AudioFlinger.cpp revision 5798d4ebf236357a4b13246f40e52b90a34d09a4
13bbc90992645934df523762f7dcdb097eae366d5Joseph Wen/* 23bbc90992645934df523762f7dcdb097eae366d5Joseph Wen** 33bbc90992645934df523762f7dcdb097eae366d5Joseph Wen** Copyright 2007, The Android Open Source Project 43bbc90992645934df523762f7dcdb097eae366d5Joseph Wen** 53bbc90992645934df523762f7dcdb097eae366d5Joseph Wen** Licensed under the Apache License, Version 2.0 (the "License"); 63bbc90992645934df523762f7dcdb097eae366d5Joseph Wen** you may not use this file except in compliance with the License. 73bbc90992645934df523762f7dcdb097eae366d5Joseph Wen** You may obtain a copy of the License at 83bbc90992645934df523762f7dcdb097eae366d5Joseph Wen** 93bbc90992645934df523762f7dcdb097eae366d5Joseph Wen** http://www.apache.org/licenses/LICENSE-2.0 103bbc90992645934df523762f7dcdb097eae366d5Joseph Wen** 113bbc90992645934df523762f7dcdb097eae366d5Joseph Wen** Unless required by applicable law or agreed to in writing, software 123bbc90992645934df523762f7dcdb097eae366d5Joseph Wen** distributed under the License is distributed on an "AS IS" BASIS, 133bbc90992645934df523762f7dcdb097eae366d5Joseph Wen** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. 143bbc90992645934df523762f7dcdb097eae366d5Joseph Wen** See the License for the specific language governing permissions and 153bbc90992645934df523762f7dcdb097eae366d5Joseph Wen** limitations under the License. 163bbc90992645934df523762f7dcdb097eae366d5Joseph Wen*/ 173bbc90992645934df523762f7dcdb097eae366d5Joseph Wen 183bbc90992645934df523762f7dcdb097eae366d5Joseph Wen 193bbc90992645934df523762f7dcdb097eae366d5Joseph Wen#define LOG_TAG "AudioFlinger" 203bbc90992645934df523762f7dcdb097eae366d5Joseph Wen//#define LOG_NDEBUG 0 213bbc90992645934df523762f7dcdb097eae366d5Joseph Wen 223bbc90992645934df523762f7dcdb097eae366d5Joseph Wen#include <math.h> 233bbc90992645934df523762f7dcdb097eae366d5Joseph Wen#include <signal.h> 243bbc90992645934df523762f7dcdb097eae366d5Joseph Wen#include <sys/time.h> 253bbc90992645934df523762f7dcdb097eae366d5Joseph Wen#include <sys/resource.h> 263bbc90992645934df523762f7dcdb097eae366d5Joseph Wen 273bbc90992645934df523762f7dcdb097eae366d5Joseph Wen#include <binder/IPCThreadState.h> 283bbc90992645934df523762f7dcdb097eae366d5Joseph Wen#include <binder/IServiceManager.h> 293bbc90992645934df523762f7dcdb097eae366d5Joseph Wen#include <utils/Log.h> 303bbc90992645934df523762f7dcdb097eae366d5Joseph Wen#include <binder/Parcel.h> 313bbc90992645934df523762f7dcdb097eae366d5Joseph Wen#include <binder/IPCThreadState.h> 323bbc90992645934df523762f7dcdb097eae366d5Joseph Wen#include <utils/String16.h> 333bbc90992645934df523762f7dcdb097eae366d5Joseph Wen#include <utils/threads.h> 343bbc90992645934df523762f7dcdb097eae366d5Joseph Wen#include <utils/Atomic.h> 353bbc90992645934df523762f7dcdb097eae366d5Joseph Wen 363bbc90992645934df523762f7dcdb097eae366d5Joseph Wen#include <cutils/bitops.h> 373bbc90992645934df523762f7dcdb097eae366d5Joseph Wen#include <cutils/properties.h> 383bbc90992645934df523762f7dcdb097eae366d5Joseph Wen#include <cutils/compiler.h> 393bbc90992645934df523762f7dcdb097eae366d5Joseph Wen 403bbc90992645934df523762f7dcdb097eae366d5Joseph Wen#include <media/IMediaPlayerService.h> 413bbc90992645934df523762f7dcdb097eae366d5Joseph Wen#include <media/IMediaDeathNotifier.h> 423bbc90992645934df523762f7dcdb097eae366d5Joseph Wen 433bbc90992645934df523762f7dcdb097eae366d5Joseph Wen#include <private/media/AudioTrackShared.h> 443bbc90992645934df523762f7dcdb097eae366d5Joseph Wen#include <private/media/AudioEffectShared.h> 453bbc90992645934df523762f7dcdb097eae366d5Joseph Wen 463bbc90992645934df523762f7dcdb097eae366d5Joseph Wen#include <system/audio.h> 473bbc90992645934df523762f7dcdb097eae366d5Joseph Wen#include <hardware/audio.h> 483bbc90992645934df523762f7dcdb097eae366d5Joseph Wen 493bbc90992645934df523762f7dcdb097eae366d5Joseph Wen#include "AudioMixer.h" 503bbc90992645934df523762f7dcdb097eae366d5Joseph Wen#include "AudioFlinger.h" 513bbc90992645934df523762f7dcdb097eae366d5Joseph Wen#include "ServiceUtilities.h" 523bbc90992645934df523762f7dcdb097eae366d5Joseph Wen 533bbc90992645934df523762f7dcdb097eae366d5Joseph Wen#include <media/EffectsFactoryApi.h> 543bbc90992645934df523762f7dcdb097eae366d5Joseph Wen#include <audio_effects/effect_visualizer.h> 553bbc90992645934df523762f7dcdb097eae366d5Joseph Wen#include <audio_effects/effect_ns.h> 563bbc90992645934df523762f7dcdb097eae366d5Joseph Wen#include <audio_effects/effect_aec.h> 573bbc90992645934df523762f7dcdb097eae366d5Joseph Wen 583bbc90992645934df523762f7dcdb097eae366d5Joseph Wen#include <audio_utils/primitives.h> 593bbc90992645934df523762f7dcdb097eae366d5Joseph Wen 603bbc90992645934df523762f7dcdb097eae366d5Joseph Wen#include <cpustats/ThreadCpuUsage.h> 613bbc90992645934df523762f7dcdb097eae366d5Joseph Wen#include <powermanager/PowerManager.h> 623bbc90992645934df523762f7dcdb097eae366d5Joseph Wen// #define DEBUG_CPU_USAGE 10 // log statistics every n wall clock seconds 633bbc90992645934df523762f7dcdb097eae366d5Joseph Wen 643bbc90992645934df523762f7dcdb097eae366d5Joseph Wen#include <common_time/cc_helper.h> 653bbc90992645934df523762f7dcdb097eae366d5Joseph Wen#include <common_time/local_clock.h> 663bbc90992645934df523762f7dcdb097eae366d5Joseph Wen 673bbc90992645934df523762f7dcdb097eae366d5Joseph Wen// ---------------------------------------------------------------------------- 683bbc90992645934df523762f7dcdb097eae366d5Joseph Wen 693bbc90992645934df523762f7dcdb097eae366d5Joseph Wen 703bbc90992645934df523762f7dcdb097eae366d5Joseph Wennamespace android { 713bbc90992645934df523762f7dcdb097eae366d5Joseph Wen 723bbc90992645934df523762f7dcdb097eae366d5Joseph Wenstatic const char kDeadlockedString[] = "AudioFlinger may be deadlocked\n"; 733bbc90992645934df523762f7dcdb097eae366d5Joseph Wenstatic const char kHardwareLockedString[] = "Hardware lock is taken\n"; 743bbc90992645934df523762f7dcdb097eae366d5Joseph Wen 753bbc90992645934df523762f7dcdb097eae366d5Joseph Wenstatic const float MAX_GAIN = 4096.0f; 763bbc90992645934df523762f7dcdb097eae366d5Joseph Wenstatic const uint32_t MAX_GAIN_INT = 0x1000; 773bbc90992645934df523762f7dcdb097eae366d5Joseph Wen 783bbc90992645934df523762f7dcdb097eae366d5Joseph Wen// retry counts for buffer fill timeout 793bbc90992645934df523762f7dcdb097eae366d5Joseph Wen// 50 * ~20msecs = 1 second 803bbc90992645934df523762f7dcdb097eae366d5Joseph Wenstatic const int8_t kMaxTrackRetries = 50; 813bbc90992645934df523762f7dcdb097eae366d5Joseph Wenstatic const int8_t kMaxTrackStartupRetries = 50; 823bbc90992645934df523762f7dcdb097eae366d5Joseph Wen// allow less retry attempts on direct output thread. 833bbc90992645934df523762f7dcdb097eae366d5Joseph Wen// direct outputs can be a scarce resource in audio hardware and should 843bbc90992645934df523762f7dcdb097eae366d5Joseph Wen// be released as quickly as possible. 853bbc90992645934df523762f7dcdb097eae366d5Joseph Wenstatic const int8_t kMaxTrackRetriesDirect = 2; 863bbc90992645934df523762f7dcdb097eae366d5Joseph Wen 873bbc90992645934df523762f7dcdb097eae366d5Joseph Wenstatic const int kDumpLockRetries = 50; 883bbc90992645934df523762f7dcdb097eae366d5Joseph Wenstatic const int kDumpLockSleepUs = 20000; 893bbc90992645934df523762f7dcdb097eae366d5Joseph Wen 903bbc90992645934df523762f7dcdb097eae366d5Joseph Wen// don't warn about blocked writes or record buffer overflows more often than this 913bbc90992645934df523762f7dcdb097eae366d5Joseph Wenstatic const nsecs_t kWarningThrottleNs = seconds(5); 923bbc90992645934df523762f7dcdb097eae366d5Joseph Wen 933bbc90992645934df523762f7dcdb097eae366d5Joseph Wen// RecordThread loop sleep time upon application overrun or audio HAL read error 943bbc90992645934df523762f7dcdb097eae366d5Joseph Wenstatic const int kRecordThreadSleepUs = 5000; 95 96// maximum time to wait for setParameters to complete 97static const nsecs_t kSetParametersTimeoutNs = seconds(2); 98 99// minimum sleep time for the mixer thread loop when tracks are active but in underrun 100static const uint32_t kMinThreadSleepTimeUs = 5000; 101// maximum divider applied to the active sleep time in the mixer thread loop 102static const uint32_t kMaxThreadSleepTimeShift = 2; 103 104nsecs_t AudioFlinger::mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs; 105 106// ---------------------------------------------------------------------------- 107 108// To collect the amplifier usage 109static void addBatteryData(uint32_t params) { 110 sp<IMediaPlayerService> service = IMediaDeathNotifier::getMediaPlayerService(); 111 if (service == NULL) { 112 // it already logged 113 return; 114 } 115 116 service->addBatteryData(params); 117} 118 119static int load_audio_interface(const char *if_name, const hw_module_t **mod, 120 audio_hw_device_t **dev) 121{ 122 int rc; 123 124 rc = hw_get_module_by_class(AUDIO_HARDWARE_MODULE_ID, if_name, mod); 125 if (rc) 126 goto out; 127 128 rc = audio_hw_device_open(*mod, dev); 129 ALOGE_IF(rc, "couldn't open audio hw device in %s.%s (%s)", 130 AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc)); 131 if (rc) 132 goto out; 133 134 return 0; 135 136out: 137 *mod = NULL; 138 *dev = NULL; 139 return rc; 140} 141 142static const char * const audio_interfaces[] = { 143 "primary", 144 "a2dp", 145 "usb", 146}; 147#define ARRAY_SIZE(x) (sizeof((x))/sizeof(((x)[0]))) 148 149// ---------------------------------------------------------------------------- 150 151AudioFlinger::AudioFlinger() 152 : BnAudioFlinger(), 153 mPrimaryHardwareDev(NULL), 154 mHardwareStatus(AUDIO_HW_IDLE), // see also onFirstRef() 155 mMasterVolume(1.0f), 156 mMasterVolumeSupportLvl(MVS_NONE), 157 mMasterMute(false), 158 mNextUniqueId(1), 159 mMode(AUDIO_MODE_INVALID), 160 mBtNrecIsOff(false) 161{ 162} 163 164void AudioFlinger::onFirstRef() 165{ 166 int rc = 0; 167 168 Mutex::Autolock _l(mLock); 169 170 /* TODO: move all this work into an Init() function */ 171 char val_str[PROPERTY_VALUE_MAX] = { 0 }; 172 if (property_get("ro.audio.flinger_standbytime_ms", val_str, NULL) >= 0) { 173 uint32_t int_val; 174 if (1 == sscanf(val_str, "%u", &int_val)) { 175 mStandbyTimeInNsecs = milliseconds(int_val); 176 ALOGI("Using %u mSec as standby time.", int_val); 177 } else { 178 mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs; 179 ALOGI("Using default %u mSec as standby time.", 180 (uint32_t)(mStandbyTimeInNsecs / 1000000)); 181 } 182 } 183 184 for (size_t i = 0; i < ARRAY_SIZE(audio_interfaces); i++) { 185 const hw_module_t *mod; 186 audio_hw_device_t *dev; 187 188 rc = load_audio_interface(audio_interfaces[i], &mod, &dev); 189 if (rc) 190 continue; 191 192 ALOGI("Loaded %s audio interface from %s (%s)", audio_interfaces[i], 193 mod->name, mod->id); 194 mAudioHwDevs.push(dev); 195 196 if (mPrimaryHardwareDev == NULL) { 197 mPrimaryHardwareDev = dev; 198 ALOGI("Using '%s' (%s.%s) as the primary audio interface", 199 mod->name, mod->id, audio_interfaces[i]); 200 } 201 } 202 203 if (mPrimaryHardwareDev == NULL) { 204 ALOGE("Primary audio interface not found"); 205 // proceed, all later accesses to mPrimaryHardwareDev verify it's safe with initCheck() 206 } 207 208 // Currently (mPrimaryHardwareDev == NULL) == (mAudioHwDevs.size() == 0), but the way the 209 // primary HW dev is selected can change so these conditions might not always be equivalent. 210 // When that happens, re-visit all the code that assumes this. 211 212 AutoMutex lock(mHardwareLock); 213 214 // Determine the level of master volume support the primary audio HAL has, 215 // and set the initial master volume at the same time. 216 float initialVolume = 1.0; 217 mMasterVolumeSupportLvl = MVS_NONE; 218 if (0 == mPrimaryHardwareDev->init_check(mPrimaryHardwareDev)) { 219 audio_hw_device_t *dev = mPrimaryHardwareDev; 220 221 mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME; 222 if ((NULL != dev->get_master_volume) && 223 (NO_ERROR == dev->get_master_volume(dev, &initialVolume))) { 224 mMasterVolumeSupportLvl = MVS_FULL; 225 } else { 226 mMasterVolumeSupportLvl = MVS_SETONLY; 227 initialVolume = 1.0; 228 } 229 230 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 231 if ((NULL == dev->set_master_volume) || 232 (NO_ERROR != dev->set_master_volume(dev, initialVolume))) { 233 mMasterVolumeSupportLvl = MVS_NONE; 234 } 235 mHardwareStatus = AUDIO_HW_IDLE; 236 } 237 238 // Set the mode for each audio HAL, and try to set the initial volume (if 239 // supported) for all of the non-primary audio HALs. 240 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 241 audio_hw_device_t *dev = mAudioHwDevs[i]; 242 243 mHardwareStatus = AUDIO_HW_INIT; 244 rc = dev->init_check(dev); 245 mHardwareStatus = AUDIO_HW_IDLE; 246 if (rc == 0) { 247 mMode = AUDIO_MODE_NORMAL; // assigned multiple times with same value 248 mHardwareStatus = AUDIO_HW_SET_MODE; 249 dev->set_mode(dev, mMode); 250 251 if ((dev != mPrimaryHardwareDev) && 252 (NULL != dev->set_master_volume)) { 253 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 254 dev->set_master_volume(dev, initialVolume); 255 } 256 257 mHardwareStatus = AUDIO_HW_IDLE; 258 } 259 } 260 261 mMasterVolumeSW = (MVS_NONE == mMasterVolumeSupportLvl) 262 ? initialVolume 263 : 1.0; 264 mMasterVolume = initialVolume; 265 mHardwareStatus = AUDIO_HW_IDLE; 266} 267 268AudioFlinger::~AudioFlinger() 269{ 270 271 while (!mRecordThreads.isEmpty()) { 272 // closeInput() will remove first entry from mRecordThreads 273 closeInput(mRecordThreads.keyAt(0)); 274 } 275 while (!mPlaybackThreads.isEmpty()) { 276 // closeOutput() will remove first entry from mPlaybackThreads 277 closeOutput(mPlaybackThreads.keyAt(0)); 278 } 279 280 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 281 // no mHardwareLock needed, as there are no other references to this 282 audio_hw_device_close(mAudioHwDevs[i]); 283 } 284} 285 286audio_hw_device_t* AudioFlinger::findSuitableHwDev_l(uint32_t devices) 287{ 288 /* first matching HW device is returned */ 289 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 290 audio_hw_device_t *dev = mAudioHwDevs[i]; 291 if ((dev->get_supported_devices(dev) & devices) == devices) 292 return dev; 293 } 294 return NULL; 295} 296 297status_t AudioFlinger::dumpClients(int fd, const Vector<String16>& args) 298{ 299 const size_t SIZE = 256; 300 char buffer[SIZE]; 301 String8 result; 302 303 result.append("Clients:\n"); 304 for (size_t i = 0; i < mClients.size(); ++i) { 305 sp<Client> client = mClients.valueAt(i).promote(); 306 if (client != 0) { 307 snprintf(buffer, SIZE, " pid: %d\n", client->pid()); 308 result.append(buffer); 309 } 310 } 311 312 result.append("Global session refs:\n"); 313 result.append(" session pid count\n"); 314 for (size_t i = 0; i < mAudioSessionRefs.size(); i++) { 315 AudioSessionRef *r = mAudioSessionRefs[i]; 316 snprintf(buffer, SIZE, " %7d %3d %3d\n", r->mSessionid, r->mPid, r->mCnt); 317 result.append(buffer); 318 } 319 write(fd, result.string(), result.size()); 320 return NO_ERROR; 321} 322 323 324status_t AudioFlinger::dumpInternals(int fd, const Vector<String16>& args) 325{ 326 const size_t SIZE = 256; 327 char buffer[SIZE]; 328 String8 result; 329 hardware_call_state hardwareStatus = mHardwareStatus; 330 331 snprintf(buffer, SIZE, "Hardware status: %d\n" 332 "Standby Time mSec: %u\n", 333 hardwareStatus, 334 (uint32_t)(mStandbyTimeInNsecs / 1000000)); 335 result.append(buffer); 336 write(fd, result.string(), result.size()); 337 return NO_ERROR; 338} 339 340status_t AudioFlinger::dumpPermissionDenial(int fd, const Vector<String16>& args) 341{ 342 const size_t SIZE = 256; 343 char buffer[SIZE]; 344 String8 result; 345 snprintf(buffer, SIZE, "Permission Denial: " 346 "can't dump AudioFlinger from pid=%d, uid=%d\n", 347 IPCThreadState::self()->getCallingPid(), 348 IPCThreadState::self()->getCallingUid()); 349 result.append(buffer); 350 write(fd, result.string(), result.size()); 351 return NO_ERROR; 352} 353 354static bool tryLock(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 = tryLock(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 = tryLock(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 dumpClients(fd, args); 390 dumpInternals(fd, args); 391 392 // dump playback threads 393 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 394 mPlaybackThreads.valueAt(i)->dump(fd, args); 395 } 396 397 // dump record threads 398 for (size_t i = 0; i < mRecordThreads.size(); i++) { 399 mRecordThreads.valueAt(i)->dump(fd, args); 400 } 401 402 // dump all hardware devs 403 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 404 audio_hw_device_t *dev = mAudioHwDevs[i]; 405 dev->dump(dev, fd); 406 } 407 if (locked) mLock.unlock(); 408 } 409 return NO_ERROR; 410} 411 412sp<AudioFlinger::Client> AudioFlinger::registerPid_l(pid_t pid) 413{ 414 // If pid is already in the mClients wp<> map, then use that entry 415 // (for which promote() is always != 0), otherwise create a new entry and Client. 416 sp<Client> client = mClients.valueFor(pid).promote(); 417 if (client == 0) { 418 client = new Client(this, pid); 419 mClients.add(pid, client); 420 } 421 422 return client; 423} 424 425// IAudioFlinger interface 426 427 428sp<IAudioTrack> AudioFlinger::createTrack( 429 pid_t pid, 430 audio_stream_type_t streamType, 431 uint32_t sampleRate, 432 audio_format_t format, 433 uint32_t channelMask, 434 int frameCount, 435 // FIXME dead, remove from IAudioFlinger 436 uint32_t flags, 437 const sp<IMemory>& sharedBuffer, 438 audio_io_handle_t output, 439 bool isTimed, 440 int *sessionId, 441 status_t *status) 442{ 443 sp<PlaybackThread::Track> track; 444 sp<TrackHandle> trackHandle; 445 sp<Client> client; 446 status_t lStatus; 447 int lSessionId; 448 449 // client AudioTrack::set already implements AUDIO_STREAM_DEFAULT => AUDIO_STREAM_MUSIC, 450 // but if someone uses binder directly they could bypass that and cause us to crash 451 if (uint32_t(streamType) >= AUDIO_STREAM_CNT) { 452 ALOGE("createTrack() invalid stream type %d", streamType); 453 lStatus = BAD_VALUE; 454 goto Exit; 455 } 456 457 { 458 Mutex::Autolock _l(mLock); 459 PlaybackThread *thread = checkPlaybackThread_l(output); 460 PlaybackThread *effectThread = NULL; 461 if (thread == NULL) { 462 ALOGE("unknown output thread"); 463 lStatus = BAD_VALUE; 464 goto Exit; 465 } 466 467 client = registerPid_l(pid); 468 469 ALOGV("createTrack() sessionId: %d", (sessionId == NULL) ? -2 : *sessionId); 470 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 471 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 472 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 473 if (mPlaybackThreads.keyAt(i) != output) { 474 // prevent same audio session on different output threads 475 uint32_t sessions = t->hasAudioSession(*sessionId); 476 if (sessions & PlaybackThread::TRACK_SESSION) { 477 ALOGE("createTrack() session ID %d already in use", *sessionId); 478 lStatus = BAD_VALUE; 479 goto Exit; 480 } 481 // check if an effect with same session ID is waiting for a track to be created 482 if (sessions & PlaybackThread::EFFECT_SESSION) { 483 effectThread = t.get(); 484 } 485 } 486 } 487 lSessionId = *sessionId; 488 } else { 489 // if no audio session id is provided, create one here 490 lSessionId = nextUniqueId(); 491 if (sessionId != NULL) { 492 *sessionId = lSessionId; 493 } 494 } 495 ALOGV("createTrack() lSessionId: %d", lSessionId); 496 497 track = thread->createTrack_l(client, streamType, sampleRate, format, 498 channelMask, frameCount, sharedBuffer, lSessionId, isTimed, &lStatus); 499 500 // move effect chain to this output thread if an effect on same session was waiting 501 // for a track to be created 502 if (lStatus == NO_ERROR && effectThread != NULL) { 503 Mutex::Autolock _dl(thread->mLock); 504 Mutex::Autolock _sl(effectThread->mLock); 505 moveEffectChain_l(lSessionId, effectThread, thread, true); 506 } 507 } 508 if (lStatus == NO_ERROR) { 509 trackHandle = new TrackHandle(track); 510 } else { 511 // remove local strong reference to Client before deleting the Track so that the Client 512 // destructor is called by the TrackBase destructor with mLock held 513 client.clear(); 514 track.clear(); 515 } 516 517Exit: 518 if(status) { 519 *status = lStatus; 520 } 521 return trackHandle; 522} 523 524uint32_t AudioFlinger::sampleRate(audio_io_handle_t output) const 525{ 526 Mutex::Autolock _l(mLock); 527 PlaybackThread *thread = checkPlaybackThread_l(output); 528 if (thread == NULL) { 529 ALOGW("sampleRate() unknown thread %d", output); 530 return 0; 531 } 532 return thread->sampleRate(); 533} 534 535int AudioFlinger::channelCount(audio_io_handle_t output) const 536{ 537 Mutex::Autolock _l(mLock); 538 PlaybackThread *thread = checkPlaybackThread_l(output); 539 if (thread == NULL) { 540 ALOGW("channelCount() unknown thread %d", output); 541 return 0; 542 } 543 return thread->channelCount(); 544} 545 546audio_format_t AudioFlinger::format(audio_io_handle_t output) const 547{ 548 Mutex::Autolock _l(mLock); 549 PlaybackThread *thread = checkPlaybackThread_l(output); 550 if (thread == NULL) { 551 ALOGW("format() unknown thread %d", output); 552 return AUDIO_FORMAT_INVALID; 553 } 554 return thread->format(); 555} 556 557size_t AudioFlinger::frameCount(audio_io_handle_t output) const 558{ 559 Mutex::Autolock _l(mLock); 560 PlaybackThread *thread = checkPlaybackThread_l(output); 561 if (thread == NULL) { 562 ALOGW("frameCount() unknown thread %d", output); 563 return 0; 564 } 565 return thread->frameCount(); 566} 567 568uint32_t AudioFlinger::latency(audio_io_handle_t output) const 569{ 570 Mutex::Autolock _l(mLock); 571 PlaybackThread *thread = checkPlaybackThread_l(output); 572 if (thread == NULL) { 573 ALOGW("latency() unknown thread %d", output); 574 return 0; 575 } 576 return thread->latency(); 577} 578 579status_t AudioFlinger::setMasterVolume(float value) 580{ 581 status_t ret = initCheck(); 582 if (ret != NO_ERROR) { 583 return ret; 584 } 585 586 // check calling permissions 587 if (!settingsAllowed()) { 588 return PERMISSION_DENIED; 589 } 590 591 float swmv = value; 592 593 // when hw supports master volume, don't scale in sw mixer 594 if (MVS_NONE != mMasterVolumeSupportLvl) { 595 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 596 AutoMutex lock(mHardwareLock); 597 audio_hw_device_t *dev = mAudioHwDevs[i]; 598 599 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 600 if (NULL != dev->set_master_volume) { 601 dev->set_master_volume(dev, value); 602 } 603 mHardwareStatus = AUDIO_HW_IDLE; 604 } 605 606 swmv = 1.0; 607 } 608 609 Mutex::Autolock _l(mLock); 610 mMasterVolume = value; 611 mMasterVolumeSW = swmv; 612 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 613 mPlaybackThreads.valueAt(i)->setMasterVolume(swmv); 614 615 return NO_ERROR; 616} 617 618status_t AudioFlinger::setMode(audio_mode_t mode) 619{ 620 status_t ret = initCheck(); 621 if (ret != NO_ERROR) { 622 return ret; 623 } 624 625 // check calling permissions 626 if (!settingsAllowed()) { 627 return PERMISSION_DENIED; 628 } 629 if (uint32_t(mode) >= AUDIO_MODE_CNT) { 630 ALOGW("Illegal value: setMode(%d)", mode); 631 return BAD_VALUE; 632 } 633 634 { // scope for the lock 635 AutoMutex lock(mHardwareLock); 636 mHardwareStatus = AUDIO_HW_SET_MODE; 637 ret = mPrimaryHardwareDev->set_mode(mPrimaryHardwareDev, mode); 638 mHardwareStatus = AUDIO_HW_IDLE; 639 } 640 641 if (NO_ERROR == ret) { 642 Mutex::Autolock _l(mLock); 643 mMode = mode; 644 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 645 mPlaybackThreads.valueAt(i)->setMode(mode); 646 } 647 648 return ret; 649} 650 651status_t AudioFlinger::setMicMute(bool state) 652{ 653 status_t ret = initCheck(); 654 if (ret != NO_ERROR) { 655 return ret; 656 } 657 658 // check calling permissions 659 if (!settingsAllowed()) { 660 return PERMISSION_DENIED; 661 } 662 663 AutoMutex lock(mHardwareLock); 664 mHardwareStatus = AUDIO_HW_SET_MIC_MUTE; 665 ret = mPrimaryHardwareDev->set_mic_mute(mPrimaryHardwareDev, state); 666 mHardwareStatus = AUDIO_HW_IDLE; 667 return ret; 668} 669 670bool AudioFlinger::getMicMute() const 671{ 672 status_t ret = initCheck(); 673 if (ret != NO_ERROR) { 674 return false; 675 } 676 677 bool state = AUDIO_MODE_INVALID; 678 AutoMutex lock(mHardwareLock); 679 mHardwareStatus = AUDIO_HW_GET_MIC_MUTE; 680 mPrimaryHardwareDev->get_mic_mute(mPrimaryHardwareDev, &state); 681 mHardwareStatus = AUDIO_HW_IDLE; 682 return state; 683} 684 685status_t AudioFlinger::setMasterMute(bool muted) 686{ 687 // check calling permissions 688 if (!settingsAllowed()) { 689 return PERMISSION_DENIED; 690 } 691 692 Mutex::Autolock _l(mLock); 693 // This is an optimization, so PlaybackThread doesn't have to look at the one from AudioFlinger 694 mMasterMute = muted; 695 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 696 mPlaybackThreads.valueAt(i)->setMasterMute(muted); 697 698 return NO_ERROR; 699} 700 701float AudioFlinger::masterVolume() const 702{ 703 Mutex::Autolock _l(mLock); 704 return masterVolume_l(); 705} 706 707float AudioFlinger::masterVolumeSW() const 708{ 709 Mutex::Autolock _l(mLock); 710 return masterVolumeSW_l(); 711} 712 713bool AudioFlinger::masterMute() const 714{ 715 Mutex::Autolock _l(mLock); 716 return masterMute_l(); 717} 718 719float AudioFlinger::masterVolume_l() const 720{ 721 if (MVS_FULL == mMasterVolumeSupportLvl) { 722 float ret_val; 723 AutoMutex lock(mHardwareLock); 724 725 mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME; 726 ALOG_ASSERT((NULL != mPrimaryHardwareDev) && 727 (NULL != mPrimaryHardwareDev->get_master_volume), 728 "can't get master volume"); 729 730 mPrimaryHardwareDev->get_master_volume(mPrimaryHardwareDev, &ret_val); 731 mHardwareStatus = AUDIO_HW_IDLE; 732 return ret_val; 733 } 734 735 return mMasterVolume; 736} 737 738status_t AudioFlinger::setStreamVolume(audio_stream_type_t stream, float value, 739 audio_io_handle_t output) 740{ 741 // check calling permissions 742 if (!settingsAllowed()) { 743 return PERMISSION_DENIED; 744 } 745 746 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 747 ALOGE("setStreamVolume() invalid stream %d", stream); 748 return BAD_VALUE; 749 } 750 751 AutoMutex lock(mLock); 752 PlaybackThread *thread = NULL; 753 if (output) { 754 thread = checkPlaybackThread_l(output); 755 if (thread == NULL) { 756 return BAD_VALUE; 757 } 758 } 759 760 mStreamTypes[stream].volume = value; 761 762 if (thread == NULL) { 763 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 764 mPlaybackThreads.valueAt(i)->setStreamVolume(stream, value); 765 } 766 } else { 767 thread->setStreamVolume(stream, value); 768 } 769 770 return NO_ERROR; 771} 772 773status_t AudioFlinger::setStreamMute(audio_stream_type_t stream, bool muted) 774{ 775 // check calling permissions 776 if (!settingsAllowed()) { 777 return PERMISSION_DENIED; 778 } 779 780 if (uint32_t(stream) >= AUDIO_STREAM_CNT || 781 uint32_t(stream) == AUDIO_STREAM_ENFORCED_AUDIBLE) { 782 ALOGE("setStreamMute() invalid stream %d", stream); 783 return BAD_VALUE; 784 } 785 786 AutoMutex lock(mLock); 787 mStreamTypes[stream].mute = muted; 788 for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) 789 mPlaybackThreads.valueAt(i)->setStreamMute(stream, muted); 790 791 return NO_ERROR; 792} 793 794float AudioFlinger::streamVolume(audio_stream_type_t stream, audio_io_handle_t output) const 795{ 796 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 797 return 0.0f; 798 } 799 800 AutoMutex lock(mLock); 801 float volume; 802 if (output) { 803 PlaybackThread *thread = checkPlaybackThread_l(output); 804 if (thread == NULL) { 805 return 0.0f; 806 } 807 volume = thread->streamVolume(stream); 808 } else { 809 volume = streamVolume_l(stream); 810 } 811 812 return volume; 813} 814 815bool AudioFlinger::streamMute(audio_stream_type_t stream) const 816{ 817 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 818 return true; 819 } 820 821 AutoMutex lock(mLock); 822 return streamMute_l(stream); 823} 824 825status_t AudioFlinger::setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs) 826{ 827 ALOGV("setParameters(): io %d, keyvalue %s, tid %d, calling pid %d", 828 ioHandle, keyValuePairs.string(), gettid(), IPCThreadState::self()->getCallingPid()); 829 // check calling permissions 830 if (!settingsAllowed()) { 831 return PERMISSION_DENIED; 832 } 833 834 // ioHandle == 0 means the parameters are global to the audio hardware interface 835 if (ioHandle == 0) { 836 status_t final_result = NO_ERROR; 837 { 838 AutoMutex lock(mHardwareLock); 839 mHardwareStatus = AUDIO_HW_SET_PARAMETER; 840 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 841 audio_hw_device_t *dev = mAudioHwDevs[i]; 842 status_t result = dev->set_parameters(dev, keyValuePairs.string()); 843 final_result = result ?: final_result; 844 } 845 mHardwareStatus = AUDIO_HW_IDLE; 846 } 847 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 848 AudioParameter param = AudioParameter(keyValuePairs); 849 String8 value; 850 if (param.get(String8(AUDIO_PARAMETER_KEY_BT_NREC), value) == NO_ERROR) { 851 Mutex::Autolock _l(mLock); 852 bool btNrecIsOff = (value == AUDIO_PARAMETER_VALUE_OFF); 853 if (mBtNrecIsOff != btNrecIsOff) { 854 for (size_t i = 0; i < mRecordThreads.size(); i++) { 855 sp<RecordThread> thread = mRecordThreads.valueAt(i); 856 RecordThread::RecordTrack *track = thread->track(); 857 if (track != NULL) { 858 audio_devices_t device = (audio_devices_t)( 859 thread->device() & AUDIO_DEVICE_IN_ALL); 860 bool suspend = audio_is_bluetooth_sco_device(device) && btNrecIsOff; 861 thread->setEffectSuspended(FX_IID_AEC, 862 suspend, 863 track->sessionId()); 864 thread->setEffectSuspended(FX_IID_NS, 865 suspend, 866 track->sessionId()); 867 } 868 } 869 mBtNrecIsOff = btNrecIsOff; 870 } 871 } 872 return final_result; 873 } 874 875 // hold a strong ref on thread in case closeOutput() or closeInput() is called 876 // and the thread is exited once the lock is released 877 sp<ThreadBase> thread; 878 { 879 Mutex::Autolock _l(mLock); 880 thread = checkPlaybackThread_l(ioHandle); 881 if (thread == NULL) { 882 thread = checkRecordThread_l(ioHandle); 883 } else if (thread == primaryPlaybackThread_l()) { 884 // indicate output device change to all input threads for pre processing 885 AudioParameter param = AudioParameter(keyValuePairs); 886 int value; 887 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 888 for (size_t i = 0; i < mRecordThreads.size(); i++) { 889 mRecordThreads.valueAt(i)->setParameters(keyValuePairs); 890 } 891 } 892 } 893 } 894 if (thread != 0) { 895 return thread->setParameters(keyValuePairs); 896 } 897 return BAD_VALUE; 898} 899 900String8 AudioFlinger::getParameters(audio_io_handle_t ioHandle, const String8& keys) const 901{ 902// ALOGV("getParameters() io %d, keys %s, tid %d, calling pid %d", 903// ioHandle, keys.string(), gettid(), IPCThreadState::self()->getCallingPid()); 904 905 if (ioHandle == 0) { 906 String8 out_s8; 907 908 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 909 char *s; 910 { 911 AutoMutex lock(mHardwareLock); 912 mHardwareStatus = AUDIO_HW_GET_PARAMETER; 913 audio_hw_device_t *dev = mAudioHwDevs[i]; 914 s = dev->get_parameters(dev, keys.string()); 915 mHardwareStatus = AUDIO_HW_IDLE; 916 } 917 out_s8 += String8(s ? s : ""); 918 free(s); 919 } 920 return out_s8; 921 } 922 923 Mutex::Autolock _l(mLock); 924 925 PlaybackThread *playbackThread = checkPlaybackThread_l(ioHandle); 926 if (playbackThread != NULL) { 927 return playbackThread->getParameters(keys); 928 } 929 RecordThread *recordThread = checkRecordThread_l(ioHandle); 930 if (recordThread != NULL) { 931 return recordThread->getParameters(keys); 932 } 933 return String8(""); 934} 935 936size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, audio_format_t format, int channelCount) const 937{ 938 status_t ret = initCheck(); 939 if (ret != NO_ERROR) { 940 return 0; 941 } 942 943 AutoMutex lock(mHardwareLock); 944 mHardwareStatus = AUDIO_HW_GET_INPUT_BUFFER_SIZE; 945 size_t size = mPrimaryHardwareDev->get_input_buffer_size(mPrimaryHardwareDev, sampleRate, format, channelCount); 946 mHardwareStatus = AUDIO_HW_IDLE; 947 return size; 948} 949 950unsigned int AudioFlinger::getInputFramesLost(audio_io_handle_t ioHandle) const 951{ 952 if (ioHandle == 0) { 953 return 0; 954 } 955 956 Mutex::Autolock _l(mLock); 957 958 RecordThread *recordThread = checkRecordThread_l(ioHandle); 959 if (recordThread != NULL) { 960 return recordThread->getInputFramesLost(); 961 } 962 return 0; 963} 964 965status_t AudioFlinger::setVoiceVolume(float