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 83 namespace android { 84 85 static const char kDeadlockedString[] = "AudioFlinger may be deadlocked\n"; 86 static const char kHardwareLockedString[] = "Hardware lock is taken\n"; 87 static const char kClientLockedString[] = "Client lock is taken\n"; 88 89 90 nsecs_t AudioFlinger::mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs; 91 92 uint32_t AudioFlinger::mScreenState; 93 94 #ifdef TEE_SINK 95 bool AudioFlinger::mTeeSinkInputEnabled = false; 96 bool AudioFlinger::mTeeSinkOutputEnabled = false; 97 bool AudioFlinger::mTeeSinkTrackEnabled = false; 98 99 size_t AudioFlinger::mTeeSinkInputFrames = kTeeSinkInputFramesDefault; 100 size_t AudioFlinger::mTeeSinkOutputFrames = kTeeSinkOutputFramesDefault; 101 size_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. 106 static const nsecs_t kMinGlobalEffectEnabletimeNs = seconds(7200); 107 108 // ---------------------------------------------------------------------------- 109 110 const 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 } 143 144 static 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; 167 168 out: 169 *dev = NULL; 170 return rc; 171 } 172 173 // ---------------------------------------------------------------------------- 174 175 AudioFlinger::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 } 229 230 void 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); 249 250 mMode = AUDIO_MODE_NORMAL; 251 } 252 253 AudioFlinger::~AudioFlinger() 254 { 255 while (!mRecordThreads.isEmpty()) { 256 // closeInput_nonvirtual() will remove specified entry from mRecordThreads 257 closeInput_nonvirtual(mRecordThreads.keyAt(0)); 258 } 259 while (!mPlaybackThreads.isEmpty()) { 260 // closeOutput_nonvirtual() will remove specified entry from mPlaybackThreads 261 closeOutput_nonvirtual(mPlaybackThreads.keyAt(0)); 262 } 263 264 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 265 // no mHardwareLock needed, as there are no other references to this 266 audio_hw_device_close(mAudioHwDevs.valueAt(i)->hwDevice()); 267 delete mAudioHwDevs.valueAt(i); 268 } 269 270 // Tell media.log service about any old writers that still need to be unregistered 271 if (mLogMemoryDealer != 0) { 272 sp<IBinder> binder = defaultServiceManager()->getService(String16("media.log")); 273 if (binder != 0) { 274 sp<IMediaLogService> mediaLogService(interface_cast<IMediaLogService>(binder)); 275 for (size_t count = mUnregisteredWriters.size(); count > 0; count--) { 276 sp<IMemory> iMemory(mUnregisteredWriters.top()->getIMemory()); 277 mUnregisteredWriters.pop(); 278 mediaLogService->unregisterWriter(iMemory); 279 } 280 } 281 } 282 } 283 284 static const char * const audio_interfaces[] = { 285 AUDIO_HARDWARE_MODULE_ID_PRIMARY, 286 AUDIO_HARDWARE_MODULE_ID_A2DP, 287 AUDIO_HARDWARE_MODULE_ID_USB, 288 }; 289 #define ARRAY_SIZE(x) (sizeof((x))/sizeof(((x)[0]))) 290 291 AudioHwDevice* AudioFlinger::findSuitableHwDev_l( 292 audio_module_handle_t module, 293 audio_devices_t devices) 294 { 295 // if module is 0, the request comes from an old policy manager and we should load 296 // well known modules 297 if (module == 0) { 298 ALOGW("findSuitableHwDev_l() loading well know audio hw modules"); 299 for (size_t i = 0; i < ARRAY_SIZE(audio_interfaces); i++) { 300 loadHwModule_l(audio_interfaces[i]); 301 } 302 // then try to find a module supporting the requested device. 303 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 304 AudioHwDevice *audioHwDevice = mAudioHwDevs.valueAt(i); 305 audio_hw_device_t *dev = audioHwDevice->hwDevice(); 306 if ((dev->get_supported_devices != NULL) && 307 (dev->get_supported_devices(dev) & devices) == devices) 308 return audioHwDevice; 309 } 310 } else { 311 // check a match for the requested module handle 312 AudioHwDevice *audioHwDevice = mAudioHwDevs.valueFor(module); 313 if (audioHwDevice != NULL) { 314 return audioHwDevice; 315 } 316 } 317 318 return NULL; 319 } 320 321 void AudioFlinger::dumpClients(int fd, const Vector<String16>& args __unused) 322 { 323 const size_t SIZE = 256; 324 char buffer[SIZE]; 325 String8 result; 326 327 result.append("Clients:\n"); 328 for (size_t i = 0; i < mClients.size(); ++i) { 329 sp<Client> client = mClients.valueAt(i).promote(); 330 if (client != 0) { 331 snprintf(buffer, SIZE, " pid: %d\n", client->pid()); 332 result.append(buffer); 333 } 334 } 335 336 result.append("Notification Clients:\n"); 337 for (size_t i = 0; i < mNotificationClients.size(); ++i) { 338 snprintf(buffer, SIZE, " pid: %d\n", mNotificationClients.keyAt(i)); 339 result.append(buffer); 340 } 341 342 result.append("Global session refs:\n"); 343 result.append(" session pid count\n"); 344 for (size_t i = 0; i < mAudioSessionRefs.size(); i++) { 345 AudioSessionRef *r = mAudioSessionRefs[i]; 346 snprintf(buffer, SIZE, " %7d %5d %5d\n", r->mSessionid, r->mPid, r->mCnt); 347 result.append(buffer); 348 } 349 write(fd, result.string(), result.size()); 350 } 351 352 353 void AudioFlinger::dumpInternals(int fd, const Vector<String16>& args __unused) 354 { 355 const size_t SIZE = 256; 356 char buffer[SIZE]; 357 String8 result; 358 hardware_call_state hardwareStatus = mHardwareStatus; 359 360 snprintf(buffer, SIZE, "Hardware status: %d\n" 361 "Standby Time mSec: %u\n", 362 hardwareStatus, 363 (uint32_t)(mStandbyTimeInNsecs / 1000000)); 364 result.append(buffer); 365 write(fd, result.string(), result.size()); 366 } 367 368 void AudioFlinger::dumpPermissionDenial(int fd, const Vector<String16>& args __unused) 369 { 370 const size_t SIZE = 256; 371 char buffer[SIZE]; 372 String8 result; 373 snprintf(buffer, SIZE, "Permission Denial: " 374 "can't dump AudioFlinger from pid=%d, uid=%d\n", 375 IPCThreadState::self()->getCallingPid(), 376 IPCThreadState::self()->getCallingUid()); 377 result.append(buffer); 378 write(fd, result.string(), result.size()); 379 } 380 381 bool AudioFlinger::dumpTryLock(Mutex& mutex) 382 { 383 bool locked = false; 384 for (int i = 0; i < kDumpLockRetries; ++i) { 385 if (mutex.tryLock() == NO_ERROR) { 386 locked = true; 387 break; 388 } 389 usleep(kDumpLockSleepUs); 390 } 391 return locked; 392 } 393 394 status_t AudioFlinger::dump(int fd, const Vector<String16>& args) 395 { 396 if (!dumpAllowed()) { 397 dumpPermissionDenial(fd, args); 398 } else { 399 // get state of hardware lock 400 bool hardwareLocked = dumpTryLock(mHardwareLock); 401 if (!hardwareLocked) { 402 String8 result(kHardwareLockedString); 403 write(fd, result.string(), result.size()); 404 } else { 405 mHardwareLock.unlock(); 406 } 407 408 bool locked = dumpTryLock(mLock); 409 410 // failed to lock - AudioFlinger is probably deadlocked 411 if (!locked) { 412 String8 result(kDeadlockedString); 413 write(fd, result.string(), result.size()); 414 } 415 416 bool clientLocked = dumpTryLock(mClientLock); 417 if (!clientLocked) { 418 String8 result(kClientLockedString); 419 write(fd, result.string(), result.size()); 420 } 421 422 EffectDumpEffects(fd); 423 424 dumpClients(fd, args); 425 if (clientLocked) { 426 mClientLock.unlock(); 427 } 428 429 dumpInternals(fd, args); 430 431 // dump playback threads 432 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 433 mPlaybackThreads.valueAt(i)->dump(fd, args); 434 } 435 436 // dump record threads 437 for (size_t i = 0; i < mRecordThreads.size(); i++) { 438 mRecordThreads.valueAt(i)->dump(fd, args); 439 } 440 441 // dump orphan effect chains 442 if (mOrphanEffectChains.size() != 0) { 443 write(fd, " Orphan Effect Chains\n", strlen(" Orphan Effect Chains\n")); 444 for (size_t i = 0; i < mOrphanEffectChains.size(); i++) { 445 mOrphanEffectChains.valueAt(i)->dump(fd, args); 446 } 447 } 448 // dump all hardware devs 449 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 450 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 451 dev->dump(dev, fd); 452 } 453 454 #ifdef TEE_SINK 455 // dump the serially shared record tee sink 456 if (mRecordTeeSource != 0) { 457 dumpTee(fd, mRecordTeeSource); 458 } 459 #endif 460 461 if (locked) { 462 mLock.unlock(); 463 } 464 465 // append a copy of media.log here by forwarding fd to it, but don't attempt 466 // to lookup the service if it's not running, as it will block for a second 467 if (mLogMemoryDealer != 0) { 468 sp<IBinder> binder = defaultServiceManager()->getService(String16("media.log")); 469 if (binder != 0) { 470 dprintf(fd, "\nmedia.log:\n"); 471 Vector<String16> args; 472 binder->dump(fd, args); 473 } 474 } 475 476 // check for optional arguments 477 bool dumpMem = false; 478 bool unreachableMemory = false; 479 for (const auto &arg : args) { 480 if (arg == String16("-m")) { 481 dumpMem = true; 482 } else if (arg == String16("--unreachable")) { 483 unreachableMemory = true; 484 } 485 } 486 487 if (dumpMem) { 488 dprintf(fd, "\nDumping memory:\n"); 489 std::string s = dumpMemoryAddresses(100 /* limit */); 490 write(fd, s.c_str(), s.size()); 491 } 492 if (unreachableMemory) { 493 dprintf(fd, "\nDumping unreachable memory:\n"); 494 // TODO - should limit be an argument parameter? 495 std::string s = GetUnreachableMemoryString(true /* contents */, 100 /* limit */); 496 write(fd, s.c_str(), s.size()); 497 } 498 } 499 return NO_ERROR; 500 } 501 502 sp<AudioFlinger::Client> AudioFlinger::registerPid(pid_t pid) 503 { 504 Mutex::Autolock _cl(mClientLock); 505 // If pid is already in the mClients wp<> map, then use that entry 506 // (for which promote() is always != 0), otherwise create a new entry and Client. 507 sp<Client> client = mClients.valueFor(pid).promote(); 508 if (client == 0) { 509 client = new Client(this, pid); 510 mClients.add(pid, client); 511 } 512 513 return client; 514 } 515 516 sp<NBLog::Writer> AudioFlinger::newWriter_l(size_t size, const char *name) 517 { 518 // If there is no memory allocated for logs, return a dummy writer that does nothing 519 if (mLogMemoryDealer == 0) { 520 return new NBLog::Writer(); 521 } 522 sp<IBinder> binder = defaultServiceManager()->getService(String16("media.log")); 523 // Similarly if we can't contact the media.log service, also return a dummy writer 524 if (binder == 0) { 525 return new NBLog::Writer(); 526 } 527 sp<IMediaLogService> mediaLogService(interface_cast<IMediaLogService>(binder)); 528 sp<IMemory> shared = mLogMemoryDealer->allocate(NBLog::Timeline::sharedSize(size)); 529 // If allocation fails, consult the vector of previously unregistered writers 530 // and garbage-collect one or more them until an allocation succeeds 531 if (shared == 0) { 532 Mutex::Autolock _l(mUnregisteredWritersLock); 533 for (size_t count = mUnregisteredWriters.size(); count > 0; count--) { 534 { 535 // Pick the oldest stale writer to garbage-collect 536 sp<IMemory> iMemory(mUnregisteredWriters[0]->getIMemory()); 537 mUnregisteredWriters.removeAt(0); 538 mediaLogService->unregisterWriter(iMemory); 539 // Now the media.log remote reference to IMemory is gone. When our last local 540 // reference to IMemory also drops to zero at end of this block, 541 // the IMemory destructor will deallocate the region from mLogMemoryDealer. 542 } 543 // Re-attempt the allocation 544 shared = mLogMemoryDealer->allocate(NBLog::Timeline::sharedSize(size)); 545 if (shared != 0) { 546 goto success; 547 } 548 } 549 // Even after garbage-collecting all old writers, there is still not enough memory, 550 // so return a dummy writer 551 return new NBLog::Writer(); 552 } 553 success: 554 mediaLogService->registerWriter(shared, size, name); 555 return new NBLog::Writer(size, shared); 556 } 557 558 void AudioFlinger::unregisterWriter(const sp<NBLog::Writer>& writer) 559 { 560 if (writer == 0) { 561 return; 562 } 563 sp<IMemory> iMemory(writer->getIMemory()); 564 if (iMemory == 0) { 565 return; 566 } 567 // Rather than removing the writer immediately, append it to a queue of old writers to 568 // be garbage-collected later. This allows us to continue to view old logs for a while. 