AudioFlinger.cpp revision 1c333e252cbca3337c1bedbc57a005f3b7d23fdb
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 <utils/String16.h> 35#include <utils/threads.h> 36#include <utils/Atomic.h> 37 38#include <cutils/bitops.h> 39#include <cutils/properties.h> 40 41#include <system/audio.h> 42#include <hardware/audio.h> 43 44#include "AudioMixer.h" 45#include "AudioFlinger.h" 46#include "ServiceUtilities.h" 47 48#include <media/EffectsFactoryApi.h> 49#include <audio_effects/effect_visualizer.h> 50#include <audio_effects/effect_ns.h> 51#include <audio_effects/effect_aec.h> 52 53#include <audio_utils/primitives.h> 54 55#include <powermanager/PowerManager.h> 56 57#include <common_time/cc_helper.h> 58 59#include <media/IMediaLogService.h> 60 61#include <media/nbaio/Pipe.h> 62#include <media/nbaio/PipeReader.h> 63#include <media/AudioParameter.h> 64#include <private/android_filesystem_config.h> 65 66// ---------------------------------------------------------------------------- 67 68// Note: the following macro is used for extremely verbose logging message. In 69// order to run with ALOG_ASSERT turned on, we need to have LOG_NDEBUG set to 70// 0; but one side effect of this is to turn all LOGV's as well. Some messages 71// are so verbose that we want to suppress them even when we have ALOG_ASSERT 72// turned on. Do not uncomment the #def below unless you really know what you 73// are doing and want to see all of the extremely verbose messages. 74//#define VERY_VERY_VERBOSE_LOGGING 75#ifdef VERY_VERY_VERBOSE_LOGGING 76#define ALOGVV ALOGV 77#else 78#define ALOGVV(a...) do { } while(0) 79#endif 80 81namespace android { 82 83static const char kDeadlockedString[] = "AudioFlinger may be deadlocked\n"; 84static const char kHardwareLockedString[] = "Hardware lock is taken\n"; 85static const char kClientLockedString[] = "Client lock is taken\n"; 86 87 88nsecs_t AudioFlinger::mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs; 89 90uint32_t AudioFlinger::mScreenState; 91 92#ifdef TEE_SINK 93bool AudioFlinger::mTeeSinkInputEnabled = false; 94bool AudioFlinger::mTeeSinkOutputEnabled = false; 95bool AudioFlinger::mTeeSinkTrackEnabled = false; 96 97size_t AudioFlinger::mTeeSinkInputFrames = kTeeSinkInputFramesDefault; 98size_t AudioFlinger::mTeeSinkOutputFrames = kTeeSinkOutputFramesDefault; 99size_t AudioFlinger::mTeeSinkTrackFrames = kTeeSinkTrackFramesDefault; 100#endif 101 102// In order to avoid invalidating offloaded tracks each time a Visualizer is turned on and off 103// we define a minimum time during which a global effect is considered enabled. 104static const nsecs_t kMinGlobalEffectEnabletimeNs = seconds(7200); 105 106// ---------------------------------------------------------------------------- 107 108const char *formatToString(audio_format_t format) { 109 switch(format) { 110 case AUDIO_FORMAT_PCM_SUB_8_BIT: return "pcm8"; 111 case AUDIO_FORMAT_PCM_SUB_16_BIT: return "pcm16"; 112 case AUDIO_FORMAT_PCM_SUB_32_BIT: return "pcm32"; 113 case AUDIO_FORMAT_PCM_SUB_8_24_BIT: return "pcm8.24"; 114 case AUDIO_FORMAT_PCM_SUB_24_BIT_PACKED: return "pcm24"; 115 case AUDIO_FORMAT_PCM_SUB_FLOAT: return "pcmfloat"; 116 case AUDIO_FORMAT_MP3: return "mp3"; 117 case AUDIO_FORMAT_AMR_NB: return "amr-nb"; 118 case AUDIO_FORMAT_AMR_WB: return "amr-wb"; 119 case AUDIO_FORMAT_AAC: return "aac"; 120 case AUDIO_FORMAT_HE_AAC_V1: return "he-aac-v1"; 121 case AUDIO_FORMAT_HE_AAC_V2: return "he-aac-v2"; 122 case AUDIO_FORMAT_VORBIS: return "vorbis"; 123 default: 124 break; 125 } 126 return "unknown"; 127} 128 129static int load_audio_interface(const char *if_name, audio_hw_device_t **dev) 130{ 131 const hw_module_t *mod; 132 int rc; 133 134 rc = hw_get_module_by_class(AUDIO_HARDWARE_MODULE_ID, if_name, &mod); 135 ALOGE_IF(rc, "%s couldn't load audio hw module %s.%s (%s)", __func__, 136 AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc)); 137 if (rc) { 138 goto out; 139 } 140 rc = audio_hw_device_open(mod, dev); 141 ALOGE_IF(rc, "%s couldn't open audio hw device in %s.%s (%s)", __func__, 142 AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc)); 143 if (rc) { 144 goto out; 145 } 146 if ((*dev)->common.version < AUDIO_DEVICE_API_VERSION_MIN) { 147 ALOGE("%s wrong audio hw device version %04x", __func__, (*dev)->common.version); 148 rc = BAD_VALUE; 149 goto out; 150 } 151 return 0; 152 153out: 154 *dev = NULL; 155 return rc; 156} 157 158// ---------------------------------------------------------------------------- 159 160AudioFlinger::AudioFlinger() 161 : BnAudioFlinger(), 162 mPrimaryHardwareDev(NULL), 163 mAudioHwDevs(NULL), 164 mHardwareStatus(AUDIO_HW_IDLE), 165 mMasterVolume(1.0f), 166 mMasterMute(false), 167 mNextUniqueId(1), 168 mMode(AUDIO_MODE_INVALID), 169 mBtNrecIsOff(false), 170 mIsLowRamDevice(true), 171 mIsDeviceTypeKnown(false), 172 mGlobalEffectEnableTime(0) 173{ 174 getpid_cached = getpid(); 175 char value[PROPERTY_VALUE_MAX]; 176 bool doLog = (property_get("ro.test_harness", value, "0") > 0) && (atoi(value) == 1); 177 if (doLog) { 178 mLogMemoryDealer = new MemoryDealer(kLogMemorySize, "LogWriters", MemoryHeapBase::READ_ONLY); 179 } 180 181#ifdef TEE_SINK 182 (void) property_get("ro.debuggable", value, "0"); 183 int debuggable = atoi(value); 184 int teeEnabled = 0; 185 if (debuggable) { 186 (void) property_get("af.tee", value, "0"); 187 teeEnabled = atoi(value); 188 } 189 // FIXME symbolic constants here 190 if (teeEnabled & 1) { 191 mTeeSinkInputEnabled = true; 192 } 193 if (teeEnabled & 2) { 194 mTeeSinkOutputEnabled = true; 195 } 196 if (teeEnabled & 4) { 197 mTeeSinkTrackEnabled = true; 198 } 199#endif 200} 201 202void AudioFlinger::onFirstRef() 203{ 204 int rc = 0; 205 206 Mutex::Autolock _l(mLock); 207 208 /* TODO: move all this work into an Init() function */ 209 char val_str[PROPERTY_VALUE_MAX] = { 0 }; 210 if (property_get("ro.audio.flinger_standbytime_ms", val_str, NULL) >= 0) { 211 uint32_t int_val; 212 if (1 == sscanf(val_str, "%u", &int_val)) { 213 mStandbyTimeInNsecs = milliseconds(int_val); 214 ALOGI("Using %u mSec as standby time.", int_val); 215 } else { 216 mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs; 217 ALOGI("Using default %u mSec as standby time.", 218 (uint32_t)(mStandbyTimeInNsecs / 1000000)); 219 } 220 } 221 222 mPatchPanel = new PatchPanel(this); 223 224 mMode = AUDIO_MODE_NORMAL; 225} 226 227AudioFlinger::~AudioFlinger() 228{ 229 while (!mRecordThreads.isEmpty()) { 230 // closeInput_nonvirtual() will remove specified entry from mRecordThreads 231 closeInput_nonvirtual(mRecordThreads.keyAt(0)); 232 } 233 while (!mPlaybackThreads.isEmpty()) { 234 // closeOutput_nonvirtual() will remove specified entry from mPlaybackThreads 235 closeOutput_nonvirtual(mPlaybackThreads.keyAt(0)); 236 } 237 238 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 239 // no mHardwareLock needed, as there are no other references to this 240 audio_hw_device_close(mAudioHwDevs.valueAt(i)->hwDevice()); 241 delete mAudioHwDevs.valueAt(i); 242 } 243 244 // Tell media.log service about any old writers that still need to be unregistered 245 sp<IBinder> binder = defaultServiceManager()->getService(String16("media.log")); 246 if (binder != 0) { 247 sp<IMediaLogService> mediaLogService(interface_cast<IMediaLogService>(binder)); 248 for (size_t count = mUnregisteredWriters.size(); count > 0; count--) { 249 sp<IMemory> iMemory(mUnregisteredWriters.top()->getIMemory()); 250 mUnregisteredWriters.pop(); 251 mediaLogService->unregisterWriter(iMemory); 252 } 253 } 254 255} 256 257static const char * const audio_interfaces[] = { 258 AUDIO_HARDWARE_MODULE_ID_PRIMARY, 259 AUDIO_HARDWARE_MODULE_ID_A2DP, 260 AUDIO_HARDWARE_MODULE_ID_USB, 261}; 262#define ARRAY_SIZE(x) (sizeof((x))/sizeof(((x)[0]))) 263 264AudioFlinger::AudioHwDevice* AudioFlinger::findSuitableHwDev_l( 265 audio_module_handle_t module, 266 audio_devices_t devices) 267{ 268 // if module is 0, the request comes from an old policy manager and we should load 269 // well known modules 270 if (module == 0) { 271 ALOGW("findSuitableHwDev_l() loading well know audio hw modules"); 272 for (size_t i = 0; i < ARRAY_SIZE(audio_interfaces); i++) { 273 loadHwModule_l(audio_interfaces[i]); 274 } 275 // then try to find a module supporting the requested device. 276 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 277 AudioHwDevice *audioHwDevice = mAudioHwDevs.valueAt(i); 278 audio_hw_device_t *dev = audioHwDevice->hwDevice(); 279 if ((dev->get_supported_devices != NULL) && 280 (dev->get_supported_devices(dev) & devices) == devices) 281 return audioHwDevice; 282 } 283 } else { 284 // check a match for the requested module handle 285 AudioHwDevice *audioHwDevice = mAudioHwDevs.valueFor(module); 286 if (audioHwDevice != NULL) { 287 return audioHwDevice; 288 } 289 } 290 291 return NULL; 292} 293 294void AudioFlinger::dumpClients(int fd, const Vector<String16>& args __unused) 295{ 296 const size_t SIZE = 256; 297 char buffer[SIZE]; 298 String8 result; 299 300 result.append("Clients:\n"); 301 for (size_t i = 0; i < mClients.size(); ++i) { 302 sp<Client> client = mClients.valueAt(i).promote(); 303 if (client != 0) { 304 snprintf(buffer, SIZE, " pid: %d\n", client->pid()); 305 result.append(buffer); 306 } 307 } 308 309 result.append("Notification Clients:\n"); 310 for (size_t i = 0; i < mNotificationClients.size(); ++i) { 311 snprintf(buffer, SIZE, " pid: %d\n", mNotificationClients.keyAt(i)); 312 result.append(buffer); 313 } 314 315 result.append("Global session refs:\n"); 316 result.append(" session pid count\n"); 317 for (size_t i = 0; i < mAudioSessionRefs.size(); i++) { 318 AudioSessionRef *r = mAudioSessionRefs[i]; 319 snprintf(buffer, SIZE, " %7d %5d %5d\n", r->mSessionid, r->mPid, r->mCnt); 320 result.append(buffer); 321 } 322 write(fd, result.string(), result.size()); 323} 324 325 326void AudioFlinger::dumpInternals(int fd, const Vector<String16>& args __unused) 327{ 328 const size_t SIZE = 256; 329 char buffer[SIZE]; 330 String8 result; 331 hardware_call_state hardwareStatus = mHardwareStatus; 332 333 snprintf(buffer, SIZE, "Hardware status: %d\n" 334 "Standby Time mSec: %u\n", 335 hardwareStatus, 336 (uint32_t)(mStandbyTimeInNsecs / 1000000)); 337 result.append(buffer); 338 write(fd, result.string(), result.size()); 339} 340 341void AudioFlinger::dumpPermissionDenial(int fd, const Vector<String16>& args __unused) 342{ 343 const size_t SIZE = 256; 344 char buffer[SIZE]; 345 String8 result; 346 snprintf(buffer, SIZE, "Permission Denial: " 347 "can't dump AudioFlinger from pid=%d, uid=%d\n", 348 IPCThreadState::self()->getCallingPid(), 