Threads.cpp revision dce27d0ebab31e82543bb777ed3eb04955cd18ff
13527fd6f0df794207215790321824b7844cc712dDavid 'Digit' Turner/* 23527fd6f0df794207215790321824b7844cc712dDavid 'Digit' Turner** 33527fd6f0df794207215790321824b7844cc712dDavid 'Digit' Turner** Copyright 2012, The Android Open Source Project 43527fd6f0df794207215790321824b7844cc712dDavid 'Digit' Turner** 53527fd6f0df794207215790321824b7844cc712dDavid 'Digit' Turner** Licensed under the Apache License, Version 2.0 (the "License"); 63527fd6f0df794207215790321824b7844cc712dDavid 'Digit' Turner** you may not use this file except in compliance with the License. 73527fd6f0df794207215790321824b7844cc712dDavid 'Digit' Turner** You may obtain a copy of the License at 83527fd6f0df794207215790321824b7844cc712dDavid 'Digit' Turner** 93527fd6f0df794207215790321824b7844cc712dDavid 'Digit' Turner** http://www.apache.org/licenses/LICENSE-2.0 103527fd6f0df794207215790321824b7844cc712dDavid 'Digit' Turner** 113527fd6f0df794207215790321824b7844cc712dDavid 'Digit' Turner** Unless required by applicable law or agreed to in writing, software 123527fd6f0df794207215790321824b7844cc712dDavid 'Digit' Turner** distributed under the License is distributed on an "AS IS" BASIS, 133527fd6f0df794207215790321824b7844cc712dDavid 'Digit' Turner** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. 143527fd6f0df794207215790321824b7844cc712dDavid 'Digit' Turner** See the License for the specific language governing permissions and 153527fd6f0df794207215790321824b7844cc712dDavid 'Digit' Turner** limitations under the License. 163527fd6f0df794207215790321824b7844cc712dDavid 'Digit' Turner*/ 173527fd6f0df794207215790321824b7844cc712dDavid 'Digit' Turner 183527fd6f0df794207215790321824b7844cc712dDavid 'Digit' Turner 193527fd6f0df794207215790321824b7844cc712dDavid 'Digit' Turner#define LOG_TAG "AudioFlinger" 203527fd6f0df794207215790321824b7844cc712dDavid 'Digit' Turner//#define LOG_NDEBUG 0 213527fd6f0df794207215790321824b7844cc712dDavid 'Digit' Turner#define ATRACE_TAG ATRACE_TAG_AUDIO 223527fd6f0df794207215790321824b7844cc712dDavid 'Digit' Turner 233527fd6f0df794207215790321824b7844cc712dDavid 'Digit' Turner#include "Configuration.h" 243527fd6f0df794207215790321824b7844cc712dDavid 'Digit' Turner#include <math.h> 253527fd6f0df794207215790321824b7844cc712dDavid 'Digit' Turner#include <fcntl.h> 263527fd6f0df794207215790321824b7844cc712dDavid 'Digit' Turner#include <sys/stat.h> 273527fd6f0df794207215790321824b7844cc712dDavid 'Digit' Turner#include <cutils/properties.h> 283527fd6f0df794207215790321824b7844cc712dDavid 'Digit' Turner#include <media/AudioParameter.h> 293527fd6f0df794207215790321824b7844cc712dDavid 'Digit' Turner#include <media/AudioResamplerPublic.h> 303527fd6f0df794207215790321824b7844cc712dDavid 'Digit' Turner#include <utils/Log.h> 313527fd6f0df794207215790321824b7844cc712dDavid 'Digit' Turner#include <utils/Trace.h> 323527fd6f0df794207215790321824b7844cc712dDavid 'Digit' Turner 333527fd6f0df794207215790321824b7844cc712dDavid 'Digit' Turner#include <private/media/AudioTrackShared.h> 343527fd6f0df794207215790321824b7844cc712dDavid 'Digit' Turner#include <hardware/audio.h> 353527fd6f0df794207215790321824b7844cc712dDavid 'Digit' Turner#include <audio_effects/effect_ns.h> 363527fd6f0df794207215790321824b7844cc712dDavid 'Digit' Turner#include <audio_effects/effect_aec.h> 373527fd6f0df794207215790321824b7844cc712dDavid 'Digit' Turner#include <audio_utils/primitives.h> 383527fd6f0df794207215790321824b7844cc712dDavid 'Digit' Turner#include <audio_utils/format.h> 393527fd6f0df794207215790321824b7844cc712dDavid 'Digit' Turner#include <audio_utils/minifloat.h> 403527fd6f0df794207215790321824b7844cc712dDavid 'Digit' Turner 413527fd6f0df794207215790321824b7844cc712dDavid 'Digit' Turner// NBAIO implementations 423527fd6f0df794207215790321824b7844cc712dDavid 'Digit' Turner#include <media/nbaio/AudioStreamInSource.h> 433527fd6f0df794207215790321824b7844cc712dDavid 'Digit' Turner#include <media/nbaio/AudioStreamOutSink.h> 443527fd6f0df794207215790321824b7844cc712dDavid 'Digit' Turner#include <media/nbaio/MonoPipe.h> 453527fd6f0df794207215790321824b7844cc712dDavid 'Digit' Turner#include <media/nbaio/MonoPipeReader.h> 463527fd6f0df794207215790321824b7844cc712dDavid 'Digit' Turner#include <media/nbaio/Pipe.h> 473527fd6f0df794207215790321824b7844cc712dDavid 'Digit' Turner#include <media/nbaio/PipeReader.h> 483527fd6f0df794207215790321824b7844cc712dDavid 'Digit' Turner#include <media/nbaio/SourceAudioBufferProvider.h> 493527fd6f0df794207215790321824b7844cc712dDavid 'Digit' Turner 503527fd6f0df794207215790321824b7844cc712dDavid 'Digit' Turner#include <powermanager/PowerManager.h> 513527fd6f0df794207215790321824b7844cc712dDavid 'Digit' Turner 52#include <common_time/cc_helper.h> 53#include <common_time/local_clock.h> 54 55#include "AudioFlinger.h" 56#include "AudioMixer.h" 57#include "FastMixer.h" 58#include "FastCapture.h" 59#include "ServiceUtilities.h" 60#include "SchedulingPolicyService.h" 61 62#ifdef ADD_BATTERY_DATA 63#include <media/IMediaPlayerService.h> 64#include <media/IMediaDeathNotifier.h> 65#endif 66 67#ifdef DEBUG_CPU_USAGE 68#include <cpustats/CentralTendencyStatistics.h> 69#include <cpustats/ThreadCpuUsage.h> 70#endif 71 72// ---------------------------------------------------------------------------- 73 74// Note: the following macro is used for extremely verbose logging message. In 75// order to run with ALOG_ASSERT turned on, we need to have LOG_NDEBUG set to 76// 0; but one side effect of this is to turn all LOGV's as well. Some messages 77// are so verbose that we want to suppress them even when we have ALOG_ASSERT 78// turned on. Do not uncomment the #def below unless you really know what you 79// are doing and want to see all of the extremely verbose messages. 80//#define VERY_VERY_VERBOSE_LOGGING 81#ifdef VERY_VERY_VERBOSE_LOGGING 82#define ALOGVV ALOGV 83#else 84#define ALOGVV(a...) do { } while(0) 85#endif 86 87#define max(a, b) ((a) > (b) ? (a) : (b)) 88 89namespace android { 90 91// retry counts for buffer fill timeout 92// 50 * ~20msecs = 1 second 93static const int8_t kMaxTrackRetries = 50; 94static const int8_t kMaxTrackStartupRetries = 50; 95// allow less retry attempts on direct output thread. 96// direct outputs can be a scarce resource in audio hardware and should 97// be released as quickly as possible. 98static const int8_t kMaxTrackRetriesDirect = 2; 99 100// don't warn about blocked writes or record buffer overflows more often than this 101static const nsecs_t kWarningThrottleNs = seconds(5); 102 103// RecordThread loop sleep time upon application overrun or audio HAL read error 104static const int kRecordThreadSleepUs = 5000; 105 106// maximum time to wait in sendConfigEvent_l() for a status to be received 107static const nsecs_t kConfigEventTimeoutNs = seconds(2); 108 109// minimum sleep time for the mixer thread loop when tracks are active but in underrun 110static const uint32_t kMinThreadSleepTimeUs = 5000; 111// maximum divider applied to the active sleep time in the mixer thread loop 112static const uint32_t kMaxThreadSleepTimeShift = 2; 113 114// minimum normal sink buffer size, expressed in milliseconds rather than frames 115static const uint32_t kMinNormalSinkBufferSizeMs = 20; 116// maximum normal sink buffer size 117static const uint32_t kMaxNormalSinkBufferSizeMs = 24; 118 119// Offloaded output thread standby delay: allows track transition without going to standby 120static const nsecs_t kOffloadStandbyDelayNs = seconds(1); 121 122// Whether to use fast mixer 123static const enum { 124 FastMixer_Never, // never initialize or use: for debugging only 125 FastMixer_Always, // always initialize and use, even if not needed: for debugging only 126 // normal mixer multiplier is 1 127 FastMixer_Static, // initialize if needed, then use all the time if initialized, 128 // multiplier is calculated based on min & max normal mixer buffer size 129 FastMixer_Dynamic, // initialize if needed, then use dynamically depending on track load, 130 // multiplier is calculated based on min & max normal mixer buffer size 131 // FIXME for FastMixer_Dynamic: 132 // Supporting this option will require fixing HALs that can't handle large writes. 133 // For example, one HAL implementation returns an error from a large write, 134 // and another HAL implementation corrupts memory, possibly in the sample rate converter. 135 // We could either fix the HAL implementations, or provide a wrapper that breaks 136 // up large writes into smaller ones, and the wrapper would need to deal with scheduler. 137} kUseFastMixer = FastMixer_Static; 138 139// Whether to use fast capture 140static const enum { 141 FastCapture_Never, // never initialize or use: for debugging only 142 FastCapture_Always, // always initialize and use, even if not needed: for debugging only 143 FastCapture_Static, // initialize if needed, then use all the time if initialized 144} kUseFastCapture = FastCapture_Static; 145 146// Priorities for requestPriority 147static const int kPriorityAudioApp = 2; 148static const int kPriorityFastMixer = 3; 149static const int kPriorityFastCapture = 3; 150 151// IAudioFlinger::createTrack() reports back to client the total size of shared memory area 152// for the track. The client then sub-divides this into smaller buffers for its use. 153// Currently the client uses N-buffering by default, but doesn't tell us about the value of N. 154// So for now we just assume that client is double-buffered for fast tracks. 155// FIXME It would be better for client to tell AudioFlinger the value of N, 156// so AudioFlinger could allocate the right amount of memory. 157// See the client's minBufCount and mNotificationFramesAct calculations for details. 158 159// This is the default value, if not specified by property. 160static const int kFastTrackMultiplier = 2; 161 162// The minimum and maximum allowed values 163static const int kFastTrackMultiplierMin = 1; 164static const int kFastTrackMultiplierMax = 2; 165 166// The actual value to use, which can be specified per-device via property af.fast_track_multiplier. 167static int sFastTrackMultiplier = kFastTrackMultiplier; 168 169// See Thread::readOnlyHeap(). 170// Initially this heap is used to allocate client buffers for "fast" AudioRecord. 171// Eventually it will be the single buffer that FastCapture writes into via HAL read(), 172// and that all "fast" AudioRecord clients read from. In either case, the size can be small. 173static const size_t kRecordThreadReadOnlyHeapSize = 0x2000; 174 175// Returns the source frames needed to resample to destination frames. This is not a precise 176// value and depends on the resampler (and possibly how it handles rounding internally). 177// If srcSampleRate and dstSampleRate are equal, then it returns destination frames, which 178// may not be a true if the resampler is asynchronous. 179static inline size_t sourceFramesNeeded( 180 uint32_t srcSampleRate, size_t dstFramesRequired, uint32_t dstSampleRate) { 181 // +1 for rounding - always do this even if matched ratio 182 // +1 for additional sample needed for interpolation 183 return srcSampleRate == dstSampleRate ? dstFramesRequired : 184 size_t((uint64_t)dstFramesRequired * srcSampleRate / dstSampleRate + 1 + 1); 185} 186 187// ---------------------------------------------------------------------------- 188 189static pthread_once_t sFastTrackMultiplierOnce = PTHREAD_ONCE_INIT; 190 191static void sFastTrackMultiplierInit() 192{ 193 char value[PROPERTY_VALUE_MAX]; 194 if (property_get("af.fast_track_multiplier", value, NULL) > 0) { 195 char *endptr; 196 unsigned long ul = strtoul(value, &endptr, 0); 197 if (*endptr == '\0' && kFastTrackMultiplierMin <= ul && ul <= kFastTrackMultiplierMax) { 198 sFastTrackMultiplier = (int) ul; 199 } 200 } 201} 202 203// ---------------------------------------------------------------------------- 204 205#ifdef ADD_BATTERY_DATA 206// To collect the amplifier usage 207static void addBatteryData(uint32_t params) { 208 sp<IMediaPlayerService> service = IMediaDeathNotifier::getMediaPlayerService(); 209 if (service == NULL) { 210 // it already logged 211 return; 212 } 213 214 service->addBatteryData(params); 215} 216#endif 217 218 219// ---------------------------------------------------------------------------- 220// CPU Stats 221// ---------------------------------------------------------------------------- 222 223class CpuStats { 224public: 225 CpuStats(); 226 void sample(const String8 &title); 227#ifdef DEBUG_CPU_USAGE 228private: 229 ThreadCpuUsage mCpuUsage; // instantaneous thread CPU usage in wall clock ns 230 CentralTendencyStatistics mWcStats; // statistics on thread CPU usage in wall clock ns 231 232 CentralTendencyStatistics mHzStats; // statistics on thread CPU usage in cycles 233 234 int mCpuNum; // thread's current CPU number 235 int mCpukHz; // frequency of thread's current CPU in kHz 236#endif 237}; 238 239CpuStats::CpuStats() 240#ifdef DEBUG_CPU_USAGE 241 : mCpuNum(-1), mCpukHz(-1) 242#endif 243{ 244} 245 246void CpuStats::sample(const String8 &title 247#ifndef DEBUG_CPU_USAGE 248 __unused 249#endif 250 ) { 251#ifdef DEBUG_CPU_USAGE 252 // get current thread's delta CPU time in wall clock ns 253 double wcNs; 254 bool valid = mCpuUsage.sampleAndEnable(wcNs); 255 256 // record sample for wall clock statistics 257 if (valid) { 258 mWcStats.sample(wcNs); 259 } 260 261 // get the current CPU number 262 int cpuNum = sched_getcpu(); 263 264 // get the current CPU frequency in kHz 265 int cpukHz = mCpuUsage.getCpukHz(cpuNum); 266 267 // check if either CPU number or frequency changed 268 if (cpuNum != mCpuNum || cpukHz != mCpukHz) { 269 mCpuNum = cpuNum; 270 mCpukHz = cpukHz; 271 // ignore sample for purposes of cycles 272 valid = false; 273 } 274 275 // if no change in CPU number or frequency, then record sample for cycle statistics 276 if (valid && mCpukHz > 0) { 277 double cycles = wcNs * cpukHz * 0.000001; 278 mHzStats.sample(cycles); 279 } 280 281 unsigned n = mWcStats.n(); 282 // mCpuUsage.elapsed() is expensive, so don't call it every loop 283 if ((n & 127) == 1) { 284 long long elapsed = mCpuUsage.elapsed(); 285 if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) { 286 double perLoop = elapsed / (double) n; 287 double perLoop100 = perLoop * 0.01; 288 double perLoop1k = perLoop * 0.001; 289 double mean = mWcStats.mean(); 290 double stddev = mWcStats.stddev(); 291 double minimum = mWcStats.minimum(); 292 double maximum = mWcStats.maximum(); 293 double meanCycles = mHzStats.mean(); 294 double stddevCycles = mHzStats.stddev(); 295 double minCycles = mHzStats.minimum(); 296 double maxCycles = mHzStats.maximum(); 297 mCpuUsage.resetElapsed(); 298 mWcStats.reset(); 299 mHzStats.reset(); 300 ALOGD("CPU usage for %s over past %.1f secs\n" 301 " (%u mixer loops at %.1f mean ms per loop):\n" 302 " us per mix loop: mean=%.0f stddev=%.0f min=%.0f max=%.0f\n" 303 " %% of wall: mean=%.1f stddev=%.1f min=%.1f max=%.1f\n" 304 " MHz: mean=%.1f, stddev=%.1f, min=%.1f max=%.1f", 305 title.string(), 306 elapsed * .000000001, n, perLoop * .000001, 307 mean * .001, 308 stddev * .001, 309 minimum * .001, 310 maximum * .001, 311 mean / perLoop100, 312 stddev / perLoop100, 313 minimum / perLoop100, 314 maximum / perLoop100, 315 meanCycles / perLoop1k, 316 stddevCycles / perLoop1k, 317 minCycles / perLoop1k, 318 maxCycles / perLoop1k); 319 320 } 321 } 322#endif 323}; 324 325// ---------------------------------------------------------------------------- 326// ThreadBase 327// ---------------------------------------------------------------------------- 328 329// static 330const char *AudioFlinger::ThreadBase::threadTypeToString(AudioFlinger::ThreadBase::type_t type) 331{ 332 switch (type) { 333 case MIXER: 334 return "MIXER"; 335 case DIRECT: 336 return "DIRECT"; 337 case DUPLICATING: 338 return "DUPLICATING"; 339 case RECORD: 340 return "RECORD"; 341 case OFFLOAD: 342 return "OFFLOAD"; 343 default: 344 return "unknown"; 345 } 346} 347 348static String8 outputFlagsToString(audio_output_flags_t flags) 349{ 350 static const struct mapping { 351 audio_output_flags_t mFlag; 352 const char * mString; 353 } mappings[] = { 354 AUDIO_OUTPUT_FLAG_DIRECT, "DIRECT", 355 AUDIO_OUTPUT_FLAG_PRIMARY, "PRIMARY", 356 AUDIO_OUTPUT_FLAG_FAST, "FAST", 357 AUDIO_OUTPUT_FLAG_DEEP_BUFFER, "DEEP_BUFFER", 358 AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD, "COMPRESS_OFFLOAAD", 359 AUDIO_OUTPUT_FLAG_NON_BLOCKING, "NON_BLOCKING", 360 AUDIO_OUTPUT_FLAG_HW_AV_SYNC, "HW_AV_SYNC", 361 AUDIO_OUTPUT_FLAG_NONE, "NONE", // must be last 362 }; 363 String8 result; 364 audio_output_flags_t allFlags = AUDIO_OUTPUT_FLAG_NONE; 365 const mapping *entry; 366 for (entry = mappings; entry->mFlag != AUDIO_OUTPUT_FLAG_NONE; entry++) { 367 allFlags = (audio_output_flags_t) (allFlags | entry->mFlag); 368 if (flags & entry->mFlag) { 369 if (!result.isEmpty()) { 370 result.append("|"); 371 } 372 result.append(entry->mString); 373 } 374 } 375 if (flags & ~allFlags) { 376 if (!result.isEmpty()) { 377 result.append("|"); 378 } 379 result.appendFormat("0x%X", flags & ~allFlags); 380 } 381 if (result.isEmpty()) { 382 result.append(entry->mString); 383 } 384 return result; 385} 386 387AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, 388 audio_devices_t outDevice, audio_devices_t inDevice, type_t type) 389 : Thread(false /*canCallJava*/), 390 mType(type), 391 mAudioFlinger(audioFlinger), 392 // mSampleRate, mFrameCount, mChannelMask, mChannelCount, mFrameSize, mFormat, mBufferSize 393 // are set by PlaybackThread::readOutputParameters_l() or 394 // RecordThread::readInputParameters_l() 395 //FIXME: mStandby should be true here. Is this some kind of hack? 396 mStandby(false), mOutDevice(outDevice), mInDevice(inDevice), 397 mAudioSource(AUDIO_SOURCE_DEFAULT), mId(id), 398 // mName will be set by concrete (non-virtual) subclass 399 mDeathRecipient(new PMDeathRecipient(this)) 400{ 401} 402 403AudioFlinger::ThreadBase::~ThreadBase() 404{ 405 // mConfigEvents should be empty, but just in case it isn't, free the memory it owns 406 mConfigEvents.clear(); 407 408 // do not lock the mutex in destructor 409 releaseWakeLock_l(); 410 if (mPowerManager != 0) { 411 sp<IBinder> binder = IInterface::asBinder(mPowerManager); 412 binder->unlinkToDeath(mDeathRecipient); 413 } 414} 415 416status_t AudioFlinger::ThreadBase::readyToRun() 417{ 418 status_t status = initCheck(); 419 if (status == NO_ERROR) { 420 ALOGI("AudioFlinger's thread %p ready to run", this); 421 } else { 422 ALOGE("No working audio driver found."); 423 } 424 return status; 425} 426 427void AudioFlinger::ThreadBase::exit() 428{ 429 ALOGV("ThreadBase::exit"); 430 // do any cleanup required for exit to succeed 431 preExit(); 432 { 433 // This lock prevents the following race in thread (uniprocessor for illustration): 434 // if (!exitPending()) { 435 // // context switch from here to exit() 436 // // exit() calls requestExit(), what exitPending() observes 437 // // exit() calls signal(), which is dropped since no waiters 438 // // context switch back from exit() to here 439 // mWaitWorkCV.wait(...); 440 // // now thread is hung 441 // } 442 AutoMutex lock(mLock); 443 requestExit(); 444 mWaitWorkCV.broadcast(); 445 } 446 // When Thread::requestExitAndWait is made virtual and this method is renamed to 447 // "virtual status_t requestExitAndWait()", replace by "return Thread::requestExitAndWait();" 448 requestExitAndWait(); 449} 450 451status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs) 452{ 453 status_t status; 454 455 ALOGV("ThreadBase::setParameters() %s", keyValuePairs.string()); 456 Mutex::Autolock _l(mLock); 457 458 return sendSetParameterConfigEvent_l(keyValuePairs); 459} 460 461// sendConfigEvent_l() must be called with ThreadBase::mLock held 462// Can temporarily release the lock if waiting for a reply from processConfigEvents_l(). 463status_t AudioFlinger::ThreadBase::sendConfigEvent_l(sp<ConfigEvent>& event) 464{ 465 status_t status = NO_ERROR; 466 467 mConfigEvents.add(event); 468 ALOGV("sendConfigEvent_l() num events %d event %d", mConfigEvents.size(), event->mType); 469 mWaitWorkCV.signal(); 470 mLock.unlock(); 471 { 472 Mutex::Autolock _l(event->mLock); 473 while (event->mWaitStatus) { 474 if (event->mCond.waitRelative(event->mLock, kConfigEventTimeoutNs) != NO_ERROR) { 475 event->mStatus = TIMED_OUT; 476 event->mWaitStatus = false; 477 } 478 } 479 status = event->mStatus; 480 } 481 mLock.lock(); 482 return status; 483} 484 485void AudioFlinger::ThreadBase::sendIoConfigEvent(int event, int param) 486{ 487 Mutex::Autolock _l(mLock); 488 sendIoConfigEvent_l(event, param); 489} 490 491// sendIoConfigEvent_l() must be called with ThreadBase::mLock held 492void AudioFlinger::ThreadBase::sendIoConfigEvent_l(int event, int param) 493{ 494 sp<ConfigEvent> configEvent = (ConfigEvent *)new IoConfigEvent(event, param); 495 sendConfigEvent_l(configEvent); 496} 497 498// sendPrioConfigEvent_l() must be called with ThreadBase::mLock held 499void AudioFlinger::ThreadBase::sendPrioConfigEvent_l(pid_t pid, pid_t tid, int32_t prio) 500{ 501 sp<ConfigEvent> configEvent = (ConfigEvent *)new PrioConfigEvent(pid, tid, prio); 502 sendConfigEvent_l(configEvent); 503} 504 505// sendSetParameterConfigEvent_l() must be called with ThreadBase::mLock held 506status_t AudioFlinger::ThreadBase::sendSetParameterConfigEvent_l(const String8& keyValuePair) 507{ 508 sp<ConfigEvent> configEvent = (ConfigEvent *)new SetParameterConfigEvent(keyValuePair); 509 return sendConfigEvent_l(configEvent); 510} 511 512status_t AudioFlinger::ThreadBase::sendCreateAudioPatchConfigEvent( 513 const struct audio_patch *patch, 514 audio_patch_handle_t *handle) 515{ 516 Mutex::Autolock _l(mLock); 517 sp<ConfigEvent> configEvent = (ConfigEvent *)new CreateAudioPatchConfigEvent(*patch, *handle); 518 status_t status = sendConfigEvent_l(configEvent); 519 if (status == NO_ERROR) { 520 CreateAudioPatchConfigEventData *data = 521 (CreateAudioPatchConfigEventData *)configEvent->mData.get(); 522 *handle = data->mHandle; 523 } 524 return status; 525} 526 527status_t AudioFlinger::ThreadBase::sendReleaseAudioPatchConfigEvent( 528 const audio_patch_handle_t handle) 529{ 530 Mutex::Autolock _l(mLock); 531 sp<ConfigEvent> configEvent = (ConfigEvent *)new ReleaseAudioPatchConfigEvent(handle); 532 return sendConfigEvent_l(configEvent); 533} 534 535 536// post condition: mConfigEvents.isEmpty() 537void AudioFlinger::ThreadBase::processConfigEvents_l() 538{ 539 bool configChanged = false; 540 541 while (!mConfigEvents.isEmpty()) { 542 ALOGV("processConfigEvents_l() remaining events %d", mConfigEvents.size()); 543 sp<ConfigEvent> event = mConfigEvents[0]; 544 mConfigEvents.removeAt(0); 545 switch (event->mType) { 546 case CFG_EVENT_PRIO: { 547 PrioConfigEventData *data = (PrioConfigEventData *)event->mData.get(); 548 // FIXME Need to understand why this has to be done asynchronously 549 int err = requestPriority(data->mPid, data->mTid, data->mPrio, 550 true /*asynchronous*/); 551 if (err != 0) { 552 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 553 data->mPrio, data->mPid, data->mTid, err); 554 } 555 } break; 556 case CFG_EVENT_IO: { 557 IoConfigEventData *data = (IoConfigEventData *)event->mData.get(); 558 audioConfigChanged(data->mEvent, data->mParam); 559 } break; 560 case CFG_EVENT_SET_PARAMETER: { 561 SetParameterConfigEventData *data = (SetParameterConfigEventData *)event->mData.get(); 562 if (checkForNewParameter_l(data->mKeyValuePairs, event->mStatus)) { 563 configChanged = true; 564 } 565 } break; 566 case CFG_EVENT_CREATE_AUDIO_PATCH: { 567 CreateAudioPatchConfigEventData *data = 568 (CreateAudioPatchConfigEventData *)event->mData.get(); 569 event->mStatus = createAudioPatch_l(&data->mPatch, &data->mHandle); 570 } break; 571 case CFG_EVENT_RELEASE_AUDIO_PATCH: { 572 ReleaseAudioPatchConfigEventData *data = 573 (ReleaseAudioPatchConfigEventData *)event->mData.get(); 574 event->mStatus = releaseAudioPatch_l(data->mHandle); 575 } break; 576 default: 577 ALOG_ASSERT(false, "processConfigEvents_l() unknown event type %d", event->mType); 578 break; 579 } 580 { 581 Mutex::Autolock _l(event->mLock); 582 if (event->mWaitStatus) { 583 event->mWaitStatus = false; 584 event->mCond.signal(); 585 } 586 } 587 ALOGV_IF(mConfigEvents.isEmpty(), "processConfigEvents_l() DONE thread %p", this); 588 } 589 590 if (configChanged) { 591 cacheParameters_l(); 592 } 593} 594 595String8 channelMaskToString(audio_channel_mask_t mask, bool output) { 596 String8 s; 597 if (output) { 598 if (mask & AUDIO_CHANNEL_OUT_FRONT_LEFT) s.append("front-left, "); 599 if (mask & AUDIO_CHANNEL_OUT_FRONT_RIGHT) s.append("front-right, "); 600 if (mask & AUDIO_CHANNEL_OUT_FRONT_CENTER) s.append("front-center, "); 601 if (mask & AUDIO_CHANNEL_OUT_LOW_FREQUENCY) s.append("low freq, "); 602 if (mask & AUDIO_CHANNEL_OUT_BACK_LEFT) s.append("back-left, "); 603 if (mask & AUDIO_CHANNEL_OUT_BACK_RIGHT) s.append("back-right, "); 604 if (mask & AUDIO_CHANNEL_OUT_FRONT_LEFT_OF_CENTER) s.append("front-left-of-center, "); 605 if (mask & AUDIO_CHANNEL_OUT_FRONT_RIGHT_OF_CENTER) s.append("front-right-of-center, "); 606 if (mask & AUDIO_CHANNEL_OUT_BACK_CENTER) s.append("back-center, "); 607 if (mask & AUDIO_CHANNEL_OUT_SIDE_LEFT) s.append("side-left, "); 608 if (mask & AUDIO_CHANNEL_OUT_SIDE_RIGHT) s.append("side-right, "); 609 if (mask & AUDIO_CHANNEL_OUT_TOP_CENTER) s.append("top-center ,"); 610 if (mask & AUDIO_CHANNEL_OUT_TOP_FRONT_LEFT) s.append("top-front-left, "); 611 if (mask & AUDIO_CHANNEL_OUT_TOP_FRONT_CENTER) s.append("top-front-center, "); 612 if (mask & AUDIO_CHANNEL_OUT_TOP_FRONT_RIGHT) s.append("top-front-right, "); 613 if (mask & AUDIO_CHANNEL_OUT_TOP_BACK_LEFT) s.append("top-back-left, "); 614 if (mask & AUDIO_CHANNEL_OUT_TOP_BACK_CENTER) s.append("top-back-center, " ); 615 if (mask & AUDIO_CHANNEL_OUT_TOP_BACK_RIGHT) s.append("top-back-right, " ); 616 if (mask & ~AUDIO_CHANNEL_OUT_ALL) s.append("unknown, "); 617 } else { 618 if (mask & AUDIO_CHANNEL_IN_LEFT) s.append("left, "); 619 if (mask & AUDIO_CHANNEL_IN_RIGHT) s.append("right, "); 620 if (mask & AUDIO_CHANNEL_IN_FRONT) s.append("front, "); 621 if (mask & AUDIO_CHANNEL_IN_BACK) s.append("back, "); 622 if (mask & AUDIO_CHANNEL_IN_LEFT_PROCESSED) s.append("left-processed, "); 623 if (mask & AUDIO_CHANNEL_IN_RIGHT_PROCESSED) s.append("right-processed, "); 624 if (mask & AUDIO_CHANNEL_IN_FRONT_PROCESSED) s.append("front-processed, "); 625 if (mask & AUDIO_CHANNEL_IN_BACK_PROCESSED) s.append("back-processed, "); 626 if (mask & AUDIO_CHANNEL_IN_PRESSURE) s.append("pressure, "); 627 if (mask & AUDIO_CHANNEL_IN_X_AXIS) s.append("X, "); 628 if (mask & AUDIO_CHANNEL_IN_Y_AXIS) s.append("Y, "); 629 if (mask & AUDIO_CHANNEL_IN_Z_AXIS) s.append("Z, "); 630 if (mask & AUDIO_CHANNEL_IN_VOICE_UPLINK) s.append("voice-uplink, "); 631 if (mask & AUDIO_CHANNEL_IN_VOICE_DNLINK) s.append("voice-dnlink, "); 632 if (mask & ~AUDIO_CHANNEL_IN_ALL) s.append("unknown, "); 633 } 634 int len = s.length(); 635 if (s.length() > 2) { 636 char *str = s.lockBuffer(len); 637 s.unlockBuffer(len - 2); 638 } 639 return s; 640} 641 642void AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args __unused) 643{ 644 const size_t SIZE = 256; 645 char buffer[SIZE]; 646 String8 result; 647 648 bool locked = AudioFlinger::dumpTryLock(mLock); 649 if (!locked) { 650 dprintf(fd, "thread %p may be deadlocked\n", this); 651 } 652 653 dprintf(fd, " I/O handle: %d\n", mId); 654 dprintf(fd, " TID: %d\n", getTid()); 655 dprintf(fd, " Standby: %s\n", mStandby ? "yes" : "no"); 656 dprintf(fd, " Sample rate: %u Hz\n", mSampleRate); 657 dprintf(fd, " HAL frame count: %zu\n", mFrameCount); 658 dprintf(fd, " HAL format: 0x%x (%s)\n", mHALFormat, formatToString(mHALFormat)); 659 dprintf(fd, " HAL buffer size: %u bytes\n", mBufferSize); 660 dprintf(fd, " Channel count: %u\n", mChannelCount); 661 dprintf(fd, " Channel mask: 0x%08x (%s)\n", mChannelMask, 662 channelMaskToString(mChannelMask, mType != RECORD).string()); 663 dprintf(fd, " Format: 0x%x (%s)\n", mFormat, formatToString(mFormat)); 664 dprintf(fd, " Frame size: %zu bytes\n", mFrameSize); 665 dprintf(fd, " Pending config events:"); 666 size_t numConfig = mConfigEvents.size(); 667 if (numConfig) { 668 for (size_t i = 0; i < numConfig; i++) { 669 