev->get_supported_devices != NULL) && 310 (dev->get_supported_devices(dev) & devices) == devices) 311 return audioHwDevice; 312 } 313 } else { 314 // check a match for the requested module handle 315 AudioHwDevice *audioHwDevice = mAudioHwDevs.valueFor(module); 316 if (audioHwDevice != NULL) { 317 return audioHwDevice; 318 } 319 } 320 321 return NULL; 322} 323 324void AudioFlinger::dumpClients(int fd, const Vector<String16>& args __unused) 325{ 326 const size_t SIZE = 256; 327 char buffer[SIZE]; 328 String8 result; 329 330 result.append("Clients:\n"); 331 for (size_t i = 0; i < mClients.size(); ++i) { 332 sp<Client> client = mClients.valueAt(i).promote(); 333 if (client != 0) { 334 snprintf(buffer, SIZE, " pid: %d\n", client->pid()); 335 result.append(buffer); 336 } 337 } 338 339 result.append("Notification Clients:\n"); 340 for (size_t i = 0; i < mNotificationClients.size(); ++i) { 341 snprintf(buffer, SIZE, " pid: %d\n", mNotificationClients.keyAt(i)); 342 result.append(buffer); 343 } 344 345 result.append("Global session refs:\n"); 346 result.append(" session pid count\n"); 347 for (size_t i = 0; i < mAudioSessionRefs.size(); i++) { 348 AudioSessionRef *r = mAudioSessionRefs[i]; 349 snprintf(buffer, SIZE, " %7d %5d %5d\n", r->mSessionid, r->mPid, r->mCnt); 350 result.append(buffer); 351 } 352 write(fd, result.string(), result.size()); 353} 354 355 356void AudioFlinger::dumpInternals(int fd, const Vector<String16>& args __unused) 357{ 358 const size_t SIZE = 256; 359 char buffer[SIZE]; 360 String8 result; 361 hardware_call_state hardwareStatus = mHardwareStatus; 362 363 snprintf(buffer, SIZE, "Hardware status: %d\n" 364 "Standby Time mSec: %u\n", 365 hardwareStatus, 366 (uint32_t)(mStandbyTimeInNsecs / 1000000)); 367 result.append(buffer); 368 write(fd, result.string(), result.size()); 369} 370 371void AudioFlinger::dumpPermissionDenial(int fd, const Vector<String16>& args __unused) 372{ 373 const size_t SIZE = 256; 374 char buffer[SIZE]; 375 String8 result; 376 snprintf(buffer, SIZE, "Permission Denial: " 377 "can't dump AudioFlinger from pid=%d, uid=%d\n", 378 IPCThreadState::self()->getCallingPid(), 379 IPCThreadState::self()->getCallingUid()); 380 result.append(buffer); 381 write(fd, result.string(), result.size()); 382} 383 384bool AudioFlinger::dumpTryLock(Mutex& mutex) 385{ 386 bool locked = false; 387 for (int i = 0; i < kDumpLockRetries; ++i) { 388 if (mutex.tryLock() == NO_ERROR) { 389 locked = true; 390 break; 391 } 392 usleep(kDumpLockSleepUs); 393 } 394 return locked; 395} 396 397status_t AudioFlinger::dump(int fd, const Vector<String16>& args) 398{ 399 if (!dumpAllowed()) { 400 dumpPermissionDenial(fd, args); 401 } else { 402 // get state of hardware lock 403 bool hardwareLocked = dumpTryLock(mHardwareLock); 404 if (!hardwareLocked) { 405 String8 result(kHardwareLockedString); 406 write(fd, result.string(), result.size()); 407 } else { 408 mHardwareLock.unlock(); 409 } 410 411 bool locked = dumpTryLock(mLock); 412 413 // failed to lock - AudioFlinger is probably deadlocked 414 if (!locked) { 415 String8 result(kDeadlockedString); 416 write(fd, result.string(), result.size()); 417 } 418 419 bool clientLocked = dumpTryLock(mClientLock); 420 if (!clientLocked) { 421 String8 result(kClientLockedString); 422 write(fd, result.string(), result.size()); 423 } 424 425 EffectDumpEffects(fd); 426 427 dumpClients(fd, args); 428 if (clientLocked) { 429 mClientLock.unlock(); 430 } 431 432 dumpInternals(fd, args); 433 434 // dump playback threads 435 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 436 mPlaybackThreads.valueAt(i)->dump(fd, args); 437 } 438 439 // dump record threads 440 for (size_t i = 0; i < mRecordThreads.size(); i++) { 441 mRecordThreads.valueAt(i)->dump(fd, args); 442 } 443 444 // dump orphan effect chains 445 if (mOrphanEffectChains.size() != 0) { 446 write(fd, " Orphan Effect Chains\n", strlen(" Orphan Effect Chains\n")); 447 for (size_t i = 0; i < mOrphanEffectChains.size(); i++) { 448 mOrphanEffectChains.valueAt(i)->dump(fd, args); 449 } 450 } 451 // dump all hardware devs 452 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 453 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 454 dev->dump(dev, fd); 455 } 456 457#ifdef TEE_SINK 458 // dump the serially shared record tee sink 459 if (mRecordTeeSource != 0) { 460 dumpTee(fd, mRecordTeeSource); 461 } 462#endif 463 464 if (locked) { 465 mLock.unlock(); 466 } 467 468 // append a copy of media.log here by forwarding fd to it, but don't attempt 469 // to lookup the service if it's not running, as it will block for a second 470 if (mLogMemoryDealer != 0) { 471 sp<IBinder> binder = defaultServiceManager()->getService(String16("media.log")); 472 if (binder != 0) { 473 dprintf(fd, "\nmedia.log:\n"); 474 Vector<String16> args; 475 binder->dump(fd, args); 476 } 477 } 478 479 // check for optional arguments 480 bool dumpMem = false; 481 bool unreachableMemory = false; 482 for (const auto &arg : args) { 483 if (arg == String16("-m")) { 484 dumpMem = true; 485 } else if (arg == String16("--unreachable")) { 486 unreachableMemory = true; 487 } 488 } 489 490 if (dumpMem) { 491 dprintf(fd, "\nDumping memory:\n"); 492 std::string s = dumpMemoryAddresses(100 /* limit */); 493 write(fd, s.c_str(), s.size()); 494 } 495 if (unreachableMemory) { 496 dprintf(fd, "\nDumping unreachable memory:\n"); 497 // TODO - should limit be an argument parameter? 498 std::string s = GetUnreachableMemoryString(true /* contents */, 100 /* limit */); 499 write(fd, s.c_str(), s.size()); 500 } 501 } 502 return NO_ERROR; 503} 504 505sp<AudioFlinger::Client> AudioFlinger::registerPid(pid_t pid) 506{ 507 Mutex::Autolock _cl(mClientLock); 508 // If pid is already in the mClients wp<> map, then use that entry 509 // (for which promote() is always != 0), otherwise create a new entry and Client. 510 sp<Client> client = mClients.valueFor(pid).promote(); 511 if (client == 0) { 512 client = new Client(this, pid); 513 mClients.add(pid, client); 514 } 515 516 return client; 517} 518 519sp<NBLog::Writer> AudioFlinger::newWriter_l(size_t size, const char *name) 520{ 521 // If there is no memory allocated for logs, return a dummy writer that does nothing 522 if (mLogMemoryDealer == 0) { 523 return new NBLog::Writer(); 524 } 525 sp<IBinder> binder = defaultServiceManager()->getService(String16("media.log")); 526 // Similarly if we can't contact the media.log service, also return a dummy writer 527 if (binder == 0) { 528 return new NBLog::Writer(); 529 } 530 sp<IMediaLogService> mediaLogService(interface_cast<IMediaLogService>(binder)); 531 sp<IMemory> shared = mLogMemoryDealer->allocate(NBLog::Timeline::sharedSize(size)); 532 // If allocation fails, consult the vector of previously unregistered writers 533 // and garbage-collect one or more them until an allocation succeeds 534 if (shared == 0) { 535 Mutex::Autolock _l(mUnregisteredWritersLock); 536 for (size_t count = mUnregisteredWriters.size(); count > 0; count--) { 537 { 538 // Pick the oldest stale writer to garbage-collect 539 sp<IMemory> iMemory(mUnregisteredWriters[0]->getIMemory()); 540 mUnregisteredWriters.removeAt(0); 541 mediaLogService->unregisterWriter(iMemory); 542 // Now the media.log remote reference to IMemory is gone. When our last local 543 // reference to IMemory also drops to zero at end of this block, 544 // the IMemory destructor will deallocate the region from mLogMemoryDealer. 545 } 546 // Re-attempt the allocation 547 shared = mLogMemoryDealer->allocate(NBLog::Timeline::sharedSize(size)); 548 if (shared != 0) { 549 goto success; 550 } 551 } 552 // Even after garbage-collecting all old writers, there is still not enough memory, 553 // so return a dummy writer 554 return new NBLog::Writer(); 555 } 556success: 557 mediaLogService->registerWriter(shared, size, name); 558 return new NBLog::Writer(size, shared); 559} 560 561void AudioFlinger::unregisterWriter(const sp<NBLog::Writer>& writer) 562{ 563 if (writer == 0) { 564 return; 565 } 566 sp<IMemory> iMemory(writer->getIMemory()); 567 if (iMemory == 0) { 568 return; 569 } 570 // Rather than removing the writer immediately, append it to a queue of old writers to 571 // be garbage-collected later. This allows us to continue to view old logs for a while. 572 Mutex::Autolock _l(mUnregisteredWritersLock); 573 mUnregisteredWriters.push(writer); 574} 575 576// IAudioFlinger interface 577 578 579sp<IAudioTrack> AudioFlinger::createTrack( 580 audio_stream_type_t streamType, 581 uint32_t sampleRate, 582 audio_format_t format, 583 audio_channel_mask_t channelMask, 584 size_t *frameCount, 585 audio_output_flags_t *flags, 586 const sp<IMemory>& sharedBuffer, 587 audio_io_handle_t output, 588 pid_t pid, 589 pid_t tid, 590 audio_session_t *sessionId, 591 int clientUid, 592 status_t *status) 593{ 594 sp<PlaybackThread::Track> track; 595 sp<TrackHandle> trackHandle; 596 sp<Client> client; 597 status_t lStatus; 598 audio_session_t lSessionId; 599 600 const uid_t callingUid = IPCThreadState::self()->getCallingUid(); 601 if (pid == -1 || !isTrustedCallingUid(callingUid)) { 602 const pid_t callingPid = IPCThreadState::self()->getCallingPid(); 603 ALOGW_IF(pid != -1 && pid != callingPid, 604 "%s uid %d pid %d tried to pass itself off as pid %d", 605 __func__, callingUid, callingPid, pid); 606 pid = callingPid; 607 } 608 609 // client AudioTrack::set already implements AUDIO_STREAM_DEFAULT => AUDIO_STREAM_MUSIC, 610 // but if someone uses binder directly they could bypass that and cause us to crash 611 if (uint32_t(streamType) >= AUDIO_STREAM_CNT) { 612 ALOGE("createTrack() invalid stream type %d", streamType); 613 lStatus = BAD_VALUE; 614 goto Exit; 615 } 616 617 // further sample rate checks are performed by createTrack_l() depending on the thread type 618 if (sampleRate == 0) { 619 ALOGE("createTrack() invalid sample rate %u", sampleRate); 620 lStatus = BAD_VALUE; 621 goto Exit; 622 } 623 624 // further channel mask checks are performed by createTrack_l() depending on the thread type 625 if (!audio_is_output_channel(channelMask)) { 626 ALOGE("createTrack() invalid channel mask %#x", channelMask); 627 lStatus = BAD_VALUE; 628 goto Exit; 629 } 630 631 // further format checks are performed by createTrack_l() depending on the thread type 632 if (!audio_is_valid_format(format)) { 633 ALOGE("createTrack() invalid format %#x", format); 634 lStatus = BAD_VALUE; 635 goto Exit; 636 } 637 638 if (sharedBuffer != 0 && sharedBuffer->pointer() == NULL) { 639 ALOGE("createTrack() sharedBuffer is non-0 but has NULL pointer()"); 640 lStatus = BAD_VALUE; 641 goto Exit; 642 } 643 644 { 645 Mutex::Autolock _l(mLock); 646 PlaybackThread *thread = checkPlaybackThread_l(output); 647 if (thread == NULL) { 648 ALOGE("no playback thread found for output handle %d", output); 649 lStatus = BAD_VALUE; 650 goto Exit; 651 } 652 653 client = registerPid(pid); 654 655 PlaybackThread *effectThread = NULL; 656 if (sessionId != NULL && *sessionId != AUDIO_SESSION_ALLOCATE) { 657 if (audio_unique_id_get_use(*sessionId) != AUDIO_UNIQUE_ID_USE_SESSION) { 658 ALOGE("createTrack() invalid session ID %d", *sessionId); 659 lStatus = BAD_VALUE; 660 goto Exit; 661 } 662 lSessionId = *sessionId; 663 // check if an effect chain with the same session ID is present on another 664 // output thread and move it here. 665 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 666 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 667 if (mPlaybackThreads.keyAt(i) != output) { 668 uint32_t sessions = t->hasAudioSession(lSessionId); 669 if (sessions & ThreadBase::EFFECT_SESSION) { 670 effectThread = t.get(); 671 break; 672 } 673 } 674 } 675 } else { 676 // if no audio session