569 Mutex::Autolock _l(mUnregisteredWritersLock); 570 mUnregisteredWriters.push(writer); 571 } 572 573 // IAudioFlinger interface 574 575 576 sp<IAudioTrack> AudioFlinger::createTrack( 577 audio_stream_type_t streamType, 578 uint32_t sampleRate, 579 audio_format_t format, 580 audio_channel_mask_t channelMask, 581 size_t *frameCount, 582 audio_output_flags_t *flags, 583 const sp<IMemory>& sharedBuffer, 584 audio_io_handle_t output, 585 pid_t pid, 586 pid_t tid, 587 audio_session_t *sessionId, 588 int clientUid, 589 status_t *status) 590 { 591 sp<PlaybackThread::Track> track; 592 sp<TrackHandle> trackHandle; 593 sp<Client> client; 594 status_t lStatus; 595 audio_session_t lSessionId; 596 597 const uid_t callingUid = IPCThreadState::self()->getCallingUid(); 598 if (pid == -1 || !isTrustedCallingUid(callingUid)) { 599 const pid_t callingPid = IPCThreadState::self()->getCallingPid(); 600 ALOGW_IF(pid != -1 && pid != callingPid, 601 "%s uid %d pid %d tried to pass itself off as pid %d", 602 __func__, callingUid, callingPid, pid); 603 pid = callingPid; 604 } 605 606 // client AudioTrack::set already implements AUDIO_STREAM_DEFAULT => AUDIO_STREAM_MUSIC, 607 // but if someone uses binder directly they could bypass that and cause us to crash 608 if (uint32_t(streamType) >= AUDIO_STREAM_CNT) { 609 ALOGE("createTrack() invalid stream type %d", streamType); 610 lStatus = BAD_VALUE; 611 goto Exit; 612 } 613 614 // further sample rate checks are performed by createTrack_l() depending on the thread type 615 if (sampleRate == 0) { 616 ALOGE("createTrack() invalid sample rate %u", sampleRate); 617 lStatus = BAD_VALUE; 618 goto Exit; 619 } 620 621 // further channel mask checks are performed by createTrack_l() depending on the thread type 622 if (!audio_is_output_channel(channelMask)) { 623 ALOGE("createTrack() invalid channel mask %#x", channelMask); 624 lStatus = BAD_VALUE; 625 goto Exit; 626 } 627 628 // further format checks are performed by createTrack_l() depending on the thread type 629 if (!audio_is_valid_format(format)) { 630 ALOGE("createTrack() invalid format %#x", format); 631 lStatus = BAD_VALUE; 632 goto Exit; 633 } 634 635 if (sharedBuffer != 0 && sharedBuffer->pointer() == NULL) { 636 ALOGE("createTrack() sharedBuffer is non-0 but has NULL pointer()"); 637 lStatus = BAD_VALUE; 638 goto Exit; 639 } 640 641 { 642 Mutex::Autolock _l(mLock); 643 PlaybackThread *thread = checkPlaybackThread_l(output); 644 if (thread == NULL) { 645 ALOGE("no playback thread found for output handle %d", output); 646 lStatus = BAD_VALUE; 647 goto Exit; 648 } 649 650 client = registerPid(pid); 651 652 PlaybackThread *effectThread = NULL; 653 if (sessionId != NULL && *sessionId != AUDIO_SESSION_ALLOCATE) { 654 if (audio_unique_id_get_use(*sessionId) != AUDIO_UNIQUE_ID_USE_SESSION) { 655 ALOGE("createTrack() invalid session ID %d", *sessionId); 656 lStatus = BAD_VALUE; 657 goto Exit; 658 } 659 lSessionId = *sessionId; 660 // check if an effect chain with the same session ID is present on another 661 // output thread and move it here. 662 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 663 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 664 if (mPlaybackThreads.keyAt(i) != output) { 665 uint32_t sessions = t->hasAudioSession(lSessionId); 666 if (sessions & ThreadBase::EFFECT_SESSION) { 667 effectThread = t.get(); 668 break; 669 } 670 } 671 } 672 } else { 673 // if no audio session id is provided, create one here 674 lSessionId = (audio_session_t) nextUniqueId(AUDIO_UNIQUE_ID_USE_SESSION); 675 if (sessionId != NULL) { 676 *sessionId = lSessionId; 677 } 678 } 679 ALOGV("createTrack() lSessionId: %d", lSessionId); 680 681 track = thread->createTrack_l(client, streamType, sampleRate, format, 682 channelMask, frameCount, sharedBuffer, lSessionId, flags, tid, clientUid, &lStatus); 683 LOG_ALWAYS_FATAL_IF((lStatus == NO_ERROR) && (track == 0)); 684 // we don't abort yet if lStatus != NO_ERROR; there is still work to be done regardless 685 686 // move effect chain to this output thread if an effect on same session was waiting 687 // for a track to be created 688 if (lStatus == NO_ERROR && effectThread != NULL) { 689 // no risk of deadlock because AudioFlinger::mLock is held 690 Mutex::Autolock _dl(thread->mLock); 691 Mutex::Autolock _sl(effectThread->mLock); 692 moveEffectChain_l(lSessionId, effectThread, thread, true); 693 } 694 695 // Look for sync events awaiting for a session to be used. 696 for (size_t i = 0; i < mPendingSyncEvents.size(); i++) { 697 if (mPendingSyncEvents[i]->triggerSession() == lSessionId) { 698 if (thread->isValidSyncEvent(mPendingSyncEvents[i])) { 699 if (lStatus == NO_ERROR) { 700 (void) track->setSyncEvent(mPendingSyncEvents[i]); 701 } else { 702 mPendingSyncEvents[i]->cancel(); 703 } 704 mPendingSyncEvents.removeAt(i); 705 i--; 706 } 707 } 708 } 709 710 setAudioHwSyncForSession_l(thread, lSessionId); 711 } 712 713 if (lStatus != NO_ERROR) { 714 // remove local strong reference to Client before deleting the Track so that the 715 // Client destructor is called by the TrackBase destructor with mClientLock held 716 // Don't hold mClientLock when releasing the reference on the track as the 717 // destructor will acquire it. 718 { 719 Mutex::Autolock _cl(mClientLock); 720 client.clear(); 721 } 722 track.clear(); 723 goto Exit; 724 } 725 726 // return handle to client 727 trackHandle = new TrackHandle(track); 728 729 Exit: 730 *status = lStatus; 731 return trackHandle; 732 } 733 734 uint32_t AudioFlinger::sampleRate(audio_io_handle_t ioHandle) const 735 { 736 Mutex::Autolock _l(mLock); 737 ThreadBase *thread = checkThread_l(ioHandle); 738 if (thread == NULL) { 739 ALOGW("sampleRate() unknown thread %d", ioHandle); 740 return 0; 741 } 742 return thread->sampleRate(); 743 } 744 745 audio_format_t AudioFlinger::format(audio_io_handle_t output) const 746 { 747 Mutex::Autolock _l(mLock); 748 PlaybackThread *thread = checkPlaybackThread_l(output); 749 if (thread == NULL) { 750 ALOGW("format() unknown thread %d", output); 751 return AUDIO_FORMAT_INVALID; 752 } 753 return thread->format(); 754 } 755 756 size_t AudioFlinger::frameCount(audio_io_handle_t ioHandle) const 757 { 758 Mutex::Autolock _l(mLock); 759 ThreadBase *thread = checkThread_l(ioHandle); 760 if (thread == NULL) { 761 ALOGW("frameCount() unknown thread %d", ioHandle); 762 return 0; 763 } 764 // FIXME currently returns the normal mixer's frame count to avoid confusing legacy callers; 765 // should examine all callers and fix them to handle smaller counts 766 return thread->frameCount(); 767 } 768 769 size_t AudioFlinger::frameCountHAL(audio_io_handle_t ioHandle) const 770 { 771 Mutex::Autolock _l(mLock); 772 ThreadBase *thread = checkThread_l(ioHandle); 773 if (thread == NULL) { 774 ALOGW("frameCountHAL() unknown thread %d", ioHandle); 775 return 0; 776 } 777 return thread->frameCountHAL(); 778 } 779 780 uint32_t AudioFlinger::latency(audio_io_handle_t output) const 781 { 782 Mutex::Autolock _l(mLock); 783 PlaybackThread *thread = checkPlaybackThread_l(output); 784 if (thread == NULL) { 785 ALOGW("latency(): no playback thread found for output handle %d", output); 786 return 0; 787 } 788 return thread->latency(); 789 } 790 791 status_t AudioFlinger::setMasterVolume(float value) 792 { 793 status_t ret = initCheck(); 794 if (ret != NO_ERROR) { 795 return ret; 796 } 797 798 // check calling permissions 799 if (!settingsAllowed()) { 800 return PERMISSION_DENIED; 801 } 802 803 Mutex::Autolock _l(mLock); 804 mMasterVolume = value; 805 806 // Set master volume in the HALs which support it. 807 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 808 AutoMutex lock(mHardwareLock); 809 AudioHwDevice *dev = mAudioHwDevs.valueAt(i); 810 811 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 812 if (dev->canSetMasterVolume()) { 813 dev->hwDevice()->set_master_volume(dev->hwDevice(), value); 814 } 815 mHardwareStatus = AUDIO_HW_IDLE; 816 } 817 818 // Now set the master volume in each playback thread. Playback threads 819 // assigned to HALs which do not have master volume support will apply 820 // master volume during the mix operation. Threads with HALs which do 821 // support master volume will simply ignore the setting. 822 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 823 if (mPlaybackThreads.valueAt(i)->isDuplicating()) { 824 continue; 825 } 826 mPlaybackThreads.valueAt(i)->setMasterVolume(value); 827 } 828 829 return NO_ERROR; 830 } 831 832 status_t AudioFlinger::setMode(audio_mode_t mode) 833 { 834 status_t ret = initCheck(); 835 if (ret != NO_ERROR) { 836 return ret; 837 } 838 839 // check calling permissions 840 if (!settingsAllowed()) { 841 return PERMISSION_DENIED; 842 } 843 if (uint32_t(mode) >= AUDIO_MODE_CNT) { 844 ALOGW("Illegal value: setMode(%d)", mode); 845 return BAD_VALUE; 846 } 847 848 { // scope for the lock 849 AutoMutex lock(mHardwareLock); 850 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 851 mHardwareStatus = AUDIO_HW_SET_MODE; 852 ret = dev->set_mode(dev, mode); 853 mHardwareStatus = AUDIO_HW_IDLE; 854 } 855 856 if (NO_ERROR == ret) { 857 Mutex::Autolock _l(mLock); 858 mMode = mode; 859 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 860 mPlaybackThreads.valueAt(i)->setMode(mode); 861 } 862 863 return ret; 864 } 865 866 status_t AudioFlinger::setMicMute(bool state) 867 { 868 status_t ret = initCheck(); 869 if (ret != NO_ERROR) { 870 return ret; 871 } 872 873 // check calling permissions 874 if (!settingsAllowed()) { 875 return PERMISSION_DENIED; 876 } 877 878 AutoMutex lock(mHardwareLock); 879 mHardwareStatus = AUDIO_HW_SET_MIC_MUTE; 880 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 881 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 882 status_t result = dev->set_mic_mute(dev, state); 883 if (result != NO_ERROR) { 884 ret = result; 885 } 886 } 887 mHardwareStatus = AUDIO_HW_IDLE; 888 return ret; 889 } 890 891 bool AudioFlinger::getMicMute() const 892 { 893 status_t ret = initCheck(); 894 if (ret != NO_ERROR) { 895 return false; 896 } 897 bool mute = true; 898 bool state = AUDIO_MODE_INVALID; 899 AutoMutex lock(mHardwareLock); 900 mHardwareStatus = AUDIO_HW_GET_MIC_MUTE; 901 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 902 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 903 status_t result = dev->get_mic_mute(dev, &state); 904 if (result == NO_ERROR) { 905 mute = mute && state; 906 } 907 } 908 mHardwareStatus = AUDIO_HW_IDLE; 909 910 return mute; 911 } 912 913 status_t AudioFlinger::setMasterMute(bool muted) 914 { 915 status_t ret = initCheck(); 916 if (ret != NO_ERROR) { 917 return ret; 918 } 919 920 // check calling permissions 921 if (!settingsAllowed()) { 922 return PERMISSION_DENIED; 923 } 924 925 Mutex::Autolock _l(mLock); 926 mMasterMute = muted; 927 928 // Set master mute in the HALs which support it. 929 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 930 AutoMutex lock(mHardwareLock); 931 AudioHwDevice *dev = mAudioHwDevs.valueAt(i); 932 933 mHardwareStatus = AUDIO_HW_SET_MASTER_MUTE; 934 if (dev->canSetMasterMute()) { 935 dev->hwDevice()->set_master_mute(dev->hwDevice(), muted); 936 } 937 mHardwareStatus = AUDIO_HW_IDLE; 938 } 939 940 // Now set the master mute in each playback thread. Playback threads 941 // assigned to HALs which do not have master mute support will apply master 942 // mute during the mix operation. Threads with HALs which do support master 943 // mute will simply ignore the setting. 