349 IPCThreadState::self()->getCallingUid()); 350 result.append(buffer); 351 write(fd, result.string(), result.size()); 352} 353 354bool AudioFlinger::dumpTryLock(Mutex& mutex) 355{ 356 bool locked = false; 357 for (int i = 0; i < kDumpLockRetries; ++i) { 358 if (mutex.tryLock() == NO_ERROR) { 359 locked = true; 360 break; 361 } 362 usleep(kDumpLockSleepUs); 363 } 364 return locked; 365} 366 367status_t AudioFlinger::dump(int fd, const Vector<String16>& args) 368{ 369 if (!dumpAllowed()) { 370 dumpPermissionDenial(fd, args); 371 } else { 372 // get state of hardware lock 373 bool hardwareLocked = dumpTryLock(mHardwareLock); 374 if (!hardwareLocked) { 375 String8 result(kHardwareLockedString); 376 write(fd, result.string(), result.size()); 377 } else { 378 mHardwareLock.unlock(); 379 } 380 381 bool locked = dumpTryLock(mLock); 382 383 // failed to lock - AudioFlinger is probably deadlocked 384 if (!locked) { 385 String8 result(kDeadlockedString); 386 write(fd, result.string(), result.size()); 387 } 388 389 bool clientLocked = dumpTryLock(mClientLock); 390 if (!clientLocked) { 391 String8 result(kClientLockedString); 392 write(fd, result.string(), result.size()); 393 } 394 dumpClients(fd, args); 395 if (clientLocked) { 396 mClientLock.unlock(); 397 } 398 399 dumpInternals(fd, args); 400 401 // dump playback threads 402 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 403 mPlaybackThreads.valueAt(i)->dump(fd, args); 404 } 405 406 // dump record threads 407 for (size_t i = 0; i < mRecordThreads.size(); i++) { 408 mRecordThreads.valueAt(i)->dump(fd, args); 409 } 410 411 // dump all hardware devs 412 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 413 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 414 dev->dump(dev, fd); 415 } 416 417#ifdef TEE_SINK 418 // dump the serially shared record tee sink 419 if (mRecordTeeSource != 0) { 420 dumpTee(fd, mRecordTeeSource); 421 } 422#endif 423 424 if (locked) { 425 mLock.unlock(); 426 } 427 428 // append a copy of media.log here by forwarding fd to it, but don't attempt 429 // to lookup the service if it's not running, as it will block for a second 430 if (mLogMemoryDealer != 0) { 431 sp<IBinder> binder = defaultServiceManager()->getService(String16("media.log")); 432 if (binder != 0) { 433 dprintf(fd, "\nmedia.log:\n"); 434 Vector<String16> args; 435 binder->dump(fd, args); 436 } 437 } 438 } 439 return NO_ERROR; 440} 441 442sp<AudioFlinger::Client> AudioFlinger::registerPid(pid_t pid) 443{ 444 Mutex::Autolock _cl(mClientLock); 445 // If pid is already in the mClients wp<> map, then use that entry 446 // (for which promote() is always != 0), otherwise create a new entry and Client. 447 sp<Client> client = mClients.valueFor(pid).promote(); 448 if (client == 0) { 449 client = new Client(this, pid); 450 mClients.add(pid, client); 451 } 452 453 return client; 454} 455 456sp<NBLog::Writer> AudioFlinger::newWriter_l(size_t size, const char *name) 457{ 458 // If there is no memory allocated for logs, return a dummy writer that does nothing 459 if (mLogMemoryDealer == 0) { 460 return new NBLog::Writer(); 461 } 462 sp<IBinder> binder = defaultServiceManager()->getService(String16("media.log")); 463 // Similarly if we can't contact the media.log service, also return a dummy writer 464 if (binder == 0) { 465 return new NBLog::Writer(); 466 } 467 sp<IMediaLogService> mediaLogService(interface_cast<IMediaLogService>(binder)); 468 sp<IMemory> shared = mLogMemoryDealer->allocate(NBLog::Timeline::sharedSize(size)); 469 // If allocation fails, consult the vector of previously unregistered writers 470 // and garbage-collect one or more them until an allocation succeeds 471 if (shared == 0) { 472 Mutex::Autolock _l(mUnregisteredWritersLock); 473 for (size_t count = mUnregisteredWriters.size(); count > 0; count--) { 474 { 475 // Pick the oldest stale writer to garbage-collect 476 sp<IMemory> iMemory(mUnregisteredWriters[0]->getIMemory()); 477 mUnregisteredWriters.removeAt(0); 478 mediaLogService->unregisterWriter(iMemory); 479 // Now the media.log remote reference to IMemory is gone. When our last local 480 // reference to IMemory also drops to zero at end of this block, 481 // the IMemory destructor will deallocate the region from mLogMemoryDealer. 482 } 483 // Re-attempt the allocation 484 shared = mLogMemoryDealer->allocate(NBLog::Timeline::sharedSize(size)); 485 if (shared != 0) { 486 goto success; 487 } 488 } 489 // Even after garbage-collecting all old writers, there is still not enough memory, 490 // so return a dummy writer 491 return new NBLog::Writer(); 492 } 493success: 494 mediaLogService->registerWriter(shared, size, name); 495 return new NBLog::Writer(size, shared); 496} 497 498void AudioFlinger::unregisterWriter(const sp<NBLog::Writer>& writer) 499{ 500 if (writer == 0) { 501 return; 502 } 503 sp<IMemory> iMemory(writer->getIMemory()); 504 if (iMemory == 0) { 505 return; 506 } 507 // Rather than removing the writer immediately, append it to a queue of old writers to 508 // be garbage-collected later. This allows us to continue to view old logs for a while. 509 Mutex::Autolock _l(mUnregisteredWritersLock); 510 mUnregisteredWriters.push(writer); 511} 512 513// IAudioFlinger interface 514 515 516sp<IAudioTrack> AudioFlinger::createTrack( 517 audio_stream_type_t streamType, 518 uint32_t sampleRate, 519 audio_format_t format, 520 audio_channel_mask_t channelMask, 521 size_t *frameCount, 522 IAudioFlinger::track_flags_t *flags, 523 const sp<IMemory>& sharedBuffer, 524 audio_io_handle_t output, 525 pid_t tid, 526 int *sessionId, 527 int clientUid, 528 status_t *status) 529{ 530 sp<PlaybackThread::Track> track; 531 sp<TrackHandle> trackHandle; 532 sp<Client> client; 533 status_t lStatus; 534 int lSessionId; 535 536 // client AudioTrack::set already implements AUDIO_STREAM_DEFAULT => AUDIO_STREAM_MUSIC, 537 // but if someone uses binder directly they could bypass that and cause us to crash 538 if (uint32_t(streamType) >= AUDIO_STREAM_CNT) { 539 ALOGE("createTrack() invalid stream type %d", streamType); 540 lStatus = BAD_VALUE; 541 goto Exit; 542 } 543 544 // further sample rate checks are performed by createTrack_l() depending on the thread type 545 if (sampleRate == 0) { 546 ALOGE("createTrack() invalid sample rate %u", sampleRate); 547 lStatus = BAD_VALUE; 548 goto Exit; 549 } 550 551 // further channel mask checks are performed by createTrack_l() depending on the thread type 552 if (!audio_is_output_channel(channelMask)) { 553 ALOGE("createTrack() invalid channel mask %#x", channelMask); 554 lStatus = BAD_VALUE; 555 goto Exit; 556 } 557 558 // further format checks are performed by createTrack_l() depending on the thread type 559 if (!audio_is_valid_format(format)) { 560 ALOGE("createTrack() invalid format %#x", format); 561 lStatus = BAD_VALUE; 562 goto Exit; 563 } 564 565 if (sharedBuffer != 0 && sharedBuffer->pointer() == NULL) { 566 ALOGE("createTrack() sharedBuffer is non-0 but has NULL pointer()"); 567 lStatus = BAD_VALUE; 568 goto Exit; 569 } 570 571 { 572 Mutex::Autolock _l(mLock); 573 PlaybackThread *thread = checkPlaybackThread_l(output); 574 if (thread == NULL) { 575 ALOGE("no playback thread found for output handle %d", output); 576 lStatus = BAD_VALUE; 577 goto Exit; 578 } 579 580 pid_t pid = IPCThreadState::self()->getCallingPid(); 581 client = registerPid(pid); 582 583 PlaybackThread *effectThread = NULL; 584 if (sessionId != NULL && *sessionId != AUDIO_SESSION_ALLOCATE) { 585 lSessionId = *sessionId; 586 // check if an effect chain with the same session ID is present on another 587 // output thread and move it here. 588 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 589 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 590 if (mPlaybackThreads.keyAt(i) != output) { 591 uint32_t sessions = t->hasAudioSession(lSessionId); 592 if (sessions & PlaybackThread::EFFECT_SESSION) { 593 effectThread = t.get(); 594 break; 595 } 596 } 597 } 598 } else { 599 // if no audio session id is provided, create one here 600 lSessionId = nextUniqueId(); 601 if (sessionId != NULL) { 602 *sessionId = lSessionId; 603 } 604 } 605 ALOGV("createTrack() lSessionId: %d", lSessionId); 606 607 track = thread->createTrack_l(client, streamType, sampleRate, format, 608 channelMask, frameCount, sharedBuffer, lSessionId, flags, tid, clientUid, &lStatus); 609 LOG_ALWAYS_FATAL_IF((lStatus == NO_ERROR) && (track == 0)); 610 // we don't abort yet if lStatus != NO_ERROR; there is still work to be done regardless 611 612 // move effect chain to this output thread if an effect on same session was waiting 613 // for a track to be created 614 if (lStatus == NO_ERROR && effectThread != NULL) { 615 // no risk of deadlock because AudioFlinger::mLock is held 616 Mutex::Autolock _dl(thread->mLock); 617 Mutex::Autolock _sl(effectThread->mLock); 618 moveEffectChain_l(lSessionId, effectThread, thread, true); 619 } 620 621 // Look for sync events awaiting for a session to be used. 622 for (size_t i = 0; i < mPendingSyncEvents.size(); i++) { 623 if (mPendingSyncEvents[i]->triggerSession() == lSessionId) { 624 if (thread->isValidSyncEvent(mPendingSyncEvents[i])) { 625 if (lStatus == NO_ERROR) { 626 (void) track->setSyncEvent(mPendingSyncEvents[i]); 627 } else { 628 mPendingSyncEvents[i]->cancel(); 629 } 630 mPendingSyncEvents.removeAt(i); 631 i--; 632 } 633 } 634 } 635 636 } 637 638 if (lStatus != NO_ERROR) { 639 // remove local strong reference to Client before deleting the Track so that the 640 // Client destructor is called by the TrackBase destructor with mClientLock held 641 Mutex::Autolock _cl(mClientLock); 642 client.clear(); 643 track.clear(); 644 goto Exit; 645 } 646 647 // return handle to client 648 trackHandle = new TrackHandle(track); 649 650Exit: 651 *status = lStatus; 652 return trackHandle; 653} 654 655uint32_t AudioFlinger::sampleRate(audio_io_handle_t output) const 656{ 657 Mutex::Autolock _l(mLock); 658 PlaybackThread *thread = checkPlaybackThread_l(output); 659 if (thread == NULL) { 660 ALOGW("sampleRate() unknown thread %d", output); 661 return 0; 662 } 663 return thread->sampleRate(); 664} 665 666int AudioFlinger::channelCount(audio_io_handle_t output) const 667{ 668 Mutex::Autolock _l(mLock); 669 PlaybackThread *thread = checkPlaybackThread_l(output); 670 if (thread == NULL) { 671 ALOGW("channelCount() unknown thread %d", output); 672 return 0; 673 } 674 return thread->channelCount(); 675} 676 677audio_format_t AudioFlinger::format(audio_io_handle_t output) const 678{ 679 Mutex::Autolock _l(mLock); 680 PlaybackThread *thread = checkPlaybackThread_l(output); 