id is provided, create one here 677 lSessionId = (audio_session_t) nextUniqueId(AUDIO_UNIQUE_ID_USE_SESSION); 678 if (sessionId != NULL) { 679 *sessionId = lSessionId; 680 } 681 } 682 ALOGV("createTrack() lSessionId: %d", lSessionId); 683 684 track = thread->createTrack_l(client, streamType, sampleRate, format, 685 channelMask, frameCount, sharedBuffer, lSessionId, flags, tid, clientUid, &lStatus); 686 LOG_ALWAYS_FATAL_IF((lStatus == NO_ERROR) && (track == 0)); 687 // we don't abort yet if lStatus != NO_ERROR; there is still work to be done regardless 688 689 // move effect chain to this output thread if an effect on same session was waiting 690 // for a track to be created 691 if (lStatus == NO_ERROR && effectThread != NULL) { 692 // no risk of deadlock because AudioFlinger::mLock is held 693 Mutex::Autolock _dl(thread->mLock); 694 Mutex::Autolock _sl(effectThread->mLock); 695 moveEffectChain_l(lSessionId, effectThread, thread, true); 696 } 697 698 // Look for sync events awaiting for a session to be used. 699 for (size_t i = 0; i < mPendingSyncEvents.size(); i++) { 700 if (mPendingSyncEvents[i]->triggerSession() == lSessionId) { 701 if (thread->isValidSyncEvent(mPendingSyncEvents[i])) { 702 if (lStatus == NO_ERROR) { 703 (void) track->setSyncEvent(mPendingSyncEvents[i]); 704 } else { 705 mPendingSyncEvents[i]->cancel(); 706 } 707 mPendingSyncEvents.removeAt(i); 708 i--; 709 } 710 } 711 } 712 713 setAudioHwSyncForSession_l(thread, lSessionId); 714 } 715 716 if (lStatus != NO_ERROR) { 717 // remove local strong reference to Client before deleting the Track so that the 718 // Client destructor is called by the TrackBase destructor with mClientLock held 719 // Don't hold mClientLock when releasing the reference on the track as the 720 // destructor will acquire it. 721 { 722 Mutex::Autolock _cl(mClientLock); 723 client.clear(); 724 } 725 track.clear(); 726 goto Exit; 727 } 728 729 // return handle to client 730 trackHandle = new TrackHandle(track); 731 732Exit: 733 *status = lStatus; 734 return trackHandle; 735} 736 737uint32_t AudioFlinger::sampleRate(audio_io_handle_t ioHandle) const 738{ 739 Mutex::Autolock _l(mLock); 740 ThreadBase *thread = checkThread_l(ioHandle); 741 if (thread == NULL) { 742 ALOGW("sampleRate() unknown thread %d", ioHandle); 743 return 0; 744 } 745 return thread->sampleRate(); 746} 747 748audio_format_t AudioFlinger::format(audio_io_handle_t output) const 749{ 750 Mutex::Autolock _l(mLock); 751 PlaybackThread *thread = checkPlaybackThread_l(output); 752 if (thread == NULL) { 753 ALOGW("format() unknown thread %d", output); 754 return AUDIO_FORMAT_INVALID; 755 } 756 return thread->format(); 757} 758 759size_t AudioFlinger::frameCount(audio_io_handle_t ioHandle) const 760{ 761 Mutex::Autolock _l(mLock); 762 ThreadBase *thread = checkThread_l(ioHandle); 763 if (thread == NULL) { 764 ALOGW("frameCount() unknown thread %d", ioHandle); 765 return 0; 766 } 767 // FIXME currently returns the normal mixer's frame count to avoid confusing legacy callers; 768 // should examine all callers and fix them to handle smaller counts 769 return thread->frameCount(); 770} 771 772size_t AudioFlinger::frameCountHAL(audio_io_handle_t ioHandle) const 773{ 774 Mutex::Autolock _l(mLock); 775 ThreadBase *thread = checkThread_l(ioHandle); 776 if (thread == NULL) { 777 ALOGW("frameCountHAL() unknown thread %d", ioHandle); 778 return 0; 779 } 780 return thread->frameCountHAL(); 781} 782 783uint32_t AudioFlinger::latency(audio_io_handle_t output) const 784{ 785 Mutex::Autolock _l(mLock); 786 PlaybackThread *thread = checkPlaybackThread_l(output); 787 if (thread == NULL) { 788 ALOGW("latency(): no playback thread found for output handle %d", output); 789 return 0; 790 } 791 return thread->latency(); 792} 793 794status_t AudioFlinger::setMasterVolume(float value) 795{ 796 status_t ret = initCheck(); 797 if (ret != NO_ERROR) { 798 return ret; 799 } 800 801 // check calling permissions 802 if (!settingsAllowed()) { 803 return PERMISSION_DENIED; 804 } 805 806 Mutex::Autolock _l(mLock); 807 mMasterVolume = value; 808 809 // Set master volume in the HALs which support it. 810 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 811 AutoMutex lock(mHardwareLock); 812 AudioHwDevice *dev = mAudioHwDevs.valueAt(i); 813 814 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 815 if (dev->canSetMasterVolume()) { 816 dev->hwDevice()->set_master_volume(dev->hwDevice(), value); 817 } 818 mHardwareStatus = AUDIO_HW_IDLE; 819 } 820 821 // Now set the master volume in each playback thread. Playback threads 822 // assigned to HALs which do not have master volume support will apply 823 // master volume during the mix operation. Threads with HALs which do 824 // support master volume will simply ignore the setting. 825 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 826 if (mPlaybackThreads.valueAt(i)->isDuplicating()) { 827 continue; 828 } 829 mPlaybackThreads.valueAt(i)->setMasterVolume(value); 830 } 831 832 return NO_ERROR; 833} 834 835status_t AudioFlinger::setMode(audio_mode_t mode) 836{ 837 status_t ret = initCheck(); 838 if (ret != NO_ERROR) { 839 return ret; 840 } 841 842 // check calling permissions 843 if (!settingsAllowed()) { 844 return PERMISSION_DENIED; 845 } 846 if (uint32_t(mode) >= AUDIO_MODE_CNT) { 847 ALOGW("Illegal value: setMode(%d)", mode); 848 return BAD_VALUE; 849 } 850 851 { // scope for the lock 852 AutoMutex lock(mHardwareLock); 853 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 854 mHardwareStatus = AUDIO_HW_SET_MODE; 855 ret = dev->set_mode(dev, mode); 856 mHardwareStatus = AUDIO_HW_IDLE; 857 } 858 859 if (NO_ERROR == ret) { 860 Mutex::Autolock _l(mLock); 861 mMode = mode; 862 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 863 mPlaybackThreads.valueAt(i)->setMode(mode); 864 } 865 866 return ret; 867} 868 869status_t AudioFlinger::setMicMute(bool state) 870{ 871 status_t ret = initCheck(); 872 if (ret != NO_ERROR) { 873 return ret; 874 } 875 876 // check calling permissions 877 if (!settingsAllowed()) { 878 return PERMISSION_DENIED; 879 } 880 881 AutoMutex lock(mHardwareLock); 882 mHardwareStatus = AUDIO_HW_SET_MIC_MUTE; 883 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 884 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 885 status_t result = dev->set_mic_mute(dev, state); 886 if (result != NO_ERROR) { 887 ret = result; 888 } 889 } 890 mHardwareStatus = AUDIO_HW_IDLE; 891 return ret; 892} 893 894bool AudioFlinger::getMicMute() const 895{ 896 status_t ret = initCheck(); 897 if (ret != NO_ERROR) { 898 return false; 899 } 900 bool mute = true; 901 bool state = AUDIO_MODE_INVALID; 902 AutoMutex lock(mHardwareLock); 903 mHardwareStatus = AUDIO_HW_GET_MIC_MUTE; 904 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 905 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 906 status_t result = dev->get_mic_mute(dev, &state); 907 if (result == NO_ERROR) { 908 mute = mute && state; 909 } 910 } 911 mHardwareStatus = AUDIO_HW_IDLE; 912 913 return mute; 914} 915 916status_t AudioFlinger::setMasterMute(bool muted) 917{ 918 status_t ret = initCheck(); 919 if (ret != NO_ERROR) { 920 return ret; 921 } 922 923 // check calling permissions 924 if (!settingsAllowed()) { 925 return PERMISSION_DENIED; 926 } 927 928 Mutex::Autolock _l(mLock); 929 mMasterMute = muted; 930 931 // Set master mute in the HALs which support it. 932 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 933 AutoMutex lock(mHardwareLock); 934 AudioHwDevice *dev = mAudioHwDevs.valueAt(i); 935 936 mHardwareStatus = AUDIO_HW_SET_MASTER_MUTE; 937 if (dev->canSetMasterMute()) { 938 dev->hwDevice()->set_master_mute(dev->hwDevice(), muted); 939 } 940 mHardwareStatus = AUDIO_HW_IDLE; 941 } 942 943 // Now set the master mute in each playback thread. Playback threads 944 // assigned to HALs which do not have master mute support will apply master 945 // mute during the mix operation. Threads with HALs which do support master 946 // mute will simply ignore the setting. 947 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 948 if (mPlaybackThreads.valueAt(i)->isDuplicating()) { 949 continue; 950 } 951 mPlaybackThreads.valueAt(i)->setMasterMute(muted); 952 } 953 954 return NO_ERROR; 955} 956 957float AudioFlinger::masterVolume() const 958{ 959 Mutex::Autolock _l(mLock); 960 return masterVolume_l(); 961} 962 963bool AudioFlinger::masterMute() const 964{ 965 Mutex::Autolock _l(mLock); 966 return masterMute_l(); 967} 968 969float AudioFlinger::masterVolume_l() const 970{ 971 return mMasterVolume; 972} 973 974bool AudioFlinger::masterMute_l() const 975{ 976 return mMasterMute; 977} 978 979status_t AudioFlinger::checkStreamType(audio_stream_type_t stream) const 980{ 981 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 982 ALOGW("setStreamVolume() invalid stream %d", stream); 983 return BAD_VALUE; 984 } 985 pid_t caller = IPCThreadState::self()->getCallingPid(); 986 if (uint32_t(stream) >= AUDIO_STREAM_PUBLIC_CNT && caller != getpid_cached) { 987 ALOGW("setStreamVolume() pid %d cannot use internal stream type %d", caller, stream); 988 return PERMISSION_DENIED; 989 } 990 991 return NO_ERROR; 992} 993 994status_t AudioFlinger::setStreamVolume(audio_stream_type_t stream, float value, 995 audio_io_handle_t output) 996{ 997 // check calling permissions 998 if (!settingsAllowed()) { 999 return PERMISSION_DENIED; 1000 } 1001 1002 status_t status = checkStreamType(stream); 1003 if (status != NO_ERROR) { 1004 return status; 1005 } 1006 ALOG_ASSERT(stream != AUDIO_STREAM_PATCH, "attempt to change AUDIO_STREAM_PATCH volume"); 1007 1008 AutoMutex lock(mLock); 1009 PlaybackThread *thread = NULL; 1010 if (output != AUDIO_IO_HANDLE_NONE) { 1011 thread = checkPlaybackThread_l(output); 1012 if (thread == NULL) { 1013 return BAD_VALUE; 1014 } 1015 } 1016 1017 mStreamTypes[stream].volume = value; 1018 1019 if (thread == NULL) { 1020 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1021 mPlaybackThreads.valueAt(i)->setStreamVolume(stream, value); 1022 } 1023 } else { 1024 thread->setStreamVolume(stream, value); 1025 } 1026 1027 return NO_ERROR; 1028} 1029 1030status_t AudioFlinger::setStreamMute(audio_stream_type_t stream, bool muted) 1031{ 1032 // check calling permissions 1033 if (!settingsAllowed()) { 1034 return PERMISSION_DENIED; 1035 } 1036 1037 status_t status = checkStreamType(stream); 1038 if (status != NO_ERROR) { 1039 return status; 1040 } 1041 ALOG_ASSERT(stream != AUDIO_STREAM_PATCH, "attempt to mute AUDIO_STREAM_PATCH"); 1042 1043 if (uint32_t(stream) == AUDIO_STREAM_ENFORCED_AUDIBLE) { 1044 ALOGE("setStreamMute() invalid stream %d", stream); 1045 return BAD_VALUE; 1046 } 1047 1048 AutoMutex lock(mLock); 1049 mStreamTypes[stream].mute = muted; 1050 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 1051 mPlaybackThreads.valueAt(i)->setStreamMute(stream, muted); 1052 1053 return NO_ERROR; 1054} 1055 1056float AudioFlinger::streamVolume(audio_stream_type_t stream, audio_io_handle_t output) const 1057{ 1058 status_t status = checkStreamType(stream); 1059 if (status != NO_ERROR) { 1060 return 0.0f; 1061 } 1062 1063 AutoMutex lock(mLock); 1064 float volume; 1065 if (output != AUDIO_IO_HANDLE_NONE) { 1066 PlaybackThread *thread = checkPlaybackThread_l(output); 1067 if (thread == NULL) { 1068 return 0.0f; 1069 } 1070 volume = thread->streamVolume(stream); 1071 } else { 1072 volume = streamVolume_l(stream); 1073 } 1074 1075 return volume; 1076} 1077 1078bool AudioFlinger::streamMute(audio_stream_type_t stream) const 1079{ 1080 status_t status = checkStreamType(stream); 1081 if (status != NO_ERROR) { 1082 return true; 1083 } 1084 1085 AutoMutex lock(mLock); 1086 return streamMute_l(stream); 