944 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 945 if (mPlaybackThreads.valueAt(i)->isDuplicating()) { 946 continue; 947 } 948 mPlaybackThreads.valueAt(i)->setMasterMute(muted); 949 } 950 951 return NO_ERROR; 952 } 953 954 float AudioFlinger::masterVolume() const 955 { 956 Mutex::Autolock _l(mLock); 957 return masterVolume_l(); 958 } 959 960 bool AudioFlinger::masterMute() const 961 { 962 Mutex::Autolock _l(mLock); 963 return masterMute_l(); 964 } 965 966 float AudioFlinger::masterVolume_l() const 967 { 968 return mMasterVolume; 969 } 970 971 bool AudioFlinger::masterMute_l() const 972 { 973 return mMasterMute; 974 } 975 976 status_t AudioFlinger::checkStreamType(audio_stream_type_t stream) const 977 { 978 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 979 ALOGW("setStreamVolume() invalid stream %d", stream); 980 return BAD_VALUE; 981 } 982 pid_t caller = IPCThreadState::self()->getCallingPid(); 983 if (uint32_t(stream) >= AUDIO_STREAM_PUBLIC_CNT && caller != getpid_cached) { 984 ALOGW("setStreamVolume() pid %d cannot use internal stream type %d", caller, stream); 985 return PERMISSION_DENIED; 986 } 987 988 return NO_ERROR; 989 } 990 991 status_t AudioFlinger::setStreamVolume(audio_stream_type_t stream, float value, 992 audio_io_handle_t output) 993 { 994 // check calling permissions 995 if (!settingsAllowed()) { 996 return PERMISSION_DENIED; 997 } 998 999 status_t status = checkStreamType(stream); 1000 if (status != NO_ERROR) { 1001 return status; 1002 } 1003 ALOG_ASSERT(stream != AUDIO_STREAM_PATCH, "attempt to change AUDIO_STREAM_PATCH volume"); 1004 1005 AutoMutex lock(mLock); 1006 PlaybackThread *thread = NULL; 1007 if (output != AUDIO_IO_HANDLE_NONE) { 1008 thread = checkPlaybackThread_l(output); 1009 if (thread == NULL) { 1010 return BAD_VALUE; 1011 } 1012 } 1013 1014 mStreamTypes[stream].volume = value; 1015 1016 if (thread == NULL) { 1017 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1018 mPlaybackThreads.valueAt(i)->setStreamVolume(stream, value); 1019 } 1020 } else { 1021 thread->setStreamVolume(stream, value); 1022 } 1023 1024 return NO_ERROR; 1025 } 1026 1027 status_t AudioFlinger::setStreamMute(audio_stream_type_t stream, bool muted) 1028 { 1029 // check calling permissions 1030 if (!settingsAllowed()) { 1031 return PERMISSION_DENIED; 1032 } 1033 1034 status_t status = checkStreamType(stream); 1035 if (status != NO_ERROR) { 1036 return status; 1037 } 1038 ALOG_ASSERT(stream != AUDIO_STREAM_PATCH, "attempt to mute AUDIO_STREAM_PATCH"); 1039 1040 if (uint32_t(stream) == AUDIO_STREAM_ENFORCED_AUDIBLE) { 1041 ALOGE("setStreamMute() invalid stream %d", stream); 1042 return BAD_VALUE; 1043 } 1044 1045 AutoMutex lock(mLock); 1046 mStreamTypes[stream].mute = muted; 1047 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 1048 mPlaybackThreads.valueAt(i)->setStreamMute(stream, muted); 1049 1050 return NO_ERROR; 1051 } 1052 1053 float AudioFlinger::streamVolume(audio_stream_type_t stream, audio_io_handle_t output) const 1054 { 1055 status_t status = checkStreamType(stream); 1056 if (status != NO_ERROR) { 1057 return 0.0f; 1058 } 1059 1060 AutoMutex lock(mLock); 1061 float volume; 1062 if (output != AUDIO_IO_HANDLE_NONE) { 1063 PlaybackThread *thread = checkPlaybackThread_l(output); 1064 if (thread == NULL) { 1065 return 0.0f; 1066 } 1067 volume = thread->streamVolume(stream); 1068 } else { 1069 volume = streamVolume_l(stream); 1070 } 1071 1072 return volume; 1073 } 1074 1075 bool AudioFlinger::streamMute(audio_stream_type_t stream) const 1076 { 1077 status_t status = checkStreamType(stream); 1078 if (status != NO_ERROR) { 1079 return true; 1080 } 1081 1082 AutoMutex lock(mLock); 1083 return streamMute_l(stream); 1084 } 1085 1086 1087 void AudioFlinger::broacastParametersToRecordThreads_l(const String8& keyValuePairs) 1088 { 1089 for (size_t i = 0; i < mRecordThreads.size(); i++) { 1090 mRecordThreads.valueAt(i)->setParameters(keyValuePairs); 1091 } 1092 } 1093 1094 status_t AudioFlinger::setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs) 1095 { 1096 ALOGV("setParameters(): io %d, keyvalue %s, calling pid %d", 1097 ioHandle, keyValuePairs.string(), IPCThreadState::self()->getCallingPid()); 1098 1099 // check calling permissions 1100 if (!settingsAllowed()) { 1101 return PERMISSION_DENIED; 1102 } 1103 1104 // AUDIO_IO_HANDLE_NONE means the parameters are global to the audio hardware interface 1105 if (ioHandle == AUDIO_IO_HANDLE_NONE) { 1106 Mutex::Autolock _l(mLock); 1107 // result will remain NO_INIT if no audio device is present 1108 status_t final_result = NO_INIT; 1109 { 1110 AutoMutex lock(mHardwareLock); 1111 mHardwareStatus = AUDIO_HW_SET_PARAMETER; 1112 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 1113 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 1114 status_t result = dev->set_parameters(dev, keyValuePairs.string()); 1115 // return success if at least one audio device accepts the parameters as not all 1116 // HALs are requested to support all parameters. If no audio device supports the 1117 // requested parameters, the last error is reported. 1118 if (final_result != NO_ERROR) { 1119 final_result = result; 1120 } 1121 } 1122 mHardwareStatus = AUDIO_HW_IDLE; 1123 } 1124 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 1125 AudioParameter param = AudioParameter(keyValuePairs); 1126 String8 value; 1127 if (param.get(String8(AUDIO_PARAMETER_KEY_BT_NREC), value) == NO_ERROR) { 1128 bool btNrecIsOff = (value == AUDIO_PARAMETER_VALUE_OFF); 1129 if (mBtNrecIsOff != btNrecIsOff) { 1130 for (size_t i = 0; i < mRecordThreads.size(); i++) { 1131 sp<RecordThread> thread = mRecordThreads.valueAt(i); 1132 audio_devices_t device = thread->inDevice(); 1133 bool suspend = audio_is_bluetooth_sco_device(device) && btNrecIsOff; 1134 // collect all of the thread's session IDs 1135 KeyedVector<audio_session_t, bool> ids = thread->sessionIds(); 1136 // suspend effects associated with those session IDs 1137 for (size_t j = 0; j < ids.size(); ++j) { 1138 audio_session_t sessionId = ids.keyAt(j); 1139 thread->setEffectSuspended(FX_IID_AEC, 1140 suspend, 1141 sessionId); 1142 thread->setEffectSuspended(FX_IID_NS, 1143 suspend, 1144 sessionId); 1145 } 1146 } 1147 mBtNrecIsOff = btNrecIsOff; 1148 } 1149 } 1150 String8 screenState; 1151 if (param.get(String8(AudioParameter::keyScreenState), screenState) == NO_ERROR) { 1152 bool isOff = screenState == "off"; 1153 if (isOff != (AudioFlinger::mScreenState & 1)) { 1154 AudioFlinger::mScreenState = ((AudioFlinger::mScreenState & ~1) + 2) | isOff; 1155 } 1156 } 1157 return final_result; 1158 } 1159 1160 // hold a strong ref on thread in case closeOutput() or closeInput() is called 1161 // and the thread is exited once the lock is released 1162 sp<ThreadBase> thread; 1163 { 1164 Mutex::Autolock _l(mLock); 1165 thread = checkPlaybackThread_l(ioHandle); 1166 if (thread == 0) { 1167 thread = checkRecordThread_l(ioHandle); 1168 } else if (thread == primaryPlaybackThread_l()) { 1169 // indicate output device change to all input threads for pre processing 1170 AudioParameter param = AudioParameter(keyValuePairs); 1171 int value; 1172 if ((param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) && 1173 (value != 0)) { 1174 broacastParametersToRecordThreads_l(keyValuePairs); 1175 } 1176 } 1177 } 1178 if (thread != 0) { 1179 return thread->setParameters(keyValuePairs); 1180 } 1181 return BAD_VALUE; 1182 } 1183 1184 String8 AudioFlinger::getParameters(audio_io_handle_t ioHandle, const String8& keys) const 1185 { 1186 ALOGVV("getParameters() io %d, keys %s, calling pid %d", 1187 ioHandle, keys.string(), IPCThreadState::self()->getCallingPid()); 1188 1189 Mutex::Autolock _l(mLock); 1190 1191 if (ioHandle == AUDIO_IO_HANDLE_NONE) { 1192 String8 out_s8; 1193 1194 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 1195 char *s; 1196 { 1197 AutoMutex lock(mHardwareLock); 1198 mHardwareStatus = AUDIO_HW_GET_PARAMETER; 1199 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 1200 s = dev->get_parameters(dev, keys.string()); 1201 mHardwareStatus = AUDIO_HW_IDLE; 1202 } 1203 out_s8 += String8(s ? s : ""); 1204 free(s); 1205 } 1206 return out_s8; 1207 } 1208 1209 PlaybackThread *playbackThread = checkPlaybackThread_l(ioHandle); 1210 if (playbackThread != NULL) { 1211 return playbackThread->getParameters(keys); 1212 } 1213 RecordThread *recordThread = checkRecordThread_l(ioHandle); 1214 if (recordThread != NULL) { 1215 return recordThread->getParameters(keys); 1216 } 1217 return String8(""); 1218 } 1219 1220 size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, audio_format_t format, 1221 audio_channel_mask_t channelMask) const 1222 { 1223 status_t ret = initCheck(); 1224 if (ret != NO_ERROR) { 1225 return 0; 1226 } 1227 if ((sampleRate == 0) || 1228 !audio_is_valid_format(format) || !audio_has_proportional_frames(format) || 1229 !audio_is_input_channel(channelMask)) { 1230 return 0; 1231 } 1232 1233 AutoMutex lock(mHardwareLock); 1234 mHardwareStatus = AUDIO_HW_GET_INPUT_BUFFER_SIZE; 1235 audio_config_t config, proposed; 1236 memset(&proposed, 0, sizeof(proposed)); 1237 proposed.sample_rate = sampleRate; 1238 proposed.channel_mask = channelMask; 1239 proposed.format = format; 1240 1241 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 1242 size_t frames; 1243 for (;;) { 1244 // Note: config is currently a const parameter for get_input_buffer_size() 1245 // but we use a copy from proposed in case config changes from the call. 1246 config = proposed; 1247 frames = dev->get_input_buffer_size(dev, &config); 1248 if (frames != 0) { 1249 break; // hal success, config is the result 1250 } 1251 // change one parameter of the configuration each iteration to a more "common" value 1252 // to see if the device will support it. 1253 if (proposed.format != AUDIO_FORMAT_PCM_16_BIT) { 1254 proposed.format = AUDIO_FORMAT_PCM_16_BIT; 1255 } else if (proposed.sample_rate != 44100) { // 44.1 is claimed as must in CDD as well as 1256 proposed.sample_rate = 44100; // legacy AudioRecord.java. TODO: Query hw? 1257 } else { 1258 ALOGW("getInputBufferSize failed with minimum buffer size sampleRate %u, " 1259 "format %#x, channelMask 0x%X", 1260 sampleRate, format, channelMask); 1261 break; // retries failed, break out of loop with frames == 0. 1262 } 1263 } 1264 mHardwareStatus = AUDIO_HW_IDLE; 1265 if (frames > 0 && config.sample_rate != sampleRate) { 1266 frames = destinationFramesPossible(frames, sampleRate, config.sample_rate); 1267 } 1268 return frames; // may be converted to bytes at the Java level. 1269 } 1270 1271 uint32_t AudioFlinger::getInputFramesLost(audio_io_handle_t ioHandle) const 1272 { 1273 Mutex::Autolock _l(mLock); 1274 1275 RecordThread *recordThread = checkRecordThread_l(ioHandle); 1276 if (recordThread != NULL) { 1277 return recordThread->getInputFramesLost(); 1278 } 1279 return 0; 1280 } 1281 1282 status_t AudioFlinger::setVoiceVolume(float value) 1283 { 1284 status_t ret = initCheck(); 1285 if (ret != NO_ERROR) { 1286 return ret; 1287 } 1288 1289 // check calling permissions 1290 if (!settingsAllowed()) { 1291 return PERMISSION_DENIED; 1292 } 1293 1294 AutoMutex lock(mHardwareLock); 1295 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 1296 mHardwareStatus = AUDIO_HW_SET_VOICE_VOLUME; 1297 ret = dev->set_voice_volume(dev, value); 1298 mHardwareStatus = AUDIO_HW_IDLE; 1299 1300 return ret; 1301 } 1302 1303 status_t AudioFlinger::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, 1304 audio_io_handle_t output) const 1305 { 1306 Mutex::Autolock _l(mLock); 1307 1308 PlaybackThread *playbackThread = checkPlaybackThread_l(output); 1309 if (playbackThread != NULL) { 1310 return playbackThread->getRenderPosition(halFrames, dspFrames); 1311 } 1312 1313 return BAD_VALUE; 1314 } 1315 1316 void AudioFlinger::registerClient(const sp<IAudioFlingerClient>& client) 1317 { 1318 Mutex::Autolock _l(mLock); 1319 if (client == 0) { 1320 return; 1321 } 1322 pid_t pid = IPCThreadState::self()->getCallingPid(); 1323 { 1324 Mutex::Autolock _cl(mClientLock); 1325 if (mNotificationClients.indexOfKey(pid) < 0) { 1326 sp<NotificationClient> notificationClient = new NotificationClient(this, 1327 client, 1328 pid); 1329 ALOGV("registerClient() client %p, pid %d", notificationClient.get(), pid); 1330 1331 mNotificationClients.add(pid, notificationClient); 1332 1333 sp<IBinder> binder = IInterface::asBinder(client); 1334 binder->linkToDeath(notificationClient); 1335 } 1336 } 1337 1338 // mClientLock should not be held here because ThreadBase::sendIoConfigEvent() will lock the 1339 // ThreadBase mutex and the locking order is ThreadBase::mLock then AudioFlinger::mClientLock. 