Threads.cpp revision 6770c6faa3467c92eabc5ec9b23d60eb556a0d03
1b9b088375d33a87b201cdbe18be71802e2607717Dan Stoza/* 2b9b088375d33a87b201cdbe18be71802e2607717Dan Stoza** 3b9b088375d33a87b201cdbe18be71802e2607717Dan Stoza** Copyright 2012, The Android Open Source Project 4b9b088375d33a87b201cdbe18be71802e2607717Dan Stoza** 5b9b088375d33a87b201cdbe18be71802e2607717Dan Stoza** Licensed under the Apache License, Version 2.0 (the "License"); 6b9b088375d33a87b201cdbe18be71802e2607717Dan Stoza** you may not use this file except in compliance with the License. 7b9b088375d33a87b201cdbe18be71802e2607717Dan Stoza** You may obtain a copy of the License at 8b9b088375d33a87b201cdbe18be71802e2607717Dan Stoza** 9b9b088375d33a87b201cdbe18be71802e2607717Dan Stoza** http://www.apache.org/licenses/LICENSE-2.0 10b9b088375d33a87b201cdbe18be71802e2607717Dan Stoza** 11b9b088375d33a87b201cdbe18be71802e2607717Dan Stoza** Unless required by applicable law or agreed to in writing, software 12b9b088375d33a87b201cdbe18be71802e2607717Dan Stoza** distributed under the License is distributed on an "AS IS" BASIS, 13b9b088375d33a87b201cdbe18be71802e2607717Dan Stoza** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. 14b9b088375d33a87b201cdbe18be71802e2607717Dan Stoza** See the License for the specific language governing permissions and 15b9b088375d33a87b201cdbe18be71802e2607717Dan Stoza** limitations under the License. 16b9b088375d33a87b201cdbe18be71802e2607717Dan Stoza*/ 17b9b088375d33a87b201cdbe18be71802e2607717Dan Stoza 18b9b088375d33a87b201cdbe18be71802e2607717Dan Stoza 19b9b088375d33a87b201cdbe18be71802e2607717Dan Stoza#define LOG_TAG "AudioFlinger" 20b9b088375d33a87b201cdbe18be71802e2607717Dan Stoza//#define LOG_NDEBUG 0 21b9b088375d33a87b201cdbe18be71802e2607717Dan Stoza#define ATRACE_TAG ATRACE_TAG_AUDIO 22b9b088375d33a87b201cdbe18be71802e2607717Dan Stoza 23b9b088375d33a87b201cdbe18be71802e2607717Dan Stoza#include "Configuration.h" 24f0eaf25e9247edf4d124bedaeb863f7abdf35a3eDan Stoza#include <math.h> 25b9b088375d33a87b201cdbe18be71802e2607717Dan Stoza#include <fcntl.h> 26b9b088375d33a87b201cdbe18be71802e2607717Dan Stoza#include <linux/futex.h> 27b9b088375d33a87b201cdbe18be71802e2607717Dan Stoza#include <sys/stat.h> 28b9b088375d33a87b201cdbe18be71802e2607717Dan Stoza#include <sys/syscall.h> 29b9b088375d33a87b201cdbe18be71802e2607717Dan Stoza#include <cutils/properties.h> 30b3d0bdf0dbc19f0a0d7d924693025371e24828fdDan Stoza#include <media/AudioParameter.h> 31b9b088375d33a87b201cdbe18be71802e2607717Dan Stoza#include <media/AudioResamplerPublic.h> 32b3d0bdf0dbc19f0a0d7d924693025371e24828fdDan Stoza#include <utils/Log.h> 33b9b088375d33a87b201cdbe18be71802e2607717Dan Stoza#include <utils/Trace.h> 34b9b088375d33a87b201cdbe18be71802e2607717Dan Stoza 35b9b088375d33a87b201cdbe18be71802e2607717Dan Stoza#include <private/media/AudioTrackShared.h> 36b9b088375d33a87b201cdbe18be71802e2607717Dan Stoza#include <hardware/audio.h> 37b9b088375d33a87b201cdbe18be71802e2607717Dan Stoza#include <audio_effects/effect_ns.h> 38fa455354557f6283ff3a7d76979e52fd251c155fPablo Ceballos#include <audio_effects/effect_aec.h> 39fa455354557f6283ff3a7d76979e52fd251c155fPablo Ceballos#include <audio_utils/primitives.h> 40567dbbb6dd42be5013fcde0dadb3316d85f2fa0dPablo Ceballos#include <audio_utils/format.h> 41567dbbb6dd42be5013fcde0dadb3316d85f2fa0dPablo Ceballos#include <audio_utils/minifloat.h> 42b9b088375d33a87b201cdbe18be71802e2607717Dan Stoza 43d9822a3843017444364899afc3c23fb5be6b9cb9Dan Stoza// NBAIO implementations 44d9822a3843017444364899afc3c23fb5be6b9cb9Dan Stoza#include <media/nbaio/AudioStreamInSource.h> 