AudioFlinger.cpp revision f517d276a8bfe4034106abf8009fcd6cc28a0838
1/* 2** 3** Copyright 2007, The Android Open Source Project 4** 5** Licensed under the Apache License, Version 2.0 (the "License"); 6** you may not use this file except in compliance with the License. 7** You may obtain a copy of the License at 8** 9** http://www.apache.org/licenses/LICENSE-2.0 10** 11** Unless required by applicable law or agreed to in writing, software 12** distributed under the License is distributed on an "AS IS" BASIS, 13** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. 14** See the License for the specific language governing permissions and 15** limitations under the License. 16*/ 17 18 19#define LOG_TAG "AudioFlinger" 20//#define LOG_NDEBUG 0 21 22#include "Configuration.h" 23#include <dirent.h> 24#include <math.h> 25#include <signal.h> 26#include <sys/time.h> 27#include <sys/resource.h> 28 29#include <binder/IPCThreadState.h> 30#include <binder/IServiceManager.h> 31#include <utils/Log.h> 32#include <utils/Trace.h> 33#include <binder/Parcel.h> 34#include <memunreachable/memunreachable.h> 35#include <utils/String16.h> 36#include <utils/threads.h> 37#include <utils/Atomic.h> 38 39#include <cutils/bitops.h> 40#include <cutils/properties.h> 41 42#include <system/audio.h> 43#include <hardware/audio.h> 44 45#include "AudioMixer.h" 46#include "AudioFlinger.h" 47#include "ServiceUtilities.h" 48 49#include <media/AudioResamplerPublic.h> 50 51#include <media/EffectsFactoryApi.h> 52#include <audio_effects/effect_visualizer.h> 53#include <audio_effects/effect_ns.h> 54#include <audio_effects/effect_aec.h> 55 56#include <audio_utils/primitives.h> 57 58#include <powermanager/PowerManager.h> 59 60#include <media/IMediaLogService.h> 61#include <media/MemoryLeakTrackUtil.h> 62#include <media/nbaio/Pipe.h> 63#include <media/nbaio/PipeReader.h> 64#include <media/AudioParameter.h> 65#include <mediautils/BatteryNotifier.h> 66#include <private/android_filesystem_config.h> 67 68// ---------------------------------------------------------------------------- 69 70// Note: the following macro is used for extremely verbose logging message. In 71// order to run with ALOG_ASSERT turned on, we need to have LOG_NDEBUG set to 72// 0; but one side effect of this is to turn all LOGV's as well. Some messages 73// are so verbose that we want to suppress them even when we have ALOG_ASSERT 74// turned on. Do not uncomment the #def below unless you really know what you 75// are doing and want to see all of the extremely verbose messages. 76//#define VERY_VERY_VERBOSE_LOGGING 77#ifdef VERY_VERY_VERBOSE_LOGGING 78#define ALOGVV ALOGV 79#else 80#define ALOGVV(a...) do { } while(0) 81#endif 82 83namespace android { 84 85static const char kDeadlockedString[] = "AudioFlinger may be deadlocked\n"; 86static const char kHardwareLockedString[] = "Hardware lock is taken\n"; 87static const char kClientLockedString[] = "Client lock is taken\n"; 88 89 90nsecs_t AudioFlinger::mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs; 91 92uint32_t AudioFlinger::mScreenState; 93 94#ifdef TEE_SINK 95bool AudioFlinger::mTeeSinkInputEnabled = false; 96bool AudioFlinger::mTeeSinkOutputEnabled = false; 97bool AudioFlinger::mTeeSinkTrackEnabled = false; 98 99size_t AudioFlinger::mTeeSinkInputFrames = kTeeSinkInputFramesDefault; 100size_t AudioFlinger::mTeeSinkOutputFrames = kTeeSinkOutputFramesDefault; 101size_t AudioFlinger::mTeeSinkTrackFrames = kTeeSinkTrackFramesDefault; 102#endif 103 104// In order to avoid invalidating offloaded tracks each time a Visualizer is turned on and off 105// we define a minimum time during which a global effect is considered enabled. 106static const nsecs_t kMinGlobalEffectEnabletimeNs = seconds(7200); 107 108// ---------------------------------------------------------------------------- 109 110const char *formatToString(audio_format_t format) { 111 switch (audio_get_main_format(format)) { 112 case AUDIO_FORMAT_PCM: 113 switch (format) { 114 case AUDIO_FORMAT_PCM_16_BIT: return "pcm16"; 115 case AUDIO_FORMAT_PCM_8_BIT: return "pcm8"; 116 case AUDIO_FORMAT_PCM_32_BIT: return "pcm32"; 117 case AUDIO_FORMAT_PCM_8_24_BIT: return "pcm8.24"; 118 case AUDIO_FORMAT_PCM_FLOAT: return "pcmfloat"; 119 case AUDIO_FORMAT_PCM_24_BIT_PACKED: return "pcm24"; 120 default: 121 break; 122 } 123 break; 124 case AUDIO_FORMAT_MP3: return "mp3"; 125 case AUDIO_FORMAT_AMR_NB: return "amr-nb"; 126 case AUDIO_FORMAT_AMR_WB: return "amr-wb"; 127 case AUDIO_FORMAT_AAC: return "aac"; 128 case AUDIO_FORMAT_HE_AAC_V1: return "he-aac-v1"; 129 case AUDIO_FORMAT_HE_AAC_V2: return "he-aac-v2"; 130 case AUDIO_FORMAT_VORBIS: return "vorbis"; 131 case AUDIO_FORMAT_OPUS: return "opus"; 132 case AUDIO_FORMAT_AC3: return "ac-3"; 133 case AUDIO_FORMAT_E_AC3: return "e-ac-3"; 134 case AUDIO_FORMAT_IEC61937: return "iec61937"; 135 case AUDIO_FORMAT_DTS: return "dts"; 136 case AUDIO_FORMAT_DTS_HD: return "dts-hd"; 137 case AUDIO_FORMAT_DOLBY_TRUEHD: return "dolby-truehd"; 138 default: 139 break; 140 } 141 return "unknown"; 142} 143 144static int load_audio_interface(const char *if_name, audio_hw_device_t **dev) 145{ 146 const hw_module_t *mod; 147 int rc; 148 149 rc = hw_get_module_by_class(AUDIO_HARDWARE_MODULE_ID, if_name, &mod); 150 ALOGE_IF(rc, "%s couldn't load audio hw module %s.%s (%s)", __func__, 151 AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc)); 152 if (rc) { 153 goto out; 154 } 155 rc = audio_hw_device_open(mod, dev); 156 ALOGE_IF(rc, "%s couldn't open audio hw device in %s.%s (%s)", __func__, 157 AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc)); 158 if (rc) { 159 goto out; 160 } 161 if ((*dev)->common.version < AUDIO_DEVICE_API_VERSION_MIN) { 162 ALOGE("%s wrong audio hw device version %04x", __func__, (*dev)->common.version); 163 rc = BAD_VALUE; 164 goto out; 165 } 166 return 0; 167 168out: 169 *dev = NULL; 170 return rc; 171} 172 173// ---------------------------------------------------------------------------- 174 175AudioFlinger::AudioFlinger() 176 : BnAudioFlinger(), 177 mPrimaryHardwareDev(NULL), 178 mAudioHwDevs(NULL), 179 mHardwareStatus(AUDIO_HW_IDLE), 180 mMasterVolume(1.0f), 181 mMasterMute(false), 182 // mNextUniqueId(AUDIO_UNIQUE_ID_USE_MAX), 183 mMode(AUDIO_MODE_INVALID), 184 mBtNrecIsOff(false), 185 mIsLowRamDevice(true), 186 mIsDeviceTypeKnown(false), 187 mGlobalEffectEnableTime(0), 188 mSystemReady(false) 189{ 190 // unsigned instead of audio_unique_id_use_t, because ++ operator is unavailable for enum 191 for (unsigned use = AUDIO_UNIQUE_ID_USE_UNSPECIFIED; use < AUDIO_UNIQUE_ID_USE_MAX; use++) { 192 // zero ID has a special meaning, so unavailable 193 mNextUniqueIds[use] = AUDIO_UNIQUE_ID_USE_MAX; 194 } 195 196 getpid_cached = getpid(); 197 const bool doLog = property_get_bool("ro.test_harness", false); 198 if (doLog) { 199 mLogMemoryDealer = new MemoryDealer(kLogMemorySize, "LogWriters", 200 MemoryHeapBase::READ_ONLY); 201 } 202 203 // reset battery stats. 204 // if the audio service has crashed, battery stats could be left 205 // in bad state, reset the state upon service start. 206 BatteryNotifier::getInstance().noteResetAudio(); 207 208#ifdef TEE_SINK 209 char value[PROPERTY_VALUE_MAX]; 210 (void) property_get("ro.debuggable", value, "0"); 211 int debuggable = atoi(value); 212 int teeEnabled = 0; 213 if (debuggable) { 214 (void) property_get("af.tee", value, "0"); 215 teeEnabled = atoi(value); 216 } 217 // FIXME symbolic constants here 218 if (teeEnabled & 1) { 219 mTeeSinkInputEnabled = true; 220 } 221 if (teeEnabled & 2) { 222 mTeeSinkOutputEnabled = true; 223 } 224 if (teeEnabled & 4) { 225 mTeeSinkTrackEnabled = true; 226 } 227#endif 228} 229 230void AudioFlinger::onFirstRef() 231{ 232 Mutex::Autolock _l(mLock); 233 234 /* TODO: move all this work into an Init() function */ 235 char val_str[PROPERTY_VALUE_MAX] = { 0 }; 236 if (property_get("ro.audio.flinger_standbytime_ms", val_str, NULL) >= 0) { 237 uint32_t int_val; 238 if (1 == sscanf(val_str, "%u", &int_val)) { 239 mStandbyTimeInNsecs = milliseconds(int_val); 240 ALOGI("Using %u mSec as standby time.", int_val); 241 } else { 242 mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs; 243 ALOGI("Using default %u mSec as standby time.", 244 (uint32_t)(mStandbyTimeInNsecs / 1000000)); 245 } 246 } 247 248 mPatchPanel = new PatchPanel(this); 249 250 mMode = AUDIO_MODE_NORMAL; 251} 252 253AudioFlinger::~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 284static 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 291AudioHwDevice* 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 321void 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 353void 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 368void 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 381bool 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 394status_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 502sp<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 516sp<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 } 553success: 554 mediaLogService->registerWriter(shared, size, name); 555 return new NBLog::Writer(size, shared); 556} 557 558void 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 576sp<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 729Exit: 730 *status = lStatus; 731 return trackHandle; 732} 733 734uint32_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 745audio_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 756size_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 769size_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 780uint32_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 791status_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 832status_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 866status_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 891bool 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 913status_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 954float AudioFlinger::masterVolume() const 955{ 956 Mutex::Autolock _l(mLock); 957 return masterVolume_l(); 958} 959 960bool AudioFlinger::masterMute() const 961{ 962 Mutex::Autolock _l(mLock); 963 return masterMute_l(); 964} 965 966float AudioFlinger::masterVolume_l() const 967{ 968 return mMasterVolume; 969} 970 971bool AudioFlinger::masterMute_l() const 972{ 973 return mMasterMute; 974} 975 976status_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 991status_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 1027status_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 1053float 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 1075bool 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 1087void 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 1094status_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 status_t final_result = NO_ERROR; 1108 { 1109 AutoMutex lock(mHardwareLock); 1110 mHardwareStatus = AUDIO_HW_SET_PARAMETER; 1111 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 1112 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 1113 status_t result = dev->set_parameters(dev, keyValuePairs.string()); 1114 final_result = result ?: final_result; 1115 } 1116 mHardwareStatus = AUDIO_HW_IDLE; 1117 } 1118 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 1119 AudioParameter param = AudioParameter(keyValuePairs); 1120 String8 value; 1121 if (param.get(String8(AUDIO_PARAMETER_KEY_BT_NREC), value) == NO_ERROR) { 1122 bool btNrecIsOff = (value == AUDIO_PARAMETER_VALUE_OFF); 1123 if (mBtNrecIsOff != btNrecIsOff) { 1124 for (size_t i = 0; i < mRecordThreads.size(); i++) { 1125 sp<RecordThread> thread = mRecordThreads.valueAt(i); 1126 audio_devices_t device = thread->inDevice(); 1127 bool suspend = audio_is_bluetooth_sco_device(device) && btNrecIsOff; 1128 // collect all of the thread's session IDs 1129 KeyedVector<audio_session_t, bool> ids = thread->sessionIds(); 1130 // suspend effects associated with those session IDs 1131 for (size_t j = 0; j < ids.size(); ++j) { 1132 audio_session_t sessionId = ids.keyAt(j); 1133 thread->setEffectSuspended(FX_IID_AEC, 1134 suspend, 1135 sessionId); 1136 thread->setEffectSuspended(FX_IID_NS, 1137 suspend, 1138 sessionId); 1139 } 1140 } 1141 mBtNrecIsOff = btNrecIsOff; 1142 } 1143 } 1144 String8 screenState; 1145 if (param.get(String8(AudioParameter::keyScreenState), screenState) == NO_ERROR) { 1146 bool isOff = screenState == "off"; 1147 if (isOff != (AudioFlinger::mScreenState & 1)) { 1148 AudioFlinger::mScreenState = ((AudioFlinger::mScreenState & ~1) + 2) | isOff; 1149 } 1150 } 1151 return final_result; 1152 } 1153 1154 // hold a strong ref on thread in case closeOutput() or closeInput() is called 1155 // and the thread is exited once the lock is released 1156 sp<ThreadBase> thread; 1157 { 1158 Mutex::Autolock _l(mLock); 1159 thread = checkPlaybackThread_l(ioHandle); 1160 if (thread == 0) { 1161 thread = checkRecordThread_l(ioHandle); 1162 } else if (thread == primaryPlaybackThread_l()) { 1163 // indicate output device change to all input threads for pre processing 1164 AudioParameter param = AudioParameter(keyValuePairs); 1165 int value; 1166 if ((param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) && 1167 (value != 0)) { 1168 broacastParametersToRecordThreads_l(keyValuePairs); 1169 } 1170 } 1171 } 1172 if (thread != 0) { 1173 return thread->setParameters(keyValuePairs); 1174 } 1175 return BAD_VALUE; 1176} 1177 1178String8 AudioFlinger::getParameters(audio_io_handle_t ioHandle, const String8& keys) const 1179{ 1180 ALOGVV("getParameters() io %d, keys %s, calling pid %d", 1181 ioHandle, keys.string(), IPCThreadState::self()->getCallingPid()); 1182 1183 Mutex::Autolock _l(mLock); 1184 1185 if (ioHandle == AUDIO_IO_HANDLE_NONE) { 1186 String8 out_s8; 1187 1188 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 1189 char *s; 1190 { 1191 AutoMutex lock(mHardwareLock); 1192 mHardwareStatus = AUDIO_HW_GET_PARAMETER; 1193 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 1194 s = dev->get_parameters(dev, keys.string()); 1195 mHardwareStatus = AUDIO_HW_IDLE; 1196 } 1197 out_s8 += String8(s ? s : ""); 1198 free(s); 1199 } 1200 return out_s8; 1201 } 1202 1203 PlaybackThread *playbackThread = checkPlaybackThread_l(ioHandle); 1204 if (playbackThread != NULL) { 1205 return playbackThread->getParameters(keys); 1206 } 1207 RecordThread *recordThread = checkRecordThread_l(ioHandle); 1208 if (recordThread != NULL) { 1209 return recordThread->getParameters(keys); 1210 } 1211 return String8(""); 1212} 1213 1214size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, audio_format_t format, 1215 audio_channel_mask_t channelMask) const 1216{ 1217 status_t ret = initCheck(); 1218 if (ret != NO_ERROR) { 1219 return 0; 1220 } 1221 if ((sampleRate == 0) || 1222 !audio_is_valid_format(format) || !audio_has_proportional_frames(format) || 1223 !audio_is_input_channel(channelMask)) { 1224 return 0; 1225 } 1226 1227 AutoMutex lock(mHardwareLock); 1228 mHardwareStatus = AUDIO_HW_GET_INPUT_BUFFER_SIZE; 1229 audio_config_t config, proposed; 1230 memset(&proposed, 0, sizeof(proposed)); 1231 proposed.sample_rate = sampleRate; 1232 proposed.channel_mask = channelMask; 1233 proposed.format = format; 1234 1235 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 1236 size_t frames; 1237 for (;;) { 1238 // Note: config is currently a const parameter for get_input_buffer_size() 1239 // but we use a copy from proposed in case config changes from the call. 1240 config = proposed; 1241 frames = dev->get_input_buffer_size(dev, &config); 1242 if (frames != 0) { 1243 break; // hal success, config is the result 1244 } 1245 // change one parameter of the configuration each iteration to a more "common" value 1246 // to see if the device will support it. 1247 if (proposed.format != AUDIO_FORMAT_PCM_16_BIT) { 1248 proposed.format = AUDIO_FORMAT_PCM_16_BIT; 1249 } else if (proposed.sample_rate != 44100) { // 44.1 is claimed as must in CDD as well as 1250 proposed.sample_rate = 44100; // legacy AudioRecord.java. TODO: Query hw? 1251 } else { 1252 ALOGW("getInputBufferSize failed with minimum buffer size sampleRate %u, " 1253 "format %#x, channelMask 0x%X", 1254 sampleRate, format, channelMask); 1255 break; // retries failed, break out of loop with frames == 0. 1256 } 1257 } 1258 mHardwareStatus = AUDIO_HW_IDLE; 1259 if (frames > 0 && config.sample_rate != sampleRate) { 1260 frames = destinationFramesPossible(frames, sampleRate, config.sample_rate); 1261 } 1262 return frames; // may be converted to bytes at the Java level. 1263} 1264 1265uint32_t AudioFlinger::getInputFramesLost(audio_io_handle_t ioHandle) const 1266{ 1267 Mutex::Autolock _l(mLock); 1268 1269 RecordThread *recordThread = checkRecordThread_l(ioHandle); 1270 if (recordThread != NULL) { 1271 return recordThread->getInputFramesLost(); 1272 } 1273 return 0; 1274} 1275 1276status_t AudioFlinger::setVoiceVolume(float value) 1277{ 1278 status_t ret = initCheck(); 1279 if (ret != NO_ERROR) { 1280 return ret; 1281 } 1282 1283 // check calling permissions 1284 if (!settingsAllowed()) { 1285 return PERMISSION_DENIED; 1286 } 1287 1288 AutoMutex lock(mHardwareLock); 1289 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 1290 mHardwareStatus = AUDIO_HW_SET_VOICE_VOLUME; 1291 ret = dev->set_voice_volume(dev, value); 1292 mHardwareStatus = AUDIO_HW_IDLE; 1293 1294 return ret; 1295} 1296 1297status_t AudioFlinger::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, 1298 audio_io_handle_t output) const 1299{ 1300 Mutex::Autolock _l(mLock); 1301 1302 PlaybackThread *playbackThread = checkPlaybackThread_l(output); 1303 if (playbackThread != NULL) { 1304 return playbackThread->getRenderPosition(halFrames, dspFrames); 1305 } 1306 1307 return BAD_VALUE; 1308} 1309 1310void AudioFlinger::registerClient(const sp<IAudioFlingerClient>& client) 1311{ 1312 Mutex::Autolock _l(mLock); 1313 if (client == 0) { 1314 return; 1315 } 1316 pid_t pid = IPCThreadState::self()->getCallingPid(); 1317 { 1318 Mutex::Autolock _cl(mClientLock); 1319 if (mNotificationClients.indexOfKey(pid) < 0) { 1320 sp<NotificationClient> notificationClient = new NotificationClient(this, 1321 client, 1322 pid); 1323 ALOGV("registerClient() client %p, pid %d", notificationClient.get(), pid); 1324 1325 mNotificationClients.add(pid, notificationClient); 1326 1327 sp<IBinder> binder = IInterface::asBinder(client); 1328 binder->linkToDeath(notificationClient); 1329 } 1330 } 1331 1332 // mClientLock should not be held here because ThreadBase::sendIoConfigEvent() will lock the 1333 // ThreadBase mutex and the locking order is ThreadBase::mLock then AudioFlinger::mClientLock. 