1340 // the config change is always sent from playback or record threads to avoid deadlock 1341 // with AudioSystem::gLock 1342 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1343 mPlaybackThreads.valueAt(i)->sendIoConfigEvent(AUDIO_OUTPUT_OPENED, pid); 1344 } 1345 1346 for (size_t i = 0; i < mRecordThreads.size(); i++) { 1347 mRecordThreads.valueAt(i)->sendIoConfigEvent(AUDIO_INPUT_OPENED, pid); 1348 } 1349 } 1350 1351 void AudioFlinger::removeNotificationClient(pid_t pid) 1352 { 1353 Mutex::Autolock _l(mLock); 1354 { 1355 Mutex::Autolock _cl(mClientLock); 1356 mNotificationClients.removeItem(pid); 1357 } 1358 1359 ALOGV("%d died, releasing its sessions", pid); 1360 size_t num = mAudioSessionRefs.size(); 1361 bool removed = false; 1362 for (size_t i = 0; i< num; ) { 1363 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 1364 ALOGV(" pid %d @ %zu", ref->mPid, i); 1365 if (ref->mPid == pid) { 1366 ALOGV(" removing entry for pid %d session %d", pid, ref->mSessionid); 1367 mAudioSessionRefs.removeAt(i); 1368 delete ref; 1369 removed = true; 1370 num--; 1371 } else { 1372 i++; 1373 } 1374 } 1375 if (removed) { 1376 purgeStaleEffects_l(); 1377 } 1378 } 1379 1380 void AudioFlinger::ioConfigChanged(audio_io_config_event event, 1381 const sp<AudioIoDescriptor>& ioDesc, 1382 pid_t pid) 1383 { 1384 Mutex::Autolock _l(mClientLock); 1385 size_t size = mNotificationClients.size(); 1386 for (size_t i = 0; i < size; i++) { 1387 if ((pid == 0) || (mNotificationClients.keyAt(i) == pid)) { 1388 mNotificationClients.valueAt(i)->audioFlingerClient()->ioConfigChanged(event, ioDesc); 1389 } 1390 } 1391 } 1392 1393 // removeClient_l() must be called with AudioFlinger::mClientLock held 1394 void AudioFlinger::removeClient_l(pid_t pid) 1395 { 1396 ALOGV("removeClient_l() pid %d, calling pid %d", pid, 1397 IPCThreadState::self()->getCallingPid()); 1398 mClients.removeItem(pid); 1399 } 1400 1401 // getEffectThread_l() must be called with AudioFlinger::mLock held 1402 sp<AudioFlinger::PlaybackThread> AudioFlinger::getEffectThread_l(audio_session_t sessionId, 1403 int EffectId) 1404 { 1405 sp<PlaybackThread> thread; 1406 1407 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1408 if (mPlaybackThreads.valueAt(i)->getEffect(sessionId, EffectId) != 0) { 1409 ALOG_ASSERT(thread == 0); 1410 thread = mPlaybackThreads.valueAt(i); 1411 } 1412 } 1413 1414 return thread; 1415 } 1416 1417 1418 1419 // ---------------------------------------------------------------------------- 1420 1421 AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid) 1422 : RefBase(), 1423 mAudioFlinger(audioFlinger), 1424 mPid(pid) 1425 { 1426 size_t heapSize = kClientSharedHeapSizeBytes; 1427 // Increase heap size on non low ram devices to limit risk of reconnection failure for 1428 // invalidated tracks 1429 if (!audioFlinger->isLowRamDevice()) { 1430 heapSize *= kClientSharedHeapSizeMultiplier; 1431 } 1432 mMemoryDealer = new MemoryDealer(heapSize, "AudioFlinger::Client"); 1433 } 1434 1435 // Client destructor must be called with AudioFlinger::mClientLock held 1436 AudioFlinger::Client::~Client() 1437 { 1438 mAudioFlinger->removeClient_l(mPid); 1439 } 1440 1441 sp<MemoryDealer> AudioFlinger::Client::heap() const 1442 { 1443 return mMemoryDealer; 1444 } 1445 1446 // ---------------------------------------------------------------------------- 1447 1448 AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger, 1449 const sp<IAudioFlingerClient>& client, 1450 pid_t pid) 1451 : mAudioFlinger(audioFlinger), mPid(pid), mAudioFlingerClient(client) 1452 { 1453 } 1454 1455 AudioFlinger::NotificationClient::~NotificationClient() 1456 { 1457 } 1458 1459 void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who __unused) 1460 { 1461 sp<NotificationClient> keep(this); 1462 mAudioFlinger->removeNotificationClient(mPid); 1463 } 1464 1465 1466 // ---------------------------------------------------------------------------- 1467 1468 sp<IAudioRecord> AudioFlinger::openRecord( 1469 audio_io_handle_t input, 1470 uint32_t sampleRate, 1471 audio_format_t format, 1472 audio_channel_mask_t channelMask, 1473 const String16& opPackageName, 1474 size_t *frameCount, 1475 audio_input_flags_t *flags, 1476 pid_t pid, 1477 pid_t tid, 1478 int clientUid, 1479 audio_session_t *sessionId, 1480 size_t *notificationFrames, 1481 sp<IMemory>& cblk, 1482 sp<IMemory>& buffers, 1483 status_t *status) 1484 { 1485 sp<RecordThread::RecordTrack> recordTrack; 1486 sp<RecordHandle> recordHandle; 1487 sp<Client> client; 1488 status_t lStatus; 1489 audio_session_t lSessionId; 1490 1491 cblk.clear(); 1492 buffers.clear(); 1493 1494 bool updatePid = (pid == -1); 1495 const uid_t callingUid = IPCThreadState::self()->getCallingUid(); 1496 if (!isTrustedCallingUid(callingUid)) { 1497 ALOGW_IF((uid_t)clientUid != callingUid, 1498 "%s uid %d tried to pass itself off as %d", __FUNCTION__, callingUid, clientUid); 1499 clientUid = callingUid; 1500 updatePid = true; 1501 } 1502 1503 if (updatePid) { 1504 const pid_t callingPid = IPCThreadState::self()->getCallingPid(); 1505 ALOGW_IF(pid != -1 && pid != callingPid, 1506 "%s uid %d pid %d tried to pass itself off as pid %d", 1507 __func__, callingUid, callingPid, pid); 1508 pid = callingPid; 1509 } 1510 1511 // check calling permissions 1512 if (!recordingAllowed(opPackageName, tid, clientUid)) { 1513 ALOGE("openRecord() permission denied: recording not allowed"); 1514 lStatus = PERMISSION_DENIED; 1515 goto Exit; 1516 } 1517 1518 // further sample rate checks are performed by createRecordTrack_l() 1519 if (sampleRate == 0) { 1520 ALOGE("openRecord() invalid sample rate %u", sampleRate); 1521 lStatus = BAD_VALUE; 1522 goto Exit; 1523 } 1524 1525 // we don't yet support anything other than linear PCM 1526 if (!audio_is_valid_format(format) || !audio_is_linear_pcm(format)) { 1527 ALOGE("openRecord() invalid format %#x", format); 1528 lStatus = BAD_VALUE; 1529 goto Exit; 1530 } 1531 1532 // further channel mask checks are performed by createRecordTrack_l() 1533 if (!audio_is_input_channel(channelMask)) { 1534 ALOGE("openRecord() invalid channel mask %#x", channelMask); 1535 lStatus = BAD_VALUE; 1536 goto Exit; 1537 } 1538 1539 { 1540 Mutex::Autolock _l(mLock); 1541 RecordThread *thread = checkRecordThread_l(input); 1542 if (thread == NULL) { 1543 ALOGE("openRecord() checkRecordThread_l failed"); 1544 lStatus = BAD_VALUE; 1545 goto Exit; 1546 } 1547 1548 client = registerPid(pid); 1549 1550 if (sessionId != NULL && *sessionId != AUDIO_SESSION_ALLOCATE) { 1551 if (audio_unique_id_get_use(*sessionId) != AUDIO_UNIQUE_ID_USE_SESSION) { 1552 lStatus = BAD_VALUE; 1553 goto Exit; 1554 } 1555 lSessionId = *sessionId; 1556 } else { 1557 // if no audio session id is provided, create one here 1558 lSessionId = (audio_session_t) nextUniqueId(AUDIO_UNIQUE_ID_USE_SESSION); 1559 if (sessionId != NULL) { 1560 *sessionId = lSessionId; 1561 } 1562 } 1563 ALOGV("openRecord() lSessionId: %d input %d", lSessionId, input); 1564 1565 recordTrack = thread->createRecordTrack_l(client, sampleRate, format, channelMask, 1566 frameCount, lSessionId, notificationFrames, 1567 clientUid, flags, tid, &lStatus); 1568 LOG_ALWAYS_FATAL_IF((lStatus == NO_ERROR) && (recordTrack == 0)); 1569 1570 if (lStatus == NO_ERROR) { 1571 // Check if one effect chain was awaiting for an AudioRecord to be created on this 1572 // session and move it to this thread. 1573 sp<EffectChain> chain = getOrphanEffectChain_l(lSessionId); 1574 if (chain != 0) { 1575 Mutex::Autolock _l(thread->mLock); 1576 thread->addEffectChain_l(chain); 1577 } 1578 } 1579 } 1580 1581 if (lStatus != NO_ERROR) { 1582 // remove local strong reference to Client before deleting the RecordTrack so that the 1583 // Client destructor is called by the TrackBase destructor with mClientLock held 1584 // Don't hold mClientLock when releasing the reference on the track as the 1585 // destructor will acquire it. 1586 { 1587 Mutex::Autolock _cl(mClientLock); 1588 client.clear(); 1589 } 1590 recordTrack.clear(); 1591 goto Exit; 1592 } 1593 1594 cblk = recordTrack->getCblk(); 1595 buffers = recordTrack->getBuffers(); 1596 1597 // return handle to client 1598 recordHandle = new RecordHandle(recordTrack); 1599 1600 Exit: 1601 *status = lStatus; 1602 return recordHandle; 1603 } 1604 1605 1606 1607 // ---------------------------------------------------------------------------- 1608 1609 audio_module_handle_t AudioFlinger::loadHwModule(const char *name) 1610 { 1611 if (name == NULL) { 1612 return AUDIO_MODULE_HANDLE_NONE; 1613 } 1614 if (!settingsAllowed()) { 1615 return AUDIO_MODULE_HANDLE_NONE; 1616 } 1617 Mutex::Autolock _l(mLock); 1618 return loadHwModule_l(name); 1619 } 1620 1621 // loadHwModule_l() must be called with AudioFlinger::mLock held 1622 audio_module_handle_t AudioFlinger::loadHwModule_l(const char *name) 1623 { 1624 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 1625 if (strncmp(mAudioHwDevs.valueAt(i)->moduleName(), name, strlen(name)) == 0) { 1626 ALOGW("loadHwModule() module %s already loaded", name); 1627 return mAudioHwDevs.keyAt(i); 1628 } 1629 } 1630 1631 audio_hw_device_t *dev; 1632 1633 int rc = load_audio_interface(name, &dev); 1634 if (rc) { 1635 ALOGE("loadHwModule() error %d loading module %s", rc, name); 1636 return AUDIO_MODULE_HANDLE_NONE; 1637 } 1638 1639 mHardwareStatus = AUDIO_HW_INIT; 1640 rc = dev->init_check(dev); 1641 mHardwareStatus = AUDIO_HW_IDLE; 1642 if (rc) { 1643 ALOGE("loadHwModule() init check error %d for module %s", rc, name); 1644 return AUDIO_MODULE_HANDLE_NONE; 1645 } 1646 1647 // Check and cache this HAL's level of support for master mute and master 1648 // volume. If this is the first HAL opened, and it supports the get 1649 // methods, use the initial values provided by the HAL as the current 1650 // master mute and volume settings. 1651 1652 AudioHwDevice::Flags flags = static_cast<AudioHwDevice::Flags>(0); 1653 { // scope for auto-lock pattern 1654 AutoMutex lock(mHardwareLock); 1655 1656 if (0 == mAudioHwDevs.size()) { 1657 mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME; 1658 if (NULL != dev->get_master_volume) { 1659 float mv; 1660 if (OK == dev->get_master_volume(dev, &mv)) { 1661 mMasterVolume = mv; 1662 } 1663 } 1664 1665 mHardwareStatus = AUDIO_HW_GET_MASTER_MUTE; 1666 if (NULL != dev->get_master_mute) { 1667 bool mm; 1668 if (OK == dev->get_master_mute(dev, &mm)) { 1669 mMasterMute = mm; 1670 } 1671 } 1672 } 1673 1674 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 1675 if ((NULL != dev->set_master_volume) && 1676 (OK == dev->set_master_volume(dev, mMasterVolume))) { 1677 flags = static_cast<AudioHwDevice::Flags>(flags | 1678 AudioHwDevice::AHWD_CAN_SET_MASTER_VOLUME); 1679 } 1680 1681 mHardwareStatus = AUDIO_HW_SET_MASTER_MUTE; 1682 if ((NULL != dev->set_master_mute) && 1683 (OK == dev->set_master_mute(dev, mMasterMute))) { 1684 flags = static_cast<AudioHwDevice::Flags>(flags | 1685 AudioHwDevice::AHWD_CAN_SET_MASTER_MUTE); 1686 } 1687 1688 mHardwareStatus = AUDIO_HW_IDLE; 1689 } 1690 1691 audio_module_handle_t handle = (audio_module_handle_t) nextUniqueId(AUDIO_UNIQUE_ID_USE_MODULE); 1692 mAudioHwDevs.add(handle, new AudioHwDevice(handle, name, dev, flags)); 1693 1694 ALOGI("loadHwModule() Loaded %s audio interface from %s (%s) handle %d", 1695 name, dev->common.module->name, dev->common.module->id, handle); 1696 1697 return handle; 1698 1699 } 1700 1701 // ---------------------------------------------------------------------------- 1702 1703 uint32_t AudioFlinger::getPrimaryOutputSamplingRate() 1704 { 1705 Mutex::Autolock _l(mLock); 1706 PlaybackThread *thread = fastPlaybackThread_l(); 1707 return thread != NULL ? thread->sampleRate() : 0; 1708 } 1709 1710 size_t AudioFlinger::getPrimaryOutputFrameCount() 1711 { 1712 Mutex::Autolock _l(mLock); 1713 PlaybackThread *thread = fastPlaybackThread_l(); 1714 return thread != NULL ? thread->frameCountHAL() : 0; 1715 } 1716 1717 // ---------------------------------------------------------------------------- 1718 1719 status_t AudioFlinger::setLowRamDevice(bool isLowRamDevice) 1720 { 1721 uid_t uid = IPCThreadState::self()->getCallingUid(); 1722 if (uid != AID_SYSTEM) { 1723 return PERMISSION_DENIED; 1724 } 1725 Mutex::Autolock _l(mLock); 1726 if (mIsDeviceTypeKnown) { 1727 return INVALID_OPERATION; 1728 } 1729 mIsLowRamDevice = isLowRamDevice; 1730 mIsDeviceTypeKnown = true; 1731 return NO_ERROR; 1732 } 1733 1734 audio_hw_sync_t AudioFlinger::getAudioHwSyncForSession(audio_session_t sessionId) 1735 { 1736 Mutex::Autolock _l(mLock); 1737 1738 ssize_t index = mHwAvSyncIds.indexOfKey(sessionId); 1739 if (index >= 0) { 1740 ALOGV("getAudioHwSyncForSession found ID %d for session %d", 1741 mHwAvSyncIds.valueAt(index), sessionId); 1742 return mHwAvSyncIds.valueAt(index); 1743 } 1744 1745 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 