mConfigEvents[i]->dump(buffer, SIZE); 670 dprintf(fd, "\n %s", buffer); 671 } 672 dprintf(fd, "\n"); 673 } else { 674 dprintf(fd, " none\n"); 675 } 676 677 if (locked) { 678 mLock.unlock(); 679 } 680} 681 682void AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args) 683{ 684 const size_t SIZE = 256; 685 char buffer[SIZE]; 686 String8 result; 687 688 size_t numEffectChains = mEffectChains.size(); 689 snprintf(buffer, SIZE, " %zu Effect Chains\n", numEffectChains); 690 write(fd, buffer, strlen(buffer)); 691 692 for (size_t i = 0; i < numEffectChains; ++i) { 693 sp<EffectChain> chain = mEffectChains[i]; 694 if (chain != 0) { 695 chain->dump(fd, args); 696 } 697 } 698} 699 700void AudioFlinger::ThreadBase::acquireWakeLock(int uid) 701{ 702 Mutex::Autolock _l(mLock); 703 acquireWakeLock_l(uid); 704} 705 706String16 AudioFlinger::ThreadBase::getWakeLockTag() 707{ 708 switch (mType) { 709 case MIXER: 710 return String16("AudioMix"); 711 case DIRECT: 712 return String16("AudioDirectOut"); 713 case DUPLICATING: 714 return String16("AudioDup"); 715 case RECORD: 716 return String16("AudioIn"); 717 case OFFLOAD: 718 return String16("AudioOffload"); 719 default: 720 ALOG_ASSERT(false); 721 return String16("AudioUnknown"); 722 } 723} 724 725void AudioFlinger::ThreadBase::acquireWakeLock_l(int uid) 726{ 727 getPowerManager_l(); 728 if (mPowerManager != 0) { 729 sp<IBinder> binder = new BBinder(); 730 status_t status; 731 if (uid >= 0) { 732 status = mPowerManager->acquireWakeLockWithUid(POWERMANAGER_PARTIAL_WAKE_LOCK, 733 binder, 734 getWakeLockTag(), 735 String16("media"), 736 uid, 737 true /* FIXME force oneway contrary to .aidl */); 738 } else { 739 status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK, 740 binder, 741 getWakeLockTag(), 742 String16("media"), 743 true /* FIXME force oneway contrary to .aidl */); 744 } 745 if (status == NO_ERROR) { 746 mWakeLockToken = binder; 747 } 748 ALOGV("acquireWakeLock_l() %s status %d", mName, status); 749 } 750} 751 752void AudioFlinger::ThreadBase::releaseWakeLock() 753{ 754 Mutex::Autolock _l(mLock); 755 releaseWakeLock_l(); 756} 757 758void AudioFlinger::ThreadBase::releaseWakeLock_l() 759{ 760 if (mWakeLockToken != 0) { 761 ALOGV("releaseWakeLock_l() %s", mName); 762 if (mPowerManager != 0) { 763 mPowerManager->releaseWakeLock(mWakeLockToken, 0, 764 true /* FIXME force oneway contrary to .aidl */); 765 } 766 mWakeLockToken.clear(); 767 } 768} 769 770void AudioFlinger::ThreadBase::updateWakeLockUids(const SortedVector<int> &uids) { 771 Mutex::Autolock _l(mLock); 772 updateWakeLockUids_l(uids); 773} 774 775void AudioFlinger::ThreadBase::getPowerManager_l() { 776 777 if (mPowerManager == 0) { 778 // use checkService() to avoid blocking if power service is not up yet 779 sp<IBinder> binder = 780 defaultServiceManager()->checkService(String16("power")); 781 if (binder == 0) { 782 ALOGW("Thread %s cannot connect to the power manager service", mName); 783 } else { 784 mPowerManager = interface_cast<IPowerManager>(binder); 785 binder->linkToDeath(mDeathRecipient); 786 } 787 } 788} 789 790void AudioFlinger::ThreadBase::updateWakeLockUids_l(const SortedVector<int> &uids) { 791 792 getPowerManager_l(); 793 if (mWakeLockToken == NULL) { 794 ALOGE("no wake lock to update!"); 795 return; 796 } 797 if (mPowerManager != 0) { 798 sp<IBinder> binder = new BBinder(); 799 status_t status; 800 status = mPowerManager->updateWakeLockUids(mWakeLockToken, uids.size(), uids.array(), 801 true /* FIXME force oneway contrary to .aidl */); 802 ALOGV("acquireWakeLock_l() %s status %d", mName, status); 803 } 804} 805 806void AudioFlinger::ThreadBase::clearPowerManager() 807{ 808 Mutex::Autolock _l(mLock); 809 releaseWakeLock_l(); 810 mPowerManager.clear(); 811} 812 813void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who __unused) 814{ 815 sp<ThreadBase> thread = mThread.promote(); 816 if (thread != 0) { 817 thread->clearPowerManager(); 818 } 819 ALOGW("power manager service died !!!"); 820} 821 822void AudioFlinger::ThreadBase::setEffectSuspended( 823 const effect_uuid_t *type, bool suspend, int sessionId) 824{ 825 Mutex::Autolock _l(mLock); 826 setEffectSuspended_l(type, suspend, sessionId); 827} 828 829void AudioFlinger::ThreadBase::setEffectSuspended_l( 830 const effect_uuid_t *type, bool suspend, int sessionId) 831{ 832 sp<EffectChain> chain = getEffectChain_l(sessionId); 833 if (chain != 0) { 834 if (type != NULL) { 835 chain->setEffectSuspended_l(type, suspend); 836 } else { 837 chain->setEffectSuspendedAll_l(suspend); 838 } 839 } 840 841 updateSuspendedSessions_l(type, suspend, sessionId); 842} 843 844void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain) 845{ 846 ssize_t index = mSuspendedSessions.indexOfKey(chain->sessionId()); 847 if (index < 0) { 848 return; 849 } 850 851 const KeyedVector <int, sp<SuspendedSessionDesc> >& sessionEffects = 852 mSuspendedSessions.valueAt(index); 853 854 for (size_t i = 0; i < sessionEffects.size(); i++) { 855 sp<SuspendedSessionDesc> desc = sessionEffects.valueAt(i); 856 for (int j = 0; j < desc->mRefCount; j++) { 857 if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) { 858 chain->setEffectSuspendedAll_l(true); 859 } else { 860 ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x", 861 desc->mType.timeLow); 862 chain->setEffectSuspended_l(&desc->mType, true); 863 } 864 } 865 } 866} 867 868void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type, 869 bool suspend, 870 int sessionId) 871{ 872 ssize_t index = mSuspendedSessions.indexOfKey(sessionId); 873 874 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects; 875 876 if (suspend) { 877 if (index >= 0) { 878 sessionEffects = mSuspendedSessions.valueAt(index); 879 } else { 880 mSuspendedSessions.add(sessionId, sessionEffects); 881 } 882 } else { 883 if (index < 0) { 884 return; 885 } 886 sessionEffects = mSuspendedSessions.valueAt(index); 887 } 888 889 890 int key = EffectChain::kKeyForSuspendAll; 891 if (type != NULL) { 892 key = type->timeLow; 893 } 894 index = sessionEffects.indexOfKey(key); 895 896 sp<SuspendedSessionDesc> desc; 897 if (suspend) { 898 if (index >= 0) { 899 desc = sessionEffects.valueAt(index); 900 } else { 901 desc = new SuspendedSessionDesc(); 902 if (type != NULL) { 903 desc->mType = *type; 904 } 905 sessionEffects.add(key, desc); 906 ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key); 907 } 908 desc->mRefCount++; 909 } else { 910 if (index < 0) { 911 return; 912 } 913 desc = sessionEffects.valueAt(index); 914 if (--desc->mRefCount == 0) { 915 ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key); 916 sessionEffects.removeItemsAt(index); 917 if (sessionEffects.isEmpty()) { 918 ALOGV("updateSuspendedSessions_l() restore removing session %d", 919 sessionId); 920 mSuspendedSessions.removeItem(sessionId); 921 } 922 } 923 } 924 if (!sessionEffects.isEmpty()) { 925 mSuspendedSessions.replaceValueFor(sessionId, sessionEffects); 926 } 927} 928 929void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 930 bool enabled, 931 int sessionId) 932{ 933 Mutex::Autolock _l(mLock); 934 checkSuspendOnEffectEnabled_l(effect, enabled, sessionId); 935} 936 937void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 938 bool enabled, 939 int sessionId) 940{ 941 if (mType != RECORD) { 942 // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on 943 // another session. This gives the priority to well behaved effect control panels 944 // and applications not using global effects. 945 // Enabling post processing in AUDIO_SESSION_OUTPUT_STAGE session does not affect 946 // global effects 947 if ((sessionId != AUDIO_SESSION_OUTPUT_MIX) && (sessionId != AUDIO_SESSION_OUTPUT_STAGE)) { 948 setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX); 949 } 950 } 951 952 sp<EffectChain> chain = getEffectChain_l(sessionId); 953 if (chain != 0) { 954 chain->checkSuspendOnEffectEnabled(effect, enabled); 955 } 956} 957 958// ThreadBase::createEffect_l() must be called with AudioFlinger::mLock held 959sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 960 const sp<AudioFlinger::Client>& client, 961 const sp<IEffectClient>& effectClient, 962 int32_t priority, 963 int sessionId, 964 effect_descriptor_t *desc, 965 int *enabled, 966 status_t *status) 967{ 968 sp<EffectModule> effect; 969 sp<EffectHandle> handle; 970 status_t lStatus; 971 sp<EffectChain> chain; 972 bool chainCreated = false; 973 bool effectCreated = false; 974 bool effectRegistered = false; 975 976 lStatus = initCheck(); 977 if (lStatus != NO_ERROR) { 978 ALOGW("createEffect_l() Audio driver not initialized."); 979 goto Exit; 980 } 981 982 // Reject any effect on Direct output threads for now, since the format of 983 // mSinkBuffer is not guaranteed to be compatible with effect processing (PCM 16 stereo). 984 if (mType == DIRECT) { 985 ALOGW("createEffect_l() Cannot add effect %s on Direct output type thread %s", 986 desc->name, mName); 987 lStatus = BAD_VALUE; 988 goto Exit; 989 } 990 991 // Reject any effect on mixer or duplicating multichannel sinks. 992 // TODO: fix both format and multichannel issues with effects. 993 if ((mType == MIXER || mType == DUPLICATING) && mChannelCount != FCC_2) { 994 ALOGW("createEffect_l() Cannot add effect %s for multichannel(%d) %s threads", 995 desc->name, mChannelCount, mType == MIXER ? "MIXER" : "DUPLICATING"); 996 lStatus = BAD_VALUE; 997 goto Exit; 998 } 999 1000 // Allow global effects only on offloaded and mixer threads 1001 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 1002 switch (mType) { 1003 case MIXER: 1004 case OFFLOAD: 1005 break; 1006 case DIRECT: 1007 case DUPLICATING: 1008 case RECORD: 1009 default: 1010 ALOGW("createEffect_l() Cannot add global effect %s on thread %s", desc->name, mName); 1011 lStatus = BAD_VALUE; 1012 goto Exit; 1013 } 1014 } 1015 1016 // Only Pre processor effects are allowed on input threads and only on input threads 1017 if ((mType == RECORD) != ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { 1018 ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d", 1019 desc->name, desc->flags, mType); 1020 lStatus = BAD_VALUE; 1021 goto Exit; 1022 } 1023 1024 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 1025 1026 { // scope for mLock 1027 Mutex::Autolock _l(mLock); 1028 1029 // check for existing effect chain with the requested audio session 1030 chain = getEffectChain_l(sessionId); 1031 if (chain == 0) { 1032 // create a new chain for this session 1033 ALOGV("createEffect_l() new effect chain for session %d", sessionId); 1034 chain = new EffectChain(this, sessionId); 1035 addEffectChain_l(chain); 1036 chain->setStrategy(getStrategyForSession_l(sessionId)); 1037 chainCreated = true; 1038 } else { 1039 effect = chain->getEffectFromDesc_l(desc); 1040 } 1041 1042 ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get()); 1043 1044 if (effect == 0) { 1045 int id = mAudioFlinger->nextUniqueId(); 1046 // Check CPU and memory usage 1047 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 1048 if (lStatus != NO_ERROR) { 1049 goto Exit; 1050 } 1051 effectRegistered = true; 1052 // create a new effect module if none present in the chain 1053 effect = new EffectModule(this, chain, desc, id, sessionId); 1054 lStatus = effect->status(); 1055 if (lStatus != NO_ERROR) { 1056 goto Exit; 1057 } 1058 effect->setOffloaded(mType == OFFLOAD, mId); 1059 1060 lStatus = chain->addEffect_l(effect); 1061 if (lStatus != NO_ERROR) { 1062 goto Exit; 1063 } 1064 effectCreated = true; 1065 1066 effect->setDevice(mOutDevice); 1067 effect->setDevice(mInDevice); 1068 effect->setMode(mAudioFlinger->getMode()); 1069 effect->setAudioSource(mAudioSource); 1070 } 1071 // create effect handle and connect it to effect module 1072 handle = new EffectHandle(effect, client, effectClient, priority); 1073 lStatus = handle->initCheck(); 1074 if (lStatus == OK) { 1075 lStatus = effect->addHandle(handle.get()); 1076 } 1077 if (enabled != NULL) { 1078 *enabled = (int)effect->isEnabled(); 1079 } 1080 } 1081 1082Exit: 1083 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 1084 Mutex::Autolock _l(mLock); 1085 if (effectCreated) { 1086 chain->removeEffect_l(effect); 1087 } 1088 if (effectRegistered) { 1089 AudioSystem::unregisterEffect(effect->id()); 1090 } 1091 if (chainCreated) { 1092 removeEffectChain_l(chain); 1093 } 1094 handle.clear(); 1095 } 1096 1097 *status = lStatus; 1098 return handle; 1099} 1100 1101sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect(int sessionId, int effectId) 1102{ 1103 Mutex::Autolock _l(mLock); 1104 return getEffect_l(sessionId, effectId); 1105} 1106 1107sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId) 1108{ 1109 sp<EffectChain> chain = getEffectChain_l(sessionId); 1110 return chain != 0 ? chain->getEffectFromId_l(effectId) : 0; 1111} 1112 1113// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 1114// PlaybackThread::mLock held 1115status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 1116{ 1117 // check for existing effect chain with the requested audio session 1118 int sessionId = effect->sessionId(); 1119 sp<EffectChain> chain = getEffectChain_l(sessionId); 1120 bool chainCreated = false; 1121 1122 ALOGD_IF((mType == OFFLOAD) && !effect->isOffloadable(), 1123 "addEffect_l() on offloaded thread %p: effect %s does not support offload flags %x", 1124 this, effect->desc().name, effect->desc().flags); 1125 1126 if (chain == 0) { 1127 // create a new chain for this session 1128 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 1129 chain = new EffectChain(this, sessionId); 1130 addEffectChain_l(chain); 1131 chain->setStrategy(getStrategyForSession_l(sessionId)); 1132 chainCreated = true; 1133 } 1134 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 1135 1136 if (chain->getEffectFromId_l(effect->id()) != 0) { 1137 ALOGW("addEffect_l() %p effect %s already present in chain %p", 1138 this, effect->desc().name, chain.get()); 1139 return BAD_VALUE; 1140 } 1141 1142 effect->setOffloaded(mType == OFFLOAD, mId); 1143 1144 status_t status = chain->addEffect_l(effect); 1145 if (status != NO_ERROR) { 1146 if (chainCreated) { 1147 removeEffectChain_l(chain); 1148 } 1149 return status; 1150 } 1151 1152 effect->setDevice(mOutDevice); 1153 effect->setDevice(mInDevice); 1154 effect->setMode(mAudioFlinger->getMode()); 1155 effect->setAudioSource(mAudioSource); 1156 return NO_ERROR; 1157} 1158 1159void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 1160 1161 ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 1162 effect_descriptor_t desc = effect->desc(); 1163 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 1164 detachAuxEffect_l(effect->id()); 1165 } 1166 1167 sp<EffectChain> chain = effect->chain().promote(); 1168 if (chain != 0) { 1169 // remove effect chain if removing last effect 1170 if (chain->removeEffect_l(effect) == 0) { 1171 removeEffectChain_l(chain); 1172 } 1173 } else { 1174 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 1175 } 1176} 1177 1178void AudioFlinger::ThreadBase::lockEffectChains_l( 1179 Vector< sp<AudioFlinger::EffectChain> >& effectChains) 1180{ 1181 effectChains = mEffectChains; 1182 for (size_t i = 0; i < mEffectChains.size(); i++) { 1183 mEffectChains[i]->lock(); 1184 } 1185} 1186 1187void AudioFlinger::ThreadBase::unlockEffectChains( 1188 const Vector< sp<AudioFlinger::EffectChain> >& effectChains) 1189{ 1190 for (size_t i = 0; i < effectChains.size(); i++) { 1191 effectChains[i]->unlock(); 1192 } 1193} 1194 1195sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId) 1196{ 1197 Mutex::Autolock _l(mLock); 1198 return getEffectChain_l(sessionId); 1199} 1200 1201sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) const 1202{ 1203 size_t size = mEffectChains.size(); 1204 for (size_t i = 0; i < size; i++) { 1205 if (mEffectChains[i]->sessionId() == sessionId) { 1206 return mEffectChains[i]; 1207 } 1208 } 1209 return 0; 1210} 1211 1212void AudioFlinger::ThreadBase::setMode(audio_mode_t mode) 1213{ 1214 Mutex::Autolock _l(mLock); 1215 size_t size = mEffectChains.size(); 1216 for (size_t i = 0; i < size; i++) { 1217 mEffectChains[i]->setMode_l(mode); 1218 } 1219} 1220 1221void AudioFlinger::ThreadBase::getAudioPortConfig(struct audio_port_config *config) 1222{ 1223 config->type = AUDIO_PORT_TYPE_MIX; 1224 config->ext.mix.handle = mId; 1225 config->sample_rate = mSampleRate; 1226 config->format = mFormat; 1227 config->channel_mask = mChannelMask; 1228 config->config_mask = AUDIO_PORT_CONFIG_SAMPLE_RATE|AUDIO_PORT_CONFIG_CHANNEL_MASK| 1229 AUDIO_PORT_CONFIG_FORMAT; 1230} 1231 1232 1233// ---------------------------------------------------------------------------- 1234// Playback 1235// ---------------------------------------------------------------------------- 1236 1237AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, 1238 AudioStreamOut* output, 1239 audio_io_handle_t id, 1240 audio_devices_t device, 1241 type_t type) 1242 : ThreadBase(audioFlinger, id, device, AUDIO_DEVICE_NONE, type), 1243 mNormalFrameCount(0), mSinkBuffer(NULL), 1244 mMixerBufferEnabled(AudioFlinger::kEnableExtendedPrecision), 1245 mMixerBuffer(NULL), 1246 mMixerBufferSize(0), 1247 mMixerBufferFormat(AUDIO_FORMAT_INVALID), 1248 mMixerBufferValid(false), 1249 mEffectBufferEnabled(AudioFlinger::kEnableExtendedPrecision), 1250 mEffectBuffer(NULL), 1251 mEffectBufferSize(0), 1252 mEffectBufferFormat(AUDIO_FORMAT_INVALID), 1253 mEffectBufferValid(false), 1254 mSuspended(0), mBytesWritten(0), 1255 mActiveTracksGeneration(0), 1256 // mStreamTypes[] initialized in constructor body 1257 mOutput(output), 1258 mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false), 1259 mMixerStatus(MIXER_IDLE), 1260 mMixerStatusIgnoringFastTracks(MIXER_IDLE), 1261 standbyDelay(AudioFlinger::mStandbyTimeInNsecs), 1262 mBytesRemaining(0), 1263 mCurrentWriteLength(0), 1264 mUseAsyncWrite(false), 1265 mWriteAckSequence(0), 1266 mDrainSequence(0), 1267 mSignalPending(false), 1268 mScreenState(AudioFlinger::mScreenState), 1269 // index 0 is reserved for normal mixer's submix 1270 mFastTrackAvailMask(((1 << FastMixerState::kMaxFastTracks) - 1) & ~1), 1271 mHwSupportsPause(false), mHwPaused(false), mFlushPending(false), 1272 // mLatchD, mLatchQ, 1273 mLatchDValid(false), mLatchQValid(false) 1274{ 1275 snprintf(mName, kNameLength, "AudioOut_%X", id); 1276 mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mName); 1277 1278 // Assumes constructor is called by AudioFlinger with it's mLock held, but 1279 // it would be safer to explicitly pass initial masterVolume/masterMute as 1280 // parameter. 1281 // 1282 // If the HAL we are using has support for master volume or master mute, 1283 // then do not attenuate or mute during mixing (just leave the volume at 1.0 1284 // and the mute set to false). 1285 mMasterVolume = audioFlinger->masterVolume_l(); 1286 mMasterMute = audioFlinger->masterMute_l(); 1287 if (mOutput && mOutput->audioHwDev) { 1288 if (mOutput->audioHwDev->canSetMasterVolume()) { 1289 mMasterVolume = 1.0; 1290 } 1291 1292 if (mOutput->audioHwDev->canSetMasterMute()) { 1293 mMasterMute = false; 1294 } 1295 } 1296 1297 readOutputParameters_l(); 1298 1299 // ++ operator does not compile 1300 for (audio_stream_type_t stream = AUDIO_STREAM_MIN; stream < AUDIO_STREAM_CNT; 1301 stream = (audio_stream_type_t) (stream + 1)) { 1302 mStreamTypes[stream].volume = mAudioFlinger->streamVolume_l(stream); 1303 mStreamTypes[stream].mute = mAudioFlinger->streamMute_l(stream); 1304 } 1305} 1306 1307AudioFlinger::PlaybackThread::~PlaybackThread() 1308{ 1309 mAudioFlinger->unregisterWriter(mNBLogWriter); 1310 free(mSinkBuffer); 1311 free(mMixerBuffer); 1312 free(mEffectBuffer); 1313} 1314 1315void AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) 1316{ 1317 dumpInternals(fd, args); 1318 dumpTracks(fd, args); 1319 dumpEffectChains(fd, args); 1320} 1321 1322void AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args __unused) 1323{ 1324 const size_t SIZE = 256; 1325 char buffer[SIZE]; 1326 String8 result; 1327 1328 result.appendFormat(" Stream volumes in dB: "); 1329 for (int i = 0; i < AUDIO_STREAM_CNT; ++i) { 1330 const stream_type_t *st = &mStreamTypes[i]; 1331 if (i > 0) { 1332 result.appendFormat(", "); 1333 } 1334 result.appendFormat("%d:%.2g", i, 20.0 * log10(st->volume)); 1335 if (st->mute) { 1336 result.append("M"); 1337 } 1338 } 1339 result.append("\n"); 1340 write(fd, result.string(), result.length()); 1341 result.clear(); 1342 1343 // These values are "raw"; they will wrap around. See prepareTracks_l() for a better way. 1344 FastTrackUnderruns underruns = getFastTrackUnderruns(0); 1345 dprintf(fd, " Normal mixer raw underrun counters: partial=%u empty=%u\n", 1346 underruns.mBitFields.mPartial, underruns.mBitFields.mEmpty); 1347 1348 size_t numtracks = mTracks.size(); 1349 size_t numactive = mActiveTracks.size(); 1350 dprintf(fd, " %d Tracks", numtracks); 1351 size_t numactiveseen = 0; 1352 if (numtracks) { 1353 dprintf(fd, " of which %d are active\n", numactive); 1354 Track::appendDumpHeader(result); 1355 for (size_t i = 0; i < numtracks; ++i) { 1356 sp<Track> track = mTracks[i]; 1357 if (track != 0) { 1358 bool active = mActiveTracks.indexOf(track) >= 0; 1359 if (active) { 1360 numactiveseen++; 1361 } 1362 track->dump(buffer, SIZE, active); 1363 result.append(buffer); 1364 } 1365 } 1366 } else { 1367 result.append("\n"); 1368 } 1369 if (numactiveseen != numactive) { 1370 // some tracks in the active list were not in the tracks list 1371 snprintf(buffer, SIZE, " The following tracks are in the active list but" 1372 " not in the track list\n"); 1373 result.append(buffer); 1374 Track::appendDumpHeader(result); 1375 for (size_t i = 0; i < numactive; ++i) { 1376 sp<Track> track = mActiveTracks[i].promote(); 1377 if (track != 0 && mTracks.indexOf(track) < 0) { 1378 track->dump(buffer, SIZE, true); 1379 result.append(buffer); 1380 } 1381 } 1382 } 1383 1384 write(fd, result.string(), result.size()); 1385} 1386 1387void AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 1388{ 1389 dprintf(fd, "\nOutput thread %p type %d (%s):\n", this, type(), threadTypeToString(type())); 1390 dprintf(fd, " Normal frame count: %zu\n", mNormalFrameCount); 1391 dprintf(fd, " Last write occurred (msecs): %llu\n", ns2ms(systemTime() - mLastWriteTime)); 1392 dprintf(fd, " Total writes: %d\n", mNumWrites); 1393 dprintf(fd, " Delayed writes: %d\n", mNumDelayedWrites); 1394 dprintf(fd, " Blocked in write: %s\n", mInWrite ? "yes" : "no"); 1395 dprintf(fd, " Suspend count: %d\n", mSuspended); 1396 dprintf(fd, " Sink buffer : %p\n", mSinkBuffer); 1397 dprintf(fd, " Mixer buffer: %p\n", mMixerBuffer); 1398 dprintf(fd, " Effect buffer: %p\n", mEffectBuffer); 1399 dprintf(fd, " Fast track availMask=%#x\n", mFastTrackAvailMask); 1400 AudioStreamOut *output = mOutput; 1401 audio_output_flags_t flags = output != NULL ? output->flags : AUDIO_OUTPUT_FLAG_NONE; 1402 String8 flagsAsString = outputFlagsToString(flags); 1403 dprintf(fd, " AudioStreamOut: %p flags %#x (%s)\n", output, flags, flagsAsString.string()); 1404 1405 dumpBase(fd, args); 1406} 1407 1408// Thread virtuals 1409 1410void AudioFlinger::PlaybackThread::onFirstRef() 1411{ 1412 run(mName, ANDROID_PRIORITY_URGENT_AUDIO); 1413} 1414 1415// ThreadBase virtuals 1416void AudioFlinger::PlaybackThread::preExit() 1417{ 1418 ALOGV(" preExit()"); 1419 // FIXME this is using hard-coded strings but in the future, this functionality will be 1420 // converted to use audio HAL extensions required to support tunneling 1421 mOutput->stream->common.set_parameters(&mOutput->stream->common, "exiting=1"); 1422} 1423 1424// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 1425sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 1426 const sp<AudioFlinger::Client>& client, 1427 audio_stream_type_t streamType, 1428 uint32_t sampleRate, 1429 audio_format_t format, 1430 audio_channel_mask_t channelMask, 1431 size_t *pFrameCount, 1432 const sp<IMemory>& sharedBuffer, 1433 int sessionId, 1434 IAudioFlinger::track_flags_t *flags, 1435 pid_t tid, 1436 int uid, 1437 status_t *status) 1438{ 1439 size_t frameCount = *pFrameCount; 1440 sp<Track> track; 1441 status_t lStatus; 1442 1443 bool isTimed = (*flags & IAudioFlinger::TRACK_TIMED) != 0; 1444 1445 // client expresses a preference for FAST, but we get the final say 1446 if (*flags & IAudioFlinger::TRACK_FAST) { 1447 if ( 1448 // not timed 1449 (!isTimed) && 1450 // either of these use cases: 1451 ( 1452 // use case 1: shared buffer with any frame count 1453 ( 1454 (sharedBuffer != 0) 1455 ) || 1456 // use case 2: callback handler and frame count is default or at least as large as HAL 1457 ( 1458 (tid != -1) && 1459 ((frameCount == 0) || 1460 (frameCount >= mFrameCount)) 1461 ) 1462 ) && 1463 // PCM data 1464 audio_is_linear_pcm(format) && 1465 // identical channel mask to sink, or mono in and stereo sink 1466 (channelMask == mChannelMask || 1467 (channelMask == AUDIO_CHANNEL_OUT_MONO && 1468 mChannelMask == AUDIO_CHANNEL_OUT_STEREO)) && 1469 // hardware sample rate 1470 (sampleRate == mSampleRate) && 1471 // normal mixer has an associated fast mixer 1472 hasFastMixer() && 1473 // there are sufficient fast track slots available 1474 (mFastTrackAvailMask != 0) 1475 // FIXME test that MixerThread for this fast track has a capable output HAL 1476 // FIXME add a permission test also? 1477 ) { 1478 // if frameCount not specified, then it defaults to fast mixer (HAL) frame count 1479 if (frameCount == 0) { 1480 // read the fast track multiplier property the first time it is needed 1481 int ok = pthread_once(&sFastTrackMultiplierOnce, sFastTrackMultiplierInit); 1482 if (ok != 0) { 1483 ALOGE("%s pthread_once failed: %d", __func__, ok); 1484 } 1485 frameCount = mFrameCount * sFastTrackMultiplier; 1486 } 1487 ALOGV("AUDIO_OUTPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d", 1488 frameCount, mFrameCount); 1489 } else { 1490 ALOGV("AUDIO_OUTPUT_FLAG_FAST denied: isTimed=%d sharedBuffer=%p frameCount=%d " 1491 "mFrameCount=%d format=%#x mFormat=%#x isLinear=%d channelMask=%#x " 1492 "sampleRate=%u mSampleRate=%u " 1493 "hasFastMixer=%d tid=%d fastTrackAvailMask=%#x", 1494 isTimed, sharedBuffer.get(), frameCount, mFrameCount, format, mFormat, 1495 audio_is_linear_pcm(format), 1496 channelMask, sampleRate, mSampleRate, hasFastMixer(), tid, mFastTrackAvailMask); 1497 *flags &= ~IAudioFlinger::TRACK_FAST; 1498 // For compatibility with AudioTrack calculation, buffer depth is forced 1499 // to be at least 2 x the normal mixer frame count and cover audio hardware latency. 1500 // This is probably too conservative, but legacy application code may depend on it. 1501 // If you change this calculation, also review the start threshold which is related. 