1334 // the config change is always sent from playback or record threads to avoid deadlock 1335 // with AudioSystem::gLock 1336 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1337 mPlaybackThreads.valueAt(i)->sendIoConfigEvent(AUDIO_OUTPUT_OPENED, pid); 1338 } 1339 1340 for (size_t i = 0; i < mRecordThreads.size(); i++) { 1341 mRecordThreads.valueAt(i)->sendIoConfigEvent(AUDIO_INPUT_OPENED, pid); 1342 } 1343} 1344 1345void AudioFlinger::removeNotificationClient(pid_t pid) 1346{ 1347 Mutex::Autolock _l(mLock); 1348 { 1349 Mutex::Autolock _cl(mClientLock); 1350 mNotificationClients.removeItem(pid); 1351 } 1352 1353 ALOGV("%d died, releasing its sessions", pid); 1354 size_t num = mAudioSessionRefs.size(); 1355 bool removed = false; 1356 for (size_t i = 0; i< num; ) { 1357 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 1358 ALOGV(" pid %d @ %zu", ref->mPid, i); 1359 if (ref->mPid == pid) { 1360 ALOGV(" removing entry for pid %d session %d", pid, ref->mSessionid); 1361 mAudioSessionRefs.removeAt(i); 1362 delete ref; 1363 removed = true; 1364 num--; 1365 } else { 1366 i++; 1367 } 1368 } 1369 if (removed) { 1370 purgeStaleEffects_l(); 1371 } 1372} 1373 1374void AudioFlinger::ioConfigChanged(audio_io_config_event event, 1375 const sp<AudioIoDescriptor>& ioDesc, 1376 pid_t pid) 1377{ 1378 Mutex::Autolock _l(mClientLock); 1379 size_t size = mNotificationClients.size(); 1380 for (size_t i = 0; i < size; i++) { 1381 if ((pid == 0) || (mNotificationClients.keyAt(i) == pid)) { 1382 mNotificationClients.valueAt(i)->audioFlingerClient()->ioConfigChanged(event, ioDesc); 1383 } 1384 } 1385} 1386 1387// removeClient_l() must be called with AudioFlinger::mClientLock held 1388void AudioFlinger::removeClient_l(pid_t pid) 1389{ 1390 ALOGV("removeClient_l() pid %d, calling pid %d", pid, 1391 IPCThreadState::self()->getCallingPid()); 1392 mClients.removeItem(pid); 1393} 1394 1395// getEffectThread_l() must be called with AudioFlinger::mLock held 1396sp<AudioFlinger::PlaybackThread> AudioFlinger::getEffectThread_l(audio_session_t sessionId, 1397 int EffectId) 1398{ 1399 sp<PlaybackThread> thread; 1400 1401 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1402 if (mPlaybackThreads.valueAt(i)->getEffect(sessionId, EffectId) != 0) { 1403 ALOG_ASSERT(thread == 0); 1404 thread = mPlaybackThreads.valueAt(i); 1405 } 1406 } 1407 1408 return thread; 1409} 1410 1411 1412 1413// ---------------------------------------------------------------------------- 1414 1415AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid) 1416 : RefBase(), 1417 mAudioFlinger(audioFlinger), 1418 mPid(pid) 1419{ 1420 size_t heapSize = property_get_int32("ro.af.client_heap_size_kbyte", 0); 1421 heapSize *= 1024; 1422 if (!heapSize) { 1423 heapSize = kClientSharedHeapSizeBytes; 1424 // Increase heap size on non low ram devices to limit risk of reconnection failure for 1425 // invalidated tracks 1426 if (!audioFlinger->isLowRamDevice()) { 1427 heapSize *= kClientSharedHeapSizeMultiplier; 1428 } 1429 } 1430 mMemoryDealer = new MemoryDealer(heapSize, "AudioFlinger::Client"); 1431} 1432 1433// Client destructor must be called with AudioFlinger::mClientLock held 1434AudioFlinger::Client::~Client() 1435{ 1436 mAudioFlinger->removeClient_l(mPid); 1437} 1438 1439sp<MemoryDealer> AudioFlinger::Client::heap() const 1440{ 1441 return mMemoryDealer; 1442} 1443 1444// ---------------------------------------------------------------------------- 1445 1446AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger, 1447 const sp<IAudioFlingerClient>& client, 1448 pid_t pid) 1449 : mAudioFlinger(audioFlinger), mPid(pid), mAudioFlingerClient(client) 1450{ 1451} 1452 1453AudioFlinger::NotificationClient::~NotificationClient() 1454{ 1455} 1456 1457void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who __unused) 1458{ 1459 sp<NotificationClient> keep(this); 1460 mAudioFlinger->removeNotificationClient(mPid); 1461} 1462 1463 1464// ---------------------------------------------------------------------------- 1465 1466sp<IAudioRecord> AudioFlinger::openRecord( 1467 audio_io_handle_t input, 1468 uint32_t sampleRate, 1469 audio_format_t format, 1470 audio_channel_mask_t channelMask, 1471 const String16& opPackageName, 1472 size_t *frameCount, 1473 audio_input_flags_t *flags, 1474 pid_t pid, 1475 pid_t tid, 1476 int clientUid, 1477 audio_session_t *sessionId, 1478 size_t *notificationFrames, 1479 sp<IMemory>& cblk, 1480 sp<IMemory>& buffers, 1481 status_t *status) 1482{ 1483 sp<RecordThread::RecordTrack> recordTrack; 1484 sp<RecordHandle> recordHandle; 1485 sp<Client> client; 1486 status_t lStatus; 1487 audio_session_t lSessionId; 1488 1489 cblk.clear(); 1490 buffers.clear(); 1491 1492 bool updatePid = (pid == -1); 1493 const uid_t callingUid = IPCThreadState::self()->getCallingUid(); 1494 if (!isTrustedCallingUid(callingUid)) { 1495 ALOGW_IF((uid_t)clientUid != callingUid, 1496 "%s uid %d tried to pass itself off as %d", __FUNCTION__, callingUid, clientUid); 1497 clientUid = callingUid; 1498 updatePid = true; 1499 } 1500 1501 if (updatePid) { 1502 const pid_t callingPid = IPCThreadState::self()->getCallingPid(); 1503 ALOGW_IF(pid != -1 && pid != callingPid, 1504 "%s uid %d pid %d tried to pass itself off as pid %d", 1505 __func__, callingUid, callingPid, pid); 1506 pid = callingPid; 1507 } 1508 1509 // check calling permissions 1510 if (!recordingAllowed(opPackageName, tid, clientUid)) { 1511 ALOGE("openRecord() permission denied: recording not allowed"); 1512 lStatus = PERMISSION_DENIED; 1513 goto Exit; 1514 } 1515 1516 // further sample rate checks are performed by createRecordTrack_l() 1517 if (sampleRate == 0) { 1518 ALOGE("openRecord() invalid sample rate %u", sampleRate); 1519 lStatus = BAD_VALUE; 1520 goto Exit; 1521 } 1522 1523 // we don't yet support anything other than linear PCM 1524 if (!audio_is_valid_format(format) || !audio_is_linear_pcm(format)) { 1525 ALOGE("openRecord() invalid format %#x", format); 1526 lStatus = BAD_VALUE; 1527 goto Exit; 1528 } 1529 1530 // further channel mask checks are performed by createRecordTrack_l() 1531 if (!audio_is_input_channel(channelMask)) { 1532 ALOGE("openRecord() invalid channel mask %#x", channelMask); 1533 lStatus = BAD_VALUE; 1534 goto Exit; 1535 } 1536 1537 { 1538 Mutex::Autolock _l(mLock); 1539 RecordThread *thread = checkRecordThread_l(input); 1540 if (thread == NULL) { 1541 ALOGE("openRecord() checkRecordThread_l failed"); 1542 lStatus = BAD_VALUE; 1543 goto Exit; 1544 } 1545 1546 client = registerPid(pid); 1547 1548 if (sessionId != NULL && *sessionId != AUDIO_SESSION_ALLOCATE) { 1549 if (audio_unique_id_get_use(*sessionId) != AUDIO_UNIQUE_ID_USE_SESSION) { 1550 lStatus = BAD_VALUE; 1551 goto Exit; 1552 } 1553 lSessionId = *sessionId; 1554 } else { 1555 // if no audio session id is provided, create one here 1556 lSessionId = (audio_session_t) nextUniqueId(AUDIO_UNIQUE_ID_USE_SESSION); 1557 if (sessionId != NULL) { 1558 *sessionId = lSessionId; 1559 } 1560 } 1561 ALOGV("openRecord() lSessionId: %d input %d", lSessionId, input); 1562 1563 recordTrack = thread->createRecordTrack_l(client, sampleRate, format, channelMask, 1564 frameCount, lSessionId, notificationFrames, 1565 clientUid, flags, tid, &lStatus); 1566 LOG_ALWAYS_FATAL_IF((lStatus == NO_ERROR) && (recordTrack == 0)); 1567 1568 if (lStatus == NO_ERROR) { 1569 // Check if one effect chain was awaiting for an AudioRecord to be created on this 1570 // session and move it to this thread. 1571 sp<EffectChain> chain = getOrphanEffectChain_l(lSessionId); 1572 if (chain != 0) { 1573 Mutex::Autolock _l(thread->mLock); 1574 thread->addEffectChain_l(chain); 1575 } 1576 } 1577 } 1578 1579 if (lStatus != NO_ERROR) { 1580 // remove local strong reference to Client before deleting the RecordTrack so that the 1581 // Client destructor is called by the TrackBase destructor with mClientLock held 1582 // Don't hold mClientLock when releasing the reference on the track as the 1583 // destructor will acquire it. 1584 { 1585 Mutex::Autolock _cl(mClientLock); 1586 client.clear(); 1587 } 1588 recordTrack.clear(); 1589 goto Exit; 1590 } 1591 1592 cblk = recordTrack->getCblk(); 1593 buffers = recordTrack->getBuffers(); 1594 1595 // return handle to client 1596 recordHandle = new RecordHandle(recordTrack); 1597 1598Exit: 1599 *status = lStatus; 1600 return recordHandle; 1601} 1602 1603 1604 1605// ---------------------------------------------------------------------------- 1606 1607audio_module_handle_t AudioFlinger::loadHwModule(const char *name) 1608{ 1609 if (name == NULL) { 1610 return AUDIO_MODULE_HANDLE_NONE; 1611 } 1612 if (!settingsAllowed()) { 1613 return AUDIO_MODULE_HANDLE_NONE; 1614 } 1615 Mutex::Autolock _l(mLock); 1616 return loadHwModule_l(name); 1617} 1618 1619// loadHwModule_l() must be called with AudioFlinger::mLock held 1620audio_module_handle_t AudioFlinger::loadHwModule_l(const char *name) 1621{ 1622 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 1623 if (strncmp(mAudioHwDevs.valueAt(i)->moduleName(), name, strlen(name)) == 0) { 1624 ALOGW("loadHwModule() module %s already loaded", name); 1625 return mAudioHwDevs.keyAt(i); 1626 } 1627 } 1628 1629 audio_hw_device_t *dev; 1630 1631 int rc = load_audio_interface(name, &dev); 1632 if (rc) { 1633 ALOGE("loadHwModule() error %d loading module %s", rc, name); 1634 return AUDIO_MODULE_HANDLE_NONE; 1635 } 1636 1637 mHardwareStatus = AUDIO_HW_INIT; 1638 rc = dev->init_check(dev); 1639 mHardwareStatus = AUDIO_HW_IDLE; 1640 if (rc) { 1641 ALOGE("loadHwModule() init check error %d for module %s", rc, name); 1642 return AUDIO_MODULE_HANDLE_NONE; 1643 } 1644 1645 // Check and cache this HAL's level of support for master mute and master 1646 // volume. If this is the first HAL opened, and it supports the get 1647 // methods, use the initial values provided by the HAL as the current 1648 // master mute and volume settings. 1649 1650 AudioHwDevice::Flags flags = static_cast<AudioHwDevice::Flags>(0); 1651 { // scope for auto-lock pattern 1652 AutoMutex lock(mHardwareLock); 1653 1654 if (0 == mAudioHwDevs.size()) { 1655 mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME; 1656 if (NULL != dev->get_master_volume) { 1657 float mv; 1658 if (OK == dev->get_master_volume(dev, &mv)) { 1659 mMasterVolume = mv; 1660 } 1661 } 1662 1663 mHardwareStatus = AUDIO_HW_GET_MASTER_MUTE; 1664 if (NULL != dev->get_master_mute) { 1665 bool mm; 1666 if (OK == dev->get_master_mute(dev, &mm)) { 1667 mMasterMute = mm; 1668 } 1669 } 1670 } 1671 1672 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 1673 if ((NULL != dev->set_master_volume) && 1674 (OK == dev->set_master_volume(dev, mMasterVolume))) { 1675 flags = static_cast<AudioHwDevice::Flags>(flags | 1676 AudioHwDevice::AHWD_CAN_SET_MASTER_VOLUME); 1677 } 1678 1679 mHardwareStatus = AUDIO_HW_SET_MASTER_MUTE; 1680 if ((NULL != dev->set_master_mute) && 1681 (OK == dev->set_master_mute(dev, mMasterMute))) { 1682 flags = static_cast<AudioHwDevice::Flags>(flags | 1683 AudioHwDevice::AHWD_CAN_SET_MASTER_MUTE); 1684 } 1685 1686 mHardwareStatus = AUDIO_HW_IDLE; 1687 } 1688 1689 audio_module_handle_t handle = (audio_module_handle_t) nextUniqueId(AUDIO_UNIQUE_ID_USE_MODULE); 1690 mAudioHwDevs.add(handle, new AudioHwDevice(handle, name, dev, flags)); 1691 1692 ALOGI("loadHwModule() Loaded %s audio interface from %s (%s) handle %d", 1693 name, dev->common.module->name, dev->common.module->id, handle); 1694 