1746 if (dev == NULL) { 1747 return AUDIO_HW_SYNC_INVALID; 1748 } 1749 char *reply = dev->get_parameters(dev, AUDIO_PARAMETER_HW_AV_SYNC); 1750 AudioParameter param = AudioParameter(String8(reply)); 1751 free(reply); 1752 1753 int value; 1754 if (param.getInt(String8(AUDIO_PARAMETER_HW_AV_SYNC), value) != NO_ERROR) { 1755 ALOGW("getAudioHwSyncForSession error getting sync for session %d", sessionId); 1756 return AUDIO_HW_SYNC_INVALID; 1757 } 1758 1759 // allow only one session for a given HW A/V sync ID. 1760 for (size_t i = 0; i < mHwAvSyncIds.size(); i++) { 1761 if (mHwAvSyncIds.valueAt(i) == (audio_hw_sync_t)value) { 1762 ALOGV("getAudioHwSyncForSession removing ID %d for session %d", 1763 value, mHwAvSyncIds.keyAt(i)); 1764 mHwAvSyncIds.removeItemsAt(i); 1765 break; 1766 } 1767 } 1768 1769 mHwAvSyncIds.add(sessionId, value); 1770 1771 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1772 sp<PlaybackThread> thread = mPlaybackThreads.valueAt(i); 1773 uint32_t sessions = thread->hasAudioSession(sessionId); 1774 if (sessions & ThreadBase::TRACK_SESSION) { 1775 AudioParameter param = AudioParameter(); 1776 param.addInt(String8(AUDIO_PARAMETER_STREAM_HW_AV_SYNC), value); 1777 thread->setParameters(param.toString()); 1778 break; 1779 } 1780 } 1781 1782 ALOGV("getAudioHwSyncForSession adding ID %d for session %d", value, sessionId); 1783 return (audio_hw_sync_t)value; 1784 } 1785 1786 status_t AudioFlinger::systemReady() 1787 { 1788 Mutex::Autolock _l(mLock); 1789 ALOGI("%s", __FUNCTION__); 1790 if (mSystemReady) { 1791 ALOGW("%s called twice", __FUNCTION__); 1792 return NO_ERROR; 1793 } 1794 mSystemReady = true; 1795 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1796 ThreadBase *thread = (ThreadBase *)mPlaybackThreads.valueAt(i).get(); 1797 thread->systemReady(); 1798 } 1799 for (size_t i = 0; i < mRecordThreads.size(); i++) { 1800 ThreadBase *thread = (ThreadBase *)mRecordThreads.valueAt(i).get(); 1801 thread->systemReady(); 1802 } 1803 return NO_ERROR; 1804 } 1805 1806 // setAudioHwSyncForSession_l() must be called with AudioFlinger::mLock held 1807 void AudioFlinger::setAudioHwSyncForSession_l(PlaybackThread *thread, audio_session_t sessionId) 1808 { 1809 ssize_t index = mHwAvSyncIds.indexOfKey(sessionId); 1810 if (index >= 0) { 1811 audio_hw_sync_t syncId = mHwAvSyncIds.valueAt(index); 1812 ALOGV("setAudioHwSyncForSession_l found ID %d for session %d", syncId, sessionId); 1813 AudioParameter param = AudioParameter(); 1814 param.addInt(String8(AUDIO_PARAMETER_STREAM_HW_AV_SYNC), syncId); 1815 thread->setParameters(param.toString()); 1816 } 1817 } 1818 1819 1820 // ---------------------------------------------------------------------------- 1821 1822 1823 sp<AudioFlinger::PlaybackThread> AudioFlinger::openOutput_l(audio_module_handle_t module, 1824 audio_io_handle_t *output, 1825 audio_config_t *config, 1826 audio_devices_t devices, 1827 const String8& address, 1828 audio_output_flags_t flags) 1829 { 1830 AudioHwDevice *outHwDev = findSuitableHwDev_l(module, devices); 1831 if (outHwDev == NULL) { 1832 return 0; 1833 } 1834 1835 if (*output == AUDIO_IO_HANDLE_NONE) { 1836 *output = nextUniqueId(AUDIO_UNIQUE_ID_USE_OUTPUT); 1837 } else { 1838 // Audio Policy does not currently request a specific output handle. 1839 // If this is ever needed, see openInput_l() for example code. 1840 ALOGE("openOutput_l requested output handle %d is not AUDIO_IO_HANDLE_NONE", *output); 1841 return 0; 1842 } 1843 1844 mHardwareStatus = AUDIO_HW_OUTPUT_OPEN; 1845 1846 // FOR TESTING ONLY: 1847 // This if statement allows overriding the audio policy settings 1848 // and forcing a specific format or channel mask to the HAL/Sink device for testing. 1849 if (!(flags & (AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD | AUDIO_OUTPUT_FLAG_DIRECT))) { 1850 // Check only for Normal Mixing mode 1851 if (kEnableExtendedPrecision) { 1852 // Specify format (uncomment one below to choose) 1853 //config->format = AUDIO_FORMAT_PCM_FLOAT; 1854 //config->format = AUDIO_FORMAT_PCM_24_BIT_PACKED; 1855 //config->format = AUDIO_FORMAT_PCM_32_BIT; 1856 //config->format = AUDIO_FORMAT_PCM_8_24_BIT; 1857 // ALOGV("openOutput_l() upgrading format to %#08x", config->format); 1858 } 1859 if (kEnableExtendedChannels) { 1860 // Specify channel mask (uncomment one below to choose) 1861 //config->channel_mask = audio_channel_out_mask_from_count(4); // for USB 4ch 1862 //config->channel_mask = audio_channel_mask_from_representation_and_bits( 1863 // AUDIO_CHANNEL_REPRESENTATION_INDEX, (1 << 4) - 1); // another 4ch example 1864 } 1865 } 1866 1867 AudioStreamOut *outputStream = NULL; 1868 status_t status = outHwDev->openOutputStream( 1869 &outputStream, 1870 *output, 1871 devices, 1872 flags, 1873 config, 1874 address.string()); 1875 1876 mHardwareStatus = AUDIO_HW_IDLE; 1877 1878 if (status == NO_ERROR) { 1879 1880 PlaybackThread *thread; 1881 if (flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) { 1882 thread = new OffloadThread(this, outputStream, *output, devices, mSystemReady); 1883 ALOGV("openOutput_l() created offload output: ID %d thread %p", *output, thread); 1884 } else if ((flags & AUDIO_OUTPUT_FLAG_DIRECT) 1885 || !isValidPcmSinkFormat(config->format) 1886 || !isValidPcmSinkChannelMask(config->channel_mask)) { 1887 thread = new DirectOutputThread(this, outputStream, *output, devices, mSystemReady); 1888 ALOGV("openOutput_l() created direct output: ID %d thread %p", *output, thread); 1889 } else { 1890 thread = new MixerThread(this, outputStream, *output, devices, mSystemReady); 1891 ALOGV("openOutput_l() created mixer output: ID %d thread %p", *output, thread); 1892 } 1893 mPlaybackThreads.add(*output, thread); 1894 return thread; 1895 } 1896 1897 return 0; 1898 } 1899 1900 status_t AudioFlinger::openOutput(audio_module_handle_t module, 1901 audio_io_handle_t *output, 1902 audio_config_t *config, 1903 audio_devices_t *devices, 1904 const String8& address, 1905 uint32_t *latencyMs, 1906 audio_output_flags_t flags) 1907 { 1908 ALOGI("openOutput(), module %d Device %x, SamplingRate %d, Format %#08x, Channels %x, flags %x", 1909 module, 1910 (devices != NULL) ? *devices : 0, 1911 config->sample_rate, 1912 config->format, 1913 config->channel_mask, 1914 flags); 1915 1916 if (*devices == AUDIO_DEVICE_NONE) { 1917 return BAD_VALUE; 1918 } 1919 1920 Mutex::Autolock _l(mLock); 1921 1922 sp<PlaybackThread> thread = openOutput_l(module, output, config, *devices, address, flags); 1923 if (thread != 0) { 1924 *latencyMs = thread->latency(); 1925 1926 // notify client processes of the new output creation 1927 thread->ioConfigChanged(AUDIO_OUTPUT_OPENED); 1928 1929 // the first primary output opened designates the primary hw device 1930 if ((mPrimaryHardwareDev == NULL) && (flags & AUDIO_OUTPUT_FLAG_PRIMARY)) { 1931 ALOGI("Using module %d has the primary audio interface", module); 1932 mPrimaryHardwareDev = thread->getOutput()->audioHwDev; 1933 1934 AutoMutex lock(mHardwareLock); 1935 mHardwareStatus = AUDIO_HW_SET_MODE; 1936 mPrimaryHardwareDev->hwDevice()->set_mode(mPrimaryHardwareDev->hwDevice(), mMode); 1937 mHardwareStatus = AUDIO_HW_IDLE; 1938 } 1939 return NO_ERROR; 1940 } 1941 1942 return NO_INIT; 1943 } 1944 1945 audio_io_handle_t AudioFlinger::openDuplicateOutput(audio_io_handle_t output1, 1946 audio_io_handle_t output2) 1947 { 1948 Mutex::Autolock _l(mLock); 1949 MixerThread *thread1 = checkMixerThread_l(output1); 1950 MixerThread *thread2 = checkMixerThread_l(output2); 1951 1952 if (thread1 == NULL || thread2 == NULL) { 1953 ALOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1, 1954 output2); 1955 return AUDIO_IO_HANDLE_NONE; 1956 } 1957 1958 audio_io_handle_t id = nextUniqueId(AUDIO_UNIQUE_ID_USE_OUTPUT); 1959 DuplicatingThread *thread = new DuplicatingThread(this, thread1, id, mSystemReady); 1960 thread->addOutputTrack(thread2); 1961 mPlaybackThreads.add(id, thread); 1962 // notify client processes of the new output creation 1963 thread->ioConfigChanged(AUDIO_OUTPUT_OPENED); 1964 return id; 1965 } 1966 1967 status_t AudioFlinger::closeOutput(audio_io_handle_t output) 1968 { 1969 return closeOutput_nonvirtual(output); 1970 } 1971 1972 status_t AudioFlinger::closeOutput_nonvirtual(audio_io_handle_t output) 1973 { 1974 // keep strong reference on the playback thread so that 1975 // it is not destroyed while exit() is executed 1976 sp<PlaybackThread> thread; 1977 { 1978 Mutex::Autolock _l(mLock); 1979 thread = checkPlaybackThread_l(output); 1980 if (thread == NULL) { 1981 return BAD_VALUE; 1982 } 1983 1984 ALOGV("closeOutput() %d", output); 1985 1986 if (thread->type() == ThreadBase::MIXER) { 1987 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1988 if (mPlaybackThreads.valueAt(i)->isDuplicating()) { 1989 DuplicatingThread *dupThread = 1990 (DuplicatingThread *)mPlaybackThreads.valueAt(i).get(); 1991 dupThread->removeOutputTrack((MixerThread *)thread.get()); 1992 } 1993 } 1994 } 1995 1996 1997 mPlaybackThreads.removeItem(output); 1998 // save all effects to the default thread 1999 if (mPlaybackThreads.size()) { 2000 PlaybackThread *dstThread = checkPlaybackThread_l(mPlaybackThreads.keyAt(0)); 2001 if (dstThread != NULL) { 2002 // audioflinger lock is held here so the acquisition order of thread locks does not 2003 // matter 2004 Mutex::Autolock _dl(dstThread->mLock); 2005 Mutex::Autolock _sl(thread->mLock); 2006 Vector< sp<EffectChain> > effectChains = thread->getEffectChains_l(); 2007 for (size_t i = 0; i < effectChains.size(); i ++) { 2008 moveEffectChain_l(effectChains[i]->sessionId(), thread.get(), dstThread, true); 2009 } 2010 } 2011 } 2012 const sp<AudioIoDescriptor> ioDesc = new AudioIoDescriptor(); 2013 ioDesc->mIoHandle = output; 2014 ioConfigChanged(AUDIO_OUTPUT_CLOSED, ioDesc); 2015 } 2016 thread->exit(); 2017 // The thread entity (active unit of execution) is no longer running here, 2018 // but the ThreadBase container still exists. 2019 2020 if (!thread->isDuplicating()) { 2021 closeOutputFinish(thread); 2022 } 2023 2024 return NO_ERROR; 2025 } 2026 2027 void AudioFlinger::closeOutputFinish(sp<PlaybackThread> thread) 2028 { 2029 AudioStreamOut *out = thread->clearOutput(); 2030 ALOG_ASSERT(out != NULL, "out shouldn't be NULL"); 2031 // from now on thread->mOutput is NULL 2032 out->hwDev()->close_output_stream(out->hwDev(), out->stream); 2033 delete out; 2034 } 2035 2036 void AudioFlinger::closeOutputInternal_l(sp<PlaybackThread> thread) 2037 { 2038 mPlaybackThreads.removeItem(thread->mId); 2039 thread->exit(); 2040 closeOutputFinish(thread); 2041 } 2042 2043 status_t AudioFlinger::suspendOutput(audio_io_handle_t output) 2044 { 2045 Mutex::Autolock _l(mLock); 2046 PlaybackThread *thread = checkPlaybackThread_l(output); 2047 2048 if (thread == NULL) { 2049 return BAD_VALUE; 2050 } 2051 2052 ALOGV("suspendOutput() %d", output); 2053 thread->suspend(); 2054 2055 return NO_ERROR; 2056 } 2057 2058 status_t AudioFlinger::restoreOutput(audio_io_handle_t output) 2059 { 2060 Mutex::Autolock _l(mLock); 2061 PlaybackThread *thread = checkPlaybackThread_l(output); 2062 2063 if (thread == NULL) { 2064 return BAD_VALUE; 2065 } 2066 2067 ALOGV("restoreOutput() %d", output); 2068 2069 thread->restore(); 2070 2071 return NO_ERROR; 2072 } 2073 2074 status_t AudioFlinger::openInput(audio_module_handle_t module, 2075 audio_io_handle_t *input, 2076 audio_config_t *config, 2077 audio_devices_t *devices, 2078 const String8& address, 2079 audio_source_t source, 2080 audio_input_flags_t flags) 2081 { 2082 Mutex::Autolock _l(mLock); 2083 2084 if (*devices == AUDIO_DEVICE_NONE) { 2085 return BAD_VALUE; 2086 } 2087 2088 sp<RecordThread> thread = openInput_l(module, input, config, *devices, address, source, flags); 2089 2090 if (thread != 0) { 2091 // notify client processes of the new input creation 2092 thread->ioConfigChanged(AUDIO_INPUT_OPENED); 2093 return NO_ERROR; 2094 } 2095 return NO_INIT; 2096 } 2097 2098 sp<AudioFlinger::RecordThread> AudioFlinger::openInput_l(audio_module_handle_t module, 2099 audio_io_handle_t *input, 2100 audio_config_t *config, 2101 audio_devices_t devices, 2102 const String8& address, 2103 audio_source_t source, 2104 audio_input_flags_t flags) 2105 { 2106 AudioHwDevice *inHwDev = findSuitableHwDev_l(module, devices); 2107 if (inHwDev == NULL) { 2108 *input = AUDIO_IO_HANDLE_NONE; 2109 return 0; 2110 } 2111 2112 // Audio Policy can request a specific handle for hardware hotword. 