45b9b088375d33a87b201cdbe18be71802e2607717Dan Stoza#include <media/nbaio/AudioStreamOutSink.h> 46b9b088375d33a87b201cdbe18be71802e2607717Dan Stoza#include <media/nbaio/MonoPipe.h> 47b9b088375d33a87b201cdbe18be71802e2607717Dan Stoza#include <media/nbaio/MonoPipeReader.h> 48b9b088375d33a87b201cdbe18be71802e2607717Dan Stoza#include <media/nbaio/Pipe.h> 49583b1b32191992d6ada58b3c61c71932a71c0c4bPablo Ceballos#include <media/nbaio/PipeReader.h> 50b9b088375d33a87b201cdbe18be71802e2607717Dan Stoza#include <media/nbaio/SourceAudioBufferProvider.h> 51f0eaf25e9247edf4d124bedaeb863f7abdf35a3eDan Stoza 52b9b088375d33a87b201cdbe18be71802e2607717Dan Stoza#include <powermanager/PowerManager.h> 53b9b088375d33a87b201cdbe18be71802e2607717Dan Stoza 54b9b088375d33a87b201cdbe18be71802e2607717Dan Stoza#include <common_time/cc_helper.h> 55567dbbb6dd42be5013fcde0dadb3316d85f2fa0dPablo Ceballos#include <common_time/local_clock.h> 563be1c6b60a188dc10025e2ce156c11fac050625dDan Stoza 579de7293b0a1b01ebe6fb1ab4a498f144adc8029fDan Stoza#include "AudioFlinger.h" 58812ed0644f8f8f71ca403f4e5793f0dbc1fcf9b2Dan Stoza#include "AudioMixer.h" 59c6f30bdee1f634eb90d68cb76efe935b6535a1e8Dan Stoza#include "FastMixer.h" 607dde599bf1a0dbef7390d91c2689d506371cdbd7Dan Stoza#include "FastCapture.h" 61127fc63e8a15366b4395f1363e8e18eb058d1709Dan Stoza#include "ServiceUtilities.h" 6250101d02a8eae555887282a5f761fdec57bdaf30Dan Stoza#include "SchedulingPolicyService.h" 631a61da5e28fa16ad556a58193c8bbeb32a5f636dJohn Reck 64b3d0bdf0dbc19f0a0d7d924693025371e24828fdDan Stoza#ifdef ADD_BATTERY_DATA 653559fbf93801e2c0d9d8fb246fb9b867a361b464Pablo Ceballos#include <media/IMediaPlayerService.h> 66ff95aabbcc6e8606acbd7933c90eeb9b8b382a21Pablo Ceballos#include <media/IMediaDeathNotifier.h> 67eb7980c224a54f860b7af5ecf30cbc633ae41289Pablo Ceballos#endif 68b9b088375d33a87b201cdbe18be71802e2607717Dan Stoza 69b9b088375d33a87b201cdbe18be71802e2607717Dan Stoza#ifdef DEBUG_CPU_USAGE 70b3d0bdf0dbc19f0a0d7d924693025371e24828fdDan Stoza#include <cpustats/CentralTendencyStatistics.h> 71b9b088375d33a87b201cdbe18be71802e2607717Dan Stoza#include <cpustats/ThreadCpuUsage.h> 72b9b088375d33a87b201cdbe18be71802e2607717Dan Stoza#endif 73b9b088375d33a87b201cdbe18be71802e2607717Dan Stoza 74b9b088375d33a87b201cdbe18be71802e2607717Dan Stoza// ---------------------------------------------------------------------------- 75b9b088375d33a87b201cdbe18be71802e2607717Dan Stoza 76b9b088375d33a87b201cdbe18be71802e2607717Dan Stoza// Note: the following macro is used for extremely verbose logging message. In 77// order to run with ALOG_ASSERT turned on, we need to have LOG_NDEBUG set to 78// 0; but one side effect of this is to turn all LOGV's as well. Some messages 79// are so verbose that we want to suppress them even when we have ALOG_ASSERT 80// turned on. Do not uncomment the #def below unless you really know what you 81// are doing and want to see all of the extremely verbose messages. 82//#define VERY_VERY_VERBOSE_LOGGING 83#ifdef VERY_VERY_VERBOSE_LOGGING 84#define ALOGVV ALOGV 85#else 86#define ALOGVV(a...) do { } while(0) 87#endif 88 89// TODO: Move these macro/inlines to a header file. 90#define max(a, b) ((a) > (b) ? (a) : (b)) 91template <typename T> 92static inline T min(const T& a, const T& b) 93{ 94 return a < b ? a : b; 95} 96 97namespace android { 98 99// retry counts for buffer fill timeout 100// 50 * ~20msecs = 1 second 101static const int8_t kMaxTrackRetries = 50; 102static const int8_t kMaxTrackStartupRetries = 50; 103// allow less retry attempts on direct output thread. 