value) 966{ 967 status_t ret = initCheck(); 968 if (ret != NO_ERROR) { 969 return ret; 970 } 971 972 // check calling permissions 973 if (!settingsAllowed()) { 974 return PERMISSION_DENIED; 975 } 976 977 AutoMutex lock(mHardwareLock); 978 mHardwareStatus = AUDIO_HW_SET_VOICE_VOLUME; 979 ret = mPrimaryHardwareDev->set_voice_volume(mPrimaryHardwareDev, value); 980 mHardwareStatus = AUDIO_HW_IDLE; 981 982 return ret; 983} 984 985status_t AudioFlinger::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, 986 audio_io_handle_t output) const 987{ 988 status_t status; 989 990 Mutex::Autolock _l(mLock); 991 992 PlaybackThread *playbackThread = checkPlaybackThread_l(output); 993 if (playbackThread != NULL) { 994 return playbackThread->getRenderPosition(halFrames, dspFrames); 995 } 996 997 return BAD_VALUE; 998} 999 1000void AudioFlinger::registerClient(const sp<IAudioFlingerClient>& client) 1001{ 1002 1003 Mutex::Autolock _l(mLock); 1004 1005 pid_t pid = IPCThreadState::self()->getCallingPid(); 1006 if (mNotificationClients.indexOfKey(pid) < 0) { 1007 sp<NotificationClient> notificationClient = new NotificationClient(this, 1008 client, 1009 pid); 1010 ALOGV("registerClient() client %p, pid %d", notificationClient.get(), pid); 1011 1012 mNotificationClients.add(pid, notificationClient); 1013 1014 sp<IBinder> binder = client->asBinder(); 1015 binder->linkToDeath(notificationClient); 1016 1017 // the config change is always sent from playback or record threads to avoid deadlock 1018 // with AudioSystem::gLock 1019 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1020 mPlaybackThreads.valueAt(i)->sendConfigEvent(AudioSystem::OUTPUT_OPENED); 1021 } 1022 1023 for (size_t i = 0; i < mRecordThreads.size(); i++) { 1024 mRecordThreads.valueAt(i)->sendConfigEvent(AudioSystem::INPUT_OPENED); 1025 } 1026 } 1027} 1028 1029void AudioFlinger::removeNotificationClient(pid_t pid) 1030{ 1031 Mutex::Autolock _l(mLock); 1032 1033 mNotificationClients.removeItem(pid); 1034 1035 ALOGV("%d died, releasing its sessions", pid); 1036 size_t num = mAudioSessionRefs.size(); 1037 bool removed = false; 1038 for (size_t i = 0; i< num; ) { 1039 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 1040 ALOGV(" pid %d @ %d", ref->mPid, i); 1041 if (ref->mPid == pid) { 1042 ALOGV(" removing entry for pid %d session %d", pid, ref->mSessionid); 1043 mAudioSessionRefs.removeAt(i); 1044 delete ref; 1045 removed = true; 1046 num--; 1047 } else { 1048 i++; 1049 } 1050 } 1051 if (removed) { 1052 purgeStaleEffects_l(); 1053 } 1054} 1055 1056// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1057void AudioFlinger::audioConfigChanged_l(int event, audio_io_handle_t ioHandle, const void *param2) 1058{ 1059 size_t size = mNotificationClients.size(); 1060 for (size_t i = 0; i < size; i++) { 1061 mNotificationClients.valueAt(i)->audioFlingerClient()->ioConfigChanged(event, ioHandle, 1062 param2); 1063 } 1064} 1065 1066// removeClient_l() must be called with AudioFlinger::mLock held 1067void AudioFlinger::removeClient_l(pid_t pid) 1068{ 1069 ALOGV("removeClient_l() pid %d, tid %d, calling tid %d", pid, gettid(), IPCThreadState::self()->getCallingPid()); 1070 mClients.removeItem(pid); 1071} 1072 1073 1074// ---------------------------------------------------------------------------- 1075 1076AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, 1077 uint32_t device, type_t type) 1078 : Thread(false), 1079 mType(type), 1080 mAudioFlinger(audioFlinger), mSampleRate(0), mFrameCount(0), 1081 // mChannelMask 1082 mChannelCount(0), 1083 mFrameSize(1), mFormat(AUDIO_FORMAT_INVALID), 1084 mParamStatus(NO_ERROR), 1085 mStandby(false), mId(id), 1086 mDevice(device), 1087 mDeathRecipient(new PMDeathRecipient(this)) 1088{ 1089} 1090 1091AudioFlinger::ThreadBase::~ThreadBase() 1092{ 1093 mParamCond.broadcast(); 1094 // do not lock the mutex in destructor 1095 releaseWakeLock_l(); 1096 if (mPowerManager != 0) { 1097 sp<IBinder> binder = mPowerManager->asBinder(); 1098 binder->unlinkToDeath(mDeathRecipient); 1099 } 1100} 1101 1102void AudioFlinger::ThreadBase::exit() 1103{ 1104 ALOGV("ThreadBase::exit"); 1105 { 1106 // This lock prevents the following race in thread (uniprocessor for illustration): 1107 // if (!exitPending()) { 1108 // // context switch from here to exit() 1109 // // exit() calls requestExit(), what exitPending() observes 1110 // // exit() calls signal(), which is dropped since no waiters 1111 // // context switch back from exit() to here 1112 // mWaitWorkCV.wait(...); 1113 // // now thread is hung 1114 // } 1115 AutoMutex lock(mLock); 1116 requestExit(); 1117 mWaitWorkCV.signal(); 1118 } 1119 // When Thread::requestExitAndWait is made virtual and this method is renamed to 1120 // "virtual status_t requestExitAndWait()", replace by "return Thread::requestExitAndWait();" 1121 requestExitAndWait(); 1122} 1123 1124status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs) 1125{ 1126 status_t status; 1127 1128 ALOGV("ThreadBase::setParameters() %s", keyValuePairs.string()); 1129 Mutex::Autolock _l(mLock); 1130 1131 mNewParameters.add(keyValuePairs); 1132 mWaitWorkCV.signal(); 1133 // wait condition with timeout in case the thread loop has exited 1134 // before the request could be processed 1135 if (mParamCond.waitRelative(mLock, kSetParametersTimeoutNs) == NO_ERROR) { 1136 status = mParamStatus; 1137 mWaitWorkCV.signal(); 1138 } else { 1139 status = TIMED_OUT; 1140 } 1141 return status; 1142} 1143 1144void AudioFlinger::ThreadBase::sendConfigEvent(int event, int param) 1145{ 1146 Mutex::Autolock _l(mLock); 1147 sendConfigEvent_l(event, param); 1148} 1149 1150// sendConfigEvent_l() must be called with ThreadBase::mLock held 1151void AudioFlinger::ThreadBase::sendConfigEvent_l(int event, int param) 1152{ 1153 ConfigEvent configEvent; 1154 configEvent.mEvent = event; 1155 configEvent.mParam = param; 1156 mConfigEvents.add(configEvent); 1157 ALOGV("sendConfigEvent() num events %d event %d, param %d", mConfigEvents.size(), event, param); 1158 mWaitWorkCV.signal(); 1159} 1160 1161void AudioFlinger::ThreadBase::processConfigEvents() 1162{ 1163 mLock.lock(); 1164 while(!mConfigEvents.isEmpty()) { 1165 ALOGV("processConfigEvents() remaining events %d", mConfigEvents.size()); 1166 ConfigEvent configEvent = mConfigEvents[0]; 1167 mConfigEvents.removeAt(0); 1168 // release mLock before locking AudioFlinger mLock: lock order is always 1169 // AudioFlinger then ThreadBase to avoid cross deadlock 1170 mLock.unlock(); 1171 mAudioFlinger->mLock.lock(); 1172 audioConfigChanged_l(configEvent.mEvent, configEvent.mParam); 1173 mAudioFlinger->mLock.unlock(); 1174 mLock.lock(); 1175 } 1176 mLock.unlock(); 1177} 1178 1179status_t AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args) 1180{ 1181 const size_t SIZE = 256; 1182 char buffer[SIZE]; 1183 String8 result; 1184 1185 bool locked = tryLock(mLock); 1186 if (!locked) { 1187 snprintf(buffer, SIZE, "thread %p maybe dead locked\n", this); 1188 write(fd, buffer, strlen(buffer)); 1189 } 1190 1191 snprintf(buffer, SIZE, "standby: %d\n", mStandby); 1192 result.append(buffer); 1193 snprintf(buffer, SIZE, "Sample rate: %d\n", mSampleRate); 1194 result.append(buffer); 1195 snprintf(buffer, SIZE, "Frame count: %d\n", mFrameCount); 1196 result.append(buffer); 1197 snprintf(buffer, SIZE, "Channel Count: %d\n", mChannelCount); 1198 result.append(buffer); 1199 snprintf(buffer, SIZE, "Channel Mask: 0x%08x\n", mChannelMask); 1200 result.append(buffer); 1201 snprintf(buffer, SIZE, "Format: %d\n", mFormat); 1202 result.append(buffer); 1203 snprintf(buffer, SIZE, "Frame size: %u\n", mFrameSize); 1204 result.append(buffer); 1205 1206 snprintf(buffer, SIZE, "\nPending setParameters commands: \n"); 1207 result.append(buffer); 1208 result.append(" Index Command"); 1209 for (size_t i = 0; i < mNewParameters.size(); ++i) { 1210 snprintf(buffer, SIZE, "\n %02d ", i); 1211 result.append(buffer); 1212 result.append(mNewParameters[i]); 1213 } 1214 1215 snprintf(buffer, SIZE, "\n\nPending config events: \n"); 1216 result.append(buffer); 1217 snprintf(buffer, SIZE, " Index event param\n"); 1218 result.append(buffer); 1219 for (size_t i = 0; i < mConfigEvents.size(); i++) { 1220 snprintf(buffer, SIZE, " %02d %02d %d\n", i, mConfigEvents[i].mEvent, mConfigEvents[i].mParam); 1221 result.append(buffer); 1222 } 1223 result.append("\n"); 1224 1225 write(fd, result.string(), result.size()); 1226 1227 if (locked) { 1228 mLock.unlock(); 1229 } 1230 return NO_ERROR; 1231} 1232 1233status_t AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args) 1234{ 1235 const size_t SIZE = 256; 1236 char buffer[SIZE]; 1237 String8 result; 1238 1239 snprintf(buffer, SIZE, "\n- %d Effect Chains:\n", mEffectChains.size()); 1240 write(fd, buffer, strlen(buffer)); 1241 1242 for (size_t i = 0; i < mEffectChains.size(); ++i) { 1243 sp<EffectChain> chain = mEffectChains[i]; 1244 if (chain != 0) { 1245 chain->dump(fd, args); 1246 } 1247 } 1248 return NO_ERROR; 1249} 1250 1251void AudioFlinger::ThreadBase::acquireWakeLock() 1252{ 1253 Mutex::Autolock _l(mLock); 1254 acquireWakeLock_l(); 1255} 1256 1257void AudioFlinger::ThreadBase::acquireWakeLock_l() 1258{ 1259 if (mPowerManager == 0) { 1260 // use checkService() to avoid blocking if power service is not up yet 1261 sp<IBinder> binder = 1262 defaultServiceManager()->checkService(String16("power")); 1263 if (binder == 0) { 1264 ALOGW("Thread %s cannot connect to the power manager service", mName); 1265 } else { 1266 mPowerManager = interface_cast<IPowerManager>(binder); 1267 binder->linkToDeath(mDeathRecipient); 1268 } 1269 } 1270 if (mPowerManager != 0) { 1271 sp<IBinder> binder = new BBinder(); 1272 status_t status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK, 1273 binder, 1274 String16(mName)); 1275 if (status == NO_ERROR) { 1276 mWakeLockToken = binder; 1277 } 1278 ALOGV("acquireWakeLock_l() %s status %d", mName, status); 1279 } 1280} 1281 1282void AudioFlinger::ThreadBase::releaseWakeLock() 1283{ 1284 Mutex::Autolock _l(mLock); 1285 releaseWakeLock_l(); 1286} 1287 1288void AudioFlinger::ThreadBase::releaseWakeLock_l() 1289{ 1290 if (mWakeLockToken != 0) { 1291 ALOGV("releaseWakeLock_l() %s", mName); 1292 if (mPowerManager != 0) { 1293 mPowerManager->releaseWakeLock(mWakeLockToken, 0); 1294 } 1295 mWakeLockToken.clear(); 1296 } 1297} 1298 1299void AudioFlinger::ThreadBase::clearPowerManager() 1300{ 1301 Mutex::Autolock _l(mLock); 1302 releaseWakeLock_l(); 1303 mPowerManager.clear(); 1304} 1305 1306void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who) 1307{ 1308 sp<ThreadBase> thread = mThread.promote(); 1309 if (thread != 0) { 1310 thread->clearPowerManager(); 1311 } 1312 ALOGW("power manager service died !!!"); 1313} 1314 1315void AudioFlinger::ThreadBase::setEffectSuspended( 1316 const effect_uuid_t *type, bool suspend, int sessionId) 1317{ 1318 Mutex::Autolock _l(mLock); 1319 setEffectSuspended_l(type, suspend, sessionId); 1320} 1321 1322void AudioFlinger::ThreadBase::setEffectSuspended_l( 1323 const effect_uuid_t *type, bool suspend, int sessionId) 1324{ 1325 sp<EffectChain> chain = getEffectChain_l(sessionId); 1326 if (chain != 0) { 1327 if (type != NULL) { 1328 chain->setEffectSuspended_l(type, suspend); 1329 } else { 1330 chain->setEffectSuspendedAll_l(suspend); 1331 } 1332 } 1333 1334 updateSuspendedSessions_l(type, suspend, sessionId); 1335} 1336 1337void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain) 1338{ 1339 ssize_t index = mSuspendedSessions.indexOfKey(chain->sessionId()); 1340 if (index < 0) { 1341 return; 1342 } 1343 1344 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects = 1345 mSuspendedSessions.editValueAt(index); 1346 1347 for (size_t i = 0; i < sessionEffects.size(); i++) { 1348 sp <SuspendedSessionDesc> desc = sessionEffects.valueAt(i); 1349 for (int j = 0; j < desc->mRefCount; j++) { 1350 if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) { 1351 chain->setEffectSuspendedAll_l(true); 1352 } else { 1353 ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x", 1354 desc->mType.timeLow); 1355 chain->setEffectSuspended_l(&desc->mType, true); 1356 } 1357 } 1358 } 1359} 1360 1361void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type, 1362 bool suspend, 1363 int sessionId) 1364{ 1365 ssize_t index = mSuspendedSessions.indexOfKey(sessionId); 1366 1367 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects; 1368 1369 if (suspend) { 1370 if (index >= 0) { 1371 sessionEffects = mSuspendedSessions.editValueAt(index); 1372 } else { 1373 mSuspendedSessions.add(sessionId, sessionEffects); 1374 } 1375 } else { 1376 if (index < 0) { 1377 return; 1378 } 1379 sessionEffects = mSuspendedSessions.editValueAt(index); 1380 } 1381 1382 1383 int key = EffectChain::kKeyForSuspendAll; 1384 if (type != NULL) { 1385 key = type->timeLow; 1386 } 1387 index = sessionEffects.indexOfKey(key); 1388 1389 sp <SuspendedSessionDesc> desc; 1390 if (suspend) { 1391 if (index >= 0) { 1392 desc = sessionEffects.valueAt(index); 1393 } else { 1394 desc = new SuspendedSessionDesc(); 1395 if (type != NULL) { 1396 memcpy(&desc->mType, type, sizeof(effect_uuid_t)); 1397 } 1398 sessionEffects.add(key, desc); 1399 ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key); 1400 } 1401 desc->mRefCount++; 1402 } else { 1403 if (index < 0) { 1404 return; 1405 } 1406 desc = sessionEffects.valueAt(index); 1407 if (--desc->mRefCount == 0) { 1408 ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key); 1409 sessionEffects.removeItemsAt(index); 1410 if (sessionEffects.isEmpty()) { 1411 ALOGV("updateSuspendedSessions_l() restore removing session %d", 1412 sessionId); 1413 mSuspendedSessions.removeItem(sessionId); 1414 } 1415 } 1416 } 1417 if (!sessionEffects.isEmpty()) { 1418 mSuspendedSessions.replaceValueFor(sessionId, sessionEffects); 1419 } 1420} 1421 1422void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 1423 bool enabled, 1424 int sessionId) 1425{ 1426 Mutex::Autolock _l(mLock); 1427 checkSuspendOnEffectEnabled_l(effect, enabled, sessionId); 1428} 1429 1430void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 1431 bool enabled, 1432 int sessionId) 1433{ 1434 if (mType != RECORD) { 1435 // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on 1436 // another session. This gives the priority to well behaved effect control panels 1437 // and applications not using global effects. 1438 if (sessionId != AUDIO_SESSION_OUTPUT_MIX) { 1439 setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX); 1440 } 1441 } 1442 1443 sp<EffectChain> chain = getEffectChain_l(sessionId); 1444 if (chain != 0) { 1445 chain->checkSuspendOnEffectEnabled(effect, enabled); 1446 } 1447} 1448 1449// ---------------------------------------------------------------------------- 1450 1451AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, 1452 AudioStreamOut* output, 1453 audio_io_handle_t id, 1454 uint32_t device, 1455 type_t type) 1456 : ThreadBase(audioFlinger, id, device, type), 1457 mMixBuffer(NULL), mSuspended(0), mBytesWritten(0), 1458 // Assumes constructor is called by AudioFlinger with it's mLock held, 1459 // but it would be safer to explicitly pass initial masterMute as parameter 1460 mMasterMute(audioFlinger->masterMute_l()), 1461 // mStreamTypes[] initialized in constructor body 1462 mOutput(output), 1463 // Assumes constructor is called by AudioFlinger with it's mLock held, 1464 // but it would be safer to explicitly pass initial masterVolume as parameter 1465 mMasterVolume(audioFlinger->masterVolumeSW_l()), 1466 mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false), 1467 // mMixerStatus 1468 mPrevMixerStatus(MIXER_IDLE), 1469 standbyDelay(AudioFlinger::mStandbyTimeInNsecs) 1470{ 1471 snprintf(mName, kNameLength, "AudioOut_%X", id); 1472 1473 readOutputParameters(); 1474 1475 // mStreamTypes[AUDIO_STREAM_CNT] is initialized by stream_type_t default constructor 1476 // There is no AUDIO_STREAM_MIN, and ++ operator does not compile 1477 for (audio_stream_type_t stream = (audio_stream_type_t) 0; stream < AUDIO_STREAM_CNT; 1478 stream = (audio_stream_type_t) (stream + 1)) { 1479 mStreamTypes[stream].volume = mAudioFlinger->streamVolume_l(stream); 1480 mStreamTypes[stream].mute = mAudioFlinger->streamMute_l(stream); 1481 // initialized by stream_type_t default constructor 1482 // mStreamTypes[stream].valid = true; 1483 } 1484 // mStreamTypes[AUDIO_STREAM_CNT] exists but isn't explicitly initialized here, 1485 // because mAudioFlinger doesn't have one to copy from 1486} 1487 1488AudioFlinger::PlaybackThread::~PlaybackThread() 1489{ 1490 delete [] mMixBuffer; 1491} 1492 1493status_t AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) 1494{ 1495 dumpInternals(fd, args); 1496 dumpTracks(fd, args); 1497 dumpEffectChains(fd, args); 1498 return NO_ERROR; 1499} 1500 1501status_t AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args) 1502{ 1503 const size_t SIZE = 256; 1504 char buffer[SIZE]; 1505 String8 result; 1506 1507 snprintf(buffer, SIZE, "Output thread %p tracks\n", this); 1508 result.append(buffer); 1509 result.append(" Name Clien Typ Fmt Chn mask Session Buf S M F SRate LeftV RighV Serv User Main buf Aux Buf\n"); 1510 for (size_t i = 0; i < mTracks.size(); ++i) { 1511 sp<Track> track = mTracks[i]; 1512 if (track != 0) { 1513 track->dump(buffer, SIZE); 1514 result.append(buffer); 1515 } 1516 } 1517 1518 snprintf(buffer, SIZE, "Output thread %p active tracks\n", this); 1519 result.append(buffer); 1520 result.append(" Name Clien Typ Fmt Chn mask Session Buf S M F SRate LeftV RighV Serv User Main buf Aux Buf\n"); 1521 for (size_t i = 0; i < mActiveTracks.size(); ++i) { 1522 sp<Track> track = mActiveTracks[i].promote(); 1523 if (track != 0) { 1524 track->dump(buffer, SIZE); 1525 result.append(buffer); 1526 } 1527 } 1528 write(fd, result.string(), result.size()); 1529 return NO_ERROR; 1530} 1531 1532status_t AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 1533{ 1534 const size_t SIZE = 256; 1535 char buffer[SIZE]; 1536 String8 result; 1537 1538 snprintf(buffer, SIZE, "\nOutput thread %p internals\n", this); 1539 result.append(buffer); 1540 snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n", ns2ms(systemTime() - mLastWriteTime)); 1541 result.append(buffer); 1542 snprintf(buffer, SIZE, "total writes: %d\n", mNumWrites); 1543 result.append(buffer); 1544 snprintf(buffer, SIZE, "delayed writes: %d\n", mNumDelayedWrites); 1545 result.append(buffer); 1546 snprintf(buffer, SIZE, "blocked in write: %d\n", mInWrite); 1547 result.append(buffer); 1548 snprintf(buffer, SIZE, "suspend count: %d\n", mSuspended); 1549 result.append(buffer); 1550 snprintf(buffer, SIZE, "mix buffer : %p\n", mMixBuffer); 1551 result.append(buffer); 1552 write(fd, result.string(), result.size()); 1553 1554 dumpBase(fd, args); 1555 1556 return NO_ERROR; 1557} 1558 1559// Thread virtuals 1560status_t AudioFlinger::PlaybackThread::readyToRun() 1561{ 1562 status_t status = initCheck(); 1563 if (status == NO_ERROR) { 1564 ALOGI("AudioFlinger's thread %p ready to run", this); 1565 } else { 1566 ALOGE("No working audio driver found."); 1567 } 1568 return status; 1569} 1570 1571void AudioFlinger::PlaybackThread::onFirstRef() 1572{ 1573 run(mName, ANDROID_PRIORITY_URGENT_AUDIO); 1574} 1575 1576// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 1577sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 1578 const sp<AudioFlinger::Client>& client, 1579 audio_stream_type_t streamType, 1580 uint32_t sampleRate, 1581 audio_format_t format, 1582 uint32_t channelMask, 1583 int frameCount, 1584 const sp<IMemory>& sharedBuffer, 1585 int sessionId, 1586 bool isTimed, 1587 status_t *status) 1588{ 1589 sp<Track> track; 1590 status_t lStatus; 1591 1592 if (mType == DIRECT) { 1593 if ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) { 1594 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1595 ALOGE("createTrack_l() Bad parameter: sampleRate %d format %d, channelMask 0x%08x \"" 1596 "for output %p with format %d", 1597 sampleRate, format, channelMask, mOutput, mFormat); 1598 lStatus = BAD_VALUE; 1599 goto Exit; 1600 } 1601 } 1602 } else { 1603 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 1604 if (sampleRate > mSampleRate*2) { 1605 ALOGE("Sample rate out of range: %d mSampleRate %d", sampleRate, mSampleRate); 1606 lStatus = BAD_VALUE; 1607 goto Exit; 1608 } 1609 } 1610 1611 lStatus = initCheck(); 1612 if (lStatus != NO_ERROR) { 1613 ALOGE("Audio driver not initialized."); 1614 goto Exit; 1615 } 1616 1617 { // scope for mLock 1618 Mutex::Autolock _l(mLock); 1619 1620 // all tracks in same audio session must share the same routing strategy otherwise 1621 // conflicts will happen when tracks are moved from one output to another by audio policy 1622 // manager 1623 uint32_t strategy = AudioSystem::getStrategyForStream(streamType); 1624 for (size_t i = 0; i < mTracks.size(); ++i) { 1625 sp<Track> t = mTracks[i]; 1626 if (t != 0 && !t->isOutputTrack()) { 1627 uint32_t actual = AudioSystem::getStrategyForStream(t->streamType()); 1628 if (sessionId == t->sessionId() && strategy != actual) { 1629 ALOGE("createTrack_l() mismatched strategy; expected %u but found %u", 1630 strategy, actual); 1631 lStatus = BAD_VALUE; 1632 goto Exit; 1633 } 1634 } 1635 } 1636 1637 if (!isTimed) { 1638 track = new Track(this, client, streamType, sampleRate, format, 1639 channelMask, frameCount, sharedBuffer, sessionId); 1640 } else { 1641 track = TimedTrack::create(this, client, streamType, sampleRate, format, 1642 channelMask, frameCount, sharedBuffer, sessionId); 1643 } 1644 if (track == NULL || track->getCblk() == NULL || track->name() < 0) { 1645 lStatus = NO_MEMORY; 1646 goto Exit; 1647 } 1648 mTracks.add(track); 1649 1650 sp<EffectChain> chain = getEffectChain_l(sessionId); 1651 if (chain != 0) { 1652 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1653 track->setMainBuffer(chain->inBuffer()); 1654 chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType())); 1655 chain->incTrackCnt(); 1656 } 1657 1658 // invalidate track immediately if the stream type was moved to another thread since 1659 // createTrack() was called by the client process. 1660 if (!mStreamTypes[streamType].valid) { 1661 ALOGW("createTrack_l() on thread %p: invalidating track on stream %d", 1662 this, streamType); 1663 android_atomic_or(CBLK_INVALID_ON, &track->mCblk->flags); 1664 } 1665 } 1666 lStatus = NO_ERROR; 1667 1668Exit: 1669 if(status) { 1670 *status = lStatus; 1671 } 1672 return track; 1673} 1674 1675uint32_t AudioFlinger::PlaybackThread::latency() const 1676{ 1677 Mutex::Autolock _l(mLock); 1678 if (initCheck() == NO_ERROR) { 1679 return mOutput->stream->get_latency(mOutput->stream); 1680 } else { 1681 return 0; 1682 } 1683} 1684 1685void AudioFlinger::PlaybackThread::setMasterVolume(float value) 1686{ 1687 Mutex::Autolock _l(mLock); 1688 mMasterVolume = value; 1689} 1690 1691void AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1692{ 1693 Mutex::Autolock _l(mLock); 1694 setMasterMute_l(muted); 1695} 1696 1697void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) 1698{ 1699 Mutex::Autolock _l(mLock); 1700 mStreamTypes[stream].volume = value; 1701} 1702 1703void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) 1704{ 1705 Mutex::Autolock _l(mLock); 1706 mStreamTypes[stream].mute = muted; 1707} 1708 1709float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const 1710{ 1711 Mutex::Autolock _l(mLock); 1712 return mStreamTypes[stream].volume; 1713} 1714 1715// addTrack_l() must be called with ThreadBase::mLock held 1716status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 1717{ 1718 status_t status = ALREADY_EXISTS; 1719 1720 // set retry count for buffer fill 1721 track->mRetryCount = kMaxTrackStartupRetries; 1722 if (mActiveTracks.indexOf(track) < 0) { 1723 // the track is newly added, make sure it fills up all its 1724 // buffers before playing. This is to ensure the client will 1725 // effectively get the latency it requested. 1726 track->mFillingUpStatus = Track::FS_FILLING; 1727 track->mResetDone = false; 1728 mActiveTracks.add(track); 1729 if (track->mainBuffer() != mMixBuffer) { 1730 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1731 if (chain != 0) { 1732 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), track->sessionId()); 1733 chain->incActiveTrackCnt(); 1734 } 1735 } 1736 1737 status = NO_ERROR; 1738 } 1739 1740 ALOGV("mWaitWorkCV.broadcast"); 1741 mWaitWorkCV.broadcast(); 1742 1743 return status; 1744} 1745 1746// destroyTrack_l() must be called with ThreadBase::mLock held 1747void AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 1748{ 1749 track->mState = TrackBase::TERMINATED; 1750 if (mActiveTracks.indexOf(track) < 0) { 1751 removeTrack_l(track); 1752 } 1753} 1754 1755void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 1756{ 1757 mTracks.remove(track); 1758 deleteTrackName_l(track->name()); 1759 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1760 if (chain != 0) { 1761 chain->decTrackCnt(); 1762 } 1763} 1764 1765String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 1766{ 1767 String8 out_s8 = String8(""); 1768 char *s; 1769 1770 Mutex::Autolock _l(mLock); 1771 if (initCheck() != NO_ERROR) { 1772 return out_s8; 1773 } 1774 1775 s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 1776 out_s8 = String8(s); 1777 free(s); 1778 return out_s8; 1779} 1780 1781// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1782void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) { 1783 AudioSystem::OutputDescriptor desc; 1784 void *param2 = NULL; 1785 1786 ALOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event, param); 1787 1788 switch (event) { 1789 case AudioSystem::OUTPUT_OPENED: 1790 case AudioSystem::OUTPUT_CONFIG_CHANGED: 1791 desc.channels = mChannelMask; 1792 desc.samplingRate = mSampleRate; 1793 desc.format = mFormat; 1794 desc.frameCount = mFrameCount; 1795 desc.latency = latency(); 1796 param2 = &desc; 1797 break; 1798 1799 case AudioSystem::STREAM_CONFIG_CHANGED: 1800 param2 = ¶m; 1801 case AudioSystem::OUTPUT_CLOSED: 1802 default: 1803 break; 1804 } 1805 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 1806} 1807 1808void AudioFlinger::PlaybackThread::readOutputParameters() 1809{ 1810 mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common); 1811 mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common); 1812 mChannelCount = (uint16_t)popcount(mChannelMask); 1813 mFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 1814 mFrameSize = audio_stream_frame_size(&mOutput->stream->common); 1815 mFrameCount = mOutput->stream->common.get_buffer_size(&mOutput->stream->common) / mFrameSize; 1816 1817 // FIXME - Current mixer implementation only supports stereo output: Always 1818 // Allocate a stereo buffer even if HW output is mono. 1819 delete[] mMixBuffer; 1820 mMixBuffer = new int16_t[mFrameCount * 2]; 1821 memset(mMixBuffer, 0, mFrameCount * 2 * sizeof(int16_t)); 1822 1823 // force reconfiguration of effect chains and engines to take new buffer size and audio 1824 // parameters into account 1825 // Note that mLock is not held when readOutputParameters() is called from the constructor 1826 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 1827 // matter. 1828 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 1829 Vector< sp<EffectChain> > effectChains = mEffectChains; 1830 for (size_t i = 0; i < effectChains.size(); i ++) { 1831 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 1832 } 1833} 1834 1835status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames) 1836{ 1837 if (halFrames == NULL || dspFrames == NULL) { 1838 return BAD_VALUE; 1839 } 1840 Mutex::Autolock _l(mLock); 1841 if (initCheck() != NO_ERROR) { 1842 return INVALID_OPERATION; 1843 } 1844 *halFrames = mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 1845 1846 return mOutput->stream->get_render_position(mOutput->stream, dspFrames); 1847} 1848 1849uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) 1850{ 1851 Mutex::Autolock _l(mLock); 1852 uint32_t result = 0; 1853 if (getEffectChain_l(sessionId) != 0) { 1854 result = EFFECT_SESSION; 1855 } 1856 1857 for (size_t i = 0; i < mTracks.size(); ++i) { 1858 sp<Track> track = mTracks[i]; 1859 if (sessionId == track->sessionId() && 1860 !(track->mCblk->flags & CBLK_INVALID_MSK)) { 1861 result |= TRACK_SESSION; 1862 break; 1863 } 1864 } 1865 1866 return result; 1867} 1868 1869uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 1870{ 1871 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 1872 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 1873 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 1874 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1875 } 1876 for (size_t i = 0; i < mTracks.size(); i++) { 1877 sp<Track> track = mTracks[i]; 1878 if (sessionId == track->sessionId() && 1879 !(track->mCblk->flags & CBLK_INVALID_MSK)) { 1880 return AudioSystem::getStrategyForStream(track->streamType()); 1881 } 1882 } 1883 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1884} 1885 1886 1887AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 1888{ 1889 Mutex::Autolock _l(mLock); 1890 return mOutput; 1891} 1892 1893AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 1894{ 1895 Mutex::Autolock _l(mLock); 1896 AudioStreamOut *output = mOutput; 1897 mOutput = NULL; 1898 return output; 1899} 1900 1901// this method must always be called either with ThreadBase mLock held or inside the thread loop 1902audio_stream_t* AudioFlinger::PlaybackThread::stream() 1903{ 1904 if (mOutput == NULL) { 1905 return NULL; 1906 } 1907 return &mOutput->stream->common; 1908} 1909 1910uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() 1911{ 1912 // A2DP output latency is not due only to buffering capacity. It also reflects encoding, 1913 // decoding and transfer time. So sleeping for half of the latency would likely cause 1914 // underruns 1915 if (audio_is_a2dp_device((audio_devices_t)mDevice)) { 1916 return (uint32_t)((uint32_t)((mFrameCount * 1000) / mSampleRate) * 1000); 1917 } else { 1918 return (uint32_t)(mOutput->stream->get_latency(mOutput->stream) * 1000) / 2; 1919 } 1920} 1921 1922// ---------------------------------------------------------------------------- 1923 1924AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 1925 audio_io_handle_t id, uint32_t device, type_t type) 1926 : PlaybackThread(audioFlinger, output, id, device, type) 1927{ 1928 mAudioMixer = new AudioMixer(mFrameCount, mSampleRate); 1929 // FIXME - Current mixer implementation only supports stereo output 1930 if (mChannelCount == 1) { 1931 ALOGE("Invalid audio hardware channel count"); 1932 } 1933} 1934 1935AudioFlinger::MixerThread::~MixerThread() 1936{ 1937 delete mAudioMixer; 1938} 1939 1940class CpuStats { 1941public: 1942 void sample(); 1943#ifdef DEBUG_CPU_USAGE 1944private: 1945 ThreadCpuUsage mCpu; 1946#endif 1947}; 1948 1949void CpuStats::sample() { 1950#ifdef DEBUG_CPU_USAGE 1951 const CentralTendencyStatistics& stats = mCpu.statistics(); 1952 mCpu.sampleAndEnable(); 1953 unsigned n = stats.n(); 1954 // mCpu.elapsed() is expensive, so don't call it every loop 1955 if ((n & 127) == 1) { 1956 long long elapsed = mCpu.elapsed(); 1957 if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) { 1958 double perLoop = elapsed / (double) n; 1959 double perLoop100 = perLoop * 0.01; 1960 double mean = stats.mean(); 1961 double stddev = stats.stddev(); 1962 double minimum = stats.minimum(); 1963 double maximum = stats.maximum(); 1964 mCpu.resetStatistics(); 1965 ALOGI("CPU usage over past %.1f secs (%u mixer loops at %.1f mean ms per loop):\n us per mix loop: mean=%.0f stddev=%.0f min=%.0f max=%.0f\n %% of wall: mean=%.1f stddev=%.1f min=%.1f max=%.1f", 1966 elapsed * .000000001, n, perLoop * .000001, 1967 mean * .001, 1968 stddev * .001, 1969 minimum * .001, 1970 maximum * .001, 1971 mean / perLoop100, 1972 stddev / perLoop100, 1973 minimum / perLoop100, 1974 maximum / perLoop100); 1975 } 1976 } 1977#endif 1978}; 1979 1980void AudioFlinger::PlaybackThread::checkSilentMode_l() 1981{ 1982 if (!mMasterMute) { 1983 char value[PROPERTY_VALUE_MAX]; 1984 if (property_get("ro.audio.silent", value, "0") > 0) { 1985 char *endptr; 1986 unsigned long ul = strtoul(value, &endptr, 0); 1987 if (*endptr == '\0' && ul != 0) { 1988 ALOGD("Silence is golden"); 1989 // The setprop command will not allow a property to be changed after 1990 // the first time it is set, so we don't have to worry about un-muting. 