2113 // The goal here is not to re-open an already opened input. 2114 // It is to use a pre-assigned I/O handle. 2115 if (*input == AUDIO_IO_HANDLE_NONE) { 2116 *input = nextUniqueId(AUDIO_UNIQUE_ID_USE_INPUT); 2117 } else if (audio_unique_id_get_use(*input) != AUDIO_UNIQUE_ID_USE_INPUT) { 2118 ALOGE("openInput_l() requested input handle %d is invalid", *input); 2119 return 0; 2120 } else if (mRecordThreads.indexOfKey(*input) >= 0) { 2121 // This should not happen in a transient state with current design. 2122 ALOGE("openInput_l() requested input handle %d is already assigned", *input); 2123 return 0; 2124 } 2125 2126 audio_config_t halconfig = *config; 2127 audio_hw_device_t *inHwHal = inHwDev->hwDevice(); 2128 audio_stream_in_t *inStream = NULL; 2129 status_t status = inHwHal->open_input_stream(inHwHal, *input, devices, &halconfig, 2130 &inStream, flags, address.string(), source); 2131 ALOGV("openInput_l() openInputStream returned input %p, SamplingRate %d" 2132 ", Format %#x, Channels %x, flags %#x, status %d addr %s", 2133 inStream, 2134 halconfig.sample_rate, 2135 halconfig.format, 2136 halconfig.channel_mask, 2137 flags, 2138 status, address.string()); 2139 2140 // If the input could not be opened with the requested parameters and we can handle the 2141 // conversion internally, try to open again with the proposed parameters. 2142 if (status == BAD_VALUE && 2143 audio_is_linear_pcm(config->format) && 2144 audio_is_linear_pcm(halconfig.format) && 2145 (halconfig.sample_rate <= AUDIO_RESAMPLER_DOWN_RATIO_MAX * config->sample_rate) && 2146 (audio_channel_count_from_in_mask(halconfig.channel_mask) <= FCC_8) && 2147 (audio_channel_count_from_in_mask(config->channel_mask) <= FCC_8)) { 2148 // FIXME describe the change proposed by HAL (save old values so we can log them here) 2149 ALOGV("openInput_l() reopening with proposed sampling rate and channel mask"); 2150 inStream = NULL; 2151 status = inHwHal->open_input_stream(inHwHal, *input, devices, &halconfig, 2152 &inStream, flags, address.string(), source); 2153 // FIXME log this new status; HAL should not propose any further changes 2154 } 2155 2156 if (status == NO_ERROR && inStream != NULL) { 2157 2158 #ifdef TEE_SINK 2159 // Try to re-use most recently used Pipe to archive a copy of input for dumpsys, 2160 // or (re-)create if current Pipe is idle and does not match the new format 2161 sp<NBAIO_Sink> teeSink; 2162 enum { 2163 TEE_SINK_NO, // don't copy input 2164 TEE_SINK_NEW, // copy input using a new pipe 2165 TEE_SINK_OLD, // copy input using an existing pipe 2166 } kind; 2167 NBAIO_Format format = Format_from_SR_C(halconfig.sample_rate, 2168 audio_channel_count_from_in_mask(halconfig.channel_mask), halconfig.format); 2169 if (!mTeeSinkInputEnabled) { 2170 kind = TEE_SINK_NO; 2171 } else if (!Format_isValid(format)) { 2172 kind = TEE_SINK_NO; 2173 } else if (mRecordTeeSink == 0) { 2174 kind = TEE_SINK_NEW; 2175 } else if (mRecordTeeSink->getStrongCount() != 1) { 2176 kind = TEE_SINK_NO; 2177 } else if (Format_isEqual(format, mRecordTeeSink->format())) { 2178 kind = TEE_SINK_OLD; 2179 } else { 2180 kind = TEE_SINK_NEW; 2181 } 2182 switch (kind) { 2183 case TEE_SINK_NEW: { 2184 Pipe *pipe = new Pipe(mTeeSinkInputFrames, format); 2185 size_t numCounterOffers = 0; 2186 const NBAIO_Format offers[1] = {format}; 2187 ssize_t index = pipe->negotiate(offers, 1, NULL, numCounterOffers); 2188 ALOG_ASSERT(index == 0); 2189 PipeReader *pipeReader = new PipeReader(*pipe); 2190 numCounterOffers = 0; 2191 index = pipeReader->negotiate(offers, 1, NULL, numCounterOffers); 2192 ALOG_ASSERT(index == 0); 2193 mRecordTeeSink = pipe; 2194 mRecordTeeSource = pipeReader; 2195 teeSink = pipe; 2196 } 2197 break; 2198 case TEE_SINK_OLD: 2199 teeSink = mRecordTeeSink; 2200 break; 2201 case TEE_SINK_NO: 2202 default: 2203 break; 2204 } 2205 #endif 2206 2207 AudioStreamIn *inputStream = new AudioStreamIn(inHwDev, inStream, flags); 2208 2209 // Start record thread 2210 // RecordThread requires both input and output device indication to forward to audio 2211 // pre processing modules 2212 sp<RecordThread> thread = new RecordThread(this, 2213 inputStream, 2214 *input, 2215 primaryOutputDevice_l(), 2216 devices, 2217 mSystemReady 2218 #ifdef TEE_SINK 2219 , teeSink 2220 #endif 2221 ); 2222 mRecordThreads.add(*input, thread); 2223 ALOGV("openInput_l() created record thread: ID %d thread %p", *input, thread.get()); 2224 return thread; 2225 } 2226 2227 *input = AUDIO_IO_HANDLE_NONE; 2228 return 0; 2229 } 2230 2231 status_t AudioFlinger::closeInput(audio_io_handle_t input) 2232 { 2233 return closeInput_nonvirtual(input); 2234 } 2235 2236 status_t AudioFlinger::closeInput_nonvirtual(audio_io_handle_t input) 2237 { 2238 // keep strong reference on the record thread so that 2239 // it is not destroyed while exit() is executed 2240 sp<RecordThread> thread; 2241 { 2242 Mutex::Autolock _l(mLock); 2243 thread = checkRecordThread_l(input); 2244 if (thread == 0) { 2245 return BAD_VALUE; 2246 } 2247 2248 ALOGV("closeInput() %d", input); 2249 2250 // If we still have effect chains, it means that a client still holds a handle 2251 // on at least one effect. We must either move the chain to an existing thread with the 2252 // same session ID or put it aside in case a new record thread is opened for a 2253 // new capture on the same session 2254 sp<EffectChain> chain; 2255 { 2256 Mutex::Autolock _sl(thread->mLock); 2257 Vector< sp<EffectChain> > effectChains = thread->getEffectChains_l(); 2258 // Note: maximum one chain per record thread 2259 if (effectChains.size() != 0) { 2260 chain = effectChains[0]; 2261 } 2262 } 2263 if (chain != 0) { 2264 // first check if a record thread is already opened with a client on the same session. 2265 // This should only happen in case of overlap between one thread tear down and the 2266 // creation of its replacement 2267 size_t i; 2268 for (i = 0; i < mRecordThreads.size(); i++) { 2269 sp<RecordThread> t = mRecordThreads.valueAt(i); 2270 if (t == thread) { 2271 continue; 2272 } 2273 if (t->hasAudioSession(chain->sessionId()) != 0) { 2274 Mutex::Autolock _l(t->mLock); 2275 ALOGV("closeInput() found thread %d for effect session %d", 2276 t->id(), chain->sessionId()); 2277 t->addEffectChain_l(chain); 2278 break; 2279 } 2280 } 2281 // put the chain aside if we could not find a record thread with the same session id. 2282 if (i == mRecordThreads.size()) { 2283 putOrphanEffectChain_l(chain); 2284 } 2285 } 2286 const sp<AudioIoDescriptor> ioDesc = new AudioIoDescriptor(); 2287 ioDesc->mIoHandle = input; 2288 ioConfigChanged(AUDIO_INPUT_CLOSED, ioDesc); 2289 mRecordThreads.removeItem(input); 2290 } 2291 // FIXME: calling thread->exit() without mLock held should not be needed anymore now that 2292 // we have a different lock for notification client 2293 closeInputFinish(thread); 2294 return NO_ERROR; 2295 } 2296 2297 void AudioFlinger::closeInputFinish(sp<RecordThread> thread) 2298 { 2299 thread->exit(); 2300 AudioStreamIn *in = thread->clearInput(); 2301 ALOG_ASSERT(in != NULL, "in shouldn't be NULL"); 2302 // from now on thread->mInput is NULL 2303 in->hwDev()->close_input_stream(in->hwDev(), in->stream); 2304 delete in; 2305 } 2306 2307 void AudioFlinger::closeInputInternal_l(sp<RecordThread> thread) 2308 { 2309 mRecordThreads.removeItem(thread->mId); 2310 closeInputFinish(thread); 2311 } 2312 2313 status_t AudioFlinger::invalidateStream(audio_stream_type_t stream) 2314 { 2315 Mutex::Autolock _l(mLock); 2316 ALOGV("invalidateStream() stream %d", stream); 2317 2318 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2319 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 2320 thread->invalidateTracks(stream); 2321 } 2322 2323 return NO_ERROR; 2324 } 2325 2326 2327 audio_unique_id_t AudioFlinger::newAudioUniqueId(audio_unique_id_use_t use) 2328 { 2329 // This is a binder API, so a malicious client could pass in a bad parameter. 2330 // Check for that before calling the internal API nextUniqueId(). 2331 if ((unsigned) use >= (unsigned) AUDIO_UNIQUE_ID_USE_MAX) { 2332 ALOGE("newAudioUniqueId invalid use %d", use); 2333 return AUDIO_UNIQUE_ID_ALLOCATE; 2334 } 2335 return nextUniqueId(use); 2336 } 2337 2338 void AudioFlinger::acquireAudioSessionId(audio_session_t audioSession, pid_t pid) 2339 { 2340 Mutex::Autolock _l(mLock); 2341 pid_t caller = IPCThreadState::self()->getCallingPid(); 2342 ALOGV("acquiring %d from %d, for %d", audioSession, caller, pid); 2343 if (pid != -1 && (caller == getpid_cached)) { 2344 caller = pid; 2345 } 2346 2347 { 2348 Mutex::Autolock _cl(mClientLock); 2349 // Ignore requests received from processes not known as notification client. The request 2350 // is likely proxied by mediaserver (e.g CameraService) and releaseAudioSessionId() can be 2351 // called from a different pid leaving a stale session reference. Also we don't know how 2352 // to clear this reference if the client process dies. 2353 if (mNotificationClients.indexOfKey(caller) < 0) { 2354 ALOGW("acquireAudioSessionId() unknown client %d for session %d", caller, audioSession); 2355 return; 2356 } 2357 } 2358 2359 size_t num = mAudioSessionRefs.size(); 2360 for (size_t i = 0; i< num; i++) { 2361 AudioSessionRef *ref = mAudioSessionRefs.editItemAt(i); 2362 if (ref->mSessionid == audioSession && ref->mPid == caller) { 2363 ref->mCnt++; 2364 ALOGV(" incremented refcount to %d", ref->mCnt); 2365 return; 2366 } 2367 } 2368 mAudioSessionRefs.push(new AudioSessionRef(audioSession, caller)); 2369 ALOGV(" added new entry for %d", audioSession); 2370 } 2371 2372 void AudioFlinger::releaseAudioSessionId(audio_session_t audioSession, pid_t pid) 2373 { 2374 Mutex::Autolock _l(mLock); 2375 pid_t caller = IPCThreadState::self()->getCallingPid(); 2376 ALOGV("releasing %d from %d for %d", audioSession, caller, pid); 2377 if (pid != -1 && (caller == getpid_cached)) { 2378 caller = pid; 2379 } 2380 size_t num = mAudioSessionRefs.size(); 2381 for (size_t i = 0; i< num; i++) { 2382 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 2383 if (ref->mSessionid == audioSession && ref->mPid == caller) { 2384 ref->mCnt--; 2385 ALOGV(" decremented refcount to %d", ref->mCnt); 2386 if (ref->mCnt == 0) { 2387 mAudioSessionRefs.removeAt(i); 2388 delete ref; 2389 purgeStaleEffects_l(); 2390 } 2391 return; 2392 } 2393 } 2394 // If the caller is mediaserver it is likely that the session being released was acquired 2395 // on behalf of a process not in notification clients and we ignore the warning. 