1695 return handle; 1696 1697} 1698 1699// ---------------------------------------------------------------------------- 1700 1701uint32_t AudioFlinger::getPrimaryOutputSamplingRate() 1702{ 1703 Mutex::Autolock _l(mLock); 1704 PlaybackThread *thread = fastPlaybackThread_l(); 1705 return thread != NULL ? thread->sampleRate() : 0; 1706} 1707 1708size_t AudioFlinger::getPrimaryOutputFrameCount() 1709{ 1710 Mutex::Autolock _l(mLock); 1711 PlaybackThread *thread = fastPlaybackThread_l(); 1712 return thread != NULL ? thread->frameCountHAL() : 0; 1713} 1714 1715// ---------------------------------------------------------------------------- 1716 1717status_t AudioFlinger::setLowRamDevice(bool isLowRamDevice) 1718{ 1719 uid_t uid = IPCThreadState::self()->getCallingUid(); 1720 if (uid != AID_SYSTEM) { 1721 return PERMISSION_DENIED; 1722 } 1723 Mutex::Autolock _l(mLock); 1724 if (mIsDeviceTypeKnown) { 1725 return INVALID_OPERATION; 1726 } 1727 mIsLowRamDevice = isLowRamDevice; 1728 mIsDeviceTypeKnown = true; 1729 return NO_ERROR; 1730} 1731 1732audio_hw_sync_t AudioFlinger::getAudioHwSyncForSession(audio_session_t sessionId) 1733{ 1734 Mutex::Autolock _l(mLock); 1735 1736 ssize_t index = mHwAvSyncIds.indexOfKey(sessionId); 1737 if (index >= 0) { 1738 ALOGV("getAudioHwSyncForSession found ID %d for session %d", 1739 mHwAvSyncIds.valueAt(index), sessionId); 1740 return mHwAvSyncIds.valueAt(index); 1741 } 1742 1743 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 1744 if (dev == NULL) { 1745 return AUDIO_HW_SYNC_INVALID; 1746 } 1747 char *reply = dev->get_parameters(dev, AUDIO_PARAMETER_HW_AV_SYNC); 1748 AudioParameter param = AudioParameter(String8(reply)); 1749 free(reply); 1750 1751 int value; 1752 if (param.getInt(String8(AUDIO_PARAMETER_HW_AV_SYNC), value) != NO_ERROR) { 1753 ALOGW("getAudioHwSyncForSession error getting sync for session %d", sessionId); 1754 return AUDIO_HW_SYNC_INVALID; 1755 } 1756 1757 // allow only one session for a given HW A/V sync ID. 1758 for (size_t i = 0; i < mHwAvSyncIds.size(); i++) { 1759 if (mHwAvSyncIds.valueAt(i) == (audio_hw_sync_t)value) { 1760 ALOGV("getAudioHwSyncForSession removing ID %d for session %d", 1761 value, mHwAvSyncIds.keyAt(i)); 1762 mHwAvSyncIds.removeItemsAt(i); 1763 break; 1764 } 1765 } 1766 1767 mHwAvSyncIds.add(sessionId, value); 1768 1769 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1770 sp<PlaybackThread> thread = mPlaybackThreads.valueAt(i); 1771 uint32_t sessions = thread->hasAudioSession(sessionId); 1772 if (sessions & ThreadBase::TRACK_SESSION) { 1773 AudioParameter param = AudioParameter(); 1774 param.addInt(String8(AUDIO_PARAMETER_STREAM_HW_AV_SYNC), value); 1775 thread->setParameters(param.toString()); 1776 break; 1777 } 1778 } 1779 1780 ALOGV("getAudioHwSyncForSession adding ID %d for session %d", value, sessionId); 1781 return (audio_hw_sync_t)value; 1782} 1783 1784status_t AudioFlinger::systemReady() 1785{ 1786 Mutex::Autolock _l(mLock); 1787 ALOGI("%s", __FUNCTION__); 1788 if (mSystemReady) { 1789 ALOGW("%s called twice", __FUNCTION__); 1790 return NO_ERROR; 1791 } 1792 mSystemReady = true; 1793 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1794 ThreadBase *thread = (ThreadBase *)mPlaybackThreads.valueAt(i).get(); 1795 thread->systemReady(); 1796 } 1797 for (size_t i = 0; i < mRecordThreads.size(); i++) { 1798 ThreadBase *thread = (ThreadBase *)mRecordThreads.valueAt(i).get(); 1799 thread->systemReady(); 1800 } 1801 return NO_ERROR; 1802} 1803 1804// setAudioHwSyncForSession_l() must be called with AudioFlinger::mLock held 1805void AudioFlinger::setAudioHwSyncForSession_l(PlaybackThread *thread, audio_session_t sessionId) 1806{ 1807 ssize_t index = mHwAvSyncIds.indexOfKey(sessionId); 1808 if (index >= 0) { 1809 audio_hw_sync_t syncId = mHwAvSyncIds.valueAt(index); 1810 ALOGV("setAudioHwSyncForSession_l found ID %d for session %d", syncId, sessionId); 1811 AudioParameter param = AudioParameter(); 1812 param.addInt(String8(AUDIO_PARAMETER_STREAM_HW_AV_SYNC), syncId); 1813 thread->setParameters(param.toString()); 1814 } 1815} 1816 1817 1818// ---------------------------------------------------------------------------- 1819 1820 1821sp<AudioFlinger::PlaybackThread> AudioFlinger::openOutput_l(audio_module_handle_t module, 1822 audio_io_handle_t *output, 1823 audio_config_t *config, 1824 audio_devices_t devices, 1825 const String8& address, 1826 audio_output_flags_t flags) 1827{ 1828 AudioHwDevice *outHwDev = findSuitableHwDev_l(module, devices); 1829 if (outHwDev == NULL) { 1830 return 0; 1831 } 1832 1833 if (*output == AUDIO_IO_HANDLE_NONE) { 1834 *output = nextUniqueId(AUDIO_UNIQUE_ID_USE_OUTPUT); 1835 } else { 1836 // Audio Policy does not currently request a specific output handle. 1837 // If this is ever needed, see openInput_l() for example code. 1838 ALOGE("openOutput_l requested output handle %d is not AUDIO_IO_HANDLE_NONE", *output); 1839 return 0; 1840 } 1841 1842 mHardwareStatus = AUDIO_HW_OUTPUT_OPEN; 1843 1844 // FOR TESTING ONLY: 1845 // This if statement allows overriding the audio policy settings 1846 // and forcing a specific format or channel mask to the HAL/Sink device for testing. 1847 if (!(flags & (AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD | AUDIO_OUTPUT_FLAG_DIRECT))) { 1848 // Check only for Normal Mixing mode 1849 if (kEnableExtendedPrecision) { 1850 // Specify format (uncomment one below to choose) 1851 //config->format = AUDIO_FORMAT_PCM_FLOAT; 1852 //config->format = AUDIO_FORMAT_PCM_24_BIT_PACKED; 1853 //config->format = AUDIO_FORMAT_PCM_32_BIT; 1854 //config->format = AUDIO_FORMAT_PCM_8_24_BIT; 1855 // ALOGV("openOutput_l() upgrading format to %#08x", config->format); 1856 } 1857 if (kEnableExtendedChannels) { 1858 // Specify channel mask (uncomment one below to choose) 1859 //config->channel_mask = audio_channel_out_mask_from_count(4); // for USB 4ch 1860 //config->channel_mask = audio_channel_mask_from_representation_and_bits( 1861 // AUDIO_CHANNEL_REPRESENTATION_INDEX, (1 << 4) - 1); // another 4ch example 1862 } 1863 } 1864 1865 AudioStreamOut *outputStream = NULL; 1866 status_t status = outHwDev->openOutputStream( 1867 &outputStream, 1868 *output, 1869 devices, 1870 flags, 1871 config, 1872 address.string()); 1873 1874 mHardwareStatus = AUDIO_HW_IDLE; 1875 1876 if (status == NO_ERROR) { 1877 1878 PlaybackThread *thread; 1879 if (flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) { 1880 thread = new OffloadThread(this, outputStream, *output, devices, mSystemReady); 1881 ALOGV("openOutput_l() created offload output: ID %d thread %p", *output, thread); 1882 } else if ((flags & AUDIO_OUTPUT_FLAG_DIRECT) 1883 || !isValidPcmSinkFormat(config->format) 1884 || !isValidPcmSinkChannelMask(config->channel_mask)) { 1885 thread = new DirectOutputThread(this, outputStream, *output, devices, mSystemReady); 1886 ALOGV("openOutput_l() created direct output: ID %d thread %p", *output, thread); 1887 } else { 1888 thread = new MixerThread(this, outputStream, *output, devices, mSystemReady); 1889 ALOGV("openOutput_l() created mixer output: ID %d thread %p", *output, thread); 1890 } 1891 mPlaybackThreads.add(*output, thread); 1892 return thread; 1893 } 1894 1895 return 0; 1896} 1897 1898status_t AudioFlinger::openOutput(audio_module_handle_t module, 1899 audio_io_handle_t *output, 1900 audio_config_t *config, 1901 audio_devices_t *devices, 1902 const String8& address, 1903 uint32_t *latencyMs, 1904 audio_output_flags_t flags) 1905{ 1906 ALOGI("openOutput(), module %d Device %x, SamplingRate %d, Format %#08x, Channels %x, flags %x", 1907 module, 1908 (devices != NULL) ? *devices : 0, 1909 config->sample_rate, 1910 config->format, 1911 config->channel_mask, 1912 flags); 1913 1914 if (*devices == AUDIO_DEVICE_NONE) { 1915 return BAD_VALUE; 1916 } 1917 1918 Mutex::Autolock _l(mLock); 1919 1920 sp<PlaybackThread> thread = openOutput_l(module, output, config, *devices, address, flags); 1921 if (thread != 0) { 1922 *latencyMs = thread->latency(); 1923 1924 // notify client processes of the new output creation 1925 thread->ioConfigChanged(AUDIO_OUTPUT_OPENED); 1926 1927 // the first primary output opened designates the primary hw device 1928 if ((mPrimaryHardwareDev == NULL) && (flags & AUDIO_OUTPUT_FLAG_PRIMARY)) { 1929 ALOGI("Using module %d has the primary audio interface", module); 1930 mPrimaryHardwareDev = thread->getOutput()->audioHwDev; 1931 1932 AutoMutex lock(mHardwareLock); 1933 mHardwareStatus = AUDIO_HW_SET_MODE; 1934 mPrimaryHardwareDev->hwDevice()->set_mode(mPrimaryHardwareDev->hwDevice(), mMode); 1935 mHardwareStatus = AUDIO_HW_IDLE; 1936 } 1937 return NO_ERROR; 1938 } 1939 1940 return NO_INIT; 1941} 1942 1943audio_io_handle_t AudioFlinger::openDuplicateOutput(audio_io_handle_t output1, 1944 audio_io_handle_t output2) 1945{ 1946 Mutex::Autolock _l(mLock); 1947 MixerThread *thread1 = checkMixerThread_l(output1); 1948 MixerThread *thread2 = checkMixerThread_l(output2); 1949 1950 if (thread1 == NULL || thread2 == NULL) { 1951 ALOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1, 1952 output2); 1953 return AUDIO_IO_HANDLE_NONE; 1954 } 1955 1956 audio_io_handle_t id = nextUniqueId(AUDIO_UNIQUE_ID_USE_OUTPUT); 1957 DuplicatingThread *thread = new DuplicatingThread(this, thread1, id, mSystemReady); 1958 thread->addOutputTrack(thread2); 1959 mPlaybackThreads.add(id, thread); 1960 // notify client processes of the new output creation 1961 thread->ioConfigChanged(AUDIO_OUTPUT_OPENED); 1962 return id; 1963} 1964 1965status_t AudioFlinger::closeOutput(audio_io_handle_t output) 1966{ 1967 return closeOutput_nonvirtual(output); 1968} 1969 1970status_t AudioFlinger::closeOutput_nonvirtual(audio_io_handle_t output) 1971{ 1972 // keep strong reference on the playback thread so that 1973 // it is not destroyed while exit() is executed 1974 sp<PlaybackThread> thread; 1975 { 1976 Mutex::Autolock _l(mLock); 1977 thread = checkPlaybackThread_l(output); 1978 if (thread == NULL) { 1979 return BAD_VALUE; 1980 } 1981 1982 ALOGV("closeOutput() %d", output); 1983 1984 if (thread->type() == ThreadBase::MIXER) { 1985 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1986 if (mPlaybackThreads.valueAt(i)->isDuplicating()) { 1987 DuplicatingThread *dupThread = 1988 (DuplicatingThread *)mPlaybackThreads.valueAt(i).get(); 1989 dupThread->removeOutputTrack((MixerThread *)thread.get()); 1990 } 1991 } 1992 } 1993 1994 1995 mPlaybackThreads.removeItem(output); 1996 // save all effects to the default thread 1997 if (mPlaybackThreads.size()) { 1998 PlaybackThread *dstThread = checkPlaybackThread_l(mPlaybackThreads.keyAt(0)); 1999 if (dstThread != NULL) { 2000 // audioflinger lock is held here so the acquisition order of thread locks does not 2001 // matter 2002 Mutex::Autolock _dl(dstThread->mLock); 2003 Mutex::Autolock _sl(thread->mLock); 2004 Vector< sp<EffectChain> > effectChains = thread->getEffectChains_l(); 2005 for (size_t i = 0; i < effectChains.size(); i ++) { 2006 moveEffectChain_l(effectChains[i]->sessionId(), thread.get(), dstThread, true); 2007 } 2008 } 2009 } 2010 const sp<AudioIoDescriptor> ioDesc = new AudioIoDescriptor(); 2011 ioDesc->mIoHandle = output; 2012 ioConfigChanged(AUDIO_OUTPUT_CLOSED, ioDesc); 2013 } 2014 thread->exit(); 2015 // The thread entity (active unit of execution) is no longer running here, 2016 // but the ThreadBase container still exists. 