1991 setMasterMute_l(true); 1992 } 1993 } 1994 } 1995} 1996 1997bool AudioFlinger::PlaybackThread::threadLoop() 1998{ 1999 Vector< sp<Track> > tracksToRemove; 2000 2001 standbyTime = systemTime(); 2002 2003 // MIXER 2004 nsecs_t lastWarning = 0; 2005if (mType == MIXER) { 2006 longStandbyExit = false; 2007} 2008 2009 // DUPLICATING 2010 // FIXME could this be made local to while loop? 2011 writeFrames = 0; 2012 2013 cacheParameters_l(); 2014 sleepTime = idleSleepTime; 2015 2016if (mType == MIXER) { 2017 sleepTimeShift = 0; 2018} 2019 2020 // MIXER 2021 CpuStats cpuStats; 2022 2023 acquireWakeLock(); 2024 2025 while (!exitPending()) 2026 { 2027if (mType == MIXER) { 2028 cpuStats.sample(); 2029} 2030 2031 Vector< sp<EffectChain> > effectChains; 2032 2033 processConfigEvents(); 2034 2035 mMixerStatus = MIXER_IDLE; 2036 { // scope for mLock 2037 2038 Mutex::Autolock _l(mLock); 2039 2040 if (checkForNewParameters_l()) { 2041 cacheParameters_l(); 2042 } 2043 2044 saveOutputTracks(); 2045 2046 // put audio hardware into standby after short delay 2047 if (CC_UNLIKELY((!mActiveTracks.size() && systemTime() > standbyTime) || 2048 mSuspended > 0)) { 2049 if (!mStandby) { 2050 2051 threadLoop_standby(); 2052 2053 mStandby = true; 2054 mBytesWritten = 0; 2055 } 2056 2057 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2058 // we're about to wait, flush the binder command buffer 2059 IPCThreadState::self()->flushCommands(); 2060 2061 clearOutputTracks(); 2062 2063 if (exitPending()) break; 2064 2065 releaseWakeLock_l(); 2066 // wait until we have something to do... 2067 ALOGV("Thread %p type %d TID %d going to sleep", this, mType, gettid()); 2068 mWaitWorkCV.wait(mLock); 2069 ALOGV("Thread %p type %d TID %d waking up", this, mType, gettid()); 2070 acquireWakeLock_l(); 2071 2072 mPrevMixerStatus = MIXER_IDLE; 2073 2074 checkSilentMode_l(); 2075 2076 standbyTime = systemTime() + standbyDelay; 2077 sleepTime = idleSleepTime; 2078 if (mType == MIXER) { 2079 sleepTimeShift = 0; 2080 } 2081 2082 continue; 2083 } 2084 } 2085 2086 mixer_state newMixerStatus = prepareTracks_l(&tracksToRemove); 2087 // Shift in the new status; this could be a queue if it's 2088 // useful to filter the mixer status over several cycles. 2089 mPrevMixerStatus = mMixerStatus; 2090 mMixerStatus = newMixerStatus; 2091 2092 // prevent any changes in effect chain list and in each effect chain 2093 // during mixing and effect process as the audio buffers could be deleted 2094 // or modified if an effect is created or deleted 2095 lockEffectChains_l(effectChains); 2096 } 2097 2098 if (CC_LIKELY(mMixerStatus == MIXER_TRACKS_READY)) { 2099 threadLoop_mix(); 2100 } else { 2101 threadLoop_sleepTime(); 2102 } 2103 2104 if (mSuspended > 0) { 2105 sleepTime = suspendSleepTimeUs(); 2106 } 2107 2108 // only process effects if we're going to write 2109 if (sleepTime == 0) { 2110 for (size_t i = 0; i < effectChains.size(); i ++) { 2111 effectChains[i]->process_l(); 2112 } 2113 } 2114 2115 // enable changes in effect chain 2116 unlockEffectChains(effectChains); 2117 2118 // sleepTime == 0 means we must write to audio hardware 2119 if (sleepTime == 0) { 2120 2121 threadLoop_write(); 2122 2123if (mType == MIXER) { 2124 // write blocked detection 2125 nsecs_t now = systemTime(); 2126 nsecs_t delta = now - mLastWriteTime; 2127 if (!mStandby && delta > maxPeriod) { 2128 mNumDelayedWrites++; 2129 if ((now - lastWarning) > kWarningThrottleNs) { 2130 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2131 ns2ms(delta), mNumDelayedWrites, this); 2132 lastWarning = now; 2133 } 2134 // FIXME this is broken: longStandbyExit should be handled out of the if() and with 2135 // a different threshold. Or completely removed for what it is worth anyway... 2136 if (mStandby) { 2137 longStandbyExit = true; 2138 } 2139 } 2140} 2141 2142 mStandby = false; 2143 } else { 2144 usleep(sleepTime); 2145 } 2146 2147 // finally let go of removed track(s), without the lock held 2148 // since we can't guarantee the destructors won't acquire that 2149 // same lock. 2150 tracksToRemove.clear(); 2151 2152 // FIXME I don't understand the need for this here; 2153 // it was in the original code but maybe the 2154 // assignment in saveOutputTracks() makes this unnecessary? 2155 clearOutputTracks(); 2156 2157 // Effect chains will be actually deleted here if they were removed from 2158 // mEffectChains list during mixing or effects processing 2159 effectChains.clear(); 2160 2161 // FIXME Note that the above .clear() is no longer necessary since effectChains 2162 // is now local to this block, but will keep it for now (at least until merge done). 2163 } 2164 2165if (mType == MIXER || mType == DIRECT) { 2166 // put output stream into standby mode 2167 if (!mStandby) { 2168 mOutput->stream->common.standby(&mOutput->stream->common); 2169 } 2170} 2171if (mType == DUPLICATING) { 2172 // for DuplicatingThread, standby mode is handled by the outputTracks 2173} 2174 2175 releaseWakeLock(); 2176 2177 ALOGV("Thread %p type %d exiting", this, mType); 2178 return false; 2179} 2180 2181// shared by MIXER and DIRECT, overridden by DUPLICATING 2182void AudioFlinger::PlaybackThread::threadLoop_write() 2183{ 2184 // FIXME rewrite to reduce number of system calls 2185 mLastWriteTime = systemTime(); 2186 mInWrite = true; 2187 mBytesWritten += mixBufferSize; 2188 int bytesWritten = (int)mOutput->stream->write(mOutput->stream, mMixBuffer, mixBufferSize); 2189 if (bytesWritten < 0) mBytesWritten -= mixBufferSize; 2190 mNumWrites++; 2191 mInWrite = false; 2192} 2193 2194// shared by MIXER and DIRECT, overridden by DUPLICATING 2195void AudioFlinger::PlaybackThread::threadLoop_standby() 2196{ 2197 ALOGV("Audio hardware entering standby, mixer %p, suspend count %u", this, mSuspended); 2198 mOutput->stream->common.standby(&mOutput->stream->common); 2199} 2200 2201void AudioFlinger::MixerThread::threadLoop_mix() 2202{ 2203 // obtain the presentation timestamp of the next output buffer 2204 int64_t pts; 2205 status_t status = INVALID_OPERATION; 2206 2207 if (NULL != mOutput->stream->get_next_write_timestamp) { 2208 status = mOutput->stream->get_next_write_timestamp( 2209 mOutput->stream, &pts); 2210 } 2211 2212 if (status != NO_ERROR) { 2213 pts = AudioBufferProvider::kInvalidPTS; 2214 } 2215 2216 // mix buffers... 2217 mAudioMixer->process(pts); 2218 // increase sleep time progressively when application underrun condition clears. 2219 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 2220 // that a steady state of alternating ready/not ready conditions keeps the sleep time 2221 // such that we would underrun the audio HAL. 2222 if ((sleepTime == 0) && (sleepTimeShift > 0)) { 2223 sleepTimeShift--; 2224 } 2225 sleepTime = 0; 2226 standbyTime = systemTime() + standbyDelay; 2227 //TODO: delay standby when effects have a tail 2228} 2229 2230void AudioFlinger::MixerThread::threadLoop_sleepTime() 2231{ 2232 // If no tracks are ready, sleep once for the duration of an output 2233 // buffer size, then write 0s to the output 2234 if (sleepTime == 0) { 2235 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 2236 sleepTime = activeSleepTime >> sleepTimeShift; 2237 if (sleepTime < kMinThreadSleepTimeUs) { 2238 sleepTime = kMinThreadSleepTimeUs; 2239 } 2240 // reduce sleep time in case of consecutive application underruns to avoid 2241 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 2242 // duration we would end up writing less data than needed by the audio HAL if 2243 // the condition persists. 2244 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 2245 sleepTimeShift++; 2246 } 2247 } else { 2248 sleepTime = idleSleepTime; 2249 } 2250 } else if (mBytesWritten != 0 || 2251 (mMixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)) { 2252 memset (mMixBuffer, 0, mixBufferSize); 2253 sleepTime = 0; 2254 ALOGV_IF((mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)), "anticipated start"); 2255 } 2256 // TODO add standby time extension fct of effect tail 2257} 2258 2259// prepareTracks_l() must be called with ThreadBase::mLock held 2260AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 2261 Vector< sp<Track> > *tracksToRemove) 2262{ 2263 2264 mixer_state mixerStatus = MIXER_IDLE; 2265 // find out which tracks need to be processed 2266 size_t count = mActiveTracks.size(); 2267 size_t mixedTracks = 0; 2268 size_t tracksWithEffect = 0; 2269 2270 float masterVolume = mMasterVolume; 2271 bool masterMute = mMasterMute; 2272 2273 if (masterMute) { 2274 masterVolume = 0; 2275 } 2276 // Delegate master volume control to effect in output mix effect chain if needed 2277 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2278 if (chain != 0) { 2279 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 2280 chain->setVolume_l(&v, &v); 2281 masterVolume = (float)((v + (1 << 23)) >> 24); 2282 chain.clear(); 2283 } 2284 2285 for (size_t i=0 ; i<count ; i++) { 2286 sp<Track> t = mActiveTracks[i].promote(); 2287 if (t == 0) continue; 2288 2289 // this const just means the local variable doesn't change 2290 Track* const track = t.get(); 2291 audio_track_cblk_t* cblk = track->cblk(); 2292 2293 // The first time a track is added we wait 2294 // for all its buffers to be filled before processing it 2295 int name = track->name(); 2296 // make sure that we have enough frames to mix one full buffer. 2297 // enforce this condition only once to enable draining the buffer in case the client 2298 // app does not call stop() and relies on underrun to stop: 2299 // hence the test on (mPrevMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 2300 // during last round 2301 uint32_t minFrames = 1; 2302 if (!track->isStopped() && !track->isPausing() && 2303 (mPrevMixerStatus == MIXER_TRACKS_READY)) { 2304 if (t->sampleRate() == (int)mSampleRate) { 2305 minFrames = mFrameCount; 2306 } else { 2307 // +1 for rounding and +1 for additional sample needed for interpolation 2308 minFrames = (mFrameCount * t->sampleRate()) / mSampleRate + 1 + 1; 2309 // add frames already consumed but not yet released by the resampler 2310 // because cblk->framesReady() will include these frames 2311 minFrames += mAudioMixer->getUnreleasedFrames(track->name()); 2312 // the minimum track buffer size is normally twice the number of frames necessary 2313 // to fill one buffer and the resampler should not leave more than one buffer worth 2314 // of unreleased frames after each pass, but just in case... 2315 ALOG_ASSERT(minFrames <= cblk->frameCount); 2316 } 2317 } 2318 if ((track->framesReady() >= minFrames) && track->isReady() && 2319 !track->isPaused() && !track->isTerminated()) 2320 { 2321 //ALOGV("track %d u=%08x, s=%08x [OK] on thread %p", name, cblk->user, cblk->server, this); 2322 2323 mixedTracks++; 2324 2325 // track->mainBuffer() != mMixBuffer means there is an effect chain 2326 // connected to the track 2327 chain.clear(); 2328 if (track->mainBuffer() != mMixBuffer) { 2329 chain = getEffectChain_l(track->sessionId()); 2330 // Delegate volume control to effect in track effect chain if needed 2331 if (chain != 0) { 2332 tracksWithEffect++; 2333 } else { 2334 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on session %d", 2335 name, track->sessionId()); 2336 } 2337 } 2338 2339 2340 int param = AudioMixer::VOLUME; 2341 if (track->mFillingUpStatus == Track::FS_FILLED) { 2342 // no ramp for the first volume setting 2343 track->mFillingUpStatus = Track::FS_ACTIVE; 2344 if (track->mState == TrackBase::RESUMING) { 2345 track->mState = TrackBase::ACTIVE; 2346 param = AudioMixer::RAMP_VOLUME; 2347 } 2348 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 2349 } else if (cblk->server != 0) { 2350 // If the track is stopped before the first frame was mixed, 2351 // do not apply ramp 2352 param = AudioMixer::RAMP_VOLUME; 2353 } 2354 2355 // compute volume for this track 2356 uint32_t vl, vr, va; 2357 if (track->isMuted() || track->isPausing() || 2358 mStreamTypes[track->streamType()].mute) { 2359 vl = vr = va = 0; 2360 if (track->isPausing()) { 2361 track->setPaused(); 2362 } 2363 } else { 2364 2365 // read original volumes with volume control 2366 float typeVolume = mStreamTypes[track->streamType()].volume; 2367 float v = masterVolume * typeVolume; 2368 uint32_t vlr = cblk->getVolumeLR(); 2369 vl = vlr & 0xFFFF; 2370 vr = vlr >> 16; 2371 // track volumes come from shared memory, so can't be trusted and must be clamped 2372 if (vl > MAX_GAIN_INT) { 2373 ALOGV("Track left volume out of range: %04X", vl); 2374 vl = MAX_GAIN_INT; 2375 } 2376 if (vr > MAX_GAIN_INT) { 2377 ALOGV("Track right volume out of range: %04X", vr); 2378 vr = MAX_GAIN_INT; 2379 } 2380 // now apply the master volume and stream type volume 2381 vl = (uint32_t)(v * vl) << 12; 2382 vr = (uint32_t)(v * vr) << 12; 2383 // assuming master volume and stream type volume each go up to 1.0, 2384 // vl and vr are now in 8.24 format 2385 2386 uint16_t sendLevel = cblk->getSendLevel_U4_12(); 2387 // send level comes from shared memory and so may be corrupt 2388 if (sendLevel > MAX_GAIN_INT) { 2389 ALOGV("Track send level out of range: %04X", sendLevel); 2390 sendLevel = MAX_GAIN_INT; 2391 } 2392 va = (uint32_t)(v * sendLevel); 2393 } 2394 // Delegate volume control to effect in track effect chain if needed 2395 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 2396 // Do not ramp volume if volume is controlled by effect 2397 param = AudioMixer::VOLUME; 2398 track->mHasVolumeController = true; 2399 } else { 2400 // force no volume ramp when volume controller was just disabled or removed 2401 // from effect chain to avoid volume spike 2402 if (track->mHasVolumeController) { 2403 param = AudioMixer::VOLUME; 2404 } 2405 track->mHasVolumeController = false; 2406 } 2407 2408 // Convert volumes from 8.24 to 4.12 format 2409 // This additional clamping is needed in case chain->setVolume_l() overshot 2410 vl = (vl + (1 << 11)) >> 12; 2411 if (vl > MAX_GAIN_INT) vl = MAX_GAIN_INT; 2412 vr = (vr + (1 << 11)) >> 12; 2413 if (vr > MAX_GAIN_INT) vr = MAX_GAIN_INT; 2414 2415 if (va > MAX_GAIN_INT) va = MAX_GAIN_INT; // va is uint32_t, so no need to check for - 2416 2417 // XXX: these things DON'T need to be done each time 2418 mAudioMixer->setBufferProvider(name, track); 2419 mAudioMixer->enable(name); 2420 2421 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, (void *)vl); 2422 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)vr); 2423 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)va); 2424 mAudioMixer->setParameter( 2425 name, 2426 AudioMixer::TRACK, 2427 AudioMixer::FORMAT, (void *)track->format()); 2428 mAudioMixer->setParameter( 2429 name, 2430 AudioMixer::TRACK, 2431 AudioMixer::CHANNEL_MASK, (void *)track->channelMask()); 2432 mAudioMixer->setParameter( 2433 name, 2434 AudioMixer::RESAMPLE, 2435 AudioMixer::SAMPLE_RATE, 2436 (void *)(cblk->sampleRate)); 2437 mAudioMixer->setParameter( 2438 name, 2439 AudioMixer::TRACK, 2440 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 2441 mAudioMixer->setParameter( 2442 name, 2443 AudioMixer::TRACK, 2444 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 2445 2446 // reset retry count 2447 track->mRetryCount = kMaxTrackRetries; 2448 // If one track is ready, set the mixer ready if: 2449 // - the mixer was not ready during previous round OR 2450 // - no other track is not ready 2451 if (mPrevMixerStatus != MIXER_TRACKS_READY || 2452 mixerStatus != MIXER_TRACKS_ENABLED) { 2453 mixerStatus = MIXER_TRACKS_READY; 2454 } 2455 } else { 2456 //ALOGV("track %d u=%08x, s=%08x [NOT READY] on thread %p", name, cblk->user, cblk->server, this); 2457 if (track->isStopped()) { 2458 track->reset(); 2459 } 2460 if (track->isTerminated() || track->isStopped() || track->isPaused()) { 2461 // We have consumed all the buffers of this track. 2462 // Remove it from the list of active tracks. 2463 tracksToRemove->add(track); 2464 } else { 2465 // No buffers for this track. Give it a few chances to 2466 // fill a buffer, then remove it from active list. 2467 if (--(track->mRetryCount) <= 0) { 2468 ALOGV("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 2469 tracksToRemove->add(track); 2470 // indicate to client process that the track was disabled because of underrun 2471 android_atomic_or(CBLK_DISABLED_ON, &cblk->flags); 2472 // If one track is not ready, mark the mixer also not ready if: 2473 // - the mixer was ready during previous round OR 2474 // - no other track is ready 2475 } else if (mPrevMixerStatus == MIXER_TRACKS_READY || 2476 mixerStatus != MIXER_TRACKS_READY) { 2477 mixerStatus = MIXER_TRACKS_ENABLED; 2478 } 2479 } 2480 mAudioMixer->disable(name); 2481 } 2482 } 2483 2484 // remove all the tracks that need to be... 2485 count = tracksToRemove->size(); 2486 if (CC_UNLIKELY(count)) { 2487 for (size_t i=0 ; i<count ; i++) { 2488 const sp<Track>& track = tracksToRemove->itemAt(i); 2489 mActiveTracks.remove(track); 2490 if (track->mainBuffer() != mMixBuffer) { 2491 chain = getEffectChain_l(track->sessionId()); 2492 if (chain != 0) { 2493 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), track->sessionId()); 2494 chain->decActiveTrackCnt(); 2495 } 2496 } 2497 if (track->isTerminated()) { 2498 removeTrack_l(track); 2499 } 2500 } 2501 } 2502 2503 // mix buffer must be cleared if all tracks are connected to an 2504 // effect chain as in this case the mixer will not write to 2505 // mix buffer and track effects will accumulate into it 2506 if (mixedTracks != 0 && mixedTracks == tracksWithEffect) { 2507 memset(mMixBuffer, 0, mFrameCount * mChannelCount * sizeof(int16_t)); 2508 } 2509 2510 return mixerStatus; 2511} 2512 2513/* 2514The derived values that are cached: 2515 - mixBufferSize from frame count * frame size 2516 - activeSleepTime from activeSleepTimeUs() 2517 - idleSleepTime from idleSleepTimeUs() 2518 - standbyDelay from mActiveSleepTimeUs (DIRECT only) 2519 - maxPeriod from frame count and sample rate (MIXER only) 2520 2521The parameters that affect these derived values are: 2522 - frame count 2523 - frame size 2524 - sample rate 2525 - device type: A2DP or not 2526 - device latency 2527 - format: PCM or not 2528 - active sleep time 2529 - idle sleep time 2530*/ 2531 2532void AudioFlinger::PlaybackThread::cacheParameters_l() 2533{ 2534 mixBufferSize = mFrameCount * mFrameSize; 2535 activeSleepTime = activeSleepTimeUs(); 2536 idleSleepTime = idleSleepTimeUs(); 2537} 2538 2539void AudioFlinger::MixerThread::invalidateTracks(audio_stream_type_t streamType) 2540{ 2541 ALOGV ("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 2542 this, streamType, mTracks.size()); 2543 Mutex::Autolock _l(mLock); 2544 2545 size_t size = mTracks.size(); 2546 for (size_t i = 0; i < size; i++) { 2547 sp<Track> t = mTracks[i]; 2548 if (t->streamType() == streamType) { 2549 android_atomic_or(CBLK_INVALID_ON, &t->mCblk->flags); 2550 t->mCblk->cv.signal(); 2551 } 2552 } 2553} 2554 2555void AudioFlinger::PlaybackThread::setStreamValid(audio_stream_type_t streamType, bool valid) 2556{ 2557 ALOGV ("PlaybackThread::setStreamValid() thread %p, streamType %d, valid %d", 2558 this, streamType, valid); 2559 Mutex::Autolock _l(mLock); 2560 2561 mStreamTypes[streamType].valid = valid; 2562} 2563 2564// getTrackName_l() must be called with ThreadBase::mLock held 2565int AudioFlinger::MixerThread::getTrackName_l() 2566{ 2567 return mAudioMixer->getTrackName(); 2568} 2569 2570// deleteTrackName_l() must be called with ThreadBase::mLock held 2571void AudioFlinger::MixerThread::deleteTrackName_l(int name) 2572{ 2573 ALOGV("remove track (%d) and delete from mixer", name); 2574 mAudioMixer->deleteTrackName(name); 2575} 2576 2577// checkForNewParameters_l() must be called with ThreadBase::mLock held 2578bool AudioFlinger::MixerThread::checkForNewParameters_l() 2579{ 2580 bool reconfig = false; 2581 2582 while (!mNewParameters.isEmpty()) { 2583 status_t status = NO_ERROR; 2584 String8 keyValuePair = mNewParameters[0]; 2585 AudioParameter param = AudioParameter(keyValuePair); 2586 int value; 2587 2588 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 2589 reconfig = true; 2590 } 2591 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 2592 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 2593 status = BAD_VALUE; 2594 } else { 2595 reconfig = true; 2596 } 2597 } 2598 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 2599 if (value != AUDIO_CHANNEL_OUT_STEREO) { 2600 status = BAD_VALUE; 2601 } else { 2602 reconfig = true; 2603 } 2604 } 2605 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 2606 // do not accept frame count changes if tracks are open as the track buffer 2607 // size depends on frame count and correct behavior would not be guaranteed 2608 // if frame count is changed after track creation 2609 if (!mTracks.isEmpty()) { 2610 status = INVALID_OPERATION; 2611 } else { 2612 reconfig = true; 2613 } 2614 } 2615 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 2616 // when changing the audio output device, call addBatteryData to notify 2617 // the change 2618 if ((int)mDevice != value) { 2619 uint32_t params = 0; 2620 // check whether speaker is on 2621 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 2622 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 2623 } 2624 2625 int deviceWithoutSpeaker 2626 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 2627 // check if any other device (except speaker) is on 2628 if (value & deviceWithoutSpeaker ) { 2629 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 2630 } 2631 2632 if (params != 0) { 2633 addBatteryData(params); 2634 } 2635 } 2636 2637 // forward device change to effects that have requested to be 2638 // aware of attached audio device. 2639 mDevice = (uint32_t)value; 2640 for (size_t i = 0; i < mEffectChains.size(); i++) { 2641 mEffectChains[i]->setDevice_l(mDevice); 2642 } 2643 } 2644 2645 if (status == NO_ERROR) { 2646 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 2647 keyValuePair.string()); 2648 if (!mStandby && status == INVALID_OPERATION) { 2649 mOutput->stream->common.standby(&mOutput->stream->common); 2650 mStandby = true; 2651 mBytesWritten = 0; 2652 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 2653 keyValuePair.string()); 2654 } 2655 if (status == NO_ERROR && reconfig) { 2656 delete mAudioMixer; 2657 // for safety in case readOutputParameters() accesses mAudioMixer (it doesn't) 2658 mAudioMixer = NULL; 2659 readOutputParameters(); 2660 mAudioMixer = new AudioMixer(mFrameCount, mSampleRate); 2661 for (size_t i = 0; i < mTracks.size() ; i++) { 2662 int name = getTrackName_l(); 2663 if (name < 0) break; 2664 mTracks[i]->mName = name; 2665 // limit track sample rate to 2 x new output sample rate 2666 if (mTracks[i]->mCblk->sampleRate > 2 * sampleRate()) { 2667 mTracks[i]->mCblk->sampleRate = 2 * sampleRate(); 2668 } 2669 } 2670 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 2671 } 2672 } 2673 2674 mNewParameters.removeAt(0); 2675 2676 mParamStatus = status; 2677 mParamCond.signal(); 2678 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 2679 // already timed out waiting for the status and will never signal the condition. 2680 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 2681 } 2682 return reconfig; 2683} 2684 2685status_t AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 2686{ 2687 const size_t SIZE = 256; 2688 char buffer[SIZE]; 2689 String8 result; 2690 2691 PlaybackThread::dumpInternals(fd, args); 2692 2693 snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames()); 2694 result.append(buffer); 2695 write(fd, result.string(), result.size()); 2696 return NO_ERROR; 2697} 2698 2699uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() 2700{ 2701 return (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 2702} 2703 2704uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() 2705{ 2706 return (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 2707} 2708 2709void AudioFlinger::MixerThread::cacheParameters_l() 2710{ 2711 PlaybackThread::cacheParameters_l(); 2712 2713 // FIXME: Relaxed timing because of a certain device that can't meet latency 2714 // Should be reduced to 2x after the vendor fixes the driver issue 2715 // increase threshold again due to low power audio mode. The way this warning 2716 // threshold is calculated and its usefulness should be reconsidered anyway. 2717 maxPeriod = seconds(mFrameCount) / mSampleRate * 15; 2718} 2719 2720// ---------------------------------------------------------------------------- 2721AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 2722 AudioStreamOut* output, audio_io_handle_t id, uint32_t device) 2723 : PlaybackThread(audioFlinger, output, id, device, DIRECT) 2724 // mLeftVolFloat, mRightVolFloat 2725 // mLeftVolShort, mRightVolShort 2726{ 2727} 2728 2729AudioFlinger::DirectOutputThread::~DirectOutputThread() 2730{ 2731} 2732 2733AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l( 2734 Vector< sp<Track> > *tracksToRemove 2735) 2736{ 2737 sp<Track> trackToRemove; 2738 2739 mixer_state mixerStatus = MIXER_IDLE; 2740 2741 // find out which tracks need to be processed 2742 if (mActiveTracks.size() != 0) { 2743 sp<Track> t = mActiveTracks[0].promote(); 2744 // The track died recently 2745 if (t == 0) return MIXER_IDLE; 2746 2747 Track* const track = t.get(); 2748 audio_track_cblk_t* cblk = track->cblk(); 2749 2750 // The first time a track is added we wait 2751 // for all its buffers to be filled before processing it 2752 if (cblk->framesReady() && track->isReady() && 2753 !track->isPaused() && !track->isTerminated()) 2754 { 2755 //ALOGV("track %d u=%08x, s=%08x [OK]", track->name(), cblk->user, cblk->server); 2756 2757 if (track->mFillingUpStatus == Track::FS_FILLED) { 2758 track->mFillingUpStatus = Track::FS_ACTIVE; 2759 mLeftVolFloat = mRightVolFloat = 0; 2760 mLeftVolShort = mRightVolShort = 0; 2761 if (track->mState == TrackBase::RESUMING) { 2762 track->mState = TrackBase::ACTIVE; 2763 rampVolume = true; 2764 } 2765 } else if (cblk->server != 0) { 2766 // If the track is stopped before the first frame was mixed, 2767 // do not apply ramp 2768 rampVolume = true; 2769 } 2770 // compute volume for this track 2771 float left, right; 2772 if (track->isMuted() || mMasterMute || track->isPausing() || 2773 mStreamTypes[track->streamType()].mute) { 2774 left = right = 0; 2775 if (track->isPausing()) { 2776 track->setPaused(); 2777 } 2778 } else { 2779 float typeVolume = mStreamTypes[track->streamType()].volume; 2780 float v = mMasterVolume * typeVolume; 2781 uint32_t vlr = cblk->getVolumeLR(); 2782 float v_clamped = v * (vlr & 0xFFFF); 2783 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 2784 left = v_clamped/MAX_GAIN; 2785 v_clamped = v * (vlr >> 16); 2786 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 2787 right = v_clamped/MAX_GAIN; 2788 } 2789 2790 if (left != mLeftVolFloat || right != mRightVolFloat) { 2791 mLeftVolFloat = left; 2792 mRightVolFloat = right; 2793 2794 // If audio HAL implements volume control, 2795 // force software volume to nominal value 2796 if (mOutput->stream->set_volume(mOutput->stream, left, right) == NO_ERROR) { 2797 left = 1.0f; 2798 right = 1.0f; 2799 } 2800 2801 // Convert volumes from float to 8.24 2802 uint32_t vl = (uint32_t)(left * (1 << 24)); 2803 uint32_t vr = (uint32_t)(right * (1 << 24)); 2804 2805 // Delegate volume control to effect in track effect chain if needed 2806 // only one effect chain can be present on DirectOutputThread, so if 2807 // there is one, the track is connected to it 2808 if (!mEffectChains.isEmpty()) { 2809 // Do not ramp volume if volume is controlled by effect 2810 if (mEffectChains[0]->setVolume_l(&vl, &vr)) { 2811 rampVolume = false; 2812 } 2813 } 2814 2815 // Convert volumes from 8.24 to 4.12 format 2816 uint32_t v_clamped = (vl + (1 << 11)) >> 12; 2817 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 2818 leftVol = (uint16_t)v_clamped; 2819 v_clamped = (vr + (1 << 11)) >> 12; 2820 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 2821 rightVol = (uint16_t)v_clamped; 2822 } else { 2823 leftVol = mLeftVolShort; 2824 rightVol = mRightVolShort; 2825 rampVolume = false; 2826 } 2827 2828 // reset retry count 2829 track->mRetryCount = kMaxTrackRetriesDirect; 2830 mActiveTrack = t; 2831 mixerStatus = MIXER_TRACKS_READY; 2832 } else { 2833 //ALOGV("track %d u=%08x, s=%08x [NOT READY]", track->name(), cblk->user, cblk->server); 2834 if (track->isStopped()) { 2835 track->reset(); 2836 } 2837 if (track->isTerminated() || track->isStopped() || track->isPaused()) { 2838 // We have consumed all the buffers of this track. 2839 // Remove it from the list of active tracks. 2840 trackToRemove = track; 2841 } else { 2842 // No buffers for this track. Give it a few chances to 2843 // fill a buffer, then remove it from active list. 2844 if (--(track->mRetryCount) <= 0) { 2845 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 2846 trackToRemove = track; 2847 } else { 2848 mixerStatus = MIXER_TRACKS_ENABLED; 2849 } 2850 } 2851 } 2852 } 2853 2854 // FIXME merge this with similar code for removing multiple tracks 2855 // remove all the tracks that need to be... 2856 if (CC_UNLIKELY(trackToRemove != 0)) { 2857 tracksToRemove->add(trackToRemove); 2858 mActiveTracks.remove(trackToRemove); 2859 if (!mEffectChains.isEmpty()) { 2860 ALOGV("stopping track on chain %p for session Id: %d", mEffectChains[0].get(), 2861 trackToRemove->sessionId()); 2862 mEffectChains[0]->decActiveTrackCnt(); 2863 } 2864 if (trackToRemove->isTerminated()) { 2865 removeTrack_l(trackToRemove); 2866 } 2867 } 2868 2869 return mixerStatus; 2870} 2871 2872void AudioFlinger::DirectOutputThread::threadLoop_mix() 2873{ 2874 AudioBufferProvider::Buffer buffer; 2875 size_t frameCount = mFrameCount; 2876 int8_t *curBuf = (int8_t *)mMixBuffer; 2877 // output audio to hardware 2878 while (frameCount) { 2879 buffer.frameCount = frameCount; 2880 mActiveTrack->getNextBuffer(&buffer); 2881 if (CC_UNLIKELY(buffer.raw == NULL)) { 2882 memset(curBuf, 0, frameCount * mFrameSize); 2883 break; 2884 } 2885 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 2886 frameCount -= buffer.frameCount; 2887 curBuf += buffer.frameCount * mFrameSize; 2888 mActiveTrack->releaseBuffer(&buffer); 2889 } 2890 sleepTime = 0; 2891 standbyTime = systemTime() + standbyDelay; 2892 mActiveTrack.clear(); 2893 2894 // apply volume 2895 2896 // Do not apply volume on compressed audio 2897 if (!audio_is_linear_pcm(mFormat)) { 2898 return; 2899 } 2900 2901 // convert to signed 16 bit before volume calculation 2902 if (mFormat == AUDIO_FORMAT_PCM_8_BIT) { 2903 size_t count = mFrameCount * mChannelCount; 2904 uint8_t *src = (uint8_t *)mMixBuffer + count-1; 2905 int16_t *dst = mMixBuffer + count-1; 2906 while(count--) { 2907 *dst-- = (int16_t)(*src--^0x80) << 8; 2908 } 2909 } 2910 2911 frameCount = mFrameCount; 2912 int16_t *out = mMixBuffer; 2913 if (rampVolume) { 2914 if (mChannelCount == 1) { 2915 int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16; 2916 int32_t vlInc = d / (int32_t)frameCount; 2917 int32_t vl = ((int32_t)mLeftVolShort << 16); 2918 do { 2919 out[0] = clamp16(mul(out[0], vl >> 16) >> 12); 2920 out++; 2921 vl += vlInc; 2922 } while (--frameCount); 2923 2924 } else { 2925 int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16; 2926 int32_t vlInc = d / (int32_t)frameCount; 2927 d = ((int32_t)rightVol - (int32_t)mRightVolShort) << 16; 2928 int32_t vrInc = d / (int32_t)frameCount; 2929 int32_t vl = ((int32_t)mLeftVolShort << 16); 2930 int32_t vr = ((int32_t)mRightVolShort << 16); 2931 do { 2932 out[0] = clamp16(mul(out[0], vl >> 16) >> 12); 2933 out[1] = clamp16(mul(out[1], vr >> 16) >> 12); 2934 out += 2; 2935 vl += vlInc; 2936 vr += vrInc; 2937 } while (--frameCount); 2938 } 2939 } else { 2940 if (mChannelCount == 1) { 2941 do { 2942 out[0] = clamp16(mul(out[0], leftVol) >> 12); 2943 out++; 2944 } while (--frameCount); 2945 } else { 2946 do { 2947 out[0] = clamp16(mul(out[0], leftVol) >> 12); 2948 out[1] = clamp16(mul(out[1], rightVol) >> 12); 2949 out += 2; 2950 } while (--frameCount); 2951 } 2952 } 2953 2954 // convert back to unsigned 8 bit after volume calculation 2955 if (mFormat == AUDIO_FORMAT_PCM_8_BIT) { 2956 size_t count = mFrameCount * mChannelCount; 2957 int16_t *src = mMixBuffer; 2958 uint8_t *dst = (uint8_t *)mMixBuffer; 2959 while(count--) { 2960 *dst++ = (uint8_t)(((int32_t)*src++ + (1<<7)) >> 8)^0x80; 2961 } 2962 } 2963 2964 mLeftVolShort = leftVol; 2965 mRightVolShort = rightVol; 2966} 2967 2968void AudioFlinger::DirectOutputThread::threadLoop_sleepTime() 2969{ 2970 if (sleepTime == 0) { 2971 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 2972 sleepTime = activeSleepTime; 2973 } else { 2974 sleepTime = idleSleepTime; 2975 } 2976 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 2977 memset (mMixBuffer, 0, mFrameCount * mFrameSize); 2978 sleepTime = 0; 2979 } 2980} 2981 2982// getTrackName_l() must be called with ThreadBase::mLock held 2983int AudioFlinger::DirectOutputThread::getTrackName_l() 2984{ 2985 return 0; 2986} 2987 2988// deleteTrackName_l() must be called with ThreadBase::mLock held 2989void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name) 2990{ 2991} 2992 2993// checkForNewParameters_l() must be called with ThreadBase::mLock held 2994bool AudioFlinger::DirectOutputThread::checkForNewParameters_l() 2995{ 2996 bool reconfig = false; 2997 2998 while (!mNewParameters.isEmpty()) { 2999 status_t status = NO_ERROR; 3000 String8 keyValuePair = mNewParameters[0]; 3001 AudioParameter param = AudioParameter(keyValuePair); 3002 int value; 3003 3004 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3005 // do not accept frame count changes if tracks are open as the track buffer 3006 // size depends on frame count and correct behavior would not be garantied 3007 // if frame count is changed after track creation 3008 if (!mTracks.isEmpty()) { 3009 status = INVALID_OPERATION; 3010 } else { 3011 reconfig = true; 3012 } 3013 } 3014 if (status == NO_ERROR) { 3015 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3016 keyValuePair.string()); 3017 if (!mStandby && status == INVALID_OPERATION) { 3018 mOutput->stream->common.standby(&mOutput->stream->common); 3019 mStandby = true; 3020 mBytesWritten = 0; 3021 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3022 keyValuePair.string()); 3023 } 3024 if (status == NO_ERROR && reconfig) { 3025 readOutputParameters(); 3026 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3027 } 3028 } 3029 3030 mNewParameters.removeAt(0); 3031 3032 mParamStatus = status; 3033 mParamCond.signal(); 3034 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3035 // already timed out waiting for the status and will never signal the condition. 