2396 ALOGW_IF(caller != getpid_cached, "session id %d not found for pid %d", audioSession, caller); 2397 } 2398 2399 void AudioFlinger::purgeStaleEffects_l() { 2400 2401 ALOGV("purging stale effects"); 2402 2403 Vector< sp<EffectChain> > chains; 2404 2405 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2406 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 2407 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 2408 sp<EffectChain> ec = t->mEffectChains[j]; 2409 if (ec->sessionId() > AUDIO_SESSION_OUTPUT_MIX) { 2410 chains.push(ec); 2411 } 2412 } 2413 } 2414 for (size_t i = 0; i < mRecordThreads.size(); i++) { 2415 sp<RecordThread> t = mRecordThreads.valueAt(i); 2416 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 2417 sp<EffectChain> ec = t->mEffectChains[j]; 2418 chains.push(ec); 2419 } 2420 } 2421 2422 for (size_t i = 0; i < chains.size(); i++) { 2423 sp<EffectChain> ec = chains[i]; 2424 int sessionid = ec->sessionId(); 2425 sp<ThreadBase> t = ec->mThread.promote(); 2426 if (t == 0) { 2427 continue; 2428 } 2429 size_t numsessionrefs = mAudioSessionRefs.size(); 2430 bool found = false; 2431 for (size_t k = 0; k < numsessionrefs; k++) { 2432 AudioSessionRef *ref = mAudioSessionRefs.itemAt(k); 2433 if (ref->mSessionid == sessionid) { 2434 ALOGV(" session %d still exists for %d with %d refs", 2435 sessionid, ref->mPid, ref->mCnt); 2436 found = true; 2437 break; 2438 } 2439 } 2440 if (!found) { 2441 Mutex::Autolock _l(t->mLock); 2442 // remove all effects from the chain 2443 while (ec->mEffects.size()) { 2444 sp<EffectModule> effect = ec->mEffects[0]; 2445 effect->unPin(); 2446 t->removeEffect_l(effect); 2447 if (effect->purgeHandles()) { 2448 t->checkSuspendOnEffectEnabled_l(effect, false, effect->sessionId()); 2449 } 2450 AudioSystem::unregisterEffect(effect->id()); 2451 } 2452 } 2453 } 2454 return; 2455 } 2456 2457 // checkThread_l() must be called with AudioFlinger::mLock held 2458 AudioFlinger::ThreadBase *AudioFlinger::checkThread_l(audio_io_handle_t ioHandle) const 2459 { 2460 ThreadBase *thread = NULL; 2461 switch (audio_unique_id_get_use(ioHandle)) { 2462 case AUDIO_UNIQUE_ID_USE_OUTPUT: 2463 thread = checkPlaybackThread_l(ioHandle); 2464 break; 2465 case AUDIO_UNIQUE_ID_USE_INPUT: 2466 thread = checkRecordThread_l(ioHandle); 2467 break; 2468 default: 2469 break; 2470 } 2471 return thread; 2472 } 2473 2474 // checkPlaybackThread_l() must be called with AudioFlinger::mLock held 2475 AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(audio_io_handle_t output) const 2476 { 2477 return mPlaybackThreads.valueFor(output).get(); 2478 } 2479 2480 // checkMixerThread_l() must be called with AudioFlinger::mLock held 2481 AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(audio_io_handle_t output) const 2482 { 2483 PlaybackThread *thread = checkPlaybackThread_l(output); 2484 return thread != NULL && thread->type() != ThreadBase::DIRECT ? (MixerThread *) thread : NULL; 2485 } 2486 2487 // checkRecordThread_l() must be called with AudioFlinger::mLock held 2488 AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(audio_io_handle_t input) const 2489 { 2490 return mRecordThreads.valueFor(input).get(); 2491 } 2492 2493 audio_unique_id_t AudioFlinger::nextUniqueId(audio_unique_id_use_t use) 2494 { 2495 // This is the internal API, so it is OK to assert on bad parameter. 2496 LOG_ALWAYS_FATAL_IF((unsigned) use >= (unsigned) AUDIO_UNIQUE_ID_USE_MAX); 2497 const int maxRetries = use == AUDIO_UNIQUE_ID_USE_SESSION ? 3 : 1; 2498 for (int retry = 0; retry < maxRetries; retry++) { 2499 // The cast allows wraparound from max positive to min negative instead of abort 2500 uint32_t base = (uint32_t) atomic_fetch_add_explicit(&mNextUniqueIds[use], 2501 (uint_fast32_t) AUDIO_UNIQUE_ID_USE_MAX, memory_order_acq_rel); 2502 ALOG_ASSERT(audio_unique_id_get_use(base) == AUDIO_UNIQUE_ID_USE_UNSPECIFIED); 2503 // allow wrap by skipping 0 and -1 for session ids 2504 if (!(base == 0 || base == (~0u & ~AUDIO_UNIQUE_ID_USE_MASK))) { 2505 ALOGW_IF(retry != 0, "unique ID overflow for use %d", use); 2506 return (audio_unique_id_t) (base | use); 2507 } 2508 } 2509 // We have no way of recovering from wraparound 2510 LOG_ALWAYS_FATAL("unique ID overflow for use %d", use); 2511 // TODO Use a floor after wraparound. This may need a mutex. 2512 } 2513 2514 AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l() const 2515 { 2516 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2517 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 2518 if(thread->isDuplicating()) { 2519 continue; 2520 } 2521 AudioStreamOut *output = thread->getOutput(); 2522 if (output != NULL && output->audioHwDev == mPrimaryHardwareDev) { 2523 return thread; 2524 } 2525 } 2526 return NULL; 2527 } 2528 2529 audio_devices_t AudioFlinger::primaryOutputDevice_l() const 2530 { 2531 PlaybackThread *thread = primaryPlaybackThread_l(); 2532 2533 if (thread == NULL) { 2534 return 0; 2535 } 2536 2537 return thread->outDevice(); 2538 } 2539 2540 AudioFlinger::PlaybackThread *AudioFlinger::fastPlaybackThread_l() const 2541 { 2542 size_t minFrameCount = 0; 2543 PlaybackThread *minThread = NULL; 2544 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2545 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 2546 if (!thread->isDuplicating()) { 2547 size_t frameCount = thread->frameCountHAL(); 2548 if (frameCount != 0 && (minFrameCount == 0 || frameCount < minFrameCount || 2549 (frameCount == minFrameCount && thread->hasFastMixer() && 2550 /*minThread != NULL &&*/ !minThread->hasFastMixer()))) { 2551 minFrameCount = frameCount; 2552 minThread = thread; 2553 } 2554 } 2555 } 2556 return minThread; 2557 } 2558 2559 sp<AudioFlinger::SyncEvent> AudioFlinger::createSyncEvent(AudioSystem::sync_event_t type, 2560 audio_session_t triggerSession, 2561 audio_session_t listenerSession, 2562 sync_event_callback_t callBack, 2563 wp<RefBase> cookie) 2564 { 2565 Mutex::Autolock _l(mLock); 2566 2567 sp<SyncEvent> event = new SyncEvent(type, triggerSession, listenerSession, callBack, cookie); 2568 status_t playStatus = NAME_NOT_FOUND; 2569 status_t recStatus = NAME_NOT_FOUND; 2570 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2571 playStatus = mPlaybackThreads.valueAt(i)->setSyncEvent(event); 2572 if (playStatus == NO_ERROR) { 2573 return event; 2574 } 2575 } 2576 for (size_t i = 0; i < mRecordThreads.size(); i++) { 2577 recStatus = mRecordThreads.valueAt(i)->setSyncEvent(event); 2578 if (recStatus == NO_ERROR) { 2579 return event; 2580 } 2581 } 2582 if (playStatus == NAME_NOT_FOUND || recStatus == NAME_NOT_FOUND) { 2583 mPendingSyncEvents.add(event); 2584 } else { 2585 ALOGV("createSyncEvent() invalid event %d", event->type()); 2586 event.clear(); 2587 } 2588 return event; 2589 } 2590 2591 // ---------------------------------------------------------------------------- 2592 // Effect management 2593 // ---------------------------------------------------------------------------- 2594 2595 2596 status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects) const 2597 { 2598 Mutex::Autolock _l(mLock); 2599 return EffectQueryNumberEffects(numEffects); 2600 } 2601 2602 status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor) const 2603 { 2604 Mutex::Autolock _l(mLock); 2605 return EffectQueryEffect(index, descriptor); 2606 } 2607 2608 status_t AudioFlinger::getEffectDescriptor(const effect_uuid_t *pUuid, 2609 effect_descriptor_t *descriptor) const 2610 { 2611 Mutex::Autolock _l(mLock); 2612 return EffectGetDescriptor(pUuid, descriptor); 2613 } 2614 2615 2616 sp<IEffect> AudioFlinger::createEffect( 2617 effect_descriptor_t *pDesc, 2618 const sp<IEffectClient>& effectClient, 2619 int32_t priority, 2620 audio_io_handle_t io, 2621 audio_session_t sessionId, 2622 const String16& opPackageName, 2623 status_t *status, 2624 int *id, 2625 int *enabled) 2626 { 2627 status_t lStatus = NO_ERROR; 2628 sp<EffectHandle> handle; 2629 effect_descriptor_t desc; 2630 2631 pid_t pid = IPCThreadState::self()->getCallingPid(); 2632 ALOGV("createEffect pid %d, effectClient %p, priority %d, sessionId %d, io %d", 2633 pid, effectClient.get(), priority, sessionId, io); 2634 2635 if (pDesc == NULL) { 2636 lStatus = BAD_VALUE; 2637 goto Exit; 2638 } 2639 2640 // check audio settings permission for global effects 2641 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && !settingsAllowed()) { 2642 lStatus = PERMISSION_DENIED; 2643 goto Exit; 2644 } 2645 2646 // Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects 2647 // that can only be created by audio policy manager (running in same process) 2648 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid_cached != pid) { 2649 lStatus = PERMISSION_DENIED; 2650 goto Exit; 2651 } 2652 2653 { 2654 if (!EffectIsNullUuid(&pDesc->uuid)) { 2655 // if uuid is specified, request effect descriptor 2656 lStatus = EffectGetDescriptor(&pDesc->uuid, &desc); 2657 if (lStatus < 0) { 2658 ALOGW("createEffect() error %d from EffectGetDescriptor", lStatus); 2659 goto Exit; 2660 } 2661 } else { 2662 // if uuid is not specified, look for an available implementation 2663 // of the required type in effect factory 2664 if (EffectIsNullUuid(&pDesc->type)) { 2665 ALOGW("createEffect() no effect type"); 2666 lStatus = BAD_VALUE; 2667 goto Exit; 2668 } 2669 uint32_t numEffects = 0; 2670 effect_descriptor_t d; 2671 d.flags = 0; // prevent compiler warning 2672 bool found = false; 2673 2674 lStatus = EffectQueryNumberEffects(&numEffects); 2675 if (lStatus < 0) { 2676 ALOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus); 2677 goto Exit; 2678 } 2679 for (uint32_t i = 0; i < numEffects; i++) { 2680 lStatus = EffectQueryEffect(i, &desc); 2681 if (lStatus < 0) { 2682 ALOGW("createEffect() error %d from EffectQueryEffect", lStatus); 2683 continue; 2684 } 2685 if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) { 2686 // If matching type found save effect descriptor. If the session is 2687 // 0 and the effect is not auxiliary, continue enumeration in case 2688 // an auxiliary version of this effect type is available 2689 found = true; 2690 d = desc; 2691 if (sessionId != AUDIO_SESSION_OUTPUT_MIX || 2692 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 2693 break; 2694 } 2695 } 2696 } 2697 if (!found) { 2698 lStatus = BAD_VALUE; 2699 ALOGW("createEffect() effect not found"); 2700 goto Exit; 2701 } 2702 // For same effect type, chose auxiliary version over insert version if 2703 // connect to output mix (Compliance to OpenSL ES) 2704 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && 2705 (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) { 2706 desc = d; 2707 } 2708 } 2709 2710 // Do not allow auxiliary effects on a session different from 0 (output mix) 2711 if (sessionId != AUDIO_SESSION_OUTPUT_MIX && 2712 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 2713 lStatus = INVALID_OPERATION; 2714 goto Exit; 2715 } 2716 2717 // check recording permission for visualizer 2718 if ((memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) && 2719 !recordingAllowed(opPackageName, pid, IPCThreadState::self()->getCallingUid())) { 2720 lStatus = PERMISSION_DENIED; 2721 goto Exit; 2722 } 2723 2724 // return effect descriptor 2725 *pDesc = desc; 2726 if (io == AUDIO_IO_HANDLE_NONE && sessionId == AUDIO_SESSION_OUTPUT_MIX) { 2727 // if the output returned by getOutputForEffect() is removed before we lock the 2728 // mutex below, the call to checkPlaybackThread_l(io) below will detect it 2729 // and we will exit safely 2730 io = AudioSystem::getOutputForEffect(&desc); 2731 ALOGV("createEffect got output %d", io); 2732 } 2733 2734 Mutex::Autolock _l(mLock); 2735 2736 // If output is not specified try to find a matching audio session ID in one of the 2737 // output threads. 2738 // If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX 2739 // because of code checking output when entering the function. 