104// direct outputs can be a scarce resource in audio hardware and should 105// be released as quickly as possible. 106static const int8_t kMaxTrackRetriesDirect = 2; 107 108// don't warn about blocked writes or record buffer overflows more often than this 109static const nsecs_t kWarningThrottleNs = seconds(5); 110 111// RecordThread loop sleep time upon application overrun or audio HAL read error 112static const int kRecordThreadSleepUs = 5000; 113 114// maximum time to wait in sendConfigEvent_l() for a status to be received 115static const nsecs_t kConfigEventTimeoutNs = seconds(2); 116 117// minimum sleep time for the mixer thread loop when tracks are active but in underrun 118static const uint32_t kMinThreadSleepTimeUs = 5000; 119// maximum divider applied to the active sleep time in the mixer thread loop 120static const uint32_t kMaxThreadSleepTimeShift = 2; 121 122// minimum normal sink buffer size, expressed in milliseconds rather than frames 123static const uint32_t kMinNormalSinkBufferSizeMs = 20; 124// maximum normal sink buffer size 125static const uint32_t kMaxNormalSinkBufferSizeMs = 24; 126 127// Offloaded output thread standby delay: allows track transition without going to standby 128static const nsecs_t kOffloadStandbyDelayNs = seconds(1); 129 130// Whether to use fast mixer 131static const enum { 132 FastMixer_Never, // never initialize or use: for debugging only 133 FastMixer_Always, // always initialize and use, even if not needed: for debugging only 134 // normal mixer multiplier is 1 135 FastMixer_Static, // initialize if needed, then use all the time if initialized, 136 // multiplier is calculated based on min & max normal mixer buffer size 137 FastMixer_Dynamic, // initialize if needed, then use dynamically depending on track load, 138 // multiplier is calculated based on min & max normal mixer buffer size 139 // FIXME for FastMixer_Dynamic: 140 // Supporting this option will require fixing HALs that can't handle large writes. 141 // For example, one HAL implementation returns an error from a large write, 142 // and another HAL implementation corrupts memory, possibly in the sample rate converter. 143 // We could either fix the HAL implementations, or provide a wrapper that breaks 144 // up large writes into smaller ones, and the wrapper would need to deal with scheduler. 145} kUseFastMixer = FastMixer_Static; 146 147// Whether to use fast capture 148static const enum { 149 FastCapture_Never, // never initialize or use: for debugging only 150 FastCapture_Always, // always initialize and use, even if not needed: for debugging only 151 FastCapture_Static, // initialize if needed, then use all the time if initialized 152} kUseFastCapture = FastCapture_Static; 153 154// Priorities for requestPriority 155static const int kPriorityAudioApp = 2; 156static const int kPriorityFastMixer = 3; 157static const int kPriorityFastCapture = 3; 158 159// IAudioFlinger::createTrack() reports back to client the total size of shared memory area 160// for the track. The client then sub-divides this into smaller buffers for its use. 161// Currently the client uses N-buffering by default, but doesn't tell us about the value of N. 162// So for now we just assume that client is double-buffered for fast tracks. 163// FIXME It would be better for client to tell AudioFlinger the value of N, 164// so AudioFlinger could allocate the right amount of memory. 165// See the client's minBufCount and mNotificationFramesAct calculations for details. 166 167// This is the default value, if not specified by property. 