1502 uint32_t latencyMs = mOutput->stream->get_latency(mOutput->stream); 1503 uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate); 1504 if (minBufCount < 2) { 1505 minBufCount = 2; 1506 } 1507 size_t minFrameCount = mNormalFrameCount * minBufCount; 1508 if (frameCount < minFrameCount) { 1509 frameCount = minFrameCount; 1510 } 1511 } 1512 } 1513 *pFrameCount = frameCount; 1514 1515 switch (mType) { 1516 1517 case DIRECT: 1518 if (audio_is_linear_pcm(format)) { 1519 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1520 ALOGE("createTrack_l() Bad parameter: sampleRate %u format %#x, channelMask 0x%08x " 1521 "for output %p with format %#x", 1522 sampleRate, format, channelMask, mOutput, mFormat); 1523 lStatus = BAD_VALUE; 1524 goto Exit; 1525 } 1526 } 1527 break; 1528 1529 case OFFLOAD: 1530 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1531 ALOGE("createTrack_l() Bad parameter: sampleRate %d format %#x, channelMask 0x%08x \"" 1532 "for output %p with format %#x", 1533 sampleRate, format, channelMask, mOutput, mFormat); 1534 lStatus = BAD_VALUE; 1535 goto Exit; 1536 } 1537 break; 1538 1539 default: 1540 if (!audio_is_linear_pcm(format)) { 1541 ALOGE("createTrack_l() Bad parameter: format %#x \"" 1542 "for output %p with format %#x", 1543 format, mOutput, mFormat); 1544 lStatus = BAD_VALUE; 1545 goto Exit; 1546 } 1547 if (sampleRate > mSampleRate * AUDIO_RESAMPLER_DOWN_RATIO_MAX) { 1548 ALOGE("Sample rate out of range: %u mSampleRate %u", sampleRate, mSampleRate); 1549 lStatus = BAD_VALUE; 1550 goto Exit; 1551 } 1552 break; 1553 1554 } 1555 1556 lStatus = initCheck(); 1557 if (lStatus != NO_ERROR) { 1558 ALOGE("createTrack_l() audio driver not initialized"); 1559 goto Exit; 1560 } 1561 1562 { // scope for mLock 1563 Mutex::Autolock _l(mLock); 1564 1565 // all tracks in same audio session must share the same routing strategy otherwise 1566 // conflicts will happen when tracks are moved from one output to another by audio policy 1567 // manager 1568 uint32_t strategy = AudioSystem::getStrategyForStream(streamType); 1569 for (size_t i = 0; i < mTracks.size(); ++i) { 1570 sp<Track> t = mTracks[i]; 1571 if (t != 0 && t->isExternalTrack()) { 1572 uint32_t actual = AudioSystem::getStrategyForStream(t->streamType()); 1573 if (sessionId == t->sessionId() && strategy != actual) { 1574 ALOGE("createTrack_l() mismatched strategy; expected %u but found %u", 1575 strategy, actual); 1576 lStatus = BAD_VALUE; 1577 goto Exit; 1578 } 1579 } 1580 } 1581 1582 if (!isTimed) { 1583 track = new Track(this, client, streamType, sampleRate, format, 1584 channelMask, frameCount, NULL, sharedBuffer, 1585 sessionId, uid, *flags, TrackBase::TYPE_DEFAULT); 1586 } else { 1587 track = TimedTrack::create(this, client, streamType, sampleRate, format, 1588 channelMask, frameCount, sharedBuffer, sessionId, uid); 1589 } 1590 1591 // new Track always returns non-NULL, 1592 // but TimedTrack::create() is a factory that could fail by returning NULL 1593 lStatus = track != 0 ? track->initCheck() : (status_t) NO_MEMORY; 1594 if (lStatus != NO_ERROR) { 1595 ALOGE("createTrack_l() initCheck failed %d; no control block?", lStatus); 1596 // track must be cleared from the caller as the caller has the AF lock 1597 goto Exit; 1598 } 1599 mTracks.add(track); 1600 1601 sp<EffectChain> chain = getEffectChain_l(sessionId); 1602 if (chain != 0) { 1603 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1604 track->setMainBuffer(chain->inBuffer()); 1605 chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType())); 1606 chain->incTrackCnt(); 1607 } 1608 1609 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 1610 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 1611 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 1612 // so ask activity manager to do this on our behalf 1613 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 1614 } 1615 } 1616 1617 lStatus = NO_ERROR; 1618 1619Exit: 1620 *status = lStatus; 1621 return track; 1622} 1623 1624uint32_t AudioFlinger::PlaybackThread::correctLatency_l(uint32_t latency) const 1625{ 1626 return latency; 1627} 1628 1629uint32_t AudioFlinger::PlaybackThread::latency() const 1630{ 1631 Mutex::Autolock _l(mLock); 1632 return latency_l(); 1633} 1634uint32_t AudioFlinger::PlaybackThread::latency_l() const 1635{ 1636 if (initCheck() == NO_ERROR) { 1637 return correctLatency_l(mOutput->stream->get_latency(mOutput->stream)); 1638 } else { 1639 return 0; 1640 } 1641} 1642 1643void AudioFlinger::PlaybackThread::setMasterVolume(float value) 1644{ 1645 Mutex::Autolock _l(mLock); 1646 // Don't apply master volume in SW if our HAL can do it for us. 1647 if (mOutput && mOutput->audioHwDev && 1648 mOutput->audioHwDev->canSetMasterVolume()) { 1649 mMasterVolume = 1.0; 1650 } else { 1651 mMasterVolume = value; 1652 } 1653} 1654 1655void AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1656{ 1657 Mutex::Autolock _l(mLock); 1658 // Don't apply master mute in SW if our HAL can do it for us. 1659 if (mOutput && mOutput->audioHwDev && 1660 mOutput->audioHwDev->canSetMasterMute()) { 1661 mMasterMute = false; 1662 } else { 1663 mMasterMute = muted; 1664 } 1665} 1666 1667void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) 1668{ 1669 Mutex::Autolock _l(mLock); 1670 mStreamTypes[stream].volume = value; 1671 broadcast_l(); 1672} 1673 1674void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) 1675{ 1676 Mutex::Autolock _l(mLock); 1677 mStreamTypes[stream].mute = muted; 1678 broadcast_l(); 1679} 1680 1681float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const 1682{ 1683 Mutex::Autolock _l(mLock); 1684 return mStreamTypes[stream].volume; 1685} 1686 1687// addTrack_l() must be called with ThreadBase::mLock held 1688status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 1689{ 1690 status_t status = ALREADY_EXISTS; 1691 1692 // set retry count for buffer fill 1693 track->mRetryCount = kMaxTrackStartupRetries; 1694 if (mActiveTracks.indexOf(track) < 0) { 1695 // the track is newly added, make sure it fills up all its 1696 // buffers before playing. This is to ensure the client will 1697 // effectively get the latency it requested. 1698 if (track->isExternalTrack()) { 1699 TrackBase::track_state state = track->mState; 1700 mLock.unlock(); 1701 status = AudioSystem::startOutput(mId, track->streamType(), 1702 (audio_session_t)track->sessionId()); 1703 mLock.lock(); 1704 // abort track was stopped/paused while we released the lock 1705 if (state != track->mState) { 1706 if (status == NO_ERROR) { 1707 mLock.unlock(); 1708 AudioSystem::stopOutput(mId, track->streamType(), 1709 (audio_session_t)track->sessionId()); 1710 mLock.lock(); 1711 } 1712 return INVALID_OPERATION; 1713 } 1714 // abort if start is rejected by audio policy manager 1715 if (status != NO_ERROR) { 1716 return PERMISSION_DENIED; 1717 } 1718#ifdef ADD_BATTERY_DATA 1719 // to track the speaker usage 1720 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 1721#endif 1722 } 1723 1724 track->mFillingUpStatus = track->sharedBuffer() != 0 ? Track::FS_FILLED : Track::FS_FILLING; 1725 track->mResetDone = false; 1726 track->mPresentationCompleteFrames = 0; 1727 mActiveTracks.add(track); 1728 mWakeLockUids.add(track->uid()); 1729 mActiveTracksGeneration++; 1730 mLatestActiveTrack = track; 1731 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1732 if (chain != 0) { 1733 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), 1734 track->sessionId()); 1735 chain->incActiveTrackCnt(); 1736 } 1737 1738 status = NO_ERROR; 1739 } 1740 1741 onAddNewTrack_l(); 1742 return status; 1743} 1744 1745bool AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 1746{ 1747 track->terminate(); 1748 // active tracks are removed by threadLoop() 1749 bool trackActive = (mActiveTracks.indexOf(track) >= 0); 1750 track->mState = TrackBase::STOPPED; 1751 if (!trackActive) { 1752 removeTrack_l(track); 1753 } else if (track->isFastTrack() || track->isOffloaded() || track->isDirect()) { 1754 track->mState = TrackBase::STOPPING_1; 1755 } 1756 1757 return trackActive; 1758} 1759 1760void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 1761{ 1762 track->triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 1763 mTracks.remove(track); 1764 deleteTrackName_l(track->name()); 1765 // redundant as track is about to be destroyed, for dumpsys only 1766 track->mName = -1; 1767 if (track->isFastTrack()) { 1768 int index = track->mFastIndex; 1769 ALOG_ASSERT(0 < index && index < (int)FastMixerState::kMaxFastTracks); 1770 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << index))); 1771 mFastTrackAvailMask |= 1 << index; 1772 // redundant as track is about to be destroyed, for dumpsys only 1773 track->mFastIndex = -1; 1774 } 1775 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1776 if (chain != 0) { 1777 chain->decTrackCnt(); 1778 } 1779} 1780 1781void AudioFlinger::PlaybackThread::broadcast_l() 1782{ 1783 // Thread could be blocked waiting for async 1784 // so signal it to handle state changes immediately 1785 // If threadLoop is currently unlocked a signal of mWaitWorkCV will 1786 // be lost so we also flag to prevent it blocking on mWaitWorkCV 1787 mSignalPending = true; 1788 mWaitWorkCV.broadcast(); 1789} 1790 1791String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 1792{ 1793 Mutex::Autolock _l(mLock); 1794 if (initCheck() != NO_ERROR) { 1795 return String8(); 1796 } 1797 1798 char *s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 1799 const String8 out_s8(s); 1800 free(s); 1801 return out_s8; 1802} 1803 1804void AudioFlinger::PlaybackThread::audioConfigChanged(int event, int param) { 1805 AudioSystem::OutputDescriptor desc; 1806 void *param2 = NULL; 1807 1808 ALOGV("PlaybackThread::audioConfigChanged, thread %p, event %d, param %d", this, event, 1809 param); 1810 1811 switch (event) { 1812 case AudioSystem::OUTPUT_OPENED: 1813 case AudioSystem::OUTPUT_CONFIG_CHANGED: 1814 desc.channelMask = mChannelMask; 1815 desc.samplingRate = mSampleRate; 1816 desc.format = mFormat; 1817 desc.frameCount = mNormalFrameCount; // FIXME see 1818 // AudioFlinger::frameCount(audio_io_handle_t) 1819 desc.latency = latency_l(); 1820 param2 = &desc; 1821 break; 1822 1823 case AudioSystem::STREAM_CONFIG_CHANGED: 1824 param2 = ¶m; 1825 case AudioSystem::OUTPUT_CLOSED: 1826 default: 1827 break; 1828 } 1829 mAudioFlinger->audioConfigChanged(event, mId, param2); 1830} 1831 1832void AudioFlinger::PlaybackThread::writeCallback() 1833{ 1834 ALOG_ASSERT(mCallbackThread != 0); 1835 mCallbackThread->resetWriteBlocked(); 1836} 1837 1838void AudioFlinger::PlaybackThread::drainCallback() 1839{ 1840 ALOG_ASSERT(mCallbackThread != 0); 1841 mCallbackThread->resetDraining(); 1842} 1843 1844void AudioFlinger::PlaybackThread::resetWriteBlocked(uint32_t sequence) 1845{ 1846 Mutex::Autolock _l(mLock); 1847 // reject out of sequence requests 1848 if ((mWriteAckSequence & 1) && (sequence == mWriteAckSequence)) { 1849 mWriteAckSequence &= ~1; 1850 mWaitWorkCV.signal(); 1851 } 1852} 1853 1854void AudioFlinger::PlaybackThread::resetDraining(uint32_t sequence) 1855{ 1856 Mutex::Autolock _l(mLock); 1857 // reject out of sequence requests 1858 if ((mDrainSequence & 1) && (sequence == mDrainSequence)) { 1859 mDrainSequence &= ~1; 1860 mWaitWorkCV.signal(); 1861 } 1862} 1863 1864// static 1865int AudioFlinger::PlaybackThread::asyncCallback(stream_callback_event_t event, 1866 void *param __unused, 1867 void *cookie) 1868{ 1869 AudioFlinger::PlaybackThread *me = (AudioFlinger::PlaybackThread *)cookie; 1870 ALOGV("asyncCallback() event %d", event); 1871 switch (event) { 1872 case STREAM_CBK_EVENT_WRITE_READY: 1873 me->writeCallback(); 1874 break; 1875 case STREAM_CBK_EVENT_DRAIN_READY: 1876 me->drainCallback(); 1877 break; 1878 default: 1879 ALOGW("asyncCallback() unknown event %d", event); 1880 break; 1881 } 1882 return 0; 1883} 1884 1885void AudioFlinger::PlaybackThread::readOutputParameters_l() 1886{ 1887 // unfortunately we have no way of recovering from errors here, hence the LOG_ALWAYS_FATAL 1888 mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common); 1889 mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common); 1890 if (!audio_is_output_channel(mChannelMask)) { 1891 LOG_ALWAYS_FATAL("HAL channel mask %#x not valid for output", mChannelMask); 1892 } 1893 if ((mType == MIXER || mType == DUPLICATING) 1894 && !isValidPcmSinkChannelMask(mChannelMask)) { 1895 LOG_ALWAYS_FATAL("HAL channel mask %#x not supported for mixed output", 1896 mChannelMask); 1897 } 1898 mChannelCount = audio_channel_count_from_out_mask(mChannelMask); 1899 mHALFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 1900 mFormat = mHALFormat; 1901 if (!audio_is_valid_format(mFormat)) { 1902 LOG_ALWAYS_FATAL("HAL format %#x not valid for output", mFormat); 1903 } 1904 if ((mType == MIXER || mType == DUPLICATING) 1905 && !isValidPcmSinkFormat(mFormat)) { 1906 LOG_FATAL("HAL format %#x not supported for mixed output", 1907 mFormat); 1908 } 1909 mFrameSize = audio_stream_out_frame_size(mOutput->stream); 1910 mBufferSize = mOutput->stream->common.get_buffer_size(&mOutput->stream->common); 1911 mFrameCount = mBufferSize / mFrameSize; 1912 if (mFrameCount & 15) { 1913 ALOGW("HAL output buffer size is %u frames but AudioMixer requires multiples of 16 frames", 1914 mFrameCount); 1915 } 1916 1917 if ((mOutput->flags & AUDIO_OUTPUT_FLAG_NON_BLOCKING) && 1918 (mOutput->stream->set_callback != NULL)) { 1919 if (mOutput->stream->set_callback(mOutput->stream, 1920 AudioFlinger::PlaybackThread::asyncCallback, this) == 0) { 1921 mUseAsyncWrite = true; 1922 mCallbackThread = new AudioFlinger::AsyncCallbackThread(this); 1923 } 1924 } 1925 1926 mHwSupportsPause = false; 1927 if (mOutput->flags & AUDIO_OUTPUT_FLAG_DIRECT) { 1928 if (mOutput->stream->pause != NULL) { 1929 if (mOutput->stream->resume != NULL) { 1930 mHwSupportsPause = true; 1931 } else { 1932 ALOGW("direct output implements pause but not resume"); 1933 } 1934 } else if (mOutput->stream->resume != NULL) { 1935 ALOGW("direct output implements resume but not pause"); 1936 } 1937 } 1938 1939 if (mType == DUPLICATING && mMixerBufferEnabled && mEffectBufferEnabled) { 1940 // For best precision, we use float instead of the associated output 1941 // device format (typically PCM 16 bit). 1942 1943 mFormat = AUDIO_FORMAT_PCM_FLOAT; 1944 mFrameSize = mChannelCount * audio_bytes_per_sample(mFormat); 1945 mBufferSize = mFrameSize * mFrameCount; 1946 1947 // TODO: We currently use the associated output device channel mask and sample rate. 1948 // (1) Perhaps use the ORed channel mask of all downstream MixerThreads 1949 // (if a valid mask) to avoid premature downmix. 1950 // (2) Perhaps use the maximum sample rate of all downstream MixerThreads 1951 // instead of the output device sample rate to avoid loss of high frequency information. 1952 // This may need to be updated as MixerThread/OutputTracks are added and not here. 1953 } 1954 1955 // Calculate size of normal sink buffer relative to the HAL output buffer size 1956 double multiplier = 1.0; 1957 if (mType == MIXER && (kUseFastMixer == FastMixer_Static || 1958 kUseFastMixer == FastMixer_Dynamic)) { 1959 size_t minNormalFrameCount = (kMinNormalSinkBufferSizeMs * mSampleRate) / 1000; 1960 size_t maxNormalFrameCount = (kMaxNormalSinkBufferSizeMs * mSampleRate) / 1000; 1961 // round up minimum and round down maximum to nearest 16 frames to satisfy AudioMixer 1962 minNormalFrameCount = (minNormalFrameCount + 15) & ~15; 1963 maxNormalFrameCount = maxNormalFrameCount & ~15; 1964 if (maxNormalFrameCount < minNormalFrameCount) { 1965 maxNormalFrameCount = minNormalFrameCount; 1966 } 1967 multiplier = (double) minNormalFrameCount / (double) mFrameCount; 1968 if (multiplier <= 1.0) { 1969 multiplier = 1.0; 1970 } else if (multiplier <= 2.0) { 1971 if (2 * mFrameCount <= maxNormalFrameCount) { 1972 multiplier = 2.0; 1973 } else { 1974 multiplier = (double) maxNormalFrameCount / (double) mFrameCount; 1975 } 1976 } else { 1977 // prefer an even multiplier, for compatibility with doubling of fast tracks due to HAL 1978 // SRC (it would be unusual for the normal sink buffer size to not be a multiple of fast 1979 // track, but we sometimes have to do this to satisfy the maximum frame count 1980 // constraint) 1981 // FIXME this rounding up should not be done if no HAL SRC 1982 uint32_t truncMult = (uint32_t) multiplier; 1983 if ((truncMult & 1)) { 1984 if ((truncMult + 1) * mFrameCount <= maxNormalFrameCount) { 1985 ++truncMult; 1986 } 1987 } 1988 multiplier = (double) truncMult; 1989 } 1990 } 1991 mNormalFrameCount = multiplier * mFrameCount; 1992 // round up to nearest 16 frames to satisfy AudioMixer 1993 if (mType == MIXER || mType == DUPLICATING) { 1994 mNormalFrameCount = (mNormalFrameCount + 15) & ~15; 1995 } 1996 ALOGI("HAL output buffer size %u frames, normal sink buffer size %u frames", mFrameCount, 1997 mNormalFrameCount); 1998 1999 // mSinkBuffer is the sink buffer. Size is always multiple-of-16 frames. 2000 // Originally this was int16_t[] array, need to remove legacy implications. 2001 free(mSinkBuffer); 2002 mSinkBuffer = NULL; 2003 // For sink buffer size, we use the frame size from the downstream sink to avoid problems 2004 // with non PCM formats for compressed music, e.g. AAC, and Offload threads. 2005 const size_t sinkBufferSize = mNormalFrameCount * mFrameSize; 2006 (void)posix_memalign(&mSinkBuffer, 32, sinkBufferSize); 2007 2008 // We resize the mMixerBuffer according to the requirements of the sink buffer which 2009 // drives the output. 2010 free(mMixerBuffer); 2011 mMixerBuffer = NULL; 2012 if (mMixerBufferEnabled) { 2013 mMixerBufferFormat = AUDIO_FORMAT_PCM_FLOAT; // also valid: AUDIO_FORMAT_PCM_16_BIT. 2014 mMixerBufferSize = mNormalFrameCount * mChannelCount 2015 * audio_bytes_per_sample(mMixerBufferFormat); 2016 (void)posix_memalign(&mMixerBuffer, 32, mMixerBufferSize); 2017 } 2018 free(mEffectBuffer); 2019 mEffectBuffer = NULL; 2020 if (mEffectBufferEnabled) { 2021 mEffectBufferFormat = AUDIO_FORMAT_PCM_16_BIT; // Note: Effects support 16b only 2022 mEffectBufferSize = mNormalFrameCount * mChannelCount 2023 * audio_bytes_per_sample(mEffectBufferFormat); 2024 (void)posix_memalign(&mEffectBuffer, 32, mEffectBufferSize); 2025 } 2026 2027 // force reconfiguration of effect chains and engines to take new buffer size and audio 2028 // parameters into account 2029 // Note that mLock is not held when readOutputParameters_l() is called from the constructor 2030 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 2031 // matter. 2032 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 2033 Vector< sp<EffectChain> > effectChains = mEffectChains; 2034 for (size_t i = 0; i < effectChains.size(); i ++) { 2035 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 2036 } 2037} 2038 2039 2040status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames) 2041{ 2042 if (halFrames == NULL || dspFrames == NULL) { 2043 return BAD_VALUE; 2044 } 2045 Mutex::Autolock _l(mLock); 2046 if (initCheck() != NO_ERROR) { 2047 return INVALID_OPERATION; 2048 } 2049 size_t framesWritten = mBytesWritten / mFrameSize; 2050 *halFrames = framesWritten; 2051 2052 if (isSuspended()) { 2053 // return an estimation of rendered frames when the output is suspended 2054 size_t latencyFrames = (latency_l() * mSampleRate) / 1000; 2055 *dspFrames = framesWritten >= latencyFrames ? framesWritten - latencyFrames : 0; 2056 return NO_ERROR; 2057 } else { 2058 status_t status; 2059 uint32_t frames; 2060 status = mOutput->stream->get_render_position(mOutput->stream, &frames); 2061 *dspFrames = (size_t)frames; 2062 return status; 2063 } 2064} 2065 2066uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) const 2067{ 2068 Mutex::Autolock _l(mLock); 2069 uint32_t result = 0; 2070 if (getEffectChain_l(sessionId) != 0) { 2071 result = EFFECT_SESSION; 2072 } 2073 2074 for (size_t i = 0; i < mTracks.size(); ++i) { 2075 sp<Track> track = mTracks[i]; 2076 if (sessionId == track->sessionId() && !track->isInvalid()) { 2077 result |= TRACK_SESSION; 2078 break; 2079 } 2080 } 2081 2082 return result; 2083} 2084 2085uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 2086{ 2087 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 2088 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 2089 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 2090 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 2091 } 2092 for (size_t i = 0; i < mTracks.size(); i++) { 2093 sp<Track> track = mTracks[i]; 2094 if (sessionId == track->sessionId() && !track->isInvalid()) { 2095 return AudioSystem::getStrategyForStream(track->streamType()); 2096 } 2097 } 2098 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 2099} 2100 2101 2102AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 2103{ 2104 Mutex::Autolock _l(mLock); 2105 return mOutput; 2106} 2107 2108AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 2109{ 2110 Mutex::Autolock _l(mLock); 2111 AudioStreamOut *output = mOutput; 2112 mOutput = NULL; 2113 // FIXME FastMixer might also have a raw ptr to mOutputSink; 2114 // must push a NULL and wait for ack 2115 mOutputSink.clear(); 2116 mPipeSink.clear(); 2117 mNormalSink.clear(); 2118 return output; 2119} 2120 2121// this method must always be called either with ThreadBase mLock held or inside the thread loop 2122audio_stream_t* AudioFlinger::PlaybackThread::stream() const 2123{ 2124 if (mOutput == NULL) { 2125 return NULL; 2126 } 2127 return &mOutput->stream->common; 2128} 2129 2130uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() const 2131{ 2132 return (uint32_t)((uint32_t)((mNormalFrameCount * 1000) / mSampleRate) * 1000); 2133} 2134 2135status_t AudioFlinger::PlaybackThread::setSyncEvent(const sp<SyncEvent>& event) 2136{ 2137 if (!isValidSyncEvent(event)) { 2138 return BAD_VALUE; 2139 } 2140 2141 Mutex::Autolock _l(mLock); 2142 2143 for (size_t i = 0; i < mTracks.size(); ++i) { 2144 sp<Track> track = mTracks[i]; 2145 if (event->triggerSession() == track->sessionId()) { 2146 (void) track->setSyncEvent(event); 2147 return NO_ERROR; 2148 } 2149 } 2150 2151 return NAME_NOT_FOUND; 2152} 2153 2154bool AudioFlinger::PlaybackThread::isValidSyncEvent(const sp<SyncEvent>& event) const 2155{ 2156 return event->type() == AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE; 2157} 2158 2159void AudioFlinger::PlaybackThread::threadLoop_removeTracks( 2160 const Vector< sp<Track> >& tracksToRemove) 2161{ 2162 size_t count = tracksToRemove.size(); 2163 if (count > 0) { 2164 for (size_t i = 0 ; i < count ; i++) { 2165 const sp<Track>& track = tracksToRemove.itemAt(i); 2166 if (track->isExternalTrack()) { 2167 AudioSystem::stopOutput(mId, track->streamType(), 2168 (audio_session_t)track->sessionId()); 2169#ifdef ADD_BATTERY_DATA 2170 // to track the speaker usage 2171 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 2172#endif 2173 if (track->isTerminated()) { 2174 AudioSystem::releaseOutput(mId, track->streamType(), 2175 (audio_session_t)track->sessionId()); 2176 } 2177 } 2178 } 2179 } 2180} 2181 2182void AudioFlinger::PlaybackThread::checkSilentMode_l() 2183{ 2184 if (!mMasterMute) { 2185 char value[PROPERTY_VALUE_MAX]; 2186 if (property_get("ro.audio.silent", value, "0") > 0) { 2187 char *endptr; 2188 unsigned long ul = strtoul(value, &endptr, 0); 2189 if (*endptr == '\0' && ul != 0) { 2190 ALOGD("Silence is golden"); 2191 // The setprop command will not allow a property to be changed after 2192 // the first time it is set, so we don't have to worry about un-muting. 2193 setMasterMute_l(true); 2194 } 2195 } 2196 } 2197} 2198 2199// shared by MIXER and DIRECT, overridden by DUPLICATING 2200ssize_t AudioFlinger::PlaybackThread::threadLoop_write() 2201{ 2202 // FIXME rewrite to reduce number of system calls 2203 mLastWriteTime = systemTime(); 2204 mInWrite = true; 2205 ssize_t bytesWritten; 2206 const size_t offset = mCurrentWriteLength - mBytesRemaining; 2207 2208 // If an NBAIO sink is present, use it to write the normal mixer's submix 2209 if (mNormalSink != 0) { 2210 2211 const size_t count = mBytesRemaining / mFrameSize; 2212 2213 ATRACE_BEGIN("write"); 2214 // update the setpoint when AudioFlinger::mScreenState changes 2215 uint32_t screenState = AudioFlinger::mScreenState; 2216 if (screenState != mScreenState) { 2217 mScreenState = screenState; 2218 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 2219 if (pipe != NULL) { 2220 pipe->setAvgFrames((mScreenState & 1) ? 2221 (pipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2222 } 2223 } 2224 ssize_t framesWritten = mNormalSink->write((char *)mSinkBuffer + offset, count); 2225 ATRACE_END(); 2226 if (framesWritten > 0) { 2227 bytesWritten = framesWritten * mFrameSize; 2228 } else { 2229 bytesWritten = framesWritten; 2230 } 2231 mLatchDValid = false; 2232 status_t status = mNormalSink->getTimestamp(mLatchD.mTimestamp); 2233 if (status == NO_ERROR) { 2234 size_t totalFramesWritten = mNormalSink->framesWritten(); 2235 if (totalFramesWritten >= mLatchD.mTimestamp.mPosition) { 2236 mLatchD.mUnpresentedFrames = totalFramesWritten - mLatchD.mTimestamp.mPosition; 2237 // mLatchD.mFramesReleased is set immediately before D is clocked into Q 2238 mLatchDValid = true; 2239 } 2240 } 2241 // otherwise use the HAL / AudioStreamOut directly 2242 } else { 2243 // Direct output and offload threads 2244 2245 if (mUseAsyncWrite) { 2246 ALOGW_IF(mWriteAckSequence & 1, "threadLoop_write(): out of sequence write request"); 2247 mWriteAckSequence += 2; 2248 mWriteAckSequence |= 1; 2249 ALOG_ASSERT(mCallbackThread != 0); 2250 mCallbackThread->setWriteBlocked(mWriteAckSequence); 2251 } 2252 // FIXME We should have an implementation of timestamps for direct output threads. 2253 // They are used e.g for multichannel PCM playback over HDMI. 2254 bytesWritten = mOutput->stream->write(mOutput->stream, 2255 (char *)mSinkBuffer + offset, mBytesRemaining); 2256 if (mUseAsyncWrite && 2257 ((bytesWritten < 0) || (bytesWritten == (ssize_t)mBytesRemaining))) { 2258 // do not wait for async callback in case of error of full write 2259 mWriteAckSequence &= ~1; 2260 ALOG_ASSERT(mCallbackThread != 0); 2261 mCallbackThread->setWriteBlocked(mWriteAckSequence); 2262 } 2263 } 2264 2265 mNumWrites++; 2266 mInWrite = false; 2267 mStandby = false; 2268 return bytesWritten; 2269} 2270 2271void AudioFlinger::PlaybackThread::threadLoop_drain() 2272{ 2273 if (mOutput->stream->drain) { 2274 ALOGV("draining %s", (mMixerStatus == MIXER_DRAIN_TRACK) ? "early" : "full"); 2275 if (mUseAsyncWrite) { 2276 ALOGW_IF(mDrainSequence & 1, "threadLoop_drain(): out of sequence drain request"); 2277 mDrainSequence |= 1; 2278 ALOG_ASSERT(mCallbackThread != 0); 2279 mCallbackThread->setDraining(mDrainSequence); 2280 } 2281 mOutput->stream->drain(mOutput->stream, 2282 (mMixerStatus == MIXER_DRAIN_TRACK) ? AUDIO_DRAIN_EARLY_NOTIFY 2283 : AUDIO_DRAIN_ALL); 2284 } 2285} 2286 2287void AudioFlinger::PlaybackThread::threadLoop_exit() 2288{ 2289 { 2290 Mutex::Autolock _l(mLock); 2291 for (size_t i = 0; i < mTracks.size(); i++) { 2292 sp<Track> track = mTracks[i]; 2293 track->invalidate(); 2294 } 2295 } 2296} 2297 2298/* 2299The derived values that are cached: 2300 - mSinkBufferSize from frame count * frame size 2301 - activeSleepTime from activeSleepTimeUs() 2302 - idleSleepTime from idleSleepTimeUs() 2303 - standbyDelay from mActiveSleepTimeUs (DIRECT only) 2304 - maxPeriod from frame count and sample rate (MIXER only) 2305 2306The parameters that affect these derived values are: 2307 - frame count 2308 - frame size 2309 - sample rate 2310 - device type: A2DP or not 2311 - device latency 2312 - format: PCM or not 2313 - active sleep time 2314 - idle sleep time 2315*/ 2316 2317void AudioFlinger::PlaybackThread::cacheParameters_l() 2318{ 2319 mSinkBufferSize = mNormalFrameCount * mFrameSize; 2320 activeSleepTime = activeSleepTimeUs(); 2321 idleSleepTime = idleSleepTimeUs(); 2322} 2323 2324void AudioFlinger::PlaybackThread::invalidateTracks(audio_stream_type_t streamType) 2325{ 2326 ALOGV("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 2327 this, streamType, mTracks.size()); 2328 Mutex::Autolock _l(mLock); 2329 2330 size_t size = mTracks.size(); 2331 for (size_t i = 0; i < size; i++) { 2332 sp<Track> t = mTracks[i]; 2333 if (t->streamType() == streamType) { 2334 t->invalidate(); 2335 } 2336 } 2337} 2338 2339status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 2340{ 2341 int session = chain->sessionId(); 2342 int16_t* buffer = reinterpret_cast<int16_t*>(mEffectBufferEnabled 2343 ? mEffectBuffer : mSinkBuffer); 2344 bool ownsBuffer = false; 2345 2346 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 2347 if (session > 0) { 2348 // Only one effect chain can be present in direct output thread and it uses 2349 // the sink buffer as input 2350 if (mType != DIRECT) { 2351 size_t numSamples = mNormalFrameCount * mChannelCount; 2352 buffer = new int16_t[numSamples]; 2353 memset(buffer, 0, numSamples * sizeof(int16_t)); 2354 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 2355 ownsBuffer = true; 2356 } 2357 2358 // Attach all tracks with same session ID to this chain. 