2740 // Note: io is never 0 when creating an effect on an input 2741 if (io == AUDIO_IO_HANDLE_NONE) { 2742 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) { 2743 // output must be specified by AudioPolicyManager when using session 2744 // AUDIO_SESSION_OUTPUT_STAGE 2745 lStatus = BAD_VALUE; 2746 goto Exit; 2747 } 2748 // look for the thread where the specified audio session is present 2749 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2750 if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 2751 io = mPlaybackThreads.keyAt(i); 2752 break; 2753 } 2754 } 2755 if (io == 0) { 2756 for (size_t i = 0; i < mRecordThreads.size(); i++) { 2757 if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 2758 io = mRecordThreads.keyAt(i); 2759 break; 2760 } 2761 } 2762 } 2763 // If no output thread contains the requested session ID, default to 2764 // first output. The effect chain will be moved to the correct output 2765 // thread when a track with the same session ID is created 2766 if (io == AUDIO_IO_HANDLE_NONE && mPlaybackThreads.size() > 0) { 2767 io = mPlaybackThreads.keyAt(0); 2768 } 2769 ALOGV("createEffect() got io %d for effect %s", io, desc.name); 2770 } 2771 ThreadBase *thread = checkRecordThread_l(io); 2772 if (thread == NULL) { 2773 thread = checkPlaybackThread_l(io); 2774 if (thread == NULL) { 2775 ALOGE("createEffect() unknown output thread"); 2776 lStatus = BAD_VALUE; 2777 goto Exit; 2778 } 2779 } else { 2780 // Check if one effect chain was awaiting for an effect to be created on this 2781 // session and used it instead of creating a new one. 2782 sp<EffectChain> chain = getOrphanEffectChain_l(sessionId); 2783 if (chain != 0) { 2784 Mutex::Autolock _l(thread->mLock); 2785 thread->addEffectChain_l(chain); 2786 } 2787 } 2788 2789 sp<Client> client = registerPid(pid); 2790 2791 // create effect on selected output thread 2792 handle = thread->createEffect_l(client, effectClient, priority, sessionId, 2793 &desc, enabled, &lStatus); 2794 if (handle != 0 && id != NULL) { 2795 *id = handle->id(); 2796 } 2797 if (handle == 0) { 2798 // remove local strong reference to Client with mClientLock held 2799 Mutex::Autolock _cl(mClientLock); 2800 client.clear(); 2801 } 2802 } 2803 2804 Exit: 2805 *status = lStatus; 2806 return handle; 2807 } 2808 2809 status_t AudioFlinger::moveEffects(audio_session_t sessionId, audio_io_handle_t srcOutput, 2810 audio_io_handle_t dstOutput) 2811 { 2812 ALOGV("moveEffects() session %d, srcOutput %d, dstOutput %d", 2813 sessionId, srcOutput, dstOutput); 2814 Mutex::Autolock _l(mLock); 2815 if (srcOutput == dstOutput) { 2816 ALOGW("moveEffects() same dst and src outputs %d", dstOutput); 2817 return NO_ERROR; 2818 } 2819 PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput); 2820 if (srcThread == NULL) { 2821 ALOGW("moveEffects() bad srcOutput %d", srcOutput); 2822 return BAD_VALUE; 2823 } 2824 PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput); 2825 if (dstThread == NULL) { 2826 ALOGW("moveEffects() bad dstOutput %d", dstOutput); 2827 return BAD_VALUE; 2828 } 2829 2830 Mutex::Autolock _dl(dstThread->mLock); 2831 Mutex::Autolock _sl(srcThread->mLock); 2832 return moveEffectChain_l(sessionId, srcThread, dstThread, false); 2833 } 2834 2835 // moveEffectChain_l must be called with both srcThread and dstThread mLocks held 2836 status_t AudioFlinger::moveEffectChain_l(audio_session_t sessionId, 2837 AudioFlinger::PlaybackThread *srcThread, 2838 AudioFlinger::PlaybackThread *dstThread, 2839 bool reRegister) 2840 { 2841 ALOGV("moveEffectChain_l() session %d from thread %p to thread %p", 2842 sessionId, srcThread, dstThread); 2843 2844 sp<EffectChain> chain = srcThread->getEffectChain_l(sessionId); 2845 if (chain == 0) { 2846 ALOGW("moveEffectChain_l() effect chain for session %d not on source thread %p", 2847 sessionId, srcThread); 2848 return INVALID_OPERATION; 2849 } 2850 2851 // Check whether the destination thread and all effects in the chain are compatible 2852 if (!chain->isCompatibleWithThread_l(dstThread)) { 2853 ALOGW("moveEffectChain_l() effect chain failed because" 2854 " destination thread %p is not compatible with effects in the chain", 2855 dstThread); 2856 return INVALID_OPERATION; 2857 } 2858 2859 // remove chain first. This is useful only if reconfiguring effect chain on same output thread, 2860 // so that a new chain is created with correct parameters when first effect is added. This is 2861 // otherwise unnecessary as removeEffect_l() will remove the chain when last effect is 2862 // removed. 2863 srcThread->removeEffectChain_l(chain); 2864 2865 // transfer all effects one by one so that new effect chain is created on new thread with 2866 // correct buffer sizes and audio parameters and effect engines reconfigured accordingly 2867 sp<EffectChain> dstChain; 2868 uint32_t strategy = 0; // prevent compiler warning 2869 sp<EffectModule> effect = chain->getEffectFromId_l(0); 2870 Vector< sp<EffectModule> > removed; 2871 status_t status = NO_ERROR; 2872 while (effect != 0) { 2873 srcThread->removeEffect_l(effect); 2874 removed.add(effect); 2875 status = dstThread->addEffect_l(effect); 2876 if (status != NO_ERROR) { 2877 break; 2878 } 2879 // removeEffect_l() has stopped the effect if it was active so it must be restarted 2880 if (effect->state() == EffectModule::ACTIVE || 2881 effect->state() == EffectModule::STOPPING) { 2882 effect->start(); 2883 } 2884 // if the move request is not received from audio policy manager, the effect must be 2885 // re-registered with the new strategy and output 2886 if (dstChain == 0) { 2887 dstChain = effect->chain().promote(); 2888 if (dstChain == 0) { 2889 ALOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get()); 2890 status = NO_INIT; 2891 break; 2892 } 2893 strategy = dstChain->strategy(); 2894 } 2895 if (reRegister) { 2896 AudioSystem::unregisterEffect(effect->id()); 2897 AudioSystem::registerEffect(&effect->desc(), 2898 dstThread->id(), 2899 strategy, 2900 sessionId, 2901 effect->id()); 2902 AudioSystem::setEffectEnabled(effect->id(), effect->isEnabled()); 2903 } 2904 effect = chain->getEffectFromId_l(0); 2905 } 2906 2907 if (status != NO_ERROR) { 2908 for (size_t i = 0; i < removed.size(); i++) { 2909 srcThread->addEffect_l(removed[i]); 2910 if (dstChain != 0 && reRegister) { 2911 AudioSystem::unregisterEffect(removed[i]->id()); 2912 AudioSystem::registerEffect(&removed[i]->desc(), 2913 srcThread->id(), 2914 strategy, 2915 sessionId, 2916 removed[i]->id()); 2917 AudioSystem::setEffectEnabled(effect->id(), effect->isEnabled()); 2918 } 2919 } 2920 } 2921 2922 return status; 2923 } 2924 2925 bool AudioFlinger::isNonOffloadableGlobalEffectEnabled_l() 2926 { 2927 if (mGlobalEffectEnableTime != 0 && 2928 ((systemTime() - mGlobalEffectEnableTime) < kMinGlobalEffectEnabletimeNs)) { 2929 return true; 2930 } 2931 2932 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2933 sp<EffectChain> ec = 2934 mPlaybackThreads.valueAt(i)->getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2935 if (ec != 0 && ec->isNonOffloadableEnabled()) { 2936 return true; 2937 } 2938 } 2939 return false; 2940 } 2941 2942 void AudioFlinger::onNonOffloadableGlobalEffectEnable() 2943 { 2944 Mutex::Autolock _l(mLock); 2945 2946 mGlobalEffectEnableTime = systemTime(); 2947 2948 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2949 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 2950 if (t->mType == ThreadBase::OFFLOAD) { 2951 t->invalidateTracks(AUDIO_STREAM_MUSIC); 2952 } 2953 } 2954 2955 } 2956 2957 status_t AudioFlinger::putOrphanEffectChain_l(const sp<AudioFlinger::EffectChain>& chain) 2958 { 2959 audio_session_t session = chain->sessionId(); 2960 ssize_t index = mOrphanEffectChains.indexOfKey(session); 2961 ALOGV("putOrphanEffectChain_l session %d index %zd", session, index); 2962 if (index >= 0) { 2963 ALOGW("putOrphanEffectChain_l chain for session %d already present", session); 2964 return ALREADY_EXISTS; 2965 } 2966 mOrphanEffectChains.add(session, chain); 2967 return NO_ERROR; 2968 } 2969 2970 sp<AudioFlinger::EffectChain> AudioFlinger::getOrphanEffectChain_l(audio_session_t session) 2971 { 2972 sp<EffectChain> chain; 2973 ssize_t index = mOrphanEffectChains.indexOfKey(session); 2974 ALOGV("getOrphanEffectChain_l session %d index %zd", session, index); 2975 if (index >= 0) { 2976 chain = mOrphanEffectChains.valueAt(index); 2977 mOrphanEffectChains.removeItemsAt(index); 2978 } 2979 return chain; 2980 } 2981 2982 bool AudioFlinger::updateOrphanEffectChains(const sp<AudioFlinger::EffectModule>& effect) 2983 { 2984 Mutex::Autolock _l(mLock); 2985 audio_session_t session = effect->sessionId(); 2986 ssize_t index = mOrphanEffectChains.indexOfKey(session); 2987 ALOGV("updateOrphanEffectChains session %d index %zd", session, index); 2988 if (index >= 0) { 2989 sp<EffectChain> chain = mOrphanEffectChains.valueAt(index); 2990 if (chain->removeEffect_l(effect) == 0) { 2991 ALOGV("updateOrphanEffectChains removing effect chain at index %zd", index); 2992 mOrphanEffectChains.removeItemsAt(index); 2993 } 2994 return true; 2995 } 2996 return false; 2997 } 2998 2999 3000 struct Entry { 3001 #define TEE_MAX_FILENAME 32 // %Y%m%d%H%M%S_%d.wav = 4+2+2+2+2+2+1+1+4+1 = 21 3002 char mFileName[TEE_MAX_FILENAME]; 3003 }; 3004 3005 int comparEntry(const void *p1, const void *p2) 3006 { 3007 return strcmp(((const Entry *) p1)->mFileName, ((const Entry *) p2)->mFileName); 3008 } 3009 3010 #ifdef TEE_SINK 3011 void AudioFlinger::dumpTee(int fd, const sp<NBAIO_Source>& source, audio_io_handle_t id) 3012 { 3013 NBAIO_Source *teeSource = source.get(); 3014 if (teeSource != NULL) { 3015 // .wav rotation 3016 // There is a benign race condition if 2 threads call this simultaneously. 3017 // They would both traverse the directory, but the result would simply be 3018 // failures at unlink() which are ignored. It's also unlikely since 3019 // normally dumpsys is only done by bugreport or from the command line. 3020 char teePath[32+256]; 3021 strcpy(teePath, "/data/misc/audioserver"); 3022 size_t teePathLen = strlen(teePath); 3023 DIR *dir = opendir(teePath); 3024 teePath[teePathLen++] = '/'; 3025 if (dir != NULL) { 3026 #define TEE_MAX_SORT 20 // number of entries to sort 3027 #define TEE_MAX_KEEP 10 // number of entries to keep 3028 struct Entry entries[TEE_MAX_SORT]; 3029 size_t entryCount = 0; 3030 while (entryCount < TEE_MAX_SORT) { 3031 struct dirent de; 3032 struct dirent *result = NULL; 3033 int rc = readdir_r(dir, &de, &result); 3034 if (rc != 0) { 3035 ALOGW("readdir_r failed %d", rc); 3036 break; 3037 } 3038 if (result == NULL) { 3039 break; 3040 } 3041 if (result != &de) { 3042 ALOGW("readdir_r returned unexpected result %p != %p", result, &de); 3043 break; 3044 } 3045 // ignore non .wav file entries 3046 size_t nameLen = strlen(de.d_name); 3047 if (nameLen <= 4 || nameLen >= TEE_MAX_FILENAME || 3048 strcmp(&de.d_name[nameLen - 4], ".wav")) { 3049 continue; 3050 } 3051 strcpy(entries[entryCount++].mFileName, de.d_name); 3052 } 3053 (void) closedir(dir); 3054 if (entryCount > TEE_MAX_KEEP) { 3055 qsort(entries, entryCount, sizeof(Entry), comparEntry); 3056 for (size_t i = 0; i < entryCount - TEE_MAX_KEEP; ++i) { 3057 strcpy(&teePath[teePathLen], entries[i].mFileName); 3058 (void) unlink(teePath); 3059 } 3060 } 3061 } else { 3062 if (fd >= 0) { 3063 dprintf(fd, "unable to rotate tees in %.*s: %s\n", (int) teePathLen, teePath, 3064 strerror(errno)); 3065 } 3066 } 3067 char teeTime[16]; 3068 struct timeval tv; 3069 gettimeofday(&tv, NULL); 3070 struct tm tm; 3071 localtime_r(&tv.tv_sec, &tm); 3072 strftime(teeTime, sizeof(teeTime), "%Y%m%d%H%M%S", &tm); 3073 snprintf(&teePath[teePathLen], sizeof(teePath) - teePathLen, "%s_%d.wav", teeTime, id); 3074 // if 2 dumpsys are done within 1 second, and rotation didn't work, then discard 2nd 3075 int teeFd = open(teePath, O_WRONLY | O_CREAT | O_EXCL | O_NOFOLLOW, S_IRUSR | S_IWUSR); 3076 if (teeFd >= 0) { 3077 // FIXME use libsndfile 3078 char wavHeader[44]; 3079 memcpy(wavHeader, 3080 "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", 3081 sizeof(wavHeader)); 3082 NBAIO_Format format = teeSource->format(); 3083 unsigned channelCount = Format_channelCount(format); 3084 uint32_t sampleRate = Format_sampleRate(format); 3085 size_t frameSize = Format_frameSize(format); 3086 wavHeader[22] = channelCount; // number of channels 3087 wavHeader[24] = sampleRate; // sample rate 3088 wavHeader[25] = sampleRate >> 8; 3089 wavHeader[32] = frameSize; // block alignment 3090 wavHeader[33] = frameSize >> 8; 3091 write(teeFd, wavHeader, sizeof(wavHeader)); 3092 size_t total = 0; 3093 bool firstRead = true; 3094 #define TEE_SINK_READ 1024 // frames per I/O operation 3095 void *buffer = malloc(TEE_SINK_READ * frameSize); 3096 for (;;) { 3097 size_t count = TEE_SINK_READ; 3098 ssize_t actual = teeSource->read(buffer, count); 3099 bool wasFirstRead = firstRead; 3100 firstRead = false; 3101 if (actual <= 0) { 3102 if (actual == (ssize_t) OVERRUN && wasFirstRead) { 3103 continue; 3104 } 3105 break; 3106 } 3107 ALOG_ASSERT(actual <= (ssize_t)count); 3108 write(teeFd, buffer, actual * frameSize); 3109 total += actual; 3110 } 3111 free(buffer); 3112 lseek(teeFd, (off_t) 4, SEEK_SET); 3113 uint32_t temp = 44 + total * frameSize - 8; 3114 // FIXME not big-endian safe 3115 write(teeFd, &temp, sizeof(temp)); 3116 lseek(teeFd, (off_t) 40, SEEK_SET); 3117 temp = total * frameSize; 3118 // FIXME not big-endian safe 3119 write(teeFd, &temp, sizeof(temp)); 3120 close(teeFd); 3121 if (fd >= 0) { 3122 dprintf(fd, "tee copied to %s\n", teePath); 3123 } 3124 } else { 3125 if (fd >= 0) { 3126 dprintf(fd, "unable to create tee %s: %s\n", teePath, strerror(errno)); 3127 } 3128 } 3129 } 3130 } 3131 #endif 3132 3133 // ---------------------------------------------------------------------------- 3134 3135 status_t AudioFlinger::onTransact( 3136 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 3137 { 3138 return BnAudioFlinger::onTransact(code, data, reply, flags); 3139 } 3140 3141 } // namespace android 3142