1087} 1088 1089 1090void AudioFlinger::broacastParametersToRecordThreads_l(const String8& keyValuePairs) 1091{ 1092 for (size_t i = 0; i < mRecordThreads.size(); i++) { 1093 mRecordThreads.valueAt(i)->setParameters(keyValuePairs); 1094 } 1095} 1096 1097status_t AudioFlinger::setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs) 1098{ 1099 ALOGV("setParameters(): io %d, keyvalue %s, calling pid %d", 1100 ioHandle, keyValuePairs.string(), IPCThreadState::self()->getCallingPid()); 1101 1102 // check calling permissions 1103 if (!settingsAllowed()) { 1104 return PERMISSION_DENIED; 1105 } 1106 1107 // AUDIO_IO_HANDLE_NONE means the parameters are global to the audio hardware interface 1108 if (ioHandle == AUDIO_IO_HANDLE_NONE) { 1109 Mutex::Autolock _l(mLock); 1110 status_t final_result = NO_ERROR; 1111 { 1112 AutoMutex lock(mHardwareLock); 1113 mHardwareStatus = AUDIO_HW_SET_PARAMETER; 1114 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 1115 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 1116 status_t result = dev->set_parameters(dev, keyValuePairs.string()); 1117 final_result = result ?: final_result; 1118 } 1119 mHardwareStatus = AUDIO_HW_IDLE; 1120 } 1121 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 1122 AudioParameter param = AudioParameter(keyValuePairs); 1123 String8 value; 1124 if (param.get(String8(AUDIO_PARAMETER_KEY_BT_NREC), value) == NO_ERROR) { 1125 bool btNrecIsOff = (value == AUDIO_PARAMETER_VALUE_OFF); 1126 if (mBtNrecIsOff != btNrecIsOff) { 1127 for (size_t i = 0; i < mRecordThreads.size(); i++) { 1128 sp<RecordThread> thread = mRecordThreads.valueAt(i); 1129 audio_devices_t device = thread->inDevice(); 1130 bool suspend = audio_is_bluetooth_sco_device(device) && btNrecIsOff; 1131 // collect all of the thread's session IDs 1132 KeyedVector<audio_session_t, bool> ids = thread->sessionIds(); 1133 // suspend effects associated with those session IDs 1134 for (size_t j = 0; j < ids.size(); ++j) { 1135 audio_session_t sessionId = ids.keyAt(j); 1136 thread->setEffectSuspended(FX_IID_AEC, 1137 suspend, 1138 sessionId); 1139 thread->setEffectSuspended(FX_IID_NS, 1140 suspend, 1141 sessionId); 1142 } 1143 } 1144 mBtNrecIsOff = btNrecIsOff; 1145 } 1146 } 1147 String8 screenState; 1148 if (param.get(String8(AudioParameter::keyScreenState), screenState) == NO_ERROR) { 1149 bool isOff = screenState == "off"; 1150 if (isOff != (AudioFlinger::mScreenState & 1)) { 1151 AudioFlinger::mScreenState = ((AudioFlinger::mScreenState & ~1) + 2) | isOff; 1152 } 1153 } 1154 return final_result; 1155 } 1156 1157 // hold a strong ref on thread in case closeOutput() or closeInput() is called 1158 // and the thread is exited once the lock is released 1159 sp<ThreadBase> thread; 1160 { 1161 Mutex::Autolock _l(mLock); 1162 thread = checkPlaybackThread_l(ioHandle); 1163 if (thread == 0) { 1164 thread = checkRecordThread_l(ioHandle); 1165 } else if (thread == primaryPlaybackThread_l()) { 1166 // indicate output device change to all input threads for pre processing 1167 AudioParameter param = AudioParameter(keyValuePairs); 1168 int value; 1169 if ((param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) && 1170 (value != 0)) { 1171 broacastParametersToRecordThreads_l(keyValuePairs); 1172 } 1173 } 1174 } 1175 if (thread != 0) { 1176 return thread->setParameters(keyValuePairs); 1177 } 1178 return BAD_VALUE; 1179} 1180 1181String8 AudioFlinger::getParameters(audio_io_handle_t ioHandle, const String8& keys) const 1182{ 1183 ALOGVV("getParameters() io %d, keys %s, calling pid %d", 1184 ioHandle, keys.string(), IPCThreadState::self()->getCallingPid()); 1185 1186 Mutex::Autolock _l(mLock); 1187 1188 if (ioHandle == AUDIO_IO_HANDLE_NONE) { 1189 String8 out_s8; 1190 1191 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 1192 char *s; 1193 { 1194 AutoMutex lock(mHardwareLock); 1195 mHardwareStatus = AUDIO_HW_GET_PARAMETER; 1196 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 1197 s = dev->get_parameters(dev, keys.string()); 1198 mHardwareStatus = AUDIO_HW_IDLE; 1199 } 1200 out_s8 += String8(s ? s : ""); 1201 free(s); 1202 } 1203 return out_s8; 1204 } 1205 1206 PlaybackThread *playbackThread = checkPlaybackThread_l(ioHandle); 1207 if (playbackThread != NULL) { 1208 return playbackThread->getParameters(keys); 1209 } 1210 RecordThread *recordThread = checkRecordThread_l(ioHandle); 1211 if (recordThread != NULL) { 1212 return recordThread->getParameters(keys); 1213 } 1214 return String8(""); 1215} 1216 1217size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, audio_format_t format, 1218 audio_channel_mask_t channelMask) const 1219{ 1220 status_t ret = initCheck(); 1221 if (ret != NO_ERROR) { 1222 return 0; 1223 } 1224 if ((sampleRate == 0) || 1225 !audio_is_valid_format(format) || !audio_has_proportional_frames(format) || 1226 !audio_is_input_channel(channelMask)) { 1227 return 0; 1228 } 1229 1230 AutoMutex lock(mHardwareLock); 1231 mHardwareStatus = AUDIO_HW_GET_INPUT_BUFFER_SIZE; 1232 audio_config_t config, proposed; 1233 memset(&proposed, 0, sizeof(proposed)); 1234 proposed.sample_rate = sampleRate; 1235 proposed.channel_mask = channelMask; 1236 proposed.format = format; 1237 1238 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 1239 size_t frames; 1240 for (;;) { 1241 // Note: config is currently a const parameter for get_input_buffer_size() 1242 // but we use a copy from proposed in case config changes from the call. 1243 config = proposed; 1244 frames = dev->get_input_buffer_size(dev, &config); 1245 if (frames != 0) { 1246 break; // hal success, config is the result 1247 } 1248 // change one parameter of the configuration each iteration to a more "common" value 1249 // to see if the device will support it. 1250 if (proposed.format != AUDIO_FORMAT_PCM_16_BIT) { 1251 proposed.format = AUDIO_FORMAT_PCM_16_BIT; 1252 } else if (proposed.sample_rate != 44100) { // 44.1 is claimed as must in CDD as well as 1253 proposed.sample_rate = 44100; // legacy AudioRecord.java. TODO: Query hw? 1254 } else { 1255 ALOGW("getInputBufferSize failed with minimum buffer size sampleRate %u, " 1256 "format %#x, channelMask 0x%X", 1257 sampleRate, format, channelMask); 1258 break; // retries failed, break out of loop with frames == 0. 1259 } 1260 } 1261 mHardwareStatus = AUDIO_HW_IDLE; 1262 if (frames > 0 && config.sample_rate != sampleRate) { 1263 frames = destinationFramesPossible(frames, sampleRate, config.sample_rate); 1264 } 1265 return frames; // may be converted to bytes at the Java level. 1266} 1267 1268uint32_t AudioFlinger::getInputFramesLost(audio_io_handle_t ioHandle) const 1269{ 1270 Mutex::Autolock _l(mLock); 1271 1272 RecordThread *recordThread = checkRecordThread_l(ioHandle); 1273 if (recordThread != NULL) { 1274 return recordThread->getInputFramesLost(); 1275 } 1276 return 0; 1277} 1278 1279status_t AudioFlinger::setVoiceVolume(float value) 1280{ 1281 status_t ret = initCheck(); 1282 if (ret != NO_ERROR) { 1283 return ret; 1284 } 1285 1286 // check calling permissions 1287 if (!settingsAllowed()) { 1288 return PERMISSION_DENIED; 1289 } 1290 1291 AutoMutex lock(mHardwareLock); 1292 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 1293 mHardwareStatus = AUDIO_HW_SET_VOICE_VOLUME; 1294 ret = dev->set_voice_volume(dev, value); 1295 mHardwareStatus = AUDIO_HW_IDLE; 1296 1297 return ret; 1298} 1299 1300status_t AudioFlinger::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, 1301 audio_io_handle_t output) const 1302{ 1303 Mutex::Autolock _l(mLock); 1304 1305 PlaybackThread *playbackThread = checkPlaybackThread_l(output); 1306 if (playbackThread != NULL) { 1307 return playbackThread->getRenderPosition(halFrames, dspFrames); 1308 } 1309 1310 return BAD_VALUE; 1311} 1312 1313void AudioFlinger::registerClient(const sp<IAudioFlingerClient>& client) 1314{ 1315 Mutex::Autolock _l(mLock); 1316 if (client == 0) { 1317 return; 1318 } 1319 pid_t pid = IPCThreadState::self()->getCallingPid(); 1320 { 1321 Mutex::Autolock _cl(mClientLock); 1322 if (mNotificationClients.indexOfKey(pid) < 0) { 1323 sp<NotificationClient> notificationClient = new NotificationClient(this, 1324 client, 1325 pid); 1326 ALOGV("registerClient() client %p, pid %d", notificationClient.get(), pid); 1327 1328 mNotificationClients.add(pid, notificationClient); 1329 1330 sp<IBinder> binder = IInterface::asBinder(client); 1331 binder->linkToDeath(notificationClient); 1332 } 1333 } 1334 1335 // mClientLock should not be held here because ThreadBase::sendIoConfigEvent() will lock the 1336 // ThreadBase mutex and the locking order is ThreadBase::mLock then AudioFlinger::mClientLock. 1337 // the config change is always sent from playback or record threads to avoid deadlock 1338 // with AudioSystem::gLock 1339 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1340 mPlaybackThreads.valueAt(i)->sendIoConfigEvent(AUDIO_OUTPUT_OPENED, pid); 1341 } 1342 1343 for (size_t i = 0; i < mRecordThreads.size(); i++) { 1344 mRecordThreads.valueAt(i)->sendIoConfigEvent(AUDIO_INPUT_OPENED, pid); 1345 } 1346} 1347 1348void AudioFlinger::removeNotificationClient(pid_t pid) 1349{ 1350 Mutex::Autolock _l(mLock); 1351 { 1352 Mutex::Autolock _cl(mClientLock); 1353 mNotificationClients.removeItem(pid); 1354 } 1355 1356 ALOGV("%d died, releasing its sessions", pid); 1357 size_t num = mAudioSessionRefs.size(); 1358 bool removed = false; 1359 for (size_t i = 0; i< num; ) { 1360 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 1361 ALOGV(" pid %d @ %zu", ref->mPid, i); 1362 if (ref->mPid == pid) { 1363 ALOGV(" removing entry for pid %d session %d", pid, ref->mSessionid); 1364 mAudioSessionRefs.removeAt(i); 1365 delete ref; 1366 removed = true; 1367 num--; 1368 } else { 1369 i++; 1370 } 1371 } 1372 if (removed) { 1373 purgeStaleEffects_l(); 1374 } 1375} 1376 1377void AudioFlinger::ioConfigChanged(audio_io_config_event event, 1378 const sp<AudioIoDescriptor>& ioDesc, 1379 pid_t pid) 1380{ 1381 Mutex::Autolock _l(mClientLock); 1382 size_t size = mNotificationClients.size(); 1383 for (size_t i = 0; i < size; i++) { 1384 if ((pid == 0) || (mNotificationClients.keyAt(i) == pid)) { 1385 mNotificationClients.valueAt(i)->audioFlingerClient()->ioConfigChanged(event, ioDesc); 1386 } 1387 } 1388} 1389 1390// removeClient_l() must be called with AudioFlinger::mClientLock held 1391void AudioFlinger::removeClient_l(pid_t pid) 1392{ 1393 ALOGV("removeClient_l() pid %d, calling pid %d", pid, 1394 IPCThreadState::self()->getCallingPid()); 1395 mClients.removeItem(pid); 1396} 1397 1398// getEffectThread_l() must be called with AudioFlinger::mLock held 1399sp<AudioFlinger::PlaybackThread> AudioFlinger::getEffectThread_l(audio_session_t sessionId, 1400 int EffectId) 1401{ 1402 sp<PlaybackThread> thread; 1403 1404 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1405 if (mPlaybackThreads.valueAt(i)->getEffect(sessionId, EffectId) != 0) { 1406 ALOG_ASSERT(thread == 0); 1407 thread = mPlaybackThreads.valueAt(i); 1408 } 1409 } 1410 1411 return thread; 1412} 1413 1414 1415 1416// ---------------------------------------------------------------------------- 1417 1418AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid) 1419 : RefBase(), 1420 mAudioFlinger(audioFlinger), 1421 mPid(pid) 1422{ 1423 size_t heapSize = kClientSharedHeapSizeBytes; 1424 // Increase heap size on non low ram devices to limit risk of reconnection failure for 1425 // invalidated tracks 1426 if (!audioFlinger->isLowRamDevice()) { 1427 heapSize *= kClientSharedHeapSizeMultiplier; 1428 } 1429 mMemoryDealer = new MemoryDealer(heapSize, "AudioFlinger::Client"); 1430} 1431 1432// Client destructor must be called with AudioFlinger::mClientLock held 