2359 for (size_t i = 0; i < mTracks.size(); ++i) { 2360 sp<Track> track = mTracks[i]; 2361 if (session == track->sessionId()) { 2362 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), 2363 buffer); 2364 track->setMainBuffer(buffer); 2365 chain->incTrackCnt(); 2366 } 2367 } 2368 2369 // indicate all active tracks in the chain 2370 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 2371 sp<Track> track = mActiveTracks[i].promote(); 2372 if (track == 0) { 2373 continue; 2374 } 2375 if (session == track->sessionId()) { 2376 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 2377 chain->incActiveTrackCnt(); 2378 } 2379 } 2380 } 2381 chain->setThread(this); 2382 chain->setInBuffer(buffer, ownsBuffer); 2383 chain->setOutBuffer(reinterpret_cast<int16_t*>(mEffectBufferEnabled 2384 ? mEffectBuffer : mSinkBuffer)); 2385 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 2386 // chains list in order to be processed last as it contains output stage effects 2387 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 2388 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 2389 // after track specific effects and before output stage 2390 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 2391 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 2392 // Effect chain for other sessions are inserted at beginning of effect 2393 // chains list to be processed before output mix effects. Relative order between other 2394 // sessions is not important 2395 size_t size = mEffectChains.size(); 2396 size_t i = 0; 2397 for (i = 0; i < size; i++) { 2398 if (mEffectChains[i]->sessionId() < session) { 2399 break; 2400 } 2401 } 2402 mEffectChains.insertAt(chain, i); 2403 checkSuspendOnAddEffectChain_l(chain); 2404 2405 return NO_ERROR; 2406} 2407 2408size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 2409{ 2410 int session = chain->sessionId(); 2411 2412 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 2413 2414 for (size_t i = 0; i < mEffectChains.size(); i++) { 2415 if (chain == mEffectChains[i]) { 2416 mEffectChains.removeAt(i); 2417 // detach all active tracks from the chain 2418 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 2419 sp<Track> track = mActiveTracks[i].promote(); 2420 if (track == 0) { 2421 continue; 2422 } 2423 if (session == track->sessionId()) { 2424 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 2425 chain.get(), session); 2426 chain->decActiveTrackCnt(); 2427 } 2428 } 2429 2430 // detach all tracks with same session ID from this chain 2431 for (size_t i = 0; i < mTracks.size(); ++i) { 2432 sp<Track> track = mTracks[i]; 2433 if (session == track->sessionId()) { 2434 track->setMainBuffer(reinterpret_cast<int16_t*>(mSinkBuffer)); 2435 chain->decTrackCnt(); 2436 } 2437 } 2438 break; 2439 } 2440 } 2441 return mEffectChains.size(); 2442} 2443 2444status_t AudioFlinger::PlaybackThread::attachAuxEffect( 2445 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2446{ 2447 Mutex::Autolock _l(mLock); 2448 return attachAuxEffect_l(track, EffectId); 2449} 2450 2451status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 2452 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2453{ 2454 status_t status = NO_ERROR; 2455 2456 if (EffectId == 0) { 2457 track->setAuxBuffer(0, NULL); 2458 } else { 2459 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 2460 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 2461 if (effect != 0) { 2462 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 2463 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 2464 } else { 2465 status = INVALID_OPERATION; 2466 } 2467 } else { 2468 status = BAD_VALUE; 2469 } 2470 } 2471 return status; 2472} 2473 2474void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 2475{ 2476 for (size_t i = 0; i < mTracks.size(); ++i) { 2477 sp<Track> track = mTracks[i]; 2478 if (track->auxEffectId() == effectId) { 2479 attachAuxEffect_l(track, 0); 2480 } 2481 } 2482} 2483 2484bool AudioFlinger::PlaybackThread::threadLoop() 2485{ 2486 Vector< sp<Track> > tracksToRemove; 2487 2488 standbyTime = systemTime(); 2489 2490 // MIXER 2491 nsecs_t lastWarning = 0; 2492 2493 // DUPLICATING 2494 // FIXME could this be made local to while loop? 2495 writeFrames = 0; 2496 2497 int lastGeneration = 0; 2498 2499 cacheParameters_l(); 2500 sleepTime = idleSleepTime; 2501 2502 if (mType == MIXER) { 2503 sleepTimeShift = 0; 2504 } 2505 2506 CpuStats cpuStats; 2507 const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid())); 2508 2509 acquireWakeLock(); 2510 2511 // mNBLogWriter->log can only be called while thread mutex mLock is held. 2512 // So if you need to log when mutex is unlocked, set logString to a non-NULL string, 2513 // and then that string will be logged at the next convenient opportunity. 2514 const char *logString = NULL; 2515 2516 checkSilentMode_l(); 2517 2518 while (!exitPending()) 2519 { 2520 cpuStats.sample(myName); 2521 2522 Vector< sp<EffectChain> > effectChains; 2523 2524 { // scope for mLock 2525 2526 Mutex::Autolock _l(mLock); 2527 2528 processConfigEvents_l(); 2529 2530 if (logString != NULL) { 2531 mNBLogWriter->logTimestamp(); 2532 mNBLogWriter->log(logString); 2533 logString = NULL; 2534 } 2535 2536 // Gather the framesReleased counters for all active tracks, 2537 // and latch them atomically with the timestamp. 2538 // FIXME We're using raw pointers as indices. A unique track ID would be a better index. 2539 mLatchD.mFramesReleased.clear(); 2540 size_t size = mActiveTracks.size(); 2541 for (size_t i = 0; i < size; i++) { 2542 sp<Track> t = mActiveTracks[i].promote(); 2543 if (t != 0) { 2544 mLatchD.mFramesReleased.add(t.get(), 2545 t->mAudioTrackServerProxy->framesReleased()); 2546 } 2547 } 2548 if (mLatchDValid) { 2549 mLatchQ = mLatchD; 2550 mLatchDValid = false; 2551 mLatchQValid = true; 2552 } 2553 2554 saveOutputTracks(); 2555 if (mSignalPending) { 2556 // A signal was raised while we were unlocked 2557 mSignalPending = false; 2558 } else if (waitingAsyncCallback_l()) { 2559 if (exitPending()) { 2560 break; 2561 } 2562 releaseWakeLock_l(); 2563 mWakeLockUids.clear(); 2564 mActiveTracksGeneration++; 2565 ALOGV("wait async completion"); 2566 mWaitWorkCV.wait(mLock); 2567 ALOGV("async completion/wake"); 2568 acquireWakeLock_l(); 2569 standbyTime = systemTime() + standbyDelay; 2570 sleepTime = 0; 2571 2572 continue; 2573 } 2574 if ((!mActiveTracks.size() && systemTime() > standbyTime) || 2575 isSuspended()) { 2576 // put audio hardware into standby after short delay 2577 if (shouldStandby_l()) { 2578 2579 threadLoop_standby(); 2580 2581 mStandby = true; 2582 } 2583 2584 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2585 // we're about to wait, flush the binder command buffer 2586 IPCThreadState::self()->flushCommands(); 2587 2588 clearOutputTracks(); 2589 2590 if (exitPending()) { 2591 break; 2592 } 2593 2594 releaseWakeLock_l(); 2595 mWakeLockUids.clear(); 2596 mActiveTracksGeneration++; 2597 // wait until we have something to do... 2598 ALOGV("%s going to sleep", myName.string()); 2599 mWaitWorkCV.wait(mLock); 2600 ALOGV("%s waking up", myName.string()); 2601 acquireWakeLock_l(); 2602 2603 mMixerStatus = MIXER_IDLE; 2604 mMixerStatusIgnoringFastTracks = MIXER_IDLE; 2605 mBytesWritten = 0; 2606 mBytesRemaining = 0; 2607 checkSilentMode_l(); 2608 2609 standbyTime = systemTime() + standbyDelay; 2610 sleepTime = idleSleepTime; 2611 if (mType == MIXER) { 2612 sleepTimeShift = 0; 2613 } 2614 2615 continue; 2616 } 2617 } 2618 // mMixerStatusIgnoringFastTracks is also updated internally 2619 mMixerStatus = prepareTracks_l(&tracksToRemove); 2620 2621 // compare with previously applied list 2622 if (lastGeneration != mActiveTracksGeneration) { 2623 // update wakelock 2624 updateWakeLockUids_l(mWakeLockUids); 2625 lastGeneration = mActiveTracksGeneration; 2626 } 2627 2628 // prevent any changes in effect chain list and in each effect chain 2629 // during mixing and effect process as the audio buffers could be deleted 2630 // or modified if an effect is created or deleted 2631 lockEffectChains_l(effectChains); 2632 } // mLock scope ends 2633 2634 if (mBytesRemaining == 0) { 2635 mCurrentWriteLength = 0; 2636 if (mMixerStatus == MIXER_TRACKS_READY) { 2637 // threadLoop_mix() sets mCurrentWriteLength 2638 threadLoop_mix(); 2639 } else if ((mMixerStatus != MIXER_DRAIN_TRACK) 2640 && (mMixerStatus != MIXER_DRAIN_ALL)) { 2641 // threadLoop_sleepTime sets sleepTime to 0 if data 2642 // must be written to HAL 2643 threadLoop_sleepTime(); 2644 if (sleepTime == 0) { 2645 mCurrentWriteLength = mSinkBufferSize; 2646 } 2647 } 2648 // Either threadLoop_mix() or threadLoop_sleepTime() should have set 2649 // mMixerBuffer with data if mMixerBufferValid is true and sleepTime == 0. 2650 // Merge mMixerBuffer data into mEffectBuffer (if any effects are valid) 2651 // or mSinkBuffer (if there are no effects). 2652 // 2653 // This is done pre-effects computation; if effects change to 2654 // support higher precision, this needs to move. 2655 // 2656 // mMixerBufferValid is only set true by MixerThread::prepareTracks_l(). 2657 // TODO use sleepTime == 0 as an additional condition. 2658 if (mMixerBufferValid) { 2659 void *buffer = mEffectBufferValid ? mEffectBuffer : mSinkBuffer; 2660 audio_format_t format = mEffectBufferValid ? mEffectBufferFormat : mFormat; 2661 2662 memcpy_by_audio_format(buffer, format, mMixerBuffer, mMixerBufferFormat, 2663 mNormalFrameCount * mChannelCount); 2664 } 2665 2666 mBytesRemaining = mCurrentWriteLength; 2667 if (isSuspended()) { 2668 sleepTime = suspendSleepTimeUs(); 2669 // simulate write to HAL when suspended 2670 mBytesWritten += mSinkBufferSize; 2671 mBytesRemaining = 0; 2672 } 2673 2674 // only process effects if we're going to write 2675 if (sleepTime == 0 && mType != OFFLOAD) { 2676 for (size_t i = 0; i < effectChains.size(); i ++) { 2677 effectChains[i]->process_l(); 2678 } 2679 } 2680 } 2681 // Process effect chains for offloaded thread even if no audio 2682 // was read from audio track: process only updates effect state 2683 // and thus does have to be synchronized with audio writes but may have 2684 // to be called while waiting for async write callback 2685 if (mType == OFFLOAD) { 2686 for (size_t i = 0; i < effectChains.size(); i ++) { 2687 effectChains[i]->process_l(); 2688 } 2689 } 2690 2691 // Only if the Effects buffer is enabled and there is data in the 2692 // Effects buffer (buffer valid), we need to 2693 // copy into the sink buffer. 2694 // TODO use sleepTime == 0 as an additional condition. 2695 if (mEffectBufferValid) { 2696 //ALOGV("writing effect buffer to sink buffer format %#x", mFormat); 2697 memcpy_by_audio_format(mSinkBuffer, mFormat, mEffectBuffer, mEffectBufferFormat, 2698 mNormalFrameCount * mChannelCount); 2699 } 2700 2701 // enable changes in effect chain 2702 unlockEffectChains(effectChains); 2703 2704 if (!waitingAsyncCallback()) { 2705 // sleepTime == 0 means we must write to audio hardware 2706 if (sleepTime == 0) { 2707 if (mBytesRemaining) { 2708 ssize_t ret = threadLoop_write(); 2709 if (ret < 0) { 2710 mBytesRemaining = 0; 2711 } else { 2712 mBytesWritten += ret; 2713 mBytesRemaining -= ret; 2714 } 2715 } else if ((mMixerStatus == MIXER_DRAIN_TRACK) || 2716 (mMixerStatus == MIXER_DRAIN_ALL)) { 2717 threadLoop_drain(); 2718 } 2719 if (mType == MIXER) { 2720 // write blocked detection 2721 nsecs_t now = systemTime(); 2722 nsecs_t delta = now - mLastWriteTime; 2723 if (!mStandby && delta > maxPeriod) { 2724 mNumDelayedWrites++; 2725 if ((now - lastWarning) > kWarningThrottleNs) { 2726 ATRACE_NAME("underrun"); 2727 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2728 ns2ms(delta), mNumDelayedWrites, this); 2729 lastWarning = now; 2730 } 2731 } 2732 } 2733 2734 } else { 2735 ATRACE_BEGIN("sleep"); 2736 usleep(sleepTime); 2737 ATRACE_END(); 2738 } 2739 } 2740 2741 // Finally let go of removed track(s), without the lock held 2742 // since we can't guarantee the destructors won't acquire that 2743 // same lock. This will also mutate and push a new fast mixer state. 2744 threadLoop_removeTracks(tracksToRemove); 2745 tracksToRemove.clear(); 2746 2747 // FIXME I don't understand the need for this here; 2748 // it was in the original code but maybe the 2749 // assignment in saveOutputTracks() makes this unnecessary? 2750 clearOutputTracks(); 2751 2752 // Effect chains will be actually deleted here if they were removed from 2753 // mEffectChains list during mixing or effects processing 2754 effectChains.clear(); 2755 2756 // FIXME Note that the above .clear() is no longer necessary since effectChains 2757 // is now local to this block, but will keep it for now (at least until merge done). 2758 } 2759 2760 threadLoop_exit(); 2761 2762 if (!mStandby) { 2763 threadLoop_standby(); 2764 mStandby = true; 2765 } 2766 2767 releaseWakeLock(); 2768 mWakeLockUids.clear(); 2769 mActiveTracksGeneration++; 2770 2771 ALOGV("Thread %p type %d exiting", this, mType); 2772 return false; 2773} 2774 2775// removeTracks_l() must be called with ThreadBase::mLock held 2776void AudioFlinger::PlaybackThread::removeTracks_l(const Vector< sp<Track> >& tracksToRemove) 2777{ 2778 size_t count = tracksToRemove.size(); 2779 if (count > 0) { 2780 for (size_t i=0 ; i<count ; i++) { 2781 const sp<Track>& track = tracksToRemove.itemAt(i); 2782 mActiveTracks.remove(track); 2783 mWakeLockUids.remove(track->uid()); 2784 mActiveTracksGeneration++; 2785 ALOGV("removeTracks_l removing track on session %d", track->sessionId()); 2786 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 2787 if (chain != 0) { 2788 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), 2789 track->sessionId()); 2790 chain->decActiveTrackCnt(); 2791 } 2792 if (track->isTerminated()) { 2793 removeTrack_l(track); 2794 } 2795 } 2796 } 2797 2798} 2799 2800status_t AudioFlinger::PlaybackThread::getTimestamp_l(AudioTimestamp& timestamp) 2801{ 2802 if (mNormalSink != 0) { 2803 return mNormalSink->getTimestamp(timestamp); 2804 } 2805 if ((mType == OFFLOAD || mType == DIRECT) 2806 && mOutput != NULL && mOutput->stream->get_presentation_position) { 2807 uint64_t position64; 2808 int ret = mOutput->stream->get_presentation_position( 2809 mOutput->stream, &position64, ×tamp.mTime); 2810 if (ret == 0) { 2811 timestamp.mPosition = (uint32_t)position64; 2812 return NO_ERROR; 2813 } 2814 } 2815 return INVALID_OPERATION; 2816} 2817 2818status_t AudioFlinger::PlaybackThread::createAudioPatch_l(const struct audio_patch *patch, 2819 audio_patch_handle_t *handle) 2820{ 2821 status_t status = NO_ERROR; 2822 if (mOutput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 2823 // store new device and send to effects 2824 audio_devices_t type = AUDIO_DEVICE_NONE; 2825 for (unsigned int i = 0; i < patch->num_sinks; i++) { 2826 type |= patch->sinks[i].ext.device.type; 2827 } 2828 mOutDevice = type; 2829 for (size_t i = 0; i < mEffectChains.size(); i++) { 2830 mEffectChains[i]->setDevice_l(mOutDevice); 2831 } 2832 2833 audio_hw_device_t *hwDevice = mOutput->audioHwDev->hwDevice(); 2834 status = hwDevice->create_audio_patch(hwDevice, 2835 patch->num_sources, 2836 patch->sources, 2837 patch->num_sinks, 2838 patch->sinks, 2839 handle); 2840 } else { 2841 ALOG_ASSERT(false, "createAudioPatch_l() called on a pre 3.0 HAL"); 2842 } 2843 return status; 2844} 2845 2846status_t AudioFlinger::PlaybackThread::releaseAudioPatch_l(const audio_patch_handle_t handle) 2847{ 2848 status_t status = NO_ERROR; 2849 if (mOutput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 2850 audio_hw_device_t *hwDevice = mOutput->audioHwDev->hwDevice(); 2851 status = hwDevice->release_audio_patch(hwDevice, handle); 2852 } else { 2853 ALOG_ASSERT(false, "releaseAudioPatch_l() called on a pre 3.0 HAL"); 2854 } 2855 return status; 2856} 2857 2858void AudioFlinger::PlaybackThread::addPatchTrack(const sp<PatchTrack>& track) 2859{ 2860 Mutex::Autolock _l(mLock); 2861 mTracks.add(track); 2862} 2863 2864void AudioFlinger::PlaybackThread::deletePatchTrack(const sp<PatchTrack>& track) 2865{ 2866 Mutex::Autolock _l(mLock); 2867 destroyTrack_l(track); 2868} 2869 2870void AudioFlinger::PlaybackThread::getAudioPortConfig(struct audio_port_config *config) 2871{ 2872 ThreadBase::getAudioPortConfig(config); 2873 config->role = AUDIO_PORT_ROLE_SOURCE; 2874 config->ext.mix.hw_module = mOutput->audioHwDev->handle(); 2875 config->ext.mix.usecase.stream = AUDIO_STREAM_DEFAULT; 2876} 2877 2878// ---------------------------------------------------------------------------- 2879 2880AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 2881 audio_io_handle_t id, audio_devices_t device, type_t type) 2882 : PlaybackThread(audioFlinger, output, id, device, type), 2883 // mAudioMixer below 2884 // mFastMixer below 2885 mFastMixerFutex(0) 2886 // mOutputSink below 2887 // mPipeSink below 2888 // mNormalSink below 2889{ 2890 ALOGV("MixerThread() id=%d device=%#x type=%d", id, device, type); 2891 ALOGV("mSampleRate=%u, mChannelMask=%#x, mChannelCount=%u, mFormat=%d, mFrameSize=%u, " 2892 "mFrameCount=%d, mNormalFrameCount=%d", 2893 mSampleRate, mChannelMask, mChannelCount, mFormat, mFrameSize, mFrameCount, 2894 mNormalFrameCount); 2895 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 2896 2897 if (type == DUPLICATING) { 2898 // The Duplicating thread uses the AudioMixer and delivers data to OutputTracks 2899 // (downstream MixerThreads) in DuplicatingThread::threadLoop_write(). 2900 // Do not create or use mFastMixer, mOutputSink, mPipeSink, or mNormalSink. 2901 return; 2902 } 2903 // create an NBAIO sink for the HAL output stream, and negotiate 2904 mOutputSink = new AudioStreamOutSink(output->stream); 2905 size_t numCounterOffers = 0; 2906 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount, mFormat)}; 2907 ssize_t index = mOutputSink->negotiate(offers, 1, NULL, numCounterOffers); 2908 ALOG_ASSERT(index == 0); 2909 2910 // initialize fast mixer depending on configuration 2911 bool initFastMixer; 2912 switch (kUseFastMixer) { 2913 case FastMixer_Never: 2914 initFastMixer = false; 2915 break; 2916 case FastMixer_Always: 2917 initFastMixer = true; 2918 break; 2919 case FastMixer_Static: 2920 case FastMixer_Dynamic: 2921 initFastMixer = mFrameCount < mNormalFrameCount; 2922 break; 2923 } 2924 if (initFastMixer) { 2925 audio_format_t fastMixerFormat; 2926 if (mMixerBufferEnabled && mEffectBufferEnabled) { 2927 fastMixerFormat = AUDIO_FORMAT_PCM_FLOAT; 2928 } else { 2929 fastMixerFormat = AUDIO_FORMAT_PCM_16_BIT; 2930 } 2931 if (mFormat != fastMixerFormat) { 2932 // change our Sink format to accept our intermediate precision 2933 mFormat = fastMixerFormat; 2934 free(mSinkBuffer); 2935 mFrameSize = mChannelCount * audio_bytes_per_sample(mFormat); 2936 const size_t sinkBufferSize = mNormalFrameCount * mFrameSize; 2937 (void)posix_memalign(&mSinkBuffer, 32, sinkBufferSize); 2938 } 2939 2940 // create a MonoPipe to connect our submix to FastMixer 2941 NBAIO_Format format = mOutputSink->format(); 2942 NBAIO_Format origformat = format; 2943 // adjust format to match that of the Fast Mixer 2944 ALOGV("format changed from %d to %d", format.mFormat, fastMixerFormat); 2945 format.mFormat = fastMixerFormat; 2946 format.mFrameSize = audio_bytes_per_sample(format.mFormat) * format.mChannelCount; 2947 2948 // This pipe depth compensates for scheduling latency of the normal mixer thread. 2949 // When it wakes up after a maximum latency, it runs a few cycles quickly before 2950 // finally blocking. Note the pipe implementation rounds up the request to a power of 2. 2951 MonoPipe *monoPipe = new MonoPipe(mNormalFrameCount * 4, format, true /*writeCanBlock*/); 2952 const NBAIO_Format offers[1] = {format}; 2953 size_t numCounterOffers = 0; 2954 ssize_t index = monoPipe->negotiate(offers, 1, NULL, numCounterOffers); 2955 ALOG_ASSERT(index == 0); 2956 monoPipe->setAvgFrames((mScreenState & 1) ? 2957 (monoPipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2958 mPipeSink = monoPipe; 2959 2960#ifdef TEE_SINK 2961 if (mTeeSinkOutputEnabled) { 2962 // create a Pipe to archive a copy of FastMixer's output for dumpsys 2963 Pipe *teeSink = new Pipe(mTeeSinkOutputFrames, origformat); 2964 const NBAIO_Format offers2[1] = {origformat}; 2965 numCounterOffers = 0; 2966 index = teeSink->negotiate(offers2, 1, NULL, numCounterOffers); 2967 ALOG_ASSERT(index == 0); 2968 mTeeSink = teeSink; 2969 PipeReader *teeSource = new PipeReader(*teeSink); 2970 numCounterOffers = 0; 2971 index = teeSource->negotiate(offers2, 1, NULL, numCounterOffers); 2972 ALOG_ASSERT(index == 0); 2973 mTeeSource = teeSource; 2974 } 2975#endif 2976 2977 // create fast mixer and configure it initially with just one fast track for our submix 2978 mFastMixer = new FastMixer(); 2979 FastMixerStateQueue *sq = mFastMixer->sq(); 2980#ifdef STATE_QUEUE_DUMP 2981 sq->setObserverDump(&mStateQueueObserverDump); 2982 sq->setMutatorDump(&mStateQueueMutatorDump); 2983#endif 2984 FastMixerState *state = sq->begin(); 2985 FastTrack *fastTrack = &state->mFastTracks[0]; 2986 // wrap the source side of the MonoPipe to make it an AudioBufferProvider 2987 fastTrack->mBufferProvider = new SourceAudioBufferProvider(new MonoPipeReader(monoPipe)); 2988 fastTrack->mVolumeProvider = NULL; 2989 fastTrack->mChannelMask = mChannelMask; // mPipeSink channel mask for audio to FastMixer 2990 fastTrack->mFormat = mFormat; // mPipeSink format for audio to FastMixer 2991 fastTrack->mGeneration++; 2992 state->mFastTracksGen++; 2993 state->mTrackMask = 1; 2994 // fast mixer will use the HAL output sink 2995 state->mOutputSink = mOutputSink.get(); 2996 state->mOutputSinkGen++; 2997 state->mFrameCount = mFrameCount; 2998 state->mCommand = FastMixerState::COLD_IDLE; 2999 // already done in constructor initialization list 3000 //mFastMixerFutex = 0; 3001 state->mColdFutexAddr = &mFastMixerFutex; 3002 state->mColdGen++; 3003 state->mDumpState = &mFastMixerDumpState; 3004#ifdef TEE_SINK 3005 state->mTeeSink = mTeeSink.get(); 3006#endif 3007 mFastMixerNBLogWriter = audioFlinger->newWriter_l(kFastMixerLogSize, "FastMixer"); 3008 state->mNBLogWriter = mFastMixerNBLogWriter.get(); 3009 sq->end(); 3010 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3011 3012 // start the fast mixer 3013 mFastMixer->run("FastMixer", PRIORITY_URGENT_AUDIO); 3014 pid_t tid = mFastMixer->getTid(); 3015 int err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 3016 if (err != 0) { 3017 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 3018 kPriorityFastMixer, getpid_cached, tid, err); 3019 } 3020 3021#ifdef AUDIO_WATCHDOG 3022 // create and start the watchdog 3023 mAudioWatchdog = new AudioWatchdog(); 3024 mAudioWatchdog->setDump(&mAudioWatchdogDump); 3025 mAudioWatchdog->run("AudioWatchdog", PRIORITY_URGENT_AUDIO); 3026 tid = mAudioWatchdog->getTid(); 3027 err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 3028 if (err != 0) { 3029 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 3030 kPriorityFastMixer, getpid_cached, tid, err); 3031 } 3032#endif 3033 3034 } 3035 3036 switch (kUseFastMixer) { 3037 case FastMixer_Never: 3038 case FastMixer_Dynamic: 3039 mNormalSink = mOutputSink; 3040 break; 3041 case FastMixer_Always: 3042 mNormalSink = mPipeSink; 3043 break; 3044 case FastMixer_Static: 3045 mNormalSink = initFastMixer ? mPipeSink : mOutputSink; 3046 break; 3047 } 3048} 3049 3050AudioFlinger::MixerThread::~MixerThread() 3051{ 3052 if (mFastMixer != 0) { 3053 FastMixerStateQueue *sq = mFastMixer->sq(); 3054 FastMixerState *state = sq->begin(); 3055 if (state->mCommand == FastMixerState::COLD_IDLE) { 3056 int32_t old = android_atomic_inc(&mFastMixerFutex); 3057 if (old == -1) { 3058 (void) syscall(__NR_futex, &mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 3059 } 3060 } 3061 state->mCommand = FastMixerState::EXIT; 3062 sq->end(); 3063 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3064 mFastMixer->join(); 3065 // Though the fast mixer thread has exited, it's state queue is still valid. 3066 // We'll use that extract the final state which contains one remaining fast track 3067 // corresponding to our sub-mix. 3068 state = sq->begin(); 3069 ALOG_ASSERT(state->mTrackMask == 1); 3070 FastTrack *fastTrack = &state->mFastTracks[0]; 3071 ALOG_ASSERT(fastTrack->mBufferProvider != NULL); 3072 delete fastTrack->mBufferProvider; 3073 sq->end(false /*didModify*/); 3074 mFastMixer.clear(); 3075#ifdef AUDIO_WATCHDOG 3076 if (mAudioWatchdog != 0) { 3077 mAudioWatchdog->requestExit(); 3078 mAudioWatchdog->requestExitAndWait(); 3079 mAudioWatchdog.clear(); 3080 } 3081#endif 3082 } 3083 mAudioFlinger->unregisterWriter(mFastMixerNBLogWriter); 3084 delete mAudioMixer; 3085} 3086 3087 3088uint32_t AudioFlinger::MixerThread::correctLatency_l(uint32_t latency) const 3089{ 3090 if (mFastMixer != 0) { 3091 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 3092 latency += (pipe->getAvgFrames() * 1000) / mSampleRate; 3093 } 3094 return latency; 3095} 3096 3097 3098void AudioFlinger::MixerThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) 3099{ 3100 PlaybackThread::threadLoop_removeTracks(tracksToRemove); 3101} 3102 3103ssize_t AudioFlinger::MixerThread::threadLoop_write() 3104{ 3105 // FIXME we should only do one push per cycle; confirm this is true 3106 // Start the fast mixer if it's not already running 3107 if (mFastMixer != 0) { 3108 FastMixerStateQueue *sq = mFastMixer->sq(); 3109 FastMixerState *state = sq->begin(); 3110 if (state->mCommand != FastMixerState::MIX_WRITE && 3111 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)) { 3112 if (state->mCommand == FastMixerState::COLD_IDLE) { 3113 int32_t old = android_atomic_inc(&mFastMixerFutex); 3114 if (old == -1) { 3115 (void) syscall(__NR_futex, &mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 3116 } 3117#ifdef AUDIO_WATCHDOG 3118 if (mAudioWatchdog != 0) { 3119 mAudioWatchdog->resume(); 3120 } 3121#endif 3122 } 3123 state->mCommand = FastMixerState::MIX_WRITE; 3124 mFastMixerDumpState.increaseSamplingN(mAudioFlinger->isLowRamDevice() ? 3125 FastMixerDumpState::kSamplingNforLowRamDevice : FastMixerDumpState::kSamplingN); 3126 sq->end(); 3127 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3128 if (kUseFastMixer == FastMixer_Dynamic) { 3129 mNormalSink = mPipeSink; 3130 } 3131 } else { 3132 sq->end(false /*didModify*/); 3133 } 3134 } 3135 return PlaybackThread::threadLoop_write(); 3136} 3137 3138void AudioFlinger::MixerThread::threadLoop_standby() 3139{ 3140 // Idle the fast mixer if it's currently running 3141 if (mFastMixer != 0) { 3142 FastMixerStateQueue *sq = mFastMixer->sq(); 3143 FastMixerState *state = sq->begin(); 3144 if (!(state->mCommand & FastMixerState::IDLE)) { 3145 state->mCommand = FastMixerState::COLD_IDLE; 3146 state->mColdFutexAddr = &mFastMixerFutex; 3147 state->mColdGen++; 3148 mFastMixerFutex = 0; 3149 sq->end(); 3150 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 3151 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3152 if (kUseFastMixer == FastMixer_Dynamic) { 3153 mNormalSink = mOutputSink; 3154 } 3155#ifdef AUDIO_WATCHDOG 3156 if (mAudioWatchdog != 0) { 3157 mAudioWatchdog->pause(); 3158 } 3159#endif 3160 } else { 3161 sq->end(false /*didModify*/); 3162 } 3163 } 3164 PlaybackThread::threadLoop_standby(); 3165} 3166 3167bool AudioFlinger::PlaybackThread::waitingAsyncCallback_l() 3168{ 3169 return false; 3170} 3171 3172bool AudioFlinger::PlaybackThread::shouldStandby_l() 3173{ 3174 return !mStandby; 3175} 3176 3177bool AudioFlinger::PlaybackThread::waitingAsyncCallback() 3178{ 3179 Mutex::Autolock _l(mLock); 3180 return waitingAsyncCallback_l(); 3181} 3182 3183// shared by MIXER and DIRECT, overridden by DUPLICATING 3184void AudioFlinger::PlaybackThread::threadLoop_standby() 3185{ 3186 ALOGV("Audio hardware entering standby, mixer %p, suspend count %d", this, mSuspended); 3187 mOutput->stream->common.standby(&mOutput->stream->common); 3188 if (mUseAsyncWrite != 0) { 3189 // discard any pending drain or write ack by incrementing sequence 3190 mWriteAckSequence = (mWriteAckSequence + 2) & ~1; 3191 mDrainSequence = (mDrainSequence + 2) & ~1; 3192 ALOG_ASSERT(mCallbackThread != 0); 3193 mCallbackThread->setWriteBlocked(mWriteAckSequence); 3194 mCallbackThread->setDraining(mDrainSequence); 3195 } 3196 mHwPaused = false; 3197} 3198 3199void AudioFlinger::PlaybackThread::onAddNewTrack_l() 3200{ 3201 ALOGV("signal playback thread"); 3202 broadcast_l(); 3203} 3204 3205void AudioFlinger::MixerThread::threadLoop_mix() 3206{ 3207 // obtain the presentation timestamp of the next output buffer 3208 int64_t pts; 3209 status_t status = INVALID_OPERATION; 3210 3211 if (mNormalSink != 0) { 3212 status = mNormalSink->getNextWriteTimestamp(&pts); 3213 } else { 3214 status = mOutputSink->getNextWriteTimestamp(&pts); 3215 } 3216 3217 if (status != NO_ERROR) { 3218 pts = AudioBufferProvider::kInvalidPTS; 3219 } 3220 3221 // mix buffers... 3222 mAudioMixer->process(pts); 3223 mCurrentWriteLength = mSinkBufferSize; 3224 // increase sleep time progressively when application underrun condition clears. 