2017 2018 if (!thread->isDuplicating()) { 2019 closeOutputFinish(thread); 2020 } 2021 2022 return NO_ERROR; 2023} 2024 2025void AudioFlinger::closeOutputFinish(sp<PlaybackThread> thread) 2026{ 2027 AudioStreamOut *out = thread->clearOutput(); 2028 ALOG_ASSERT(out != NULL, "out shouldn't be NULL"); 2029 // from now on thread->mOutput is NULL 2030 out->hwDev()->close_output_stream(out->hwDev(), out->stream); 2031 delete out; 2032} 2033 2034void AudioFlinger::closeOutputInternal_l(sp<PlaybackThread> thread) 2035{ 2036 mPlaybackThreads.removeItem(thread->mId); 2037 thread->exit(); 2038 closeOutputFinish(thread); 2039} 2040 2041status_t AudioFlinger::suspendOutput(audio_io_handle_t output) 2042{ 2043 Mutex::Autolock _l(mLock); 2044 PlaybackThread *thread = checkPlaybackThread_l(output); 2045 2046 if (thread == NULL) { 2047 return BAD_VALUE; 2048 } 2049 2050 ALOGV("suspendOutput() %d", output); 2051 thread->suspend(); 2052 2053 return NO_ERROR; 2054} 2055 2056status_t AudioFlinger::restoreOutput(audio_io_handle_t output) 2057{ 2058 Mutex::Autolock _l(mLock); 2059 PlaybackThread *thread = checkPlaybackThread_l(output); 2060 2061 if (thread == NULL) { 2062 return BAD_VALUE; 2063 } 2064 2065 ALOGV("restoreOutput() %d", output); 2066 2067 thread->restore(); 2068 2069 return NO_ERROR; 2070} 2071 2072status_t AudioFlinger::openInput(audio_module_handle_t module, 2073 audio_io_handle_t *input, 2074 audio_config_t *config, 2075 audio_devices_t *devices, 2076 const String8& address, 2077 audio_source_t source, 2078 audio_input_flags_t flags) 2079{ 2080 Mutex::Autolock _l(mLock); 2081 2082 if (*devices == AUDIO_DEVICE_NONE) { 2083 return BAD_VALUE; 2084 } 2085 2086 sp<RecordThread> thread = openInput_l(module, input, config, *devices, address, source, flags); 2087 2088 if (thread != 0) { 2089 // notify client processes of the new input creation 2090 thread->ioConfigChanged(AUDIO_INPUT_OPENED); 2091 return NO_ERROR; 2092 } 2093 return NO_INIT; 2094} 2095 2096sp<AudioFlinger::RecordThread> AudioFlinger::openInput_l(audio_module_handle_t module, 2097 audio_io_handle_t *input, 2098 audio_config_t *config, 2099 audio_devices_t devices, 2100 const String8& address, 2101 audio_source_t source, 2102 audio_input_flags_t flags) 2103{ 2104 AudioHwDevice *inHwDev = findSuitableHwDev_l(module, devices); 2105 if (inHwDev == NULL) { 2106 *input = AUDIO_IO_HANDLE_NONE; 2107 return 0; 2108 } 2109 2110 // Audio Policy can request a specific handle for hardware hotword. 2111 // The goal here is not to re-open an already opened input. 2112 // It is to use a pre-assigned I/O handle. 2113 if (*input == AUDIO_IO_HANDLE_NONE) { 2114 *input = nextUniqueId(AUDIO_UNIQUE_ID_USE_INPUT); 2115 } else if (audio_unique_id_get_use(*input) != AUDIO_UNIQUE_ID_USE_INPUT) { 2116 ALOGE("openInput_l() requested input handle %d is invalid", *input); 2117 return 0; 2118 } else if (mRecordThreads.indexOfKey(*input) >= 0) { 2119 // This should not happen in a transient state with current design. 2120 ALOGE("openInput_l() requested input handle %d is already assigned", *input); 2121 return 0; 2122 } 2123 2124 audio_config_t halconfig = *config; 2125 audio_hw_device_t *inHwHal = inHwDev->hwDevice(); 2126 audio_stream_in_t *inStream = NULL; 2127 status_t status = inHwHal->open_input_stream(inHwHal, *input, devices, &halconfig, 2128 &inStream, flags, address.string(), source); 2129 ALOGV("openInput_l() openInputStream returned input %p, SamplingRate %d" 2130 ", Format %#x, Channels %x, flags %#x, status %d addr %s", 2131 inStream, 2132 halconfig.sample_rate, 2133 halconfig.format, 2134 halconfig.channel_mask, 2135 flags, 2136 status, address.string()); 2137 2138 // If the input could not be opened with the requested parameters and we can handle the 2139 // conversion internally, try to open again with the proposed parameters. 2140 if (status == BAD_VALUE && 2141 audio_is_linear_pcm(config->format) && 2142 audio_is_linear_pcm(halconfig.format) && 2143 (halconfig.sample_rate <= AUDIO_RESAMPLER_DOWN_RATIO_MAX * config->sample_rate) && 2144 (audio_channel_count_from_in_mask(halconfig.channel_mask) <= FCC_8) && 2145 (audio_channel_count_from_in_mask(config->channel_mask) <= FCC_8)) { 2146 // FIXME describe the change proposed by HAL (save old values so we can log them here) 2147 ALOGV("openInput_l() reopening with proposed sampling rate and channel mask"); 2148 inStream = NULL; 2149 status = inHwHal->open_input_stream(inHwHal, *input, devices, &halconfig, 2150 &inStream, flags, address.string(), source); 2151 // FIXME log this new status; HAL should not propose any further changes 2152 } 2153 2154 if (status == NO_ERROR && inStream != NULL) { 2155 2156#ifdef TEE_SINK 2157 // Try to re-use most recently used Pipe to archive a copy of input for dumpsys, 2158 // or (re-)create if current Pipe is idle and does not match the new format 2159 sp<NBAIO_Sink> teeSink; 2160 enum { 2161 TEE_SINK_NO, // don't copy input 2162 TEE_SINK_NEW, // copy input using a new pipe 2163 TEE_SINK_OLD, // copy input using an existing pipe 2164 } kind; 2165 NBAIO_Format format = Format_from_SR_C(halconfig.sample_rate, 2166 audio_channel_count_from_in_mask(halconfig.channel_mask), halconfig.format); 2167 if (!mTeeSinkInputEnabled) { 2168 kind = TEE_SINK_NO; 2169 } else if (!Format_isValid(format)) { 2170 kind = TEE_SINK_NO; 2171 } else if (mRecordTeeSink == 0) { 2172 kind = TEE_SINK_NEW; 2173 } else if (mRecordTeeSink->getStrongCount() != 1) { 2174 kind = TEE_SINK_NO; 2175 } else if (Format_isEqual(format, mRecordTeeSink->format())) { 2176 kind = TEE_SINK_OLD; 2177 } else { 2178 kind = TEE_SINK_NEW; 2179 } 2180 switch (kind) { 2181 case TEE_SINK_NEW: { 2182 Pipe *pipe = new Pipe(mTeeSinkInputFrames, format); 2183 size_t numCounterOffers = 0; 2184 const NBAIO_Format offers[1] = {format}; 2185 ssize_t index = pipe->negotiate(offers, 1, NULL, numCounterOffers); 2186 ALOG_ASSERT(index == 0); 2187 PipeReader *pipeReader = new PipeReader(*pipe); 2188 numCounterOffers = 0; 2189 index = pipeReader->negotiate(offers, 1, NULL, numCounterOffers); 2190 ALOG_ASSERT(index == 0); 2191 mRecordTeeSink = pipe; 2192 mRecordTeeSource = pipeReader; 2193 teeSink = pipe; 2194 } 2195 break; 2196 case TEE_SINK_OLD: 2197 teeSink = mRecordTeeSink; 2198 break; 2199 case TEE_SINK_NO: 2200 default: 2201 break; 2202 } 2203#endif 2204 2205 AudioStreamIn *inputStream = new AudioStreamIn(inHwDev, inStream, flags); 2206 2207 // Start record thread 2208 // RecordThread requires both input and output device indication to forward to audio 2209 // pre processing modules 2210 sp<RecordThread> thread = new RecordThread(this, 2211 inputStream, 2212 *input, 2213 primaryOutputDevice_l(), 2214 devices, 2215 mSystemReady 2216#ifdef TEE_SINK 2217 , teeSink 2218#endif 2219 ); 2220 mRecordThreads.add(*input, thread); 2221 ALOGV("openInput_l() created record thread: ID %d thread %p", *input, thread.get()); 2222 return thread; 2223 } 2224 2225 *input = AUDIO_IO_HANDLE_NONE; 2226 return 0; 2227} 2228 2229status_t AudioFlinger::closeInput(audio_io_handle_t input) 2230{ 2231 return closeInput_nonvirtual(input); 2232} 2233 2234status_t AudioFlinger::closeInput_nonvirtual(audio_io_handle_t input) 2235{ 2236 // keep strong reference on the record thread so that 2237 // it is not destroyed while exit() is executed 2238 sp<RecordThread> thread; 2239 { 2240 Mutex::Autolock _l(mLock); 2241 thread = checkRecordThread_l(input); 2242 if (thread == 0) { 2243 return BAD_VALUE; 2244 } 2245 2246 ALOGV("closeInput() %d", input); 2247 2248 // If we still have effect chains, it means that a client still holds a handle 2249 // on at least one effect. We must either move the chain to an existing thread with the 2250 // same session ID or put it aside in case a new record thread is opened for a 2251 // new capture on the same session 2252 sp<EffectChain> chain; 2253 { 2254 Mutex::Autolock _sl(thread->mLock); 2255 Vector< sp<EffectChain> > effectChains = thread->getEffectChains_l(); 2256 // Note: maximum one chain per record thread 2257 if (effectChains.size() != 0) { 2258 chain = effectChains[0]; 2259 } 2260 } 2261 if (chain != 0) { 2262 // first check if a record thread is already opened with a client on the same session. 