168static const int kFastTrackMultiplier = 2; 169 170// The minimum and maximum allowed values 171static const int kFastTrackMultiplierMin = 1; 172static const int kFastTrackMultiplierMax = 2; 173 174// The actual value to use, which can be specified per-device via property af.fast_track_multiplier. 175static int sFastTrackMultiplier = kFastTrackMultiplier; 176 177// See Thread::readOnlyHeap(). 178// Initially this heap is used to allocate client buffers for "fast" AudioRecord. 179// Eventually it will be the single buffer that FastCapture writes into via HAL read(), 180// and that all "fast" AudioRecord clients read from. In either case, the size can be small. 181static const size_t kRecordThreadReadOnlyHeapSize = 0x2000; 182 183// ---------------------------------------------------------------------------- 184 185static pthread_once_t sFastTrackMultiplierOnce = PTHREAD_ONCE_INIT; 186 187static void sFastTrackMultiplierInit() 188{ 189 char value[PROPERTY_VALUE_MAX]; 190 if (property_get("af.fast_track_multiplier", value, NULL) > 0) { 191 char *endptr; 192 unsigned long ul = strtoul(value, &endptr, 0); 193 if (*endptr == '\0' && kFastTrackMultiplierMin <= ul && ul <= kFastTrackMultiplierMax) { 194 sFastTrackMultiplier = (int) ul; 195 } 196 } 197} 198 199// ---------------------------------------------------------------------------- 200 201#ifdef ADD_BATTERY_DATA 202// To collect the amplifier usage 203static void addBatteryData(uint32_t params) { 204 sp<IMediaPlayerService> service = IMediaDeathNotifier::getMediaPlayerService(); 205 if (service == NULL) { 206 // it already logged 207 return; 208 } 209 210 service->addBatteryData(params); 211} 212#endif 213 214 215// ---------------------------------------------------------------------------- 216// CPU Stats 217// ---------------------------------------------------------------------------- 218 219class CpuStats { 220public: 221 CpuStats(); 222 void sample(const String8 &title); 223#ifdef DEBUG_CPU_USAGE 224private: 225 ThreadCpuUsage mCpuUsage; // instantaneous thread CPU usage in wall clock ns 226 CentralTendencyStatistics mWcStats; // statistics on thread CPU usage in wall clock ns 227 228 CentralTendencyStatistics mHzStats; // statistics on thread CPU usage in cycles 229 230 int mCpuNum; // thread's current CPU number 231 int mCpukHz; // frequency of thread's current CPU in kHz 232#endif 233}; 234 235CpuStats::CpuStats() 236#ifdef DEBUG_CPU_USAGE 237 : mCpuNum(-1), mCpukHz(-1) 238#endif 239{ 240} 241 242void CpuStats::sample(const String8 &title 243#ifndef DEBUG_CPU_USAGE 244 __unused 245#endif 246 ) { 247#ifdef DEBUG_CPU_USAGE 248 // get current thread's delta CPU time in wall clock ns 249 double wcNs; 250 bool valid = mCpuUsage.sampleAndEnable(wcNs); 251 252 // record sample for wall clock statistics 253 if (valid) { 254 mWcStats.sample(wcNs); 255 } 256 257 // get the current CPU number 258 int cpuNum = sched_getcpu(); 259 260 // get the current CPU frequency in kHz 261 int cpukHz = mCpuUsage.getCpukHz(cpuNum); 262 263 // check if either CPU number or frequency changed 264 if (cpuNum != mCpuNum || cpukHz != mCpukHz) { 265 mCpuNum = cpuNum; 266 mCpukHz = cpukHz; 267 // ignore sample for purposes of cycles 268 valid = false; 269 } 270 271 // if no change in CPU number or frequency, then record sample for cycle statistics 272 if (valid && mCpukHz > 0) { 273 double cycles = wcNs * cpukHz * 0.000001; 274 mHzStats.sample(cycles); 275 } 276 277 unsigned n = mWcStats.n(); 278 // mCpuUsage.elapsed() is expensive, so don't call it every loop 279 if ((n & 127) == 1) { 280 long long elapsed = mCpuUsage.elapsed(); 281 if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) { 