3225 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 3226 // that a steady state of alternating ready/not ready conditions keeps the sleep time 3227 // such that we would underrun the audio HAL. 3228 if ((sleepTime == 0) && (sleepTimeShift > 0)) { 3229 sleepTimeShift--; 3230 } 3231 sleepTime = 0; 3232 standbyTime = systemTime() + standbyDelay; 3233 //TODO: delay standby when effects have a tail 3234 3235} 3236 3237void AudioFlinger::MixerThread::threadLoop_sleepTime() 3238{ 3239 // If no tracks are ready, sleep once for the duration of an output 3240 // buffer size, then write 0s to the output 3241 if (sleepTime == 0) { 3242 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3243 sleepTime = activeSleepTime >> sleepTimeShift; 3244 if (sleepTime < kMinThreadSleepTimeUs) { 3245 sleepTime = kMinThreadSleepTimeUs; 3246 } 3247 // reduce sleep time in case of consecutive application underruns to avoid 3248 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 3249 // duration we would end up writing less data than needed by the audio HAL if 3250 // the condition persists. 3251 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 3252 sleepTimeShift++; 3253 } 3254 } else { 3255 sleepTime = idleSleepTime; 3256 } 3257 } else if (mBytesWritten != 0 || (mMixerStatus == MIXER_TRACKS_ENABLED)) { 3258 // clear out mMixerBuffer or mSinkBuffer, to ensure buffers are cleared 3259 // before effects processing or output. 3260 if (mMixerBufferValid) { 3261 memset(mMixerBuffer, 0, mMixerBufferSize); 3262 } else { 3263 memset(mSinkBuffer, 0, mSinkBufferSize); 3264 } 3265 sleepTime = 0; 3266 ALOGV_IF(mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED), 3267 "anticipated start"); 3268 } 3269 // TODO add standby time extension fct of effect tail 3270} 3271 3272// prepareTracks_l() must be called with ThreadBase::mLock held 3273AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 3274 Vector< sp<Track> > *tracksToRemove) 3275{ 3276 3277 mixer_state mixerStatus = MIXER_IDLE; 3278 // find out which tracks need to be processed 3279 size_t count = mActiveTracks.size(); 3280 size_t mixedTracks = 0; 3281 size_t tracksWithEffect = 0; 3282 // counts only _active_ fast tracks 3283 size_t fastTracks = 0; 3284 uint32_t resetMask = 0; // bit mask of fast tracks that need to be reset 3285 3286 float masterVolume = mMasterVolume; 3287 bool masterMute = mMasterMute; 3288 3289 if (masterMute) { 3290 masterVolume = 0; 3291 } 3292 // Delegate master volume control to effect in output mix effect chain if needed 3293 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 3294 if (chain != 0) { 3295 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 3296 chain->setVolume_l(&v, &v); 3297 masterVolume = (float)((v + (1 << 23)) >> 24); 3298 chain.clear(); 3299 } 3300 3301 // prepare a new state to push 3302 FastMixerStateQueue *sq = NULL; 3303 FastMixerState *state = NULL; 3304 bool didModify = false; 3305 FastMixerStateQueue::block_t block = FastMixerStateQueue::BLOCK_UNTIL_PUSHED; 3306 if (mFastMixer != 0) { 3307 sq = mFastMixer->sq(); 3308 state = sq->begin(); 3309 } 3310 3311 mMixerBufferValid = false; // mMixerBuffer has no valid data until appropriate tracks found. 3312 mEffectBufferValid = false; // mEffectBuffer has no valid data until tracks found. 3313 3314 for (size_t i=0 ; i<count ; i++) { 3315 const sp<Track> t = mActiveTracks[i].promote(); 3316 if (t == 0) { 3317 continue; 3318 } 3319 3320 // this const just means the local variable doesn't change 3321 Track* const track = t.get(); 3322 3323 // process fast tracks 3324 if (track->isFastTrack()) { 3325 3326 // It's theoretically possible (though unlikely) for a fast track to be created 3327 // and then removed within the same normal mix cycle. This is not a problem, as 3328 // the track never becomes active so it's fast mixer slot is never touched. 3329 // The converse, of removing an (active) track and then creating a new track 3330 // at the identical fast mixer slot within the same normal mix cycle, 3331 // is impossible because the slot isn't marked available until the end of each cycle. 3332 int j = track->mFastIndex; 3333 ALOG_ASSERT(0 < j && j < (int)FastMixerState::kMaxFastTracks); 3334 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << j))); 3335 FastTrack *fastTrack = &state->mFastTracks[j]; 3336 3337 // Determine whether the track is currently in underrun condition, 3338 // and whether it had a recent underrun. 3339 FastTrackDump *ftDump = &mFastMixerDumpState.mTracks[j]; 3340 FastTrackUnderruns underruns = ftDump->mUnderruns; 3341 uint32_t recentFull = (underruns.mBitFields.mFull - 3342 track->mObservedUnderruns.mBitFields.mFull) & UNDERRUN_MASK; 3343 uint32_t recentPartial = (underruns.mBitFields.mPartial - 3344 track->mObservedUnderruns.mBitFields.mPartial) & UNDERRUN_MASK; 3345 uint32_t recentEmpty = (underruns.mBitFields.mEmpty - 3346 track->mObservedUnderruns.mBitFields.mEmpty) & UNDERRUN_MASK; 3347 uint32_t recentUnderruns = recentPartial + recentEmpty; 3348 track->mObservedUnderruns = underruns; 3349 // don't count underruns that occur while stopping or pausing 3350 // or stopped which can occur when flush() is called while active 3351 if (!(track->isStopping() || track->isPausing() || track->isStopped()) && 3352 recentUnderruns > 0) { 3353 // FIXME fast mixer will pull & mix partial buffers, but we count as a full underrun 3354 track->mAudioTrackServerProxy->tallyUnderrunFrames(recentUnderruns * mFrameCount); 3355 } 3356 3357 // This is similar to the state machine for normal tracks, 3358 // with a few modifications for fast tracks. 3359 bool isActive = true; 3360 switch (track->mState) { 3361 case TrackBase::STOPPING_1: 3362 // track stays active in STOPPING_1 state until first underrun 3363 if (recentUnderruns > 0 || track->isTerminated()) { 3364 track->mState = TrackBase::STOPPING_2; 3365 } 3366 break; 3367 case TrackBase::PAUSING: 3368 // ramp down is not yet implemented 3369 track->setPaused(); 3370 break; 3371 case TrackBase::RESUMING: 3372 // ramp up is not yet implemented 3373 track->mState = TrackBase::ACTIVE; 3374 break; 3375 case TrackBase::ACTIVE: 3376 if (recentFull > 0 || recentPartial > 0) { 3377 // track has provided at least some frames recently: reset retry count 3378 track->mRetryCount = kMaxTrackRetries; 3379 } 3380 if (recentUnderruns == 0) { 3381 // no recent underruns: stay active 3382 break; 3383 } 3384 // there has recently been an underrun of some kind 3385 if (track->sharedBuffer() == 0) { 3386 // were any of the recent underruns "empty" (no frames available)? 3387 if (recentEmpty == 0) { 3388 // no, then ignore the partial underruns as they are allowed indefinitely 3389 break; 3390 } 3391 // there has recently been an "empty" underrun: decrement the retry counter 3392 if (--(track->mRetryCount) > 0) { 3393 break; 3394 } 3395 // indicate to client process that the track was disabled because of underrun; 3396 // it will then automatically call start() when data is available 3397 android_atomic_or(CBLK_DISABLED, &track->mCblk->mFlags); 3398 // remove from active list, but state remains ACTIVE [confusing but true] 3399 isActive = false; 3400 break; 3401 } 3402 // fall through 3403 case TrackBase::STOPPING_2: 3404 case TrackBase::PAUSED: 3405 case TrackBase::STOPPED: 3406 case TrackBase::FLUSHED: // flush() while active 3407 // Check for presentation complete if track is inactive 3408 // We have consumed all the buffers of this track. 3409 // This would be incomplete if we auto-paused on underrun 3410 { 3411 size_t audioHALFrames = 3412 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 3413 size_t framesWritten = mBytesWritten / mFrameSize; 3414 if (!(mStandby || track->presentationComplete(framesWritten, audioHALFrames))) { 3415 // track stays in active list until presentation is complete 3416 break; 3417 } 3418 } 3419 if (track->isStopping_2()) { 3420 track->mState = TrackBase::STOPPED; 3421 } 3422 if (track->isStopped()) { 3423 // Can't reset directly, as fast mixer is still polling this track 3424 // track->reset(); 3425 // So instead mark this track as needing to be reset after push with ack 3426 resetMask |= 1 << i; 3427 } 3428 isActive = false; 3429 break; 3430 case TrackBase::IDLE: 3431 default: 3432 LOG_ALWAYS_FATAL("unexpected track state %d", track->mState); 3433 } 3434 3435 if (isActive) { 3436 // was it previously inactive? 3437 if (!(state->mTrackMask & (1 << j))) { 3438 ExtendedAudioBufferProvider *eabp = track; 3439 VolumeProvider *vp = track; 3440 fastTrack->mBufferProvider = eabp; 3441 fastTrack->mVolumeProvider = vp; 3442 fastTrack->mChannelMask = track->mChannelMask; 3443 fastTrack->mFormat = track->mFormat; 3444 fastTrack->mGeneration++; 3445 state->mTrackMask |= 1 << j; 3446 didModify = true; 3447 // no acknowledgement required for newly active tracks 3448 } 3449 // cache the combined master volume and stream type volume for fast mixer; this 3450 // lacks any synchronization or barrier so VolumeProvider may read a stale value 3451 track->mCachedVolume = masterVolume * mStreamTypes[track->streamType()].volume; 3452 ++fastTracks; 3453 } else { 3454 // was it previously active? 3455 if (state->mTrackMask & (1 << j)) { 3456 fastTrack->mBufferProvider = NULL; 3457 fastTrack->mGeneration++; 3458 state->mTrackMask &= ~(1 << j); 3459 didModify = true; 3460 // If any fast tracks were removed, we must wait for acknowledgement 3461 // because we're about to decrement the last sp<> on those tracks. 3462 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3463 } else { 3464 LOG_ALWAYS_FATAL("fast track %d should have been active", j); 3465 } 3466 tracksToRemove->add(track); 3467 // Avoids a misleading display in dumpsys 3468 track->mObservedUnderruns.mBitFields.mMostRecent = UNDERRUN_FULL; 3469 } 3470 continue; 3471 } 3472 3473 { // local variable scope to avoid goto warning 3474 3475 audio_track_cblk_t* cblk = track->cblk(); 3476 3477 // The first time a track is added we wait 3478 // for all its buffers to be filled before processing it 3479 int name = track->name(); 3480 // make sure that we have enough frames to mix one full buffer. 3481 // enforce this condition only once to enable draining the buffer in case the client 3482 // app does not call stop() and relies on underrun to stop: 3483 // hence the test on (mMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 3484 // during last round 3485 size_t desiredFrames; 3486 uint32_t sr = track->sampleRate(); 3487 if (sr == mSampleRate) { 3488 desiredFrames = mNormalFrameCount; 3489 } else { 3490 desiredFrames = sourceFramesNeeded(sr, mNormalFrameCount, mSampleRate); 3491 // add frames already consumed but not yet released by the resampler 3492 // because mAudioTrackServerProxy->framesReady() will include these frames 3493 desiredFrames += mAudioMixer->getUnreleasedFrames(track->name()); 3494#if 0 3495 // the minimum track buffer size is normally twice the number of frames necessary 3496 // to fill one buffer and the resampler should not leave more than one buffer worth 3497 // of unreleased frames after each pass, but just in case... 3498 ALOG_ASSERT(desiredFrames <= cblk->frameCount_); 3499#endif 3500 } 3501 uint32_t minFrames = 1; 3502 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() && 3503 (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) { 3504 minFrames = desiredFrames; 3505 } 3506 3507 size_t framesReady = track->framesReady(); 3508 if (ATRACE_ENABLED()) { 3509 // I wish we had formatted trace names 3510 char traceName[16]; 3511 strcpy(traceName, "nRdy"); 3512 int name = track->name(); 3513 if (AudioMixer::TRACK0 <= name && 3514 name < (int) (AudioMixer::TRACK0 + AudioMixer::MAX_NUM_TRACKS)) { 3515 name -= AudioMixer::TRACK0; 3516 traceName[4] = (name / 10) + '0'; 3517 traceName[5] = (name % 10) + '0'; 3518 } else { 3519 traceName[4] = '?'; 3520 traceName[5] = '?'; 3521 } 3522 traceName[6] = '\0'; 3523 ATRACE_INT(traceName, framesReady); 3524 } 3525 if ((framesReady >= minFrames) && track->isReady() && 3526 !track->isPaused() && !track->isTerminated()) 3527 { 3528 ALOGVV("track %d s=%08x [OK] on thread %p", name, cblk->mServer, this); 3529 3530 mixedTracks++; 3531 3532 // track->mainBuffer() != mSinkBuffer or mMixerBuffer means 3533 // there is an effect chain connected to the track 3534 chain.clear(); 3535 if (track->mainBuffer() != mSinkBuffer && 3536 track->mainBuffer() != mMixerBuffer) { 3537 if (mEffectBufferEnabled) { 3538 mEffectBufferValid = true; // Later can set directly. 3539 } 3540 chain = getEffectChain_l(track->sessionId()); 3541 // Delegate volume control to effect in track effect chain if needed 3542 if (chain != 0) { 3543 tracksWithEffect++; 3544 } else { 3545 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on " 3546 "session %d", 3547 name, track->sessionId()); 3548 } 3549 } 3550 3551 3552 int param = AudioMixer::VOLUME; 3553 if (track->mFillingUpStatus == Track::FS_FILLED) { 3554 // no ramp for the first volume setting 3555 track->mFillingUpStatus = Track::FS_ACTIVE; 3556 if (track->mState == TrackBase::RESUMING) { 3557 track->mState = TrackBase::ACTIVE; 3558 param = AudioMixer::RAMP_VOLUME; 3559 } 3560 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 3561 // FIXME should not make a decision based on mServer 3562 } else if (cblk->mServer != 0) { 3563 // If the track is stopped before the first frame was mixed, 3564 // do not apply ramp 3565 param = AudioMixer::RAMP_VOLUME; 3566 } 3567 3568 // compute volume for this track 3569 uint32_t vl, vr; // in U8.24 integer format 3570 float vlf, vrf, vaf; // in [0.0, 1.0] float format 3571 if (track->isPausing() || mStreamTypes[track->streamType()].mute) { 3572 vl = vr = 0; 3573 vlf = vrf = vaf = 0.; 3574 if (track->isPausing()) { 3575 track->setPaused(); 3576 } 3577 } else { 3578 3579 // read original volumes with volume control 3580 float typeVolume = mStreamTypes[track->streamType()].volume; 3581 float v = masterVolume * typeVolume; 3582 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 3583 gain_minifloat_packed_t vlr = proxy->getVolumeLR(); 3584 vlf = float_from_gain(gain_minifloat_unpack_left(vlr)); 3585 vrf = float_from_gain(gain_minifloat_unpack_right(vlr)); 3586 // track volumes come from shared memory, so can't be trusted and must be clamped 3587 if (vlf > GAIN_FLOAT_UNITY) { 3588 ALOGV("Track left volume out of range: %.3g", vlf); 3589 vlf = GAIN_FLOAT_UNITY; 3590 } 3591 if (vrf > GAIN_FLOAT_UNITY) { 3592 ALOGV("Track right volume out of range: %.3g", vrf); 3593 vrf = GAIN_FLOAT_UNITY; 3594 } 3595 // now apply the master volume and stream type volume 3596 vlf *= v; 3597 vrf *= v; 3598 // assuming master volume and stream type volume each go up to 1.0, 3599 // then derive vl and vr as U8.24 versions for the effect chain 3600 const float scaleto8_24 = MAX_GAIN_INT * MAX_GAIN_INT; 3601 vl = (uint32_t) (scaleto8_24 * vlf); 3602 vr = (uint32_t) (scaleto8_24 * vrf); 3603 // vl and vr are now in U8.24 format 3604 uint16_t sendLevel = proxy->getSendLevel_U4_12(); 3605 // send level comes from shared memory and so may be corrupt 3606 if (sendLevel > MAX_GAIN_INT) { 3607 ALOGV("Track send level out of range: %04X", sendLevel); 3608 sendLevel = MAX_GAIN_INT; 3609 } 3610 // vaf is represented as [0.0, 1.0] float by rescaling sendLevel 3611 vaf = v * sendLevel * (1. / MAX_GAIN_INT); 3612 } 3613 3614 // Delegate volume control to effect in track effect chain if needed 3615 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 3616 // Do not ramp volume if volume is controlled by effect 3617 param = AudioMixer::VOLUME; 3618 // Update remaining floating point volume levels 3619 vlf = (float)vl / (1 << 24); 3620 vrf = (float)vr / (1 << 24); 3621 track->mHasVolumeController = true; 3622 } else { 3623 // force no volume ramp when volume controller was just disabled or removed 3624 // from effect chain to avoid volume spike 3625 if (track->mHasVolumeController) { 3626 param = AudioMixer::VOLUME; 3627 } 3628 track->mHasVolumeController = false; 3629 } 3630 3631 // XXX: these things DON'T need to be done each time 3632 mAudioMixer->setBufferProvider(name, track); 3633 mAudioMixer->enable(name); 3634 3635 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, &vlf); 3636 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, &vrf); 3637 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, &vaf); 3638 mAudioMixer->setParameter( 3639 name, 3640 AudioMixer::TRACK, 3641 AudioMixer::FORMAT, (void *)track->format()); 3642 mAudioMixer->setParameter( 3643 name, 3644 AudioMixer::TRACK, 3645 AudioMixer::CHANNEL_MASK, (void *)(uintptr_t)track->channelMask()); 3646 mAudioMixer->setParameter( 3647 name, 3648 AudioMixer::TRACK, 3649 AudioMixer::MIXER_CHANNEL_MASK, (void *)(uintptr_t)mChannelMask); 3650 // limit track sample rate to 2 x output sample rate, which changes at re-configuration 3651 uint32_t maxSampleRate = mSampleRate * AUDIO_RESAMPLER_DOWN_RATIO_MAX; 3652 uint32_t reqSampleRate = track->mAudioTrackServerProxy->getSampleRate(); 3653 if (reqSampleRate == 0) { 3654 reqSampleRate = mSampleRate; 3655 } else if (reqSampleRate > maxSampleRate) { 3656 reqSampleRate = maxSampleRate; 3657 } 3658 mAudioMixer->setParameter( 3659 name, 3660 AudioMixer::RESAMPLE, 3661 AudioMixer::SAMPLE_RATE, 3662 (void *)(uintptr_t)reqSampleRate); 3663 /* 3664 * Select the appropriate output buffer for the track. 3665 * 3666 * Tracks with effects go into their own effects chain buffer 3667 * and from there into either mEffectBuffer or mSinkBuffer. 3668 * 3669 * Other tracks can use mMixerBuffer for higher precision 3670 * channel accumulation. If this buffer is enabled 3671 * (mMixerBufferEnabled true), then selected tracks will accumulate 3672 * into it. 3673 * 3674 */ 3675 if (mMixerBufferEnabled 3676 && (track->mainBuffer() == mSinkBuffer 3677 || track->mainBuffer() == mMixerBuffer)) { 3678 mAudioMixer->setParameter( 3679 name, 3680 AudioMixer::TRACK, 3681 AudioMixer::MIXER_FORMAT, (void *)mMixerBufferFormat); 3682 mAudioMixer->setParameter( 3683 name, 3684 AudioMixer::TRACK, 3685 AudioMixer::MAIN_BUFFER, (void *)mMixerBuffer); 3686 // TODO: override track->mainBuffer()? 3687 mMixerBufferValid = true; 3688 } else { 3689 mAudioMixer->setParameter( 3690 name, 3691 AudioMixer::TRACK, 3692 AudioMixer::MIXER_FORMAT, (void *)AUDIO_FORMAT_PCM_16_BIT); 3693 mAudioMixer->setParameter( 3694 name, 3695 AudioMixer::TRACK, 3696 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 3697 } 3698 mAudioMixer->setParameter( 3699 name, 3700 AudioMixer::TRACK, 3701 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 3702 3703 // reset retry count 3704 track->mRetryCount = kMaxTrackRetries; 3705 3706 // If one track is ready, set the mixer ready if: 3707 // - the mixer was not ready during previous round OR 3708 // - no other track is not ready 3709 if (mMixerStatusIgnoringFastTracks != MIXER_TRACKS_READY || 3710 mixerStatus != MIXER_TRACKS_ENABLED) { 3711 mixerStatus = MIXER_TRACKS_READY; 3712 } 3713 } else { 3714 if (framesReady < desiredFrames && !track->isStopped() && !track->isPaused()) { 3715 track->mAudioTrackServerProxy->tallyUnderrunFrames(desiredFrames); 3716 } 3717 // clear effect chain input buffer if an active track underruns to avoid sending 3718 // previous audio buffer again to effects 3719 chain = getEffectChain_l(track->sessionId()); 3720 if (chain != 0) { 3721 chain->clearInputBuffer(); 3722 } 3723 3724 ALOGVV("track %d s=%08x [NOT READY] on thread %p", name, cblk->mServer, this); 3725 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3726 track->isStopped() || track->isPaused()) { 3727 // We have consumed all the buffers of this track. 3728 // Remove it from the list of active tracks. 3729 // TODO: use actual buffer filling status instead of latency when available from 3730 // audio HAL 3731 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 3732 size_t framesWritten = mBytesWritten / mFrameSize; 3733 if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) { 3734 if (track->isStopped()) { 3735 track->reset(); 3736 } 3737 tracksToRemove->add(track); 3738 } 3739 } else { 3740 // No buffers for this track. Give it a few chances to 3741 // fill a buffer, then remove it from active list. 3742 if (--(track->mRetryCount) <= 0) { 3743 ALOGI("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 3744 tracksToRemove->add(track); 3745 // indicate to client process that the track was disabled because of underrun; 3746 // it will then automatically call start() when data is available 3747 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 3748 // If one track is not ready, mark the mixer also not ready if: 3749 // - the mixer was ready during previous round OR 3750 // - no other track is ready 3751 } else if (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY || 3752 mixerStatus != MIXER_TRACKS_READY) { 3753 mixerStatus = MIXER_TRACKS_ENABLED; 3754 } 3755 } 3756 mAudioMixer->disable(name); 3757 } 3758 3759 } // local variable scope to avoid goto warning 3760track_is_ready: ; 3761 3762 } 3763 3764 // Push the new FastMixer state if necessary 3765 bool pauseAudioWatchdog = false; 3766 if (didModify) { 3767 state->mFastTracksGen++; 3768 // if the fast mixer was active, but now there are no fast tracks, then put it in cold idle 3769 if (kUseFastMixer == FastMixer_Dynamic && 3770 state->mCommand == FastMixerState::MIX_WRITE && state->mTrackMask <= 1) { 3771 state->mCommand = FastMixerState::COLD_IDLE; 3772 state->mColdFutexAddr = &mFastMixerFutex; 3773 state->mColdGen++; 3774 mFastMixerFutex = 0; 3775 if (kUseFastMixer == FastMixer_Dynamic) { 3776 mNormalSink = mOutputSink; 3777 } 3778 // If we go into cold idle, need to wait for acknowledgement 3779 // so that fast mixer stops doing I/O. 3780 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3781 pauseAudioWatchdog = true; 3782 } 3783 } 3784 if (sq != NULL) { 3785 sq->end(didModify); 3786 sq->push(block); 3787 } 3788#ifdef AUDIO_WATCHDOG 3789 if (pauseAudioWatchdog && mAudioWatchdog != 0) { 3790 mAudioWatchdog->pause(); 3791 } 3792#endif 3793 3794 // Now perform the deferred reset on fast tracks that have stopped 3795 while (resetMask != 0) { 3796 size_t i = __builtin_ctz(resetMask); 3797 ALOG_ASSERT(i < count); 3798 resetMask &= ~(1 << i); 3799 sp<Track> t = mActiveTracks[i].promote(); 3800 if (t == 0) { 3801 continue; 3802 } 3803 Track* track = t.get(); 3804 ALOG_ASSERT(track->isFastTrack() && track->isStopped()); 3805 track->reset(); 3806 } 3807 3808 // remove all the tracks that need to be... 3809 removeTracks_l(*tracksToRemove); 3810 3811 if (getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX) != 0) { 3812 mEffectBufferValid = true; 3813 } 3814 3815 if (mEffectBufferValid) { 3816 // as long as there are effects we should clear the effects buffer, to avoid 3817 // passing a non-clean buffer to the effect chain 3818 memset(mEffectBuffer, 0, mEffectBufferSize); 3819 } 3820 // sink or mix buffer must be cleared if all tracks are connected to an 3821 // effect chain as in this case the mixer will not write to the sink or mix buffer 3822 // and track effects will accumulate into it 3823 if ((mBytesRemaining == 0) && ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 3824 (mixedTracks == 0 && fastTracks > 0))) { 3825 // FIXME as a performance optimization, should remember previous zero status 3826 if (mMixerBufferValid) { 3827 memset(mMixerBuffer, 0, mMixerBufferSize); 3828 // TODO: In testing, mSinkBuffer below need not be cleared because 3829 // the PlaybackThread::threadLoop() copies mMixerBuffer into mSinkBuffer 3830 // after mixing. 3831 // 3832 // To enforce this guarantee: 3833 // ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 3834 // (mixedTracks == 0 && fastTracks > 0)) 3835 // must imply MIXER_TRACKS_READY. 3836 // Later, we may clear buffers regardless, and skip much of this logic. 3837 } 3838 // FIXME as a performance optimization, should remember previous zero status 3839 memset(mSinkBuffer, 0, mNormalFrameCount * mFrameSize); 3840 } 3841 3842 // if any fast tracks, then status is ready 3843 mMixerStatusIgnoringFastTracks = mixerStatus; 3844 if (fastTracks > 0) { 3845 mixerStatus = MIXER_TRACKS_READY; 3846 } 3847 return mixerStatus; 3848} 3849 3850// getTrackName_l() must be called with ThreadBase::mLock held 3851int AudioFlinger::MixerThread::getTrackName_l(audio_channel_mask_t channelMask, 3852 audio_format_t format, int sessionId) 3853{ 3854 return mAudioMixer->getTrackName(channelMask, format, sessionId); 3855} 3856 3857// deleteTrackName_l() must be called with ThreadBase::mLock held 3858void AudioFlinger::MixerThread::deleteTrackName_l(int name) 3859{ 3860 ALOGV("remove track (%d) and delete from mixer", name); 3861 mAudioMixer->deleteTrackName(name); 3862} 3863 3864// checkForNewParameter_l() must be called with ThreadBase::mLock held 3865bool AudioFlinger::MixerThread::checkForNewParameter_l(const String8& keyValuePair, 3866 status_t& status) 3867{ 3868 bool reconfig = false; 3869 3870 status = NO_ERROR; 3871 3872 // if !&IDLE, holds the FastMixer state to restore after new parameters processed 3873 FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE; 3874 if (mFastMixer != 0) { 3875 FastMixerStateQueue *sq = mFastMixer->sq(); 3876 FastMixerState *state = sq->begin(); 3877 if (!(state->mCommand & FastMixerState::IDLE)) { 3878 previousCommand = state->mCommand; 3879 state->mCommand = FastMixerState::HOT_IDLE; 3880 sq->end(); 3881 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3882 } else { 3883 sq->end(false /*didModify*/); 3884 } 3885 } 3886 3887 AudioParameter param = AudioParameter(keyValuePair); 3888 int value; 3889 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 3890 reconfig = true; 3891 } 3892 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 3893 if (!isValidPcmSinkFormat((audio_format_t) value)) { 3894 status = BAD_VALUE; 3895 } else { 3896 // no need to save value, since it's constant 3897 reconfig = true; 3898 } 3899 } 3900 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 3901 if (!isValidPcmSinkChannelMask((audio_channel_mask_t) value)) { 3902 status = BAD_VALUE; 3903 } else { 3904 // no need to save value, since it's constant 3905 reconfig = true; 3906 } 3907 } 3908 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3909 // do not accept frame count changes if tracks are open as the track buffer 3910 // size depends on frame count and correct behavior would not be guaranteed 3911 // if frame count is changed after track creation 3912 if (!mTracks.isEmpty()) { 3913 status = INVALID_OPERATION; 3914 } else { 3915 reconfig = true; 3916 } 3917 } 3918 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 3919#ifdef ADD_BATTERY_DATA 3920 // when changing the audio output device, call addBatteryData to notify 3921 // the change 3922 if (mOutDevice != value) { 3923 uint32_t params = 0; 3924 // check whether speaker is on 3925 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 3926 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 3927 } 3928 3929 audio_devices_t deviceWithoutSpeaker 3930 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 3931 // check if any other device (except speaker) is on 3932 if (value & deviceWithoutSpeaker ) { 3933 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 3934 } 3935 3936 if (params != 0) { 3937 addBatteryData(params); 3938 } 3939 } 3940#endif 3941 3942 // forward device change to effects that have requested to be 3943 // aware of attached audio device. 3944 if (value != AUDIO_DEVICE_NONE) { 3945 mOutDevice = value; 3946 for (size_t i = 0; i < mEffectChains.size(); i++) { 3947 mEffectChains[i]->setDevice_l(mOutDevice); 3948 } 3949 } 3950 } 3951 3952 if (status == NO_ERROR) { 3953 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3954 keyValuePair.string()); 3955 if (!mStandby && status == INVALID_OPERATION) { 3956 mOutput->stream->common.standby(&mOutput->stream->common); 3957 mStandby = true; 3958 mBytesWritten = 0; 3959 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3960 keyValuePair.string()); 3961 } 3962 if (status == NO_ERROR && reconfig) { 3963 readOutputParameters_l(); 3964 delete mAudioMixer; 3965 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 3966 for (size_t i = 0; i < mTracks.size() ; i++) { 3967 int name = getTrackName_l(mTracks[i]->mChannelMask, 3968 mTracks[i]->mFormat, mTracks[i]->mSessionId); 3969 if (name < 0) { 3970 break; 3971 } 3972 mTracks[i]->mName = name; 3973 } 3974 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3975 } 3976 } 3977 3978 if (!