3036 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3037 } 3038 return reconfig; 3039} 3040 3041uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() 3042{ 3043 uint32_t time; 3044 if (audio_is_linear_pcm(mFormat)) { 3045 time = PlaybackThread::activeSleepTimeUs(); 3046 } else { 3047 time = 10000; 3048 } 3049 return time; 3050} 3051 3052uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() 3053{ 3054 uint32_t time; 3055 if (audio_is_linear_pcm(mFormat)) { 3056 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 3057 } else { 3058 time = 10000; 3059 } 3060 return time; 3061} 3062 3063uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() 3064{ 3065 uint32_t time; 3066 if (audio_is_linear_pcm(mFormat)) { 3067 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 3068 } else { 3069 time = 10000; 3070 } 3071 return time; 3072} 3073 3074void AudioFlinger::DirectOutputThread::cacheParameters_l() 3075{ 3076 PlaybackThread::cacheParameters_l(); 3077 3078 // use shorter standby delay as on normal output to release 3079 // hardware resources as soon as possible 3080 standbyDelay = microseconds(activeSleepTime*2); 3081} 3082 3083// ---------------------------------------------------------------------------- 3084 3085AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 3086 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id) 3087 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->device(), DUPLICATING), 3088 mWaitTimeMs(UINT_MAX) 3089{ 3090 addOutputTrack(mainThread); 3091} 3092 3093AudioFlinger::DuplicatingThread::~DuplicatingThread() 3094{ 3095 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3096 mOutputTracks[i]->destroy(); 3097 } 3098} 3099 3100void AudioFlinger::DuplicatingThread::threadLoop_mix() 3101{ 3102 // mix buffers... 3103 if (outputsReady(outputTracks)) { 3104 mAudioMixer->process(AudioBufferProvider::kInvalidPTS); 3105 } else { 3106 memset(mMixBuffer, 0, mixBufferSize); 3107 } 3108 sleepTime = 0; 3109 writeFrames = mFrameCount; 3110} 3111 3112void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() 3113{ 3114 if (sleepTime == 0) { 3115 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3116 sleepTime = activeSleepTime; 3117 } else { 3118 sleepTime = idleSleepTime; 3119 } 3120 } else if (mBytesWritten != 0) { 3121 // flush remaining overflow buffers in output tracks 3122 for (size_t i = 0; i < outputTracks.size(); i++) { 3123 if (outputTracks[i]->isActive()) { 3124 sleepTime = 0; 3125 writeFrames = 0; 3126 memset(mMixBuffer, 0, mixBufferSize); 3127 break; 3128 } 3129 } 3130 } 3131} 3132 3133void AudioFlinger::DuplicatingThread::threadLoop_write() 3134{ 3135 standbyTime = systemTime() + standbyDelay; 3136 for (size_t i = 0; i < outputTracks.size(); i++) { 3137 outputTracks[i]->write(mMixBuffer, writeFrames); 3138 } 3139 mBytesWritten += mixBufferSize; 3140} 3141 3142void AudioFlinger::DuplicatingThread::threadLoop_standby() 3143{ 3144 // DuplicatingThread implements standby by stopping all tracks 3145 for (size_t i = 0; i < outputTracks.size(); i++) { 3146 outputTracks[i]->stop(); 3147 } 3148} 3149 3150void AudioFlinger::DuplicatingThread::saveOutputTracks() 3151{ 3152 outputTracks = mOutputTracks; 3153} 3154 3155void AudioFlinger::DuplicatingThread::clearOutputTracks() 3156{ 3157 outputTracks.clear(); 3158} 3159 3160void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 3161{ 3162 Mutex::Autolock _l(mLock); 3163 // FIXME explain this formula 3164 int frameCount = (3 * mFrameCount * mSampleRate) / thread->sampleRate(); 3165 OutputTrack *outputTrack = new OutputTrack(thread, 3166 this, 3167 mSampleRate, 3168 mFormat, 3169 mChannelMask, 3170 frameCount); 3171 if (outputTrack->cblk() != NULL) { 3172 thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f); 3173 mOutputTracks.add(outputTrack); 3174 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 3175 updateWaitTime_l(); 3176 } 3177} 3178 3179void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 3180{ 3181 Mutex::Autolock _l(mLock); 3182 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3183 if (mOutputTracks[i]->thread() == thread) { 3184 mOutputTracks[i]->destroy(); 3185 mOutputTracks.removeAt(i); 3186 updateWaitTime_l(); 3187 return; 3188 } 3189 } 3190 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 3191} 3192 3193// caller must hold mLock 3194void AudioFlinger::DuplicatingThread::updateWaitTime_l() 3195{ 3196 mWaitTimeMs = UINT_MAX; 3197 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3198 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 3199 if (strong != 0) { 3200 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 3201 if (waitTimeMs < mWaitTimeMs) { 3202 mWaitTimeMs = waitTimeMs; 3203 } 3204 } 3205 } 3206} 3207 3208 3209bool AudioFlinger::DuplicatingThread::outputsReady(const SortedVector< sp<OutputTrack> > &outputTracks) 3210{ 3211 for (size_t i = 0; i < outputTracks.size(); i++) { 3212 sp <ThreadBase> thread = outputTracks[i]->thread().promote(); 3213 if (thread == 0) { 3214 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", outputTracks[i].get()); 3215 return false; 3216 } 3217 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3218 if (playbackThread->standby() && !playbackThread->isSuspended()) { 3219 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), thread.get()); 3220 return false; 3221 } 3222 } 3223 return true; 3224} 3225 3226uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() 3227{ 3228 return (mWaitTimeMs * 1000) / 2; 3229} 3230 3231void AudioFlinger::DuplicatingThread::cacheParameters_l() 3232{ 3233 // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first 3234 updateWaitTime_l(); 3235 3236 MixerThread::cacheParameters_l(); 3237} 3238 3239// ---------------------------------------------------------------------------- 3240 3241// TrackBase constructor must be called with AudioFlinger::mLock held 3242AudioFlinger::ThreadBase::TrackBase::TrackBase( 3243 ThreadBase *thread, 3244 const sp<Client>& client, 3245 uint32_t sampleRate, 3246 audio_format_t format, 3247 uint32_t channelMask, 3248 int frameCount, 3249 const sp<IMemory>& sharedBuffer, 3250 int sessionId) 3251 : RefBase(), 3252 mThread(thread), 3253 mClient(client), 3254 mCblk(NULL), 3255 // mBuffer 3256 // mBufferEnd 3257 mFrameCount(0), 3258 mState(IDLE), 3259 mFormat(format), 3260 mStepServerFailed(false), 3261 mSessionId(sessionId) 3262 // mChannelCount 3263 // mChannelMask 3264{ 3265 ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), sharedBuffer->size()); 3266 3267 // ALOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize); 3268 size_t size = sizeof(audio_track_cblk_t); 3269 uint8_t channelCount = popcount(channelMask); 3270 size_t bufferSize = frameCount*channelCount*sizeof(int16_t); 3271 if (sharedBuffer == 0) { 3272 size += bufferSize; 3273 } 3274 3275 if (client != NULL) { 3276 mCblkMemory = client->heap()->allocate(size); 3277 if (mCblkMemory != 0) { 3278 mCblk = static_cast<audio_track_cblk_t *>(mCblkMemory->pointer()); 3279 if (mCblk != NULL) { // construct the shared structure in-place. 3280 new(mCblk) audio_track_cblk_t(); 3281 // clear all buffers 3282 mCblk->frameCount = frameCount; 3283 mCblk->sampleRate = sampleRate; 3284 mChannelCount = channelCount; 3285 mChannelMask = channelMask; 3286 if (sharedBuffer == 0) { 3287 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 3288 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 3289 // Force underrun condition to avoid false underrun callback until first data is 3290 // written to buffer (other flags are cleared) 3291 mCblk->flags = CBLK_UNDERRUN_ON; 3292 } else { 3293 mBuffer = sharedBuffer->pointer(); 3294 } 3295 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 3296 } 3297 } else { 3298 ALOGE("not enough memory for AudioTrack size=%u", size); 3299 client->heap()->dump("AudioTrack"); 3300 return; 3301 } 3302 } else { 3303 mCblk = (audio_track_cblk_t *)(new uint8_t[size]); 3304 // construct the shared structure in-place. 3305 new(mCblk) audio_track_cblk_t(); 3306 // clear all buffers 3307 mCblk->frameCount = frameCount; 3308 mCblk->sampleRate = sampleRate; 3309 mChannelCount = channelCount; 3310 mChannelMask = channelMask; 3311 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 3312 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 3313 // Force underrun condition to avoid false underrun callback until first data is 3314 // written to buffer (other flags are cleared) 3315 mCblk->flags = CBLK_UNDERRUN_ON; 3316 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 3317 } 3318} 3319 3320AudioFlinger::ThreadBase::TrackBase::~TrackBase() 3321{ 3322 if (mCblk != NULL) { 3323 if (mClient == 0) { 3324 delete mCblk; 3325 } else { 3326 mCblk->~audio_track_cblk_t(); // destroy our shared-structure. 3327 } 3328 } 3329 mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to 3330 if (mClient != 0) { 3331 // Client destructor must run with AudioFlinger mutex locked 3332 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 3333 // If the client's reference count drops to zero, the associated destructor 3334 // must run with AudioFlinger lock held. Thus the explicit clear() rather than 3335 // relying on the automatic clear() at end of scope. 3336 mClient.clear(); 3337 } 3338} 3339 3340// AudioBufferProvider interface 3341// getNextBuffer() = 0; 3342// This implementation of releaseBuffer() is used by Track and RecordTrack, but not TimedTrack 3343void AudioFlinger::ThreadBase::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer) 3344{ 3345 buffer->raw = NULL; 3346 mFrameCount = buffer->frameCount; 3347 (void) step(); // ignore return value of step() 3348 buffer->frameCount = 0; 3349} 3350 3351bool AudioFlinger::ThreadBase::TrackBase::step() { 3352 bool result; 3353 audio_track_cblk_t* cblk = this->cblk(); 3354 3355 result = cblk->stepServer(mFrameCount); 3356 if (!result) { 3357 ALOGV("stepServer failed acquiring cblk mutex"); 3358 mStepServerFailed = true; 3359 } 3360 return result; 3361} 3362 3363void AudioFlinger::ThreadBase::TrackBase::reset() { 3364 audio_track_cblk_t* cblk = this->cblk(); 3365 3366 cblk->user = 0; 3367 cblk->server = 0; 3368 cblk->userBase = 0; 3369 cblk->serverBase = 0; 3370 mStepServerFailed = false; 3371 ALOGV("TrackBase::reset"); 3372} 3373 3374int AudioFlinger::ThreadBase::TrackBase::sampleRate() const { 3375 return (int)mCblk->sampleRate; 3376} 3377 3378void* AudioFlinger::ThreadBase::TrackBase::getBuffer(uint32_t offset, uint32_t frames) const { 3379 audio_track_cblk_t* cblk = this->cblk(); 3380 size_t frameSize = cblk->frameSize; 3381 int8_t *bufferStart = (int8_t *)mBuffer + (offset-cblk->serverBase)*frameSize; 3382 int8_t *bufferEnd = bufferStart + frames * frameSize; 3383 3384 // Check validity of returned pointer in case the track control block would have been corrupted. 3385 if (bufferStart < mBuffer || bufferStart > bufferEnd || bufferEnd > mBufferEnd || 3386 ((unsigned long)bufferStart & (unsigned long)(frameSize - 1))) { 3387 ALOGE("TrackBase::getBuffer buffer out of range:\n start: %p, end %p , mBuffer %p mBufferEnd %p\n \ 3388 server %d, serverBase %d, user %d, userBase %d", 3389 bufferStart, bufferEnd, mBuffer, mBufferEnd, 3390 cblk->server, cblk->serverBase, cblk->user, cblk->userBase); 3391 return NULL; 3392 } 3393 3394 return bufferStart; 3395} 3396 3397// ---------------------------------------------------------------------------- 3398 3399// Track constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held 3400AudioFlinger::PlaybackThread::Track::Track( 3401 PlaybackThread *thread, 3402 const sp<Client>& client, 3403 audio_stream_type_t streamType, 3404 uint32_t sampleRate, 3405 audio_format_t format, 3406 uint32_t channelMask, 3407 int frameCount, 3408 const sp<IMemory>& sharedBuffer, 3409 int sessionId) 3410 : TrackBase(thread, client, sampleRate, format, channelMask, frameCount, sharedBuffer, sessionId), 3411 mMute(false), mSharedBuffer(sharedBuffer), mName(-1), mMainBuffer(NULL), mAuxBuffer(NULL), 3412 mAuxEffectId(0), mHasVolumeController(false) 3413{ 3414 if (mCblk != NULL) { 3415 if (thread != NULL) { 3416 mName = thread->getTrackName_l(); 3417 mMainBuffer = thread->mixBuffer(); 3418 } 3419 ALOGV("Track constructor name %d, calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 3420 if (mName < 0) { 3421 ALOGE("no more track names available"); 3422 } 3423 mStreamType = streamType; 3424 // NOTE: audio_track_cblk_t::frameSize for 8 bit PCM data is based on a sample size of 3425 // 16 bit because data is converted to 16 bit before being stored in buffer by AudioTrack 3426 mCblk->frameSize = audio_is_linear_pcm(format) ? mChannelCount * sizeof(int16_t) : sizeof(uint8_t); 3427 } 3428} 3429 3430AudioFlinger::PlaybackThread::Track::~Track() 3431{ 3432 ALOGV("PlaybackThread::Track destructor"); 3433 sp<ThreadBase> thread = mThread.promote(); 3434 if (thread != 0) { 3435 Mutex::Autolock _l(thread->mLock); 3436 mState = TERMINATED; 3437 } 3438} 3439 3440void AudioFlinger::PlaybackThread::Track::destroy() 3441{ 3442 // NOTE: destroyTrack_l() can remove a strong reference to this Track 3443 // by removing it from mTracks vector, so there is a risk that this Tracks's 3444 // destructor is called. As the destructor needs to lock mLock, 3445 // we must acquire a strong reference on this Track before locking mLock 3446 // here so that the destructor is called only when exiting this function. 3447 // On the other hand, as long as Track::destroy() is only called by 3448 // TrackHandle destructor, the TrackHandle still holds a strong ref on 3449 // this Track with its member mTrack. 3450 sp<Track> keep(this); 3451 { // scope for mLock 3452 sp<ThreadBase> thread = mThread.promote(); 3453 if (thread != 0) { 3454 if (!isOutputTrack()) { 3455 if (mState == ACTIVE || mState == RESUMING) { 3456 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 3457 3458 // to track the speaker usage 3459 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 3460 } 3461 AudioSystem::releaseOutput(thread->id()); 3462 } 3463 Mutex::Autolock _l(thread->mLock); 3464 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3465 playbackThread->destroyTrack_l(this); 3466 } 3467 } 3468} 3469 3470void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size) 3471{ 3472 uint32_t vlr = mCblk->getVolumeLR(); 3473 snprintf(buffer, size, " %05d %05d %03u %03u 0x%08x %05u %04u %1d %1d %1d %05u %05u %05u 0x%08x 0x%08x 0x%08x 0x%08x\n", 3474 mName - AudioMixer::TRACK0, 3475 (mClient == 0) ? getpid_cached : mClient->pid(), 3476 mStreamType, 3477 mFormat, 3478 mChannelMask, 3479 mSessionId, 3480 mFrameCount, 3481 mState, 3482 mMute, 3483 mFillingUpStatus, 3484 mCblk->sampleRate, 3485 vlr & 0xFFFF, 3486 vlr >> 16, 3487 mCblk->server, 3488 mCblk->user, 3489 (int)mMainBuffer, 3490 (int)mAuxBuffer); 3491} 3492 3493// AudioBufferProvider interface 3494status_t AudioFlinger::PlaybackThread::Track::getNextBuffer( 3495 AudioBufferProvider::Buffer* buffer, int64_t pts) 3496{ 3497 audio_track_cblk_t* cblk = this->cblk(); 3498 uint32_t framesReady; 3499 uint32_t framesReq = buffer->frameCount; 3500 3501 // Check if last stepServer failed, try to step now 3502 if (mStepServerFailed) { 3503 if (!step()) goto getNextBuffer_exit; 3504 ALOGV("stepServer recovered"); 3505 mStepServerFailed = false; 3506 } 3507 3508 framesReady = cblk->framesReady(); 3509 3510 if (CC_LIKELY(framesReady)) { 3511 uint32_t s = cblk->server; 3512 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 3513 3514 bufferEnd = (cblk->loopEnd < bufferEnd) ? cblk->loopEnd : bufferEnd; 3515 if (framesReq > framesReady) { 3516 framesReq = framesReady; 3517 } 3518 if (s + framesReq > bufferEnd) { 3519 framesReq = bufferEnd - s; 3520 } 3521 3522 buffer->raw = getBuffer(s, framesReq); 3523 if (buffer->raw == NULL) goto getNextBuffer_exit; 3524 3525 buffer->frameCount = framesReq; 3526 return NO_ERROR; 3527 } 3528 3529getNextBuffer_exit: 3530 buffer->raw = NULL; 3531 buffer->frameCount = 0; 3532 ALOGV("getNextBuffer() no more data for track %d on thread %p", mName, mThread.unsafe_get()); 3533 return NOT_ENOUGH_DATA; 3534} 3535 3536uint32_t AudioFlinger::PlaybackThread::Track::framesReady() const{ 3537 return mCblk->framesReady(); 3538} 3539 3540bool AudioFlinger::PlaybackThread::Track::isReady() const { 3541 if (mFillingUpStatus != FS_FILLING || isStopped() || isPausing()) return true; 3542 3543 if (framesReady() >= mCblk->frameCount || 3544 (mCblk->flags & CBLK_FORCEREADY_MSK)) { 3545 mFillingUpStatus = FS_FILLED; 3546 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 3547 return true; 3548 } 3549 return false; 3550} 3551 3552status_t AudioFlinger::PlaybackThread::Track::start(pid_t tid) 3553{ 3554 status_t status = NO_ERROR; 3555 ALOGV("start(%d), calling pid %d session %d tid %d", 3556 mName, IPCThreadState::self()->getCallingPid(), mSessionId, tid); 3557 sp<ThreadBase> thread = mThread.promote(); 3558 if (thread != 0) { 3559 Mutex::Autolock _l(thread->mLock); 3560 track_state state = mState; 3561 // here the track could be either new, or restarted 3562 // in both cases "unstop" the track 3563 if (mState == PAUSED) { 3564 mState = TrackBase::RESUMING; 3565 ALOGV("PAUSED => RESUMING (%d) on thread %p", mName, this); 3566 } else { 3567 mState = TrackBase::ACTIVE; 3568 ALOGV("? => ACTIVE (%d) on thread %p", mName, this); 3569 } 3570 3571 if (!isOutputTrack() && state != ACTIVE && state != RESUMING) { 3572 thread->mLock.unlock(); 3573 status = AudioSystem::startOutput(thread->id(), mStreamType, mSessionId); 3574 thread->mLock.lock(); 3575 3576 // to track the speaker usage 3577 if (status == NO_ERROR) { 3578 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 3579 } 3580 } 3581 if (status == NO_ERROR) { 3582 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3583 playbackThread->addTrack_l(this); 3584 } else { 3585 mState = state; 3586 } 3587 } else { 3588 status = BAD_VALUE; 3589 } 3590 return status; 3591} 3592 3593void AudioFlinger::PlaybackThread::Track::stop() 3594{ 3595 ALOGV("stop(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 3596 sp<ThreadBase> thread = mThread.promote(); 3597 if (thread != 0) { 3598 Mutex::Autolock _l(thread->mLock); 3599 track_state state = mState; 3600 if (mState > STOPPED) { 3601 mState = STOPPED; 3602 // If the track is not active (PAUSED and buffers full), flush buffers 3603 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3604 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 3605 reset(); 3606 } 3607 ALOGV("(> STOPPED) => STOPPED (%d) on thread %p", mName, playbackThread); 3608 } 3609 if (!isOutputTrack() && (state == ACTIVE || state == RESUMING)) { 3610 thread->mLock.unlock(); 3611 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 3612 thread->mLock.lock(); 3613 3614 // to track the speaker usage 3615 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 3616 } 3617 } 3618} 3619 3620void AudioFlinger::PlaybackThread::Track::pause() 3621{ 3622 ALOGV("pause(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 3623 sp<ThreadBase> thread = mThread.promote(); 3624 if (thread != 0) { 3625 Mutex::Autolock _l(thread->mLock); 3626 if (mState == ACTIVE || mState == RESUMING) { 3627 mState = PAUSING; 3628 ALOGV("ACTIVE/RESUMING => PAUSING (%d) on thread %p", mName, thread.get()); 3629 if (!isOutputTrack()) { 3630 thread->mLock.unlock(); 3631 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 3632 thread->mLock.lock(); 3633 3634 // to track the speaker usage 3635 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 3636 } 3637 } 3638 } 3639} 3640 3641void AudioFlinger::PlaybackThread::Track::flush() 3642{ 3643 ALOGV("flush(%d)", mName); 3644 sp<ThreadBase> thread = mThread.promote(); 3645 if (thread != 0) { 3646 Mutex::Autolock _l(thread->mLock); 3647 if (mState != STOPPED && mState != PAUSED && mState != PAUSING) { 3648 return; 3649 } 3650 // No point remaining in PAUSED state after a flush => go to 3651 // STOPPED state 3652 mState = STOPPED; 3653 3654 // do not reset the track if it is still in the process of being stopped or paused. 3655 // this will be done by prepareTracks_l() when the track is stopped. 3656 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3657 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 3658 reset(); 3659 } 3660 } 3661} 3662 3663void AudioFlinger::PlaybackThread::Track::reset() 3664{ 3665 // Do not reset twice to avoid discarding data written just after a flush and before 3666 // the audioflinger thread detects the track is stopped. 3667 if (!mResetDone) { 3668 TrackBase::reset(); 3669 // Force underrun condition to avoid false underrun callback until first data is 3670 // written to buffer 3671 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 3672 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 3673 mFillingUpStatus = FS_FILLING; 3674 mResetDone = true; 3675 } 3676} 3677 3678void AudioFlinger::PlaybackThread::Track::mute(bool muted) 3679{ 3680 mMute = muted; 3681} 3682 3683status_t AudioFlinger::PlaybackThread::Track::attachAuxEffect(int EffectId) 3684{ 3685 status_t status = DEAD_OBJECT; 3686 sp<ThreadBase> thread = mThread.promote(); 3687 if (thread != 0) { 3688 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3689 status = playbackThread->attachAuxEffect(this, EffectId); 3690 } 3691 return status; 3692} 3693 3694void AudioFlinger::PlaybackThread::Track::setAuxBuffer(int EffectId, int32_t *buffer) 3695{ 3696 mAuxEffectId = EffectId; 3697 mAuxBuffer = buffer; 3698} 3699 3700// timed audio tracks 3701 3702sp<AudioFlinger::PlaybackThread::TimedTrack> 3703AudioFlinger::PlaybackThread::TimedTrack::create( 3704 PlaybackThread *thread, 3705 const sp<Client>& client, 3706 audio_stream_type_t streamType, 3707 uint32_t sampleRate, 3708 audio_format_t format, 3709 uint32_t channelMask, 3710 int frameCount, 3711 const sp<IMemory>& sharedBuffer, 3712 int sessionId) { 3713 if (!client->reserveTimedTrack()) 3714 return NULL; 3715 3716 sp<TimedTrack> track = new TimedTrack( 3717 thread, client, streamType, sampleRate, format, channelMask, frameCount, 3718 sharedBuffer, sessionId); 3719 3720 if (track == NULL) { 3721 client->releaseTimedTrack(); 3722 return NULL; 3723 } 3724 3725 return track; 3726} 3727 3728AudioFlinger::PlaybackThread::TimedTrack::TimedTrack( 3729 PlaybackThread *thread, 3730 const sp<Client>& client, 3731 audio_stream_type_t streamType, 3732 uint32_t sampleRate, 3733 audio_format_t format, 3734 uint32_t channelMask, 3735 int frameCount, 3736 const sp<IMemory>& sharedBuffer, 3737 int sessionId) 3738 : Track(thread, client, streamType, sampleRate, format, channelMask, 3739 frameCount, sharedBuffer, sessionId), 3740 mTimedSilenceBuffer(NULL), 3741 mTimedSilenceBufferSize(0), 3742 mTimedAudioOutputOnTime(false), 3743 mMediaTimeTransformValid(false) 3744{ 3745 LocalClock lc; 3746 mLocalTimeFreq = lc.getLocalFreq(); 3747 3748 mLocalTimeToSampleTransform.a_zero = 0; 3749 mLocalTimeToSampleTransform.b_zero = 0; 3750 mLocalTimeToSampleTransform.a_to_b_numer = sampleRate; 3751 mLocalTimeToSampleTransform.a_to_b_denom = mLocalTimeFreq; 3752 LinearTransform::reduce(&mLocalTimeToSampleTransform.a_to_b_numer, 3753 &mLocalTimeToSampleTransform.a_to_b_denom); 3754} 3755 3756AudioFlinger::PlaybackThread::TimedTrack::~TimedTrack() { 3757 mClient->releaseTimedTrack(); 3758 delete [] mTimedSilenceBuffer; 3759} 3760 3761status_t AudioFlinger::PlaybackThread::TimedTrack::allocateTimedBuffer( 3762 size_t size, sp<IMemory>* buffer) { 3763 3764 Mutex::Autolock _l(mTimedBufferQueueLock); 3765 3766 trimTimedBufferQueue_l(); 3767 3768 // lazily initialize the shared memory heap for timed buffers 3769 if (mTimedMemoryDealer == NULL) { 3770 const int kTimedBufferHeapSize = 512 << 10; 3771 3772 mTimedMemoryDealer = new MemoryDealer(kTimedBufferHeapSize, 3773 "AudioFlingerTimed"); 3774 if (mTimedMemoryDealer == NULL) 3775 return NO_MEMORY; 3776 } 3777 3778 sp<IMemory> newBuffer = mTimedMemoryDealer->allocate(size); 3779 if (newBuffer == NULL) { 3780 newBuffer = mTimedMemoryDealer->allocate(size); 3781 if (newBuffer == NULL) 3782 return NO_MEMORY; 3783 } 3784 3785 *buffer = newBuffer; 3786 return NO_ERROR; 3787} 3788 3789// caller must hold mTimedBufferQueueLock 3790void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueue_l() { 3791 int64_t mediaTimeNow; 3792 { 3793 Mutex::Autolock mttLock(mMediaTimeTransformLock); 3794 if (!mMediaTimeTransformValid) 3795 return; 3796 3797 int64_t targetTimeNow; 3798 status_t res = (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) 3799 ? mCCHelper.getCommonTime(&targetTimeNow) 3800 : mCCHelper.getLocalTime(&targetTimeNow); 3801 3802 if (OK != res) 3803 return; 3804 3805 if (!mMediaTimeTransform.doReverseTransform(targetTimeNow, 3806 &mediaTimeNow)) { 3807 return; 3808 } 3809 } 3810 3811 size_t trimIndex; 3812 for (trimIndex = 0; trimIndex < mTimedBufferQueue.size(); trimIndex++) { 3813 if (mTimedBufferQueue[trimIndex].pts() > mediaTimeNow) 3814 break; 3815 } 3816 3817 if (trimIndex) { 3818 mTimedBufferQueue.removeItemsAt(0, trimIndex); 3819 } 3820} 3821 3822status_t AudioFlinger::PlaybackThread::TimedTrack::queueTimedBuffer( 3823 const sp<IMemory>& buffer, int64_t pts) { 3824 3825 { 3826 Mutex::Autolock mttLock(mMediaTimeTransformLock); 3827 if (!mMediaTimeTransformValid) 3828 return INVALID_OPERATION; 3829 } 3830 3831 Mutex::Autolock _l(mTimedBufferQueueLock); 3832 3833 mTimedBufferQueue.add(TimedBuffer(buffer, pts)); 3834 3835 return NO_ERROR; 3836} 3837 3838status_t AudioFlinger::PlaybackThread::TimedTrack::setMediaTimeTransform( 3839 const LinearTransform& xform, TimedAudioTrack::TargetTimeline target) { 3840 3841 ALOGV("%s az=%lld bz=%lld n=%d d=%u tgt=%d", __PRETTY_FUNCTION__, 3842 xform.a_zero, xform.b_zero, xform.a_to_b_numer, xform.a_to_b_denom, 3843 target); 3844 3845 if (!(target == TimedAudioTrack::LOCAL_TIME || 3846 target == TimedAudioTrack::COMMON_TIME)) { 3847 return BAD_VALUE; 3848 } 3849 3850 Mutex::Autolock lock(mMediaTimeTransformLock); 3851 mMediaTimeTransform = xform; 3852 mMediaTimeTransformTarget = target; 3853 mMediaTimeTransformValid = true; 3854 3855 return NO_ERROR; 3856} 3857 3858#define min(a, b) ((a) < (b) ? (a) : (b)) 3859 3860// implementation of getNextBuffer for tracks whose buffers have timestamps 3861status_t AudioFlinger::PlaybackThread::TimedTrack::getNextBuffer( 3862 AudioBufferProvider::Buffer* buffer, int64_t pts) 3863{ 3864 if (pts == AudioBufferProvider::kInvalidPTS) { 3865 buffer->raw = 0; 3866 buffer->frameCount = 0; 3867 return INVALID_OPERATION; 3868 } 3869 3870 Mutex::Autolock _l(mTimedBufferQueueLock); 3871 3872 while (true) { 3873 3874 // if we have no timed buffers, then fail 3875 if (mTimedBufferQueue.isEmpty()) { 3876 buffer->raw = 0; 3877 buffer->frameCount = 0; 3878 return NOT_ENOUGH_DATA; 3879 } 3880 3881 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 3882 3883 // calculate the PTS of the head of the timed buffer queue expressed in 3884 // local time 3885 int64_t headLocalPTS; 3886 { 3887 Mutex::Autolock mttLock(mMediaTimeTransformLock); 3888 3889 ALOG_ASSERT(mMediaTimeTransformValid, "media time transform invalid"); 3890 3891 if (mMediaTimeTransform.a_to_b_denom == 0) { 3892 // the transform represents a pause, so yield silence 3893 timedYieldSilence(buffer->frameCount, buffer); 3894 return NO_ERROR; 3895 } 3896 3897 int64_t transformedPTS; 3898 if (!mMediaTimeTransform.doForwardTransform(head.pts(), 3899 &transformedPTS)) { 3900 // the transform failed. this shouldn't happen, but if it does 3901 // then just drop this buffer 3902 ALOGW("timedGetNextBuffer transform failed"); 3903 buffer->raw = 0; 3904 buffer->frameCount = 0; 3905 mTimedBufferQueue.removeAt(0); 3906 return NO_ERROR; 3907 } 3908 3909 if (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) { 3910 if (OK != mCCHelper.commonTimeToLocalTime(transformedPTS, 3911 &headLocalPTS)) { 3912 buffer->raw = 0; 3913 buffer->frameCount = 0; 3914 return INVALID_OPERATION; 3915 } 3916 } else { 3917 headLocalPTS = transformedPTS; 3918 } 3919 } 3920 3921 // adjust the head buffer's PTS to reflect the portion of the head buffer 3922 // that has already been consumed 3923 int64_t effectivePTS = headLocalPTS + 3924 ((head.position() / mCblk->frameSize) * mLocalTimeFreq / sampleRate()); 3925 3926 // Calculate the delta in samples between the head of the input buffer 3927 // queue and the start of the next output buffer that will be written. 3928 // If the transformation fails because of over or underflow, it means 3929 // that the sample's position in the output stream is so far out of 3930 // whack that it should just be dropped. 3931 int64_t sampleDelta; 3932 if (llabs(effectivePTS - pts) >= (static_cast<int64_t>(1) << 31)) { 3933 ALOGV("*** head buffer is too far from PTS: dropped buffer"); 3934 mTimedBufferQueue.removeAt(0); 3935 continue; 3936 } 3937 if (!mLocalTimeToSampleTransform.doForwardTransform( 3938 (effectivePTS - pts) << 32, &sampleDelta)) { 3939 ALOGV("*** too late during sample rate transform: dropped buffer"); 3940 mTimedBufferQueue.removeAt(0); 3941 continue; 3942 } 3943 3944 ALOGV("*** %s head.pts=%lld head.pos=%d pts=%lld sampleDelta=[%d.%08x]", 3945 __PRETTY_FUNCTION__, head.pts(), head.position(), pts, 3946 static_cast<int32_t>((sampleDelta >= 0 ? 