2263 // This should only happen in case of overlap between one thread tear down and the 2264 // creation of its replacement 2265 size_t i; 2266 for (i = 0; i < mRecordThreads.size(); i++) { 2267 sp<RecordThread> t = mRecordThreads.valueAt(i); 2268 if (t == thread) { 2269 continue; 2270 } 2271 if (t->hasAudioSession(chain->sessionId()) != 0) { 2272 Mutex::Autolock _l(t->mLock); 2273 ALOGV("closeInput() found thread %d for effect session %d", 2274 t->id(), chain->sessionId()); 2275 t->addEffectChain_l(chain); 2276 break; 2277 } 2278 } 2279 // put the chain aside if we could not find a record thread with the same session id. 2280 if (i == mRecordThreads.size()) { 2281 putOrphanEffectChain_l(chain); 2282 } 2283 } 2284 const sp<AudioIoDescriptor> ioDesc = new AudioIoDescriptor(); 2285 ioDesc->mIoHandle = input; 2286 ioConfigChanged(AUDIO_INPUT_CLOSED, ioDesc); 2287 mRecordThreads.removeItem(input); 2288 } 2289 // FIXME: calling thread->exit() without mLock held should not be needed anymore now that 2290 // we have a different lock for notification client 2291 closeInputFinish(thread); 2292 return NO_ERROR; 2293} 2294 2295void AudioFlinger::closeInputFinish(sp<RecordThread> thread) 2296{ 2297 thread->exit(); 2298 AudioStreamIn *in = thread->clearInput(); 2299 ALOG_ASSERT(in != NULL, "in shouldn't be NULL"); 2300 // from now on thread->mInput is NULL 2301 in->hwDev()->close_input_stream(in->hwDev(), in->stream); 2302 delete in; 2303} 2304 2305void AudioFlinger::closeInputInternal_l(sp<RecordThread> thread) 2306{ 2307 mRecordThreads.removeItem(thread->mId); 2308 closeInputFinish(thread); 2309} 2310 2311status_t AudioFlinger::invalidateStream(audio_stream_type_t stream) 2312{ 2313 Mutex::Autolock _l(mLock); 2314 ALOGV("invalidateStream() stream %d", stream); 2315 2316 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2317 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 2318 thread->invalidateTracks(stream); 2319 } 2320 2321 return NO_ERROR; 2322} 2323 2324 2325audio_unique_id_t AudioFlinger::newAudioUniqueId(audio_unique_id_use_t use) 2326{ 2327 // This is a binder API, so a malicious client could pass in a bad parameter. 2328 // Check for that before calling the internal API nextUniqueId(). 2329 if ((unsigned) use >= (unsigned) AUDIO_UNIQUE_ID_USE_MAX) { 2330 ALOGE("newAudioUniqueId invalid use %d", use); 2331 return AUDIO_UNIQUE_ID_ALLOCATE; 2332 } 2333 return nextUniqueId(use); 2334} 2335 2336void AudioFlinger::acquireAudioSessionId(audio_session_t audioSession, pid_t pid) 2337{ 2338 Mutex::Autolock _l(mLock); 2339 pid_t caller = IPCThreadState::self()->getCallingPid(); 2340 ALOGV("acquiring %d from %d, for %d", audioSession, caller, pid); 2341 if (pid != -1 && (caller == getpid_cached)) { 2342 caller = pid; 2343 } 2344 2345 { 2346 Mutex::Autolock _cl(mClientLock); 2347 // Ignore requests received from processes not known as notification client. The request 2348 // is likely proxied by mediaserver (e.g CameraService) and releaseAudioSessionId() can be 2349 // called from a different pid leaving a stale session reference. Also we don't know how 2350 // to clear this reference if the client process dies. 2351 if (mNotificationClients.indexOfKey(caller) < 0) { 2352 ALOGW("acquireAudioSessionId() unknown client %d for session %d", caller, audioSession); 2353 return; 2354 } 2355 } 2356 2357 size_t num = mAudioSessionRefs.size(); 2358 for (size_t i = 0; i< num; i++) { 2359 AudioSessionRef *ref = mAudioSessionRefs.editItemAt(i); 2360 if (ref->mSessionid == audioSession && ref->mPid == caller) { 2361 ref->mCnt++; 2362 ALOGV(" incremented refcount to %d", ref->mCnt); 2363 return; 2364 } 2365 } 2366 mAudioSessionRefs.push(new AudioSessionRef(audioSession, caller)); 2367 ALOGV(" added new entry for %d", audioSession); 2368} 2369 2370void AudioFlinger::releaseAudioSessionId(audio_session_t audioSession, pid_t pid) 2371{ 2372 Mutex::Autolock _l(mLock); 2373 pid_t caller = IPCThreadState::self()->getCallingPid(); 2374 ALOGV("releasing %d from %d for %d", audioSession, caller, pid); 2375 if (pid != -1 && (caller == getpid_cached)) { 2376 caller = pid; 2377 } 2378 size_t num = mAudioSessionRefs.size(); 2379 for (size_t i = 0; i< num; i++) { 2380 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 2381 if (ref->mSessionid == audioSession && ref->mPid == caller) { 2382 ref->mCnt--; 2383 ALOGV(" decremented refcount to %d", ref->mCnt); 2384 if (ref->mCnt == 0) { 2385 mAudioSessionRefs.removeAt(i); 2386 delete ref; 2387 purgeStaleEffects_l(); 2388 } 2389 return; 2390 } 2391 } 2392 // If the caller is mediaserver it is likely that the session being released was acquired 2393 // on behalf of a process not in notification clients and we ignore the warning. 2394 ALOGW_IF(caller != getpid_cached, "session id %d not found for pid %d", audioSession, caller); 2395} 2396 2397void AudioFlinger::purgeStaleEffects_l() { 2398 2399 ALOGV("purging stale effects"); 2400 2401 Vector< sp<EffectChain> > chains; 2402 2403 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2404 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 2405 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 2406 sp<EffectChain> ec = t->mEffectChains[j]; 2407 if (ec->sessionId() > AUDIO_SESSION_OUTPUT_MIX) { 2408 chains.push(ec); 2409 } 2410 } 2411 } 2412 for (size_t i = 0; i < mRecordThreads.size(); i++) { 2413 sp<RecordThread> t = mRecordThreads.valueAt(i); 2414 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 2415 sp<EffectChain> ec = t->mEffectChains[j]; 2416 chains.push(ec); 2417 } 2418 } 2419 2420 for (size_t i = 0; i < chains.size(); i++) { 2421 sp<EffectChain> ec = chains[i]; 2422 int sessionid = ec->sessionId(); 2423 sp<ThreadBase> t = ec->mThread.promote(); 2424 if (t == 0) { 2425 continue; 2426 } 2427 size_t numsessionrefs = mAudioSessionRefs.size(); 2428 bool found = false; 2429 for (size_t k = 0; k < numsessionrefs; k++) { 2430 AudioSessionRef *ref = mAudioSessionRefs.itemAt(k); 2431 if (ref->mSessionid == sessionid) { 2432 ALOGV(" session %d still exists for %d with %d refs", 2433 sessionid, ref->mPid, ref->mCnt); 2434 found = true; 2435 break; 2436 } 2437 } 2438 if (!found) { 2439 Mutex::Autolock _l(t->mLock); 2440 // remove all effects from the chain 2441 while (ec->mEffects.size()) { 2442 sp<EffectModule> effect = ec->mEffects[0]; 2443 effect->unPin(); 2444 t->removeEffect_l(effect); 2445 if (effect->purgeHandles()) { 2446 t->checkSuspendOnEffectEnabled_l(effect, false, effect->sessionId()); 2447 } 2448 AudioSystem::unregisterEffect(effect->id()); 2449 } 2450 } 2451 } 2452 return; 2453} 2454 2455// checkThread_l() must be called with AudioFlinger::mLock held 2456AudioFlinger::ThreadBase *AudioFlinger::checkThread_l(audio_io_handle_t ioHandle) const 2457{ 2458 ThreadBase *thread = NULL; 2459 switch (audio_unique_id_get_use(ioHandle)) { 2460 case AUDIO_UNIQUE_ID_USE_OUTPUT: 2461 thread = checkPlaybackThread_l(ioHandle); 2462 break; 2463 case AUDIO_UNIQUE_ID_USE_INPUT: 2464 thread = checkRecordThread_l(ioHandle); 2465 break; 2466 default: 2467 break; 2468 } 2469 return thread; 2470} 2471 2472// checkPlaybackThread_l() must be called with AudioFlinger::mLock held 2473AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(audio_io_handle_t output) const 2474{ 2475 return mPlaybackThreads.valueFor(output).get(); 2476} 2477 2478// checkMixerThread_l() must be called with AudioFlinger::mLock held 2479AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(audio_io_handle_t output) const 2480{ 2481 PlaybackThread *thread = checkPlaybackThread_l(output); 2482 return thread != NULL && thread->type() != ThreadBase::DIRECT ? (MixerThread *) thread : NULL; 2483} 2484 2485// checkRecordThread_l() must be called with AudioFlinger::mLock held 2486AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(audio_io_handle_t input) const 2487{ 2488 return mRecordThreads.valueFor(input).get(); 2489} 2490 2491audio_unique_id_t AudioFlinger::nextUniqueId(audio_unique_id_use_t use) 2492{ 2493 // This is the internal API, so it is OK to assert on bad parameter. 2494 LOG_ALWAYS_FATAL_IF((unsigned) use >= (unsigned) AUDIO_UNIQUE_ID_USE_MAX); 2495 const int maxRetries = use == AUDIO_UNIQUE_ID_USE_SESSION ? 3 : 1; 2496 for (int retry = 0; retry < maxRetries; retry++) { 2497 // The cast allows wraparound from max positive to min negative instead of abort 2498 uint32_t base = (uint32_t) atomic_fetch_add_explicit(&mNextUniqueIds[use], 2499 (uint_fast32_t) AUDIO_UNIQUE_ID_USE_MAX, memory_order_acq_rel); 2500 ALOG_ASSERT(audio_unique_id_get_use(base) == AUDIO_UNIQUE_ID_USE_UNSPECIFIED); 2501 // allow wrap by skipping 0 and -1 for session ids 2502 if (!(base == 0 || base == (~0u & ~AUDIO_UNIQUE_ID_USE_MASK))) { 2503 ALOGW_IF(retry != 0, "unique ID overflow for use %d", use); 2504 return (audio_unique_id_t) (base | use); 2505 } 2506 } 2507 // We have no way of recovering from wraparound 2508 LOG_ALWAYS_FATAL("unique ID overflow for use %d", use); 2509 // TODO Use a floor after wraparound. This may need a mutex. 