1433AudioFlinger::Client::~Client() 1434{ 1435 mAudioFlinger->removeClient_l(mPid); 1436} 1437 1438sp<MemoryDealer> AudioFlinger::Client::heap() const 1439{ 1440 return mMemoryDealer; 1441} 1442 1443// ---------------------------------------------------------------------------- 1444 1445AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger, 1446 const sp<IAudioFlingerClient>& client, 1447 pid_t pid) 1448 : mAudioFlinger(audioFlinger), mPid(pid), mAudioFlingerClient(client) 1449{ 1450} 1451 1452AudioFlinger::NotificationClient::~NotificationClient() 1453{ 1454} 1455 1456void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who __unused) 1457{ 1458 sp<NotificationClient> keep(this); 1459 mAudioFlinger->removeNotificationClient(mPid); 1460} 1461 1462 1463// ---------------------------------------------------------------------------- 1464 1465sp<IAudioRecord> AudioFlinger::openRecord( 1466 audio_io_handle_t input, 1467 uint32_t sampleRate, 1468 audio_format_t format, 1469 audio_channel_mask_t channelMask, 1470 const String16& opPackageName, 1471 size_t *frameCount, 1472 audio_input_flags_t *flags, 1473 pid_t pid, 1474 pid_t tid, 1475 int clientUid, 1476 audio_session_t *sessionId, 1477 size_t *notificationFrames, 1478 sp<IMemory>& cblk, 1479 sp<IMemory>& buffers, 1480 status_t *status) 1481{ 1482 sp<RecordThread::RecordTrack> recordTrack; 1483 sp<RecordHandle> recordHandle; 1484 sp<Client> client; 1485 status_t lStatus; 1486 audio_session_t lSessionId; 1487 1488 cblk.clear(); 1489 buffers.clear(); 1490 1491 bool updatePid = (pid == -1); 1492 const uid_t callingUid = IPCThreadState::self()->getCallingUid(); 1493 if (!isTrustedCallingUid(callingUid)) { 1494 ALOGW_IF((uid_t)clientUid != callingUid, 1495 "%s uid %d tried to pass itself off as %d", __FUNCTION__, callingUid, clientUid); 1496 clientUid = callingUid; 1497 updatePid = true; 1498 } 1499 1500 if (updatePid) { 1501 const pid_t callingPid = IPCThreadState::self()->getCallingPid(); 1502 ALOGW_IF(pid != -1 && pid != callingPid, 1503 "%s uid %d pid %d tried to pass itself off as pid %d", 1504 __func__, callingUid, callingPid, pid); 1505 pid = callingPid; 1506 } 1507 1508 // check calling permissions 1509 if (!recordingAllowed(opPackageName, tid, clientUid)) { 1510 ALOGE("openRecord() permission denied: recording not allowed"); 1511 lStatus = PERMISSION_DENIED; 1512 goto Exit; 1513 } 1514 1515 // further sample rate checks are performed by createRecordTrack_l() 1516 if (sampleRate == 0) { 1517 ALOGE("openRecord() invalid sample rate %u", sampleRate); 1518 lStatus = BAD_VALUE; 1519 goto Exit; 1520 } 1521 1522 // we don't yet support anything other than linear PCM 1523 if (!audio_is_valid_format(format) || !audio_is_linear_pcm(format)) { 1524 ALOGE("openRecord() invalid format %#x", format); 1525 lStatus = BAD_VALUE; 1526 goto Exit; 1527 } 1528 1529 // further channel mask checks are performed by createRecordTrack_l() 1530 if (!audio_is_input_channel(channelMask)) { 1531 ALOGE("openRecord() invalid channel mask %#x", channelMask); 1532 lStatus = BAD_VALUE; 1533 goto Exit; 1534 } 1535 1536 { 1537 Mutex::Autolock _l(mLock); 1538 RecordThread *thread = checkRecordThread_l(input); 1539 if (thread == NULL) { 1540 ALOGE("openRecord() checkRecordThread_l failed"); 1541 lStatus = BAD_VALUE; 1542 goto Exit; 1543 } 1544 1545 client = registerPid(pid); 1546 1547 if (sessionId != NULL && *sessionId != AUDIO_SESSION_ALLOCATE) { 1548 if (audio_unique_id_get_use(*sessionId) != AUDIO_UNIQUE_ID_USE_SESSION) { 1549 lStatus = BAD_VALUE; 1550 goto Exit; 1551 } 1552 lSessionId = *sessionId; 1553 } else { 1554 // if no audio session id is provided, create one here 1555 lSessionId = (audio_session_t) nextUniqueId(AUDIO_UNIQUE_ID_USE_SESSION); 1556 if (sessionId != NULL) { 1557 *sessionId = lSessionId; 1558 } 1559 } 1560 ALOGV("openRecord() lSessionId: %d input %d", lSessionId, input); 1561 1562 recordTrack = thread->createRecordTrack_l(client, sampleRate, format, channelMask, 1563 frameCount, lSessionId, notificationFrames, 1564 clientUid, flags, tid, &lStatus); 1565 LOG_ALWAYS_FATAL_IF((lStatus == NO_ERROR) && (recordTrack == 0)); 1566 1567 if (lStatus == NO_ERROR) { 1568 // Check if one effect chain was awaiting for an AudioRecord to be created on this 1569 // session and move it to this thread. 1570 sp<EffectChain> chain = getOrphanEffectChain_l(lSessionId); 1571 if (chain != 0) { 1572 Mutex::Autolock _l(thread->mLock); 1573 thread->addEffectChain_l(chain); 1574 } 1575 } 1576 } 1577 1578 if (lStatus != NO_ERROR) { 1579 // remove local strong reference to Client before deleting the RecordTrack so that the 1580 // Client destructor is called by the TrackBase destructor with mClientLock held 1581 // Don't hold mClientLock when releasing the reference on the track as the 1582 // destructor will acquire it. 1583 { 1584 Mutex::Autolock _cl(mClientLock); 1585 client.clear(); 1586 } 1587 recordTrack.clear(); 1588 goto Exit; 1589 } 1590 1591 cblk = recordTrack->getCblk(); 1592 buffers = recordTrack->getBuffers(); 1593 1594 // return handle to client 1595 recordHandle = new RecordHandle(recordTrack); 1596 1597Exit: 1598 *status = lStatus; 1599 return recordHandle; 1600} 1601 1602 1603 1604// ---------------------------------------------------------------------------- 1605 1606audio_module_handle_t AudioFlinger::loadHwModule(const char *name) 1607{ 1608 if (name == NULL) { 1609 return AUDIO_MODULE_HANDLE_NONE; 1610 } 1611 if (!settingsAllowed()) { 1612 return AUDIO_MODULE_HANDLE_NONE; 1613 } 1614 Mutex::Autolock _l(mLock); 1615 return loadHwModule_l(name); 1616} 1617 1618// loadHwModule_l() must be called with AudioFlinger::mLock held 1619audio_module_handle_t AudioFlinger::loadHwModule_l(const char *name) 1620{ 1621 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 1622 if (strncmp(mAudioHwDevs.valueAt(i)->moduleName(), name, strlen(name)) == 0) { 1623 ALOGW("loadHwModule() module %s already loaded", name); 1624 return mAudioHwDevs.keyAt(i); 1625 } 1626 } 1627 1628 audio_hw_device_t *dev; 1629 1630 int rc = load_audio_interface(name, &dev); 1631 if (rc) { 1632 ALOGE("loadHwModule() error %d loading module %s", rc, name); 1633 return AUDIO_MODULE_HANDLE_NONE; 1634 } 1635 1636 mHardwareStatus = AUDIO_HW_INIT; 1637 rc = dev->init_check(dev); 1638 mHardwareStatus = AUDIO_HW_IDLE; 1639 if (rc) { 1640 ALOGE("loadHwModule() init check error %d for module %s", rc, name); 1641 return AUDIO_MODULE_HANDLE_NONE; 1642 } 1643 1644 // Check and cache this HAL's level of support for master mute and master 1645 // volume. If this is the first HAL opened, and it supports the get 1646 // methods, use the initial values provided by the HAL as the current 1647 // master mute and volume settings. 1648 1649 AudioHwDevice::Flags flags = static_cast<AudioHwDevice::Flags>(0); 1650 { // scope for auto-lock pattern 1651 AutoMutex lock(mHardwareLock); 1652 1653 if (0 == mAudioHwDevs.size()) { 1654 mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME; 1655 if (NULL != dev->get_master_volume) { 1656 float mv; 1657 if (OK == dev->get_master_volume(dev, &mv)) { 1658 mMasterVolume = mv; 1659 } 1660 } 1661 1662 mHardwareStatus = AUDIO_HW_GET_MASTER_MUTE; 1663 if (NULL != dev->get_master_mute) { 1664 bool mm; 1665 if (OK == dev->get_master_mute(dev, &mm)) { 1666 mMasterMute = mm; 1667 } 1668 } 1669 } 1670 1671 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 1672 if ((NULL != dev->set_master_volume) && 1673 (OK == dev->set_master_volume(dev, mMasterVolume))) { 1674 flags = static_cast<AudioHwDevice::Flags>(flags | 1675 AudioHwDevice::AHWD_CAN_SET_MASTER_VOLUME); 1676 } 1677 1678 mHardwareStatus = AUDIO_HW_SET_MASTER_MUTE; 1679 if ((NULL != dev->set_master_mute) && 1680 (OK == dev->set_master_mute(dev, mMasterMute))) { 1681 flags = static_cast<AudioHwDevice::Flags>(flags | 1682 AudioHwDevice::AHWD_CAN_SET_MASTER_MUTE); 1683 } 1684 1685 mHardwareStatus = AUDIO_HW_IDLE; 1686 } 1687 1688 audio_module_handle_t handle = (audio_module_handle_t) nextUniqueId(AUDIO_UNIQUE_ID_USE_MODULE); 1689 mAudioHwDevs.add(handle, new AudioHwDevice(handle, name, dev, flags)); 1690 1691 ALOGI("loadHwModule() Loaded %s audio interface from %s (%s) handle %d", 1692 name, dev->common.module->name, dev->common.module->id, handle); 1693 1694 return handle; 1695 1696} 1697 1698// ---------------------------------------------------------------------------- 1699 1700uint32_t AudioFlinger::getPrimaryOutputSamplingRate() 1701{ 1702 Mutex::Autolock _l(mLock); 1703 PlaybackThread *thread = fastPlaybackThread_l(); 1704 return thread != NULL ? thread->sampleRate() : 0; 1705} 1706 1707size_t AudioFlinger::getPrimaryOutputFrameCount() 1708{ 1709 Mutex::Autolock _l(mLock); 1710 PlaybackThread *thread = fastPlaybackThread_l(); 1711 return thread != NULL ? thread->frameCountHAL() : 0; 1712} 1713 1714// ---------------------------------------------------------------------------- 1715 1716status_t AudioFlinger::setLowRamDevice(bool isLowRamDevice) 1717{ 1718 uid_t uid = IPCThreadState::self()->getCallingUid(); 1719 if (uid != AID_SYSTEM) { 1720 return PERMISSION_DENIED; 1721 } 1722 Mutex::Autolock _l(mLock); 1723 if (mIsDeviceTypeKnown) { 1724 return INVALID_OPERATION; 1725 } 1726 mIsLowRamDevice = isLowRamDevice; 1727 mIsDeviceTypeKnown = true; 1728 return NO_ERROR; 1729} 1730 1731audio_hw_sync_t AudioFlinger::getAudioHwSyncForSession(audio_session_t sessionId) 1732{ 1733 Mutex::Autolock _l(mLock); 1734 1735 ssize_t index = mHwAvSyncIds.indexOfKey(sessionId); 1736 if (index >= 0) { 1737 ALOGV("getAudioHwSyncForSession found ID %d for session %d", 1738 mHwAvSyncIds.valueAt(index), sessionId); 1739 return mHwAvSyncIds.valueAt(index); 1740 } 1741 1742 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 1743 if (dev == NULL) { 1744 return AUDIO_HW_SYNC_INVALID; 1745 } 1746 char *reply = dev->get_parameters(dev, AUDIO_PARAMETER_HW_AV_SYNC); 1747 AudioParameter param = AudioParameter(String8(reply)); 1748 free(reply); 1749 1750 int value; 1751 if (param.getInt(String8(AUDIO_PARAMETER_HW_AV_SYNC), value) != NO_ERROR) { 1752 ALOGW("getAudioHwSyncForSession error getting sync for session %d", sessionId); 1753 return AUDIO_HW_SYNC_INVALID; 1754 } 1755 1756 // allow only one session for a given HW A/V sync ID. 1757 for (size_t i = 0; i < mHwAvSyncIds.size(); i++) { 1758 if (mHwAvSyncIds.valueAt(i) == (audio_hw_sync_t)value) { 1759 ALOGV("getAudioHwSyncForSession removing ID %d for session %d", 1760 value, mHwAvSyncIds.keyAt(i)); 1761 mHwAvSyncIds.removeItemsAt(i); 1762 break; 1763 } 1764 } 1765 1766 mHwAvSyncIds.add(sessionId, value); 1767 1768 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1769 sp<PlaybackThread> thread = mPlaybackThreads.valueAt(i); 1770 uint32_t sessions = thread->hasAudioSession(sessionId); 1771 if (sessions & ThreadBase::TRACK_SESSION) { 1772 AudioParameter param = AudioParameter(); 1773 param.addInt(String8(AUDIO_PARAMETER_STREAM_HW_AV_SYNC), value); 1774 thread->setParameters(param.toString()); 1775 break; 1776 } 1777 } 1778 1779 ALOGV("getAudioHwSyncForSession adding ID %d for session %d", value, sessionId); 1780 return (audio_hw_sync_t)value; 1781} 1782 1783status_t AudioFlinger::systemReady() 1784{ 1785 Mutex::Autolock _l(mLock); 1786 ALOGI("%s", __FUNCTION__); 1787 if (mSystemReady) { 1788 ALOGW("%s called twice", __FUNCTION__); 1789 return NO_ERROR; 1790 } 1791 mSystemReady = true; 1792 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1793 ThreadBase *thread = (ThreadBase *)mPlaybackThreads.valueAt(i).get(); 1794 thread->systemReady(); 1795 } 1796 for (size_t i = 0; i < mRecordThreads.size(); i++) { 1797 ThreadBase *thread = (ThreadBase *)mRecordThreads.valueAt(i).get(); 1798 thread->systemReady(); 1799 } 1800 return NO_ERROR; 1801} 1802 1803// setAudioHwSyncForSession_l() must be called with AudioFlinger::mLock held 1804void AudioFlinger::setAudioHwSyncForSession_l(PlaybackThread *thread, audio_session_t sessionId) 1805{ 1806 ssize_t index = mHwAvSyncIds.indexOfKey(sessionId); 1807 if (index >= 0) { 1808 audio_hw_sync_t syncId = mHwAvSyncIds.valueAt(index); 1809 ALOGV("setAudioHwSyncForSession_l found ID %d for session %d", syncId, sessionId); 1810 AudioParameter param = AudioParameter(); 1811 param.addInt(String8(AUDIO_PARAMETER_STREAM_HW_AV_SYNC), syncId); 1812 thread->setParameters(param.toString()); 1813 } 1814} 1815 1816 1817// ---------------------------------------------------------------------------- 1818 1819 1820sp<AudioFlinger::PlaybackThread> AudioFlinger::openOutput_l(audio_module_handle_t module, 1821 audio_io_handle_t *output, 1822 audio_config_t *config, 1823 audio_devices_t devices, 1824 const String8& address, 1825 audio_output_flags_t flags) 1826{ 1827 AudioHwDevice *outHwDev = findSuitableHwDev_l(module, devices); 1828 if (outHwDev == NULL) { 1829 return 0; 1830 } 1831 1832 if (*output == AUDIO_IO_HANDLE_NONE) { 1833 *output = nextUniqueId(AUDIO_UNIQUE_ID_USE_OUTPUT); 1834 } else { 1835 // Audio Policy does not currently request a specific output handle. 