282 double perLoop = elapsed / (double) n; 283 double perLoop100 = perLoop * 0.01; 284 double perLoop1k = perLoop * 0.001; 285 double mean = mWcStats.mean(); 286 double stddev = mWcStats.stddev(); 287 double minimum = mWcStats.minimum(); 288 double maximum = mWcStats.maximum(); 289 double meanCycles = mHzStats.mean(); 290 double stddevCycles = mHzStats.stddev(); 291 double minCycles = mHzStats.minimum(); 292 double maxCycles = mHzStats.maximum(); 293 mCpuUsage.resetElapsed(); 294 mWcStats.reset(); 295 mHzStats.reset(); 296 ALOGD("CPU usage for %s over past %.1f secs\n" 297 " (%u mixer loops at %.1f mean ms per loop):\n" 298 " us per mix loop: mean=%.0f stddev=%.0f min=%.0f max=%.0f\n" 299 " %% of wall: mean=%.1f stddev=%.1f min=%.1f max=%.1f\n" 300 " MHz: mean=%.1f, stddev=%.1f, min=%.1f max=%.1f", 301 title.string(), 302 elapsed * .000000001, n, perLoop * .000001, 303 mean * .001, 304 stddev * .001, 305 minimum * .001, 306 maximum * .001, 307 mean / perLoop100, 308 stddev / perLoop100, 309 minimum / perLoop100, 310 maximum / perLoop100, 311 meanCycles / perLoop1k, 312 stddevCycles / perLoop1k, 313 minCycles / perLoop1k, 314 maxCycles / perLoop1k); 315 316 } 317 } 318#endif 319}; 320 321// ---------------------------------------------------------------------------- 322// ThreadBase 323// ---------------------------------------------------------------------------- 324 325// static 326const char *AudioFlinger::ThreadBase::threadTypeToString(AudioFlinger::ThreadBase::type_t type) 327{ 328 switch (type) { 329 case MIXER: 330 return "MIXER"; 331 case DIRECT: 332 return "DIRECT"; 333 case DUPLICATING: 334 return "DUPLICATING"; 335 case RECORD: 336 return "RECORD"; 337 case OFFLOAD: 338 return "OFFLOAD"; 339 default: 340 return "unknown"; 341 } 342} 343 344String8 devicesToString(audio_devices_t devices) 345{ 346 static const struct mapping { 347 audio_devices_t mDevices; 348 const char * mString; 349 } mappingsOut[] = { 350 AUDIO_DEVICE_OUT_EARPIECE, "EARPIECE", 351 AUDIO_DEVICE_OUT_SPEAKER, "SPEAKER", 352 AUDIO_DEVICE_OUT_WIRED_HEADSET, "WIRED_HEADSET", 353 AUDIO_DEVICE_OUT_WIRED_HEADPHONE, "WIRED_HEADPHONE", 354 AUDIO_DEVICE_OUT_TELEPHONY_TX, "TELEPHONY_TX", 355 AUDIO_DEVICE_NONE, "NONE", // must be last 356 }, mappingsIn[] = { 357 AUDIO_DEVICE_IN_BUILTIN_MIC, "BUILTIN_MIC", 358 AUDIO_DEVICE_IN_WIRED_HEADSET, "WIRED_HEADSET", 359 AUDIO_DEVICE_IN_VOICE_CALL, "VOICE_CALL", 360 AUDIO_DEVICE_IN_REMOTE_SUBMIX, "REMOTE_SUBMIX", 361 AUDIO_DEVICE_NONE, "NONE", // must be last 362 }; 363 String8 result; 364 audio_devices_t allDevices = AUDIO_DEVICE_NONE; 365 const mapping *entry; 366 if (devices & AUDIO_DEVICE_BIT_IN) { 367 devices &= ~AUDIO_DEVICE_BIT_IN; 368 entry = mappingsIn; 369 } else { 370 entry = mappingsOut; 371 } 372 for ( ; entry->mDevices != AUDIO_DEVICE_NONE; entry++) { 373 allDevices = (audio_devices_t) (allDevices | entry->mDevices); 374 if (devices & entry->mDevices) { 375 if (!result.isEmpty()) { 376 result.append("|"); 377 } 378 result.append(entry->mString); 379 } 380 } 381 if (devices & ~allDevices) { 382 if (!result.isEmpty()) { 383 result.append("|"); 384 } 385 result.appendFormat("0x%X", devices & ~allDevices); 386 } 387 if (result.isEmpty()) { 388 result.append(entry->mString); 389 } 390 return result; 391} 392 393String8 inputFlagsToString(audio_input_flags_t flags) 394{ 395 static const struct mapping { 396 audio_input_flags_t mFlag; 397 const char * mString; 398 } mappings[] = { 399 AUDIO_INPUT_FLAG_FAST, "FAST", 400 AUDIO_INPUT_FLAG_HW_HOTWORD, "HW_HOTWORD", 401 AUDIO_INPUT_FLAG_NONE, "NONE", // must be last 402 }; 403 String8 