0 : 1) + (sampleDelta >> 32)), 3947 static_cast<uint32_t>(sampleDelta & 0xFFFFFFFF)); 3948 3949 // if the delta between the ideal placement for the next input sample and 3950 // the current output position is within this threshold, then we will 3951 // concatenate the next input samples to the previous output 3952 const int64_t kSampleContinuityThreshold = 3953 (static_cast<int64_t>(sampleRate()) << 32) / 10; 3954 3955 // if this is the first buffer of audio that we're emitting from this track 3956 // then it should be almost exactly on time. 3957 const int64_t kSampleStartupThreshold = 1LL << 32; 3958 3959 if ((mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleContinuityThreshold) || 3960 (!mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleStartupThreshold)) { 3961 // the next input is close enough to being on time, so concatenate it 3962 // with the last output 3963 timedYieldSamples(buffer); 3964 3965 ALOGV("*** on time: head.pos=%d frameCount=%u", head.position(), buffer->frameCount); 3966 return NO_ERROR; 3967 } else if (sampleDelta > 0) { 3968 // the gap between the current output position and the proper start of 3969 // the next input sample is too big, so fill it with silence 3970 uint32_t framesUntilNextInput = (sampleDelta + 0x80000000) >> 32; 3971 3972 timedYieldSilence(framesUntilNextInput, buffer); 3973 ALOGV("*** silence: frameCount=%u", buffer->frameCount); 3974 return NO_ERROR; 3975 } else { 3976 // the next input sample is late 3977 uint32_t lateFrames = static_cast<uint32_t>(-((sampleDelta + 0x80000000) >> 32)); 3978 size_t onTimeSamplePosition = 3979 head.position() + lateFrames * mCblk->frameSize; 3980 3981 if (onTimeSamplePosition > head.buffer()->size()) { 3982 // all the remaining samples in the head are too late, so 3983 // drop it and move on 3984 ALOGV("*** too late: dropped buffer"); 3985 mTimedBufferQueue.removeAt(0); 3986 continue; 3987 } else { 3988 // skip over the late samples 3989 head.setPosition(onTimeSamplePosition); 3990 3991 // yield the available samples 3992 timedYieldSamples(buffer); 3993 3994 ALOGV("*** late: head.pos=%d frameCount=%u", head.position(), buffer->frameCount); 3995 return NO_ERROR; 3996 } 3997 } 3998 } 3999} 4000 4001// Yield samples from the timed buffer queue head up to the given output 4002// buffer's capacity. 4003// 4004// Caller must hold mTimedBufferQueueLock 4005void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSamples( 4006 AudioBufferProvider::Buffer* buffer) { 4007 4008 const TimedBuffer& head = mTimedBufferQueue[0]; 4009 4010 buffer->raw = (static_cast<uint8_t*>(head.buffer()->pointer()) + 4011 head.position()); 4012 4013 uint32_t framesLeftInHead = ((head.buffer()->size() - head.position()) / 4014 mCblk->frameSize); 4015 size_t framesRequested = buffer->frameCount; 4016 buffer->frameCount = min(framesLeftInHead, framesRequested); 4017 4018 mTimedAudioOutputOnTime = true; 4019} 4020 4021// Yield samples of silence up to the given output buffer's capacity 4022// 4023// Caller must hold mTimedBufferQueueLock 4024void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSilence( 4025 uint32_t numFrames, AudioBufferProvider::Buffer* buffer) { 4026 4027 // lazily allocate a buffer filled with silence 4028 if (mTimedSilenceBufferSize < numFrames * mCblk->frameSize) { 4029 delete [] mTimedSilenceBuffer; 4030 mTimedSilenceBufferSize = numFrames * mCblk->frameSize; 4031 mTimedSilenceBuffer = new uint8_t[mTimedSilenceBufferSize]; 4032 memset(mTimedSilenceBuffer, 0, mTimedSilenceBufferSize); 4033 } 4034 4035 buffer->raw = mTimedSilenceBuffer; 4036 size_t framesRequested = buffer->frameCount; 4037 buffer->frameCount = min(numFrames, framesRequested); 4038 4039 mTimedAudioOutputOnTime = false; 4040} 4041 4042// AudioBufferProvider interface 4043void AudioFlinger::PlaybackThread::TimedTrack::releaseBuffer( 4044 AudioBufferProvider::Buffer* buffer) { 4045 4046 Mutex::Autolock _l(mTimedBufferQueueLock); 4047 4048 // If the buffer which was just released is part of the buffer at the head 4049 // of the queue, be sure to update the amt of the buffer which has been 4050 // consumed. If the buffer being returned is not part of the head of the 4051 // queue, its either because the buffer is part of the silence buffer, or 4052 // because the head of the timed queue was trimmed after the mixer called 4053 // getNextBuffer but before the mixer called releaseBuffer. 4054 if ((buffer->raw != mTimedSilenceBuffer) && mTimedBufferQueue.size()) { 4055 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 4056 4057 void* start = head.buffer()->pointer(); 4058 void* end = (char *) head.buffer()->pointer() + head.buffer()->size(); 4059 4060 if ((buffer->raw >= start) && (buffer->raw <= end)) { 4061 head.setPosition(head.position() + 4062 (buffer->frameCount * mCblk->frameSize)); 4063 if (static_cast<size_t>(head.position()) >= head.buffer()->size()) { 4064 mTimedBufferQueue.removeAt(0); 4065 } 4066 } 4067 } 4068 4069 buffer->raw = 0; 4070 buffer->frameCount = 0; 4071} 4072 4073uint32_t AudioFlinger::PlaybackThread::TimedTrack::framesReady() const { 4074 Mutex::Autolock _l(mTimedBufferQueueLock); 4075 4076 uint32_t frames = 0; 4077 for (size_t i = 0; i < mTimedBufferQueue.size(); i++) { 4078 const TimedBuffer& tb = mTimedBufferQueue[i]; 4079 frames += (tb.buffer()->size() - tb.position()) / mCblk->frameSize; 4080 } 4081 4082 return frames; 4083} 4084 4085AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer() 4086 : mPTS(0), mPosition(0) {} 4087 4088AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer( 4089 const sp<IMemory>& buffer, int64_t pts) 4090 : mBuffer(buffer), mPTS(pts), mPosition(0) {} 4091 4092// ---------------------------------------------------------------------------- 4093 4094// RecordTrack constructor must be called with AudioFlinger::mLock held 4095AudioFlinger::RecordThread::RecordTrack::RecordTrack( 4096 RecordThread *thread, 4097 const sp<Client>& client, 4098 uint32_t sampleRate, 4099 audio_format_t format, 4100 uint32_t channelMask, 4101 int frameCount, 4102 int sessionId) 4103 : TrackBase(thread, client, sampleRate, format, 4104 channelMask, frameCount, 0 /*sharedBuffer*/, sessionId), 4105 mOverflow(false) 4106{ 4107 if (mCblk != NULL) { 4108 ALOGV("RecordTrack constructor, size %d", (int)mBufferEnd - (int)mBuffer); 4109 if (format == AUDIO_FORMAT_PCM_16_BIT) { 4110 mCblk->frameSize = mChannelCount * sizeof(int16_t); 4111 } else if (format == AUDIO_FORMAT_PCM_8_BIT) { 4112 mCblk->frameSize = mChannelCount * sizeof(int8_t); 4113 } else { 4114 mCblk->frameSize = sizeof(int8_t); 4115 } 4116 } 4117} 4118 4119AudioFlinger::RecordThread::RecordTrack::~RecordTrack() 4120{ 4121 sp<ThreadBase> thread = mThread.promote(); 4122 if (thread != 0) { 4123 AudioSystem::releaseInput(thread->id()); 4124 } 4125} 4126 4127// AudioBufferProvider interface 4128status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 4129{ 4130 audio_track_cblk_t* cblk = this->cblk(); 4131 uint32_t framesAvail; 4132 uint32_t framesReq = buffer->frameCount; 4133 4134 // Check if last stepServer failed, try to step now 4135 if (mStepServerFailed) { 4136 if (!step()) goto getNextBuffer_exit; 4137 ALOGV("stepServer recovered"); 4138 mStepServerFailed = false; 4139 } 4140 4141 framesAvail = cblk->framesAvailable_l(); 4142 4143 if (CC_LIKELY(framesAvail)) { 4144 uint32_t s = cblk->server; 4145 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 4146 4147 if (framesReq > framesAvail) { 4148 framesReq = framesAvail; 4149 } 4150 if (s + framesReq > bufferEnd) { 4151 framesReq = bufferEnd - s; 4152 } 4153 4154 buffer->raw = getBuffer(s, framesReq); 4155 if (buffer->raw == NULL) goto getNextBuffer_exit; 4156 4157 buffer->frameCount = framesReq; 4158 return NO_ERROR; 4159 } 4160 4161getNextBuffer_exit: 4162 buffer->raw = NULL; 4163 buffer->frameCount = 0; 4164 return NOT_ENOUGH_DATA; 4165} 4166 4167status_t AudioFlinger::RecordThread::RecordTrack::start(pid_t tid) 4168{ 4169 sp<ThreadBase> thread = mThread.promote(); 4170 if (thread != 0) { 4171 RecordThread *recordThread = (RecordThread *)thread.get(); 4172 return recordThread->start(this, tid); 4173 } else { 4174 return BAD_VALUE; 4175 } 4176} 4177 4178void AudioFlinger::RecordThread::RecordTrack::stop() 4179{ 4180 sp<ThreadBase> thread = mThread.promote(); 4181 if (thread != 0) { 4182 RecordThread *recordThread = (RecordThread *)thread.get(); 4183 recordThread->stop(this); 4184 TrackBase::reset(); 4185 // Force overerrun condition to avoid false overrun callback until first data is 4186 // read from buffer 4187 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 4188 } 4189} 4190 4191void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size) 4192{ 4193 snprintf(buffer, size, " %05d %03u 0x%08x %05d %04u %01d %05u %08x %08x\n", 4194 (mClient == 0) ? getpid_cached : mClient->pid(), 4195 mFormat, 4196 mChannelMask, 4197 mSessionId, 4198 mFrameCount, 4199 mState, 4200 mCblk->sampleRate, 4201 mCblk->server, 4202 mCblk->user); 4203} 4204 4205 4206// ---------------------------------------------------------------------------- 4207 4208AudioFlinger::PlaybackThread::OutputTrack::OutputTrack( 4209 PlaybackThread *playbackThread, 4210 DuplicatingThread *sourceThread, 4211 uint32_t sampleRate, 4212 audio_format_t format, 4213 uint32_t channelMask, 4214 int frameCount) 4215 : Track(playbackThread, NULL, AUDIO_STREAM_CNT, sampleRate, format, channelMask, frameCount, NULL, 0), 4216 mActive(false), mSourceThread(sourceThread) 4217{ 4218 4219 if (mCblk != NULL) { 4220 mCblk->flags |= CBLK_DIRECTION_OUT; 4221 mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t); 4222 mOutBuffer.frameCount = 0; 4223 playbackThread->mTracks.add(this); 4224 ALOGV("OutputTrack constructor mCblk %p, mBuffer %p, mCblk->buffers %p, " \ 4225 "mCblk->frameCount %d, mCblk->sampleRate %d, mChannelMask 0x%08x mBufferEnd %p", 4226 mCblk, mBuffer, mCblk->buffers, 4227 mCblk->frameCount, mCblk->sampleRate, mChannelMask, mBufferEnd); 4228 } else { 4229 ALOGW("Error creating output track on thread %p", playbackThread); 4230 } 4231} 4232 4233AudioFlinger::PlaybackThread::OutputTrack::~OutputTrack() 4234{ 4235 clearBufferQueue(); 4236} 4237 4238status_t AudioFlinger::PlaybackThread::OutputTrack::start(pid_t tid) 4239{ 4240 status_t status = Track::start(tid); 4241 if (status != NO_ERROR) { 4242 return status; 4243 } 4244 4245 mActive = true; 4246 mRetryCount = 127; 4247 return status; 4248} 4249 4250void AudioFlinger::PlaybackThread::OutputTrack::stop() 4251{ 4252 Track::stop(); 4253 clearBufferQueue(); 4254 mOutBuffer.frameCount = 0; 4255 mActive = false; 4256} 4257 4258bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t frames) 4259{ 4260 Buffer *pInBuffer; 4261 Buffer inBuffer; 4262 uint32_t channelCount = mChannelCount; 4263 bool outputBufferFull = false; 4264 inBuffer.frameCount = frames; 4265 inBuffer.i16 = data; 4266 4267 uint32_t waitTimeLeftMs = mSourceThread->waitTimeMs(); 4268 4269 if (!mActive && frames != 0) { 4270 start(0); 4271 sp<ThreadBase> thread = mThread.promote(); 4272 if (thread != 0) { 4273 MixerThread *mixerThread = (MixerThread *)thread.get(); 4274 if (mCblk->frameCount > frames){ 4275 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 4276 uint32_t startFrames = (mCblk->frameCount - frames); 4277 pInBuffer = new Buffer; 4278 pInBuffer->mBuffer = new int16_t[startFrames * channelCount]; 4279 pInBuffer->frameCount = startFrames; 4280 pInBuffer->i16 = pInBuffer->mBuffer; 4281 memset(pInBuffer->raw, 0, startFrames * channelCount * sizeof(int16_t)); 4282 mBufferQueue.add(pInBuffer); 4283 } else { 4284 ALOGW ("OutputTrack::write() %p no more buffers in queue", this); 4285 } 4286 } 4287 } 4288 } 4289 4290 while (waitTimeLeftMs) { 4291 // First write pending buffers, then new data 4292 if (mBufferQueue.size()) { 4293 pInBuffer = mBufferQueue.itemAt(0); 4294 } else { 4295 pInBuffer = &inBuffer; 4296 } 4297 4298 if (pInBuffer->frameCount == 0) { 4299 break; 4300 } 4301 4302 if (mOutBuffer.frameCount == 0) { 4303 mOutBuffer.frameCount = pInBuffer->frameCount; 4304 nsecs_t startTime = systemTime(); 4305 if (obtainBuffer(&mOutBuffer, waitTimeLeftMs) == (status_t)NO_MORE_BUFFERS) { 4306 ALOGV ("OutputTrack::write() %p thread %p no more output buffers", this, mThread.unsafe_get()); 4307 outputBufferFull = true; 4308 break; 4309 } 4310 uint32_t waitTimeMs = (uint32_t)ns2ms(systemTime() - startTime); 4311 if (waitTimeLeftMs >= waitTimeMs) { 4312 waitTimeLeftMs -= waitTimeMs; 4313 } else { 4314 waitTimeLeftMs = 0; 4315 } 4316 } 4317 4318 uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount : pInBuffer->frameCount; 4319 memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channelCount * sizeof(int16_t)); 4320 mCblk->stepUser(outFrames); 4321 pInBuffer->frameCount -= outFrames; 4322 pInBuffer->i16 += outFrames * channelCount; 4323 mOutBuffer.frameCount -= outFrames; 4324 mOutBuffer.i16 += outFrames * channelCount; 4325 4326 if (pInBuffer->frameCount == 0) { 4327 if (mBufferQueue.size()) { 4328 mBufferQueue.removeAt(0); 4329 delete [] pInBuffer->mBuffer; 4330 delete pInBuffer; 4331 ALOGV("OutputTrack::write() %p thread %p released overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 4332 } else { 4333 break; 4334 } 4335 } 4336 } 4337 4338 // If we could not write all frames, allocate a buffer and queue it for next time. 4339 if (inBuffer.frameCount) { 4340 sp<ThreadBase> thread = mThread.promote(); 4341 if (thread != 0 && !thread->standby()) { 4342 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 4343 pInBuffer = new Buffer; 4344 pInBuffer->mBuffer = new int16_t[inBuffer.frameCount * channelCount]; 4345 pInBuffer->frameCount = inBuffer.frameCount; 4346 pInBuffer->i16 = pInBuffer->mBuffer; 4347 memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channelCount * sizeof(int16_t)); 4348 mBufferQueue.add(pInBuffer); 4349 ALOGV("OutputTrack::write() %p thread %p adding overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 4350 } else { 4351 ALOGW("OutputTrack::write() %p thread %p no more overflow buffers", mThread.unsafe_get(), this); 4352 } 4353 } 4354 } 4355 4356 // Calling write() with a 0 length buffer, means that no more data will be written: 4357 // If no more buffers are pending, fill output track buffer to make sure it is started 4358 // by output mixer. 4359 if (frames == 0 && mBufferQueue.size() == 0) { 4360 if (mCblk->user < mCblk->frameCount) { 4361 frames = mCblk->frameCount - mCblk->user; 4362 pInBuffer = new Buffer; 4363 pInBuffer->mBuffer = new int16_t[frames * channelCount]; 4364 pInBuffer->frameCount = frames; 4365 pInBuffer->i16 = pInBuffer->mBuffer; 4366 memset(pInBuffer->raw, 0, frames * channelCount * sizeof(int16_t)); 4367 mBufferQueue.add(pInBuffer); 4368 } else if (mActive) { 4369 stop(); 4370 } 4371 } 4372 4373 return outputBufferFull; 4374} 4375 4376status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer(AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs) 4377{ 4378 int active; 4379 status_t result; 4380 audio_track_cblk_t* cblk = mCblk; 4381 uint32_t framesReq = buffer->frameCount; 4382 4383// ALOGV("OutputTrack::obtainBuffer user %d, server %d", cblk->user, cblk->server); 4384 buffer->frameCount = 0; 4385 4386 uint32_t framesAvail = cblk->framesAvailable(); 4387 4388 4389 if (framesAvail == 0) { 4390 Mutex::Autolock _l(cblk->lock); 4391 goto start_loop_here; 4392 while (framesAvail == 0) { 4393 active = mActive; 4394 if (CC_UNLIKELY(!active)) { 4395 ALOGV("Not active and NO_MORE_BUFFERS"); 4396 return NO_MORE_BUFFERS; 4397 } 4398 result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs)); 4399 if (result != NO_ERROR) { 4400 return NO_MORE_BUFFERS; 4401 } 4402 // read the server count again 4403 start_loop_here: 4404 framesAvail = cblk->framesAvailable_l(); 4405 } 4406 } 4407 4408// if (framesAvail < framesReq) { 4409// return NO_MORE_BUFFERS; 4410// } 4411 4412 if (framesReq > framesAvail) { 4413 framesReq = framesAvail; 4414 } 4415 4416 uint32_t u = cblk->user; 4417 uint32_t bufferEnd = cblk->userBase + cblk->frameCount; 4418 4419 if (u + framesReq > bufferEnd) { 4420 framesReq = bufferEnd - u; 4421 } 4422 4423 buffer->frameCount = framesReq; 4424 buffer->raw = (void *)cblk->buffer(u); 4425 return NO_ERROR; 4426} 4427 4428 4429void AudioFlinger::PlaybackThread::OutputTrack::clearBufferQueue() 4430{ 4431 size_t size = mBufferQueue.size(); 4432 4433 for (size_t i = 0; i < size; i++) { 4434 Buffer *pBuffer = mBufferQueue.itemAt(i); 4435 delete [] pBuffer->mBuffer; 4436 delete pBuffer; 4437 } 4438 mBufferQueue.clear(); 4439} 4440 4441// ---------------------------------------------------------------------------- 4442 4443AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid) 4444 : RefBase(), 4445 mAudioFlinger(audioFlinger), 4446 // FIXME should be a "k" constant not hard-coded, in .h or ro. property, see 4 lines below 4447 mMemoryDealer(new MemoryDealer(1024*1024, "AudioFlinger::Client")), 4448 mPid(pid), 4449 mTimedTrackCount(0) 4450{ 4451 // 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer 4452} 4453 4454// Client destructor must be called with AudioFlinger::mLock held 4455AudioFlinger::Client::~Client() 4456{ 4457 mAudioFlinger->removeClient_l(mPid); 4458} 4459 4460sp<MemoryDealer> AudioFlinger::Client::heap() const 4461{ 4462 return mMemoryDealer; 4463} 4464 4465// Reserve one of the limited slots for a timed audio track associated 4466// with this client 4467bool AudioFlinger::Client::reserveTimedTrack() 4468{ 4469 const int kMaxTimedTracksPerClient = 4; 4470 4471 Mutex::Autolock _l(mTimedTrackLock); 4472 4473 if (mTimedTrackCount >= kMaxTimedTracksPerClient) { 4474 ALOGW("can not create timed track - pid %d has exceeded the limit", 4475 mPid); 4476 return false; 4477 } 4478 4479 mTimedTrackCount++; 4480 return true; 4481} 4482 4483// Release a slot for a timed audio track 4484void AudioFlinger::Client::releaseTimedTrack() 4485{ 4486 Mutex::Autolock _l(mTimedTrackLock); 4487 mTimedTrackCount--; 4488} 4489 4490// ---------------------------------------------------------------------------- 4491 4492AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger, 4493 const sp<IAudioFlingerClient>& client, 4494 pid_t pid) 4495 : mAudioFlinger(audioFlinger), mPid(pid), mAudioFlingerClient(client) 4496{ 4497} 4498 4499AudioFlinger::NotificationClient::~NotificationClient() 4500{ 4501} 4502 4503void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who) 4504{ 4505 sp<NotificationClient> keep(this); 4506 mAudioFlinger->removeNotificationClient(mPid); 4507} 4508 4509// ---------------------------------------------------------------------------- 4510 4511AudioFlinger::TrackHandle::TrackHandle(const sp<AudioFlinger::PlaybackThread::Track>& track) 4512 : BnAudioTrack(), 4513 mTrack(track) 4514{ 4515} 4516 4517AudioFlinger::TrackHandle::~TrackHandle() { 4518 // just stop the track on deletion, associated resources 4519 // will be freed from the main thread once all pending buffers have 4520 // been played. Unless it's not in the active track list, in which 4521 // case we free everything now... 4522 mTrack->destroy(); 4523} 4524 4525sp<IMemory> AudioFlinger::TrackHandle::getCblk() const { 4526 return mTrack->getCblk(); 4527} 4528 4529status_t AudioFlinger::TrackHandle::start(pid_t tid) { 4530 return mTrack->start(tid); 4531} 4532 4533void AudioFlinger::TrackHandle::stop() { 4534 mTrack->stop(); 4535} 4536 4537void AudioFlinger::TrackHandle::flush() { 4538 mTrack->flush(); 4539} 4540 4541void AudioFlinger::TrackHandle::mute(bool e) { 4542 mTrack->mute(e); 4543} 4544 4545void AudioFlinger::TrackHandle::pause() { 4546 mTrack->pause(); 4547} 4548 4549status_t AudioFlinger::TrackHandle::attachAuxEffect(int EffectId) 4550{ 4551 return mTrack->attachAuxEffect(EffectId); 4552} 4553 4554status_t AudioFlinger::TrackHandle::allocateTimedBuffer(size_t size, 4555 sp<IMemory>* buffer) { 4556 if (!mTrack->isTimedTrack()) 4557 return INVALID_OPERATION; 4558 4559 PlaybackThread::TimedTrack* tt = 4560 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 4561 return tt->allocateTimedBuffer(size, buffer); 4562} 4563 4564status_t AudioFlinger::TrackHandle::queueTimedBuffer(const sp<IMemory>& buffer, 4565 int64_t pts) { 4566 if (!mTrack->isTimedTrack()) 4567 return INVALID_OPERATION; 4568 4569 PlaybackThread::TimedTrack* tt = 4570 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 4571 return tt->queueTimedBuffer(buffer, pts); 4572} 4573 4574status_t AudioFlinger::TrackHandle::setMediaTimeTransform( 4575 const LinearTransform& xform, int target) { 4576 4577 if (!mTrack->isTimedTrack()) 4578 return INVALID_OPERATION; 4579 4580 PlaybackThread::TimedTrack* tt = 4581 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 4582 return tt->setMediaTimeTransform( 4583 xform, static_cast<TimedAudioTrack::TargetTimeline>(target)); 4584} 4585 4586status_t AudioFlinger::TrackHandle::onTransact( 4587 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 4588{ 4589 return BnAudioTrack::onTransact(code, data, reply, flags); 4590} 4591 4592// ---------------------------------------------------------------------------- 4593 4594sp<IAudioRecord> AudioFlinger::openRecord( 4595 pid_t pid, 4596 audio_io_handle_t input, 4597 uint32_t sampleRate, 4598 audio_format_t format, 4599 uint32_t channelMask, 4600 int frameCount, 4601 // FIXME dead, remove from IAudioFlinger 4602 uint32_t flags, 4603 int *sessionId, 4604 status_t *status) 4605{ 4606 sp<RecordThread::RecordTrack> recordTrack; 4607 sp<RecordHandle> recordHandle; 4608 sp<Client> client; 4609 status_t lStatus; 4610 RecordThread *thread; 4611 size_t inFrameCount; 4612 int lSessionId; 4613 4614 // check calling permissions 4615 if (!recordingAllowed()) { 4616 lStatus = PERMISSION_DENIED; 4617 goto Exit; 4618 } 4619 4620 // add client to list 4621 { // scope for mLock 4622 Mutex::Autolock _l(mLock); 4623 thread = checkRecordThread_l(input); 4624 if (thread == NULL) { 4625 lStatus = BAD_VALUE; 4626 goto Exit; 4627 } 4628 4629 client = registerPid_l(pid); 4630 4631 // If no audio session id is provided, create one here 4632 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 4633 lSessionId = *sessionId; 4634 } else { 4635 lSessionId = nextUniqueId(); 4636 if (sessionId != NULL) { 4637 *sessionId = lSessionId; 4638 } 4639 } 4640 // create new record track. The record track uses one track in mHardwareMixerThread by convention. 4641 recordTrack = thread->createRecordTrack_l(client, 4642 sampleRate, 4643 format, 4644 channelMask, 4645 frameCount, 4646 lSessionId, 4647 &lStatus); 4648 } 4649 if (lStatus != NO_ERROR) { 4650 // remove local strong reference to Client before deleting the RecordTrack so that the Client 4651 // destructor is called by the TrackBase destructor with mLock held 4652 client.clear(); 4653 recordTrack.clear(); 4654 goto Exit; 4655 } 4656 4657 // return to handle to client 4658 recordHandle = new RecordHandle(recordTrack); 4659 lStatus = NO_ERROR; 4660 4661Exit: 4662 if (status) { 4663 *status = lStatus; 4664 } 4665 return recordHandle; 4666} 4667 4668// ---------------------------------------------------------------------------- 4669 4670AudioFlinger::RecordHandle::RecordHandle(const sp<AudioFlinger::RecordThread::RecordTrack>& recordTrack) 4671 : BnAudioRecord(), 4672 mRecordTrack(recordTrack) 4673{ 4674} 4675 4676AudioFlinger::RecordHandle::~RecordHandle() { 4677 stop(); 4678} 4679 4680sp<IMemory> AudioFlinger::RecordHandle::getCblk() const { 4681 return mRecordTrack->getCblk(); 4682} 4683 4684status_t AudioFlinger::RecordHandle::start(pid_t tid) { 4685 ALOGV("RecordHandle::start()"); 4686 return mRecordTrack->start(tid); 4687} 4688 4689void AudioFlinger::RecordHandle::stop() { 4690 ALOGV("RecordHandle::stop()"); 4691 mRecordTrack->stop(); 4692} 4693 4694status_t AudioFlinger::RecordHandle::onTransact( 4695 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 4696{ 4697 return BnAudioRecord::onTransact(code, data, reply, flags); 4698} 4699 4700// ---------------------------------------------------------------------------- 4701 4702AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 4703 AudioStreamIn *input, 4704 uint32_t sampleRate, 4705 uint32_t channels, 4706 audio_io_handle_t id, 4707 uint32_t device) : 4708 ThreadBase(audioFlinger, id, device, RECORD), 4709 mInput(input), mTrack(NULL), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL), 4710 // mRsmpInIndex and mInputBytes set by readInputParameters() 4711 mReqChannelCount(popcount(channels)), 4712 mReqSampleRate(sampleRate) 4713 // mBytesRead is only meaningful while active, and so is cleared in start() 4714 // (but might be better to also clear here for dump?) 4715{ 4716 snprintf(mName, kNameLength, "AudioIn_%X", id); 4717 4718 readInputParameters(); 4719} 4720 4721 4722AudioFlinger::RecordThread::~RecordThread() 4723{ 4724 delete[] mRsmpInBuffer; 4725 delete mResampler; 4726 delete[] mRsmpOutBuffer; 4727} 4728 4729void AudioFlinger::RecordThread::onFirstRef() 4730{ 4731 run(mName, PRIORITY_URGENT_AUDIO); 4732} 4733 4734status_t AudioFlinger::RecordThread::readyToRun() 4735{ 4736 status_t status = initCheck(); 4737 ALOGW_IF(status != NO_ERROR,"RecordThread %p could not initialize", this); 4738 return status; 4739} 4740 4741bool AudioFlinger::RecordThread::threadLoop() 4742{ 4743 AudioBufferProvider::Buffer buffer; 4744 sp<RecordTrack> activeTrack; 4745 Vector< sp<EffectChain> > effectChains; 4746 4747 nsecs_t lastWarning = 0; 4748 4749 acquireWakeLock(); 4750 4751 // start recording 4752 while (!exitPending()) { 4753 4754 processConfigEvents(); 4755 4756 { // scope for mLock 4757 Mutex::Autolock _l(mLock); 4758 checkForNewParameters_l(); 4759 if (mActiveTrack == 0 && mConfigEvents.isEmpty()) { 4760 if (!mStandby) { 4761 mInput->stream->common.standby(&mInput->stream->common); 4762 mStandby = true; 4763 } 4764 4765 if (exitPending()) break; 4766 4767 releaseWakeLock_l(); 4768 ALOGV("RecordThread: loop stopping"); 4769 // go to sleep 4770 mWaitWorkCV.wait(mLock); 4771 ALOGV("RecordThread: loop starting"); 4772 acquireWakeLock_l(); 4773 continue; 4774 } 4775 if (mActiveTrack != 0) { 4776 if (mActiveTrack->mState == TrackBase::PAUSING) { 4777 if (!mStandby) { 4778 mInput->stream->common.standby(&mInput->stream->common); 4779 mStandby = true; 4780 } 4781 mActiveTrack.clear(); 4782 mStartStopCond.broadcast(); 4783 } else if (mActiveTrack->mState == TrackBase::RESUMING) { 4784 if (mReqChannelCount != mActiveTrack->channelCount()) { 4785 mActiveTrack.clear(); 4786 mStartStopCond.broadcast(); 4787 } else if (mBytesRead != 0) { 4788 // record start succeeds only if first read from audio input 4789 // succeeds 4790 if (mBytesRead > 0) { 4791 mActiveTrack->mState = TrackBase::ACTIVE; 4792 } else { 4793 mActiveTrack.clear(); 4794 } 4795 mStartStopCond.broadcast(); 4796 } 4797 mStandby = false; 4798 } 4799 } 4800 lockEffectChains_l(effectChains); 4801 } 4802 4803 if (mActiveTrack != 0) { 4804 if (mActiveTrack->mState != TrackBase::ACTIVE && 4805 mActiveTrack->mState != TrackBase::RESUMING) { 4806 unlockEffectChains(effectChains); 4807 usleep(kRecordThreadSleepUs); 4808 continue; 4809 } 4810 for (size_t i = 0; i < effectChains.size(); i ++) { 4811 effectChains[i]->process_l(); 4812 } 4813 4814 buffer.frameCount = mFrameCount; 4815 if (CC_LIKELY(mActiveTrack->getNextBuffer(&buffer) == NO_ERROR)) { 4816 size_t framesOut = buffer.frameCount; 4817 if (mResampler == NULL) { 4818 // no resampling 4819 while (framesOut) { 4820 size_t framesIn = mFrameCount - mRsmpInIndex; 4821 if (framesIn) { 4822 int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize; 4823 int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * mActiveTrack->mCblk->frameSize; 4824 if (framesIn > framesOut) 4825 framesIn = framesOut; 4826 mRsmpInIndex += framesIn; 4827 framesOut -= framesIn; 4828 if ((int)mChannelCount == mReqChannelCount || 4829 mFormat != AUDIO_FORMAT_PCM_16_BIT) { 4830 memcpy(dst, src, framesIn * mFrameSize); 4831 } else { 4832 int16_t *src16 = (int16_t *)src; 4833 int16_t *dst16 = (int16_t *)dst; 4834 if (mChannelCount == 1) { 4835 while (framesIn--) { 4836 *dst16++ = *src16; 4837 *dst16++ = *src16++; 4838 } 4839 } else { 4840 while (framesIn--) { 4841 *dst16++ = (int16_t)(((int32_t)*src16 + (int32_t)*(src16 + 1)) >> 1); 4842 src16 += 2; 4843 } 4844 } 4845 } 4846 } 4847 if (framesOut && mFrameCount == mRsmpInIndex) { 4848 if (framesOut == mFrameCount && 4849 ((int)mChannelCount == mReqChannelCount || mFormat != AUDIO_FORMAT_PCM_16_BIT)) { 4850 mBytesRead = mInput->stream->read(mInput->stream, buffer.raw, mInputBytes); 4851 framesOut = 0; 4852 } else { 4853 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 4854 mRsmpInIndex = 0; 4855 } 4856 if (mBytesRead < 0) { 4857 ALOGE("Error reading audio input"); 4858 if (mActiveTrack->mState == TrackBase::ACTIVE) { 4859 // Force input into standby so that it tries to 4860 // recover at next read attempt 4861 mInput->stream->common.standby(&mInput->stream->common); 4862 usleep(kRecordThreadSleepUs); 4863 } 4864 mRsmpInIndex = mFrameCount; 4865 framesOut = 0; 4866 buffer.frameCount = 0; 4867 } 4868 } 4869 } 4870 } else { 4871 // resampling 4872 4873 memset(mRsmpOutBuffer, 0, framesOut * 2 * sizeof(int32_t)); 4874 // alter output frame count as if we were expecting stereo samples 4875 if (mChannelCount == 1 && mReqChannelCount == 1) { 4876 framesOut >>= 1; 4877 } 4878 mResampler->resample(mRsmpOutBuffer, framesOut, this); 4879 // ditherAndClamp() works as long as all buffers returned by mActiveTrack->getNextBuffer() 4880 // are 32 bit aligned which should be always true. 4881 if (mChannelCount == 2 && mReqChannelCount == 1) { 4882 ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut); 4883 // the resampler always outputs stereo samples: do post stereo to mono conversion 4884 int16_t *src = (int16_t *)mRsmpOutBuffer; 4885 int16_t *dst = buffer.i16; 4886 while (framesOut--) { 4887 *dst++ = (int16_t)(((int32_t)*src + (int32_t)*(src + 1)) >> 1); 4888 src += 2; 4889 } 4890 } else { 4891 ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut); 4892 } 4893 4894 } 4895 mActiveTrack->releaseBuffer(&buffer); 4896 mActiveTrack->overflow(); 4897 } 4898 // client isn't retrieving buffers fast enough 4899 else { 4900 if (!mActiveTrack->setOverflow()) { 4901 nsecs_t now = systemTime(); 4902 if ((now - lastWarning) > kWarningThrottleNs) { 4903 ALOGW("RecordThread: buffer overflow"); 4904 lastWarning = now; 4905 } 4906 } 4907 // Release the processor for a while before asking for a new buffer. 4908 // This will give the application more chance to read from the buffer and 4909 // clear the overflow. 4910 usleep(kRecordThreadSleepUs); 4911 } 4912 } 4913 // enable changes in effect chain 4914 unlockEffectChains(effectChains); 4915 effectChains.clear(); 4916 } 4917 4918 if (!mStandby) { 4919 mInput->stream->common.standby(&mInput->stream->common); 4920 } 4921 mActiveTrack.clear(); 4922 4923 mStartStopCond.broadcast(); 4924 4925 releaseWakeLock(); 4926 4927 ALOGV("RecordThread %p exiting", this); 4928 return false; 4929} 4930 4931 4932sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 4933 const sp<AudioFlinger::Client>& client, 4934 uint32_t sampleRate, 4935 audio_format_t format, 4936 int channelMask, 4937 int frameCount, 4938 int sessionId, 4939 status_t *status) 4940{ 4941 sp<RecordTrack> track; 4942 status_t lStatus; 4943 4944 lStatus = initCheck(); 4945 if (lStatus != NO_ERROR) { 4946 ALOGE("Audio driver not initialized."); 4947 goto Exit; 4948 } 4949 4950 { // scope for mLock 4951 Mutex::Autolock _l(mLock); 4952 4953 track = new RecordTrack(this, client, sampleRate, 4954 format, channelMask, frameCount, sessionId); 4955 4956 if (track->getCblk() == 0) { 4957 lStatus = NO_MEMORY; 4958 goto Exit; 4959 } 4960 4961 mTrack = track.get(); 4962 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 4963 bool suspend = audio_is_bluetooth_sco_device( 4964 (audio_devices_t)(mDevice & AUDIO_DEVICE_IN_ALL)) && mAudioFlinger->btNrecIsOff(); 4965 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 4966 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 4967 } 4968 lStatus = NO_ERROR; 4969 4970Exit: 4971 if (status) { 4972 *status = lStatus; 4973 } 4974 return track; 4975} 4976 4977status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, pid_t tid) 4978{ 4979 ALOGV("RecordThread::start tid=%d", tid); 4980 sp <ThreadBase> strongMe = this; 4981 status_t status = NO_ERROR; 4982 { 4983 AutoMutex lock(mLock); 4984 if (mActiveTrack != 0) { 4985 if (recordTrack != mActiveTrack.get()) { 4986 status = -EBUSY; 4987 } else if (mActiveTrack->mState == TrackBase::PAUSING) { 4988 mActiveTrack->mState = TrackBase::ACTIVE; 4989 } 4990 return status; 4991 } 4992 4993 recordTrack->mState = TrackBase::IDLE; 4994 mActiveTrack = recordTrack; 4995 mLock.unlock(); 4996 status_t status = AudioSystem::startInput(mId); 4997 mLock.lock(); 4998 if (status != NO_ERROR) { 4999 mActiveTrack.clear(); 5000 return status; 5001 } 5002 mRsmpInIndex = mFrameCount; 5003 mBytesRead = 0; 5004 if (mResampler != NULL) { 5005 mResampler->reset(); 5006 } 5007 mActiveTrack->mState = TrackBase::RESUMING; 5008 // signal thread to start 5009 ALOGV("Signal record thread"); 5010 mWaitWorkCV.signal(); 5011 // do not wait for mStartStopCond if exiting 5012 if (exitPending()) { 5013 mActiveTrack.clear(); 5014 status = INVALID_OPERATION; 5015 goto startError; 5016 } 5017 mStartStopCond.wait(mLock); 5018 if (mActiveTrack == 0) { 5019 ALOGV("Record failed to start"); 5020 status = BAD_VALUE; 5021 goto startError; 5022 } 5023 ALOGV("Record started OK"); 5024 return status; 5025 } 5026startError: 5027 AudioSystem::stopInput(mId); 5028 return status; 5029} 5030 5031void AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 5032 ALOGV("RecordThread::stop"); 5033 sp <ThreadBase> strongMe = this; 5034 { 5035 AutoMutex lock(mLock); 5036 if (mActiveTrack != 0 && recordTrack == mActiveTrack.get()) { 5037 mActiveTrack->mState = TrackBase::PAUSING; 5038 // do not wait for mStartStopCond if exiting 5039 if (exitPending()) { 5040 return; 5041 } 5042 mStartStopCond.wait(mLock); 5043 // if we have been restarted, recordTrack == mActiveTrack.get() here 5044 if (mActiveTrack == 0 || recordTrack != mActiveTrack.get()) { 5045 mLock.unlock(); 5046 AudioSystem::stopInput(mId); 5047 mLock.lock(); 5048 ALOGV("Record stopped OK"); 5049 } 5050 } 5051 } 5052} 5053 5054status_t AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 5055{ 5056 const size_t SIZE = 256; 5057 char buffer[SIZE]; 5058 String8 result; 5059 5060 snprintf(buffer, SIZE, "\nInput thread %p internals\n", this); 5061 result.append(buffer); 5062 5063 if (mActiveTrack != 0) { 5064 result.append("Active Track:\n"); 5065 result.append(" Clien Fmt Chn mask Session Buf S SRate Serv User\n"); 5066 mActiveTrack->dump(buffer, SIZE); 5067 result.append(buffer); 5068 5069 snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex); 5070 result.append(buffer); 5071 snprintf(buffer, SIZE, "In size: %d\n", mInputBytes); 5072 result.append(buffer); 5073 snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != NULL)); 5074 result.append(buffer); 5075 snprintf(buffer, SIZE, "Out channel count: %d\n", mReqChannelCount); 5076 result.append(buffer); 5077 snprintf(buffer, SIZE, "Out sample rate: %d\n", mReqSampleRate); 5078 result.append(buffer); 5079 5080 5081 } else { 5082 result.append("No record client\n"); 5083 } 5084 write(fd, result.string(), result.size()); 5085 5086 dumpBase(fd, args); 5087 dumpEffectChains(fd, args); 5088 5089 return NO_ERROR; 5090} 5091 5092// AudioBufferProvider interface 5093status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 5094{ 5095 size_t framesReq = buffer->frameCount; 5096 size_t framesReady = mFrameCount - mRsmpInIndex; 5097 int channelCount; 5098 5099 if (framesReady == 0) { 5100 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 5101 if (mBytesRead < 0) { 5102 ALOGE("RecordThread::getNextBuffer() Error reading audio input"); 5103 if (mActiveTrack->mState == TrackBase::ACTIVE) { 5104 // Force input into standby so that it tries to 5105 // recover at next read attempt 5106 mInput->stream->common.standby(&mInput->stream->common); 5107 usleep(kRecordThreadSleepUs); 5108 } 5109 buffer->raw = NULL; 5110 buffer->frameCount = 0; 5111 return NOT_ENOUGH_DATA; 5112 } 5113 mRsmpInIndex = 0; 5114 framesReady = mFrameCount; 5115 } 5116 5117 if (framesReq > framesReady) { 5118 framesReq = framesReady; 5119 } 5120 5121 if (mChannelCount == 1 && mReqChannelCount == 2) { 5122 channelCount = 1; 5123 } else { 5124 channelCount = 2; 5125 } 5126 buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount; 5127 buffer->frameCount = framesReq; 5128 return NO_ERROR; 5129} 5130 5131// AudioBufferProvider interface 5132void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer) 5133{ 5134 mRsmpInIndex += buffer->frameCount; 5135 buffer->frameCount = 0; 5136} 5137 5138bool AudioFlinger::RecordThread::checkForNewParameters_l() 5139{ 5140 bool reconfig = false; 5141 5142 while (!mNewParameters.isEmpty()) { 5143 status_t status = NO_ERROR; 5144 String8 keyValuePair = mNewParameters[0]; 5145 AudioParameter param = AudioParameter(keyValuePair); 5146 int value; 5147 audio_format_t reqFormat = mFormat; 5148 int reqSamplingRate = mReqSampleRate; 5149 int reqChannelCount = mReqChannelCount; 5150 5151 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 5152 reqSamplingRate = value; 5153 reconfig = true; 5154 } 5155 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 5156 reqFormat = (audio_format_t) value; 5157 reconfig = true; 5158 } 5159 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 5160 reqChannelCount = popcount(value); 5161 reconfig = true; 5162 } 5163 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 5164 // do not accept frame count changes if tracks are open as the track buffer 5165 // size depends on frame count and correct behavior would not be guaranteed 5166 // if frame count is changed after track creation 5167 if (mActiveTrack != 0) { 5168 status = INVALID_OPERATION; 5169 } else { 5170 reconfig = true; 5171 } 5172 } 5173 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 5174 // forward device change to effects that have requested to be 5175 // aware of attached audio device. 5176 for (size_t i = 0; i < mEffectChains.size(); i++) { 5177 mEffectChains[i]->setDevice_l(value); 5178 } 5179 // store input device and output device but do not forward output device to audio HAL. 5180 // Note that status is ignored by the caller for output device 5181 // (see AudioFlinger::setParameters() 5182 if (value & AUDIO_DEVICE_OUT_ALL) { 5183 mDevice &= (uint32_t)~(value & AUDIO_DEVICE_OUT_ALL); 5184 status = BAD_VALUE; 5185 } else { 5186 mDevice &= (uint32_t)~(value & AUDIO_DEVICE_IN_ALL); 5187 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 5188 if (mTrack != NULL) { 5189 bool suspend = audio_is_bluetooth_sco_device( 5190 (audio_devices_t)value) && mAudioFlinger->btNrecIsOff(); 5191 setEffectSuspended_l(FX_IID_AEC, suspend, mTrack->sessionId()); 5192 setEffectSuspended_l(FX_IID_NS, suspend, mTrack->sessionId()); 5193 } 5194 } 5195 mDevice |= (uint32_t)value; 5196 } 5197 if (status == NO_ERROR) { 5198 status = mInput->stream->common.set_parameters(&mInput->stream->common, keyValuePair.string()); 5199 if (status == INVALID_OPERATION) { 5200 mInput->stream->common.standby(&mInput->stream->common); 5201 status = mInput->stream->common.set_parameters(&mInput->stream->common, keyValuePair.string()); 5202 } 5203 if (reconfig) { 5204 if (status == BAD_VALUE && 5205 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 5206 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 5207 ((int)mInput->stream->common.get_sample_rate(&mInput->stream->common) <= (2 * reqSamplingRate)) && 5208 popcount(mInput->stream->common.get_channels(&mInput->stream->common)) <= FCC_2 && 5209 (reqChannelCount <= FCC_2)) { 5210 status = NO_ERROR; 5211 } 5212 if (status == NO_ERROR) { 5213 readInputParameters(); 5214 sendConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 5215 } 5216 } 5217 } 5218 5219 mNewParameters.removeAt(0); 5220 5221 mParamStatus = status; 5222 mParamCond.signal(); 5223 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 5224 // already timed out waiting for the status and will never signal the condition. 5225 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 5226 } 5227 return reconfig; 5228} 5229 5230String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 5231{ 5232 char *s; 5233 String8 out_s8 = String8(); 5234 5235 Mutex::Autolock _l(mLock); 5236 if (initCheck() != NO_ERROR) { 5237 return out_s8; 5238 } 5239 5240 s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 5241 out_s8 = String8(s); 5242 free(s); 5243 return out_s8; 5244} 5245 5246void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) { 5247 AudioSystem::OutputDescriptor desc; 5248 void *param2 = NULL; 5249 5250 switch (event) { 5251 case AudioSystem::INPUT_OPENED: 5252 case AudioSystem::INPUT_CONFIG_CHANGED: 5253 desc.channels = mChannelMask; 5254 desc.samplingRate = mSampleRate; 5255 desc.format = mFormat; 5256 desc.frameCount = mFrameCount; 5257 desc.latency = 0; 5258 param2 = &desc; 5259 break; 5260 5261 case AudioSystem::INPUT_CLOSED: 5262 default: 5263 break; 5264 } 5265 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 5266} 5267 5268void AudioFlinger::RecordThread::readInputParameters() 5269{ 5270 delete mRsmpInBuffer; 5271 // mRsmpInBuffer is always assigned a new[] below 5272 delete mRsmpOutBuffer; 5273 mRsmpOutBuffer = NULL; 5274 delete mResampler; 5275 mResampler = NULL; 5276 5277 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 5278 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 5279 mChannelCount = (uint16_t)popcount(mChannelMask); 5280 mFormat = mInput->stream->common.get_format(&mInput->stream->common); 5281 mFrameSize = audio_stream_frame_size(&mInput->stream->common); 5282 mInputBytes = mInput->stream->common.get_buffer_size(&mInput->stream->common); 5283 mFrameCount = mInputBytes / mFrameSize; 5284 mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount]; 5285 5286 if (mSampleRate != mReqSampleRate && mChannelCount <= FCC_2 && mReqChannelCount <= FCC_2) 5287 { 5288 int channelCount; 5289 // optmization: if mono to mono, use the resampler in stereo to stereo mode to avoid 5290 // stereo to mono post process as the resampler always outputs stereo. 5291 if (mChannelCount == 1 && mReqChannelCount == 2) { 5292 channelCount = 1; 5293 } else { 5294 channelCount = 2; 5295 } 5296 mResampler = AudioResampler::create(16, channelCount, mReqSampleRate); 5297 mResampler->setSampleRate(mSampleRate); 5298 mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN); 5299 mRsmpOutBuffer = new int32_t[mFrameCount * 2]; 5300 5301 // optmization: if mono to mono, alter input frame count as if we were inputing stereo samples 5302 if (mChannelCount == 1 && mReqChannelCount == 1) { 5303 mFrameCount >>= 1; 5304 } 5305 5306 } 5307 mRsmpInIndex = mFrameCount; 5308} 5309 5310unsigned int AudioFlinger::RecordThread::getInputFramesLost() 5311{ 5312 Mutex::Autolock _l(mLock); 5313 if (initCheck() != NO_ERROR) { 5314 return 0; 5315 } 5316 5317 return mInput->stream->get_input_frames_lost(mInput->stream); 5318} 5319 5320uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) 5321{ 5322 Mutex::Autolock _l(mLock); 5323 uint32_t result = 0; 5324 if (getEffectChain_l(sessionId) != 0) { 5325 result = EFFECT_SESSION; 5326 } 5327 5328 if (mTrack != NULL && sessionId == mTrack->sessionId()) { 5329 result |= TRACK_SESSION; 5330 } 5331 5332 return result; 5333} 5334 5335AudioFlinger::RecordThread::RecordTrack* AudioFlinger::RecordThread::track() 5336{ 5337 Mutex::Autolock _l(mLock); 5338 return mTrack; 5339} 5340 5341AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::getInput() const 5342{ 5343 Mutex::Autolock _l(mLock); 5344 return mInput; 5345} 5346 5347AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 5348{ 5349 Mutex::Autolock _l(mLock); 5350 AudioStreamIn *input = mInput; 5351 mInput = NULL; 5352 return input; 5353} 5354 5355// this method must always be called either with ThreadBase mLock held or inside the thread loop 5356audio_stream_t* AudioFlinger::RecordThread::stream() 5357{ 5358 if (mInput == NULL) { 5359 return NULL; 5360 } 5361 return &mInput->stream->common; 5362} 5363 5364 5365// ---------------------------------------------------------------------------- 5366 5367audio_io_handle_t AudioFlinger::openOutput(uint32_t *pDevices, 5368 uint32_t *pSamplingRate, 5369 audio_format_t *pFormat, 5370 uint32_t *pChannels, 5371 uint32_t *pLatencyMs, 5372 audio_policy_output_flags_t flags) 5373{ 5374 status_t status; 5375 PlaybackThread *thread = NULL; 5376 uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0; 5377 audio_format_t format = pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT; 5378 uint32_t channels = pChannels ? *pChannels : 0; 5379 uint32_t latency = pLatencyMs ? *pLatencyMs : 0; 5380 audio_stream_out_t *outStream; 5381 audio_hw_device_t *outHwDev; 5382 5383 ALOGV("openOutput(), Device %x, SamplingRate %d, Format %d, Channels %x, flags %x", 5384 pDevices ? *pDevices : 0, 5385 samplingRate, 5386 format, 5387 channels, 5388 flags); 5389 5390 if (pDevices == NULL || *pDevices == 0) { 5391 return 0; 5392 } 5393 5394 Mutex::Autolock _l(mLock); 5395 5396 outHwDev = findSuitableHwDev_l(*pDevices); 5397 if (outHwDev == NULL) 5398 return 0; 5399 5400 mHardwareStatus = AUDIO_HW_OUTPUT_OPEN; 5401 status = outHwDev->open_output_stream(outHwDev, *pDevices, &format, 5402 &channels, &samplingRate, &outStream); 5403 mHardwareStatus = AUDIO_HW_IDLE; 5404 ALOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %d, Channels %x, status %d", 5405 outStream, 5406 samplingRate, 5407 format, 5408 channels, 5409 status); 5410 5411 if (outStream != NULL) { 5412 AudioStreamOut *output = new AudioStreamOut(outHwDev, outStream); 5413 audio_io_handle_t id = nextUniqueId(); 5414 5415 if ((flags & AUDIO_POLICY_OUTPUT_FLAG_DIRECT) || 5416 (format != AUDIO_FORMAT_PCM_16_BIT) || 5417 (channels != AUDIO_CHANNEL_OUT_STEREO)) { 5418 thread = new DirectOutputThread(this, output, id, *pDevices); 5419 ALOGV("openOutput() created direct output: ID %d thread %p", id, thread); 5420 } else { 5421 thread = new MixerThread(this, output, id, *pDevices); 5422 ALOGV("openOutput() created mixer output: ID %d thread %p", id, thread); 5423 } 5424 mPlaybackThreads.add(id, thread); 5425 5426 if (pSamplingRate != NULL) *pSamplingRate = samplingRate; 5427 if (pFormat != NULL) *pFormat = format; 5428 if (pChannels != NULL) *pChannels = channels; 5429 if (pLatencyMs != NULL) *pLatencyMs = thread->latency(); 5430 5431 // notify client processes of the new output creation 5432 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 5433 return id; 5434 } 5435 5436 return 0; 5437} 5438 5439audio_io_handle_t AudioFlinger::openDuplicateOutput(audio_io_handle_t output1, 5440 audio_io_handle_t output2) 5441{ 5442 Mutex::Autolock _l(mLock); 5443 MixerThread *thread1 = checkMixerThread_l(output1); 5444 MixerThread *thread2 = checkMixerThread_l(output2); 5445 5446 if (thread1 == NULL || thread2 == NULL) { 5447 ALOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1, output2); 5448 return 0; 5449 } 5450 5451 audio_io_handle_t id = nextUniqueId(); 5452 DuplicatingThread *thread = new DuplicatingThread(this, thread1, id); 5453 thread->addOutputTrack(thread2); 5454 mPlaybackThreads.add(id, thread); 5455 // notify client processes of the new output creation 5456 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 5457 return id; 5458} 5459 5460status_t AudioFlinger::closeOutput(audio_io_handle_t output) 5461{ 5462 // keep strong reference on the playback thread so that 5463 // it is not destroyed while exit() is executed 5464 sp <PlaybackThread> thread; 5465 { 5466 Mutex::Autolock _l(mLock); 5467 thread = checkPlaybackThread_l(output); 5468 if (thread == NULL) { 5469 return BAD_VALUE; 5470 } 5471 5472 ALOGV("closeOutput() %d", output); 5473 5474 if (thread->type() == ThreadBase::MIXER) { 5475 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5476 if (mPlaybackThreads.valueAt(i)->type() == ThreadBase::DUPLICATING) { 5477 DuplicatingThread *dupThread = (DuplicatingThread *)mPlaybackThreads.valueAt(i).get(); 5478 dupThread->removeOutputTrack((MixerThread *)thread.get()); 5479 } 5480 } 5481 } 5482 audioConfigChanged_l(AudioSystem::OUTPUT_CLOSED, output, NULL); 5483 mPlaybackThreads.removeItem(output); 5484 } 5485 thread->exit(); 5486 // The thread entity (active unit of execution) is no longer running here, 5487 // but the ThreadBase container still exists. 5488 5489 if (thread->type() != ThreadBase::DUPLICATING) { 5490 AudioStreamOut *out = thread->clearOutput(); 5491 ALOG_ASSERT(out != NULL, "out shouldn't be NULL"); 5492 // from now on thread->mOutput is NULL 5493 out->hwDev->close_output_stream(out->hwDev, out->stream); 5494 delete out; 5495 } 5496 return NO_ERROR; 5497} 5498 5499status_t AudioFlinger::suspendOutput(audio_io_handle_t output) 5500{ 5501 Mutex::Autolock _l(mLock); 5502 PlaybackThread *thread = checkPlaybackThread_l(output); 5503 5504 if (thread == NULL) { 5505 return BAD_VALUE; 5506 } 5507 5508 ALOGV("suspendOutput() %d", output); 5509 thread->suspend(); 5510 5511 return NO_ERROR; 5512} 5513 5514status_t AudioFlinger::restoreOutput(audio_io_handle_t output) 5515{ 5516 Mutex::Autolock _l(mLock); 5517 PlaybackThread *thread = checkPlaybackThread_l(output); 5518 5519 if (thread == NULL) { 5520 return BAD_VALUE; 5521 } 5522 5523 ALOGV("restoreOutput() %d", output); 5524 5525 thread->restore(); 5526 5527 return NO_ERROR; 5528} 5529 5530audio_io_handle_t AudioFlinger::openInput(uint32_t *pDevices, 5531 uint32_t *pSamplingRate, 5532 audio_format_t *pFormat, 5533 uint32_t *pChannels, 5534 audio_in_acoustics_t acoustics) 5535{ 5536 status_t status; 5537 RecordThread *thread = NULL; 5538 uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0; 5539 audio_format_t format = pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT; 5540 uint32_t channels = pChannels ? *pChannels : 0; 5541 uint32_t reqSamplingRate = samplingRate; 5542 audio_format_t reqFormat = format; 5543 uint32_t reqChannels = channels; 5544 audio_stream_in_t *inStream; 5545 audio_hw_device_t *inHwDev; 5546 5547 if (pDevices == NULL || *pDevices == 0) { 5548 return 0; 5549 } 5550 5551 Mutex::Autolock _l(mLock); 5552 5553 inHwDev = findSuitableHwDev_l(*pDevices); 5554 if (inHwDev == NULL) 5555 return 0; 5556 5557 status = inHwDev->open_input_stream(inHwDev, *pDevices, &format, 5558 &channels, &samplingRate, 5559 acoustics, 5560 &inStream); 5561 ALOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %d, Channels %x, acoustics %x, status %d", 5562 inStream, 5563 samplingRate, 5564 format, 5565 channels, 5566 acoustics, 5567 status); 5568 5569 // If the input could not be opened with the requested parameters and we can handle the conversion internally, 5570 // try to open again with the proposed parameters. The AudioFlinger can resample the input and do mono to stereo 5571 // or stereo to mono conversions on 16 bit PCM inputs. 5572 if (inStream == NULL && status == BAD_VALUE && 5573 reqFormat == format && format == AUDIO_FORMAT_PCM_16_BIT && 5574 (samplingRate <= 2 * reqSamplingRate) && 5575 (popcount(channels) <= FCC_2) && (popcount(reqChannels) <= FCC_2)) { 5576 ALOGV("openInput() reopening with proposed sampling rate and channels"); 5577 status = inHwDev->open_input_stream(inHwDev, *pDevices, &format, 5578 &channels, &samplingRate, 5579 acoustics, 5580 &inStream); 5581 } 5582 5583 if (inStream != NULL) { 5584 AudioStreamIn *input = new AudioStreamIn(inHwDev, inStream); 5585 5586 audio_io_handle_t id = nextUniqueId(); 5587 // Start record thread 5588 // RecorThread require both input and output device indication to forward to audio 5589 // pre processing modules 5590 uint32_t device = (*pDevices) | primaryOutputDevice_l(); 5591 thread = new RecordThread(this, 5592 input, 5593 reqSamplingRate, 5594 reqChannels, 5595 id, 5596 device); 5597 mRecordThreads.add(id, thread); 5598 ALOGV("openInput() created record thread: ID %d thread %p", id, thread); 5599 if (pSamplingRate != NULL) *pSamplingRate = reqSamplingRate; 5600 if (pFormat != NULL) *pFormat = format; 5601 if (pChannels != NULL) *pChannels = reqChannels; 5602 5603 input->stream->common.standby(&input->stream->common); 5604 5605 // notify client processes of the new input creation 5606 thread->audioConfigChanged_l(AudioSystem::INPUT_OPENED); 5607 return id; 5608 } 5609 5610 return 0; 5611} 5612 5613status_t AudioFlinger::closeInput(audio_io_handle_t input) 5614{ 5615 // keep strong reference on the record thread so that 5616 // it is not destroyed while exit() is executed 5617 sp <RecordThread> thread; 5618 { 5619 Mutex::Autolock _l(mLock); 5620 thread = checkRecordThread_l(input); 5621 if (thread == NULL) { 5622 return BAD_VALUE; 5623 } 5624 5625 ALOGV("closeInput() %d", input); 5626 audioConfigChanged_l(AudioSystem::INPUT_CLOSED, input, NULL); 5627 mRecordThreads.removeItem(input); 5628 } 5629 thread->exit(); 5630 // The thread entity (active unit of execution) is no longer running here, 5631 // but the ThreadBase container still exists. 5632 5633 AudioStreamIn *in = thread->clearInput(); 5634 ALOG_ASSERT(in != NULL, "in shouldn't be NULL"); 5635 // from now on thread->mInput is NULL 5636 in->hwDev->close_input_stream(in->hwDev, in->stream); 5637 delete in; 5638 5639 return NO_ERROR; 5640} 5641 5642status_t AudioFlinger::setStreamOutput(audio_stream_type_t stream, audio_io_handle_t output) 5643{ 5644 Mutex::Autolock _l(mLock); 5645 MixerThread *dstThread = checkMixerThread_l(output); 5646 if (dstThread == NULL) { 5647 ALOGW("setStreamOutput() bad output id %d", output); 5648 return BAD_VALUE; 5649 } 5650 5651 ALOGV("setStreamOutput() stream %d to output %d", stream, output); 5652 audioConfigChanged_l(AudioSystem::STREAM_CONFIG_CHANGED, output, &stream); 5653 5654 dstThread->setStreamValid(stream, true); 5655 5656 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5657 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 5658 if (thread != dstThread && thread->type() != ThreadBase::DIRECT) { 5659 MixerThread *srcThread = (MixerThread *)thread; 5660 srcThread->setStreamValid(stream, false); 5661 srcThread->invalidateTracks(stream); 5662 } 5663 } 5664 5665 return NO_ERROR; 5666} 5667 5668 5669int AudioFlinger::newAudioSessionId() 5670{ 5671 return nextUniqueId(); 5672} 5673 5674void AudioFlinger::acquireAudioSessionId(int audioSession) 5675{ 5676 Mutex::Autolock _l(mLock); 5677 pid_t caller = IPCThreadState::self()->getCallingPid(); 5678 ALOGV("acquiring %d from %d", audioSession, caller); 5679 size_t num = mAudioSessionRefs.size(); 5680 for (size_t i = 0; i< num; i++) { 5681 AudioSessionRef *ref = mAudioSessionRefs.editItemAt(i); 5682 if (ref->mSessionid == audioSession && ref->mPid == caller) { 5683 ref->mCnt++; 5684 ALOGV(" incremented refcount to %d", ref->mCnt); 5685 return; 5686 } 5687 } 5688 mAudioSessionRefs.push(new AudioSessionRef(audioSession, caller)); 5689 ALOGV(" added new entry for %d", audioSession); 5690} 5691 5692void AudioFlinger::releaseAudioSessionId(int audioSession) 5693{ 5694 Mutex::Autolock _l(mLock); 5695 pid_t caller = IPCThreadState::self()->getCallingPid(); 5696 ALOGV("releasing %d from %d", audioSession, caller); 5697 size_t num = mAudioSessionRefs.size(); 5698 for (size_t i = 0; i< num; i++) { 5699 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 5700 if (ref->mSessionid == audioSession && ref->mPid == caller) { 5701 ref->mCnt--; 5702 ALOGV(" decremented refcount to %d", ref->mCnt); 5703 if (ref->mCnt == 0) { 5704 mAudioSessionRefs.removeAt(i); 5705 delete ref; 5706 purgeStaleEffects_l(); 5707 } 5708 return; 5709 } 5710 } 5711 ALOGW("session id %d not found for pid %d", audioSession, caller); 5712} 5713 5714void AudioFlinger::purgeStaleEffects_l() { 5715 5716 ALOGV("purging stale effects"); 5717 5718 Vector< sp<EffectChain> > chains; 5719 5720 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5721 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 5722 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 5723 sp<EffectChain> ec = t->mEffectChains[j]; 5724 if (ec->sessionId() > AUDIO_SESSION_OUTPUT_MIX) { 5725 chains.push(ec); 5726 } 5727 } 5728 } 5729 for (size_t i = 0; i < mRecordThreads.size(); i++) { 5730 sp<RecordThread> t = mRecordThreads.valueAt(i); 5731 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 5732 sp<EffectChain> ec = t->mEffectChains[j]; 5733 chains.push(ec); 5734 } 5735 } 5736 5737 for (size_t i = 0; i < chains.size(); i++) { 5738 sp<EffectChain> ec = chains[i]; 5739 int sessionid = ec->sessionId(); 5740 sp<ThreadBase> t = ec->mThread.promote(); 5741 if (t == 0) { 5742 continue; 5743 } 5744 size_t numsessionrefs = mAudioSessionRefs.size(); 5745 bool found = false; 5746 for (size_t k = 0; k < numsessionrefs; k++) { 5747 AudioSessionRef *ref = mAudioSessionRefs.itemAt(k); 5748 if (ref->mSessionid == sessionid) { 5749 ALOGV(" session %d still exists for %d with %d refs", 5750 sessionid, ref->mPid, ref->mCnt); 5751 found = true; 5752 break; 5753 } 5754 } 5755 if (!found) { 5756 // remove all effects from the chain 5757 while (ec->mEffects.size()) { 5758 sp<EffectModule> effect = ec->mEffects[0]; 5759 effect->unPin(); 5760 Mutex::Autolock _l (t->mLock); 5761 t->removeEffect_l(effect); 5762 for (size_t j = 0; j < effect->mHandles.size(); j++) { 5763 sp<EffectHandle> handle = effect->mHandles[j].promote(); 5764 if (handle != 0) { 5765 handle->mEffect.clear(); 5766 if (handle->mHasControl && handle->mEnabled) { 5767 t->checkSuspendOnEffectEnabled_l(effect, false, effect->sessionId()); 5768 } 5769 } 5770 } 5771 AudioSystem::unregisterEffect(effect->id()); 5772 } 5773 } 5774 } 5775 return; 5776} 5777 5778// checkPlaybackThread_l() must be called with AudioFlinger::mLock held 5779AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(audio_io_handle_t output) const 5780{ 5781 return mPlaybackThreads.valueFor(output).get(); 5782} 5783 5784// checkMixerThread_l() must be called with AudioFlinger::mLock held 5785AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(audio_io_handle_t output) const 5786{ 5787 PlaybackThread *thread = checkPlaybackThread_l(output); 5788 return thread != NULL && thread->type() != ThreadBase::DIRECT ? (MixerThread *) thread : NULL; 5789} 5790 5791// checkRecordThread_l() must be called with AudioFlinger::mLock held 5792AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(audio_io_handle_t input) const 5793{ 5794 return mRecordThreads.valueFor(input).get(); 5795} 5796 5797uint32_t AudioFlinger::nextUniqueId() 5798{ 5799 return android_atomic_inc(&mNextUniqueId); 5800} 5801 5802AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l() const 5803{ 5804 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5805 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 5806 AudioStreamOut *output = thread->getOutput(); 5807 if (output != NULL && output->hwDev == mPrimaryHardwareDev) { 5808 return thread; 5809 } 5810 } 5811 return NULL; 5812} 5813 5814uint32_t AudioFlinger::primaryOutputDevice_l() const 5815{ 5816 PlaybackThread *thread = primaryPlaybackThread_l(); 5817 5818 if (thread == NULL) { 5819 return 0; 5820 } 5821 5822 return thread->device(); 5823} 5824 5825 5826// ---------------------------------------------------------------------------- 5827// Effect management 5828// ---------------------------------------------------------------------------- 5829 5830 5831status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects) const 5832{ 5833 Mutex::Autolock _l(mLock); 5834 return EffectQueryNumberEffects(numEffects); 5835} 5836 5837status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor) const 5838{ 5839 Mutex::Autolock _l(mLock); 5840 return EffectQueryEffect(index, descriptor); 5841} 5842 5843status_t AudioFlinger::getEffectDescriptor(const effect_uuid_t *pUuid, 5844 effect_descriptor_t *descriptor) const 5845{ 5846 Mutex::Autolock _l(mLock); 5847 return EffectGetDescriptor(pUuid, descriptor); 5848} 5849 5850 5851sp<IEffect> AudioFlinger::createEffect(pid_t pid, 5852 effect_descriptor_t *pDesc, 5853 const sp<IEffectClient>& effectClient, 5854 int32_t priority, 5855 audio_io_handle_t io, 5856 int sessionId, 5857 status_t *status, 5858 int *id, 5859 int *enabled) 5860{ 5861 status_t lStatus = NO_ERROR; 5862 sp<EffectHandle> handle; 5863 effect_descriptor_t desc; 5864 5865 ALOGV("createEffect pid %d, effectClient %p, priority %d, sessionId %d, io %d", 5866 pid, effectClient.get(), priority, sessionId, io); 5867 5868 if (pDesc == NULL) { 5869 lStatus = BAD_VALUE; 5870 goto Exit; 5871 } 5872 5873 // check audio settings permission for global effects 5874 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && !settingsAllowed()) { 5875 lStatus = PERMISSION_DENIED; 5876 goto Exit; 5877 } 5878 5879 // Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects 5880 // that can only be created by audio policy manager (running in same process) 5881 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid_cached != pid) { 5882 lStatus = PERMISSION_DENIED; 5883 goto Exit; 5884 } 5885 5886 if (io == 0) { 5887 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) { 5888 // output must be specified by AudioPolicyManager when using session 5889 // AUDIO_SESSION_OUTPUT_STAGE 5890 lStatus = BAD_VALUE; 5891 goto Exit; 5892 } else if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 5893 // if the output returned by getOutputForEffect() is removed before we lock the 5894 // mutex below, the call to checkPlaybackThread_l(io) below will detect it 5895 // and we will exit safely 5896 io = AudioSystem::getOutputForEffect(&desc); 5897 } 5898 } 5899 5900 { 5901 Mutex::Autolock _l(mLock); 5902 5903 5904 if (!EffectIsNullUuid(&pDesc->uuid)) { 5905 // if uuid is specified, request effect descriptor 5906 lStatus = EffectGetDescriptor(&pDesc->uuid, &desc); 5907 if (lStatus < 0) { 5908 ALOGW("createEffect() error %d from EffectGetDescriptor", lStatus); 5909 goto Exit; 5910 } 5911 } else { 5912 // if uuid is not specified, look for an available implementation 5913 // of the required type in effect factory 5914 if (EffectIsNullUuid(&pDesc->type)) { 5915 ALOGW("createEffect() no effect type"); 5916 lStatus = BAD_VALUE; 5917 goto Exit; 5918 } 5919 uint32_t numEffects = 0; 5920 effect_descriptor_t d; 5921 d.flags = 0; // prevent compiler warning 5922 bool found = false; 5923 5924 lStatus = EffectQueryNumberEffects(&numEffects); 5925 if (lStatus < 0) { 5926 ALOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus); 5927 goto Exit; 5928 } 5929 for (uint32_t i = 0; i < numEffects; i++) { 5930 lStatus = EffectQueryEffect(i, &desc); 5931 if (lStatus < 0) { 5932 ALOGW("createEffect() error %d from EffectQueryEffect", lStatus); 5933 continue; 5934 } 5935 if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) { 5936 // If matching type found save effect descriptor. If the session is 5937 // 0 and the effect is not auxiliary, continue enumeration in case 5938 // an auxiliary version of this effect type is available 5939 found = true; 5940 memcpy(&d, &desc, sizeof(effect_descriptor_t)); 5941 if (sessionId != AUDIO_SESSION_OUTPUT_MIX || 5942 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 5943 break; 5944 } 5945 } 5946 } 5947 if (!found) { 5948 lStatus = BAD_VALUE; 5949 ALOGW("createEffect() effect not found"); 5950 goto Exit; 5951 } 5952 // For same effect type, chose auxiliary version over insert version if 5953 // connect to output mix (Compliance to OpenSL ES) 5954 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && 5955 (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) { 5956 memcpy(&desc, &d, sizeof(effect_descriptor_t)); 5957 } 5958 } 5959 5960 // Do not allow auxiliary effects on a session different from 0 (output mix) 5961 if (sessionId != AUDIO_SESSION_OUTPUT_MIX && 5962 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 5963 lStatus = INVALID_OPERATION; 5964 goto Exit; 5965 } 5966 5967 // check recording permission for visualizer 5968 if ((memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) && 5969 !recordingAllowed()) { 5970 lStatus = PERMISSION_DENIED; 5971 goto Exit; 5972 } 5973 5974 // return effect descriptor 5975 memcpy(pDesc, &desc, sizeof(effect_descriptor_t)); 5976 5977 // If output is not specified try to find a matching audio session ID in one of the 5978 // output threads. 5979 // If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX 5980 // because of code checking output when entering the function. 