2510} 2511 2512AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l() const 2513{ 2514 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2515 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 2516 if(thread->isDuplicating()) { 2517 continue; 2518 } 2519 AudioStreamOut *output = thread->getOutput(); 2520 if (output != NULL && output->audioHwDev == mPrimaryHardwareDev) { 2521 return thread; 2522 } 2523 } 2524 return NULL; 2525} 2526 2527audio_devices_t AudioFlinger::primaryOutputDevice_l() const 2528{ 2529 PlaybackThread *thread = primaryPlaybackThread_l(); 2530 2531 if (thread == NULL) { 2532 return 0; 2533 } 2534 2535 return thread->outDevice(); 2536} 2537 2538AudioFlinger::PlaybackThread *AudioFlinger::fastPlaybackThread_l() const 2539{ 2540 size_t minFrameCount = 0; 2541 PlaybackThread *minThread = NULL; 2542 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2543 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 2544 if (!thread->isDuplicating()) { 2545 size_t frameCount = thread->frameCountHAL(); 2546 if (frameCount != 0 && (minFrameCount == 0 || frameCount < minFrameCount || 2547 (frameCount == minFrameCount && thread->hasFastMixer() && 2548 /*minThread != NULL &&*/ !minThread->hasFastMixer()))) { 2549 minFrameCount = frameCount; 2550 minThread = thread; 2551 } 2552 } 2553 } 2554 return minThread; 2555} 2556 2557sp<AudioFlinger::SyncEvent> AudioFlinger::createSyncEvent(AudioSystem::sync_event_t type, 2558 audio_session_t triggerSession, 2559 audio_session_t listenerSession, 2560 sync_event_callback_t callBack, 2561 wp<RefBase> cookie) 2562{ 2563 Mutex::Autolock _l(mLock); 2564 2565 sp<SyncEvent> event = new SyncEvent(type, triggerSession, listenerSession, callBack, cookie); 2566 status_t playStatus = NAME_NOT_FOUND; 2567 status_t recStatus = NAME_NOT_FOUND; 2568 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2569 playStatus = mPlaybackThreads.valueAt(i)->setSyncEvent(event); 2570 if (playStatus == NO_ERROR) { 2571 return event; 2572 } 2573 } 2574 for (size_t i = 0; i < mRecordThreads.size(); i++) { 2575 recStatus = mRecordThreads.valueAt(i)->setSyncEvent(event); 2576 if (recStatus == NO_ERROR) { 2577 return event; 2578 } 2579 } 2580 if (playStatus == NAME_NOT_FOUND || recStatus == NAME_NOT_FOUND) { 2581 mPendingSyncEvents.add(event); 2582 } else { 2583 ALOGV("createSyncEvent() invalid event %d", event->type()); 2584 event.clear(); 2585 } 2586 return event; 2587} 2588 2589// ---------------------------------------------------------------------------- 2590// Effect management 2591// ---------------------------------------------------------------------------- 2592 2593 2594status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects) const 2595{ 2596 Mutex::Autolock _l(mLock); 2597 return EffectQueryNumberEffects(numEffects); 2598} 2599 2600status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor) const 2601{ 2602 Mutex::Autolock _l(mLock); 2603 return EffectQueryEffect(index, descriptor); 2604} 2605 2606status_t AudioFlinger::getEffectDescriptor(const effect_uuid_t *pUuid, 2607 effect_descriptor_t *descriptor) const 2608{ 2609 Mutex::Autolock _l(mLock); 2610 return EffectGetDescriptor(pUuid, descriptor); 2611} 2612 2613 2614sp<IEffect> AudioFlinger::createEffect( 2615 effect_descriptor_t *pDesc, 2616 const sp<IEffectClient>& effectClient, 2617 int32_t priority, 2618 audio_io_handle_t io, 2619 audio_session_t sessionId, 2620 const String16& opPackageName, 2621 status_t *status, 2622 int *id, 2623 int *enabled) 2624{ 2625 status_t lStatus = NO_ERROR; 2626 sp<EffectHandle> handle; 2627 effect_descriptor_t desc; 2628 2629 pid_t pid = IPCThreadState::self()->getCallingPid(); 2630 ALOGV("createEffect pid %d, effectClient %p, priority %d, sessionId %d, io %d", 2631 pid, effectClient.get(), priority, sessionId, io); 2632 2633 if (pDesc == NULL) { 2634 lStatus = BAD_VALUE; 2635 goto Exit; 2636 } 2637 2638 // check audio settings permission for global effects 2639 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && !settingsAllowed()) { 2640 lStatus = PERMISSION_DENIED; 2641 goto Exit; 2642 } 2643 2644 // Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects 2645 // that can only be created by audio policy manager (running in same process) 2646 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid_cached != pid) { 2647 lStatus = PERMISSION_DENIED; 2648 goto Exit; 2649 } 2650 2651 { 2652 if (!EffectIsNullUuid(&pDesc->uuid)) { 2653 // if uuid is specified, request effect descriptor 2654 lStatus = EffectGetDescriptor(&pDesc->uuid, &desc); 2655 if (lStatus < 0) { 2656 ALOGW("createEffect() error %d from EffectGetDescriptor", lStatus); 2657 goto Exit; 2658 } 2659 } else { 2660 // if uuid is not specified, look for an available implementation 2661 // of the required type in effect factory 2662 if (EffectIsNullUuid(&pDesc->type)) { 2663 ALOGW("createEffect() no effect type"); 2664 lStatus = BAD_VALUE; 2665 goto Exit; 2666 } 2667 uint32_t numEffects = 0; 2668 effect_descriptor_t d; 2669 d.flags = 0; // prevent compiler warning 2670 bool found = false; 2671 2672 lStatus = EffectQueryNumberEffects(&numEffects); 2673 if (lStatus < 0) { 2674 ALOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus); 2675 goto Exit; 2676 } 2677 for (uint32_t i = 0; i < numEffects; i++) { 2678 lStatus = EffectQueryEffect(i, &desc); 2679 if (lStatus < 0) { 2680 ALOGW("createEffect() error %d from EffectQueryEffect", lStatus); 2681 continue; 2682 } 2683 if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) { 2684 // If matching type found save effect descriptor. If the session is 2685 // 0 and the effect is not auxiliary, continue enumeration in case 2686 // an auxiliary version of this effect type is available 2687 found = true; 2688 d = desc; 2689 if (sessionId != AUDIO_SESSION_OUTPUT_MIX || 2690 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 2691 break; 2692 } 2693 } 2694 } 2695 if (!found) { 2696 lStatus = BAD_VALUE; 2697 ALOGW("createEffect() effect not found"); 2698 goto Exit; 2699 } 2700 // For same effect type, chose auxiliary version over insert version if 2701 // connect to output mix (Compliance to OpenSL ES) 2702 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && 2703 (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) { 2704 desc = d; 2705 } 2706 } 2707 2708 // Do not allow auxiliary effects on a session different from 0 (output mix) 2709 if (sessionId != AUDIO_SESSION_OUTPUT_MIX && 2710 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 2711 lStatus = INVALID_OPERATION; 2712 goto Exit; 2713 } 2714 2715 // check recording permission for visualizer 2716 if ((memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) && 2717 !recordingAllowed(opPackageName, pid, IPCThreadState::self()->getCallingUid())) { 2718 lStatus = PERMISSION_DENIED; 2719 goto Exit; 2720 } 2721 2722 // return effect descriptor 2723 *pDesc = desc; 2724 if (io == AUDIO_IO_HANDLE_NONE && sessionId == AUDIO_SESSION_OUTPUT_MIX) { 2725 // if the output returned by getOutputForEffect() is removed before we lock the 2726 // mutex below, the call to checkPlaybackThread_l(io) below will detect it 2727 // and we will exit safely 2728 io = AudioSystem::getOutputForEffect(&desc); 2729 ALOGV("createEffect got output %d", io); 2730 } 2731 2732 Mutex::Autolock _l(mLock); 2733 2734 // If output is not specified try to find a matching audio session ID in one of the 2735 // output threads. 2736 // If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX 2737 // because of code checking output when entering the function. 2738 // Note: io is never 0 when creating an effect on an input 2739 if (io == AUDIO_IO_HANDLE_NONE) { 2740 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) { 2741 // output must be specified by AudioPolicyManager when using session 2742 // AUDIO_SESSION_OUTPUT_STAGE 2743 lStatus = BAD_VALUE; 2744 goto Exit; 2745 } 2746 // look for the thread where the specified audio session is present 2747 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2748 if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 2749 io = mPlaybackThreads.keyAt(i); 2750 break; 2751 } 2752 } 2753 if (io == 0) { 2754 for (size_t i = 0; i < mRecordThreads.size(); i++) { 2755 if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 2756 io = mRecordThreads.keyAt(i); 2757 break; 2758 } 2759 } 2760 } 2761 // If no output thread contains the requested session ID, default to 2762 // first output. The effect chain will be moved to the correct output 2763 // thread when a track with the same session ID is created 2764 if (io == AUDIO_IO_HANDLE_NONE && mPlaybackThreads.size() > 0) { 2765 io = mPlaybackThreads.keyAt(0); 2766 } 2767 ALOGV("createEffect() got io %d for effect %s", io, desc.name); 2768 } 2769 ThreadBase *thread = checkRecordThread_l(io); 2770 if (thread == NULL) { 2771 thread = checkPlaybackThread_l(io); 2772 if (thread == NULL) { 2773 ALOGE("createEffect() unknown output thread"); 2774 lStatus = BAD_VALUE; 2775 goto Exit; 2776 } 2777 } else { 2778 // Check if one effect chain was awaiting for an effect to be created on this 2779 // session and used it instead of creating a new one. 2780 sp<EffectChain> chain = getOrphanEffectChain_l(sessionId); 2781 if (chain != 0) { 2782 Mutex::Autolock _l(thread->mLock); 2783 thread->addEffectChain_l(chain); 2784 } 2785 } 2786 2787 sp<Client> client = registerPid(pid); 2788 2789 // create effect on selected output thread 2790 handle = thread->createEffect_l(client, effectClient, priority, sessionId, 2791 &desc, enabled, &lStatus); 2792 if (handle != 0 && id != NULL) { 2793 *id = handle->id(); 2794 } 2795 if (handle == 0) { 2796 // remove local strong reference to Client with mClientLock held 2797 Mutex::Autolock _cl(mClientLock); 2798 client.clear(); 2799 } 2800 } 2801 2802Exit: 2803 *status = lStatus; 2804 return handle; 2805} 2806 2807status_t AudioFlinger::moveEffects(audio_session_t sessionId, audio_io_handle_t srcOutput, 2808 audio_io_handle_t dstOutput) 2809{ 2810 ALOGV("moveEffects() session %d, srcOutput %d, dstOutput %d", 2811 sessionId, srcOutput, dstOutput); 2812 Mutex::Autolock _l(mLock); 2813 if (srcOutput == dstOutput) { 2814 ALOGW("moveEffects() same dst and src outputs %d", dstOutput); 2815 return NO_ERROR; 2816 } 2817 PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput); 2818 if (srcThread == NULL) { 2819 ALOGW("moveEffects() bad srcOutput %d", srcOutput); 2820 return BAD_VALUE; 2821 } 2822 PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput); 2823 if (dstThread == NULL) { 2824 ALOGW("moveEffects() bad dstOutput %d", dstOutput); 2825 return BAD_VALUE; 2826 } 2827 2828 Mutex::Autolock _dl(dstThread->mLock); 2829 Mutex::Autolock _sl(srcThread->mLock); 2830 return moveEffectChain_l(sessionId, srcThread, dstThread, false); 2831} 2832 2833// moveEffectChain_l must be called with both srcThread and dstThread mLocks held 2834status_t AudioFlinger::moveEffectChain_l(audio_session_t sessionId, 2835 AudioFlinger::PlaybackThread *srcThread, 2836 AudioFlinger::PlaybackThread *dstThread, 2837 bool reRegister) 2838{ 2839 ALOGV("moveEffectChain_l() session %d from thread %p to thread %p", 2840 sessionId, srcThread, dstThread); 2841 2842 sp<EffectChain> chain = srcThread->getEffectChain_l(sessionId); 2843 if (chain == 0) { 2844 ALOGW("moveEffectChain_l() effect chain for session %d not on source thread %p", 2845 sessionId, srcThread); 2846 return INVALID_OPERATION; 2847 } 2848 2849 // Check whether the destination thread and all effects in the chain are compatible 2850 if (!chain->isCompatibleWithThread_l(dstThread)) { 2851 ALOGW("moveEffectChain_l() effect chain failed because" 2852 " destination thread %p is not compatible with effects in the chain", 2853 dstThread); 2854 return INVALID_OPERATION; 2855 } 2856 2857 // remove chain first. This is useful only if reconfiguring effect chain on same output thread, 2858 // so that a new chain is created with correct parameters when first effect is added. This is 2859 // otherwise unnecessary as removeEffect_l() will remove the chain when last effect is 2860 // removed. 