(previousCommand & FastMixerState::IDLE)) { 3979 ALOG_ASSERT(mFastMixer != 0); 3980 FastMixerStateQueue *sq = mFastMixer->sq(); 3981 FastMixerState *state = sq->begin(); 3982 ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE); 3983 state->mCommand = previousCommand; 3984 sq->end(); 3985 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3986 } 3987 3988 return reconfig; 3989} 3990 3991 3992void AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 3993{ 3994 const size_t SIZE = 256; 3995 char buffer[SIZE]; 3996 String8 result; 3997 3998 PlaybackThread::dumpInternals(fd, args); 3999 4000 dprintf(fd, " AudioMixer tracks: 0x%08x\n", mAudioMixer->trackNames()); 4001 4002 // Make a non-atomic copy of fast mixer dump state so it won't change underneath us 4003 const FastMixerDumpState copy(mFastMixerDumpState); 4004 copy.dump(fd); 4005 4006#ifdef STATE_QUEUE_DUMP 4007 // Similar for state queue 4008 StateQueueObserverDump observerCopy = mStateQueueObserverDump; 4009 observerCopy.dump(fd); 4010 StateQueueMutatorDump mutatorCopy = mStateQueueMutatorDump; 4011 mutatorCopy.dump(fd); 4012#endif 4013 4014#ifdef TEE_SINK 4015 // Write the tee output to a .wav file 4016 dumpTee(fd, mTeeSource, mId); 4017#endif 4018 4019#ifdef AUDIO_WATCHDOG 4020 if (mAudioWatchdog != 0) { 4021 // Make a non-atomic copy of audio watchdog dump so it won't change underneath us 4022 AudioWatchdogDump wdCopy = mAudioWatchdogDump; 4023 wdCopy.dump(fd); 4024 } 4025#endif 4026} 4027 4028uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const 4029{ 4030 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000) / 2; 4031} 4032 4033uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() const 4034{ 4035 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000); 4036} 4037 4038void AudioFlinger::MixerThread::cacheParameters_l() 4039{ 4040 PlaybackThread::cacheParameters_l(); 4041 4042 // FIXME: Relaxed timing because of a certain device that can't meet latency 4043 // Should be reduced to 2x after the vendor fixes the driver issue 4044 // increase threshold again due to low power audio mode. The way this warning 4045 // threshold is calculated and its usefulness should be reconsidered anyway. 4046 maxPeriod = seconds(mNormalFrameCount) / mSampleRate * 15; 4047} 4048 4049// ---------------------------------------------------------------------------- 4050 4051AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 4052 AudioStreamOut* output, audio_io_handle_t id, audio_devices_t device) 4053 : PlaybackThread(audioFlinger, output, id, device, DIRECT) 4054 // mLeftVolFloat, mRightVolFloat 4055{ 4056} 4057 4058AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 4059 AudioStreamOut* output, audio_io_handle_t id, uint32_t device, 4060 ThreadBase::type_t type) 4061 : PlaybackThread(audioFlinger, output, id, device, type) 4062 // mLeftVolFloat, mRightVolFloat 4063{ 4064} 4065 4066AudioFlinger::DirectOutputThread::~DirectOutputThread() 4067{ 4068} 4069 4070void AudioFlinger::DirectOutputThread::processVolume_l(Track *track, bool lastTrack) 4071{ 4072 audio_track_cblk_t* cblk = track->cblk(); 4073 float left, right; 4074 4075 if (mMasterMute || mStreamTypes[track->streamType()].mute) { 4076 left = right = 0; 4077 } else { 4078 float typeVolume = mStreamTypes[track->streamType()].volume; 4079 float v = mMasterVolume * typeVolume; 4080 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 4081 gain_minifloat_packed_t vlr = proxy->getVolumeLR(); 4082 left = float_from_gain(gain_minifloat_unpack_left(vlr)); 4083 if (left > GAIN_FLOAT_UNITY) { 4084 left = GAIN_FLOAT_UNITY; 4085 } 4086 left *= v; 4087 right = float_from_gain(gain_minifloat_unpack_right(vlr)); 4088 if (right > GAIN_FLOAT_UNITY) { 4089 right = GAIN_FLOAT_UNITY; 4090 } 4091 right *= v; 4092 } 4093 4094 if (lastTrack) { 4095 if (left != mLeftVolFloat || right != mRightVolFloat) { 4096 mLeftVolFloat = left; 4097 mRightVolFloat = right; 4098 4099 // Convert volumes from float to 8.24 4100 uint32_t vl = (uint32_t)(left * (1 << 24)); 4101 uint32_t vr = (uint32_t)(right * (1 << 24)); 4102 4103 // Delegate volume control to effect in track effect chain if needed 4104 // only one effect chain can be present on DirectOutputThread, so if 4105 // there is one, the track is connected to it 4106 if (!mEffectChains.isEmpty()) { 4107 mEffectChains[0]->setVolume_l(&vl, &vr); 4108 left = (float)vl / (1 << 24); 4109 right = (float)vr / (1 << 24); 4110 } 4111 if (mOutput->stream->set_volume) { 4112 mOutput->stream->set_volume(mOutput->stream, left, right); 4113 } 4114 } 4115 } 4116} 4117 4118 4119AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l( 4120 Vector< sp<Track> > *tracksToRemove 4121) 4122{ 4123 size_t count = mActiveTracks.size(); 4124 mixer_state mixerStatus = MIXER_IDLE; 4125 bool doHwPause = false; 4126 bool doHwResume = false; 4127 bool flushPending = false; 4128 4129 // find out which tracks need to be processed 4130 for (size_t i = 0; i < count; i++) { 4131 sp<Track> t = mActiveTracks[i].promote(); 4132 // The track died recently 4133 if (t == 0) { 4134 continue; 4135 } 4136 4137 Track* const track = t.get(); 4138 audio_track_cblk_t* cblk = track->cblk(); 4139 // Only consider last track started for volume and mixer state control. 4140 // In theory an older track could underrun and restart after the new one starts 4141 // but as we only care about the transition phase between two tracks on a 4142 // direct output, it is not a problem to ignore the underrun case. 4143 sp<Track> l = mLatestActiveTrack.promote(); 4144 bool last = l.get() == track; 4145 4146 if (mHwSupportsPause && track->isPausing()) { 4147 track->setPaused(); 4148 if (last && !mHwPaused) { 4149 doHwPause = true; 4150 mHwPaused = true; 4151 } 4152 tracksToRemove->add(track); 4153 } else if (track->isFlushPending()) { 4154 track->flushAck(); 4155 if (last) { 4156 flushPending = true; 4157 } 4158 } else if (mHwSupportsPause && track->isResumePending()){ 4159 track->resumeAck(); 4160 if (last) { 4161 if (mHwPaused) { 4162 doHwResume = true; 4163 mHwPaused = false; 4164 } 4165 } 4166 } 4167 4168 // The first time a track is added we wait 4169 // for all its buffers to be filled before processing it. 4170 // Allow draining the buffer in case the client 4171 // app does not call stop() and relies on underrun to stop: 4172 // hence the test on (track->mRetryCount > 1). 4173 // If retryCount<=1 then track is about to underrun and be removed. 4174 uint32_t minFrames; 4175 if ((track->sharedBuffer() == 0) && !track->isStopping_1() && !track->isPausing() 4176 && (track->mRetryCount > 1)) { 4177 minFrames = mNormalFrameCount; 4178 } else { 4179 minFrames = 1; 4180 } 4181 4182 if ((track->framesReady() >= minFrames) && track->isReady() && !track->isPaused() && 4183 !track->isStopping_2() && !track->isStopped()) 4184 { 4185 ALOGVV("track %d s=%08x [OK]", track->name(), cblk->mServer); 4186 4187 if (track->mFillingUpStatus == Track::FS_FILLED) { 4188 track->mFillingUpStatus = Track::FS_ACTIVE; 4189 // make sure processVolume_l() will apply new volume even if 0 4190 mLeftVolFloat = mRightVolFloat = -1.0; 4191 if (!mHwSupportsPause) { 4192 track->resumeAck(); 4193 } 4194 } 4195 4196 // compute volume for this track 4197 processVolume_l(track, last); 4198 if (last) { 4199 // reset retry count 4200 track->mRetryCount = kMaxTrackRetriesDirect; 4201 mActiveTrack = t; 4202 mixerStatus = MIXER_TRACKS_READY; 4203 if (usesHwAvSync() && mHwPaused) { 4204 doHwResume = true; 4205 mHwPaused = false; 4206 } 4207 } 4208 } else { 4209 // clear effect chain input buffer if the last active track started underruns 4210 // to avoid sending previous audio buffer again to effects 4211 if (!mEffectChains.isEmpty() && last) { 4212 mEffectChains[0]->clearInputBuffer(); 4213 } 4214 if (track->isStopping_1()) { 4215 track->mState = TrackBase::STOPPING_2; 4216 } 4217 if ((track->sharedBuffer() != 0) || track->isStopped() || 4218 track->isStopping_2() || track->isPaused()) { 4219 // We have consumed all the buffers of this track. 4220 // Remove it from the list of active tracks. 4221 size_t audioHALFrames; 4222 if (audio_is_linear_pcm(mFormat)) { 4223 audioHALFrames = (latency_l() * mSampleRate) / 1000; 4224 } else { 4225 audioHALFrames = 0; 4226 } 4227 4228 size_t framesWritten = mBytesWritten / mFrameSize; 4229 if (mStandby || !last || 4230 track->presentationComplete(framesWritten, audioHALFrames)) { 4231 if (track->isStopping_2()) { 4232 track->mState = TrackBase::STOPPED; 4233 } 4234 if (track->isStopped()) { 4235 track->reset(); 4236 } 4237 tracksToRemove->add(track); 4238 } 4239 } else { 4240 // No buffers for this track. Give it a few chances to 4241 // fill a buffer, then remove it from active list. 4242 // Only consider last track started for mixer state control 4243 if (--(track->mRetryCount) <= 0) { 4244 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 4245 tracksToRemove->add(track); 4246 // indicate to client process that the track was disabled because of underrun; 4247 // it will then automatically call start() when data is available 4248 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 4249 } else if (last) { 4250 mixerStatus = MIXER_TRACKS_ENABLED; 4251 if (usesHwAvSync() && !mHwPaused && !mStandby) { 4252 doHwPause = true; 4253 mHwPaused = true; 4254 } 4255 } 4256 } 4257 } 4258 } 4259 4260 // if an active track did not command a flush, check for pending flush on stopped tracks 4261 if (!flushPending) { 4262 for (size_t i = 0; i < mTracks.size(); i++) { 4263 if (mTracks[i]->isFlushPending()) { 4264 mTracks[i]->flushAck(); 4265 flushPending = true; 4266 } 4267 } 4268 } 4269 4270 // make sure the pause/flush/resume sequence is executed in the right order. 4271 // If a flush is pending and a track is active but the HW is not paused, force a HW pause 4272 // before flush and then resume HW. This can happen in case of pause/flush/resume 4273 // if resume is received before pause is executed. 4274 if (mHwSupportsPause && !mStandby && 4275 (doHwPause || (flushPending && !mHwPaused && (count != 0)))) { 4276 mOutput->stream->pause(mOutput->stream); 4277 } 4278 if (flushPending) { 4279 flushHw_l(); 4280 } 4281 if (mHwSupportsPause && !mStandby && doHwResume) { 4282 mOutput->stream->resume(mOutput->stream); 4283 } 4284 // remove all the tracks that need to be... 4285 removeTracks_l(*tracksToRemove); 4286 4287 return mixerStatus; 4288} 4289 4290void AudioFlinger::DirectOutputThread::threadLoop_mix() 4291{ 4292 size_t frameCount = mFrameCount; 4293 int8_t *curBuf = (int8_t *)mSinkBuffer; 4294 // output audio to hardware 4295 while (frameCount) { 4296 AudioBufferProvider::Buffer buffer; 4297 buffer.frameCount = frameCount; 4298 mActiveTrack->getNextBuffer(&buffer); 4299 if (buffer.raw == NULL) { 4300 memset(curBuf, 0, frameCount * mFrameSize); 4301 break; 4302 } 4303 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 4304 frameCount -= buffer.frameCount; 4305 curBuf += buffer.frameCount * mFrameSize; 4306 mActiveTrack->releaseBuffer(&buffer); 4307 } 4308 mCurrentWriteLength = curBuf - (int8_t *)mSinkBuffer; 4309 sleepTime = 0; 4310 standbyTime = systemTime() + standbyDelay; 4311 mActiveTrack.clear(); 4312} 4313 4314void AudioFlinger::DirectOutputThread::threadLoop_sleepTime() 4315{ 4316 // do not write to HAL when paused 4317 if (mHwPaused || (usesHwAvSync() && mStandby)) { 4318 sleepTime = idleSleepTime; 4319 return; 4320 } 4321 if (sleepTime == 0) { 4322 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4323 sleepTime = activeSleepTime; 4324 } else { 4325 sleepTime = idleSleepTime; 4326 } 4327 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 4328 memset(mSinkBuffer, 0, mFrameCount * mFrameSize); 4329 sleepTime = 0; 4330 } 4331} 4332 4333void AudioFlinger::DirectOutputThread::threadLoop_exit() 4334{ 4335 { 4336 Mutex::Autolock _l(mLock); 4337 bool flushPending = false; 4338 for (size_t i = 0; i < mTracks.size(); i++) { 4339 if (mTracks[i]->isFlushPending()) { 4340 mTracks[i]->flushAck(); 4341 flushPending = true; 4342 } 4343 } 4344 if (flushPending) { 4345 flushHw_l(); 4346 } 4347 } 4348 PlaybackThread::threadLoop_exit(); 4349} 4350 4351// must be called with thread mutex locked 4352bool AudioFlinger::DirectOutputThread::shouldStandby_l() 4353{ 4354 bool trackPaused = false; 4355 4356 // do not put the HAL in standby when paused. AwesomePlayer clear the offloaded AudioTrack 4357 // after a timeout and we will enter standby then. 4358 if (mTracks.size() > 0) { 4359 trackPaused = mTracks[mTracks.size() - 1]->isPaused(); 4360 } 4361 4362 return !mStandby && !(trackPaused || (usesHwAvSync() && mHwPaused)); 4363} 4364 4365// getTrackName_l() must be called with ThreadBase::mLock held 4366int AudioFlinger::DirectOutputThread::getTrackName_l(audio_channel_mask_t channelMask __unused, 4367 audio_format_t format __unused, int sessionId __unused) 4368{ 4369 return 0; 4370} 4371 4372// deleteTrackName_l() must be called with ThreadBase::mLock held 4373void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name __unused) 4374{ 4375} 4376 4377// checkForNewParameter_l() must be called with ThreadBase::mLock held 4378bool AudioFlinger::DirectOutputThread::checkForNewParameter_l(const String8& keyValuePair, 4379 status_t& status) 4380{ 4381 bool reconfig = false; 4382 4383 status = NO_ERROR; 4384 4385 AudioParameter param = AudioParameter(keyValuePair); 4386 int value; 4387 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 4388 // forward device change to effects that have requested to be 4389 // aware of attached audio device. 4390 if (value != AUDIO_DEVICE_NONE) { 4391 mOutDevice = value; 4392 for (size_t i = 0; i < mEffectChains.size(); i++) { 4393 mEffectChains[i]->setDevice_l(mOutDevice); 4394 } 4395 } 4396 } 4397 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 4398 // do not accept frame count changes if tracks are open as the track buffer 4399 // size depends on frame count and correct behavior would not be garantied 4400 // if frame count is changed after track creation 4401 if (!mTracks.isEmpty()) { 4402 status = INVALID_OPERATION; 4403 } else { 4404 reconfig = true; 4405 } 4406 } 4407 if (status == NO_ERROR) { 4408 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 4409 keyValuePair.string()); 4410 if (!mStandby && status == INVALID_OPERATION) { 4411 mOutput->stream->common.standby(&mOutput->stream->common); 4412 mStandby = true; 4413 mBytesWritten = 0; 4414 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 4415 keyValuePair.string()); 4416 } 4417 if (status == NO_ERROR && reconfig) { 4418 readOutputParameters_l(); 4419 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 4420 } 4421 } 4422 4423 return reconfig; 4424} 4425 4426uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() const 4427{ 4428 uint32_t time; 4429 if (audio_is_linear_pcm(mFormat)) { 4430 time = PlaybackThread::activeSleepTimeUs(); 4431 } else { 4432 time = 10000; 4433 } 4434 return time; 4435} 4436 4437uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() const 4438{ 4439 uint32_t time; 4440 if (audio_is_linear_pcm(mFormat)) { 4441 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 4442 } else { 4443 time = 10000; 4444 } 4445 return time; 4446} 4447 4448uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() const 4449{ 4450 uint32_t time; 4451 if (audio_is_linear_pcm(mFormat)) { 4452 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 4453 } else { 4454 time = 10000; 4455 } 4456 return time; 4457} 4458 4459void AudioFlinger::DirectOutputThread::cacheParameters_l() 4460{ 4461 PlaybackThread::cacheParameters_l(); 4462 4463 // use shorter standby delay as on normal output to release 4464 // hardware resources as soon as possible 4465 if (audio_is_linear_pcm(mFormat)) { 4466 standbyDelay = microseconds(activeSleepTime*2); 4467 } else { 4468 standbyDelay = kOffloadStandbyDelayNs; 4469 } 4470} 4471 4472void AudioFlinger::DirectOutputThread::flushHw_l() 4473{ 4474 if (mOutput->stream->flush != NULL) { 4475 mOutput->stream->flush(mOutput->stream); 4476 } 4477 mHwPaused = false; 4478} 4479 4480// ---------------------------------------------------------------------------- 4481 4482AudioFlinger::AsyncCallbackThread::AsyncCallbackThread( 4483 const wp<AudioFlinger::PlaybackThread>& playbackThread) 4484 : Thread(false /*canCallJava*/), 4485 mPlaybackThread(playbackThread), 4486 mWriteAckSequence(0), 4487 mDrainSequence(0) 4488{ 4489} 4490 4491AudioFlinger::AsyncCallbackThread::~AsyncCallbackThread() 4492{ 4493} 4494 4495void AudioFlinger::AsyncCallbackThread::onFirstRef() 4496{ 4497 run("Offload Cbk", ANDROID_PRIORITY_URGENT_AUDIO); 4498} 4499 4500bool AudioFlinger::AsyncCallbackThread::threadLoop() 4501{ 4502 while (!exitPending()) { 4503 uint32_t writeAckSequence; 4504 uint32_t drainSequence; 4505 4506 { 4507 Mutex::Autolock _l(mLock); 4508 while (!((mWriteAckSequence & 1) || 4509 (mDrainSequence & 1) || 4510 exitPending())) { 4511 mWaitWorkCV.wait(mLock); 4512 } 4513 4514 if (exitPending()) { 4515 break; 4516 } 4517 ALOGV("AsyncCallbackThread mWriteAckSequence %d mDrainSequence %d", 4518 mWriteAckSequence, mDrainSequence); 4519 writeAckSequence = mWriteAckSequence; 4520 mWriteAckSequence &= ~1; 4521 drainSequence = mDrainSequence; 4522 mDrainSequence &= ~1; 4523 } 4524 { 4525 sp<AudioFlinger::PlaybackThread> playbackThread = mPlaybackThread.promote(); 4526 if (playbackThread != 0) { 4527 if (writeAckSequence & 1) { 4528 playbackThread->resetWriteBlocked(writeAckSequence >> 1); 4529 } 4530 if (drainSequence & 1) { 4531 playbackThread->resetDraining(drainSequence >> 1); 4532 } 4533 } 4534 } 4535 } 4536 return false; 4537} 4538 4539void AudioFlinger::AsyncCallbackThread::exit() 4540{ 4541 ALOGV("AsyncCallbackThread::exit"); 4542 Mutex::Autolock _l(mLock); 4543 requestExit(); 4544 mWaitWorkCV.broadcast(); 4545} 4546 4547void AudioFlinger::AsyncCallbackThread::setWriteBlocked(uint32_t sequence) 4548{ 4549 Mutex::Autolock _l(mLock); 4550 // bit 0 is cleared 4551 mWriteAckSequence = sequence << 1; 4552} 4553 4554void AudioFlinger::AsyncCallbackThread::resetWriteBlocked() 4555{ 4556 Mutex::Autolock _l(mLock); 4557 // ignore unexpected callbacks 4558 if (mWriteAckSequence & 2) { 4559 mWriteAckSequence |= 1; 4560 mWaitWorkCV.signal(); 4561 } 4562} 4563 4564void AudioFlinger::AsyncCallbackThread::setDraining(uint32_t sequence) 4565{ 4566 Mutex::Autolock _l(mLock); 4567 // bit 0 is cleared 4568 mDrainSequence = sequence << 1; 4569} 4570 4571void AudioFlinger::AsyncCallbackThread::resetDraining() 4572{ 4573 Mutex::Autolock _l(mLock); 4574 // ignore unexpected callbacks 4575 if (mDrainSequence & 2) { 4576 mDrainSequence |= 1; 4577 mWaitWorkCV.signal(); 4578 } 4579} 4580 4581 4582// ---------------------------------------------------------------------------- 4583AudioFlinger::OffloadThread::OffloadThread(const sp<AudioFlinger>& audioFlinger, 4584 AudioStreamOut* output, audio_io_handle_t id, uint32_t device) 4585 : DirectOutputThread(audioFlinger, output, id, device, OFFLOAD), 4586 mPausedBytesRemaining(0) 4587{ 4588 //FIXME: mStandby should be set to true by ThreadBase constructor 4589 mStandby = true; 4590} 4591 4592void AudioFlinger::OffloadThread::threadLoop_exit() 4593{ 4594 if (mFlushPending || mHwPaused) { 4595 // If a flush is pending or track was paused, just discard buffered data 4596 flushHw_l(); 4597 } else { 4598 mMixerStatus = MIXER_DRAIN_ALL; 4599 threadLoop_drain(); 4600 } 4601 if (mUseAsyncWrite) { 4602 ALOG_ASSERT(mCallbackThread != 0); 4603 mCallbackThread->exit(); 4604 } 4605 PlaybackThread::threadLoop_exit(); 4606} 4607 4608AudioFlinger::PlaybackThread::mixer_state AudioFlinger::OffloadThread::prepareTracks_l( 4609 Vector< sp<Track> > *tracksToRemove 4610) 4611{ 4612 size_t count = mActiveTracks.size(); 4613 4614 mixer_state mixerStatus = MIXER_IDLE; 4615 bool doHwPause = false; 4616 bool doHwResume = false; 4617 4618 ALOGV("OffloadThread::prepareTracks_l active tracks %d", count); 4619 4620 // find out which tracks need to be processed 4621 for (size_t i = 0; i < count; i++) { 4622 sp<Track> t = mActiveTracks[i].promote(); 4623 // The track died recently 4624 if (t == 0) { 4625 continue; 4626 } 4627 Track* const track = t.get(); 4628 audio_track_cblk_t* cblk = track->cblk(); 4629 // Only consider last track started for volume and mixer state control. 4630 // In theory an older track could underrun and restart after the new one starts 4631 // but as we only care about the transition phase between two tracks on a 4632 // direct output, it is not a problem to ignore the underrun case. 4633 sp<Track> l = mLatestActiveTrack.promote(); 4634 bool last = l.get() == track; 4635 4636 if (track->isInvalid()) { 4637 ALOGW("An invalidated track shouldn't be in active list"); 4638 tracksToRemove->add(track); 4639 continue; 4640 } 4641 4642 if (track->mState == TrackBase::IDLE) { 4643 ALOGW("An idle track shouldn't be in active list"); 4644 continue; 4645 } 4646 4647 if (track->isPausing()) { 4648 track->setPaused(); 4649 if (last) { 4650 if (!mHwPaused) { 4651 doHwPause = true; 4652 mHwPaused = true; 4653 } 4654 // If we were part way through writing the mixbuffer to 4655 // the HAL we must save this until we resume 4656 // BUG - this will be wrong if a different track is made active, 4657 // in that case we want to discard the pending data in the 4658 // mixbuffer and tell the client to present it again when the 4659 // track is resumed 4660 mPausedWriteLength = mCurrentWriteLength; 4661 mPausedBytesRemaining = mBytesRemaining; 4662 mBytesRemaining = 0; // stop writing 4663 } 4664 tracksToRemove->add(track); 4665 } else if (track->isFlushPending()) { 4666 track->flushAck(); 4667 if (last) { 4668 mFlushPending = true; 4669 } 4670 } else if (track->isResumePending()){ 4671 track->resumeAck(); 4672 if (last) { 4673 if (mPausedBytesRemaining) { 4674 // Need to continue write that was interrupted 4675 mCurrentWriteLength = mPausedWriteLength; 4676 mBytesRemaining = mPausedBytesRemaining; 4677 mPausedBytesRemaining = 0; 4678 } 4679 if (mHwPaused) { 4680 doHwResume = true; 4681 mHwPaused = false; 4682 // threadLoop_mix() will handle the case that we need to 4683 // resume an interrupted write 4684 } 4685 // enable write to audio HAL 4686 sleepTime = 0; 4687 4688 // Do not handle new data in this iteration even if track->framesReady() 4689 mixerStatus = MIXER_TRACKS_ENABLED; 4690 } 4691 } else if (track->framesReady() && track->isReady() && 4692 !track->isPaused() && !track->isTerminated() && !track->isStopping_2()) { 4693 ALOGVV("OffloadThread: track %d s=%08x [OK]", track->name(), cblk->mServer); 4694 if (track->mFillingUpStatus == Track::FS_FILLED) { 4695 track->mFillingUpStatus = Track::FS_ACTIVE; 4696 // make sure processVolume_l() will apply new volume even if 0 4697 mLeftVolFloat = mRightVolFloat = -1.0; 4698 } 4699 4700 if (last) { 4701 sp<Track> previousTrack = mPreviousTrack.promote(); 4702 if (previousTrack != 0) { 4703 if (track != previousTrack.get()) { 4704 // Flush any data still being written from last track 4705 mBytesRemaining = 0; 4706 if (mPausedBytesRemaining) { 4707 // Last track was paused so we also need to flush saved 4708 // mixbuffer state and invalidate track so that it will 4709 // re-submit that unwritten data when it is next resumed 4710 mPausedBytesRemaining = 0; 4711 // Invalidate is a bit drastic - would be more efficient 4712 // to have a flag to tell client that some of the 4713 // previously written data was lost 4714 previousTrack->invalidate(); 4715 } 4716 // flush data already sent to the DSP if changing audio session as audio 4717 // comes from a different source. Also invalidate previous track to force a 4718 // seek when resuming. 4719 if (previousTrack->sessionId() != track->sessionId()) { 4720 previousTrack->invalidate(); 4721 } 4722 } 4723 } 4724 mPreviousTrack = track; 4725 // reset retry count 4726 track->mRetryCount = kMaxTrackRetriesOffload; 4727 mActiveTrack = t; 4728 mixerStatus = MIXER_TRACKS_READY; 4729 } 4730 } else { 4731 ALOGVV("OffloadThread: track %d s=%08x [NOT READY]", track->name(), cblk->mServer); 4732 if (track->isStopping_1()) { 4733 // Hardware buffer can hold a large amount of audio so we must 4734 // wait for all current track's data to drain before we say 4735 // that the track is stopped. 4736 if (mBytesRemaining == 0) { 4737 // Only start draining when all data in mixbuffer 4738 // has been written 4739 ALOGV("OffloadThread: underrun and STOPPING_1 -> draining, STOPPING_2"); 4740 track->mState = TrackBase::STOPPING_2; // so presentation completes after drain 4741 // do not drain if no data was ever sent to HAL (mStandby == true) 4742 if (last && !mStandby) { 4743 // do not modify drain sequence if we are already draining. This happens 4744 // when resuming from pause after drain. 4745 if ((mDrainSequence & 1) == 0) { 4746 sleepTime = 0; 4747 standbyTime = systemTime() + standbyDelay; 4748 mixerStatus = MIXER_DRAIN_TRACK; 4749 mDrainSequence += 2; 4750 } 4751 if (mHwPaused) { 4752 // It is possible to move from PAUSED to STOPPING_1 without 4753 // a resume so we must ensure hardware is running 4754 doHwResume = true; 4755 mHwPaused = false; 4756 } 4757 } 4758 } 4759 } else if (track->isStopping_2()) { 4760 // Drain has completed or we are in standby, signal presentation complete 4761 if (!(mDrainSequence & 1) || !last || mStandby) { 4762 track->mState = TrackBase::STOPPED; 4763 size_t audioHALFrames = 4764 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 4765 size_t framesWritten = 4766 mBytesWritten / audio_stream_out_frame_size(mOutput->stream); 4767 track->presentationComplete(framesWritten, audioHALFrames); 4768 track->reset(); 4769 tracksToRemove->add(track); 4770 } 4771 } else { 4772 // No buffers for this track. Give it a few chances to 4773 // fill a buffer, then remove it from active list. 4774 if (--(track->mRetryCount) <= 0) { 4775 ALOGV("OffloadThread: BUFFER TIMEOUT: remove(%d) from active list", 4776 track->name()); 4777 tracksToRemove->add(track); 4778 // indicate to client process that the track was disabled because of underrun; 4779 // it will then automatically call start() when data is available 4780 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 4781 } else if (last){ 4782 mixerStatus = MIXER_TRACKS_ENABLED; 4783 } 4784 } 4785 } 4786 // compute volume for this track 4787 processVolume_l(track, last); 4788 } 4789 4790 // make sure the pause/flush/resume sequence is executed in the right order. 4791 // If a flush is pending and a track is active but the HW is not paused, force a HW pause 4792 // before flush and then resume HW. This can happen in case of pause/flush/resume 4793 // if resume is received before pause is executed. 4794 if (!mStandby && (doHwPause || (mFlushPending && !mHwPaused && (count != 0)))) { 4795 mOutput->stream->pause(mOutput->stream); 4796 } 4797 if (mFlushPending) { 4798 flushHw_l(); 4799 mFlushPending = false; 4800 } 4801 if (!mStandby && doHwResume) { 4802 mOutput->stream->resume(mOutput->stream); 4803 } 4804 4805 // remove all the tracks that need to be... 4806 removeTracks_l(*tracksToRemove); 4807 4808 return mixerStatus; 4809} 4810 4811// must be called with thread mutex locked 4812bool AudioFlinger::OffloadThread::waitingAsyncCallback_l() 4813{ 4814 ALOGVV("waitingAsyncCallback_l mWriteAckSequence %d mDrainSequence %d", 4815 mWriteAckSequence, mDrainSequence); 4816 if (mUseAsyncWrite && ((mWriteAckSequence & 1) || (mDrainSequence & 1))) { 4817 return true; 4818 } 4819 return false; 4820} 4821 4822bool AudioFlinger::OffloadThread::waitingAsyncCallback() 4823{ 4824 Mutex::Autolock _l(mLock); 4825 return waitingAsyncCallback_l(); 4826} 4827 4828void AudioFlinger::OffloadThread::flushHw_l() 4829{ 4830 DirectOutputThread::flushHw_l(); 4831 // Flush anything still waiting in the mixbuffer 4832 mCurrentWriteLength = 0; 4833 mBytesRemaining = 0; 4834 mPausedWriteLength = 0; 4835 mPausedBytesRemaining = 0; 4836 4837 if (mUseAsyncWrite) { 4838 // discard any pending drain or write ack by incrementing sequence 4839 mWriteAckSequence = (mWriteAckSequence + 2) & ~1; 4840 mDrainSequence = (mDrainSequence + 2) & ~1; 4841 ALOG_ASSERT(mCallbackThread != 0); 4842 mCallbackThread->setWriteBlocked(mWriteAckSequence); 4843 mCallbackThread->setDraining(mDrainSequence); 4844 } 4845} 4846 4847void AudioFlinger::OffloadThread::onAddNewTrack_l() 4848{ 4849 sp<Track> previousTrack = mPreviousTrack.promote(); 4850 sp<Track> latestTrack = mLatestActiveTrack.promote(); 4851 4852 if (previousTrack != 0 && latestTrack != 0 && 4853 (previousTrack->sessionId() != latestTrack->sessionId())) { 4854 mFlushPending = true; 4855 } 4856 PlaybackThread::onAddNewTrack_l(); 4857} 4858 4859// ---------------------------------------------------------------------------- 4860 4861AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 4862 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id) 4863 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->outDevice(), 4864 DUPLICATING), 4865 mWaitTimeMs(UINT_MAX) 4866{ 4867 addOutputTrack(mainThread); 4868} 4869 4870AudioFlinger::DuplicatingThread::~DuplicatingThread() 4871{ 4872 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4873 mOutputTracks[i]->destroy(); 4874 } 4875} 4876 4877void AudioFlinger::DuplicatingThread::threadLoop_mix() 4878{ 4879 // mix buffers... 