681 if (thread == NULL) { 682 ALOGW("format() unknown thread %d", output); 683 return AUDIO_FORMAT_INVALID; 684 } 685 return thread->format(); 686} 687 688size_t AudioFlinger::frameCount(audio_io_handle_t output) const 689{ 690 Mutex::Autolock _l(mLock); 691 PlaybackThread *thread = checkPlaybackThread_l(output); 692 if (thread == NULL) { 693 ALOGW("frameCount() unknown thread %d", output); 694 return 0; 695 } 696 // FIXME currently returns the normal mixer's frame count to avoid confusing legacy callers; 697 // should examine all callers and fix them to handle smaller counts 698 return thread->frameCount(); 699} 700 701uint32_t AudioFlinger::latency(audio_io_handle_t output) const 702{ 703 Mutex::Autolock _l(mLock); 704 PlaybackThread *thread = checkPlaybackThread_l(output); 705 if (thread == NULL) { 706 ALOGW("latency(): no playback thread found for output handle %d", output); 707 return 0; 708 } 709 return thread->latency(); 710} 711 712status_t AudioFlinger::setMasterVolume(float value) 713{ 714 status_t ret = initCheck(); 715 if (ret != NO_ERROR) { 716 return ret; 717 } 718 719 // check calling permissions 720 if (!settingsAllowed()) { 721 return PERMISSION_DENIED; 722 } 723 724 Mutex::Autolock _l(mLock); 725 mMasterVolume = value; 726 727 // Set master volume in the HALs which support it. 728 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 729 AutoMutex lock(mHardwareLock); 730 AudioHwDevice *dev = mAudioHwDevs.valueAt(i); 731 732 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 733 if (dev->canSetMasterVolume()) { 734 dev->hwDevice()->set_master_volume(dev->hwDevice(), value); 735 } 736 mHardwareStatus = AUDIO_HW_IDLE; 737 } 738 739 // Now set the master volume in each playback thread. Playback threads 740 // assigned to HALs which do not have master volume support will apply 741 // master volume during the mix operation. Threads with HALs which do 742 // support master volume will simply ignore the setting. 743 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 744 mPlaybackThreads.valueAt(i)->setMasterVolume(value); 745 746 return NO_ERROR; 747} 748 749status_t AudioFlinger::setMode(audio_mode_t mode) 750{ 751 status_t ret = initCheck(); 752 if (ret != NO_ERROR) { 753 return ret; 754 } 755 756 // check calling permissions 757 if (!settingsAllowed()) { 758 return PERMISSION_DENIED; 759 } 760 if (uint32_t(mode) >= AUDIO_MODE_CNT) { 761 ALOGW("Illegal value: setMode(%d)", mode); 762 return BAD_VALUE; 763 } 764 765 { // scope for the lock 766 AutoMutex lock(mHardwareLock); 767 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 768 mHardwareStatus = AUDIO_HW_SET_MODE; 769 ret = dev->set_mode(dev, mode); 770 mHardwareStatus = AUDIO_HW_IDLE; 771 } 772 773 if (NO_ERROR == ret) { 774 Mutex::Autolock _l(mLock); 775 mMode = mode; 776 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 777 mPlaybackThreads.valueAt(i)->setMode(mode); 778 } 779 780 return ret; 781} 782 783status_t AudioFlinger::setMicMute(bool state) 784{ 785 status_t ret = initCheck(); 786 if (ret != NO_ERROR) { 787 return ret; 788 } 789 790 // check calling permissions 791 if (!settingsAllowed()) { 792 return PERMISSION_DENIED; 793 } 794 795 AutoMutex lock(mHardwareLock); 796 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 797 mHardwareStatus = AUDIO_HW_SET_MIC_MUTE; 798 ret = dev->set_mic_mute(dev, state); 799 mHardwareStatus = AUDIO_HW_IDLE; 800 return ret; 801} 802 803bool AudioFlinger::getMicMute() const 804{ 805 status_t ret = initCheck(); 806 if (ret != NO_ERROR) { 807 return false; 808 } 809 810 bool state = AUDIO_MODE_INVALID; 811 AutoMutex lock(mHardwareLock); 812 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 813 mHardwareStatus = AUDIO_HW_GET_MIC_MUTE; 814 dev->get_mic_mute(dev, &state); 815 mHardwareStatus = AUDIO_HW_IDLE; 816 return state; 817} 818 819status_t AudioFlinger::setMasterMute(bool muted) 820{ 821 status_t ret = initCheck(); 822 if (ret != NO_ERROR) { 823 return ret; 824 } 825 826 // check calling permissions 827 if (!settingsAllowed()) { 828 return PERMISSION_DENIED; 829 } 830 831 Mutex::Autolock _l(mLock); 832 mMasterMute = muted; 833 834 // Set master mute in the HALs which support it. 835 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 836 AutoMutex lock(mHardwareLock); 837 AudioHwDevice *dev = mAudioHwDevs.valueAt(i); 838 839 mHardwareStatus = AUDIO_HW_SET_MASTER_MUTE; 840 if (dev->canSetMasterMute()) { 841 dev->hwDevice()->set_master_mute(dev->hwDevice(), muted); 842 } 843 mHardwareStatus = AUDIO_HW_IDLE; 844 } 845 846 // Now set the master mute in each playback thread. Playback threads 847 // assigned to HALs which do not have master mute support will apply master 848 // mute during the mix operation. Threads with HALs which do support master 849 // mute will simply ignore the setting. 850 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 851 mPlaybackThreads.valueAt(i)->setMasterMute(muted); 852 853 return NO_ERROR; 854} 855 856float AudioFlinger::masterVolume() const 857{ 858 Mutex::Autolock _l(mLock); 859 return masterVolume_l(); 860} 861 862bool AudioFlinger::masterMute() const 863{ 864 Mutex::Autolock _l(mLock); 865 return masterMute_l(); 866} 867 868float AudioFlinger::masterVolume_l() const 869{ 870 return mMasterVolume; 871} 872 873bool AudioFlinger::masterMute_l() const 874{ 875 return mMasterMute; 876} 877 878status_t AudioFlinger::setStreamVolume(audio_stream_type_t stream, float value, 879 audio_io_handle_t output) 880{ 881 // check calling permissions 882 if (!settingsAllowed()) { 883 return PERMISSION_DENIED; 884 } 885 886 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 887 ALOGE("setStreamVolume() invalid stream %d", stream); 888 return BAD_VALUE; 889 } 890 891 AutoMutex lock(mLock); 892 PlaybackThread *thread = NULL; 893 if (output != AUDIO_IO_HANDLE_NONE) { 894 thread = checkPlaybackThread_l(output); 895 if (thread == NULL) { 896 return BAD_VALUE; 897 } 898 } 899 900 mStreamTypes[stream].volume = value; 901 902 if (thread == NULL) { 903 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 904 mPlaybackThreads.valueAt(i)->setStreamVolume(stream, value); 905 } 906 } else { 907 thread->setStreamVolume(stream, value); 908 } 909 910 return NO_ERROR; 911} 912 913status_t AudioFlinger::setStreamMute(audio_stream_type_t stream, bool muted) 914{ 915 // check calling permissions 916 if (!settingsAllowed()) { 917 return PERMISSION_DENIED; 918 } 919 920 if (uint32_t(stream) >= AUDIO_STREAM_CNT || 921 uint32_t(stream) == AUDIO_STREAM_ENFORCED_AUDIBLE) { 922 ALOGE("setStreamMute() invalid stream %d", stream); 923 return BAD_VALUE; 924 } 925 926 AutoMutex lock(mLock); 927 mStreamTypes[stream].mute = muted; 928 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 929 mPlaybackThreads.valueAt(i)->setStreamMute(stream, muted); 930 931 return NO_ERROR; 932} 933 934float AudioFlinger::streamVolume(audio_stream_type_t stream, audio_io_handle_t output) const 935{ 936 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 937 return 0.0f; 938 } 939 940 AutoMutex lock(mLock); 941 float volume; 942 if (output != AUDIO_IO_HANDLE_NONE) { 943 PlaybackThread *thread = checkPlaybackThread_l(output); 944 if (thread == NULL) { 945 return 0.0f; 946 } 947 volume = thread->streamVolume(stream); 948 } else { 949 volume = streamVolume_l(stream); 950 } 951 952 return volume; 953} 954 955bool AudioFlinger::streamMute(audio_stream_type_t stream) const 956{ 957 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 958 return true; 959 } 960 961 AutoMutex lock(mLock); 962 return streamMute_l(stream); 963} 964 965status_t AudioFlinger::setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs) 966{ 967 ALOGV("setParameters(): io %d, keyvalue %s, calling pid %d", 968 ioHandle, keyValuePairs.string(), IPCThreadState::self()->getCallingPid()); 969 970 // check calling permissions 971 if (!settingsAllowed()) { 972 return PERMISSION_DENIED; 973 } 974 975 // AUDIO_IO_HANDLE_NONE means the parameters are global to the audio hardware interface 976 if (ioHandle == AUDIO_IO_HANDLE_NONE) { 977 Mutex::Autolock _l(mLock); 978 status_t final_result = NO_ERROR; 979 { 980 AutoMutex lock(mHardwareLock); 981 mHardwareStatus = AUDIO_HW_SET_PARAMETER; 982 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 983 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 984 status_t result = dev->set_parameters(dev, keyValuePairs.string()); 985 final_result = result ?: final_result; 986 } 987 mHardwareStatus = AUDIO_HW_IDLE; 988 } 989 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 990 AudioParameter param = AudioParameter(keyValuePairs); 991 String8 value; 992 if (param.get(String8(AUDIO_PARAMETER_KEY_BT_NREC), value) == NO_ERROR) { 993 bool btNrecIsOff = (value == AUDIO_PARAMETER_VALUE_OFF); 994 if (mBtNrecIsOff != btNrecIsOff) { 995 for (size_t i = 0; i < mRecordThreads.size(); i++) { 996 sp<RecordThread> thread = mRecordThreads.valueAt(i); 997 audio_devices_t device = thread->inDevice(); 998 bool suspend = audio_is_bluetooth_sco_device(device) && btNrecIsOff; 999 // collect all of the thread's session IDs 1000 KeyedVector<int, bool> ids = thread->sessionIds(); 1001 // suspend effects associated with those session IDs 1002 for (size_t j = 0; j < ids.size(); ++j) { 1003 int sessionId = ids.keyAt(j); 1004 thread->setEffectSuspended(FX_IID_AEC, 1005 suspend, 1006 sessionId); 1007 thread->setEffectSuspended(FX_IID_NS, 1008 suspend, 1009 sessionId); 1010 } 1011 } 1012 mBtNrecIsOff = btNrecIsOff; 1013 } 1014 } 1015 String8 screenState; 1016 if (param.get(String8(AudioParameter::keyScreenState), screenState) == NO_ERROR) { 1017 bool isOff = screenState == "off"; 1018 if (isOff != (AudioFlinger::mScreenState & 1)) { 1019 AudioFlinger::mScreenState = ((AudioFlinger::mScreenState & ~1) + 2) | isOff; 1020 } 1021 } 1022 return final_result; 1023 } 1024 1025 // hold a strong ref on thread in case closeOutput() or closeInput() is called 1026 // and the thread is exited once the lock is released 1027 sp<ThreadBase> thread; 1028 { 1029 Mutex::Autolock _l(mLock); 1030 thread = checkPlaybackThread_l(ioHandle); 1031 if (thread == 0) { 1032 thread = checkRecordThread_l(ioHandle); 1033 } else if (thread == primaryPlaybackThread_l()) { 1034 // indicate output device change to all input threads for pre processing 1035 AudioParameter param = AudioParameter(keyValuePairs); 1036 int value; 1037 if ((param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) && 1038 (value != 0)) { 1039 for (size_t i = 0; i < mRecordThreads.size(); i++) { 1040 mRecordThreads.valueAt(i)->setParameters(keyValuePairs); 1041 } 1042 } 1043 } 1044 } 1045 if (thread != 0) { 1046 return thread->setParameters(keyValuePairs); 1047 } 1048 return BAD_VALUE; 1049} 1050 1051String8 AudioFlinger::getParameters(audio_io_handle_t ioHandle, const String8& keys) const 1052{ 1053 ALOGVV("getParameters() io %d, keys %s, calling pid %d", 1054 ioHandle, keys.string(), IPCThreadState::self()->getCallingPid()); 1055 1056 Mutex::Autolock _l(mLock); 1057 1058 if (ioHandle == AUDIO_IO_HANDLE_NONE) { 1059 String8 out_s8; 1060 1061 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 1062 char *s; 1063 { 1064 AutoMutex lock(mHardwareLock); 1065 mHardwareStatus = AUDIO_HW_GET_PARAMETER; 1066 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 1067 s = dev->get_parameters(dev, keys.string()); 1068 mHardwareStatus = AUDIO_HW_IDLE; 1069 } 1070 out_s8 += String8(s ? s : ""); 1071 free(s); 1072 } 1073 return out_s8; 1074 } 1075 1076 PlaybackThread *playbackThread = checkPlaybackThread_l(ioHandle); 1077 if (playbackThread != NULL) { 1078 return playbackThread->getParameters(keys); 1079 } 1080 RecordThread *recordThread = checkRecordThread_l(ioHandle); 1081 if (recordThread != NULL) { 1082 return recordThread->getParameters(keys); 1083 } 1084 return String8(""); 1085} 1086 1087size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, audio_format_t format, 1088 audio_channel_mask_t channelMask) const 1089{ 1090 status_t ret = initCheck(); 1091 if (ret != NO_ERROR) { 1092 return 0; 1093 } 1094 1095 AutoMutex lock(mHardwareLock); 1096 mHardwareStatus = AUDIO_HW_GET_INPUT_BUFFER_SIZE; 1097 struct audio_config config; 1098 memset(&config, 0, sizeof(config)); 1099 config.sample_rate = sampleRate; 1100 config.channel_mask = channelMask; 1101 config.format = format; 1102 1103 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 1104 size_t size = dev->get_input_buffer_size(dev, &config); 1105 mHardwareStatus = AUDIO_HW_IDLE; 1106 return size; 1107} 1108 1109uint32_t AudioFlinger::getInputFramesLost(audio_io_handle_t ioHandle) const 1110{ 1111 Mutex::Autolock _l(mLock); 1112 1113 RecordThread *recordThread = checkRecordThread_l(ioHandle); 1114 if (recordThread != NULL) { 1115 return recordThread->getInputFramesLost(); 1116 } 1117 return 0; 1118} 1119 1120status_t AudioFlinger::setVoiceVolume(float value) 1121{ 1122 status_t ret = initCheck(); 1123 if (ret != NO_ERROR) { 1124 return ret; 1125 } 1126 1127 // check calling permissions 1128 if (!settingsAllowed()) { 1129 return PERMISSION_DENIED; 1130 } 1131 1132 AutoMutex lock(mHardwareLock); 1133 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 1134 mHardwareStatus = AUDIO_HW_SET_VOICE_VOLUME; 1135 ret = dev->set_voice_volume(dev, value); 1136 mHardwareStatus = AUDIO_HW_IDLE; 1137 1138 return ret; 1139} 1140 1141status_t AudioFlinger::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, 1142 audio_io_handle_t output) const 1143{ 1144 status_t status; 1145 1146 Mutex::Autolock _l(mLock); 1147 1148 PlaybackThread *playbackThread = checkPlaybackThread_l(output); 1149 if (playbackThread != NULL) { 1150 return playbackThread->getRenderPosition(halFrames, dspFrames); 1151 } 1152 1153 return BAD_VALUE; 1154} 1155 1156void AudioFlinger::registerClient(const sp<IAudioFlingerClient>& client) 1157{ 1158 Mutex::Autolock _l(mLock); 1159 bool clientAdded = false; 1160 { 1161 Mutex::Autolock _cl(mClientLock); 1162 1163 pid_t pid = IPCThreadState::self()->getCallingPid(); 1164 if (mNotificationClients.indexOfKey(pid) < 0) { 1165 sp<NotificationClient> notificationClient = new NotificationClient(this, 1166 client, 1167 pid); 1168 ALOGV("registerClient() client %p, pid %d", notificationClient.get(), pid); 1169 1170 mNotificationClients.add(pid, notificationClient); 1171 1172 sp<IBinder> binder = client->asBinder(); 1173 binder->linkToDeath(notificationClient); 1174 clientAdded = true; 1175 } 1176 } 1177 1178 // mClientLock should not be held here because ThreadBase::sendIoConfigEvent() will lock the 1179 // ThreadBase mutex and teh locknig order is ThreadBase::mLock then AudioFlinger::mClientLock. 1180 if (clientAdded) { 1181 // the config change is always sent from playback or record threads to avoid deadlock 1182 // with AudioSystem::gLock 1183 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1184 mPlaybackThreads.valueAt(i)->sendIoConfigEvent(AudioSystem::OUTPUT_OPENED); 1185 } 1186 1187 for (size_t i = 0; i < mRecordThreads.size(); i++) { 1188 mRecordThreads.valueAt(i)->sendIoConfigEvent(AudioSystem::INPUT_OPENED); 1189 } 1190 } 1191} 1192 1193void AudioFlinger::removeNotificationClient(pid_t pid) 1194{ 1195 Mutex::Autolock _l(mLock); 1196 { 1197 Mutex::Autolock _cl(mClientLock); 1198 mNotificationClients.removeItem(pid); 1199 } 1200 1201 ALOGV("%d died, releasing its sessions", pid); 1202 size_t num = mAudioSessionRefs.size(); 1203 bool removed = false; 1204 for (size_t i = 0; i< num; ) { 1205 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 1206 ALOGV(" pid %d @ %d", ref->mPid, i); 1207 if (ref->mPid == pid) { 1208 ALOGV(" removing entry for pid %d session %d", pid, ref->mSessionid); 1209 mAudioSessionRefs.removeAt(i); 1210 delete ref; 1211 removed = true; 1212 num--; 1213 } else { 1214 i++; 1215 } 1216 } 1217 if (removed) { 1218 purgeStaleEffects_l(); 1219 } 1220} 1221 1222void AudioFlinger::audioConfigChanged(int event, audio_io_handle_t ioHandle, const void *param2) 1223{ 1224 Mutex::Autolock _l(mClientLock); 1225 size_t size = mNotificationClients.size(); 1226 for (size_t i = 0; i < size; i++) { 1227 mNotificationClients.valueAt(i)->audioFlingerClient()->ioConfigChanged(event, 1228 ioHandle, 1229 param2); 1230 } 1231} 1232 1233// removeClient_l() must be called with AudioFlinger::mClientLock held 1234void AudioFlinger::removeClient_l(pid_t pid) 1235{ 1236 ALOGV("removeClient_l() pid %d, calling pid %d", pid, 1237 IPCThreadState::self()->getCallingPid()); 1238 mClients.removeItem(pid); 1239} 1240 1241// getEffectThread_l() must be called with AudioFlinger::mLock held 1242sp<AudioFlinger::PlaybackThread> AudioFlinger::getEffectThread_l(int sessionId, int EffectId) 1243{ 1244 sp<PlaybackThread> thread; 1245 1246 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1247 if (mPlaybackThreads.valueAt(i)->getEffect(sessionId, EffectId) != 0) { 1248 ALOG_ASSERT(thread == 0); 1249 thread = mPlaybackThreads.valueAt(i); 1250 } 1251 } 1252 1253 return thread; 1254} 1255 1256 1257 1258// ---------------------------------------------------------------------------- 1259 1260AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid) 1261 : RefBase(), 1262 mAudioFlinger(audioFlinger), 1263 // FIXME should be a "k" constant not hard-coded, in .h or ro. property, see 4 lines below 1264 mMemoryDealer(new MemoryDealer(1024*1024, "AudioFlinger::Client")), 1265 mPid(pid), 1266 mTimedTrackCount(0) 1267{ 1268 // 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer 1269} 1270 1271// Client destructor must be called with AudioFlinger::mClientLock held 1272AudioFlinger::Client::~Client() 1273{ 1274 mAudioFlinger->removeClient_l(mPid); 1275} 1276 1277sp<MemoryDealer> AudioFlinger::Client::heap() const 1278{ 1279 return mMemoryDealer; 1280} 1281 1282// Reserve one of the limited slots for a timed audio track associated 1283// with this client 1284bool AudioFlinger::Client::reserveTimedTrack() 1285{ 1286 const int kMaxTimedTracksPerClient = 4; 1287 1288 Mutex::Autolock _l(mTimedTrackLock); 1289 1290 if (mTimedTrackCount >= kMaxTimedTracksPerClient) { 1291 ALOGW("can not create timed track - pid %d has exceeded the limit", 1292 mPid); 1293 return false; 1294 } 1295 1296 mTimedTrackCount++; 1297 return true; 1298} 1299 1300// Release a slot for a timed audio track 1301void AudioFlinger::Client::releaseTimedTrack() 1302{ 1303 Mutex::Autolock _l(mTimedTrackLock); 1304 mTimedTrackCount--; 1305} 1306 1307// ---------------------------------------------------------------------------- 1308 1309AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger, 1310 const sp<IAudioFlingerClient>& client, 1311 pid_t pid) 1312 : mAudioFlinger(audioFlinger), mPid(pid), mAudioFlingerClient(client) 1313{ 1314} 1315 1316AudioFlinger::NotificationClient::~NotificationClient() 1317{ 1318} 1319 1320void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who __unused) 1321{ 1322 sp<NotificationClient> keep(this); 1323 mAudioFlinger->removeNotificationClient(mPid); 1324} 1325 1326 1327// ---------------------------------------------------------------------------- 1328 1329static bool deviceRequiresCaptureAudioOutputPermission(audio_devices_t inDevice) { 1330 return audio_is_remote_submix_device(inDevice); 1331} 1332 1333sp<IAudioRecord> AudioFlinger::openRecord( 1334 audio_io_handle_t input, 1335 uint32_t sampleRate, 1336 audio_format_t format, 1337 audio_channel_mask_t channelMask, 1338 size_t *frameCount, 1339 IAudioFlinger::track_flags_t *flags, 1340 pid_t tid, 1341 int *sessionId, 1342 sp<IMemory>& cblk, 1343 sp<IMemory>& buffers, 1344 status_t *status) 1345{ 1346 sp<RecordThread::RecordTrack> recordTrack; 1347 sp<RecordHandle> recordHandle; 1348 sp<Client> client; 1349 status_t lStatus; 1350 int lSessionId; 1351 1352 cblk.clear(); 1353 buffers.clear(); 1354 1355 // check calling permissions 1356 if (!recordingAllowed()) { 1357 ALOGE("openRecord() permission denied: recording not allowed"); 1358 lStatus = PERMISSION_DENIED; 1359 goto Exit; 1360 } 1361 1362 // further sample rate checks are performed by createRecordTrack_l() 1363 if (sampleRate == 0) { 1364 ALOGE("openRecord() invalid sample rate %u", sampleRate); 1365 lStatus = BAD_VALUE; 1366 goto Exit; 1367 } 1368 1369 // we don't yet support anything other than 16-bit PCM 1370 if (!