5981 // Note: io is never 0 when creating an effect on an input 5982 if (io == 0) { 5983 // look for the thread where the specified audio session is present 5984 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5985 if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 5986 io = mPlaybackThreads.keyAt(i); 5987 break; 5988 } 5989 } 5990 if (io == 0) { 5991 for (size_t i = 0; i < mRecordThreads.size(); i++) { 5992 if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 5993 io = mRecordThreads.keyAt(i); 5994 break; 5995 } 5996 } 5997 } 5998 // If no output thread contains the requested session ID, default to 5999 // first output. The effect chain will be moved to the correct output 6000 // thread when a track with the same session ID is created 6001 if (io == 0 && mPlaybackThreads.size()) { 6002 io = mPlaybackThreads.keyAt(0); 6003 } 6004 ALOGV("createEffect() got io %d for effect %s", io, desc.name); 6005 } 6006 ThreadBase *thread = checkRecordThread_l(io); 6007 if (thread == NULL) { 6008 thread = checkPlaybackThread_l(io); 6009 if (thread == NULL) { 6010 ALOGE("createEffect() unknown output thread"); 6011 lStatus = BAD_VALUE; 6012 goto Exit; 6013 } 6014 } 6015 6016 sp<Client> client = registerPid_l(pid); 6017 6018 // create effect on selected output thread 6019 handle = thread->createEffect_l(client, effectClient, priority, sessionId, 6020 &desc, enabled, &lStatus); 6021 if (handle != 0 && id != NULL) { 6022 *id = handle->id(); 6023 } 6024 } 6025 6026Exit: 6027 if(status) { 6028 *status = lStatus; 6029 } 6030 return handle; 6031} 6032 6033status_t AudioFlinger::moveEffects(int sessionId, audio_io_handle_t srcOutput, 6034 audio_io_handle_t dstOutput) 6035{ 6036 ALOGV("moveEffects() session %d, srcOutput %d, dstOutput %d", 6037 sessionId, srcOutput, dstOutput); 6038 Mutex::Autolock _l(mLock); 6039 if (srcOutput == dstOutput) { 6040 ALOGW("moveEffects() same dst and src outputs %d", dstOutput); 6041 return NO_ERROR; 6042 } 6043 PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput); 6044 if (srcThread == NULL) { 6045 ALOGW("moveEffects() bad srcOutput %d", srcOutput); 6046 return BAD_VALUE; 6047 } 6048 PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput); 6049 if (dstThread == NULL) { 6050 ALOGW("moveEffects() bad dstOutput %d", dstOutput); 6051 return BAD_VALUE; 6052 } 6053 6054 Mutex::Autolock _dl(dstThread->mLock); 6055 Mutex::Autolock _sl(srcThread->mLock); 6056 moveEffectChain_l(sessionId, srcThread, dstThread, false); 6057 6058 return NO_ERROR; 6059} 6060 6061// moveEffectChain_l must be called with both srcThread and dstThread mLocks held 6062status_t AudioFlinger::moveEffectChain_l(int sessionId, 6063 AudioFlinger::PlaybackThread *srcThread, 6064 AudioFlinger::PlaybackThread *dstThread, 6065 bool reRegister) 6066{ 6067 ALOGV("moveEffectChain_l() session %d from thread %p to thread %p", 6068 sessionId, srcThread, dstThread); 6069 6070 sp<EffectChain> chain = srcThread->getEffectChain_l(sessionId); 6071 if (chain == 0) { 6072 ALOGW("moveEffectChain_l() effect chain for session %d not on source thread %p", 6073 sessionId, srcThread); 6074 return INVALID_OPERATION; 6075 } 6076 6077 // remove chain first. This is useful only if reconfiguring effect chain on same output thread, 6078 // so that a new chain is created with correct parameters when first effect is added. This is 6079 // otherwise unnecessary as removeEffect_l() will remove the chain when last effect is 6080 // removed. 6081 srcThread->removeEffectChain_l(chain); 6082 6083 // transfer all effects one by one so that new effect chain is created on new thread with 6084 // correct buffer sizes and audio parameters and effect engines reconfigured accordingly 6085 audio_io_handle_t dstOutput = dstThread->id(); 6086 sp<EffectChain> dstChain; 6087 uint32_t strategy = 0; // prevent compiler warning 6088 sp<EffectModule> effect = chain->getEffectFromId_l(0); 6089 while (effect != 0) { 6090 srcThread->removeEffect_l(effect); 6091 dstThread->addEffect_l(effect); 6092 // removeEffect_l() has stopped the effect if it was active so it must be restarted 6093 if (effect->state() == EffectModule::ACTIVE || 6094 effect->state() == EffectModule::STOPPING) { 6095 effect->start(); 6096 } 6097 // if the move request is not received from audio policy manager, the effect must be 6098 // re-registered with the new strategy and output 6099 if (dstChain == 0) { 6100 dstChain = effect->chain().promote(); 6101 if (dstChain == 0) { 6102 ALOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get()); 6103 srcThread->addEffect_l(effect); 6104 return NO_INIT; 6105 } 6106 strategy = dstChain->strategy(); 6107 } 6108 if (reRegister) { 6109 AudioSystem::unregisterEffect(effect->id()); 6110 AudioSystem::registerEffect(&effect->desc(), 6111 dstOutput, 6112 strategy, 6113 sessionId, 6114 effect->id()); 6115 } 6116 effect = chain->getEffectFromId_l(0); 6117 } 6118 6119 return NO_ERROR; 6120} 6121 6122 6123// PlaybackThread::createEffect_l() must be called with AudioFlinger::mLock held 6124sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 6125 const sp<AudioFlinger::Client>& client, 6126 const sp<IEffectClient>& effectClient, 6127 int32_t priority, 6128 int sessionId, 6129 effect_descriptor_t *desc, 6130 int *enabled, 6131 status_t *status 6132 ) 6133{ 6134 sp<EffectModule> effect; 6135 sp<EffectHandle> handle; 6136 status_t lStatus; 6137 sp<EffectChain> chain; 6138 bool chainCreated = false; 6139 bool effectCreated = false; 6140 bool effectRegistered = false; 6141 6142 lStatus = initCheck(); 6143 if (lStatus != NO_ERROR) { 6144 ALOGW("createEffect_l() Audio driver not initialized."); 6145 goto Exit; 6146 } 6147 6148 // Do not allow effects with session ID 0 on direct output or duplicating threads 6149 // TODO: add rule for hw accelerated effects on direct outputs with non PCM format 6150 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && mType != MIXER) { 6151 ALOGW("createEffect_l() Cannot add auxiliary effect %s to session %d", 6152 desc->name, sessionId); 6153 lStatus = BAD_VALUE; 6154 goto Exit; 6155 } 6156 // Only Pre processor effects are allowed on input threads and only on input threads 6157 if ((mType == RECORD) != ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { 6158 ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d", 6159 desc->name, desc->flags, mType); 6160 lStatus = BAD_VALUE; 6161 goto Exit; 6162 } 6163 6164 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 6165 6166 { // scope for mLock 6167 Mutex::Autolock _l(mLock); 6168 6169 // check for existing effect chain with the requested audio session 6170 chain = getEffectChain_l(sessionId); 6171 if (chain == 0) { 6172 // create a new chain for this session 6173 ALOGV("createEffect_l() new effect chain for session %d", sessionId); 6174 chain = new EffectChain(this, sessionId); 6175 addEffectChain_l(chain); 6176 chain->setStrategy(getStrategyForSession_l(sessionId)); 6177 chainCreated = true; 6178 } else { 6179 effect = chain->getEffectFromDesc_l(desc); 6180 } 6181 6182 ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get()); 6183 6184 if (effect == 0) { 6185 int id = mAudioFlinger->nextUniqueId(); 6186 // Check CPU and memory usage 6187 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 6188 if (lStatus != NO_ERROR) { 6189 goto Exit; 6190 } 6191 effectRegistered = true; 6192 // create a new effect module if none present in the chain 6193 effect = new EffectModule(this, chain, desc, id, sessionId); 6194 lStatus = effect->status(); 6195 if (lStatus != NO_ERROR) { 6196 goto Exit; 6197 } 6198 lStatus = chain->addEffect_l(effect); 6199 if (lStatus != NO_ERROR) { 6200 goto Exit; 6201 } 6202 effectCreated = true; 6203 6204 effect->setDevice(mDevice); 6205 effect->setMode(mAudioFlinger->getMode()); 6206 } 6207 // create effect handle and connect it to effect module 6208 handle = new EffectHandle(effect, client, effectClient, priority); 6209 lStatus = effect->addHandle(handle); 6210 if (enabled != NULL) { 6211 *enabled = (int)effect->isEnabled(); 6212 } 6213 } 6214 6215Exit: 6216 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 6217 Mutex::Autolock _l(mLock); 6218 if (effectCreated) { 6219 chain->removeEffect_l(effect); 6220 } 6221 if (effectRegistered) { 6222 AudioSystem::unregisterEffect(effect->id()); 6223 } 6224 if (chainCreated) { 6225 removeEffectChain_l(chain); 6226 } 6227 handle.clear(); 6228 } 6229 6230 if(status) { 6231 *status = lStatus; 6232 } 6233 return handle; 6234} 6235 6236sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId) 6237{ 6238 sp<EffectChain> chain = getEffectChain_l(sessionId); 6239 return chain != 0 ? chain->getEffectFromId_l(effectId) : 0; 6240} 6241 6242// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 6243// PlaybackThread::mLock held 6244status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 6245{ 6246 // check for existing effect chain with the requested audio session 6247 int sessionId = effect->sessionId(); 6248 sp<EffectChain> chain = getEffectChain_l(sessionId); 6249 bool chainCreated = false; 6250 6251 if (chain == 0) { 6252 // create a new chain for this session 6253 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 6254 chain = new EffectChain(this, sessionId); 6255 addEffectChain_l(chain); 6256 chain->setStrategy(getStrategyForSession_l(sessionId)); 6257 chainCreated = true; 6258 } 6259 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 6260 6261 if (chain->getEffectFromId_l(effect->id()) != 0) { 6262 ALOGW("addEffect_l() %p effect %s already present in chain %p", 6263 this, effect->desc().name, chain.get()); 6264 return BAD_VALUE; 6265 } 6266 6267 status_t status = chain->addEffect_l(effect); 6268 if (status != NO_ERROR) { 6269 if (chainCreated) { 6270 removeEffectChain_l(chain); 6271 } 6272 return status; 6273 } 6274 6275 effect->setDevice(mDevice); 6276 effect->setMode(mAudioFlinger->getMode()); 6277 return NO_ERROR; 6278} 6279 6280void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 6281 6282 ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 6283 effect_descriptor_t desc = effect->desc(); 6284 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6285 detachAuxEffect_l(effect->id()); 6286 } 6287 6288 sp<EffectChain> chain = effect->chain().promote(); 6289 if (chain != 0) { 6290 // remove effect chain if removing last effect 6291 if (chain->removeEffect_l(effect) == 0) { 6292 removeEffectChain_l(chain); 6293 } 6294 } else { 6295 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 6296 } 6297} 6298 6299void AudioFlinger::ThreadBase::lockEffectChains_l( 6300 Vector<sp <AudioFlinger::EffectChain> >& effectChains) 6301{ 6302 effectChains = mEffectChains; 6303 for (size_t i = 0; i < mEffectChains.size(); i++) { 6304 mEffectChains[i]->lock(); 6305 } 6306} 6307 6308void AudioFlinger::ThreadBase::unlockEffectChains( 6309 const Vector<sp <AudioFlinger::EffectChain> >& effectChains) 6310{ 6311 for (size_t i = 0; i < effectChains.size(); i++) { 6312 effectChains[i]->unlock(); 6313 } 6314} 6315 6316sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId) 6317{ 6318 Mutex::Autolock _l(mLock); 6319 return getEffectChain_l(sessionId); 6320} 6321 6322sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) 6323{ 6324 size_t size = mEffectChains.size(); 6325 for (size_t i = 0; i < size; i++) { 6326 if (mEffectChains[i]->sessionId() == sessionId) { 6327 return mEffectChains[i]; 6328 } 6329 } 6330 return 0; 6331} 6332 6333void AudioFlinger::ThreadBase::setMode(audio_mode_t mode) 6334{ 6335 Mutex::Autolock _l(mLock); 6336 size_t size = mEffectChains.size(); 6337 for (size_t i = 0; i < size; i++) { 6338 mEffectChains[i]->setMode_l(mode); 6339 } 6340} 6341 6342void AudioFlinger::ThreadBase::disconnectEffect(const sp<EffectModule>& effect, 6343 const wp<EffectHandle>& handle, 6344 bool unpinIfLast) { 6345 6346 Mutex::Autolock _l(mLock); 6347 ALOGV("disconnectEffect() %p effect %p", this, effect.get()); 6348 // delete the effect module if removing last handle on it 6349 if (effect->removeHandle(handle) == 0) { 6350 if (!effect->isPinned() || unpinIfLast) { 6351 removeEffect_l(effect); 6352 AudioSystem::unregisterEffect(effect->id()); 6353 } 6354 } 6355} 6356 6357status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 6358{ 6359 int session = chain->sessionId(); 6360 int16_t *buffer = mMixBuffer; 6361 bool ownsBuffer = false; 6362 6363 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 6364 if (session > 0) { 6365 // Only one effect chain can be present in direct output thread and it uses 6366 // the mix buffer as input 6367 if (mType != DIRECT) { 6368 size_t numSamples = mFrameCount * mChannelCount; 6369 buffer = new int16_t[numSamples]; 6370 memset(buffer, 0, numSamples * sizeof(int16_t)); 6371 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 6372 ownsBuffer = true; 6373 } 6374 6375 // Attach all tracks with same session ID to this chain. 6376 for (size_t i = 0; i < mTracks.size(); ++i) { 6377 sp<Track> track = mTracks[i]; 6378 if (session == track->sessionId()) { 6379 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), buffer); 6380 track->setMainBuffer(buffer); 6381 chain->incTrackCnt(); 6382 } 6383 } 6384 6385 // indicate all active tracks in the chain 6386 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 6387 sp<Track> track = mActiveTracks[i].promote(); 6388 if (track == 0) continue; 6389 if (session == track->sessionId()) { 6390 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 6391 chain->incActiveTrackCnt(); 6392 } 6393 } 6394 } 6395 6396 chain->setInBuffer(buffer, ownsBuffer); 6397 chain->setOutBuffer(mMixBuffer); 6398 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 6399 // chains list in order to be processed last as it contains output stage effects 6400 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 6401 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 6402 // after track specific effects and before output stage 6403 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 6404 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 6405 // Effect chain for other sessions are inserted at beginning of effect 6406 // chains list to be processed before output mix effects. Relative order between other 6407 // sessions is not important 6408 size_t size = mEffectChains.size(); 6409 size_t i = 0; 6410 for (i = 0; i < size; i++) { 6411 if (mEffectChains[i]->sessionId() < session) break; 6412 } 6413 mEffectChains.insertAt(chain, i); 6414 checkSuspendOnAddEffectChain_l(chain); 6415 6416 return NO_ERROR; 6417} 6418 6419size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 6420{ 6421 int session = chain->sessionId(); 6422 6423 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 6424 6425 for (size_t i = 0; i < mEffectChains.size(); i++) { 6426 if (chain == mEffectChains[i]) { 6427 mEffectChains.removeAt(i); 6428 // detach all active tracks from the chain 6429 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 6430 sp<Track> track = mActiveTracks[i].promote(); 6431 if (track == 0) continue; 6432 if (session == track->sessionId()) { 6433 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 6434 chain.get(), session); 6435 chain->decActiveTrackCnt(); 6436 } 6437 } 6438 6439 // detach all tracks with same session ID from this chain 6440 for (size_t i = 0; i < mTracks.size(); ++i) { 6441 sp<Track> track = mTracks[i]; 6442 if (session == track->sessionId()) { 6443 track->setMainBuffer(mMixBuffer); 6444 chain->decTrackCnt(); 6445 } 6446 } 6447 break; 6448 } 6449 } 6450 return mEffectChains.size(); 6451} 6452 6453status_t AudioFlinger::PlaybackThread::attachAuxEffect( 6454 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 6455{ 6456 Mutex::Autolock _l(mLock); 6457 return attachAuxEffect_l(track, EffectId); 6458} 6459 6460status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 6461 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 6462{ 6463 status_t status = NO_ERROR; 6464 6465 if (EffectId == 0) { 6466 track->setAuxBuffer(0, NULL); 6467 } else { 6468 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 6469 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 6470 if (effect != 0) { 6471 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6472 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 6473 } else { 6474 status = INVALID_OPERATION; 6475 } 6476 } else { 6477 status = BAD_VALUE; 6478 } 6479 } 6480 return status; 6481} 6482 6483void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 6484{ 6485 for (size_t i = 0; i < mTracks.size(); ++i) { 6486 sp<Track> track = mTracks[i]; 6487 if (track->auxEffectId() == effectId) { 6488 attachAuxEffect_l(track, 0); 6489 } 6490 } 6491} 6492 6493status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 6494{ 6495 // only one chain per input thread 6496 if (mEffectChains.size() != 0) { 6497 return INVALID_OPERATION; 6498 } 6499 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 6500 6501 chain->setInBuffer(NULL); 6502 chain->setOutBuffer(NULL); 6503 6504 checkSuspendOnAddEffectChain_l(chain); 6505 6506 mEffectChains.add(chain); 6507 6508 return NO_ERROR; 6509} 6510 6511size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 6512{ 6513 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 6514 ALOGW_IF(mEffectChains.size() != 1, 6515 "removeEffectChain_l() %p invalid chain size %d on thread %p", 6516 chain.get(), mEffectChains.size(), this); 6517 if (mEffectChains.size() == 1) { 6518 mEffectChains.removeAt(0); 6519 } 6520 return 0; 6521} 6522 6523// ---------------------------------------------------------------------------- 6524// EffectModule implementation 6525// ---------------------------------------------------------------------------- 6526 6527#undef LOG_TAG 6528#define LOG_TAG "AudioFlinger::EffectModule" 6529 6530AudioFlinger::EffectModule::EffectModule(ThreadBase *thread, 6531 const wp<AudioFlinger::EffectChain>& chain, 6532 effect_descriptor_t *desc, 6533 int id, 6534 int sessionId) 6535 : mThread(thread), mChain(chain), mId(id), mSessionId(sessionId), mEffectInterface(NULL), 6536 mStatus(NO_INIT), mState(IDLE), mSuspended(false) 6537{ 6538 ALOGV("Constructor %p", this); 6539 int lStatus; 6540 if (thread == NULL) { 6541 return; 6542 } 6543 6544 memcpy(&mDescriptor, desc, sizeof(effect_descriptor_t)); 6545 6546 // create effect engine from effect factory 6547 mStatus = EffectCreate(&desc->uuid, sessionId, thread->id(), &mEffectInterface); 6548 6549 if (mStatus != NO_ERROR) { 6550 return; 6551 } 6552 lStatus = init(); 6553 if (lStatus < 0) { 6554 mStatus = lStatus; 6555 goto Error; 6556 } 6557 6558 if (mSessionId > AUDIO_SESSION_OUTPUT_MIX) { 6559 mPinned = true; 6560 } 6561 ALOGV("Constructor success name %s, Interface %p", mDescriptor.name, mEffectInterface); 6562 return; 6563Error: 6564 EffectRelease(mEffectInterface); 6565 mEffectInterface = NULL; 6566 ALOGV("Constructor Error %d", mStatus); 6567} 6568 6569AudioFlinger::EffectModule::~EffectModule() 6570{ 6571 ALOGV("Destructor %p", this); 6572 if (mEffectInterface != NULL) { 6573 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 6574 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC) { 6575 sp<ThreadBase> thread = mThread.promote(); 6576 if (thread != 0) { 6577 audio_stream_t *stream = thread->stream(); 6578 if (stream != NULL) { 6579 stream->remove_audio_effect(stream, mEffectInterface); 6580 } 6581 } 6582 } 6583 // release effect engine 6584 EffectRelease(mEffectInterface); 6585 } 6586} 6587 6588status_t AudioFlinger::EffectModule::addHandle(const sp<EffectHandle>& handle) 6589{ 6590 status_t status; 6591 6592 Mutex::Autolock _l(mLock); 6593 int priority = handle->priority(); 6594 size_t size = mHandles.size(); 6595 sp<EffectHandle> h; 6596 size_t i; 6597 for (i = 0; i < size; i++) { 6598 h = mHandles[i].promote(); 6599 if (h == 0) continue; 6600 if (h->priority() <= priority) break; 6601 } 6602 // if inserted in first place, move effect control from previous owner to this handle 6603 if (i == 0) { 6604 bool enabled = false; 6605 if (h != 0) { 6606 enabled = h->enabled(); 6607 h->setControl(false/*hasControl*/, true /*signal*/, enabled /*enabled*/); 6608 } 6609 handle->setControl(true /*hasControl*/, false /*signal*/, enabled /*enabled*/); 6610 status = NO_ERROR; 6611 } else { 6612 status = ALREADY_EXISTS; 6613 } 6614 ALOGV("addHandle() %p added handle %p in position %d", this, handle.get(), i); 6615 mHandles.insertAt(handle, i); 6616 return status; 6617} 6618 6619size_t AudioFlinger::EffectModule::removeHandle(const wp<EffectHandle>& handle) 6620{ 6621 Mutex::Autolock _l(mLock); 6622 size_t size = mHandles.size(); 6623 size_t i; 6624 for (i = 0; i < size; i++) { 6625 if (mHandles[i] == handle) break; 6626 } 6627 if (i == size) { 6628 return size; 6629 } 6630 ALOGV("removeHandle() %p removed handle %p in position %d", this, handle.unsafe_get(), i); 6631 6632 bool enabled = false; 6633 EffectHandle *hdl = handle.unsafe_get(); 6634 if (hdl != NULL) { 6635 ALOGV("removeHandle() unsafe_get OK"); 6636 enabled = hdl->enabled(); 6637 } 6638 mHandles.removeAt(i); 6639 size = mHandles.size(); 6640 // if removed from first place, move effect control from this handle to next in line 6641 if (i == 0 && size != 0) { 6642 sp<EffectHandle> h = mHandles[0].promote(); 6643 if (h != 0) { 6644 h->setControl(true /*hasControl*/, true /*signal*/ , enabled /*enabled*/); 6645 } 6646 } 6647 6648 // Prevent calls to process() and other functions on effect interface from now on. 6649 // The effect engine will be released by the destructor when the last strong reference on 6650 // this object is released which can happen after next process is called. 6651 if (size == 0 && !mPinned) { 6652 mState = DESTROYED; 6653 } 6654 6655 return size; 6656} 6657 6658sp<AudioFlinger::EffectHandle> AudioFlinger::EffectModule::controlHandle() 6659{ 6660 Mutex::Autolock _l(mLock); 6661 return mHandles.size() != 0 ? mHandles[0].promote() : 0; 6662} 6663 6664void AudioFlinger::EffectModule::disconnect(const wp<EffectHandle>& handle, bool unpinIfLast) 6665{ 6666 ALOGV("disconnect() %p handle %p", this, handle.unsafe_get()); 6667 // keep a strong reference on this EffectModule to avoid calling the 6668 // destructor before we exit 6669 sp<EffectModule> keep(this); 6670 { 6671 sp<ThreadBase> thread = mThread.promote(); 6672 if (thread != 0) { 6673 thread->disconnectEffect(keep, handle, unpinIfLast); 6674 } 6675 } 6676} 6677 6678void AudioFlinger::EffectModule::updateState() { 6679 Mutex::Autolock _l(mLock); 6680 6681 switch (mState) { 6682 case RESTART: 6683 reset_l(); 6684 // FALL THROUGH 6685 6686 case STARTING: 6687 // clear auxiliary effect input buffer for next accumulation 6688 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6689 memset(mConfig.inputCfg.buffer.raw, 6690 0, 6691 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 6692 } 6693 start_l(); 6694 mState = ACTIVE; 6695 break; 6696 case STOPPING: 6697 stop_l(); 6698 mDisableWaitCnt = mMaxDisableWaitCnt; 6699 mState = STOPPED; 6700 break; 6701 case STOPPED: 6702 // mDisableWaitCnt is forced to 1 by process() when the engine indicates the end of the 6703 // turn off sequence. 6704 if (--mDisableWaitCnt == 0) { 6705 reset_l(); 6706 mState = IDLE; 6707 } 6708 break; 6709 default: //IDLE , ACTIVE, DESTROYED 6710 break; 6711 } 6712} 6713 6714void AudioFlinger::EffectModule::process() 6715{ 6716 Mutex::Autolock _l(mLock); 6717 6718 if (mState == DESTROYED || mEffectInterface == NULL || 6719 mConfig.inputCfg.buffer.raw == NULL || 6720 mConfig.outputCfg.buffer.raw == NULL) { 6721 return; 6722 } 6723 6724 if (isProcessEnabled()) { 6725 // do 32 bit to 16 bit conversion for auxiliary effect input buffer 6726 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6727 ditherAndClamp(mConfig.inputCfg.buffer.s32, 6728 mConfig.inputCfg.buffer.s32, 6729 mConfig.inputCfg.buffer.frameCount/2); 6730 } 6731 6732 // do the actual processing in the effect engine 6733 int ret = (*mEffectInterface)->process(mEffectInterface, 6734 &mConfig.inputCfg.buffer, 6735 &mConfig.outputCfg.buffer); 6736 6737 // force transition to IDLE state when engine is ready 6738 if (mState == STOPPED && ret == -ENODATA) { 6739 mDisableWaitCnt = 1; 6740 } 6741 6742 // clear auxiliary effect input buffer for next accumulation 6743 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6744 memset(mConfig.inputCfg.buffer.raw, 0, 6745 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 6746 } 6747 } else if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_INSERT && 6748 mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 6749 // If an insert effect is idle and input buffer is different from output buffer, 6750 // accumulate input onto output 6751 sp<EffectChain> chain = mChain.promote(); 6752 if (chain != 0 && chain->activeTrackCnt() != 0) { 6753 size_t frameCnt = mConfig.inputCfg.buffer.frameCount * 2; //always stereo here 6754 int16_t *in = mConfig.inputCfg.buffer.s16; 6755 int16_t *out = mConfig.outputCfg.buffer.s16; 6756 for (size_t i = 0; i < frameCnt; i++) { 6757 out[i] = clamp16((int32_t)out[i] + (int32_t)in[i]); 6758 } 6759 } 6760 } 6761} 6762 6763void AudioFlinger::EffectModule::reset_l() 6764{ 6765 if (mEffectInterface == NULL) { 6766 return; 6767 } 6768 (*mEffectInterface)->command(mEffectInterface, EFFECT_CMD_RESET, 0, NULL, 0, NULL); 6769} 6770 6771status_t AudioFlinger::EffectModule::configure() 6772{ 6773 uint32_t channels; 6774 if (mEffectInterface == NULL) { 6775 return NO_INIT; 6776 } 6777 6778 sp<ThreadBase> thread = mThread.promote(); 6779 if (thread == 0) { 6780 return DEAD_OBJECT; 6781 } 6782 6783 // TODO: handle configuration of effects replacing track process 6784 if (thread->channelCount() == 1) { 6785 channels = AUDIO_CHANNEL_OUT_MONO; 6786 } else { 6787 channels = AUDIO_CHANNEL_OUT_STEREO; 6788 } 6789 6790 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6791 mConfig.inputCfg.channels = AUDIO_CHANNEL_OUT_MONO; 6792 } else { 6793 mConfig.inputCfg.channels = channels; 6794 } 6795 mConfig.outputCfg.channels = channels; 6796 mConfig.inputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 6797 mConfig.outputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 6798 mConfig.inputCfg.samplingRate = thread->sampleRate(); 6799 mConfig.outputCfg.samplingRate = mConfig.inputCfg.samplingRate; 6800 mConfig.inputCfg.bufferProvider.cookie = NULL; 6801 mConfig.inputCfg.bufferProvider.getBuffer = NULL; 6802 mConfig.inputCfg.bufferProvider.releaseBuffer = NULL; 6803 mConfig.outputCfg.bufferProvider.cookie = NULL; 6804 mConfig.outputCfg.bufferProvider.getBuffer = NULL; 6805 mConfig.outputCfg.bufferProvider.releaseBuffer = NULL; 6806 mConfig.inputCfg.accessMode = EFFECT_BUFFER_ACCESS_READ; 6807 // Insert effect: 6808 // - in session AUDIO_SESSION_OUTPUT_MIX or AUDIO_SESSION_OUTPUT_STAGE, 6809 // always overwrites output buffer: input buffer == output buffer 6810 // - in other sessions: 6811 // last effect in the chain accumulates in output buffer: input buffer != output buffer 6812 // other effect: overwrites output buffer: input buffer == output buffer 6813 // Auxiliary effect: 6814 // accumulates in output buffer: input buffer != output buffer 6815 // Therefore: accumulate <=> input buffer != output buffer 6816 if (mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 6817 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_ACCUMULATE; 6818 } else { 6819 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_WRITE; 6820 } 6821 mConfig.inputCfg.mask = EFFECT_CONFIG_ALL; 6822 mConfig.outputCfg.mask = EFFECT_CONFIG_ALL; 6823 mConfig.inputCfg.buffer.frameCount = thread->frameCount(); 6824 mConfig.outputCfg.buffer.frameCount = mConfig.inputCfg.buffer.frameCount; 6825 6826 ALOGV("configure() %p thread %p buffer %p framecount %d", 6827 this, thread.get(), mConfig.inputCfg.buffer.raw, mConfig.inputCfg.buffer.frameCount); 6828 6829 status_t cmdStatus; 6830 uint32_t size = sizeof(int); 6831 status_t status = (*mEffectInterface)->command(mEffectInterface, 6832 EFFECT_CMD_SET_CONFIG, 6833 sizeof(effect_config_t), 6834 &mConfig, 6835 &size, 6836 &cmdStatus); 6837 if (status == 0) { 6838 status = cmdStatus; 6839 } 6840 6841 mMaxDisableWaitCnt = (MAX_DISABLE_TIME_MS * mConfig.outputCfg.samplingRate) / 6842 (1000 * mConfig.outputCfg.buffer.frameCount); 6843 6844 return status; 6845} 6846 6847status_t AudioFlinger::EffectModule::init() 6848{ 6849 Mutex::Autolock _l(mLock); 6850 if (mEffectInterface == NULL) { 6851 return NO_INIT; 6852 } 6853 status_t cmdStatus; 6854 uint32_t size = sizeof(status_t); 6855 status_t status = (*mEffectInterface)->command(mEffectInterface, 6856 EFFECT_CMD_INIT, 6857 0, 6858 NULL, 6859 &size, 6860 &cmdStatus); 6861 if (status == 0) { 6862 status = cmdStatus; 6863 } 6864 return status; 6865} 6866 6867status_t AudioFlinger::EffectModule::start() 6868{ 6869 Mutex::Autolock _l(mLock); 6870 return start_l(); 6871} 6872 6873status_t AudioFlinger::EffectModule::start_l() 6874{ 6875 if (mEffectInterface == NULL) { 6876 return NO_INIT; 6877 } 6878 status_t cmdStatus; 6879 uint32_t size = sizeof(status_t); 6880 status_t status = (*mEffectInterface)->command(mEffectInterface, 6881 EFFECT_CMD_ENABLE, 6882 0, 6883 NULL, 6884 &size, 6885 &cmdStatus); 6886 if (status == 0) { 6887 status = cmdStatus; 6888 } 6889 if (status == 0 && 6890 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 6891 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 6892 sp<ThreadBase> thread = mThread.promote(); 6893 if (thread != 0) { 6894 audio_stream_t *stream = thread->stream(); 6895 if (stream != NULL) { 6896 stream->add_audio_effect(stream, mEffectInterface); 6897 } 6898 } 6899 } 6900 return status; 6901} 6902 6903status_t AudioFlinger::EffectModule::stop() 6904{ 6905 Mutex::Autolock _l(mLock); 6906 return stop_l(); 6907} 6908 6909status_t AudioFlinger::EffectModule::stop_l() 6910{ 6911 if (mEffectInterface == NULL) { 6912 return NO_INIT; 6913 } 6914 status_t cmdStatus; 6915 uint32_t size = sizeof(status_t); 6916 status_t status = (*mEffectInterface)->command(mEffectInterface, 6917 EFFECT_CMD_DISABLE, 6918 0, 6919 NULL, 6920 &size, 6921 &cmdStatus); 6922 if (status == 0) { 6923 status = cmdStatus; 6924 } 6925 if (status == 0 && 6926 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 6927 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 6928 sp<ThreadBase> thread = mThread.promote(); 6929 if (thread != 0) { 6930 audio_stream_t *stream = thread->stream(); 6931 if (stream != NULL) { 6932 stream->remove_audio_effect(stream, mEffectInterface); 6933 } 6934 } 6935 } 6936 return status; 6937} 6938 6939status_t AudioFlinger::EffectModule::command(uint32_t cmdCode, 6940 uint32_t cmdSize, 6941 void *pCmdData, 6942 uint32_t *replySize, 6943 void *pReplyData) 6944{ 6945 Mutex::Autolock _l(mLock); 6946// ALOGV("command(), cmdCode: %d, mEffectInterface: %p", cmdCode, mEffectInterface); 6947 6948 if (mState == DESTROYED || mEffectInterface == NULL) { 6949 return NO_INIT; 6950 } 6951 status_t status = (*mEffectInterface)->command(mEffectInterface, 6952 cmdCode, 6953 cmdSize, 6954 pCmdData, 6955 replySize, 6956 pReplyData); 6957 if (cmdCode != EFFECT_CMD_GET_PARAM && status == NO_ERROR) { 6958 uint32_t size = (replySize == NULL) ? 