4880 if (outputsReady(outputTracks)) { 4881 mAudioMixer->process(AudioBufferProvider::kInvalidPTS); 4882 } else { 4883 if (mMixerBufferValid) { 4884 memset(mMixerBuffer, 0, mMixerBufferSize); 4885 } else { 4886 memset(mSinkBuffer, 0, mSinkBufferSize); 4887 } 4888 } 4889 sleepTime = 0; 4890 writeFrames = mNormalFrameCount; 4891 mCurrentWriteLength = mSinkBufferSize; 4892 standbyTime = systemTime() + standbyDelay; 4893} 4894 4895void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() 4896{ 4897 if (sleepTime == 0) { 4898 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4899 sleepTime = activeSleepTime; 4900 } else { 4901 sleepTime = idleSleepTime; 4902 } 4903 } else if (mBytesWritten != 0) { 4904 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4905 writeFrames = mNormalFrameCount; 4906 memset(mSinkBuffer, 0, mSinkBufferSize); 4907 } else { 4908 // flush remaining overflow buffers in output tracks 4909 writeFrames = 0; 4910 } 4911 sleepTime = 0; 4912 } 4913} 4914 4915ssize_t AudioFlinger::DuplicatingThread::threadLoop_write() 4916{ 4917 for (size_t i = 0; i < outputTracks.size(); i++) { 4918 outputTracks[i]->write(mSinkBuffer, writeFrames); 4919 } 4920 mStandby = false; 4921 return (ssize_t)mSinkBufferSize; 4922} 4923 4924void AudioFlinger::DuplicatingThread::threadLoop_standby() 4925{ 4926 // DuplicatingThread implements standby by stopping all tracks 4927 for (size_t i = 0; i < outputTracks.size(); i++) { 4928 outputTracks[i]->stop(); 4929 } 4930} 4931 4932void AudioFlinger::DuplicatingThread::saveOutputTracks() 4933{ 4934 outputTracks = mOutputTracks; 4935} 4936 4937void AudioFlinger::DuplicatingThread::clearOutputTracks() 4938{ 4939 outputTracks.clear(); 4940} 4941 4942void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 4943{ 4944 Mutex::Autolock _l(mLock); 4945 // The downstream MixerThread consumes thread->frameCount() amount of frames per mix pass. 4946 // Adjust for thread->sampleRate() to determine minimum buffer frame count. 4947 // Then triple buffer because Threads do not run synchronously and may not be clock locked. 4948 const size_t frameCount = 4949 3 * sourceFramesNeeded(mSampleRate, thread->frameCount(), thread->sampleRate()); 4950 // TODO: Consider asynchronous sample rate conversion to handle clock disparity 4951 // from different OutputTracks and their associated MixerThreads (e.g. one may 4952 // nearly empty and the other may be dropping data). 4953 4954 sp<OutputTrack> outputTrack = new OutputTrack(thread, 4955 this, 4956 mSampleRate, 4957 mFormat, 4958 mChannelMask, 4959 frameCount, 4960 IPCThreadState::self()->getCallingUid()); 4961 if (outputTrack->cblk() != NULL) { 4962 thread->setStreamVolume(AUDIO_STREAM_PATCH, 1.0f); 4963 mOutputTracks.add(outputTrack); 4964 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack.get(), thread); 4965 updateWaitTime_l(); 4966 } 4967} 4968 4969void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 4970{ 4971 Mutex::Autolock _l(mLock); 4972 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4973 if (mOutputTracks[i]->thread() == thread) { 4974 mOutputTracks[i]->destroy(); 4975 mOutputTracks.removeAt(i); 4976 updateWaitTime_l(); 4977 return; 4978 } 4979 } 4980 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 4981} 4982 4983// caller must hold mLock 4984void AudioFlinger::DuplicatingThread::updateWaitTime_l() 4985{ 4986 mWaitTimeMs = UINT_MAX; 4987 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4988 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 4989 if (strong != 0) { 4990 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 4991 if (waitTimeMs < mWaitTimeMs) { 4992 mWaitTimeMs = waitTimeMs; 4993 } 4994 } 4995 } 4996} 4997 4998 4999bool AudioFlinger::DuplicatingThread::outputsReady( 5000 const SortedVector< sp<OutputTrack> > &outputTracks) 5001{ 5002 for (size_t i = 0; i < outputTracks.size(); i++) { 5003 sp<ThreadBase> thread = outputTracks[i]->thread().promote(); 5004 if (thread == 0) { 5005 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", 5006 outputTracks[i].get()); 5007 return false; 5008 } 5009 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 5010 // see note at standby() declaration 5011 if (playbackThread->standby() && !playbackThread->isSuspended()) { 5012 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), 5013 thread.get()); 5014 return false; 5015 } 5016 } 5017 return true; 5018} 5019 5020uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() const 5021{ 5022 return (mWaitTimeMs * 1000) / 2; 5023} 5024 5025void AudioFlinger::DuplicatingThread::cacheParameters_l() 5026{ 5027 // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first 5028 updateWaitTime_l(); 5029 5030 MixerThread::cacheParameters_l(); 5031} 5032 5033// ---------------------------------------------------------------------------- 5034// Record 5035// ---------------------------------------------------------------------------- 5036 5037AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 5038 AudioStreamIn *input, 5039 audio_io_handle_t id, 5040 audio_devices_t outDevice, 5041 audio_devices_t inDevice 5042#ifdef TEE_SINK 5043 , const sp<NBAIO_Sink>& teeSink 5044#endif 5045 ) : 5046 ThreadBase(audioFlinger, id, outDevice, inDevice, RECORD), 5047 mInput(input), mActiveTracksGen(0), mRsmpInBuffer(NULL), 5048 // mRsmpInFrames and mRsmpInFramesP2 are set by readInputParameters_l() 5049 mRsmpInRear(0) 5050#ifdef TEE_SINK 5051 , mTeeSink(teeSink) 5052#endif 5053 , mReadOnlyHeap(new MemoryDealer(kRecordThreadReadOnlyHeapSize, 5054 "RecordThreadRO", MemoryHeapBase::READ_ONLY)) 5055 // mFastCapture below 5056 , mFastCaptureFutex(0) 5057 // mInputSource 5058 // mPipeSink 5059 // mPipeSource 5060 , mPipeFramesP2(0) 5061 // mPipeMemory 5062 // mFastCaptureNBLogWriter 5063 , mFastTrackAvail(false) 5064{ 5065 snprintf(mName, kNameLength, "AudioIn_%X", id); 5066 mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mName); 5067 5068 readInputParameters_l(); 5069 5070 // create an NBAIO source for the HAL input stream, and negotiate 5071 mInputSource = new AudioStreamInSource(input->stream); 5072 size_t numCounterOffers = 0; 5073 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount, mFormat)}; 5074 ssize_t index = mInputSource->negotiate(offers, 1, NULL, numCounterOffers); 5075 ALOG_ASSERT(index == 0); 5076 5077 // initialize fast capture depending on configuration 5078 bool initFastCapture; 5079 switch (kUseFastCapture) { 5080 case FastCapture_Never: 5081 initFastCapture = false; 5082 break; 5083 case FastCapture_Always: 5084 initFastCapture = true; 5085 break; 5086 case FastCapture_Static: 5087 uint32_t primaryOutputSampleRate; 5088 { 5089 AutoMutex _l(audioFlinger->mHardwareLock); 5090 primaryOutputSampleRate = audioFlinger->mPrimaryOutputSampleRate; 5091 } 5092 initFastCapture = 5093 // either capture sample rate is same as (a reasonable) primary output sample rate 5094 (((primaryOutputSampleRate == 44100 || primaryOutputSampleRate == 48000) && 5095 (mSampleRate == primaryOutputSampleRate)) || 5096 // or primary output sample rate is unknown, and capture sample rate is reasonable 5097 ((primaryOutputSampleRate == 0) && 5098 ((mSampleRate == 44100 || mSampleRate == 48000)))) && 5099 // and the buffer size is < 12 ms 5100 (mFrameCount * 1000) / mSampleRate < 12; 5101 break; 5102 // case FastCapture_Dynamic: 5103 } 5104 5105 if (initFastCapture) { 5106 // create a Pipe for FastMixer to write to, and for us and fast tracks to read from 5107 NBAIO_Format format = mInputSource->format(); 5108 size_t pipeFramesP2 = roundup(mSampleRate / 25); // double-buffering of 20 ms each 5109 size_t pipeSize = pipeFramesP2 * Format_frameSize(format); 5110 void *pipeBuffer; 5111 const sp<MemoryDealer> roHeap(readOnlyHeap()); 5112 sp<IMemory> pipeMemory; 5113 if ((roHeap == 0) || 5114 (pipeMemory = roHeap->allocate(pipeSize)) == 0 || 5115 (pipeBuffer = pipeMemory->pointer()) == NULL) { 5116 ALOGE("not enough memory for pipe buffer size=%zu", pipeSize); 5117 goto failed; 5118 } 5119 // pipe will be shared directly with fast clients, so clear to avoid leaking old information 5120 memset(pipeBuffer, 0, pipeSize); 5121 Pipe *pipe = new Pipe(pipeFramesP2, format, pipeBuffer); 5122 const NBAIO_Format offers[1] = {format}; 5123 size_t numCounterOffers = 0; 5124 ssize_t index = pipe->negotiate(offers, 1, NULL, numCounterOffers); 5125 ALOG_ASSERT(index == 0); 5126 mPipeSink = pipe; 5127 PipeReader *pipeReader = new PipeReader(*pipe); 5128 numCounterOffers = 0; 5129 index = pipeReader->negotiate(offers, 1, NULL, numCounterOffers); 5130 ALOG_ASSERT(index == 0); 5131 mPipeSource = pipeReader; 5132 mPipeFramesP2 = pipeFramesP2; 5133 mPipeMemory = pipeMemory; 5134 5135 // create fast capture 5136 mFastCapture = new FastCapture(); 5137 FastCaptureStateQueue *sq = mFastCapture->sq(); 5138#ifdef STATE_QUEUE_DUMP 5139 // FIXME 5140#endif 5141 FastCaptureState *state = sq->begin(); 5142 state->mCblk = NULL; 5143 state->mInputSource = mInputSource.get(); 5144 state->mInputSourceGen++; 5145 state->mPipeSink = pipe; 5146 state->mPipeSinkGen++; 5147 state->mFrameCount = mFrameCount; 5148 state->mCommand = FastCaptureState::COLD_IDLE; 5149 // already done in constructor initialization list 5150 //mFastCaptureFutex = 0; 5151 state->mColdFutexAddr = &mFastCaptureFutex; 5152 state->mColdGen++; 5153 state->mDumpState = &mFastCaptureDumpState; 5154#ifdef TEE_SINK 5155 // FIXME 5156#endif 5157 mFastCaptureNBLogWriter = audioFlinger->newWriter_l(kFastCaptureLogSize, "FastCapture"); 5158 state->mNBLogWriter = mFastCaptureNBLogWriter.get(); 5159 sq->end(); 5160 sq->push(FastCaptureStateQueue::BLOCK_UNTIL_PUSHED); 5161 5162 // start the fast capture 5163 mFastCapture->run("FastCapture", ANDROID_PRIORITY_URGENT_AUDIO); 5164 pid_t tid = mFastCapture->getTid(); 5165 int err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 5166 if (err != 0) { 5167 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 5168 kPriorityFastCapture, getpid_cached, tid, err); 5169 } 5170 5171#ifdef AUDIO_WATCHDOG 5172 // FIXME 5173#endif 5174 5175 mFastTrackAvail = true; 5176 } 5177failed: ; 5178 5179 // FIXME mNormalSource 5180} 5181 5182 5183AudioFlinger::RecordThread::~RecordThread() 5184{ 5185 if (mFastCapture != 0) { 5186 FastCaptureStateQueue *sq = mFastCapture->sq(); 5187 FastCaptureState *state = sq->begin(); 5188 if (state->mCommand == FastCaptureState::COLD_IDLE) { 5189 int32_t old = android_atomic_inc(&mFastCaptureFutex); 5190 if (old == -1) { 5191 (void) syscall(__NR_futex, &mFastCaptureFutex, FUTEX_WAKE_PRIVATE, 1); 5192 } 5193 } 5194 state->mCommand = FastCaptureState::EXIT; 5195 sq->end(); 5196 sq->push(FastCaptureStateQueue::BLOCK_UNTIL_PUSHED); 5197 mFastCapture->join(); 5198 mFastCapture.clear(); 5199 } 5200 mAudioFlinger->unregisterWriter(mFastCaptureNBLogWriter); 5201 mAudioFlinger->unregisterWriter(mNBLogWriter); 5202 delete[] mRsmpInBuffer; 5203} 5204 5205void AudioFlinger::RecordThread::onFirstRef() 5206{ 5207 run(mName, PRIORITY_URGENT_AUDIO); 5208} 5209 5210bool AudioFlinger::RecordThread::threadLoop() 5211{ 5212 nsecs_t lastWarning = 0; 5213 5214 inputStandBy(); 5215 5216reacquire_wakelock: 5217 sp<RecordTrack> activeTrack; 5218 int activeTracksGen; 5219 { 5220 Mutex::Autolock _l(mLock); 5221 size_t size = mActiveTracks.size(); 5222 activeTracksGen = mActiveTracksGen; 5223 if (size > 0) { 5224 // FIXME an arbitrary choice 5225 activeTrack = mActiveTracks[0]; 5226 acquireWakeLock_l(activeTrack->uid()); 5227 if (size > 1) { 5228 SortedVector<int> tmp; 5229 for (size_t i = 0; i < size; i++) { 5230 tmp.add(mActiveTracks[i]->uid()); 5231 } 5232 updateWakeLockUids_l(tmp); 5233 } 5234 } else { 5235 acquireWakeLock_l(-1); 5236 } 5237 } 5238 5239 // used to request a deferred sleep, to be executed later while mutex is unlocked 5240 uint32_t sleepUs = 0; 5241 5242 // loop while there is work to do 5243 for (;;) { 5244 Vector< sp<EffectChain> > effectChains; 5245 5246 // sleep with mutex unlocked 5247 if (sleepUs > 0) { 5248 ATRACE_BEGIN("sleep"); 5249 usleep(sleepUs); 5250 ATRACE_END(); 5251 sleepUs = 0; 5252 } 5253 5254 // activeTracks accumulates a copy of a subset of mActiveTracks 5255 Vector< sp<RecordTrack> > activeTracks; 5256 5257 // reference to the (first and only) active fast track 5258 sp<RecordTrack> fastTrack; 5259 5260 // reference to a fast track which is about to be removed 5261 sp<RecordTrack> fastTrackToRemove; 5262 5263 { // scope for mLock 5264 Mutex::Autolock _l(mLock); 5265 5266 processConfigEvents_l(); 5267 5268 // check exitPending here because checkForNewParameters_l() and 5269 // checkForNewParameters_l() can temporarily release mLock 5270 if (exitPending()) { 5271 break; 5272 } 5273 5274 // if no active track(s), then standby and release wakelock 5275 size_t size = mActiveTracks.size(); 5276 if (size == 0) { 5277 standbyIfNotAlreadyInStandby(); 5278 // exitPending() can't become true here 5279 releaseWakeLock_l(); 5280 ALOGV("RecordThread: loop stopping"); 5281 // go to sleep 5282 mWaitWorkCV.wait(mLock); 5283 ALOGV("RecordThread: loop starting"); 5284 goto reacquire_wakelock; 5285 } 5286 5287 if (mActiveTracksGen != activeTracksGen) { 5288 activeTracksGen = mActiveTracksGen; 5289 SortedVector<int> tmp; 5290 for (size_t i = 0; i < size; i++) { 5291 tmp.add(mActiveTracks[i]->uid()); 5292 } 5293 updateWakeLockUids_l(tmp); 5294 } 5295 5296 bool doBroadcast = false; 5297 for (size_t i = 0; i < size; ) { 5298 5299 activeTrack = mActiveTracks[i]; 5300 if (activeTrack->isTerminated()) { 5301 if (activeTrack->isFastTrack()) { 5302 ALOG_ASSERT(fastTrackToRemove == 0); 5303 fastTrackToRemove = activeTrack; 5304 } 5305 removeTrack_l(activeTrack); 5306 mActiveTracks.remove(activeTrack); 5307 mActiveTracksGen++; 5308 size--; 5309 continue; 5310 } 5311 5312 TrackBase::track_state activeTrackState = activeTrack->mState; 5313 switch (activeTrackState) { 5314 5315 case TrackBase::PAUSING: 5316 mActiveTracks.remove(activeTrack); 5317 mActiveTracksGen++; 5318 doBroadcast = true; 5319 size--; 5320 continue; 5321 5322 case TrackBase::STARTING_1: 5323 sleepUs = 10000; 5324 i++; 5325 continue; 5326 5327 case TrackBase::STARTING_2: 5328 doBroadcast = true; 5329 mStandby = false; 5330 activeTrack->mState = TrackBase::ACTIVE; 5331 break; 5332 5333 case TrackBase::ACTIVE: 5334 break; 5335 5336 case TrackBase::IDLE: 5337 i++; 5338 continue; 5339 5340 default: 5341 LOG_ALWAYS_FATAL("Unexpected activeTrackState %d", activeTrackState); 5342 } 5343 5344 activeTracks.add(activeTrack); 5345 i++; 5346 5347 if (activeTrack->isFastTrack()) { 5348 ALOG_ASSERT(!mFastTrackAvail); 5349 ALOG_ASSERT(fastTrack == 0); 5350 fastTrack = activeTrack; 5351 } 5352 } 5353 if (doBroadcast) { 5354 mStartStopCond.broadcast(); 5355 } 5356 5357 // sleep if there are no active tracks to process 5358 if (activeTracks.size() == 0) { 5359 if (sleepUs == 0) { 5360 sleepUs = kRecordThreadSleepUs; 5361 } 5362 continue; 5363 } 5364 sleepUs = 0; 5365 5366 lockEffectChains_l(effectChains); 5367 } 5368 5369 // thread mutex is now unlocked, mActiveTracks unknown, activeTracks.size() > 0 5370 5371 size_t size = effectChains.size(); 5372 for (size_t i = 0; i < size; i++) { 5373 // thread mutex is not locked, but effect chain is locked 5374 effectChains[i]->process_l(); 5375 } 5376 5377 // Push a new fast capture state if fast capture is not already running, or cblk change 5378 if (mFastCapture != 0) { 5379 FastCaptureStateQueue *sq = mFastCapture->sq(); 5380 FastCaptureState *state = sq->begin(); 5381 bool didModify = false; 5382 FastCaptureStateQueue::block_t block = FastCaptureStateQueue::BLOCK_UNTIL_PUSHED; 5383 if (state->mCommand != FastCaptureState::READ_WRITE /* FIXME && 5384 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)*/) { 5385 if (state->mCommand == FastCaptureState::COLD_IDLE) { 5386 int32_t old = android_atomic_inc(&mFastCaptureFutex); 5387 if (old == -1) { 5388 (void) syscall(__NR_futex, &mFastCaptureFutex, FUTEX_WAKE_PRIVATE, 1); 5389 } 5390 } 5391 state->mCommand = FastCaptureState::READ_WRITE; 5392#if 0 // FIXME 5393 mFastCaptureDumpState.increaseSamplingN(mAudioFlinger->isLowRamDevice() ? 5394 FastCaptureDumpState::kSamplingNforLowRamDevice : 5395 FastMixerDumpState::kSamplingN); 5396#endif 5397 didModify = true; 5398 } 5399 audio_track_cblk_t *cblkOld = state->mCblk; 5400 audio_track_cblk_t *cblkNew = fastTrack != 0 ? fastTrack->cblk() : NULL; 5401 if (cblkNew != cblkOld) { 5402 state->mCblk = cblkNew; 5403 // block until acked if removing a fast track 5404 if (cblkOld != NULL) { 5405 block = FastCaptureStateQueue::BLOCK_UNTIL_ACKED; 5406 } 5407 didModify = true; 5408 } 5409 sq->end(didModify); 5410 if (didModify) { 5411 sq->push(block); 5412#if 0 5413 if (kUseFastCapture == FastCapture_Dynamic) { 5414 mNormalSource = mPipeSource; 5415 } 5416#endif 5417 } 5418 } 5419 5420 // now run the fast track destructor with thread mutex unlocked 5421 fastTrackToRemove.clear(); 5422 5423 // Read from HAL to keep up with fastest client if multiple active tracks, not slowest one. 5424 // Only the client(s) that are too slow will overrun. But if even the fastest client is too 5425 // slow, then this RecordThread will overrun by not calling HAL read often enough. 5426 // If destination is non-contiguous, first read past the nominal end of buffer, then 5427 // copy to the right place. Permitted because mRsmpInBuffer was over-allocated. 5428 5429 int32_t rear = mRsmpInRear & (mRsmpInFramesP2 - 1); 5430 ssize_t framesRead; 5431 5432 // If an NBAIO source is present, use it to read the normal capture's data 5433 if (mPipeSource != 0) { 5434 size_t framesToRead = mBufferSize / mFrameSize; 5435 framesRead = mPipeSource->read(&mRsmpInBuffer[rear * mChannelCount], 5436 framesToRead, AudioBufferProvider::kInvalidPTS); 5437 if (framesRead == 0) { 5438 // since pipe is non-blocking, simulate blocking input 5439 sleepUs = (framesToRead * 1000000LL) / mSampleRate; 5440 } 5441 // otherwise use the HAL / AudioStreamIn directly 5442 } else { 5443 ssize_t bytesRead = mInput->stream->read(mInput->stream, 5444 &mRsmpInBuffer[rear * mChannelCount], mBufferSize); 5445 if (bytesRead < 0) { 5446 framesRead = bytesRead; 5447 } else { 5448 framesRead = bytesRead / mFrameSize; 5449 } 5450 } 5451 5452 if (framesRead < 0 || (framesRead == 0 && mPipeSource == 0)) { 5453 ALOGE("read failed: framesRead=%d", framesRead); 5454 // Force input into standby so that it tries to recover at next read attempt 5455 inputStandBy(); 5456 sleepUs = kRecordThreadSleepUs; 5457 } 5458 if (framesRead <= 0) { 5459 goto unlock; 5460 } 5461 ALOG_ASSERT(framesRead > 0); 5462 5463 if (mTeeSink != 0) { 5464 (void) mTeeSink->write(&mRsmpInBuffer[rear * mChannelCount], framesRead); 5465 } 5466 // If destination is non-contiguous, we now correct for reading past end of buffer. 5467 { 5468 size_t part1 = mRsmpInFramesP2 - rear; 5469 if ((size_t) framesRead > part1) { 5470 memcpy(mRsmpInBuffer, &mRsmpInBuffer[mRsmpInFramesP2 * mChannelCount], 5471 (framesRead - part1) * mFrameSize); 5472 } 5473 } 5474 rear = mRsmpInRear += framesRead; 5475 5476 size = activeTracks.size(); 5477 // loop over each active track 5478 for (size_t i = 0; i < size; i++) { 5479 activeTrack = activeTracks[i]; 5480 5481 // skip fast tracks, as those are handled directly by FastCapture 5482 if (activeTrack->isFastTrack()) { 5483 continue; 5484 } 5485 5486 enum { 5487 OVERRUN_UNKNOWN, 5488 OVERRUN_TRUE, 5489 OVERRUN_FALSE 5490 } overrun = OVERRUN_UNKNOWN; 5491 5492 // loop over getNextBuffer to handle circular sink 5493 for (;;) { 5494 5495 activeTrack->mSink.frameCount = ~0; 5496 status_t status = activeTrack->getNextBuffer(&activeTrack->mSink); 5497 size_t framesOut = activeTrack->mSink.frameCount; 5498 LOG_ALWAYS_FATAL_IF((status == OK) != (framesOut > 0)); 5499 5500 int32_t front = activeTrack->mRsmpInFront; 5501 ssize_t filled = rear - front; 5502 size_t framesIn; 5503 5504 if (filled < 0) { 5505 // should not happen, but treat like a massive overrun and re-sync 5506 framesIn = 0; 5507 activeTrack->mRsmpInFront = rear; 5508 overrun = OVERRUN_TRUE; 5509 } else if ((size_t) filled <= mRsmpInFrames) { 5510 framesIn = (size_t) filled; 5511 } else { 5512 // client is not keeping up with server, but give it latest data 5513 framesIn = mRsmpInFrames; 5514 activeTrack->mRsmpInFront = front = rear - framesIn; 5515 overrun = OVERRUN_TRUE; 5516 } 5517 5518 if (framesOut == 0 || framesIn == 0) { 5519 break; 5520 } 5521 5522 if (activeTrack->mResampler == NULL) { 5523 // no resampling 5524 if (framesIn > framesOut) { 5525 framesIn = framesOut; 5526 } else { 5527 framesOut = framesIn; 5528 } 5529 int8_t *dst = activeTrack->mSink.i8; 5530 while (framesIn > 0) { 5531 front &= mRsmpInFramesP2 - 1; 5532 size_t part1 = mRsmpInFramesP2 - front; 5533 if (part1 > framesIn) { 5534 part1 = framesIn; 5535 } 5536 int8_t *src = (int8_t *)mRsmpInBuffer + (front * mFrameSize); 5537 if (mChannelCount == activeTrack->mChannelCount) { 5538 memcpy(dst, src, part1 * mFrameSize); 5539 } else if (mChannelCount == 1) { 5540 upmix_to_stereo_i16_from_mono_i16((int16_t *)dst, (const int16_t *)src, 5541 part1); 5542 } else { 5543 downmix_to_mono_i16_from_stereo_i16((int16_t *)dst, 5544 (const int16_t *)src, part1); 5545 } 5546 dst += part1 * activeTrack->mFrameSize; 5547 front += part1; 5548 framesIn -= part1; 5549 } 5550 activeTrack->mRsmpInFront += framesOut; 5551 5552 } else { 5553 // resampling 5554 // FIXME framesInNeeded should really be part of resampler API, and should 5555 // depend on the SRC ratio 5556 // to keep mRsmpInBuffer full so resampler always has sufficient input 5557 size_t framesInNeeded; 5558 // FIXME only re-calculate when it changes, and optimize for common ratios 5559 // Do not precompute in/out because floating point is not associative 5560 // e.g. a*b/c != a*(b/c). 5561 const double in(mSampleRate); 5562 const double out(activeTrack->mSampleRate); 5563 framesInNeeded = ceil(framesOut * in / out) + 1; 5564 ALOGV("need %u frames in to produce %u out given in/out ratio of %.4g", 5565 framesInNeeded, framesOut, in / out); 5566 // Although we theoretically have framesIn in circular buffer, some of those are 5567 // unreleased frames, and thus must be discounted for purpose of budgeting. 5568 size_t unreleased = activeTrack->mRsmpInUnrel; 5569 framesIn = framesIn > unreleased ? framesIn - unreleased : 0; 5570 if (framesIn < framesInNeeded) { 5571 ALOGV("not enough to resample: have %u frames in but need %u in to " 5572 "produce %u out given in/out ratio of %.4g", 5573 framesIn, framesInNeeded, framesOut, in / out); 5574 size_t newFramesOut = framesIn > 0 ? floor((framesIn - 1) * out / in) : 0; 5575 LOG_ALWAYS_FATAL_IF(newFramesOut >= framesOut); 5576 if (newFramesOut == 0) { 5577 break; 5578 } 5579 framesInNeeded = ceil(newFramesOut * in / out) + 1; 5580 ALOGV("now need %u frames in to produce %u out given out/in ratio of %.4g", 5581 framesInNeeded, newFramesOut, out / in); 5582 LOG_ALWAYS_FATAL_IF(framesIn < framesInNeeded); 5583 ALOGV("success 2: have %u frames in and need %u in to produce %u out " 5584 "given in/out ratio of %.4g", 5585 framesIn, framesInNeeded, newFramesOut, in / out); 5586 framesOut = newFramesOut; 5587 } else { 5588 ALOGV("success 1: have %u in and need %u in to produce %u out " 5589 "given in/out ratio of %.4g", 5590 framesIn, framesInNeeded, framesOut, in / out); 5591 } 5592 5593 // reallocate mRsmpOutBuffer as needed; we will grow but never shrink 5594 if (activeTrack->mRsmpOutFrameCount < framesOut) { 5595 // FIXME why does each track need it's own mRsmpOutBuffer? can't they share? 5596 delete[] activeTrack->mRsmpOutBuffer; 5597 // resampler always outputs stereo 5598 activeTrack->mRsmpOutBuffer = new int32_t[framesOut * FCC_2]; 5599 activeTrack->mRsmpOutFrameCount = framesOut; 5600 } 5601 5602 // resampler accumulates, but we only have one source track 5603 memset(activeTrack->mRsmpOutBuffer, 0, framesOut * FCC_2 * sizeof(int32_t)); 5604 activeTrack->mResampler->resample(activeTrack->mRsmpOutBuffer, framesOut, 5605 // FIXME how about having activeTrack implement this interface itself? 5606 activeTrack->mResamplerBufferProvider 5607 /*this*/ /* AudioBufferProvider* */); 5608 // ditherAndClamp() works as long as all buffers returned by 5609 // activeTrack->getNextBuffer() are 32 bit aligned which should be always true. 5610 if (activeTrack->mChannelCount == 1) { 5611 // temporarily type pun mRsmpOutBuffer from Q4.27 to int16_t 5612 ditherAndClamp(activeTrack->mRsmpOutBuffer, activeTrack->mRsmpOutBuffer, 5613 framesOut); 5614 // the resampler always outputs stereo samples: 5615 // do post stereo to mono conversion 5616 downmix_to_mono_i16_from_stereo_i16(activeTrack->mSink.i16, 5617 (const int16_t *)activeTrack->mRsmpOutBuffer, framesOut); 5618 } else { 5619 ditherAndClamp((int32_t *)activeTrack->mSink.raw, 5620 activeTrack->mRsmpOutBuffer, framesOut); 5621 } 5622 // now done with mRsmpOutBuffer 5623 5624 } 5625 5626 if (framesOut > 0 && (overrun == OVERRUN_UNKNOWN)) { 5627 overrun = OVERRUN_FALSE; 5628 } 5629 5630 if (activeTrack->mFramesToDrop == 0) { 5631 if (framesOut > 0) { 5632 activeTrack->mSink.frameCount = framesOut; 5633 activeTrack->releaseBuffer(&activeTrack->mSink); 5634 } 5635 } else { 5636 // FIXME could do a partial drop of framesOut 5637 if (activeTrack->mFramesToDrop > 0) { 5638 activeTrack->mFramesToDrop -= framesOut; 5639 if (activeTrack->mFramesToDrop <= 0) { 5640 activeTrack->clearSyncStartEvent(); 5641 } 5642 } else { 5643 activeTrack->mFramesToDrop += framesOut; 5644 if (activeTrack->mFramesToDrop >= 0 || activeTrack->mSyncStartEvent == 0 || 5645 activeTrack->mSyncStartEvent->isCancelled()) { 5646 ALOGW("Synced record %s, session %d, trigger session %d", 5647 (activeTrack->mFramesToDrop >= 0) ? "timed out" : "cancelled", 5648 activeTrack->sessionId(), 5649 (activeTrack->mSyncStartEvent != 0) ? 5650 activeTrack->mSyncStartEvent->triggerSession() : 0); 5651 activeTrack->clearSyncStartEvent(); 5652 } 5653 } 5654 } 5655 5656 if (framesOut == 0) { 5657 break; 5658 } 5659 } 5660 5661 switch (overrun) { 5662 case OVERRUN_TRUE: 5663 // client isn't retrieving buffers fast enough 5664 if (!activeTrack->setOverflow()) { 5665 nsecs_t now = systemTime(); 5666 // FIXME should lastWarning per track? 5667 if ((now - lastWarning) > kWarningThrottleNs) { 5668 ALOGW("RecordThread: buffer overflow"); 5669 lastWarning = now; 5670 } 5671 } 5672 break; 5673 case OVERRUN_FALSE: 5674 activeTrack->clearOverflow(); 5675 break; 5676 case OVERRUN_UNKNOWN: 5677 break; 5678 } 5679 5680 } 5681 5682unlock: 5683 // enable changes in effect chain 5684 unlockEffectChains(effectChains); 5685 // effectChains doesn't need to be cleared, since it is cleared by destructor at scope end 5686 } 5687 5688 standbyIfNotAlreadyInStandby(); 5689 5690 { 5691 Mutex::Autolock _l(mLock); 5692 for (size_t i = 0; i < mTracks.size(); i++) { 5693 sp<RecordTrack> track = mTracks[i]; 5694 track->invalidate(); 5695 } 5696 mActiveTracks.clear(); 5697 mActiveTracksGen++; 5698 mStartStopCond.broadcast(); 5699 } 5700 5701 releaseWakeLock(); 5702 5703 ALOGV("RecordThread %p exiting", this); 5704 return false; 5705} 5706 5707void AudioFlinger::RecordThread::standbyIfNotAlreadyInStandby() 5708{ 5709 if (!mStandby) { 5710 inputStandBy(); 5711 mStandby = true; 5712 } 5713} 5714 5715void AudioFlinger::RecordThread::inputStandBy() 5716{ 5717 // Idle the fast capture if it's currently running 5718 if (mFastCapture != 0) { 5719 FastCaptureStateQueue *sq = mFastCapture->sq(); 5720 FastCaptureState *state = sq->begin(); 5721 if (!