(audio_is_valid_format(format) && 1371 audio_is_linear_pcm(format) && format == AUDIO_FORMAT_PCM_16_BIT)) { 1372 ALOGE("openRecord() invalid format %#x", format); 1373 lStatus = BAD_VALUE; 1374 goto Exit; 1375 } 1376 1377 // further channel mask checks are performed by createRecordTrack_l() 1378 if (!audio_is_input_channel(channelMask)) { 1379 ALOGE("openRecord() invalid channel mask %#x", channelMask); 1380 lStatus = BAD_VALUE; 1381 goto Exit; 1382 } 1383 1384 { 1385 Mutex::Autolock _l(mLock); 1386 RecordThread *thread = checkRecordThread_l(input); 1387 if (thread == NULL) { 1388 ALOGE("openRecord() checkRecordThread_l failed"); 1389 lStatus = BAD_VALUE; 1390 goto Exit; 1391 } 1392 1393 if (deviceRequiresCaptureAudioOutputPermission(thread->inDevice()) 1394 && !captureAudioOutputAllowed()) { 1395 ALOGE("openRecord() permission denied: capture not allowed"); 1396 lStatus = PERMISSION_DENIED; 1397 goto Exit; 1398 } 1399 1400 pid_t pid = IPCThreadState::self()->getCallingPid(); 1401 client = registerPid(pid); 1402 1403 if (sessionId != NULL && *sessionId != AUDIO_SESSION_ALLOCATE) { 1404 lSessionId = *sessionId; 1405 } else { 1406 // if no audio session id is provided, create one here 1407 lSessionId = nextUniqueId(); 1408 if (sessionId != NULL) { 1409 *sessionId = lSessionId; 1410 } 1411 } 1412 ALOGV("openRecord() lSessionId: %d", lSessionId); 1413 1414 // TODO: the uid should be passed in as a parameter to openRecord 1415 recordTrack = thread->createRecordTrack_l(client, sampleRate, format, channelMask, 1416 frameCount, lSessionId, 1417 IPCThreadState::self()->getCallingUid(), 1418 flags, tid, &lStatus); 1419 LOG_ALWAYS_FATAL_IF((lStatus == NO_ERROR) && (recordTrack == 0)); 1420 } 1421 1422 if (lStatus != NO_ERROR) { 1423 // remove local strong reference to Client before deleting the RecordTrack so that the 1424 // Client destructor is called by the TrackBase destructor with mClientLock held 1425 Mutex::Autolock _cl(mClientLock); 1426 client.clear(); 1427 recordTrack.clear(); 1428 goto Exit; 1429 } 1430 1431 cblk = recordTrack->getCblk(); 1432 buffers = recordTrack->getBuffers(); 1433 1434 // return handle to client 1435 recordHandle = new RecordHandle(recordTrack); 1436 1437Exit: 1438 *status = lStatus; 1439 return recordHandle; 1440} 1441 1442 1443 1444// ---------------------------------------------------------------------------- 1445 1446audio_module_handle_t AudioFlinger::loadHwModule(const char *name) 1447{ 1448 if (!settingsAllowed()) { 1449 return 0; 1450 } 1451 Mutex::Autolock _l(mLock); 1452 return loadHwModule_l(name); 1453} 1454 1455// loadHwModule_l() must be called with AudioFlinger::mLock held 1456audio_module_handle_t AudioFlinger::loadHwModule_l(const char *name) 1457{ 1458 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 1459 if (strncmp(mAudioHwDevs.valueAt(i)->moduleName(), name, strlen(name)) == 0) { 1460 ALOGW("loadHwModule() module %s already loaded", name); 1461 return mAudioHwDevs.keyAt(i); 1462 } 1463 } 1464 1465 audio_hw_device_t *dev; 1466 1467 int rc = load_audio_interface(name, &dev); 1468 if (rc) { 1469 ALOGI("loadHwModule() error %d loading module %s ", rc, name); 1470 return 0; 1471 } 1472 1473 mHardwareStatus = AUDIO_HW_INIT; 1474 rc = dev->init_check(dev); 1475 mHardwareStatus = AUDIO_HW_IDLE; 1476 if (rc) { 1477 ALOGI("loadHwModule() init check error %d for module %s ", rc, name); 1478 return 0; 1479 } 1480 1481 // Check and cache this HAL's level of support for master mute and master 1482 // volume. If this is the first HAL opened, and it supports the get 1483 // methods, use the initial values provided by the HAL as the current 1484 // master mute and volume settings. 1485 1486 AudioHwDevice::Flags flags = static_cast<AudioHwDevice::Flags>(0); 1487 { // scope for auto-lock pattern 1488 AutoMutex lock(mHardwareLock); 1489 1490 if (0 == mAudioHwDevs.size()) { 1491 mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME; 1492 if (NULL != dev->get_master_volume) { 1493 float mv; 1494 if (OK == dev->get_master_volume(dev, &mv)) { 1495 mMasterVolume = mv; 1496 } 1497 } 1498 1499 mHardwareStatus = AUDIO_HW_GET_MASTER_MUTE; 1500 if (NULL != dev->get_master_mute) { 1501 bool mm; 1502 if (OK == dev->get_master_mute(dev, &mm)) { 1503 mMasterMute = mm; 1504 } 1505 } 1506 } 1507 1508 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 1509 if ((NULL != dev->set_master_volume) && 1510 (OK == dev->set_master_volume(dev, mMasterVolume))) { 1511 flags = static_cast<AudioHwDevice::Flags>(flags | 1512 AudioHwDevice::AHWD_CAN_SET_MASTER_VOLUME); 1513 } 1514 1515 mHardwareStatus = AUDIO_HW_SET_MASTER_MUTE; 1516 if ((NULL != dev->set_master_mute) && 1517 (OK == dev->set_master_mute(dev, mMasterMute))) { 1518 flags = static_cast<AudioHwDevice::Flags>(flags | 1519 AudioHwDevice::AHWD_CAN_SET_MASTER_MUTE); 1520 } 1521 1522 mHardwareStatus = AUDIO_HW_IDLE; 1523 } 1524 1525 audio_module_handle_t handle = nextUniqueId(); 1526 mAudioHwDevs.add(handle, new AudioHwDevice(name, dev, flags)); 1527 1528 ALOGI("loadHwModule() Loaded %s audio interface from %s (%s) handle %d", 1529 name, dev->common.module->name, dev->common.module->id, handle); 1530 1531 return handle; 1532 1533} 1534 1535// ---------------------------------------------------------------------------- 1536 1537uint32_t AudioFlinger::getPrimaryOutputSamplingRate() 1538{ 1539 Mutex::Autolock _l(mLock); 1540 PlaybackThread *thread = primaryPlaybackThread_l(); 1541 return thread != NULL ? thread->sampleRate() : 0; 1542} 1543 1544size_t AudioFlinger::getPrimaryOutputFrameCount() 1545{ 1546 Mutex::Autolock _l(mLock); 1547 PlaybackThread *thread = primaryPlaybackThread_l(); 1548 return thread != NULL ? thread->frameCountHAL() : 0; 1549} 1550 1551// ---------------------------------------------------------------------------- 1552 1553status_t AudioFlinger::setLowRamDevice(bool isLowRamDevice) 1554{ 1555 uid_t uid = IPCThreadState::self()->getCallingUid(); 1556 if (uid != AID_SYSTEM) { 1557 return PERMISSION_DENIED; 1558 } 1559 Mutex::Autolock _l(mLock); 1560 if (mIsDeviceTypeKnown) { 1561 return INVALID_OPERATION; 1562 } 1563 mIsLowRamDevice = isLowRamDevice; 1564 mIsDeviceTypeKnown = true; 1565 return NO_ERROR; 1566} 1567 1568// ---------------------------------------------------------------------------- 1569 1570audio_io_handle_t AudioFlinger::openOutput(audio_module_handle_t module, 1571 audio_devices_t *pDevices, 1572 uint32_t *pSamplingRate, 1573 audio_format_t *pFormat, 1574 audio_channel_mask_t *pChannelMask, 1575 uint32_t *pLatencyMs, 1576 audio_output_flags_t flags, 1577 const audio_offload_info_t *offloadInfo) 1578{ 1579 struct audio_config config; 1580 memset(&config, 0, sizeof(config)); 1581 config.sample_rate = (pSamplingRate != NULL) ? *pSamplingRate : 0; 1582 config.channel_mask = (pChannelMask != NULL) ? *pChannelMask : 0; 1583 config.format = (pFormat != NULL) ? *pFormat : AUDIO_FORMAT_DEFAULT; 1584 if (offloadInfo != NULL) { 1585 config.offload_info = *offloadInfo; 1586 } 1587 1588 ALOGV("openOutput(), module %d Device %x, SamplingRate %d, Format %#08x, Channels %x, flags %x", 1589 module, 1590 (pDevices != NULL) ? *pDevices : 0, 1591 config.sample_rate, 1592 config.format, 1593 config.channel_mask, 1594 flags); 1595 ALOGV("openOutput(), offloadInfo %p version 0x%04x", 1596 offloadInfo, offloadInfo == NULL ? -1 : offloadInfo->version); 1597 1598 if (pDevices == NULL || *pDevices == AUDIO_DEVICE_NONE) { 1599 return AUDIO_IO_HANDLE_NONE; 1600 } 1601 1602 Mutex::Autolock _l(mLock); 1603 1604 AudioHwDevice *outHwDev = findSuitableHwDev_l(module, *pDevices); 1605 if (outHwDev == NULL) { 1606 return AUDIO_IO_HANDLE_NONE; 1607 } 1608 1609 audio_hw_device_t *hwDevHal = outHwDev->hwDevice(); 1610 audio_io_handle_t id = nextUniqueId(); 1611 1612 mHardwareStatus = AUDIO_HW_OUTPUT_OPEN; 1613 1614 audio_stream_out_t *outStream = NULL; 1615 status_t status = hwDevHal->open_output_stream(hwDevHal, 1616 id, 1617 *pDevices, 1618 (audio_output_flags_t)flags, 1619 &config, 1620 &outStream); 1621 1622 mHardwareStatus = AUDIO_HW_IDLE; 1623 ALOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %#08x, " 1624 "Channels %x, status %d", 1625 outStream, 1626 config.sample_rate, 1627 config.format, 1628 config.channel_mask, 1629 status); 1630 1631 if (status == NO_ERROR && outStream != NULL) { 1632 AudioStreamOut *output = new AudioStreamOut(outHwDev, outStream, flags); 1633 1634 PlaybackThread *thread; 1635 if (flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) { 1636 thread = new OffloadThread(this, output, id, *pDevices); 1637 ALOGV("openOutput() created offload output: ID %d thread %p", id, thread); 1638 } else if ((flags & AUDIO_OUTPUT_FLAG_DIRECT) || 1639 (config.format != AUDIO_FORMAT_PCM_16_BIT) || 1640 (config.channel_mask != AUDIO_CHANNEL_OUT_STEREO)) { 1641 thread = new DirectOutputThread(this, output, id, *pDevices); 1642 ALOGV("openOutput() created direct output: ID %d thread %p", id, thread); 1643 } else { 1644 thread = new MixerThread(this, output, id, *pDevices); 1645 ALOGV("openOutput() created mixer output: ID %d thread %p", id, thread); 1646 } 1647 mPlaybackThreads.add(id, thread); 1648 1649 if (pSamplingRate != NULL) { 1650 *pSamplingRate = config.sample_rate; 1651 } 1652 if (pFormat != NULL) { 1653 *pFormat = config.format; 1654 } 1655 if (pChannelMask != NULL) { 1656 *pChannelMask = config.channel_mask; 1657 } 1658 if (pLatencyMs != NULL) { 1659 *pLatencyMs = thread->latency(); 1660 } 1661 1662 // notify client processes of the new output creation 1663 thread->audioConfigChanged(AudioSystem::OUTPUT_OPENED); 1664 1665 // the first primary output opened designates the primary hw device 1666 if ((mPrimaryHardwareDev == NULL) && (flags & AUDIO_OUTPUT_FLAG_PRIMARY)) { 1667 ALOGI("Using module %d has the primary audio interface", module); 1668 mPrimaryHardwareDev = outHwDev; 1669 1670 AutoMutex lock(mHardwareLock); 1671 mHardwareStatus = AUDIO_HW_SET_MODE; 1672 hwDevHal->set_mode(hwDevHal, mMode); 1673 mHardwareStatus = AUDIO_HW_IDLE; 1674 } 1675 return id; 1676 } 1677 1678 return AUDIO_IO_HANDLE_NONE; 1679} 1680 1681audio_io_handle_t AudioFlinger::openDuplicateOutput(audio_io_handle_t output1, 1682 audio_io_handle_t output2) 1683{ 1684 Mutex::Autolock _l(mLock); 1685 MixerThread *thread1 = checkMixerThread_l(output1); 1686 MixerThread *thread2 = checkMixerThread_l(output2); 1687 1688 if (thread1 == NULL || thread2 == NULL) { 1689 ALOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1, 1690 output2); 1691 return AUDIO_IO_HANDLE_NONE; 1692 } 1693 1694 audio_io_handle_t id = nextUniqueId(); 1695 DuplicatingThread *thread = new DuplicatingThread(this, thread1, id); 1696 thread->addOutputTrack(thread2); 1697 mPlaybackThreads.add(id, thread); 1698 // notify client processes of the new output creation 1699 thread->audioConfigChanged(AudioSystem::OUTPUT_OPENED); 1700 return id; 1701} 1702 1703status_t AudioFlinger::closeOutput(audio_io_handle_t output) 1704{ 1705 return closeOutput_nonvirtual(output); 1706} 1707 1708status_t AudioFlinger::closeOutput_nonvirtual(audio_io_handle_t output) 1709{ 1710 // keep strong reference on the playback thread so that 1711 // it is not destroyed while exit() is executed 1712 sp<PlaybackThread> thread; 1713 { 1714 Mutex::Autolock _l(mLock); 1715 thread = checkPlaybackThread_l(output); 1716 if (thread == NULL) { 1717 return BAD_VALUE; 1718 } 1719 1720 ALOGV("closeOutput() %d", output); 1721 1722 if (thread->type() == ThreadBase::MIXER) { 1723 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1724 if (mPlaybackThreads.valueAt(i)->type() == ThreadBase::DUPLICATING) { 1725 DuplicatingThread *dupThread = 1726 (DuplicatingThread *)mPlaybackThreads.valueAt(i).get(); 1727 dupThread->removeOutputTrack((MixerThread *)thread.get()); 1728 1729 } 1730 } 1731 } 1732 1733 1734 mPlaybackThreads.removeItem(output); 1735 // save all effects to the default thread 1736 if (mPlaybackThreads.size()) { 1737 PlaybackThread *dstThread = checkPlaybackThread_l(mPlaybackThreads.keyAt(0)); 1738 if (dstThread != NULL) { 1739 // audioflinger lock is held here so the acquisition order of thread locks does not 1740 // matter 1741 Mutex::Autolock _dl(dstThread->mLock); 1742 Mutex::Autolock _sl(thread->mLock); 1743 Vector< sp<EffectChain> > effectChains = thread->getEffectChains_l(); 1744 for (size_t i = 0; i < effectChains.size(); i ++) { 1745 moveEffectChain_l(effectChains[i]->sessionId(), thread.get(), dstThread, true); 1746 } 1747 } 1748 } 1749 audioConfigChanged(AudioSystem::OUTPUT_CLOSED, output, NULL); 1750 } 1751 thread->exit(); 1752 // The thread entity (active unit of execution) is no longer running here, 1753 // but the ThreadBase container still exists. 