1836 // If this is ever needed, see openInput_l() for example code. 1837 ALOGE("openOutput_l requested output handle %d is not AUDIO_IO_HANDLE_NONE", *output); 1838 return 0; 1839 } 1840 1841 mHardwareStatus = AUDIO_HW_OUTPUT_OPEN; 1842 1843 // FOR TESTING ONLY: 1844 // This if statement allows overriding the audio policy settings 1845 // and forcing a specific format or channel mask to the HAL/Sink device for testing. 1846 if (!(flags & (AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD | AUDIO_OUTPUT_FLAG_DIRECT))) { 1847 // Check only for Normal Mixing mode 1848 if (kEnableExtendedPrecision) { 1849 // Specify format (uncomment one below to choose) 1850 //config->format = AUDIO_FORMAT_PCM_FLOAT; 1851 //config->format = AUDIO_FORMAT_PCM_24_BIT_PACKED; 1852 //config->format = AUDIO_FORMAT_PCM_32_BIT; 1853 //config->format = AUDIO_FORMAT_PCM_8_24_BIT; 1854 // ALOGV("openOutput_l() upgrading format to %#08x", config->format); 1855 } 1856 if (kEnableExtendedChannels) { 1857 // Specify channel mask (uncomment one below to choose) 1858 //config->channel_mask = audio_channel_out_mask_from_count(4); // for USB 4ch 1859 //config->channel_mask = audio_channel_mask_from_representation_and_bits( 1860 // AUDIO_CHANNEL_REPRESENTATION_INDEX, (1 << 4) - 1); // another 4ch example 1861 } 1862 } 1863 1864 AudioStreamOut *outputStream = NULL; 1865 status_t status = outHwDev->openOutputStream( 1866 &outputStream, 1867 *output, 1868 devices, 1869 flags, 1870 config, 1871 address.string()); 1872 1873 mHardwareStatus = AUDIO_HW_IDLE; 1874 1875 if (status == NO_ERROR) { 1876 1877 PlaybackThread *thread; 1878 if (flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) { 1879 thread = new OffloadThread(this, outputStream, *output, devices, mSystemReady); 1880 ALOGV("openOutput_l() created offload output: ID %d thread %p", *output, thread); 1881 } else if ((flags & AUDIO_OUTPUT_FLAG_DIRECT) 1882 || !isValidPcmSinkFormat(config->format) 1883 || !isValidPcmSinkChannelMask(config->channel_mask)) { 1884 thread = new DirectOutputThread(this, outputStream, *output, devices, mSystemReady); 1885 ALOGV("openOutput_l() created direct output: ID %d thread %p", *output, thread); 1886 } else { 1887 thread = new MixerThread(this, outputStream, *output, devices, mSystemReady); 1888 ALOGV("openOutput_l() created mixer output: ID %d thread %p", *output, thread); 1889 } 1890 mPlaybackThreads.add(*output, thread); 1891 return thread; 1892 } 1893 1894 return 0; 1895} 1896 1897status_t AudioFlinger::openOutput(audio_module_handle_t module, 1898 audio_io_handle_t *output, 1899 audio_config_t *config, 1900 audio_devices_t *devices, 1901 const String8& address, 1902 uint32_t *latencyMs, 1903 audio_output_flags_t flags) 1904{ 1905 ALOGI("openOutput(), module %d Device %x, SamplingRate %d, Format %#08x, Channels %x, flags %x", 1906 module, 1907 (devices != NULL) ? *devices : 0, 1908 config->sample_rate, 1909 config->format, 1910 config->channel_mask, 1911 flags); 1912 1913 if (*devices == AUDIO_DEVICE_NONE) { 1914 return BAD_VALUE; 1915 } 1916 1917 Mutex::Autolock _l(mLock); 1918 1919 sp<PlaybackThread> thread = openOutput_l(module, output, config, *devices, address, flags); 1920 if (thread != 0) { 1921 *latencyMs = thread->latency(); 1922 1923 // notify client processes of the new output creation 1924 thread->ioConfigChanged(AUDIO_OUTPUT_OPENED); 1925 1926 // the first primary output opened designates the primary hw device 1927 if ((mPrimaryHardwareDev == NULL) && (flags & AUDIO_OUTPUT_FLAG_PRIMARY)) { 1928 ALOGI("Using module %d has the primary audio interface", module); 1929 mPrimaryHardwareDev = thread->getOutput()->audioHwDev; 1930 1931 AutoMutex lock(mHardwareLock); 1932 mHardwareStatus = AUDIO_HW_SET_MODE; 1933 mPrimaryHardwareDev->hwDevice()->set_mode(mPrimaryHardwareDev->hwDevice(), mMode); 1934 mHardwareStatus = AUDIO_HW_IDLE; 1935 } 1936 return NO_ERROR; 1937 } 1938 1939 return NO_INIT; 1940} 1941 1942audio_io_handle_t AudioFlinger::openDuplicateOutput(audio_io_handle_t output1, 1943 audio_io_handle_t output2) 1944{ 1945 Mutex::Autolock _l(mLock); 1946 MixerThread *thread1 = checkMixerThread_l(output1); 1947 MixerThread *thread2 = checkMixerThread_l(output2); 1948 1949 if (thread1 == NULL || thread2 == NULL) { 1950 ALOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1, 1951 output2); 1952 return AUDIO_IO_HANDLE_NONE; 1953 } 1954 1955 audio_io_handle_t id = nextUniqueId(AUDIO_UNIQUE_ID_USE_OUTPUT); 1956 DuplicatingThread *thread = new DuplicatingThread(this, thread1, id, mSystemReady); 1957 thread->addOutputTrack(thread2); 1958 mPlaybackThreads.add(id, thread); 1959 // notify client processes of the new output creation 1960 thread->ioConfigChanged(AUDIO_OUTPUT_OPENED); 1961 return id; 1962} 1963 1964status_t AudioFlinger::closeOutput(audio_io_handle_t output) 1965{ 1966 return closeOutput_nonvirtual(output); 1967} 1968 1969status_t AudioFlinger::closeOutput_nonvirtual(audio_io_handle_t output) 1970{ 1971 // keep strong reference on the playback thread so that 1972 // it is not destroyed while exit() is executed 1973 sp<PlaybackThread> thread; 1974 { 1975 Mutex::Autolock _l(mLock); 1976 thread = checkPlaybackThread_l(output); 1977 if (thread == NULL) { 1978 return BAD_VALUE; 1979 } 1980 1981 ALOGV("closeOutput() %d", output); 1982 1983 if (thread->type() == ThreadBase::MIXER) { 1984 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1985 if (mPlaybackThreads.valueAt(i)->isDuplicating()) { 1986 DuplicatingThread *dupThread = 1987 (DuplicatingThread *)mPlaybackThreads.valueAt(i).get(); 1988 dupThread->removeOutputTrack((MixerThread *)thread.get()); 1989 } 1990 } 1991 } 1992 1993 1994 mPlaybackThreads.removeItem(output); 1995 // save all effects to the default thread 1996 if (mPlaybackThreads.size()) { 1997 PlaybackThread *dstThread = checkPlaybackThread_l(mPlaybackThreads.keyAt(0)); 1998 if (dstThread != NULL) { 1999 // audioflinger lock is held here so the acquisition order of thread locks does not 2000 // matter 2001 Mutex::Autolock _dl(dstThread->mLock); 2002 Mutex::Autolock _sl(thread->mLock); 2003 Vector< sp<EffectChain> > effectChains = thread->getEffectChains_l(); 2004 for (size_t i = 0; i < effectChains.size(); i ++) { 2005 moveEffectChain_l(effectChains[i]->sessionId(), thread.get(), dstThread, true); 2006 } 2007 } 2008 } 2009 const sp<AudioIoDescriptor> ioDesc = new AudioIoDescriptor(); 2010 ioDesc->mIoHandle = output; 2011 ioConfigChanged(AUDIO_OUTPUT_CLOSED, ioDesc); 2012 } 2013 thread->exit(); 2014 // The thread entity (active unit of execution) is no longer running here, 2015 // but the ThreadBase container still exists. 2016 2017 if (!thread->isDuplicating()) { 2018 closeOutputFinish(thread); 2019 } 2020 2021 return NO_ERROR; 2022} 2023 2024void AudioFlinger::closeOutputFinish(sp<PlaybackThread> thread) 2025{ 2026 AudioStreamOut *out = thread->clearOutput(); 2027 ALOG_ASSERT(out != NULL, "out shouldn't be NULL"); 2028 // from now on thread->mOutput is NULL 2029 out->hwDev()->close_output_stream(out->hwDev(), out->stream); 2030 delete out; 2031} 2032 2033void AudioFlinger::closeOutputInternal_l(sp<PlaybackThread> thread) 2034{ 2035 mPlaybackThreads.removeItem(thread->mId); 2036 thread->exit(); 2037 closeOutputFinish(thread); 2038} 2039 2040status_t AudioFlinger::suspendOutput(audio_io_handle_t output) 2041{ 2042 Mutex::Autolock _l(mLock); 2043 PlaybackThread *thread = checkPlaybackThread_l(output); 2044 2045 if (thread == NULL) { 2046 return BAD_VALUE; 2047 } 2048 2049 ALOGV("suspendOutput() %d", output); 2050 thread->suspend(); 2051 2052 return NO_ERROR; 2053} 2054 2055status_t AudioFlinger::restoreOutput(audio_io_handle_t output) 2056{ 2057 Mutex::Autolock _l(mLock); 2058 PlaybackThread *thread = checkPlaybackThread_l(output); 2059 2060 if (thread == NULL) { 2061 return BAD_VALUE; 2062 } 2063 2064 ALOGV("restoreOutput() %d", output); 2065 2066 thread->restore(); 2067 2068 return NO_ERROR; 2069} 2070 2071status_t AudioFlinger::openInput(audio_module_handle_t module, 2072 audio_io_handle_t *input, 2073 audio_config_t *config, 2074 audio_devices_t *devices, 2075 const String8& address, 2076 audio_source_t source, 2077 audio_input_flags_t flags) 2078{ 2079 Mutex::Autolock _l(mLock); 2080 2081 if (*devices == AUDIO_DEVICE_NONE) { 2082 return BAD_VALUE; 2083 } 2084 2085 sp<RecordThread> thread = openInput_l(module, input, config, *devices, address, source, flags); 2086 2087 if (thread != 0) { 2088 // notify client processes of the new input creation 2089 thread->ioConfigChanged(AUDIO_INPUT_OPENED); 2090 return NO_ERROR; 2091 } 2092 return NO_INIT; 2093} 2094 2095sp<AudioFlinger::RecordThread> AudioFlinger::openInput_l(audio_module_handle_t module, 2096 audio_io_handle_t *input, 2097 audio_config_t *config, 2098 audio_devices_t devices, 2099 const String8& address, 2100 audio_source_t source, 2101 audio_input_flags_t flags) 2102{ 2103 AudioHwDevice *inHwDev = findSuitableHwDev_l(module, devices); 2104 if (inHwDev == NULL) { 2105 *input = AUDIO_IO_HANDLE_NONE; 2106 return 0; 2107 } 2108 2109 // Audio Policy can request a specific handle for hardware hotword. 2110 // The goal here is not to re-open an already opened input. 2111 // It is to use a pre-assigned I/O handle. 2112 if (*input == AUDIO_IO_HANDLE_NONE) { 2113 *input = nextUniqueId(AUDIO_UNIQUE_ID_USE_INPUT); 2114 } else if (audio_unique_id_get_use(*input) != AUDIO_UNIQUE_ID_USE_INPUT) { 2115 ALOGE("openInput_l() requested input handle %d is invalid", *input); 2116 return 0; 2117 } else if (mRecordThreads.indexOfKey(*input) >= 0) { 2118 // This should not happen in a transient state with current design. 2119 ALOGE("openInput_l() requested input handle %d is already assigned", *input); 2120 return 0; 2121 } 2122 2123 audio_config_t halconfig = *config; 2124 audio_hw_device_t *inHwHal = inHwDev->hwDevice(); 2125 audio_stream_in_t *inStream = NULL; 2126 status_t status = inHwHal->open_input_stream(inHwHal, *input, devices, &halconfig, 2127 &inStream, flags, address.string(), source); 2128 ALOGV("openInput_l() openInputStream returned input %p, SamplingRate %d" 2129 ", Format %#x, Channels %x, flags %#x, status %d addr %s", 2130 inStream, 2131 halconfig.sample_rate, 2132 halconfig.format, 2133 halconfig.channel_mask, 2134 flags, 2135 status, address.string()); 2136 2137 // If the input could not be opened with the requested parameters and we can handle the 2138 // conversion internally, try to open again with the proposed parameters. 