result; 404 audio_input_flags_t allFlags = AUDIO_INPUT_FLAG_NONE; 405 const mapping *entry; 406 for (entry = mappings; entry->mFlag != AUDIO_INPUT_FLAG_NONE; entry++) { 407 allFlags = (audio_input_flags_t) (allFlags | entry->mFlag); 408 if (flags & entry->mFlag) { 409 if (!result.isEmpty()) { 410 result.append("|"); 411 } 412 result.append(entry->mString); 413 } 414 } 415 if (flags & ~allFlags) { 416 if (!result.isEmpty()) { 417 result.append("|"); 418 } 419 result.appendFormat("0x%X", flags & ~allFlags); 420 } 421 if (result.isEmpty()) { 422 result.append(entry->mString); 423 } 424 return result; 425} 426 427String8 outputFlagsToString(audio_output_flags_t flags) 428{ 429 static const struct mapping { 430 audio_output_flags_t mFlag; 431 const char * mString; 432 } mappings[] = { 433 AUDIO_OUTPUT_FLAG_DIRECT, "DIRECT", 434 AUDIO_OUTPUT_FLAG_PRIMARY, "PRIMARY", 435 AUDIO_OUTPUT_FLAG_FAST, "FAST", 436 AUDIO_OUTPUT_FLAG_DEEP_BUFFER, "DEEP_BUFFER", 437 AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD, "COMPRESS_OFFLOAD", 438 AUDIO_OUTPUT_FLAG_NON_BLOCKING, "NON_BLOCKING", 439 AUDIO_OUTPUT_FLAG_HW_AV_SYNC, "HW_AV_SYNC", 440 AUDIO_OUTPUT_FLAG_NONE, "NONE", // must be last 441 }; 442 String8 result; 443 audio_output_flags_t allFlags = AUDIO_OUTPUT_FLAG_NONE; 444 const mapping *entry; 445 for (entry = mappings; entry->mFlag != AUDIO_OUTPUT_FLAG_NONE; entry++) { 446 allFlags = (audio_output_flags_t) (allFlags | entry->mFlag); 447 if (flags & entry->mFlag) { 448 if (!result.isEmpty()) { 449 result.append("|"); 450 } 451 result.append(entry->mString); 452 } 453 } 454 if (flags & ~allFlags) { 455 if (!result.isEmpty()) { 456 result.append("|"); 457 } 458 result.appendFormat("0x%X", flags & ~allFlags); 459 } 460 if (result.isEmpty()) { 461 result.append(entry->mString); 462 } 463 return result; 464} 465 466const char *sourceToString(audio_source_t source) 467{ 468 switch (source) { 469 case AUDIO_SOURCE_DEFAULT: return "default"; 470 case AUDIO_SOURCE_MIC: return "mic"; 471 case AUDIO_SOURCE_VOICE_UPLINK: return "voice uplink"; 472 case AUDIO_SOURCE_VOICE_DOWNLINK: return "voice downlink"; 473 case AUDIO_SOURCE_VOICE_CALL: return "voice call"; 474 case AUDIO_SOURCE_CAMCORDER: return "camcorder"; 475 case AUDIO_SOURCE_VOICE_RECOGNITION: return "voice recognition"; 476 case AUDIO_SOURCE_VOICE_COMMUNICATION: return "voice communication"; 477 case AUDIO_SOURCE_REMOTE_SUBMIX: return "remote submix"; 478 case AUDIO_SOURCE_FM_TUNER: return "FM tuner"; 479 case AUDIO_SOURCE_HOTWORD: return "hotword"; 480 default: return "unknown"; 481 } 482} 483 484AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, 485 audio_devices_t outDevice, audio_devices_t inDevice, type_t type) 486 : Thread(false /*canCallJava*/), 487 mType(type), 488 mAudioFlinger(audioFlinger), 489 // mSampleRate, mFrameCount, mChannelMask, mChannelCount, mFrameSize, mFormat, mBufferSize 490 // are set by PlaybackThread::readOutputParameters_l() or 491 // RecordThread::readInputParameters_l() 492 //FIXME: mStandby should be true here. Is this some kind of hack? 493 mStandby(false), mOutDevice(outDevice), mInDevice(inDevice), 494 mAudioSource(AUDIO_SOURCE_DEFAULT), mId(id), 495 // mName will be set by concrete (non-virtual) subclass 496 mDeathRecipient(new PMDeathRecipient(this)) 497{ 498} 499 500AudioFlinger::ThreadBase::~ThreadBase() 501{ 502 // mConfigEvents should be empty, but just in case it isn't, free the memory it owns 503 mConfigEvents.clear(); 504 505 // do not lock the mutex in destructor 506 releaseWakeLock_l(); 507 if (mPowerManager != 0) { 508 sp<IBinder> binder = IInterface::asBinder(mPowerManager); 