1754 1755 if (thread->type() != ThreadBase::DUPLICATING) { 1756 AudioStreamOut *out = thread->clearOutput(); 1757 ALOG_ASSERT(out != NULL, "out shouldn't be NULL"); 1758 // from now on thread->mOutput is NULL 1759 out->hwDev()->close_output_stream(out->hwDev(), out->stream); 1760 delete out; 1761 } 1762 return NO_ERROR; 1763} 1764 1765status_t AudioFlinger::suspendOutput(audio_io_handle_t output) 1766{ 1767 Mutex::Autolock _l(mLock); 1768 PlaybackThread *thread = checkPlaybackThread_l(output); 1769 1770 if (thread == NULL) { 1771 return BAD_VALUE; 1772 } 1773 1774 ALOGV("suspendOutput() %d", output); 1775 thread->suspend(); 1776 1777 return NO_ERROR; 1778} 1779 1780status_t AudioFlinger::restoreOutput(audio_io_handle_t output) 1781{ 1782 Mutex::Autolock _l(mLock); 1783 PlaybackThread *thread = checkPlaybackThread_l(output); 1784 1785 if (thread == NULL) { 1786 return BAD_VALUE; 1787 } 1788 1789 ALOGV("restoreOutput() %d", output); 1790 1791 thread->restore(); 1792 1793 return NO_ERROR; 1794} 1795 1796audio_io_handle_t AudioFlinger::openInput(audio_module_handle_t module, 1797 audio_devices_t *pDevices, 1798 uint32_t *pSamplingRate, 1799 audio_format_t *pFormat, 1800 audio_channel_mask_t *pChannelMask) 1801{ 1802 struct audio_config config; 1803 memset(&config, 0, sizeof(config)); 1804 config.sample_rate = (pSamplingRate != NULL) ? *pSamplingRate : 0; 1805 config.channel_mask = (pChannelMask != NULL) ? *pChannelMask : 0; 1806 config.format = (pFormat != NULL) ? *pFormat : AUDIO_FORMAT_DEFAULT; 1807 1808 uint32_t reqSamplingRate = config.sample_rate; 1809 audio_format_t reqFormat = config.format; 1810 audio_channel_mask_t reqChannelMask = config.channel_mask; 1811 1812 if (pDevices == NULL || *pDevices == AUDIO_DEVICE_NONE) { 1813 return 0; 1814 } 1815 1816 Mutex::Autolock _l(mLock); 1817 1818 AudioHwDevice *inHwDev = findSuitableHwDev_l(module, *pDevices); 1819 if (inHwDev == NULL) { 1820 return 0; 1821 } 1822 1823 audio_hw_device_t *inHwHal = inHwDev->hwDevice(); 1824 audio_io_handle_t id = nextUniqueId(); 1825 1826 audio_stream_in_t *inStream = NULL; 1827 status_t status = inHwHal->open_input_stream(inHwHal, id, *pDevices, &config, 1828 &inStream); 1829 ALOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %#x, Channels %x, " 1830 "status %d", 1831 inStream, 1832 config.sample_rate, 1833 config.format, 1834 config.channel_mask, 1835 status); 1836 1837 // If the input could not be opened with the requested parameters and we can handle the 1838 // conversion internally, try to open again with the proposed parameters. The AudioFlinger can 1839 // resample the input and do mono to stereo or stereo to mono conversions on 16 bit PCM inputs. 1840 if (status == BAD_VALUE && 1841 reqFormat == config.format && config.format == AUDIO_FORMAT_PCM_16_BIT && 1842 (config.sample_rate <= 2 * reqSamplingRate) && 1843 (audio_channel_count_from_in_mask(config.channel_mask) <= FCC_2) && 1844 (audio_channel_count_from_in_mask(reqChannelMask) <= FCC_2)) { 1845 // FIXME describe the change proposed by HAL (save old values so we can log them here) 1846 ALOGV("openInput() reopening with proposed sampling rate and channel mask"); 1847 inStream = NULL; 1848 status = inHwHal->open_input_stream(inHwHal, id, *pDevices, &config, &inStream); 1849 // FIXME log this new status; HAL should not propose any further changes 1850 } 1851 1852 if (status == NO_ERROR && inStream != NULL) { 1853 1854#ifdef TEE_SINK 1855 // Try to re-use most recently used Pipe to archive a copy of input for dumpsys, 1856 // or (re-)create if current Pipe is idle and does not match the new format 1857 sp<NBAIO_Sink> teeSink; 1858 enum { 1859 TEE_SINK_NO, // don't copy input 1860 TEE_SINK_NEW, // copy input using a new pipe 1861 TEE_SINK_OLD, // copy input using an existing pipe 1862 } kind; 1863 NBAIO_Format format = Format_from_SR_C(inStream->common.get_sample_rate(&inStream->common), 1864 audio_channel_count_from_in_mask( 1865 inStream->common.get_channels(&inStream->common))); 1866 if (!mTeeSinkInputEnabled) { 1867 kind = TEE_SINK_NO; 1868 } else if (!Format_isValid(format)) { 1869 kind = TEE_SINK_NO; 1870 } else if (mRecordTeeSink == 0) { 1871 kind = TEE_SINK_NEW; 1872 } else if (mRecordTeeSink->getStrongCount() != 1) { 1873 kind = TEE_SINK_NO; 1874 } else if (Format_isEqual(format, mRecordTeeSink->format())) { 1875 kind = TEE_SINK_OLD; 1876 } else { 1877 kind = TEE_SINK_NEW; 1878 } 1879 switch (kind) { 1880 case TEE_SINK_NEW: { 1881 Pipe *pipe = new Pipe(mTeeSinkInputFrames, format); 1882 size_t numCounterOffers = 0; 1883 const NBAIO_Format offers[1] = {format}; 1884 ssize_t index = pipe->negotiate(offers, 1, NULL, numCounterOffers); 1885 ALOG_ASSERT(index == 0); 1886 PipeReader *pipeReader = new PipeReader(*pipe); 1887 numCounterOffers = 0; 1888 index = pipeReader->negotiate(offers, 1, NULL, numCounterOffers); 1889 ALOG_ASSERT(index == 0); 1890 mRecordTeeSink = pipe; 1891 mRecordTeeSource = pipeReader; 1892 teeSink = pipe; 1893 } 1894 break; 1895 case TEE_SINK_OLD: 1896 teeSink = mRecordTeeSink; 1897 break; 1898 case TEE_SINK_NO: 1899 default: 1900 break; 1901 } 1902#endif 1903 1904 AudioStreamIn *input = new AudioStreamIn(inHwDev, inStream); 1905 1906 // Start record thread 1907 // RecordThread requires both input and output device indication to forward to audio 1908 // pre processing modules 1909 RecordThread *thread = new RecordThread(this, 1910 input, 1911 id, 1912 primaryOutputDevice_l(), 1913 *pDevices 1914#ifdef TEE_SINK 1915 , teeSink 1916#endif 1917 ); 1918 mRecordThreads.add(id, thread); 1919 ALOGV("openInput() created record thread: ID %d thread %p", id, thread); 1920 if (pSamplingRate != NULL) { 1921 *pSamplingRate = reqSamplingRate; 1922 } 1923 if (pFormat != NULL) { 1924 *pFormat = config.format; 1925 } 1926 if (pChannelMask != NULL) { 1927 *pChannelMask = reqChannelMask; 1928 } 1929 1930 // notify client processes of the new input creation 1931 thread->audioConfigChanged(AudioSystem::INPUT_OPENED); 1932 return id; 1933 } 1934 1935 return 0; 1936} 1937 1938status_t AudioFlinger::closeInput(audio_io_handle_t input) 1939{ 1940 return closeInput_nonvirtual(input); 1941} 1942 1943status_t AudioFlinger::closeInput_nonvirtual(audio_io_handle_t input) 1944{ 1945 // keep strong reference on the record thread so that 1946 // it is not destroyed while exit() is executed 1947 sp<RecordThread> thread; 1948 { 1949 Mutex::Autolock _l(mLock); 1950 thread = checkRecordThread_l(input); 1951 if (thread == 0) { 1952 return BAD_VALUE; 1953 } 1954 1955 ALOGV("closeInput() %d", input); 1956 audioConfigChanged(AudioSystem::INPUT_CLOSED, input, NULL); 1957 mRecordThreads.removeItem(input); 1958 } 1959 thread->exit(); 1960 // The thread entity (active unit of execution) is no longer running here, 1961 // but the ThreadBase container still exists. 1962 1963 AudioStreamIn *in = thread->clearInput(); 1964 ALOG_ASSERT(in != NULL, "in shouldn't be NULL"); 1965 // from now on thread->mInput is NULL 1966 in->hwDev()->close_input_stream(in->hwDev(), in->stream); 1967 delete in; 1968 1969 return NO_ERROR; 1970} 1971 1972status_t AudioFlinger::invalidateStream(audio_stream_type_t stream) 1973{ 1974 Mutex::Autolock _l(mLock); 1975 ALOGV("invalidateStream() stream %d", stream); 1976 1977 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1978 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 1979 thread->invalidateTracks(stream); 1980 } 1981 1982 return NO_ERROR; 1983} 1984 1985 1986int AudioFlinger::newAudioSessionId() 1987{ 1988 return nextUniqueId(); 1989} 1990 1991void AudioFlinger::acquireAudioSessionId(int audioSession, pid_t pid) 1992{ 1993 Mutex::Autolock _l(mLock); 1994 pid_t caller = IPCThreadState::self()->getCallingPid(); 1995 ALOGV("acquiring %d from %d, for %d", audioSession, caller, pid); 1996 if (pid != -1 && (caller == getpid_cached)) { 1997 caller = pid; 1998 } 1999 2000 { 2001 Mutex::Autolock _cl(mClientLock); 2002 // Ignore requests received from processes not known as notification client. The request 2003 // is likely proxied by mediaserver (e.g CameraService) and releaseAudioSessionId() can be 2004 // called from a different pid leaving a stale session reference. Also we don't know how 2005 // to clear this reference if the client process dies. 2006 if (mNotificationClients.indexOfKey(caller) < 0) { 2007 ALOGW("acquireAudioSessionId() unknown client %d for session %d", caller, audioSession); 2008 return; 2009 } 2010 } 2011 2012 size_t num = mAudioSessionRefs.size(); 2013 for (size_t i = 0; i< num; i++) { 2014 AudioSessionRef *ref = mAudioSessionRefs.editItemAt(i); 2015 if (ref->mSessionid == audioSession && ref->mPid == caller) { 2016 ref->mCnt++; 2017 ALOGV(" incremented refcount to %d", ref->mCnt); 2018 return; 2019 } 2020 } 2021 mAudioSessionRefs.push(new AudioSessionRef(audioSession, caller)); 2022 ALOGV(" added new entry for %d", audioSession); 2023} 2024 2025void AudioFlinger::releaseAudioSessionId(int audioSession, pid_t pid) 2026{ 2027 Mutex::Autolock _l(mLock); 2028 pid_t caller = IPCThreadState::self()->getCallingPid(); 2029 ALOGV("releasing %d from %d for %d", audioSession, caller, pid); 2030 if (pid != -1 && (caller == getpid_cached)) { 2031 caller = pid; 2032 } 2033 size_t num = mAudioSessionRefs.size(); 2034 for (size_t i = 0; i< num; i++) { 2035 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 2036 if (ref->mSessionid == audioSession && ref->mPid == caller) { 2037 ref->mCnt--; 2038 ALOGV(" decremented refcount to %d", ref->mCnt); 2039 if (ref->mCnt == 0) { 2040 mAudioSessionRefs.removeAt(i); 2041 delete ref; 2042 purgeStaleEffects_l(); 2043 } 2044 return; 2045 } 2046 } 2047 // If the caller is mediaserver it is likely that the session being released was acquired 2048 // on behalf of a process not in notification clients and we ignore the warning. 