0 : *replySize; 6959 for (size_t i = 1; i < mHandles.size(); i++) { 6960 sp<EffectHandle> h = mHandles[i].promote(); 6961 if (h != 0) { 6962 h->commandExecuted(cmdCode, cmdSize, pCmdData, size, pReplyData); 6963 } 6964 } 6965 } 6966 return status; 6967} 6968 6969status_t AudioFlinger::EffectModule::setEnabled(bool enabled) 6970{ 6971 6972 Mutex::Autolock _l(mLock); 6973 ALOGV("setEnabled %p enabled %d", this, enabled); 6974 6975 if (enabled != isEnabled()) { 6976 status_t status = AudioSystem::setEffectEnabled(mId, enabled); 6977 if (enabled && status != NO_ERROR) { 6978 return status; 6979 } 6980 6981 switch (mState) { 6982 // going from disabled to enabled 6983 case IDLE: 6984 mState = STARTING; 6985 break; 6986 case STOPPED: 6987 mState = RESTART; 6988 break; 6989 case STOPPING: 6990 mState = ACTIVE; 6991 break; 6992 6993 // going from enabled to disabled 6994 case RESTART: 6995 mState = STOPPED; 6996 break; 6997 case STARTING: 6998 mState = IDLE; 6999 break; 7000 case ACTIVE: 7001 mState = STOPPING; 7002 break; 7003 case DESTROYED: 7004 return NO_ERROR; // simply ignore as we are being destroyed 7005 } 7006 for (size_t i = 1; i < mHandles.size(); i++) { 7007 sp<EffectHandle> h = mHandles[i].promote(); 7008 if (h != 0) { 7009 h->setEnabled(enabled); 7010 } 7011 } 7012 } 7013 return NO_ERROR; 7014} 7015 7016bool AudioFlinger::EffectModule::isEnabled() const 7017{ 7018 switch (mState) { 7019 case RESTART: 7020 case STARTING: 7021 case ACTIVE: 7022 return true; 7023 case IDLE: 7024 case STOPPING: 7025 case STOPPED: 7026 case DESTROYED: 7027 default: 7028 return false; 7029 } 7030} 7031 7032bool AudioFlinger::EffectModule::isProcessEnabled() const 7033{ 7034 switch (mState) { 7035 case RESTART: 7036 case ACTIVE: 7037 case STOPPING: 7038 case STOPPED: 7039 return true; 7040 case IDLE: 7041 case STARTING: 7042 case DESTROYED: 7043 default: 7044 return false; 7045 } 7046} 7047 7048status_t AudioFlinger::EffectModule::setVolume(uint32_t *left, uint32_t *right, bool controller) 7049{ 7050 Mutex::Autolock _l(mLock); 7051 status_t status = NO_ERROR; 7052 7053 // Send volume indication if EFFECT_FLAG_VOLUME_IND is set and read back altered volume 7054 // if controller flag is set (Note that controller == TRUE => EFFECT_FLAG_VOLUME_CTRL set) 7055 if (isProcessEnabled() && 7056 ((mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL || 7057 (mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_IND)) { 7058 status_t cmdStatus; 7059 uint32_t volume[2]; 7060 uint32_t *pVolume = NULL; 7061 uint32_t size = sizeof(volume); 7062 volume[0] = *left; 7063 volume[1] = *right; 7064 if (controller) { 7065 pVolume = volume; 7066 } 7067 status = (*mEffectInterface)->command(mEffectInterface, 7068 EFFECT_CMD_SET_VOLUME, 7069 size, 7070 volume, 7071 &size, 7072 pVolume); 7073 if (controller && status == NO_ERROR && size == sizeof(volume)) { 7074 *left = volume[0]; 7075 *right = volume[1]; 7076 } 7077 } 7078 return status; 7079} 7080 7081status_t AudioFlinger::EffectModule::setDevice(uint32_t device) 7082{ 7083 Mutex::Autolock _l(mLock); 7084 status_t status = NO_ERROR; 7085 if (device && (mDescriptor.flags & EFFECT_FLAG_DEVICE_MASK) == EFFECT_FLAG_DEVICE_IND) { 7086 // audio pre processing modules on RecordThread can receive both output and 7087 // input device indication in the same call 7088 uint32_t dev = device & AUDIO_DEVICE_OUT_ALL; 7089 if (dev) { 7090 status_t cmdStatus; 7091 uint32_t size = sizeof(status_t); 7092 7093 status = (*mEffectInterface)->command(mEffectInterface, 7094 EFFECT_CMD_SET_DEVICE, 7095 sizeof(uint32_t), 7096 &dev, 7097 &size, 7098 &cmdStatus); 7099 if (status == NO_ERROR) { 7100 status = cmdStatus; 7101 } 7102 } 7103 dev = device & AUDIO_DEVICE_IN_ALL; 7104 if (dev) { 7105 status_t cmdStatus; 7106 uint32_t size = sizeof(status_t); 7107 7108 status_t status2 = (*mEffectInterface)->command(mEffectInterface, 7109 EFFECT_CMD_SET_INPUT_DEVICE, 7110 sizeof(uint32_t), 7111 &dev, 7112 &size, 7113 &cmdStatus); 7114 if (status2 == NO_ERROR) { 7115 status2 = cmdStatus; 7116 } 7117 if (status == NO_ERROR) { 7118 status = status2; 7119 } 7120 } 7121 } 7122 return status; 7123} 7124 7125status_t AudioFlinger::EffectModule::setMode(audio_mode_t mode) 7126{ 7127 Mutex::Autolock _l(mLock); 7128 status_t status = NO_ERROR; 7129 if ((mDescriptor.flags & EFFECT_FLAG_AUDIO_MODE_MASK) == EFFECT_FLAG_AUDIO_MODE_IND) { 7130 status_t cmdStatus; 7131 uint32_t size = sizeof(status_t); 7132 status = (*mEffectInterface)->command(mEffectInterface, 7133 EFFECT_CMD_SET_AUDIO_MODE, 7134 sizeof(audio_mode_t), 7135 &mode, 7136 &size, 7137 &cmdStatus); 7138 if (status == NO_ERROR) { 7139 status = cmdStatus; 7140 } 7141 } 7142 return status; 7143} 7144 7145void AudioFlinger::EffectModule::setSuspended(bool suspended) 7146{ 7147 Mutex::Autolock _l(mLock); 7148 mSuspended = suspended; 7149} 7150 7151bool AudioFlinger::EffectModule::suspended() const 7152{ 7153 Mutex::Autolock _l(mLock); 7154 return mSuspended; 7155} 7156 7157status_t AudioFlinger::EffectModule::dump(int fd, const Vector<String16>& args) 7158{ 7159 const size_t SIZE = 256; 7160 char buffer[SIZE]; 7161 String8 result; 7162 7163 snprintf(buffer, SIZE, "\tEffect ID %d:\n", mId); 7164 result.append(buffer); 7165 7166 bool locked = tryLock(mLock); 7167 // failed to lock - AudioFlinger is probably deadlocked 7168 if (!locked) { 7169 result.append("\t\tCould not lock Fx mutex:\n"); 7170 } 7171 7172 result.append("\t\tSession Status State Engine:\n"); 7173 snprintf(buffer, SIZE, "\t\t%05d %03d %03d 0x%08x\n", 7174 mSessionId, mStatus, mState, (uint32_t)mEffectInterface); 7175 result.append(buffer); 7176 7177 result.append("\t\tDescriptor:\n"); 7178 snprintf(buffer, SIZE, "\t\t- UUID: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 7179 mDescriptor.uuid.timeLow, mDescriptor.uuid.timeMid, mDescriptor.uuid.timeHiAndVersion, 7180 mDescriptor.uuid.clockSeq, mDescriptor.uuid.node[0], mDescriptor.uuid.node[1],mDescriptor.uuid.node[2], 7181 mDescriptor.uuid.node[3],mDescriptor.uuid.node[4],mDescriptor.uuid.node[5]); 7182 result.append(buffer); 7183 snprintf(buffer, SIZE, "\t\t- TYPE: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 7184 mDescriptor.type.timeLow, mDescriptor.type.timeMid, mDescriptor.type.timeHiAndVersion, 7185 mDescriptor.type.clockSeq, mDescriptor.type.node[0], mDescriptor.type.node[1],mDescriptor.type.node[2], 7186 mDescriptor.type.node[3],mDescriptor.type.node[4],mDescriptor.type.node[5]); 7187 result.append(buffer); 7188 snprintf(buffer, SIZE, "\t\t- apiVersion: %08X\n\t\t- flags: %08X\n", 7189 mDescriptor.apiVersion, 7190 mDescriptor.flags); 7191 result.append(buffer); 7192 snprintf(buffer, SIZE, "\t\t- name: %s\n", 7193 mDescriptor.name); 7194 result.append(buffer); 7195 snprintf(buffer, SIZE, "\t\t- implementor: %s\n", 7196 mDescriptor.implementor); 7197 result.append(buffer); 7198 7199 result.append("\t\t- Input configuration:\n"); 7200 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 7201 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 7202 (uint32_t)mConfig.inputCfg.buffer.raw, 7203 mConfig.inputCfg.buffer.frameCount, 7204 mConfig.inputCfg.samplingRate, 7205 mConfig.inputCfg.channels, 7206 mConfig.inputCfg.format); 7207 result.append(buffer); 7208 7209 result.append("\t\t- Output configuration:\n"); 7210 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 7211 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 7212 (uint32_t)mConfig.outputCfg.buffer.raw, 7213 mConfig.outputCfg.buffer.frameCount, 7214 mConfig.outputCfg.samplingRate, 7215 mConfig.outputCfg.channels, 7216 mConfig.outputCfg.format); 7217 result.append(buffer); 7218 7219 snprintf(buffer, SIZE, "\t\t%d Clients:\n", mHandles.size()); 7220 result.append(buffer); 7221 result.append("\t\t\tPid Priority Ctrl Locked client server\n"); 7222 for (size_t i = 0; i < mHandles.size(); ++i) { 7223 sp<EffectHandle> handle = mHandles[i].promote(); 7224 if (handle != 0) { 7225 handle->dump(buffer, SIZE); 7226 result.append(buffer); 7227 } 7228 } 7229 7230 result.append("\n"); 7231 7232 write(fd, result.string(), result.length()); 7233 7234 if (locked) { 7235 mLock.unlock(); 7236 } 7237 7238 return NO_ERROR; 7239} 7240 7241// ---------------------------------------------------------------------------- 7242// EffectHandle implementation 7243// ---------------------------------------------------------------------------- 7244 7245#undef LOG_TAG 7246#define LOG_TAG "AudioFlinger::EffectHandle" 7247 7248AudioFlinger::EffectHandle::EffectHandle(const sp<EffectModule>& effect, 7249 const sp<AudioFlinger::Client>& client, 7250 const sp<IEffectClient>& effectClient, 7251 int32_t priority) 7252 : BnEffect(), 7253 mEffect(effect), mEffectClient(effectClient), mClient(client), mCblk(NULL), 7254 mPriority(priority), mHasControl(false), mEnabled(false) 7255{ 7256 ALOGV("constructor %p", this); 7257 7258 if (client == 0) { 7259 return; 7260 } 7261 int bufOffset = ((sizeof(effect_param_cblk_t) - 1) / sizeof(int) + 1) * sizeof(int); 7262 mCblkMemory = client->heap()->allocate(EFFECT_PARAM_BUFFER_SIZE + bufOffset); 7263 if (mCblkMemory != 0) { 7264 mCblk = static_cast<effect_param_cblk_t *>(mCblkMemory->pointer()); 7265 7266 if (mCblk != NULL) { 7267 new(mCblk) effect_param_cblk_t(); 7268 mBuffer = (uint8_t *)mCblk + bufOffset; 7269 } 7270 } else { 7271 ALOGE("not enough memory for Effect size=%u", EFFECT_PARAM_BUFFER_SIZE + sizeof(effect_param_cblk_t)); 7272 return; 7273 } 7274} 7275 7276AudioFlinger::EffectHandle::~EffectHandle() 7277{ 7278 ALOGV("Destructor %p", this); 7279 disconnect(false); 7280 ALOGV("Destructor DONE %p", this); 7281} 7282 7283status_t AudioFlinger::EffectHandle::enable() 7284{ 7285 ALOGV("enable %p", this); 7286 if (!mHasControl) return INVALID_OPERATION; 7287 if (mEffect == 0) return DEAD_OBJECT; 7288 7289 if (mEnabled) { 7290 return NO_ERROR; 7291 } 7292 7293 mEnabled = true; 7294 7295 sp<ThreadBase> thread = mEffect->thread().promote(); 7296 if (thread != 0) { 7297 thread->checkSuspendOnEffectEnabled(mEffect, true, mEffect->sessionId()); 7298 } 7299 7300 // checkSuspendOnEffectEnabled() can suspend this same effect when enabled 7301 if (mEffect->suspended()) { 7302 return NO_ERROR; 7303 } 7304 7305 status_t status = mEffect->setEnabled(true); 7306 if (status != NO_ERROR) { 7307 if (thread != 0) { 7308 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 7309 } 7310 mEnabled = false; 7311 } 7312 return status; 7313} 7314 7315status_t AudioFlinger::EffectHandle::disable() 7316{ 7317 ALOGV("disable %p", this); 7318 if (!mHasControl) return INVALID_OPERATION; 7319 if (mEffect == 0) return DEAD_OBJECT; 7320 7321 if (!mEnabled) { 7322 return NO_ERROR; 7323 } 7324 mEnabled = false; 7325 7326 if (mEffect->suspended()) { 7327 return NO_ERROR; 7328 } 7329 7330 status_t status = mEffect->setEnabled(false); 7331 7332 sp<ThreadBase> thread = mEffect->thread().promote(); 7333 if (thread != 0) { 7334 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 7335 } 7336 7337 return status; 7338} 7339 7340void AudioFlinger::EffectHandle::disconnect() 7341{ 7342 disconnect(true); 7343} 7344 7345void AudioFlinger::EffectHandle::disconnect(bool unpinIfLast) 7346{ 7347 ALOGV("disconnect(%s)", unpinIfLast ? "true" : "false"); 7348 if (mEffect == 0) { 7349 return; 7350 } 7351 mEffect->disconnect(this, unpinIfLast); 7352 7353 if (mHasControl && mEnabled) { 7354 sp<ThreadBase> thread = mEffect->thread().promote(); 7355 if (thread != 0) { 7356 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 7357 } 7358 } 7359 7360 // release sp on module => module destructor can be called now 7361 mEffect.clear(); 7362 if (mClient != 0) { 7363 if (mCblk != NULL) { 7364 // unlike ~TrackBase(), mCblk is never a local new, so don't delete 7365 mCblk->~effect_param_cblk_t(); // destroy our shared-structure. 7366 } 7367 mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to 7368 // Client destructor must run with AudioFlinger mutex locked 7369 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 7370 mClient.clear(); 7371 } 7372} 7373 7374status_t AudioFlinger::EffectHandle::command(uint32_t cmdCode, 7375 uint32_t cmdSize, 7376 void *pCmdData, 7377 uint32_t *replySize, 7378 void *pReplyData) 7379{ 7380// ALOGV("command(), cmdCode: %d, mHasControl: %d, mEffect: %p", 7381// cmdCode, mHasControl, (mEffect == 0) ? 0 : mEffect.get()); 7382 7383 // only get parameter command is permitted for applications not controlling the effect 7384 if (!mHasControl && cmdCode != EFFECT_CMD_GET_PARAM) { 7385 return INVALID_OPERATION; 7386 } 7387 if (mEffect == 0) return DEAD_OBJECT; 7388 if (mClient == 0) return INVALID_OPERATION; 7389 7390 // handle commands that are not forwarded transparently to effect engine 7391 if (cmdCode == EFFECT_CMD_SET_PARAM_COMMIT) { 7392 // No need to trylock() here as this function is executed in the binder thread serving a particular client process: 7393 // no risk to block the whole media server process or mixer threads is we are stuck here 7394 Mutex::Autolock _l(mCblk->lock); 7395 if (mCblk->clientIndex > EFFECT_PARAM_BUFFER_SIZE || 7396 mCblk->serverIndex > EFFECT_PARAM_BUFFER_SIZE) { 7397 mCblk->serverIndex = 0; 7398 mCblk->clientIndex = 0; 7399 return BAD_VALUE; 7400 } 7401 status_t status = NO_ERROR; 7402 while (mCblk->serverIndex < mCblk->clientIndex) { 7403 int reply; 7404 uint32_t rsize = sizeof(int); 7405 int *p = (int *)(mBuffer + mCblk->serverIndex); 7406 int size = *p++; 7407 if (((uint8_t *)p + size) > mBuffer + mCblk->clientIndex) { 7408 ALOGW("command(): invalid parameter block size"); 7409 break; 7410 } 7411 effect_param_t *param = (effect_param_t *)p; 7412 if (param->psize == 0 || param->vsize == 0) { 7413 ALOGW("command(): null parameter or value size"); 7414 mCblk->serverIndex += size; 7415 continue; 7416 } 7417 uint32_t psize = sizeof(effect_param_t) + 7418 ((param->psize - 1) / sizeof(int) + 1) * sizeof(int) + 7419 param->vsize; 7420 status_t ret = mEffect->command(EFFECT_CMD_SET_PARAM, 7421 psize, 7422 p, 7423 &rsize, 7424 &reply); 7425 // stop at first error encountered 7426 if (ret != NO_ERROR) { 7427 status = ret; 7428 *(int *)pReplyData = reply; 7429 break; 7430 } else if (reply != NO_ERROR) { 7431 *(int *)pReplyData = reply; 7432 break; 7433 } 7434 mCblk->serverIndex += size; 7435 } 7436 mCblk->serverIndex = 0; 7437 mCblk->clientIndex = 0; 7438 return status; 7439 } else if (cmdCode == EFFECT_CMD_ENABLE) { 7440 *(int *)pReplyData = NO_ERROR; 7441 return enable(); 7442 } else if (cmdCode == EFFECT_CMD_DISABLE) { 7443 *(int *)pReplyData = NO_ERROR; 7444 return disable(); 7445 } 7446 7447 return mEffect->command(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 7448} 7449 7450void AudioFlinger::EffectHandle::setControl(bool hasControl, bool signal, bool enabled) 7451{ 7452 ALOGV("setControl %p control %d", this, hasControl); 7453 7454 mHasControl = hasControl; 7455 mEnabled = enabled; 7456 7457 if (signal && mEffectClient != 0) { 7458 mEffectClient->controlStatusChanged(hasControl); 7459 } 7460} 7461 7462void AudioFlinger::EffectHandle::commandExecuted(uint32_t cmdCode, 7463 uint32_t cmdSize, 7464 void *pCmdData, 7465 uint32_t replySize, 7466 void *pReplyData) 7467{ 7468 if (mEffectClient != 0) { 7469 mEffectClient->commandExecuted(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 7470 } 7471} 7472 7473 7474 7475void AudioFlinger::EffectHandle::setEnabled(bool enabled) 7476{ 7477 if (mEffectClient != 0) { 7478 mEffectClient->enableStatusChanged(enabled); 7479 } 7480} 7481 7482status_t AudioFlinger::EffectHandle::onTransact( 7483 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 7484{ 7485 return BnEffect::onTransact(code, data, reply, flags); 7486} 7487 7488 7489void AudioFlinger::EffectHandle::dump(char* buffer, size_t size) 7490{ 7491 bool locked = mCblk != NULL && tryLock(mCblk->lock); 7492 7493 snprintf(buffer, size, "\t\t\t%05d %05d %01u %01u %05u %05u\n", 7494 (mClient == 0) ? getpid_cached : mClient->pid(), 7495 mPriority, 7496 mHasControl, 7497 !locked, 7498 mCblk ? mCblk->clientIndex : 0, 7499 mCblk ? mCblk->serverIndex : 0 7500 ); 7501 7502 if (locked) { 7503 mCblk->lock.unlock(); 7504 } 7505} 7506 7507#undef LOG_TAG 7508#define LOG_TAG "AudioFlinger::EffectChain" 7509 7510AudioFlinger::EffectChain::EffectChain(ThreadBase *thread, 7511 int sessionId) 7512 : mThread(thread), mSessionId(sessionId), mActiveTrackCnt(0), mTrackCnt(0), mTailBufferCount(0), 7513 mOwnInBuffer(false), mVolumeCtrlIdx(-1), mLeftVolume(UINT_MAX), mRightVolume(UINT_MAX), 7514 mNewLeftVolume(UINT_MAX), mNewRightVolume(UINT_MAX) 7515{ 7516 mStrategy = AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 7517 if (thread == NULL) { 7518 return; 7519 } 7520 mMaxTailBuffers = ((kProcessTailDurationMs * thread->sampleRate()) / 1000) / 7521 thread->frameCount(); 7522} 7523 7524AudioFlinger::EffectChain::~EffectChain() 7525{ 7526 if (mOwnInBuffer) { 7527 delete mInBuffer; 7528 } 7529 7530} 7531 7532// getEffectFromDesc_l() must be called with ThreadBase::mLock held 7533sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromDesc_l(effect_descriptor_t *descriptor) 7534{ 7535 size_t size = mEffects.size(); 7536 7537 for (size_t i = 0; i < size; i++) { 7538 if (memcmp(&mEffects[i]->desc().uuid, &descriptor->uuid, sizeof(effect_uuid_t)) == 0) { 7539 return mEffects[i]; 7540 } 7541 } 7542 return 0; 7543} 7544 7545// getEffectFromId_l() must be called with ThreadBase::mLock held 7546sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromId_l(int id) 7547{ 7548 size_t size = mEffects.size(); 7549 7550 for (size_t i = 0; i < size; i++) { 7551 // by convention, return first effect if id provided is 0 (0 is never a valid id) 7552 if (id == 0 || mEffects[i]->id() == id) { 7553 return mEffects[i]; 7554 } 7555 } 7556 return 0; 7557} 7558 7559// getEffectFromType_l() must be called with ThreadBase::mLock held 7560sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromType_l( 7561 const effect_uuid_t *type) 7562{ 7563 size_t size = mEffects.size(); 7564 7565 for (size_t i = 0; i < size; i++) { 7566 if (memcmp(&mEffects[i]->desc().type, type, sizeof(effect_uuid_t)) == 0) { 7567 return mEffects[i]; 7568 } 7569 } 7570 return 0; 7571} 7572 7573// Must be called with EffectChain::mLock locked 7574void AudioFlinger::EffectChain::process_l() 7575{ 7576 sp<ThreadBase> thread = mThread.promote(); 7577 if (thread == 0) { 7578 ALOGW("process_l(): cannot promote mixer thread"); 7579 return; 7580 } 7581 bool isGlobalSession = (mSessionId == AUDIO_SESSION_OUTPUT_MIX) || 7582 (mSessionId == AUDIO_SESSION_OUTPUT_STAGE); 7583 // always process effects unless no more tracks are on the session and the effect tail 7584 // has been rendered 7585 bool doProcess = true; 7586 if (!isGlobalSession) { 7587 bool tracksOnSession = (trackCnt() != 0); 7588 7589 if (!tracksOnSession && mTailBufferCount == 0) { 7590 doProcess = false; 7591 } 7592 7593 if (activeTrackCnt() == 0) { 7594 // if no track is active and the effect tail has not been rendered, 7595 // the input buffer must be cleared here as the mixer process will not do it 7596 if (tracksOnSession || mTailBufferCount > 0) { 7597 size_t numSamples = thread->frameCount() * thread->channelCount(); 7598 memset(mInBuffer, 0, numSamples * sizeof(int16_t)); 7599 if (mTailBufferCount > 0) { 7600 mTailBufferCount--; 7601 } 7602 } 7603 } 7604 } 7605 7606 size_t size = mEffects.size(); 7607 if (doProcess) { 7608 for (size_t i = 0; i < size; i++) { 7609 mEffects[i]->process(); 7610 } 7611 } 7612 for (size_t i = 0; i < size; i++) { 7613 mEffects[i]->updateState(); 7614 } 7615} 7616 7617// addEffect_l() must be called with PlaybackThread::mLock held 7618status_t AudioFlinger::EffectChain::addEffect_l(const sp<EffectModule>& effect) 7619{ 7620 effect_descriptor_t desc = effect->desc(); 7621 uint32_t insertPref = desc.flags & EFFECT_FLAG_INSERT_MASK; 7622 7623 Mutex::Autolock _l(mLock); 7624 effect->setChain(this); 7625 sp<ThreadBase> thread = mThread.promote(); 7626 if (thread == 0) { 7627 return NO_INIT; 7628 } 7629 effect->setThread(thread); 7630 7631 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7632 // Auxiliary effects are inserted at the beginning of mEffects vector as 7633 // they are processed first and accumulated in chain input buffer 7634 mEffects.insertAt(effect, 0); 7635 7636 // the input buffer for auxiliary effect contains mono samples in 7637 // 32 bit format. This is to avoid saturation in AudoMixer 7638 // accumulation stage. Saturation is done in EffectModule::process() before 7639 // calling the process in effect engine 7640 size_t numSamples = thread->frameCount(); 7641 int32_t *buffer = new int32_t[numSamples]; 7642 memset(buffer, 0, numSamples * sizeof(int32_t)); 7643 effect->setInBuffer((int16_t *)buffer); 7644 // auxiliary effects output samples to chain input buffer for further processing 7645 // by insert effects 7646 effect->setOutBuffer(mInBuffer); 7647 } else { 7648 // Insert effects are inserted at the end of mEffects vector as they are processed 7649 // after track and auxiliary effects. 7650 // Insert effect order as a function of indicated preference: 7651 // if EFFECT_FLAG_INSERT_EXCLUSIVE, insert in first position or reject if 7652 // another effect is present 7653 // else if EFFECT_FLAG_INSERT_FIRST, insert in first position or after the 7654 // last effect claiming first position 7655 // else if EFFECT_FLAG_INSERT_LAST, insert in last position or before the 7656 // first effect claiming last position 7657 // else if EFFECT_FLAG_INSERT_ANY insert after first or before last 7658 // Reject insertion if an effect with EFFECT_FLAG_INSERT_EXCLUSIVE is 7659 // already present 7660 7661 size_t size = mEffects.size(); 7662 size_t idx_insert = size; 7663 ssize_t idx_insert_first = -1; 7664 ssize_t idx_insert_last = -1; 7665 7666 for (size_t i = 0; i < size; i++) { 7667 effect_descriptor_t d = mEffects[i]->desc(); 7668 uint32_t iMode = d.flags & EFFECT_FLAG_TYPE_MASK; 7669 uint32_t iPref = d.flags & EFFECT_FLAG_INSERT_MASK; 7670 if (iMode == EFFECT_FLAG_TYPE_INSERT) { 7671 // check invalid effect chaining combinations 7672 if (insertPref == EFFECT_FLAG_INSERT_EXCLUSIVE || 7673 iPref == EFFECT_FLAG_INSERT_EXCLUSIVE) { 7674 ALOGW("addEffect_l() could not insert effect %s: exclusive conflict with %s", desc.name, d.name); 7675 return INVALID_OPERATION; 7676 } 7677 // remember position of first insert effect and by default 7678 // select this as insert position for new effect 7679 if (idx_insert == size) { 7680 idx_insert = i; 7681 } 7682 // remember position of last insert effect claiming 7683 // first position 7684 if (iPref == EFFECT_FLAG_INSERT_FIRST) { 7685 idx_insert_first = i; 7686 } 7687 // remember position of first insert effect claiming 7688 // last position 7689 if (iPref == EFFECT_FLAG_INSERT_LAST && 7690 idx_insert_last == -1) { 7691 idx_insert_last = i; 7692 } 7693 } 7694 } 7695 7696 // modify idx_insert from first position if needed 7697 if (insertPref == EFFECT_FLAG_INSERT_LAST) { 7698 if (idx_insert_last != -1) { 7699 idx_insert = idx_insert_last; 7700 } else { 7701 idx_insert = size; 7702 } 7703 } else { 7704 if (idx_insert_first != -1) { 7705 idx_insert = idx_insert_first + 1; 7706 } 7707 } 7708 7709 // always read samples from chain input buffer 7710 effect->setInBuffer(mInBuffer); 7711 7712 // if last effect in the chain, output samples to chain 7713 // output buffer, otherwise to chain input buffer 7714 if (idx_insert == size) { 7715 if (idx_insert != 0) { 7716 mEffects[idx_insert-1]->setOutBuffer(mInBuffer); 7717 mEffects[idx_insert-1]->configure(); 7718 } 7719 effect->setOutBuffer(mOutBuffer); 7720 } else { 7721 effect->setOutBuffer(mInBuffer); 7722 } 7723 mEffects.insertAt(effect, idx_insert); 7724 7725 ALOGV("addEffect_l() effect %p, added in chain %p at rank %d", effect.get(), this, idx_insert); 7726 } 7727 effect->configure(); 7728 return NO_ERROR; 7729} 7730 7731// removeEffect_l() must be called with PlaybackThread::mLock held 7732size_t AudioFlinger::EffectChain::removeEffect_l(const sp<EffectModule>& effect) 7733{ 7734 Mutex::Autolock _l(mLock); 7735 size_t size = mEffects.size(); 7736 uint32_t type = effect->desc().flags & EFFECT_FLAG_TYPE_MASK; 7737 7738 for (size_t i = 0; i < size; i++) { 7739 if (effect == mEffects[i]) { 7740 // calling stop here will remove pre-processing effect from the audio HAL. 7741 // This is safe as we hold the EffectChain mutex which guarantees that we are not in 7742 // the middle of a read from audio HAL 7743 if (mEffects[i]->state() == EffectModule::ACTIVE || 7744 mEffects[i]->state() == EffectModule::STOPPING) { 7745 mEffects[i]->stop(); 7746 } 7747 if (type == EFFECT_FLAG_TYPE_AUXILIARY) { 7748 delete[] effect->inBuffer(); 7749 } else { 7750 if (i == size - 1 && i != 0) { 7751 mEffects[i - 1]->setOutBuffer(mOutBuffer); 7752 mEffects[i - 1]->configure(); 7753 } 7754 } 7755 mEffects.removeAt(i); 7756 ALOGV("removeEffect_l() effect %p, removed from chain %p at rank %d", effect.get(), this, i); 7757 break; 7758 } 7759 } 7760 7761 return mEffects.size(); 7762} 7763 7764// setDevice_l() must be called with PlaybackThread::mLock held 7765void AudioFlinger::EffectChain::setDevice_l(uint32_t device) 7766{ 7767 size_t size = mEffects.size(); 7768 for (size_t i = 0; i < size; i++) { 7769 mEffects[i]->setDevice(device); 7770 } 7771} 7772 7773// setMode_l() must be called with PlaybackThread::mLock held 7774void AudioFlinger::EffectChain::setMode_l(audio_mode_t mode) 7775{ 7776 size_t size = mEffects.size(); 7777 for (size_t i = 0; i < size; i++) { 7778 mEffects[i]->setMode(mode); 7779 } 7780} 7781 7782// setVolume_l() must be called with PlaybackThread::mLock held 7783bool AudioFlinger::EffectChain::setVolume_l(uint32_t *left, uint32_t *right) 7784{ 7785 uint32_t newLeft = *left; 7786 uint32_t newRight = *right; 7787 bool hasControl = false; 7788 int ctrlIdx = -1; 7789 size_t size = mEffects.size(); 7790 7791 // first update volume controller 7792 for (size_t i = size; i > 0; i--) { 7793 if (mEffects[i - 1]->isProcessEnabled() && 7794 (mEffects[i - 1]->desc().flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL) { 7795 ctrlIdx = i - 1; 7796 hasControl = true; 7797 break; 7798 } 7799 } 7800 7801 if (ctrlIdx == mVolumeCtrlIdx && *left == mLeftVolume && *right == mRightVolume) { 7802 if (hasControl) { 7803 *left = mNewLeftVolume; 7804 *right = mNewRightVolume; 7805 } 7806 return hasControl; 7807 } 7808 7809 mVolumeCtrlIdx = ctrlIdx; 7810 mLeftVolume = newLeft; 7811 mRightVolume = newRight; 7812 7813 // second get volume update from volume controller 7814 if (ctrlIdx >= 0) { 7815 mEffects[ctrlIdx]->setVolume(&newLeft, &newRight, true); 7816 mNewLeftVolume = newLeft; 7817 mNewRightVolume = newRight; 7818 } 7819 // then indicate volume to all other effects in chain. 7820 // Pass altered volume to effects before volume controller 7821 // and requested volume to effects after controller 7822 uint32_t lVol = newLeft; 7823 uint32_t rVol = newRight; 7824 7825 for (size_t i = 0; i < size; i++) { 7826 if ((int)i == ctrlIdx) continue; 7827 // this also works for ctrlIdx == -1 when there is no volume controller 7828 if ((int)i > ctrlIdx) { 7829 lVol = *left; 7830 rVol = *right; 7831 } 7832 mEffects[i]->setVolume(&lVol, &rVol, false); 7833 } 7834 *left = newLeft; 7835 *right = newRight; 7836 7837 return hasControl; 7838} 7839 7840status_t AudioFlinger::EffectChain::dump(int fd, const Vector<String16>& args) 7841{ 7842 const size_t SIZE = 256; 7843 char buffer[SIZE]; 7844 String8 result; 7845 7846 snprintf(buffer, SIZE, "Effects for session %d:\n", mSessionId); 7847 result.append(buffer); 7848 7849 bool locked = tryLock(mLock); 7850 // failed to lock - AudioFlinger is probably deadlocked 7851 if (!locked) { 7852 result.append("\tCould not lock mutex:\n"); 7853 } 7854 7855 result.append("\tNum fx In buffer Out buffer Active tracks:\n"); 7856 snprintf(buffer, SIZE, "\t%02d 0x%08x 0x%08x %d\n", 7857 mEffects.size(), 7858 (uint32_t)mInBuffer, 7859 (uint32_t)mOutBuffer, 7860 mActiveTrackCnt); 7861 result.append(buffer); 7862 write(fd, result.string(), result.size()); 7863 7864 for (size_t i = 0; i < mEffects.size(); ++i) { 7865 sp<EffectModule> effect = mEffects[i]; 7866 if (effect != 0) { 7867 effect->dump(fd, args); 7868 } 7869 } 7870 7871 if (locked) { 7872 mLock.unlock(); 7873 } 7874 7875 return NO_ERROR; 7876} 7877 7878// must be called with ThreadBase::mLock held 7879void AudioFlinger::EffectChain::setEffectSuspended_l( 7880 const effect_uuid_t *type, bool suspend) 7881{ 7882 sp<SuspendedEffectDesc> desc; 7883 // use effect type UUID timelow as key as there is no real risk of identical 7884 // timeLow fields among effect type UUIDs. 7885 ssize_t index = mSuspendedEffects.indexOfKey(type->timeLow); 7886 if (suspend) { 7887 if (index >= 0) { 7888 desc = mSuspendedEffects.valueAt(index); 7889 } else { 7890 desc = new SuspendedEffectDesc(); 7891 memcpy(&desc->mType, type, sizeof(effect_uuid_t)); 7892 mSuspendedEffects.add(type->timeLow, desc); 7893 ALOGV("setEffectSuspended_l() add entry for %08x", type->timeLow); 7894 } 7895 if (desc->mRefCount++ == 0) { 7896 sp<EffectModule> effect = getEffectIfEnabled(type); 7897 if (effect != 0) { 7898 desc->mEffect = effect; 7899 effect->setSuspended(true); 7900 effect->setEnabled(false); 7901 } 7902 } 7903 } else { 7904 if (index < 0) { 7905 return; 7906 } 7907 desc = mSuspendedEffects.valueAt(index); 7908 if (desc->mRefCount <= 0) { 7909 ALOGW("setEffectSuspended_l() restore refcount should not be 0 %d", desc->mRefCount); 7910 desc->mRefCount = 1; 7911 } 7912 if (--desc->mRefCount == 0) { 7913 ALOGV("setEffectSuspended_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 7914 if (desc->mEffect != 0) { 7915 sp<EffectModule> effect = desc->mEffect.promote(); 7916 if (effect != 0) { 7917 effect->setSuspended(false); 7918 sp<EffectHandle> handle = effect->controlHandle(); 7919 if (handle != 0) { 7920 effect->setEnabled(handle->enabled()); 7921 } 7922 } 7923 desc->mEffect.clear(); 7924 } 7925 mSuspendedEffects.removeItemsAt(index); 7926 } 7927 } 7928} 7929 7930// must be called with ThreadBase::mLock held 7931void AudioFlinger::EffectChain::setEffectSuspendedAll_l(bool suspend) 7932{ 7933 sp<SuspendedEffectDesc> desc; 7934 7935 ssize_t index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 7936 if (suspend) { 7937 if (index >= 0) { 7938 desc = mSuspendedEffects.valueAt(index); 7939 } else { 7940 desc = new SuspendedEffectDesc(); 7941 mSuspendedEffects.add((int)kKeyForSuspendAll, desc); 7942 ALOGV("setEffectSuspendedAll_l() add entry for 0"); 7943 } 7944 if (desc->mRefCount++ == 0) { 7945 Vector< sp<EffectModule> > effects; 7946 getSuspendEligibleEffects(effects); 7947 for (size_t i = 0; i < effects.size(); i++) { 7948 setEffectSuspended_l(&effects[i]->desc().type, true); 7949 } 7950 } 7951 } else { 7952 if (index < 0) { 7953 return; 7954 } 7955 desc = mSuspendedEffects.valueAt(index); 7956 if (desc->mRefCount <= 0) { 7957 ALOGW("setEffectSuspendedAll_l() restore refcount should not be 0 %d", desc->mRefCount); 7958 desc->mRefCount = 1; 7959 } 7960 if (--desc->mRefCount == 0) { 7961 Vector<const effect_uuid_t *> types; 7962 for (size_t i = 0; i < mSuspendedEffects.size(); i++) { 7963 if (mSuspendedEffects.keyAt(i) == (int)kKeyForSuspendAll) { 7964 continue; 7965 } 7966 types.add(&mSuspendedEffects.valueAt(i)->mType); 7967 } 7968 for (size_t i = 0; i < types.size(); i++) { 7969 setEffectSuspended_l(types[i], false); 7970 } 7971 ALOGV("setEffectSuspendedAll_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 7972 mSuspendedEffects.removeItem((int)kKeyForSuspendAll); 7973 } 7974 } 7975} 7976 7977 7978// The volume effect is used for automated tests only 7979#ifndef OPENSL_ES_H_ 7980static const effect_uuid_t SL_IID_VOLUME_ = { 0x09e8ede0, 0xddde, 0x11db, 0xb4f6, 7981 { 0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b } }; 7982const effect_uuid_t * const SL_IID_VOLUME = &SL_IID_VOLUME_; 7983#endif //OPENSL_ES_H_ 7984 7985bool AudioFlinger::EffectChain::isEffectEligibleForSuspend(const effect_descriptor_t& desc) 7986{ 7987 // auxiliary effects and visualizer are never suspended on output mix 7988 if ((mSessionId == AUDIO_SESSION_OUTPUT_MIX) && 7989 (((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) || 7990 (memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) || 7991 (memcmp(&desc.type, SL_IID_VOLUME, sizeof(effect_uuid_t)) == 0))) { 7992 return false; 7993 } 7994 return true; 7995} 7996 7997void AudioFlinger::EffectChain::getSuspendEligibleEffects(Vector< sp<AudioFlinger::EffectModule> > &effects) 7998{ 7999 effects.clear(); 8000 for (size_t i = 0; i < mEffects.size(); i++) { 8001 if (isEffectEligibleForSuspend(mEffects[i]->desc())) { 8002 effects.add(mEffects[i]); 8003 } 8004 } 8005} 8006 8007sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectIfEnabled( 8008 const effect_uuid_t *type) 8009{ 8010 sp<EffectModule> effect = getEffectFromType_l(type); 8011 return effect != 0 && effect->isEnabled() ? effect : 0; 8012} 8013 8014void AudioFlinger::EffectChain::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 8015 bool enabled) 8016{ 8017 ssize_t index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 8018 if (enabled) { 8019 if (index < 0) { 8020 // if the effect is not suspend check if all effects are suspended 8021 index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 8022 if (index < 0) { 8023 return; 8024 } 8025 if (!isEffectEligibleForSuspend(effect->desc())) { 8026 return; 8027 } 8028 setEffectSuspended_l(&effect->desc().type, enabled); 8029 index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 8030 if (index < 0) { 8031 ALOGW("checkSuspendOnEffectEnabled() Fx should be suspended here!"); 8032 return; 8033 } 8034 } 8035 ALOGV("checkSuspendOnEffectEnabled() enable suspending fx %08x", 8036 effect->desc().type.timeLow); 8037 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 8038 // if effect is requested to suspended but was not yet enabled, supend it now. 8039 if (desc->mEffect == 0) { 8040 desc->mEffect = effect; 8041 effect->setEnabled(false); 8042 effect->setSuspended(true); 8043 } 8044 } else { 8045 if (index < 0) { 8046 return; 8047 } 8048 ALOGV("checkSuspendOnEffectEnabled() disable restoring fx %08x", 8049 effect->desc().type.timeLow); 8050 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 8051 desc->mEffect.clear(); 8052 effect->setSuspended(false); 8053 } 8054} 8055 8056#undef LOG_TAG 8057#define LOG_TAG "AudioFlinger" 8058 8059// ---------------------------------------------------------------------------- 8060 8061status_t AudioFlinger::onTransact( 8062 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 8063{ 8064 return BnAudioFlinger::onTransact(code, data, reply, flags); 8065} 8066 8067}; // namespace android 8068