2861 srcThread->removeEffectChain_l(chain); 2862 2863 // transfer all effects one by one so that new effect chain is created on new thread with 2864 // correct buffer sizes and audio parameters and effect engines reconfigured accordingly 2865 sp<EffectChain> dstChain; 2866 uint32_t strategy = 0; // prevent compiler warning 2867 sp<EffectModule> effect = chain->getEffectFromId_l(0); 2868 Vector< sp<EffectModule> > removed; 2869 status_t status = NO_ERROR; 2870 while (effect != 0) { 2871 srcThread->removeEffect_l(effect); 2872 removed.add(effect); 2873 status = dstThread->addEffect_l(effect); 2874 if (status != NO_ERROR) { 2875 break; 2876 } 2877 // removeEffect_l() has stopped the effect if it was active so it must be restarted 2878 if (effect->state() == EffectModule::ACTIVE || 2879 effect->state() == EffectModule::STOPPING) { 2880 effect->start(); 2881 } 2882 // if the move request is not received from audio policy manager, the effect must be 2883 // re-registered with the new strategy and output 2884 if (dstChain == 0) { 2885 dstChain = effect->chain().promote(); 2886 if (dstChain == 0) { 2887 ALOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get()); 2888 status = NO_INIT; 2889 break; 2890 } 2891 strategy = dstChain->strategy(); 2892 } 2893 if (reRegister) { 2894 AudioSystem::unregisterEffect(effect->id()); 2895 AudioSystem::registerEffect(&effect->desc(), 2896 dstThread->id(), 2897 strategy, 2898 sessionId, 2899 effect->id()); 2900 AudioSystem::setEffectEnabled(effect->id(), effect->isEnabled()); 2901 } 2902 effect = chain->getEffectFromId_l(0); 2903 } 2904 2905 if (status != NO_ERROR) { 2906 for (size_t i = 0; i < removed.size(); i++) { 2907 srcThread->addEffect_l(removed[i]); 2908 if (dstChain != 0 && reRegister) { 2909 AudioSystem::unregisterEffect(removed[i]->id()); 2910 AudioSystem::registerEffect(&removed[i]->desc(), 2911 srcThread->id(), 2912 strategy, 2913 sessionId, 2914 removed[i]->id()); 2915 AudioSystem::setEffectEnabled(effect->id(), effect->isEnabled()); 2916 } 2917 } 2918 } 2919 2920 return status; 2921} 2922 2923bool AudioFlinger::isNonOffloadableGlobalEffectEnabled_l() 2924{ 2925 if (mGlobalEffectEnableTime != 0 && 2926 ((systemTime() - mGlobalEffectEnableTime) < kMinGlobalEffectEnabletimeNs)) { 2927 return true; 2928 } 2929 2930 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2931 sp<EffectChain> ec = 2932 mPlaybackThreads.valueAt(i)->getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2933 if (ec != 0 && ec->isNonOffloadableEnabled()) { 2934 return true; 2935 } 2936 } 2937 return false; 2938} 2939 2940void AudioFlinger::onNonOffloadableGlobalEffectEnable() 2941{ 2942 Mutex::Autolock _l(mLock); 2943 2944 mGlobalEffectEnableTime = systemTime(); 2945 2946 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2947 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 2948 if (t->mType == ThreadBase::OFFLOAD) { 2949 t->invalidateTracks(AUDIO_STREAM_MUSIC); 2950 } 2951 } 2952 2953} 2954 2955status_t AudioFlinger::putOrphanEffectChain_l(const sp<AudioFlinger::EffectChain>& chain) 2956{ 2957 audio_session_t session = chain->sessionId(); 2958 ssize_t index = mOrphanEffectChains.indexOfKey(session); 2959 ALOGV("putOrphanEffectChain_l session %d index %zd", session, index); 2960 if (index >= 0) { 2961 ALOGW("putOrphanEffectChain_l chain for session %d already present", session); 2962 return ALREADY_EXISTS; 2963 } 2964 mOrphanEffectChains.add(session, chain); 2965 return NO_ERROR; 2966} 2967 2968sp<AudioFlinger::EffectChain> AudioFlinger::getOrphanEffectChain_l(audio_session_t session) 2969{ 2970 sp<EffectChain> chain; 2971 ssize_t index = mOrphanEffectChains.indexOfKey(session); 2972 ALOGV("getOrphanEffectChain_l session %d index %zd", session, index); 2973 if (index >= 0) { 2974 chain = mOrphanEffectChains.valueAt(index); 2975 mOrphanEffectChains.removeItemsAt(index); 2976 } 2977 return chain; 2978} 2979 2980bool AudioFlinger::updateOrphanEffectChains(const sp<AudioFlinger::EffectModule>& effect) 2981{ 2982 Mutex::Autolock _l(mLock); 2983 audio_session_t session = effect->sessionId(); 2984 ssize_t index = mOrphanEffectChains.indexOfKey(session); 2985 ALOGV("updateOrphanEffectChains session %d index %zd", session, index); 2986 if (index >= 0) { 2987 sp<EffectChain> chain = mOrphanEffectChains.valueAt(index); 2988 if (chain->removeEffect_l(effect) == 0) { 2989 ALOGV("updateOrphanEffectChains removing effect chain at index %zd", index); 2990 mOrphanEffectChains.removeItemsAt(index); 2991 } 2992 return true; 2993 } 2994 return false; 2995} 2996 2997 2998struct Entry { 2999#define TEE_MAX_FILENAME 32 // %Y%m%d%H%M%S_%d.wav = 4+2+2+2+2+2+1+1+4+1 = 21 3000 char mFileName[TEE_MAX_FILENAME]; 3001}; 3002 3003int comparEntry(const void *p1, const void *p2) 3004{ 3005 return strcmp(((const Entry *) p1)->mFileName, ((const Entry *) p2)->mFileName); 3006} 3007 3008#ifdef TEE_SINK 3009void AudioFlinger::dumpTee(int fd, const sp<NBAIO_Source>& source, audio_io_handle_t id) 3010{ 3011 NBAIO_Source *teeSource = source.get(); 3012 if (teeSource != NULL) { 3013 // .wav rotation 3014 // There is a benign race condition if 2 threads call this simultaneously. 3015 // They would both traverse the directory, but the result would simply be 3016 // failures at unlink() which are ignored. It's also unlikely since 3017 // normally dumpsys is only done by bugreport or from the command line. 3018 char teePath[32+256]; 3019 strcpy(teePath, "/data/misc/audioserver"); 3020 size_t teePathLen = strlen(teePath); 3021 DIR *dir = opendir(teePath); 3022 teePath[teePathLen++] = '/'; 3023 if (dir != NULL) { 3024#define TEE_MAX_SORT 20 // number of entries to sort 3025#define TEE_MAX_KEEP 10 // number of entries to keep 3026 struct Entry entries[TEE_MAX_SORT]; 3027 size_t entryCount = 0; 3028 while (entryCount < TEE_MAX_SORT) { 3029 struct dirent de; 3030 struct dirent *result = NULL; 3031 int rc = readdir_r(dir, &de, &result); 3032 if (rc != 0) { 3033 ALOGW("readdir_r failed %d", rc); 3034 break; 3035 } 3036 if (result == NULL) { 3037 break; 3038 } 3039 if (result != &de) { 3040 ALOGW("readdir_r returned unexpected result %p != %p", result, &de); 3041 break; 3042 } 3043 // ignore non .wav file entries 3044 size_t nameLen = strlen(de.d_name); 3045 if (nameLen <= 4 || nameLen >= TEE_MAX_FILENAME || 3046 strcmp(&de.d_name[nameLen - 4], ".wav")) { 3047 continue; 3048 } 3049 strcpy(entries[entryCount++].mFileName, de.d_name); 3050 } 3051 (void) closedir(dir); 3052 if (entryCount > TEE_MAX_KEEP) { 3053 qsort(entries, entryCount, sizeof(Entry), comparEntry); 3054 for (size_t i = 0; i < entryCount - TEE_MAX_KEEP; ++i) { 3055 strcpy(&teePath[teePathLen], entries[i].mFileName); 3056 (void) unlink(teePath); 3057 } 3058 } 3059 } else { 3060 if (fd >= 0) { 3061 dprintf(fd, "unable to rotate tees in %.*s: %s\n", teePathLen, teePath, 3062 strerror(errno)); 3063 } 3064 } 3065 char teeTime[16]; 3066 struct timeval tv; 3067 gettimeofday(&tv, NULL); 3068 struct tm tm; 3069 localtime_r(&tv.tv_sec, &tm); 3070 strftime(teeTime, sizeof(teeTime), "%Y%m%d%H%M%S", &tm); 3071 snprintf(&teePath[teePathLen], sizeof(teePath) - teePathLen, "%s_%d.wav", teeTime, id); 3072 // if 2 dumpsys are done within 1 second, and rotation didn't work, then discard 2nd 3073 int teeFd = open(teePath, O_WRONLY | O_CREAT | O_EXCL | O_NOFOLLOW, S_IRUSR | S_IWUSR); 3074 if (teeFd >= 0) { 3075 // FIXME use libsndfile 3076 char wavHeader[44]; 3077 memcpy(wavHeader, 3078 "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", 3079 sizeof(wavHeader)); 3080 NBAIO_Format format = teeSource->format(); 3081 unsigned channelCount = Format_channelCount(format); 3082 uint32_t sampleRate = Format_sampleRate(format); 3083 size_t frameSize = Format_frameSize(format); 3084 wavHeader[22] = channelCount; // number of channels 3085 wavHeader[24] = sampleRate; // sample rate 3086 wavHeader[25] = sampleRate >> 8; 3087 wavHeader[32] = frameSize; // block alignment 3088 wavHeader[33] = frameSize >> 8; 3089 write(teeFd, wavHeader, sizeof(wavHeader)); 3090 size_t total = 0; 3091 bool firstRead = true; 3092#define TEE_SINK_READ 1024 // frames per I/O operation 3093 void *buffer = malloc(TEE_SINK_READ * frameSize); 3094 for (;;) { 3095 size_t count = TEE_SINK_READ; 3096 ssize_t actual = teeSource->read(buffer, count); 3097 bool wasFirstRead = firstRead; 3098 firstRead = false; 3099 if (actual <= 0) { 3100 if (actual == (ssize_t) OVERRUN && wasFirstRead) { 3101 continue; 3102 } 3103 break; 3104 } 3105 ALOG_ASSERT(actual <= (ssize_t)count); 3106 write(teeFd, buffer, actual * frameSize); 3107 total += actual; 3108 } 3109 free(buffer); 3110 lseek(teeFd, (off_t) 4, SEEK_SET); 3111 uint32_t temp = 44 + total * frameSize - 8; 3112 // FIXME not big-endian safe 3113 write(teeFd, &temp, sizeof(temp)); 3114 lseek(teeFd, (off_t) 40, SEEK_SET); 3115 temp = total * frameSize; 3116 // FIXME not big-endian safe 3117 write(teeFd, &temp, sizeof(temp)); 3118 close(teeFd); 3119 if (fd >= 0) { 3120 dprintf(fd, "tee copied to %s\n", teePath); 3121 } 3122 } else { 3123 if (fd >= 0) { 3124 dprintf(fd, "unable to create tee %s: %s\n", teePath, strerror(errno)); 3125 } 3126 } 3127 } 3128} 3129#endif 3130 3131// ---------------------------------------------------------------------------- 3132 3133status_t AudioFlinger::onTransact( 3134 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 3135{ 3136 return BnAudioFlinger::onTransact(code, data, reply, flags); 3137} 3138 3139} // namespace android 3140