(state->mCommand & FastCaptureState::IDLE)) { 5722 state->mCommand = FastCaptureState::COLD_IDLE; 5723 state->mColdFutexAddr = &mFastCaptureFutex; 5724 state->mColdGen++; 5725 mFastCaptureFutex = 0; 5726 sq->end(); 5727 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 5728 sq->push(FastCaptureStateQueue::BLOCK_UNTIL_ACKED); 5729#if 0 5730 if (kUseFastCapture == FastCapture_Dynamic) { 5731 // FIXME 5732 } 5733#endif 5734#ifdef AUDIO_WATCHDOG 5735 // FIXME 5736#endif 5737 } else { 5738 sq->end(false /*didModify*/); 5739 } 5740 } 5741 mInput->stream->common.standby(&mInput->stream->common); 5742} 5743 5744// RecordThread::createRecordTrack_l() must be called with AudioFlinger::mLock held 5745sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 5746 const sp<AudioFlinger::Client>& client, 5747 uint32_t sampleRate, 5748 audio_format_t format, 5749 audio_channel_mask_t channelMask, 5750 size_t *pFrameCount, 5751 int sessionId, 5752 size_t *notificationFrames, 5753 int uid, 5754 IAudioFlinger::track_flags_t *flags, 5755 pid_t tid, 5756 status_t *status) 5757{ 5758 size_t frameCount = *pFrameCount; 5759 sp<RecordTrack> track; 5760 status_t lStatus; 5761 5762 // client expresses a preference for FAST, but we get the final say 5763 if (*flags & IAudioFlinger::TRACK_FAST) { 5764 if ( 5765 // use case: callback handler 5766 (tid != -1) && 5767 // frame count is not specified, or is exactly the pipe depth 5768 ((frameCount == 0) || (frameCount == mPipeFramesP2)) && 5769 // PCM data 5770 audio_is_linear_pcm(format) && 5771 // native format 5772 (format == mFormat) && 5773 // native channel mask 5774 (channelMask == mChannelMask) && 5775 // native hardware sample rate 5776 (sampleRate == mSampleRate) && 5777 // record thread has an associated fast capture 5778 hasFastCapture() && 5779 // there are sufficient fast track slots available 5780 mFastTrackAvail 5781 ) { 5782 ALOGV("AUDIO_INPUT_FLAG_FAST accepted: frameCount=%u mFrameCount=%u", 5783 frameCount, mFrameCount); 5784 } else { 5785 ALOGV("AUDIO_INPUT_FLAG_FAST denied: frameCount=%u mFrameCount=%u mPipeFramesP2=%u " 5786 "format=%#x isLinear=%d channelMask=%#x sampleRate=%u mSampleRate=%u " 5787 "hasFastCapture=%d tid=%d mFastTrackAvail=%d", 5788 frameCount, mFrameCount, mPipeFramesP2, 5789 format, audio_is_linear_pcm(format), channelMask, sampleRate, mSampleRate, 5790 hasFastCapture(), tid, mFastTrackAvail); 5791 *flags &= ~IAudioFlinger::TRACK_FAST; 5792 } 5793 } 5794 5795 // compute track buffer size in frames, and suggest the notification frame count 5796 if (*flags & IAudioFlinger::TRACK_FAST) { 5797 // fast track: frame count is exactly the pipe depth 5798 frameCount = mPipeFramesP2; 5799 // ignore requested notificationFrames, and always notify exactly once every HAL buffer 5800 *notificationFrames = mFrameCount; 5801 } else { 5802 // not fast track: max notification period is resampled equivalent of one HAL buffer time 5803 // or 20 ms if there is a fast capture 5804 // TODO This could be a roundupRatio inline, and const 5805 size_t maxNotificationFrames = ((int64_t) (hasFastCapture() ? mSampleRate/50 : mFrameCount) 5806 * sampleRate + mSampleRate - 1) / mSampleRate; 5807 // minimum number of notification periods is at least kMinNotifications, 5808 // and at least kMinMs rounded up to a whole notification period (minNotificationsByMs) 5809 static const size_t kMinNotifications = 3; 5810 static const uint32_t kMinMs = 30; 5811 // TODO This could be a roundupRatio inline 5812 const size_t minFramesByMs = (sampleRate * kMinMs + 1000 - 1) / 1000; 5813 // TODO This could be a roundupRatio inline 5814 const size_t minNotificationsByMs = (minFramesByMs + maxNotificationFrames - 1) / 5815 maxNotificationFrames; 5816 const size_t minFrameCount = maxNotificationFrames * 5817 max(kMinNotifications, minNotificationsByMs); 5818 frameCount = max(frameCount, minFrameCount); 5819 if (*notificationFrames == 0 || *notificationFrames > maxNotificationFrames) { 5820 *notificationFrames = maxNotificationFrames; 5821 } 5822 } 5823 *pFrameCount = frameCount; 5824 5825 lStatus = initCheck(); 5826 if (lStatus != NO_ERROR) { 5827 ALOGE("createRecordTrack_l() audio driver not initialized"); 5828 goto Exit; 5829 } 5830 5831 { // scope for mLock 5832 Mutex::Autolock _l(mLock); 5833 5834 track = new RecordTrack(this, client, sampleRate, 5835 format, channelMask, frameCount, NULL, sessionId, uid, 5836 *flags, TrackBase::TYPE_DEFAULT); 5837 5838 lStatus = track->initCheck(); 5839 if (lStatus != NO_ERROR) { 5840 ALOGE("createRecordTrack_l() initCheck failed %d; no control block?", lStatus); 5841 // track must be cleared from the caller as the caller has the AF lock 5842 goto Exit; 5843 } 5844 mTracks.add(track); 5845 5846 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 5847 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 5848 mAudioFlinger->btNrecIsOff(); 5849 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 5850 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 5851 5852 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 5853 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 5854 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 5855 // so ask activity manager to do this on our behalf 5856 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 5857 } 5858 } 5859 5860 lStatus = NO_ERROR; 5861 5862Exit: 5863 *status = lStatus; 5864 return track; 5865} 5866 5867status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, 5868 AudioSystem::sync_event_t event, 5869 int triggerSession) 5870{ 5871 ALOGV("RecordThread::start event %d, triggerSession %d", event, triggerSession); 5872 sp<ThreadBase> strongMe = this; 5873 status_t status = NO_ERROR; 5874 5875 if (event == AudioSystem::SYNC_EVENT_NONE) { 5876 recordTrack->clearSyncStartEvent(); 5877 } else if (event != AudioSystem::SYNC_EVENT_SAME) { 5878 recordTrack->mSyncStartEvent = mAudioFlinger->createSyncEvent(event, 5879 triggerSession, 5880 recordTrack->sessionId(), 5881 syncStartEventCallback, 5882 recordTrack); 5883 // Sync event can be cancelled by the trigger session if the track is not in a 5884 // compatible state in which case we start record immediately 5885 if (recordTrack->mSyncStartEvent->isCancelled()) { 5886 recordTrack->clearSyncStartEvent(); 5887 } else { 5888 // do not wait for the event for more than AudioSystem::kSyncRecordStartTimeOutMs 5889 recordTrack->mFramesToDrop = - 5890 ((AudioSystem::kSyncRecordStartTimeOutMs * recordTrack->mSampleRate) / 1000); 5891 } 5892 } 5893 5894 { 5895 // This section is a rendezvous between binder thread executing start() and RecordThread 5896 AutoMutex lock(mLock); 5897 if (mActiveTracks.indexOf(recordTrack) >= 0) { 5898 if (recordTrack->mState == TrackBase::PAUSING) { 5899 ALOGV("active record track PAUSING -> ACTIVE"); 5900 recordTrack->mState = TrackBase::ACTIVE; 5901 } else { 5902 ALOGV("active record track state %d", recordTrack->mState); 5903 } 5904 return status; 5905 } 5906 5907 // TODO consider other ways of handling this, such as changing the state to :STARTING and 5908 // adding the track to mActiveTracks after returning from AudioSystem::startInput(), 5909 // or using a separate command thread 5910 recordTrack->mState = TrackBase::STARTING_1; 5911 mActiveTracks.add(recordTrack); 5912 mActiveTracksGen++; 5913 status_t status = NO_ERROR; 5914 if (recordTrack->isExternalTrack()) { 5915 mLock.unlock(); 5916 status = AudioSystem::startInput(mId, (audio_session_t)recordTrack->sessionId()); 5917 mLock.lock(); 5918 // FIXME should verify that recordTrack is still in mActiveTracks 5919 if (status != NO_ERROR) { 5920 mActiveTracks.remove(recordTrack); 5921 mActiveTracksGen++; 5922 recordTrack->clearSyncStartEvent(); 5923 ALOGV("RecordThread::start error %d", status); 5924 return status; 5925 } 5926 } 5927 // Catch up with current buffer indices if thread is already running. 5928 // This is what makes a new client discard all buffered data. If the track's mRsmpInFront 5929 // was initialized to some value closer to the thread's mRsmpInFront, then the track could 5930 // see previously buffered data before it called start(), but with greater risk of overrun. 5931 5932 recordTrack->mRsmpInFront = mRsmpInRear; 5933 recordTrack->mRsmpInUnrel = 0; 5934 // FIXME why reset? 5935 if (recordTrack->mResampler != NULL) { 5936 recordTrack->mResampler->reset(); 5937 } 5938 recordTrack->mState = TrackBase::STARTING_2; 5939 // signal thread to start 5940 mWaitWorkCV.broadcast(); 5941 if (mActiveTracks.indexOf(recordTrack) < 0) { 5942 ALOGV("Record failed to start"); 5943 status = BAD_VALUE; 5944 goto startError; 5945 } 5946 return status; 5947 } 5948 5949startError: 5950 if (recordTrack->isExternalTrack()) { 5951 AudioSystem::stopInput(mId, (audio_session_t)recordTrack->sessionId()); 5952 } 5953 recordTrack->clearSyncStartEvent(); 5954 // FIXME I wonder why we do not reset the state here? 5955 return status; 5956} 5957 5958void AudioFlinger::RecordThread::syncStartEventCallback(const wp<SyncEvent>& event) 5959{ 5960 sp<SyncEvent> strongEvent = event.promote(); 5961 5962 if (strongEvent != 0) { 5963 sp<RefBase> ptr = strongEvent->cookie().promote(); 5964 if (ptr != 0) { 5965 RecordTrack *recordTrack = (RecordTrack *)ptr.get(); 5966 recordTrack->handleSyncStartEvent(strongEvent); 5967 } 5968 } 5969} 5970 5971bool AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 5972 ALOGV("RecordThread::stop"); 5973 AutoMutex _l(mLock); 5974 if (mActiveTracks.indexOf(recordTrack) != 0 || recordTrack->mState == TrackBase::PAUSING) { 5975 return false; 5976 } 5977 // note that threadLoop may still be processing the track at this point [without lock] 5978 recordTrack->mState = TrackBase::PAUSING; 5979 // do not wait for mStartStopCond if exiting 5980 if (exitPending()) { 5981 return true; 5982 } 5983 // FIXME incorrect usage of wait: no explicit predicate or loop 5984 mStartStopCond.wait(mLock); 5985 // if we have been restarted, recordTrack is in mActiveTracks here 5986 if (exitPending() || mActiveTracks.indexOf(recordTrack) != 0) { 5987 ALOGV("Record stopped OK"); 5988 return true; 5989 } 5990 return false; 5991} 5992 5993bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event __unused) const 5994{ 5995 return false; 5996} 5997 5998status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event __unused) 5999{ 6000#if 0 // This branch is currently dead code, but is preserved in case it will be needed in future 6001 if (!isValidSyncEvent(event)) { 6002 return BAD_VALUE; 6003 } 6004 6005 int eventSession = event->triggerSession(); 6006 status_t ret = NAME_NOT_FOUND; 6007 6008 Mutex::Autolock _l(mLock); 6009 6010 for (size_t i = 0; i < mTracks.size(); i++) { 6011 sp<RecordTrack> track = mTracks[i]; 6012 if (eventSession == track->sessionId()) { 6013 (void) track->setSyncEvent(event); 6014 ret = NO_ERROR; 6015 } 6016 } 6017 return ret; 6018#else 6019 return BAD_VALUE; 6020#endif 6021} 6022 6023// destroyTrack_l() must be called with ThreadBase::mLock held 6024void AudioFlinger::RecordThread::destroyTrack_l(const sp<RecordTrack>& track) 6025{ 6026 track->terminate(); 6027 track->mState = TrackBase::STOPPED; 6028 // active tracks are removed by threadLoop() 6029 if (mActiveTracks.indexOf(track) < 0) { 6030 removeTrack_l(track); 6031 } 6032} 6033 6034void AudioFlinger::RecordThread::removeTrack_l(const sp<RecordTrack>& track) 6035{ 6036 mTracks.remove(track); 6037 // need anything related to effects here? 6038 if (track->isFastTrack()) { 6039 ALOG_ASSERT(!mFastTrackAvail); 6040 mFastTrackAvail = true; 6041 } 6042} 6043 6044void AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 6045{ 6046 dumpInternals(fd, args); 6047 dumpTracks(fd, args); 6048 dumpEffectChains(fd, args); 6049} 6050 6051void AudioFlinger::RecordThread::dumpInternals(int fd, const Vector<String16>& args) 6052{ 6053 dprintf(fd, "\nInput thread %p:\n", this); 6054 6055 if (mActiveTracks.size() > 0) { 6056 dprintf(fd, " Buffer size: %zu bytes\n", mBufferSize); 6057 } else { 6058 dprintf(fd, " No active record clients\n"); 6059 } 6060 dprintf(fd, " Fast capture thread: %s\n", hasFastCapture() ? "yes" : "no"); 6061 dprintf(fd, " Fast track available: %s\n", mFastTrackAvail ? "yes" : "no"); 6062 6063 dumpBase(fd, args); 6064} 6065 6066void AudioFlinger::RecordThread::dumpTracks(int fd, const Vector<String16>& args __unused) 6067{ 6068 const size_t SIZE = 256; 6069 char buffer[SIZE]; 6070 String8 result; 6071 6072 size_t numtracks = mTracks.size(); 6073 size_t numactive = mActiveTracks.size(); 6074 size_t numactiveseen = 0; 6075 dprintf(fd, " %d Tracks", numtracks); 6076 if (numtracks) { 6077 dprintf(fd, " of which %d are active\n", numactive); 6078 RecordTrack::appendDumpHeader(result); 6079 for (size_t i = 0; i < numtracks ; ++i) { 6080 sp<RecordTrack> track = mTracks[i]; 6081 if (track != 0) { 6082 bool active = mActiveTracks.indexOf(track) >= 0; 6083 if (active) { 6084 numactiveseen++; 6085 } 6086 track->dump(buffer, SIZE, active); 6087 result.append(buffer); 6088 } 6089 } 6090 } else { 6091 dprintf(fd, "\n"); 6092 } 6093 6094 if (numactiveseen != numactive) { 6095 snprintf(buffer, SIZE, " The following tracks are in the active list but" 6096 " not in the track list\n"); 6097 result.append(buffer); 6098 RecordTrack::appendDumpHeader(result); 6099 for (size_t i = 0; i < numactive; ++i) { 6100 sp<RecordTrack> track = mActiveTracks[i]; 6101 if (mTracks.indexOf(track) < 0) { 6102 track->dump(buffer, SIZE, true); 6103 result.append(buffer); 6104 } 6105 } 6106 6107 } 6108 write(fd, result.string(), result.size()); 6109} 6110 6111// AudioBufferProvider interface 6112status_t AudioFlinger::RecordThread::ResamplerBufferProvider::getNextBuffer( 6113 AudioBufferProvider::Buffer* buffer, int64_t pts __unused) 6114{ 6115 RecordTrack *activeTrack = mRecordTrack; 6116 sp<ThreadBase> threadBase = activeTrack->mThread.promote(); 6117 if (threadBase == 0) { 6118 buffer->frameCount = 0; 6119 buffer->raw = NULL; 6120 return NOT_ENOUGH_DATA; 6121 } 6122 RecordThread *recordThread = (RecordThread *) threadBase.get(); 6123 int32_t rear = recordThread->mRsmpInRear; 6124 int32_t front = activeTrack->mRsmpInFront; 6125 ssize_t filled = rear - front; 6126 // FIXME should not be P2 (don't want to increase latency) 6127 // FIXME if client not keeping up, discard 6128 LOG_ALWAYS_FATAL_IF(!(0 <= filled && (size_t) filled <= recordThread->mRsmpInFrames)); 6129 // 'filled' may be non-contiguous, so return only the first contiguous chunk 6130 front &= recordThread->mRsmpInFramesP2 - 1; 6131 size_t part1 = recordThread->mRsmpInFramesP2 - front; 6132 if (part1 > (size_t) filled) { 6133 part1 = filled; 6134 } 6135 size_t ask = buffer->frameCount; 6136 ALOG_ASSERT(ask > 0); 6137 if (part1 > ask) { 6138 part1 = ask; 6139 } 6140 if (part1 == 0) { 6141 // Higher-level should keep mRsmpInBuffer full, and not call resampler if empty 6142 LOG_ALWAYS_FATAL("RecordThread::getNextBuffer() starved"); 6143 buffer->raw = NULL; 6144 buffer->frameCount = 0; 6145 activeTrack->mRsmpInUnrel = 0; 6146 return NOT_ENOUGH_DATA; 6147 } 6148 6149 buffer->raw = recordThread->mRsmpInBuffer + front * recordThread->mChannelCount; 6150 buffer->frameCount = part1; 6151 activeTrack->mRsmpInUnrel = part1; 6152 return NO_ERROR; 6153} 6154 6155// AudioBufferProvider interface 6156void AudioFlinger::RecordThread::ResamplerBufferProvider::releaseBuffer( 6157 AudioBufferProvider::Buffer* buffer) 6158{ 6159 RecordTrack *activeTrack = mRecordTrack; 6160 size_t stepCount = buffer->frameCount; 6161 if (stepCount == 0) { 6162 return; 6163 } 6164 ALOG_ASSERT(stepCount <= activeTrack->mRsmpInUnrel); 6165 activeTrack->mRsmpInUnrel -= stepCount; 6166 activeTrack->mRsmpInFront += stepCount; 6167 buffer->raw = NULL; 6168 buffer->frameCount = 0; 6169} 6170 6171bool AudioFlinger::RecordThread::checkForNewParameter_l(const String8& keyValuePair, 6172 status_t& status) 6173{ 6174 bool reconfig = false; 6175 6176 status = NO_ERROR; 6177 6178 audio_format_t reqFormat = mFormat; 6179 uint32_t samplingRate = mSampleRate; 6180 audio_channel_mask_t channelMask = audio_channel_in_mask_from_count(mChannelCount); 6181 6182 AudioParameter param = AudioParameter(keyValuePair); 6183 int value; 6184 // TODO Investigate when this code runs. Check with audio policy when a sample rate and 6185 // channel count change can be requested. Do we mandate the first client defines the 6186 // HAL sampling rate and channel count or do we allow changes on the fly? 6187 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 6188 samplingRate = value; 6189 reconfig = true; 6190 } 6191 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 6192 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 6193 status = BAD_VALUE; 6194 } else { 6195 reqFormat = (audio_format_t) value; 6196 reconfig = true; 6197 } 6198 } 6199 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 6200 audio_channel_mask_t mask = (audio_channel_mask_t) value; 6201 if (mask != AUDIO_CHANNEL_IN_MONO && mask != AUDIO_CHANNEL_IN_STEREO) { 6202 status = BAD_VALUE; 6203 } else { 6204 channelMask = mask; 6205 reconfig = true; 6206 } 6207 } 6208 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 6209 // do not accept frame count changes if tracks are open as the track buffer 6210 // size depends on frame count and correct behavior would not be guaranteed 6211 // if frame count is changed after track creation 6212 if (mActiveTracks.size() > 0) { 6213 status = INVALID_OPERATION; 6214 } else { 6215 reconfig = true; 6216 } 6217 } 6218 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 6219 // forward device change to effects that have requested to be 6220 // aware of attached audio device. 6221 for (size_t i = 0; i < mEffectChains.size(); i++) { 6222 mEffectChains[i]->setDevice_l(value); 6223 } 6224 6225 // store input device and output device but do not forward output device to audio HAL. 6226 // Note that status is ignored by the caller for output device 6227 // (see AudioFlinger::setParameters() 6228 if (audio_is_output_devices(value)) { 6229 mOutDevice = value; 6230 status = BAD_VALUE; 6231 } else { 6232 mInDevice = value; 6233 // disable AEC and NS if the device is a BT SCO headset supporting those 6234 // pre processings 6235 if (mTracks.size() > 0) { 6236 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 6237 mAudioFlinger->btNrecIsOff(); 6238 for (size_t i = 0; i < mTracks.size(); i++) { 6239 sp<RecordTrack> track = mTracks[i]; 6240 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId()); 6241 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId()); 6242 } 6243 } 6244 } 6245 } 6246 if (param.getInt(String8(AudioParameter::keyInputSource), value) == NO_ERROR && 6247 mAudioSource != (audio_source_t)value) { 6248 // forward device change to effects that have requested to be 6249 // aware of attached audio device. 6250 for (size_t i = 0; i < mEffectChains.size(); i++) { 6251 mEffectChains[i]->setAudioSource_l((audio_source_t)value); 6252 } 6253 mAudioSource = (audio_source_t)value; 6254 } 6255 6256 if (status == NO_ERROR) { 6257 status = mInput->stream->common.set_parameters(&mInput->stream->common, 6258 keyValuePair.string()); 6259 if (status == INVALID_OPERATION) { 6260 inputStandBy(); 6261 status = mInput->stream->common.set_parameters(&mInput->stream->common, 6262 keyValuePair.string()); 6263 } 6264 if (reconfig) { 6265 if (status == BAD_VALUE && 6266 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 6267 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 6268 (mInput->stream->common.get_sample_rate(&mInput->stream->common) 6269 <= (2 * samplingRate)) && 6270 audio_channel_count_from_in_mask( 6271 mInput->stream->common.get_channels(&mInput->stream->common)) <= FCC_2 && 6272 (channelMask == AUDIO_CHANNEL_IN_MONO || 6273 channelMask == AUDIO_CHANNEL_IN_STEREO)) { 6274 status = NO_ERROR; 6275 } 6276 if (status == NO_ERROR) { 6277 readInputParameters_l(); 6278 sendIoConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 6279 } 6280 } 6281 } 6282 6283 return reconfig; 6284} 6285 6286String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 6287{ 6288 Mutex::Autolock _l(mLock); 6289 if (initCheck() != NO_ERROR) { 6290 return String8(); 6291 } 6292 6293 char *s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 6294 const String8 out_s8(s); 6295 free(s); 6296 return out_s8; 6297} 6298 6299void AudioFlinger::RecordThread::audioConfigChanged(int event, int param __unused) { 6300 AudioSystem::OutputDescriptor desc; 6301 const void *param2 = NULL; 6302 6303 switch (event) { 6304 case AudioSystem::INPUT_OPENED: 6305 case AudioSystem::INPUT_CONFIG_CHANGED: 6306 desc.channelMask = mChannelMask; 6307 desc.samplingRate = mSampleRate; 6308 desc.format = mFormat; 6309 desc.frameCount = mFrameCount; 6310 desc.latency = 0; 6311 param2 = &desc; 6312 break; 6313 6314 case AudioSystem::INPUT_CLOSED: 6315 default: 6316 break; 6317 } 6318 mAudioFlinger->audioConfigChanged(event, mId, param2); 6319} 6320 6321void AudioFlinger::RecordThread::readInputParameters_l() 6322{ 6323 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 6324 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 6325 mChannelCount = audio_channel_count_from_in_mask(mChannelMask); 6326 mHALFormat = mInput->stream->common.get_format(&mInput->stream->common); 6327 mFormat = mHALFormat; 6328 if (mFormat != AUDIO_FORMAT_PCM_16_BIT) { 6329 ALOGE("HAL format %#x not supported; must be AUDIO_FORMAT_PCM_16_BIT", mFormat); 6330 } 6331 mFrameSize = audio_stream_in_frame_size(mInput->stream); 6332 mBufferSize = mInput->stream->common.get_buffer_size(&mInput->stream->common); 6333 mFrameCount = mBufferSize / mFrameSize; 6334 // This is the formula for calculating the temporary buffer size. 6335 // With 7 HAL buffers, we can guarantee ability to down-sample the input by ratio of 6:1 to 6336 // 1 full output buffer, regardless of the alignment of the available input. 6337 // The value is somewhat arbitrary, and could probably be even larger. 6338 // A larger value should allow more old data to be read after a track calls start(), 6339 // without increasing latency. 6340 mRsmpInFrames = mFrameCount * 7; 6341 mRsmpInFramesP2 = roundup(mRsmpInFrames); 6342 delete[] mRsmpInBuffer; 6343 6344 // TODO optimize audio capture buffer sizes ... 6345 // Here we calculate the size of the sliding buffer used as a source 6346 // for resampling. mRsmpInFramesP2 is currently roundup(mFrameCount * 7). 6347 // For current HAL frame counts, this is usually 2048 = 40 ms. It would 6348 // be better to have it derived from the pipe depth in the long term. 6349 // The current value is higher than necessary. However it should not add to latency. 6350 6351 // Over-allocate beyond mRsmpInFramesP2 to permit a HAL read past end of buffer 6352 mRsmpInBuffer = new int16_t[(mRsmpInFramesP2 + mFrameCount - 1) * mChannelCount]; 6353 6354 // AudioRecord mSampleRate and mChannelCount are constant due to AudioRecord API constraints. 6355 // But if thread's mSampleRate or mChannelCount changes, how will that affect active tracks? 6356} 6357 6358uint32_t AudioFlinger::RecordThread::getInputFramesLost() 6359{ 6360 Mutex::Autolock _l(mLock); 6361 if (initCheck() != NO_ERROR) { 6362 return 0; 6363 } 6364 6365 return mInput->stream->get_input_frames_lost(mInput->stream); 6366} 6367 6368uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) const 6369{ 6370 Mutex::Autolock _l(mLock); 6371 uint32_t result = 0; 6372 if (getEffectChain_l(sessionId) != 0) { 6373 result = EFFECT_SESSION; 6374 } 6375 6376 for (size_t i = 0; i < mTracks.size(); ++i) { 6377 if (sessionId == mTracks[i]->sessionId()) { 6378 result |= TRACK_SESSION; 6379 break; 6380 } 6381 } 6382 6383 return result; 6384} 6385 6386KeyedVector<int, bool> AudioFlinger::RecordThread::sessionIds() const 6387{ 6388 KeyedVector<int, bool> ids; 6389 Mutex::Autolock _l(mLock); 6390 for (size_t j = 0; j < mTracks.size(); ++j) { 6391 sp<RecordThread::RecordTrack> track = mTracks[j]; 6392 int sessionId = track->sessionId(); 6393 if (ids.indexOfKey(sessionId) < 0) { 6394 ids.add(sessionId, true); 6395 } 6396 } 6397 return ids; 6398} 6399 6400AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 6401{ 6402 Mutex::Autolock _l(mLock); 6403 AudioStreamIn *input = mInput; 6404 mInput = NULL; 6405 return input; 6406} 6407 6408// this method must always be called either with ThreadBase mLock held or inside the thread loop 6409audio_stream_t* AudioFlinger::RecordThread::stream() const 6410{ 6411 if (mInput == NULL) { 6412 return NULL; 6413 } 6414 return &mInput->stream->common; 6415} 6416 6417status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 6418{ 6419 // only one chain per input thread 6420 if (mEffectChains.size() != 0) { 6421 ALOGW("addEffectChain_l() already one chain %p on thread %p", chain.get(), this); 6422 return INVALID_OPERATION; 6423 } 6424 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 6425 chain->setThread(this); 6426 chain->setInBuffer(NULL); 6427 chain->setOutBuffer(NULL); 6428 6429 checkSuspendOnAddEffectChain_l(chain); 6430 6431 // make sure enabled pre processing effects state is communicated to the HAL as we 6432 // just moved them to a new input stream. 6433 chain->syncHalEffectsState(); 6434 6435 mEffectChains.add(chain); 6436 6437 return NO_ERROR; 6438} 6439 6440size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 6441{ 6442 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 6443 ALOGW_IF(mEffectChains.size() != 1, 6444 "removeEffectChain_l() %p invalid chain size %d on thread %p", 6445 chain.get(), mEffectChains.size(), this); 6446 if (mEffectChains.size() == 1) { 6447 mEffectChains.removeAt(0); 6448 } 6449 return 0; 6450} 6451 6452status_t AudioFlinger::RecordThread::createAudioPatch_l(const struct audio_patch *patch, 6453 audio_patch_handle_t *handle) 6454{ 6455 status_t status = NO_ERROR; 6456 if (mInput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 6457 // store new device and send to effects 6458 mInDevice = patch->sources[0].ext.device.type; 6459 for (size_t i = 0; i < mEffectChains.size(); i++) { 6460 mEffectChains[i]->setDevice_l(mInDevice); 6461 } 6462 6463 // disable AEC and NS if the device is a BT SCO headset supporting those 6464 // pre processings 6465 if (mTracks.size() > 0) { 6466 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 6467 mAudioFlinger->btNrecIsOff(); 6468 for (size_t i = 0; i < mTracks.size(); i++) { 6469 sp<RecordTrack> track = mTracks[i]; 6470 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId()); 6471 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId()); 6472 } 6473 } 6474 6475 // store new source and send to effects 6476 if (mAudioSource != patch->sinks[0].ext.mix.usecase.source) { 6477 mAudioSource = patch->sinks[0].ext.mix.usecase.source; 6478 for (size_t i = 0; i < mEffectChains.size(); i++) { 6479 mEffectChains[i]->setAudioSource_l(mAudioSource); 6480 } 6481 } 6482 6483 audio_hw_device_t *hwDevice = mInput->audioHwDev->hwDevice(); 6484 status = hwDevice->create_audio_patch(hwDevice, 6485 patch->num_sources, 6486 patch->sources, 6487 patch->num_sinks, 6488 patch->sinks, 6489 handle); 6490 } else { 6491 ALOG_ASSERT(false, "createAudioPatch_l() called on a pre 3.0 HAL"); 6492 } 6493 return status; 6494} 6495 6496status_t AudioFlinger::RecordThread::releaseAudioPatch_l(const audio_patch_handle_t handle) 6497{ 6498 status_t status = NO_ERROR; 6499 if (mInput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 6500 audio_hw_device_t *hwDevice = mInput->audioHwDev->hwDevice(); 6501 status = hwDevice->release_audio_patch(hwDevice, handle); 6502 } else { 6503 ALOG_ASSERT(false, "releaseAudioPatch_l() called on a pre 3.0 HAL"); 6504 } 6505 return status; 6506} 6507 6508void AudioFlinger::RecordThread::addPatchRecord(const sp<PatchRecord>& record) 6509{ 6510 Mutex::Autolock _l(mLock); 6511 mTracks.add(record); 6512} 6513 6514void AudioFlinger::RecordThread::deletePatchRecord(const sp<PatchRecord>& record) 6515{ 6516 Mutex::Autolock _l(mLock); 6517 destroyTrack_l(record); 6518} 6519 6520void AudioFlinger::RecordThread::getAudioPortConfig(struct audio_port_config *config) 6521{ 6522 ThreadBase::getAudioPortConfig(config); 6523 config->role = AUDIO_PORT_ROLE_SINK; 6524 config->ext.mix.hw_module = mInput->audioHwDev->handle(); 6525 config->ext.mix.usecase.source = mAudioSource; 6526} 6527 6528}; // namespace android 6529