2139 if (status == BAD_VALUE && 2140 audio_is_linear_pcm(config->format) && 2141 audio_is_linear_pcm(halconfig.format) && 2142 (halconfig.sample_rate <= AUDIO_RESAMPLER_DOWN_RATIO_MAX * config->sample_rate) && 2143 (audio_channel_count_from_in_mask(halconfig.channel_mask) <= FCC_8) && 2144 (audio_channel_count_from_in_mask(config->channel_mask) <= FCC_8)) { 2145 // FIXME describe the change proposed by HAL (save old values so we can log them here) 2146 ALOGV("openInput_l() reopening with proposed sampling rate and channel mask"); 2147 inStream = NULL; 2148 status = inHwHal->open_input_stream(inHwHal, *input, devices, &halconfig, 2149 &inStream, flags, address.string(), source); 2150 // FIXME log this new status; HAL should not propose any further changes 2151 } 2152 2153 if (status == NO_ERROR && inStream != NULL) { 2154 2155#ifdef TEE_SINK 2156 // Try to re-use most recently used Pipe to archive a copy of input for dumpsys, 2157 // or (re-)create if current Pipe is idle and does not match the new format 2158 sp<NBAIO_Sink> teeSink; 2159 enum { 2160 TEE_SINK_NO, // don't copy input 2161 TEE_SINK_NEW, // copy input using a new pipe 2162 TEE_SINK_OLD, // copy input using an existing pipe 2163 } kind; 2164 NBAIO_Format format = Format_from_SR_C(halconfig.sample_rate, 2165 audio_channel_count_from_in_mask(halconfig.channel_mask), halconfig.format); 2166 if (!mTeeSinkInputEnabled) { 2167 kind = TEE_SINK_NO; 2168 } else if (!Format_isValid(format)) { 2169 kind = TEE_SINK_NO; 2170 } else if (mRecordTeeSink == 0) { 2171 kind = TEE_SINK_NEW; 2172 } else if (mRecordTeeSink->getStrongCount() != 1) { 2173 kind = TEE_SINK_NO; 2174 } else if (Format_isEqual(format, mRecordTeeSink->format())) { 2175 kind = TEE_SINK_OLD; 2176 } else { 2177 kind = TEE_SINK_NEW; 2178 } 2179 switch (kind) { 2180 case TEE_SINK_NEW: { 2181 Pipe *pipe = new Pipe(mTeeSinkInputFrames, format); 2182 size_t numCounterOffers = 0; 2183 const NBAIO_Format offers[1] = {format}; 2184 ssize_t index = pipe->negotiate(offers, 1, NULL, numCounterOffers); 2185 ALOG_ASSERT(index == 0); 2186 PipeReader *pipeReader = new PipeReader(*pipe); 2187 numCounterOffers = 0; 2188 index = pipeReader->negotiate(offers, 1, NULL, numCounterOffers); 2189 ALOG_ASSERT(index == 0); 2190 mRecordTeeSink = pipe; 2191 mRecordTeeSource = pipeReader; 2192 teeSink = pipe; 2193 } 2194 break; 2195 case TEE_SINK_OLD: 2196 teeSink = mRecordTeeSink; 2197 break; 2198 case TEE_SINK_NO: 2199 default: 2200 break; 2201 } 2202#endif 2203 2204 AudioStreamIn *inputStream = new AudioStreamIn(inHwDev, inStream, flags); 2205 2206 // Start record thread 2207 // RecordThread requires both input and output device indication to forward to audio 2208 // pre processing modules 2209 sp<RecordThread> thread = new RecordThread(this, 2210 inputStream, 2211 *input, 2212 primaryOutputDevice_l(), 2213 devices, 2214 mSystemReady 2215#ifdef TEE_SINK 2216 , teeSink 2217#endif 2218 ); 2219 mRecordThreads.add(*input, thread); 2220 ALOGV("openInput_l() created record thread: ID %d thread %p", *input, thread.get()); 2221 return thread; 2222 } 2223 2224 *input = AUDIO_IO_HANDLE_NONE; 2225 return 0; 2226} 2227 2228status_t AudioFlinger::closeInput(audio_io_handle_t input) 2229{ 2230 return closeInput_nonvirtual(input); 2231} 2232 2233status_t AudioFlinger::closeInput_nonvirtual(audio_io_handle_t input) 2234{ 2235 // keep strong reference on the record thread so that 2236 // it is not destroyed while exit() is executed 2237 sp<RecordThread> thread; 2238 { 2239 Mutex::Autolock _l(mLock); 2240 thread = checkRecordThread_l(input); 2241 if (thread == 0) { 2242 return BAD_VALUE; 2243 } 2244 2245 ALOGV("closeInput() %d", input); 2246 2247 // If we still have effect chains, it means that a client still holds a handle 2248 // on at least one effect. We must either move the chain to an existing thread with the 2249 // same session ID or put it aside in case a new record thread is opened for a 2250 // new capture on the same session 2251 sp<EffectChain> chain; 2252 { 2253 Mutex::Autolock _sl(thread->mLock); 2254 Vector< sp<EffectChain> > effectChains = thread->getEffectChains_l(); 2255 // Note: maximum one chain per record thread 2256 if (effectChains.size() != 0) { 2257 chain = effectChains[0]; 2258 } 2259 } 2260 if (chain != 0) { 2261 // first check if a record thread is already opened with a client on the same session. 2262 // This should only happen in case of overlap between one thread tear down and the 2263 // creation of its replacement 2264 size_t i; 2265 for (i = 0; i < mRecordThreads.size(); i++) { 2266 sp<RecordThread> t = mRecordThreads.valueAt(i); 2267 if (t == thread) { 2268 continue; 2269 } 2270 if (t->hasAudioSession(chain->sessionId()) != 0) { 2271 Mutex::Autolock _l(t->mLock); 2272 ALOGV("closeInput() found thread %d for effect session %d", 2273 t->id(), chain->sessionId()); 2274 t->addEffectChain_l(chain); 2275 break; 2276 } 2277 } 2278 // put the chain aside if we could not find a record thread with the same session id. 2279 if (i == mRecordThreads.size()) { 2280 putOrphanEffectChain_l(chain); 2281 } 2282 } 2283 const sp<AudioIoDescriptor> ioDesc = new AudioIoDescriptor(); 2284 ioDesc->mIoHandle = input; 2285 ioConfigChanged(AUDIO_INPUT_CLOSED, ioDesc); 2286 mRecordThreads.removeItem(input); 2287 } 2288 // FIXME: calling thread->exit() without mLock held should not be needed anymore now that 2289 // we have a different lock for notification client 2290 closeInputFinish(thread); 2291 return NO_ERROR; 2292} 2293 2294void AudioFlinger::closeInputFinish(sp<RecordThread> thread) 2295{ 2296 thread->exit(); 2297 AudioStreamIn *in = thread->clearInput(); 2298 ALOG_ASSERT(in != NULL, "in shouldn't be NULL"); 2299 // from now on thread->mInput is NULL 2300 in->hwDev()->close_input_stream(in->hwDev(), in->stream); 2301 delete in; 2302} 2303 2304void AudioFlinger::closeInputInternal_l(sp<RecordThread> thread) 2305{ 2306 mRecordThreads.removeItem(thread->mId); 2307 closeInputFinish(thread); 2308} 2309 2310status_t AudioFlinger::invalidateStream(audio_stream_type_t stream) 2311{ 2312 Mutex::Autolock _l(mLock); 2313 ALOGV("invalidateStream() stream %d", stream); 2314 2315 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2316 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 2317 thread->invalidateTracks(stream); 2318 } 2319 2320 return NO_ERROR; 2321} 2322 2323 2324audio_unique_id_t AudioFlinger::newAudioUniqueId(audio_unique_id_use_t use) 2325{ 2326 // This is a binder API, so a malicious client could pass in a bad parameter. 2327 // Check for that before calling the internal API nextUniqueId(). 2328 if ((unsigned) use >= (unsigned) AUDIO_UNIQUE_ID_USE_MAX) { 2329 ALOGE("newAudioUniqueId invalid use %d", use); 2330 return AUDIO_UNIQUE_ID_ALLOCATE; 2331 } 2332 return nextUniqueId(use); 2333} 2334 2335void AudioFlinger::acquireAudioSessionId(audio_session_t audioSession, pid_t pid) 2336{ 2337 Mutex::Autolock _l(mLock); 2338 pid_t caller = IPCThreadState::self()->getCallingPid(); 2339 ALOGV("acquiring %d from %d, for %d", audioSession, caller, pid); 2340 if (pid != -1 && (caller == getpid_cached)) { 2341 caller = pid; 2342 } 2343 2344 { 2345 Mutex::Autolock _cl(mClientLock); 2346 // Ignore requests received from processes not known as notification client. The request 2347 // is likely proxied by mediaserver (e.g CameraService) and releaseAudioSessionId() can be 2348 // called from a different pid leaving a stale session reference. Also we don't know how 2349 // to clear this reference if the client process dies. 2350 if (mNotificationClients.indexOfKey(caller) < 0) { 2351 ALOGW("acquireAudioSessionId() unknown client %d for session %d", caller, audioSession); 2352 return; 2353 } 2354 } 2355 2356 size_t num = mAudioSessionRefs.size(); 2357 for (size_t i = 0; i< num; i++) { 2358 AudioSessionRef *ref = mAudioSessionRefs.editItemAt(i); 2359 if (ref->mSessionid == audioSession && ref->mPid == caller) { 2360 ref->mCnt++; 2361 ALOGV(" incremented refcount to %d", ref->mCnt); 2362 return; 2363 } 2364 } 2365 mAudioSessionRefs.push(new AudioSessionRef(audioSession, caller)); 2366 ALOGV(" added new entry for %d", audioSession); 2367} 2368 2369void AudioFlinger::releaseAudioSessionId(audio_session_t audioSession, pid_t pid) 2370{ 2371 Mutex::Autolock _l(mLock); 2372 pid_t caller = IPCThreadState::self()->getCallingPid(); 2373 ALOGV("releasing %d from %d for %d", audioSession, caller, pid); 2374 if (pid != -1 && (caller == getpid_cached)) { 2375 caller = pid; 2376 } 2377 size_t num = mAudioSessionRefs.size(); 2378 for (size_t i = 0; i< num; i++) { 2379 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 2380 if (ref->mSessionid == audioSession && ref->mPid == caller) { 2381 ref->mCnt--; 2382 ALOGV(" decremented refcount to %d", ref->mCnt); 2383 if (ref->mCnt == 0) { 2384 mAudioSessionRefs.removeAt(i); 2385 delete ref; 2386 purgeStaleEffects_l(); 2387 } 2388 return; 2389 } 2390 } 2391 // If the caller is mediaserver it is likely that the session being released was acquired 2392 // on behalf of a process not in notification clients and we ignore the warning. 2393 ALOGW_IF(caller != getpid_cached, "session id %d not found for pid %d", audioSession, caller); 2394} 2395 2396void AudioFlinger::purgeStaleEffects_l() { 2397 2398 ALOGV("purging stale effects"); 2399 2400 Vector< sp<EffectChain> > chains; 2401 2402 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2403 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 2404 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 2405 sp<EffectChain> ec = t->mEffectChains[j]; 2406 if (ec->sessionId() > AUDIO_SESSION_OUTPUT_MIX) { 2407 chains.push(ec); 2408 } 2409 } 2410 } 2411 for (size_t i = 0; i < mRecordThreads.size(); i++) { 2412 sp<RecordThread> t = mRecordThreads.valueAt(i); 2413 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 2414 sp<EffectChain> ec = t->mEffectChains[j]; 2415 chains.push(ec); 2416 } 2417 } 2418 2419 for (size_t i = 0; i < chains.size(); i++) { 2420 sp<EffectChain> ec = chains[i]; 2421 int sessionid = ec->sessionId(); 2422 sp<ThreadBase> t = ec->mThread.promote(); 2423 if (t == 0) { 2424 continue; 2425 } 2426 size_t numsessionrefs = mAudioSessionRefs.size(); 2427 bool found = false; 2428 for (size_t k = 0; k < numsessionrefs; k++) { 2429 AudioSessionRef *ref = mAudioSessionRefs.itemAt(k); 2430 if (ref->mSessionid == sessionid) { 2431 ALOGV(" session %d still exists for %d with %d refs", 2432 sessionid, ref->mPid, ref->mCnt); 2433 found = true; 2434 break; 2435 } 2436 } 2437 if (!found) { 2438 Mutex::Autolock _l(t->mLock); 2439 // remove all effects from the chain 2440 while (ec->mEffects.size()) { 2441 sp<EffectModule> effect = ec->mEffects[0]; 2442 effect->unPin(); 2443 t->removeEffect_l(effect); 2444 if (effect->purgeHandles()) { 2445 t->checkSuspendOnEffectEnabled_l(effect, false, effect->sessionId()); 2446 } 2447 AudioSystem::unregisterEffect(effect->id()); 2448 } 2449 } 2450 } 2451 return; 2452} 2453 2454// checkThread_l() must be called with AudioFlinger::mLock held 2455AudioFlinger::ThreadBase *AudioFlinger::checkThread_l(audio_io_handle_t ioHandle) const 2456{ 2457 ThreadBase *thread = NULL; 2458 switch (audio_unique_id_get_use(ioHandle)) { 2459 case AUDIO_UNIQUE_ID_USE_OUTPUT: 2460 thread = checkPlaybackThread_l(ioHandle); 2461 break; 2462 case AUDIO_UNIQUE_ID_USE_INPUT: 2463 thread = checkRecordThread_l(ioHandle); 2464 break; 2465 default: 2466 break; 2467 } 2468 return thread; 2469} 2470 2471// checkPlaybackThread_l() must be called with AudioFlinger::mLock held 2472AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(audio_io_handle_t output) const 2473{ 2474 return mPlaybackThreads.valueFor(output).get(); 2475} 2476 2477// checkMixerThread_l() must be called with AudioFlinger::mLock held 2478AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(audio_io_handle_t output) const 2479{ 2480 PlaybackThread *thread = checkPlaybackThread_l(output); 2481 return thread != NULL && thread->type() != ThreadBase::DIRECT ? (MixerThread *) thread : NULL; 2482} 2483 2484// checkRecordThread_l() must be called with AudioFlinger::mLock held 2485AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(audio_io_handle_t input) const 2486{ 2487 return mRecordThreads.valueFor(input).get(); 2488} 2489 2490audio_unique_id_t AudioFlinger::nextUniqueId(audio_unique_id_use_t use) 2491{ 2492 // This is the internal API, so it is OK to assert on bad parameter. 