2049 ALOGW_IF(caller != getpid_cached, "session id %d not found for pid %d", audioSession, caller); 2050} 2051 2052void AudioFlinger::purgeStaleEffects_l() { 2053 2054 ALOGV("purging stale effects"); 2055 2056 Vector< sp<EffectChain> > chains; 2057 2058 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2059 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 2060 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 2061 sp<EffectChain> ec = t->mEffectChains[j]; 2062 if (ec->sessionId() > AUDIO_SESSION_OUTPUT_MIX) { 2063 chains.push(ec); 2064 } 2065 } 2066 } 2067 for (size_t i = 0; i < mRecordThreads.size(); i++) { 2068 sp<RecordThread> t = mRecordThreads.valueAt(i); 2069 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 2070 sp<EffectChain> ec = t->mEffectChains[j]; 2071 chains.push(ec); 2072 } 2073 } 2074 2075 for (size_t i = 0; i < chains.size(); i++) { 2076 sp<EffectChain> ec = chains[i]; 2077 int sessionid = ec->sessionId(); 2078 sp<ThreadBase> t = ec->mThread.promote(); 2079 if (t == 0) { 2080 continue; 2081 } 2082 size_t numsessionrefs = mAudioSessionRefs.size(); 2083 bool found = false; 2084 for (size_t k = 0; k < numsessionrefs; k++) { 2085 AudioSessionRef *ref = mAudioSessionRefs.itemAt(k); 2086 if (ref->mSessionid == sessionid) { 2087 ALOGV(" session %d still exists for %d with %d refs", 2088 sessionid, ref->mPid, ref->mCnt); 2089 found = true; 2090 break; 2091 } 2092 } 2093 if (!found) { 2094 Mutex::Autolock _l(t->mLock); 2095 // remove all effects from the chain 2096 while (ec->mEffects.size()) { 2097 sp<EffectModule> effect = ec->mEffects[0]; 2098 effect->unPin(); 2099 t->removeEffect_l(effect); 2100 if (effect->purgeHandles()) { 2101 t->checkSuspendOnEffectEnabled_l(effect, false, effect->sessionId()); 2102 } 2103 AudioSystem::unregisterEffect(effect->id()); 2104 } 2105 } 2106 } 2107 return; 2108} 2109 2110// checkPlaybackThread_l() must be called with AudioFlinger::mLock held 2111AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(audio_io_handle_t output) const 2112{ 2113 return mPlaybackThreads.valueFor(output).get(); 2114} 2115 2116// checkMixerThread_l() must be called with AudioFlinger::mLock held 2117AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(audio_io_handle_t output) const 2118{ 2119 PlaybackThread *thread = checkPlaybackThread_l(output); 2120 return thread != NULL && thread->type() != ThreadBase::DIRECT ? (MixerThread *) thread : NULL; 2121} 2122 2123// checkRecordThread_l() must be called with AudioFlinger::mLock held 2124AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(audio_io_handle_t input) const 2125{ 2126 return mRecordThreads.valueFor(input).get(); 2127} 2128 2129uint32_t AudioFlinger::nextUniqueId() 2130{ 2131 return (uint32_t) android_atomic_inc(&mNextUniqueId); 2132} 2133 2134AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l() const 2135{ 2136 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2137 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 2138 AudioStreamOut *output = thread->getOutput(); 2139 if (output != NULL && output->audioHwDev == mPrimaryHardwareDev) { 2140 return thread; 2141 } 2142 } 2143 return NULL; 2144} 2145 2146audio_devices_t AudioFlinger::primaryOutputDevice_l() const 2147{ 2148 PlaybackThread *thread = primaryPlaybackThread_l(); 2149 2150 if (thread == NULL) { 2151 return 0; 2152 } 2153 2154 return thread->outDevice(); 2155} 2156 2157sp<AudioFlinger::SyncEvent> AudioFlinger::createSyncEvent(AudioSystem::sync_event_t type, 2158 int triggerSession, 2159 int listenerSession, 2160 sync_event_callback_t callBack, 2161 wp<RefBase> cookie) 2162{ 2163 Mutex::Autolock _l(mLock); 2164 2165 sp<SyncEvent> event = new SyncEvent(type, triggerSession, listenerSession, callBack, cookie); 2166 status_t playStatus = NAME_NOT_FOUND; 2167 status_t recStatus = NAME_NOT_FOUND; 2168 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2169 playStatus = mPlaybackThreads.valueAt(i)->setSyncEvent(event); 2170 if (playStatus == NO_ERROR) { 2171 return event; 2172 } 2173 } 2174 for (size_t i = 0; i < mRecordThreads.size(); i++) { 2175 recStatus = mRecordThreads.valueAt(i)->setSyncEvent(event); 2176 if (recStatus == NO_ERROR) { 2177 return event; 2178 } 2179 } 2180 if (playStatus == NAME_NOT_FOUND || recStatus == NAME_NOT_FOUND) { 2181 mPendingSyncEvents.add(event); 2182 } else { 2183 ALOGV("createSyncEvent() invalid event %d", event->type()); 2184 event.clear(); 2185 } 2186 return event; 2187} 2188 2189// ---------------------------------------------------------------------------- 2190// Effect management 2191// ---------------------------------------------------------------------------- 2192 2193 2194status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects) const 2195{ 2196 Mutex::Autolock _l(mLock); 2197 return EffectQueryNumberEffects(numEffects); 2198} 2199 2200status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor) const 2201{ 2202 Mutex::Autolock _l(mLock); 2203 return EffectQueryEffect(index, descriptor); 2204} 2205 2206status_t AudioFlinger::getEffectDescriptor(const effect_uuid_t *pUuid, 2207 effect_descriptor_t *descriptor) const 2208{ 2209 Mutex::Autolock _l(mLock); 2210 return EffectGetDescriptor(pUuid, descriptor); 2211} 2212 2213 2214sp<IEffect> AudioFlinger::createEffect( 2215 effect_descriptor_t *pDesc, 2216 const sp<IEffectClient>& effectClient, 2217 int32_t priority, 2218 audio_io_handle_t io, 2219 int sessionId, 2220 status_t *status, 2221 int *id, 2222 int *enabled) 2223{ 2224 status_t lStatus = NO_ERROR; 2225 sp<EffectHandle> handle; 2226 effect_descriptor_t desc; 2227 2228 pid_t pid = IPCThreadState::self()->getCallingPid(); 2229 ALOGV("createEffect pid %d, effectClient %p, priority %d, sessionId %d, io %d", 2230 pid, effectClient.get(), priority, sessionId, io); 2231 2232 if (pDesc == NULL) { 2233 lStatus = BAD_VALUE; 2234 goto Exit; 2235 } 2236 2237 // check audio settings permission for global effects 2238 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && !settingsAllowed()) { 2239 lStatus = PERMISSION_DENIED; 2240 goto Exit; 2241 } 2242 2243 // Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects 2244 // that can only be created by audio policy manager (running in same process) 2245 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid_cached != pid) { 2246 lStatus = PERMISSION_DENIED; 2247 goto Exit; 2248 } 2249 2250 { 2251 if (!EffectIsNullUuid(&pDesc->uuid)) { 2252 // if uuid is specified, request effect descriptor 2253 lStatus = EffectGetDescriptor(&pDesc->uuid, &desc); 2254 if (lStatus < 0) { 2255 ALOGW("createEffect() error %d from EffectGetDescriptor", lStatus); 2256 goto Exit; 2257 } 2258 } else { 2259 // if uuid is not specified, look for an available implementation 2260 // of the required type in effect factory 2261 if (EffectIsNullUuid(&pDesc->type)) { 2262 ALOGW("createEffect() no effect type"); 2263 lStatus = BAD_VALUE; 2264 goto Exit; 2265 } 2266 uint32_t numEffects = 0; 2267 effect_descriptor_t d; 2268 d.flags = 0; // prevent compiler warning 2269 bool found = false; 2270 2271 lStatus = EffectQueryNumberEffects(&numEffects); 2272 if (lStatus < 0) { 2273 ALOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus); 2274 goto Exit; 2275 } 2276 for (uint32_t i = 0; i < numEffects; i++) { 2277 lStatus = EffectQueryEffect(i, &desc); 2278 if (lStatus < 0) { 2279 ALOGW("createEffect() error %d from EffectQueryEffect", lStatus); 2280 continue; 2281 } 2282 if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) { 2283 // If matching type found save effect descriptor. If the session is 2284 // 0 and the effect is not auxiliary, continue enumeration in case 2285 // an auxiliary version of this effect type is available 2286 found = true; 2287 d = desc; 2288 if (sessionId != AUDIO_SESSION_OUTPUT_MIX || 2289 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 2290 break; 2291 } 2292 } 2293 } 2294 if (!found) { 2295 lStatus = BAD_VALUE; 2296 ALOGW("createEffect() effect not found"); 2297 goto Exit; 2298 } 2299 // For same effect type, chose auxiliary version over insert version if 2300 // connect to output mix (Compliance to OpenSL ES) 2301 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && 2302 (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) { 2303 desc = d; 2304 } 2305 } 2306 2307 // Do not allow auxiliary effects on a session different from 0 (output mix) 2308 if (sessionId != AUDIO_SESSION_OUTPUT_MIX && 2309 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 2310 lStatus = INVALID_OPERATION; 2311 goto Exit; 2312 } 2313 2314 // check recording permission for visualizer 2315 if ((memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) && 2316 !recordingAllowed()) { 2317 lStatus = PERMISSION_DENIED; 2318 goto Exit; 2319 } 2320 2321 // return effect descriptor 2322 *pDesc = desc; 2323 if (io == AUDIO_IO_HANDLE_NONE && sessionId == AUDIO_SESSION_OUTPUT_MIX) { 2324 // if the output returned by getOutputForEffect() is removed before we lock the 2325 // mutex below, the call to checkPlaybackThread_l(io) below will detect it 2326 // and we will exit safely 2327 io = AudioSystem::getOutputForEffect(&desc); 2328 ALOGV("createEffect got output %d", io); 2329 } 2330 2331 Mutex::Autolock _l(mLock); 2332 2333 // If output is not specified try to find a matching audio session ID in one of the 2334 // output threads. 2335 // If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX 2336 // because of code checking output when entering the function. 