2493 LOG_ALWAYS_FATAL_IF((unsigned) use >= (unsigned) AUDIO_UNIQUE_ID_USE_MAX); 2494 const int maxRetries = use == AUDIO_UNIQUE_ID_USE_SESSION ? 3 : 1; 2495 for (int retry = 0; retry < maxRetries; retry++) { 2496 // The cast allows wraparound from max positive to min negative instead of abort 2497 uint32_t base = (uint32_t) atomic_fetch_add_explicit(&mNextUniqueIds[use], 2498 (uint_fast32_t) AUDIO_UNIQUE_ID_USE_MAX, memory_order_acq_rel); 2499 ALOG_ASSERT(audio_unique_id_get_use(base) == AUDIO_UNIQUE_ID_USE_UNSPECIFIED); 2500 // allow wrap by skipping 0 and -1 for session ids 2501 if (!(base == 0 || base == (~0u & ~AUDIO_UNIQUE_ID_USE_MASK))) { 2502 ALOGW_IF(retry != 0, "unique ID overflow for use %d", use); 2503 return (audio_unique_id_t) (base | use); 2504 } 2505 } 2506 // We have no way of recovering from wraparound 2507 LOG_ALWAYS_FATAL("unique ID overflow for use %d", use); 2508 // TODO Use a floor after wraparound. This may need a mutex. 2509} 2510 2511AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l() const 2512{ 2513 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2514 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 2515 if(thread->isDuplicating()) { 2516 continue; 2517 } 2518 AudioStreamOut *output = thread->getOutput(); 2519 if (output != NULL && output->audioHwDev == mPrimaryHardwareDev) { 2520 return thread; 2521 } 2522 } 2523 return NULL; 2524} 2525 2526audio_devices_t AudioFlinger::primaryOutputDevice_l() const 2527{ 2528 PlaybackThread *thread = primaryPlaybackThread_l(); 2529 2530 if (thread == NULL) { 2531 return 0; 2532 } 2533 2534 return thread->outDevice(); 2535} 2536 2537AudioFlinger::PlaybackThread *AudioFlinger::fastPlaybackThread_l() const 2538{ 2539 size_t minFrameCount = 0; 2540 PlaybackThread *minThread = NULL; 2541 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2542 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 2543 if (!thread->isDuplicating()) { 2544 size_t frameCount = thread->frameCountHAL(); 2545 if (frameCount != 0 && (minFrameCount == 0 || frameCount < minFrameCount || 2546 (frameCount == minFrameCount && thread->hasFastMixer() && 2547 /*minThread != NULL &&*/ !minThread->hasFastMixer()))) { 2548 minFrameCount = frameCount; 2549 minThread = thread; 2550 } 2551 } 2552 } 2553 return minThread; 2554} 2555 2556sp<AudioFlinger::SyncEvent> AudioFlinger::createSyncEvent(AudioSystem::sync_event_t type, 2557 audio_session_t triggerSession, 2558 audio_session_t listenerSession, 2559 sync_event_callback_t callBack, 2560 wp<RefBase> cookie) 2561{ 2562 Mutex::Autolock _l(mLock); 2563 2564 sp<SyncEvent> event = new SyncEvent(type, triggerSession, listenerSession, callBack, cookie); 2565 status_t playStatus = NAME_NOT_FOUND; 2566 status_t recStatus = NAME_NOT_FOUND; 2567 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2568 playStatus = mPlaybackThreads.valueAt(i)->setSyncEvent(event); 2569 if (playStatus == NO_ERROR) { 2570 return event; 2571 } 2572 } 2573 for (size_t i = 0; i < mRecordThreads.size(); i++) { 2574 recStatus = mRecordThreads.valueAt(i)->setSyncEvent(event); 2575 if (recStatus == NO_ERROR) { 2576 return event; 2577 } 2578 } 2579 if (playStatus == NAME_NOT_FOUND || recStatus == NAME_NOT_FOUND) { 2580 mPendingSyncEvents.add(event); 2581 } else { 2582 ALOGV("createSyncEvent() invalid event %d", event->type()); 2583 event.clear(); 2584 } 2585 return event; 2586} 2587 2588// ---------------------------------------------------------------------------- 2589// Effect management 2590// ---------------------------------------------------------------------------- 2591 2592 2593status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects) const 2594{ 2595 Mutex::Autolock _l(mLock); 2596 return EffectQueryNumberEffects(numEffects); 2597} 2598 2599status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor) const 2600{ 2601 Mutex::Autolock _l(mLock); 2602 return EffectQueryEffect(index, descriptor); 2603} 2604 2605status_t AudioFlinger::getEffectDescriptor(const effect_uuid_t *pUuid, 2606 effect_descriptor_t *descriptor) const 2607{ 2608 Mutex::Autolock _l(mLock); 2609 return EffectGetDescriptor(pUuid, descriptor); 2610} 2611 2612 2613sp<IEffect> AudioFlinger::createEffect( 2614 effect_descriptor_t *pDesc, 2615 const sp<IEffectClient>& effectClient, 2616 int32_t priority, 2617 audio_io_handle_t io, 2618 audio_session_t sessionId, 2619 const String16& opPackageName, 2620 status_t *status, 2621 int *id, 2622 int *enabled) 2623{ 2624 status_t lStatus = NO_ERROR; 2625 sp<EffectHandle> handle; 2626 effect_descriptor_t desc; 2627 2628 pid_t pid = IPCThreadState::self()->getCallingPid(); 2629 ALOGV("createEffect pid %d, effectClient %p, priority %d, sessionId %d, io %d", 2630 pid, effectClient.get(), priority, sessionId, io); 2631 2632 if (pDesc == NULL) { 2633 lStatus = BAD_VALUE; 2634 goto Exit; 2635 } 2636 2637 // check audio settings permission for global effects 2638 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && !settingsAllowed()) { 2639 lStatus = PERMISSION_DENIED; 2640 goto Exit; 2641 } 2642 2643 // Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects 2644 // that can only be created by audio policy manager (running in same process) 2645 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid_cached != pid) { 2646 lStatus = PERMISSION_DENIED; 2647 goto Exit; 2648 } 2649 2650 { 2651 if (!EffectIsNullUuid(&pDesc->uuid)) { 2652 // if uuid is specified, request effect descriptor 2653 lStatus = EffectGetDescriptor(&pDesc->uuid, &desc); 2654 if (lStatus < 0) { 2655 ALOGW("createEffect() error %d from EffectGetDescriptor", lStatus); 2656 goto Exit; 2657 } 2658 } else { 2659 // if uuid is not specified, look for an available implementation 2660 // of the required type in effect factory 2661 if (EffectIsNullUuid(&pDesc->type)) { 2662 ALOGW("createEffect() no effect type"); 2663 lStatus = BAD_VALUE; 2664 goto Exit; 2665 } 2666 uint32_t numEffects = 0; 2667 effect_descriptor_t d; 2668 d.flags = 0; // prevent compiler warning 2669 bool found = false; 2670 2671 lStatus = EffectQueryNumberEffects(&numEffects); 2672 if (lStatus < 0) { 2673 ALOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus); 2674 goto Exit; 2675 } 2676 for (uint32_t i = 0; i < numEffects; i++) { 2677 lStatus = EffectQueryEffect(i, &desc); 2678 if (lStatus < 0) { 2679 ALOGW("createEffect() error %d from EffectQueryEffect", lStatus); 2680 continue; 2681 } 2682 if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) { 2683 // If matching type found save effect descriptor. If the session is 2684 // 0 and the effect is not auxiliary, continue enumeration in case 2685 // an auxiliary version of this effect type is available 2686 found = true; 2687 d = desc; 2688 if (sessionId != AUDIO_SESSION_OUTPUT_MIX || 2689 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 2690 break; 2691 } 2692 } 2693 } 2694 if (!found) { 2695 lStatus = BAD_VALUE; 2696 ALOGW("createEffect() effect not found"); 2697 goto Exit; 2698 } 2699 // For same effect type, chose auxiliary version over insert version if 2700 // connect to output mix (Compliance to OpenSL ES) 2701 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && 2702 (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) { 2703 desc = d; 2704 } 2705 } 2706 2707 // Do not allow auxiliary effects on a session different from 0 (output mix) 2708 if (sessionId != AUDIO_SESSION_OUTPUT_MIX && 2709 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 2710 lStatus = INVALID_OPERATION; 2711 goto Exit; 2712 } 2713 2714 // check recording permission for visualizer 2715 if ((memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) && 2716 !recordingAllowed(opPackageName, pid, IPCThreadState::self()->getCallingUid())) { 2717 lStatus = PERMISSION_DENIED; 2718 goto Exit; 2719 } 2720 2721 // return effect descriptor 2722 *pDesc = desc; 2723 if (io == AUDIO_IO_HANDLE_NONE && sessionId == AUDIO_SESSION_OUTPUT_MIX) { 2724 // if the output returned by getOutputForEffect() is removed before we lock the 2725 // mutex below, the call to checkPlaybackThread_l(io) below will detect it 2726 // and we will exit safely 2727 io = AudioSystem::getOutputForEffect(&desc); 2728 ALOGV("createEffect got output %d", io); 2729 } 2730 2731 Mutex::Autolock _l(mLock); 2732 2733 // If output is not specified try to find a matching audio session ID in one of the 2734 // output threads. 2735 // If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX 2736 // because of code checking output when entering the function. 2737 // Note: io is never 0 when creating an effect on an input 2738 if (io == AUDIO_IO_HANDLE_NONE) { 2739 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) { 2740 // output must be specified by AudioPolicyManager when using session 2741 // AUDIO_SESSION_OUTPUT_STAGE 2742 lStatus = BAD_VALUE; 2743 goto Exit; 2744 } 2745 // look for the thread where the specified audio session is present 2746 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2747 if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 2748 io = mPlaybackThreads.keyAt(i); 2749 break; 2750 } 2751 } 2752 if (io == 0) { 2753 for (size_t i = 0; i < mRecordThreads.size(); i++) { 2754 if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 2755 io = mRecordThreads.keyAt(i); 2756 break; 2757 } 2758 } 2759 } 2760 // If no output thread contains the requested session ID, default to 2761 // first output. The effect chain will be moved to the correct output 2762 // thread when a track with the same session ID is created 2763 if (io == AUDIO_IO_HANDLE_NONE && mPlaybackThreads.size() > 0) { 2764 io = mPlaybackThreads.keyAt(0); 2765 } 2766 ALOGV("createEffect() got io %d for effect %s", io, desc.name); 2767 } 2768 ThreadBase *thread = checkRecordThread_l(io); 2769 if (thread == NULL) { 2770 thread = checkPlaybackThread_l(io); 2771 if (thread == NULL) { 2772 ALOGE("createEffect() unknown output thread"); 2773 lStatus = BAD_VALUE; 2774 goto Exit; 2775 } 2776 } else { 2777 // Check if one effect chain was awaiting for an effect to be created on this 2778 // session and used it instead of creating a new one. 2779 sp<EffectChain> chain = getOrphanEffectChain_l(sessionId); 2780 if (chain != 0) { 2781 Mutex::Autolock _l(thread->mLock); 2782 thread->addEffectChain_l(chain); 2783 } 2784 } 2785 2786 sp<Client> client = registerPid(pid); 2787 2788 // create effect on selected output thread 2789 handle = thread->createEffect_l(client, effectClient, priority, sessionId, 2790 &desc, enabled, &lStatus); 2791 if (handle != 0 && id != NULL) { 2792 *id = handle->id(); 2793 } 2794 if (handle == 0) { 2795 // remove local strong reference to Client with mClientLock held 2796 Mutex::Autolock _cl(mClientLock); 2797 client.clear(); 2798 } 2799 } 2800 2801Exit: 2802 *status = lStatus; 2803 return handle; 2804} 2805 2806status_t AudioFlinger::moveEffects(audio_session_t