2337 // Note: io is never 0 when creating an effect on an input 2338 if (io == AUDIO_IO_HANDLE_NONE) { 2339 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) { 2340 // output must be specified by AudioPolicyManager when using session 2341 // AUDIO_SESSION_OUTPUT_STAGE 2342 lStatus = BAD_VALUE; 2343 goto Exit; 2344 } 2345 // look for the thread where the specified audio session is present 2346 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2347 if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 2348 io = mPlaybackThreads.keyAt(i); 2349 break; 2350 } 2351 } 2352 if (io == 0) { 2353 for (size_t i = 0; i < mRecordThreads.size(); i++) { 2354 if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 2355 io = mRecordThreads.keyAt(i); 2356 break; 2357 } 2358 } 2359 } 2360 // If no output thread contains the requested session ID, default to 2361 // first output. The effect chain will be moved to the correct output 2362 // thread when a track with the same session ID is created 2363 if (io == AUDIO_IO_HANDLE_NONE && mPlaybackThreads.size() > 0) { 2364 io = mPlaybackThreads.keyAt(0); 2365 } 2366 ALOGV("createEffect() got io %d for effect %s", io, desc.name); 2367 } 2368 ThreadBase *thread = checkRecordThread_l(io); 2369 if (thread == NULL) { 2370 thread = checkPlaybackThread_l(io); 2371 if (thread == NULL) { 2372 ALOGE("createEffect() unknown output thread"); 2373 lStatus = BAD_VALUE; 2374 goto Exit; 2375 } 2376 } 2377 2378 sp<Client> client = registerPid(pid); 2379 2380 // create effect on selected output thread 2381 handle = thread->createEffect_l(client, effectClient, priority, sessionId, 2382 &desc, enabled, &lStatus); 2383 if (handle != 0 && id != NULL) { 2384 *id = handle->id(); 2385 } 2386 } 2387 2388Exit: 2389 *status = lStatus; 2390 return handle; 2391} 2392 2393status_t AudioFlinger::moveEffects(int sessionId, audio_io_handle_t srcOutput, 2394 audio_io_handle_t dstOutput) 2395{ 2396 ALOGV("moveEffects() session %d, srcOutput %d, dstOutput %d", 2397 sessionId, srcOutput, dstOutput); 2398 Mutex::Autolock _l(mLock); 2399 if (srcOutput == dstOutput) { 2400 ALOGW("moveEffects() same dst and src outputs %d", dstOutput); 2401 return NO_ERROR; 2402 } 2403 PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput); 2404 if (srcThread == NULL) { 2405 ALOGW("moveEffects() bad srcOutput %d", srcOutput); 2406 return BAD_VALUE; 2407 } 2408 PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput); 2409 if (dstThread == NULL) { 2410 ALOGW("moveEffects() bad dstOutput %d", dstOutput); 2411 return BAD_VALUE; 2412 } 2413 2414 Mutex::Autolock _dl(dstThread->mLock); 2415 Mutex::Autolock _sl(srcThread->mLock); 2416 return moveEffectChain_l(sessionId, srcThread, dstThread, false); 2417} 2418 2419// moveEffectChain_l must be called with both srcThread and dstThread mLocks held 2420status_t AudioFlinger::moveEffectChain_l(int sessionId, 2421 AudioFlinger::PlaybackThread *srcThread, 2422 AudioFlinger::PlaybackThread *dstThread, 2423 bool reRegister) 2424{ 2425 ALOGV("moveEffectChain_l() session %d from thread %p to thread %p", 2426 sessionId, srcThread, dstThread); 2427 2428 sp<EffectChain> chain = srcThread->getEffectChain_l(sessionId); 2429 if (chain == 0) { 2430 ALOGW("moveEffectChain_l() effect chain for session %d not on source thread %p", 2431 sessionId, srcThread); 2432 return INVALID_OPERATION; 2433 } 2434 2435 // remove chain first. This is useful only if reconfiguring effect chain on same output thread, 2436 // so that a new chain is created with correct parameters when first effect is added. This is 2437 // otherwise unnecessary as removeEffect_l() will remove the chain when last effect is 2438 // removed. 2439 srcThread->removeEffectChain_l(chain); 2440 2441 // transfer all effects one by one so that new effect chain is created on new thread with 2442 // correct buffer sizes and audio parameters and effect engines reconfigured accordingly 2443 sp<EffectChain> dstChain; 2444 uint32_t strategy = 0; // prevent compiler warning 2445 sp<EffectModule> effect = chain->getEffectFromId_l(0); 2446 Vector< sp<EffectModule> > removed; 2447 status_t status = NO_ERROR; 2448 while (effect != 0) { 2449 srcThread->removeEffect_l(effect); 2450 removed.add(effect); 2451 status = dstThread->addEffect_l(effect); 2452 if (status != NO_ERROR) { 2453 break; 2454 } 2455 // removeEffect_l() has stopped the effect if it was active so it must be restarted 2456 if (effect->state() == EffectModule::ACTIVE || 2457 effect->state() == EffectModule::STOPPING) { 2458 effect->start(); 2459 } 2460 // if the move request is not received from audio policy manager, the effect must be 2461 // re-registered with the new strategy and output 2462 if (dstChain == 0) { 2463 dstChain = effect->chain().promote(); 2464 if (dstChain == 0) { 2465 ALOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get()); 2466 status = NO_INIT; 2467 break; 2468 } 2469 strategy = dstChain->strategy(); 2470 } 2471 if (reRegister) { 2472 AudioSystem::unregisterEffect(effect->id()); 2473 AudioSystem::registerEffect(&effect->desc(), 2474 dstThread->id(), 2475 strategy, 2476 sessionId, 2477 effect->id()); 2478 AudioSystem::setEffectEnabled(effect->id(), effect->isEnabled()); 2479 } 2480 effect = chain->getEffectFromId_l(0); 2481 } 2482 2483 if (status != NO_ERROR) { 2484 for (size_t i = 0; i < removed.size(); i++) { 2485 srcThread->addEffect_l(removed[i]); 2486 if (dstChain != 0 && reRegister) { 2487 AudioSystem::unregisterEffect(removed[i]->id()); 2488 AudioSystem::registerEffect(&removed[i]->desc(), 2489 srcThread->id(), 2490 strategy, 2491 sessionId, 2492 removed[i]->id()); 2493 AudioSystem::setEffectEnabled(effect->id(), effect->isEnabled()); 2494 } 2495 } 2496 } 2497 2498 return status; 2499} 2500 2501bool AudioFlinger::isNonOffloadableGlobalEffectEnabled_l() 2502{ 2503 if (mGlobalEffectEnableTime != 0 && 2504 ((systemTime() - mGlobalEffectEnableTime) < kMinGlobalEffectEnabletimeNs)) { 2505 return true; 2506 } 2507 2508 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2509 sp<EffectChain> ec = 2510 mPlaybackThreads.valueAt(i)->getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2511 if (ec != 0 && ec->isNonOffloadableEnabled()) { 2512 return true; 2513 } 2514 } 2515 return false; 2516} 2517 2518void AudioFlinger::onNonOffloadableGlobalEffectEnable() 2519{ 2520 Mutex::Autolock _l(mLock); 2521 2522 mGlobalEffectEnableTime = systemTime(); 2523 2524 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2525 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 2526 if (t->mType == ThreadBase::OFFLOAD) { 2527 t->invalidateTracks(AUDIO_STREAM_MUSIC); 2528 } 2529 } 2530 2531} 2532 2533struct Entry { 2534#define MAX_NAME 32 // %Y%m%d%H%M%S_%d.wav 2535 char mName[MAX_NAME]; 2536}; 2537 2538int comparEntry(const void *p1, const void *p2) 2539{ 2540 return strcmp(((const Entry *) p1)->mName, ((const Entry *) p2)->mName); 2541} 2542 2543#ifdef TEE_SINK 2544void AudioFlinger::dumpTee(int fd, const sp<NBAIO_Source>& source, audio_io_handle_t id) 2545{ 2546 NBAIO_Source *teeSource = source.get(); 2547 if (teeSource != NULL) { 2548 // .wav rotation 2549 // There is a benign race condition if 2 threads call this simultaneously. 2550 // They would both traverse the directory, but the result would simply be 2551 // failures at unlink() which are ignored. It's also unlikely since 2552 // normally dumpsys is only done by bugreport or from the command line. 2553 char teePath[32+256]; 2554 strcpy(teePath, "/data/misc/media"); 2555 size_t teePathLen = strlen(teePath); 2556 DIR *dir = opendir(teePath); 2557 teePath[teePathLen++] = '/'; 2558 if (dir != NULL) { 2559#define MAX_SORT 20 // number of entries to sort 2560#define MAX_KEEP 10 // number of entries to keep 2561 struct Entry entries[MAX_SORT]; 2562 size_t entryCount = 0; 2563 while (entryCount < MAX_SORT) { 2564 struct dirent de; 2565 struct dirent *result = NULL; 2566 int rc = readdir_r(dir, &de, &result); 2567 if (rc != 0) { 2568 ALOGW("readdir_r failed %d", rc); 2569 break; 2570 } 2571 if (result == NULL) { 2572 break; 2573 } 2574 if (result != &de) { 2575 ALOGW("readdir_r returned unexpected result %p != %p", result, &de); 2576 break; 2577 } 2578 // ignore non .wav file entries 2579 size_t nameLen = strlen(de.d_name); 2580 if (nameLen <= 4 || nameLen >= MAX_NAME || 2581 strcmp(&de.d_name[nameLen - 4], ".wav")) { 2582 continue; 2583 } 2584 strcpy(entries[entryCount++].mName, de.d_name); 2585 } 2586 (void) closedir(dir); 2587 if (entryCount > MAX_KEEP) { 2588 qsort(entries, entryCount, sizeof(Entry), comparEntry); 2589 for (size_t i = 0; i < entryCount - MAX_KEEP; ++i) { 2590 strcpy(&teePath[teePathLen], entries[i].mName); 2591 (void) unlink(teePath); 2592 } 2593 } 2594 } else { 2595 if (fd >= 0) { 2596 dprintf(fd, "unable to rotate tees in %s: %s\n", teePath, strerror(errno)); 2597 } 2598 } 2599 char teeTime[16]; 2600 struct timeval tv; 2601 gettimeofday(&tv, NULL); 2602 struct tm tm; 2603 localtime_r(&tv.tv_sec, &tm); 2604 strftime(teeTime, sizeof(teeTime), "%Y%m%d%H%M%S", &tm); 2605 snprintf(&teePath[teePathLen], sizeof(teePath) - teePathLen, "%s_%d.wav", teeTime, id); 2606 // if 2 dumpsys are done within 1 second, and rotation didn't work, then discard 2nd 2607 int teeFd = open(teePath, O_WRONLY | O_CREAT | O_EXCL | O_NOFOLLOW, S_IRUSR | S_IWUSR); 2608 if (teeFd >= 0) { 2609 char wavHeader[44]; 2610 memcpy(wavHeader, 2611 "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", 2612 sizeof(wavHeader)); 2613 NBAIO_Format format = teeSource->format(); 2614 unsigned channelCount = Format_channelCount(format); 2615 ALOG_ASSERT(channelCount <= FCC_2); 2616 uint32_t sampleRate = Format_sampleRate(format); 2617 wavHeader[22] = channelCount; // number of channels 2618 wavHeader[24] = sampleRate; // sample rate 2619 wavHeader[25] = sampleRate >> 8; 2620 wavHeader[32] = channelCount * 2; // block alignment 2621 write(teeFd, wavHeader, sizeof(wavHeader)); 2622 size_t total = 0; 2623 bool firstRead = true; 2624 for (;;) { 2625#define TEE_SINK_READ 1024 2626 short buffer[TEE_SINK_READ * FCC_2]; 2627 size_t count = TEE_SINK_READ; 2628 ssize_t actual = teeSource->read(buffer, count, 2629 AudioBufferProvider::kInvalidPTS); 2630 bool wasFirstRead = firstRead; 2631 firstRead = false; 2632 if (actual <= 0) { 2633 if (actual == (ssize_t) OVERRUN && wasFirstRead) { 2634 continue; 2635 } 2636 break; 2637 } 2638 ALOG_ASSERT(actual <= (ssize_t)count); 2639 write(teeFd, buffer, actual * channelCount * sizeof(short)); 2640 total += actual; 2641 } 2642 lseek(teeFd, (off_t) 4, SEEK_SET); 2643 uint32_t temp = 44 + total * channelCount * sizeof(short) - 8; 2644 write(teeFd, &temp, sizeof(temp)); 2645 lseek(teeFd, (off_t) 40, SEEK_SET); 2646 temp = total * channelCount * sizeof(short); 2647 write(teeFd, &temp, sizeof(temp)); 2648 close(teeFd); 2649 if (fd >= 0) { 2650 dprintf(fd, "tee copied to %s\n", teePath); 2651 } 2652 } else { 2653 if (fd >= 0) { 2654 dprintf(fd, "unable to create tee %s: %s\n", teePath, strerror(errno)); 2655 } 2656 } 2657 } 2658} 2659#endif 2660 2661// ---------------------------------------------------------------------------- 2662 2663status_t AudioFlinger::onTransact( 2664 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 2665{ 2666 return BnAudioFlinger::onTransact(code, data, reply, flags); 2667} 2668 2669}; // namespace android 2670