sessionId, audio_io_handle_t srcOutput, 2807 audio_io_handle_t dstOutput) 2808{ 2809 ALOGV("moveEffects() session %d, srcOutput %d, dstOutput %d", 2810 sessionId, srcOutput, dstOutput); 2811 Mutex::Autolock _l(mLock); 2812 if (srcOutput == dstOutput) { 2813 ALOGW("moveEffects() same dst and src outputs %d", dstOutput); 2814 return NO_ERROR; 2815 } 2816 PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput); 2817 if (srcThread == NULL) { 2818 ALOGW("moveEffects() bad srcOutput %d", srcOutput); 2819 return BAD_VALUE; 2820 } 2821 PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput); 2822 if (dstThread == NULL) { 2823 ALOGW("moveEffects() bad dstOutput %d", dstOutput); 2824 return BAD_VALUE; 2825 } 2826 2827 Mutex::Autolock _dl(dstThread->mLock); 2828 Mutex::Autolock _sl(srcThread->mLock); 2829 return moveEffectChain_l(sessionId, srcThread, dstThread, false); 2830} 2831 2832// moveEffectChain_l must be called with both srcThread and dstThread mLocks held 2833status_t AudioFlinger::moveEffectChain_l(audio_session_t sessionId, 2834 AudioFlinger::PlaybackThread *srcThread, 2835 AudioFlinger::PlaybackThread *dstThread, 2836 bool reRegister) 2837{ 2838 ALOGV("moveEffectChain_l() session %d from thread %p to thread %p", 2839 sessionId, srcThread, dstThread); 2840 2841 sp<EffectChain> chain = srcThread->getEffectChain_l(sessionId); 2842 if (chain == 0) { 2843 ALOGW("moveEffectChain_l() effect chain for session %d not on source thread %p", 2844 sessionId, srcThread); 2845 return INVALID_OPERATION; 2846 } 2847 2848 // Check whether the destination thread and all effects in the chain are compatible 2849 if (!chain->isCompatibleWithThread_l(dstThread)) { 2850 ALOGW("moveEffectChain_l() effect chain failed because" 2851 " destination thread %p is not compatible with effects in the chain", 2852 dstThread); 2853 return INVALID_OPERATION; 2854 } 2855 2856 // remove chain first. This is useful only if reconfiguring effect chain on same output thread, 2857 // so that a new chain is created with correct parameters when first effect is added. This is 2858 // otherwise unnecessary as removeEffect_l() will remove the chain when last effect is 2859 // removed. 2860 srcThread->removeEffectChain_l(chain); 2861 2862 // transfer all effects one by one so that new effect chain is created on new thread with 2863 // correct buffer sizes and audio parameters and effect engines reconfigured accordingly 2864 sp<EffectChain> dstChain; 2865 uint32_t strategy = 0; // prevent compiler warning 2866 sp<EffectModule> effect = chain->getEffectFromId_l(0); 2867 Vector< sp<EffectModule> > removed; 2868 status_t status = NO_ERROR; 2869 while (effect != 0) { 2870 srcThread->removeEffect_l(effect); 2871 removed.add(effect); 2872 status = dstThread->addEffect_l(effect); 2873 if (status != NO_ERROR) { 2874 break; 2875 } 2876 // removeEffect_l() has stopped the effect if it was active so it must be restarted 2877 if (effect->state() == EffectModule::ACTIVE || 2878 effect->state() == EffectModule::STOPPING) { 2879 effect->start(); 2880 } 2881 // if the move request is not received from audio policy manager, the effect must be 2882 // re-registered with the new strategy and output 2883 if (dstChain == 0) { 2884 dstChain = effect->chain().promote(); 2885 if (dstChain == 0) { 2886 ALOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get()); 2887 status = NO_INIT; 2888 break; 2889 } 2890 strategy = dstChain->strategy(); 2891 } 2892 if (reRegister) { 2893 AudioSystem::unregisterEffect(effect->id()); 2894 AudioSystem::registerEffect(&effect->desc(), 2895 dstThread->id(), 2896 strategy, 2897 sessionId, 2898 effect->id()); 2899 AudioSystem::setEffectEnabled(effect->id(), effect->isEnabled()); 2900 } 2901 effect = chain->getEffectFromId_l(0); 2902 } 2903 2904 if (status != NO_ERROR) { 2905 for (size_t i = 0; i < removed.size(); i++) { 2906 srcThread->addEffect_l(removed[i]); 2907 if (dstChain != 0 && reRegister) { 2908 AudioSystem::unregisterEffect(removed[i]->id()); 2909 AudioSystem::registerEffect(&removed[i]->desc(), 2910 srcThread->id(), 2911 strategy, 2912 sessionId, 2913 removed[i]->id()); 2914 AudioSystem::setEffectEnabled(effect->id(), effect->isEnabled()); 2915 } 2916 } 2917 } 2918 2919 return status; 2920} 2921 2922bool AudioFlinger::isNonOffloadableGlobalEffectEnabled_l() 2923{ 2924 if (mGlobalEffectEnableTime != 0 && 2925 ((systemTime() - mGlobalEffectEnableTime) < kMinGlobalEffectEnabletimeNs)) { 2926 return true; 2927 } 2928 2929 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2930 sp<EffectChain> ec = 2931 mPlaybackThreads.valueAt(i)->getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2932 if (ec != 0 && ec->isNonOffloadableEnabled()) { 2933 return true; 2934 } 2935 } 2936 return false; 2937} 2938 2939void AudioFlinger::onNonOffloadableGlobalEffectEnable() 2940{ 2941 Mutex::Autolock _l(mLock); 2942 2943 mGlobalEffectEnableTime = systemTime(); 2944 2945 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2946 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 2947 if (t->mType == ThreadBase::OFFLOAD) { 2948 t->invalidateTracks(AUDIO_STREAM_MUSIC); 2949 } 2950 } 2951 2952} 2953 2954status_t AudioFlinger::putOrphanEffectChain_l(const sp<AudioFlinger::EffectChain>& chain) 2955{ 2956 audio_session_t session = chain->sessionId(); 2957 ssize_t index = mOrphanEffectChains.indexOfKey(session); 2958 ALOGV("putOrphanEffectChain_l session %d index %zd", session, index); 2959 if (index >= 0) { 2960 ALOGW("putOrphanEffectChain_l chain for session %d already present", session); 2961 return ALREADY_EXISTS; 2962 } 2963 mOrphanEffectChains.add(session, chain); 2964 return NO_ERROR; 2965} 2966 2967sp<AudioFlinger::EffectChain> AudioFlinger::getOrphanEffectChain_l(audio_session_t session) 2968{ 2969 sp<EffectChain> chain; 2970 ssize_t index = mOrphanEffectChains.indexOfKey(session); 2971 ALOGV("getOrphanEffectChain_l session %d index %zd", session, index); 2972 if (index >= 0) { 2973 chain = mOrphanEffectChains.valueAt(index); 2974 mOrphanEffectChains.removeItemsAt(index); 2975 } 2976 return chain; 2977} 2978 2979bool AudioFlinger::updateOrphanEffectChains(const sp<AudioFlinger::EffectModule>& effect) 2980{ 2981 Mutex::Autolock _l(mLock); 2982 audio_session_t session = effect->sessionId(); 2983 ssize_t index = mOrphanEffectChains.indexOfKey(session); 2984 ALOGV("updateOrphanEffectChains session %d index %zd", session, index); 2985 if (index >= 0) { 2986 sp<EffectChain> chain = mOrphanEffectChains.valueAt(index); 2987 if (chain->removeEffect_l(effect) == 0) { 2988 ALOGV("updateOrphanEffectChains removing effect chain at index %zd", index); 2989 mOrphanEffectChains.removeItemsAt(index); 2990 } 2991 return true; 2992 } 2993 return false; 2994} 2995 2996 2997struct Entry { 2998#define TEE_MAX_FILENAME 32 // %Y%m%d%H%M%S_%d.wav = 4+2+2+2+2+2+1+1+4+1 = 21 2999 char mFileName[TEE_MAX_FILENAME]; 3000}; 3001 3002int comparEntry(const void *p1, const void *p2) 3003{ 3004 return strcmp(((const Entry *) p1)->mFileName, ((const Entry *) p2)->mFileName); 3005} 3006 3007#ifdef TEE_SINK 3008void AudioFlinger::dumpTee(int fd, const sp<NBAIO_Source>& source, audio_io_handle_t id) 3009{ 3010 NBAIO_Source *teeSource = source.get(); 3011 if (teeSource != NULL) { 3012 // .wav rotation 3013 // There is a benign race condition if 2 threads call this simultaneously. 3014 // They would both traverse the directory, but the result would simply be 3015 // failures at unlink() which are ignored. It's also unlikely since 3016 // normally dumpsys is only done by bugreport or from the command line. 3017 char teePath[32+256]; 3018 strcpy(teePath, "/data/misc/audioserver"); 3019 size_t teePathLen = strlen(teePath); 3020 DIR *dir = opendir(teePath); 3021 teePath[teePathLen++] = '/'; 3022 if (dir != NULL) { 3023#define TEE_MAX_SORT 20 // number of entries to sort 3024#define TEE_MAX_KEEP 10 // number of entries to keep 3025 struct Entry entries[TEE_MAX_SORT]; 3026 size_t entryCount = 0; 3027 while (entryCount < TEE_MAX_SORT) { 3028 struct dirent de; 3029 struct dirent *result = NULL; 3030 int rc = readdir_r(dir, &de, &result); 3031 if (rc != 0) { 3032 ALOGW("readdir_r failed %d", rc); 3033 break; 3034 } 3035 if (result == NULL) { 3036 break; 3037 } 3038 if (result != &de) { 3039 ALOGW("readdir_r returned unexpected result %p != %p", result, &de); 3040 break; 3041 } 3042 // ignore non .wav file entries 3043 size_t nameLen = strlen(de.d_name); 3044 if (nameLen <= 4 || nameLen >= TEE_MAX_FILENAME || 3045 strcmp(&de.d_name[nameLen - 4], ".wav")) { 3046 continue; 3047 } 3048 strcpy(entries[entryCount++].mFileName, de.d_name); 3049 } 3050 (void) closedir(dir); 3051 if (entryCount > TEE_MAX_KEEP) { 3052 qsort(entries, entryCount, sizeof(Entry), comparEntry); 3053 for (size_t i = 0; i < entryCount - TEE_MAX_KEEP; ++i) { 3054 strcpy(&teePath[teePathLen], entries[i].mFileName); 3055 (void) unlink(teePath); 3056 } 3057 } 3058 } else { 3059 if (fd >= 0) { 3060 dprintf(fd, "unable to rotate tees in %.*s: %s\n", (int) teePathLen, teePath, 3061 strerror(errno)); 3062 } 3063 } 3064 char teeTime[16]; 3065 struct timeval tv; 3066 gettimeofday(&tv, NULL); 3067 struct tm tm; 3068 localtime_r(&tv.tv_sec, &tm); 3069 strftime(teeTime, sizeof(teeTime), "%Y%m%d%H%M%S", &tm); 3070 snprintf(&teePath[teePathLen], sizeof(teePath) - teePathLen, "%s_%d.wav", teeTime, id); 3071 // if 2 dumpsys are done within 1 second, and rotation didn't work, then discard 2nd 3072 int teeFd = open(teePath, O_WRONLY | O_CREAT | O_EXCL | O_NOFOLLOW, S_IRUSR | S_IWUSR); 3073 if (teeFd >= 0) { 3074 // FIXME use libsndfile 3075 char wavHeader[44]; 3076 memcpy(wavHeader, 3077 "RIFF\0\0\0\0WAVEfmt \20\0\0\0\1\0\2\0\104\254\0\0\0\0\0\0\4\0\20\0data\0\0\0\0", 3078 sizeof(wavHeader)); 3079 NBAIO_Format format = teeSource->format(); 3080 unsigned channelCount = Format_channelCount(format); 3081 uint32_t sampleRate = Format_sampleRate(format); 3082 size_t frameSize = Format_frameSize(format); 3083 wavHeader[22] = channelCount; // number of channels 3084 wavHeader[24] = sampleRate; // sample rate 3085 wavHeader[25] = sampleRate >> 8; 3086 wavHeader[32] = frameSize; // block alignment 3087 wavHeader[33] = frameSize >> 8; 3088 write(teeFd, wavHeader, sizeof(wavHeader)); 3089 size_t total = 0; 3090 bool firstRead = true; 3091#define TEE_SINK_READ 1024 // frames per I/O operation 3092 void *buffer = malloc(TEE_SINK_READ * frameSize); 3093 for (;;) { 3094 size_t count = TEE_SINK_READ; 3095 ssize_t actual = teeSource->read(buffer, count); 3096 bool wasFirstRead = firstRead; 3097 firstRead = false; 3098 if (actual <= 0) { 3099 if (actual == (ssize_t) OVERRUN && wasFirstRead) { 3100 continue; 3101 } 3102 break; 3103 } 3104 ALOG_ASSERT(actual <= (ssize_t)count); 3105 write(teeFd, buffer, actual * frameSize); 3106 total += actual; 3107 } 3108 free(buffer); 3109 lseek(teeFd, (off_t) 4, SEEK_SET); 3110 uint32_t temp = 44 + total * frameSize - 8; 3111 // FIXME not big-endian safe 3112 write(teeFd, &temp, sizeof(temp)); 3113 lseek(teeFd, (off_t) 40, SEEK_SET); 3114 temp = total * frameSize; 3115 // FIXME not big-endian safe 3116 write(teeFd, &temp, sizeof(temp)); 3117 close(teeFd); 3118 if (fd >= 0) { 3119 dprintf(fd, "tee copied to %s\n", teePath); 3120 } 3121 } else { 3122 if (fd >= 0) { 3123 dprintf(fd, "unable to create tee %s: %s\n", teePath, strerror(errno)); 3124 } 3125 } 3126 } 3127} 3128#endif 3129 3130// ---------------------------------------------------------------------------- 3131 3132status_t AudioFlinger::onTransact( 3133 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 3134{ 3135 return BnAudioFlinger::onTransact(code, data, reply, flags); 3136} 3137 3138} // namespace android 3139