509 binder->unlinkToDeath(mDeathRecipient); 510 } 511} 512 513status_t AudioFlinger::ThreadBase::readyToRun() 514{ 515 status_t status = initCheck(); 516 if (status == NO_ERROR) { 517 ALOGI("AudioFlinger's thread %p ready to run", this); 518 } else { 519 ALOGE("No working audio driver found."); 520 } 521 return status; 522} 523 524void AudioFlinger::ThreadBase::exit() 525{ 526 ALOGV("ThreadBase::exit"); 527 // do any cleanup required for exit to succeed 528 preExit(); 529 { 530 // This lock prevents the following race in thread (uniprocessor for illustration): 531 // if (!exitPending()) { 532 // // context switch from here to exit() 533 // // exit() calls requestExit(), what exitPending() observes 534 // // exit() calls signal(), which is dropped since no waiters 535 // // context switch back from exit() to here 536 // mWaitWorkCV.wait(...); 537 // // now thread is hung 538 // } 539 AutoMutex lock(mLock); 540 requestExit(); 541 mWaitWorkCV.broadcast(); 542 } 543 // When Thread::requestExitAndWait is made virtual and this method is renamed to 544 // "virtual status_t requestExitAndWait()", replace by "return Thread::requestExitAndWait();" 545 requestExitAndWait(); 546} 547 548status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs) 549{ 550 status_t status; 551 552 ALOGV("ThreadBase::setParameters() %s", keyValuePairs.string()); 553 Mutex::Autolock _l(mLock); 554 555 return sendSetParameterConfigEvent_l(keyValuePairs); 556} 557 558// sendConfigEvent_l() must be called with ThreadBase::mLock held 559// Can temporarily release the lock if waiting for a reply from processConfigEvents_l(). 560status_t AudioFlinger::ThreadBase::sendConfigEvent_l(sp<ConfigEvent>& event) 561{ 562 status_t status = NO_ERROR; 563 564 mConfigEvents.add(event); 565 ALOGV("sendConfigEvent_l() num events %d event %d", mConfigEvents.size(), event->mType); 566 mWaitWorkCV.signal(); 567 mLock.unlock(); 568 { 569 Mutex::Autolock _l(event->mLock); 570 while (event->mWaitStatus) { 571 if (event->mCond.waitRelative(event->mLock, kConfigEventTimeoutNs) != NO_ERROR) { 572 event->mStatus = TIMED_OUT; 573 event->mWaitStatus = false; 574 } 575 } 576 status = event->mStatus; 577 } 578 mLock.lock(); 579 return status; 580} 581 582void AudioFlinger::ThreadBase::sendIoConfigEvent(int event, int param) 583{ 584 Mutex::Autolock _l(mLock); 585 sendIoConfigEvent_l(event, param); 586} 587 588// sendIoConfigEvent_l() must be called with ThreadBase::mLock held 589void AudioFlinger::ThreadBase::sendIoConfigEvent_l(int event, int param) 590{ 591 sp<ConfigEvent> configEvent = (ConfigEvent *)new IoConfigEvent(event, param); 592 sendConfigEvent_l(configEvent); 593} 594 595// sendPrioConfigEvent_l() must be called with ThreadBase::mLock held 596void AudioFlinger::ThreadBase::sendPrioConfigEvent_l(pid_t pid, pid_t tid, int32_t prio) 597{ 598 sp<ConfigEvent> configEvent = (ConfigEvent *)new PrioConfigEvent(pid, tid, prio); 599 sendConfigEvent_l(configEvent); 600} 601 602// sendSetParameterConfigEvent_l() must be called with ThreadBase::mLock held 603status_t AudioFlinger::ThreadBase::sendSetParameterConfigEvent_l(const String8& keyValuePair) 604{ 605 sp<ConfigEvent> configEvent = (ConfigEvent *)new SetParameterConfigEvent(keyValuePair); 606 return sendConfigEvent_l(configEvent); 607} 608 609status_t AudioFlinger::ThreadBase::sendCreateAudioPatchConfigEvent( 610 const struct audio_patch *patch, 611 audio_patch_handle_t *handle) 612{ 613 Mutex::Autolock _l(mLock); 614 sp<ConfigEvent> configEvent = (ConfigEvent *)new CreateAudioPatchConfigEvent(*patch, *handle); 615 status_t status = sendConfigEvent_l(configEvent); 616 if (status == NO_ERROR) { 617 CreateAudioPatchConfigEventData *data = 618 (CreateAudioPatchConfigEventData *)configEvent->mData.get(); 619 *handle = data->mHandle; 620 } 621 return status; 622} 623 624status_t AudioFlinger::ThreadBase::sendReleaseAudioPatchConfigEvent( 625 const audio_patch_handle_t handle) 626{ 627 Mutex::Autolock _l(mLock); 628 sp<ConfigEvent> configEvent = (ConfigEvent *)new ReleaseAudioPatchConfigEvent(handle); 629 return sendConfigEvent_l(configEvent); 630} 631 632 633// post condition: mConfigEvents.isEmpty() 634void AudioFlinger::ThreadBase::processConfigEvents_l() 635{ 636 bool configChanged = false; 637 638 while (!mConfigEvents.isEmpty()) { 639 ALOGV("processConfigEvents_l() remaining events %d", mConfigEvents.size()); 640 sp<ConfigEvent> event = mConfigEvents[0]; 641 mConfigEvents.removeAt(0); 642 switch (event->mType) { 643 case CFG_EVENT_PRIO: { 644 PrioConfigEventData *data = (PrioConfigEventData *)event->mData.get(); 645 // FIXME Need to understand why this has to be done asynchronously 646 int err = requestPriority(data->mPid, data->mTid, data->mPrio, 647 true /*asynchronous*/); 648 if (err != 0) { 649 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 650 data->mPrio, data->mPid, data->mTid, err); 651 } 652 } break; 653 case CFG_EVENT_IO: { 654 IoConfigEventData *data = (IoConfigEventData *)event->mData.get(); 655 audioConfigChanged(data->mEvent, data->mParam); 656 } break; 657 case CFG_EVENT_SET_PARAMETER: { 658 SetParameterConfigEventData *data = (SetParameterConfigEventData *)event->mData.get(); 659 if (checkForNewParameter_l(data->mKeyValuePairs, event->mStatus)) { 660 configChanged = true; 661 } 662 } break; 663 case CFG_EVENT_CREATE_AUDIO_PATCH: { 664 CreateAudioPatchConfigEventData *data = 665 (CreateAudioPatchConfigEventData *)event->mData.get(); 666 event->mStatus = createAudioPatch_l(&data->mPatch, &data->mHandle); 667 } break; 668 case CFG_EVENT_RELEASE_AUDIO_PATCH: { 669 ReleaseAudioPatchConfigEventData *data = 670 (ReleaseAudioPatchConfigEventData *)event->mData.get(); 671 event->mStatus = releaseAudioPatch_l(data->mHandle); 672 } break; 673 default: 674 ALOG_ASSERT(false, "processConfigEvents_l() unknown event type %d", event->mType); 675 break; 676 } 677 { 678 Mutex::Autolock _l(event->mLock); 679 if (event->mWaitStatus) { 680 event->mWaitStatus = false; 681 event->mCond.signal(); 682 } 683 } 684 ALOGV_IF(mConfigEvents.isEmpty(), "processConfigEvents_l() DONE thread %p", this); 685 } 686 687 if (configChanged) { 688 cacheParameters_l(); 689 } 690} 691 692String8 channelMaskToString(audio_channel_mask_t mask, bool output) { 693 String8 s; 694 if (output) { 695 if (mask & AUDIO_CHANNEL_OUT_FRONT_LEFT) s.append("front-left, "); 696 if (mask & AUDIO_CHANNEL_OUT_FRONT_RIGHT) s.append("front-right, "); 697 if (mask & AUDIO_CHANNEL_OUT_FRONT_CENTER) s.append("front-center, "); 698 if (mask & AUDIO_CHANNEL_OUT_LOW_FREQUENCY) s.append("low freq, "); 699 if (mask & AUDIO_CHANNEL_OUT_BACK_LEFT) s.append("back-left, "); 700 if (mask & AUDIO_CHANNEL_OUT_BACK_RIGHT) s.append("back-right, "); 701 if (mask & AUDIO_CHANNEL_OUT_FRONT_LEFT_OF_CENTER) s.append("front-left-of-center, "); 702 if (mask & AUDIO_CHANNEL_OUT_FRONT_RIGHT_OF_CENTER) s.append("front-right-of-center, "); 703 if (mask & AUDIO_CHANNEL_OUT_BACK_CENTER) s.append("back-center, "); 704 if (mask & AUDIO_CHANNEL_OUT_SIDE_LEFT) s.append("side-left, "); 705 if (mask & AUDIO_CHANNEL_OUT_SIDE_RIGHT) s.append("side-right, "); 706 if (mask & AUDIO_CHANNEL_OUT_TOP_CENTER) s.append("top-center ,"); 707 if (mask & AUDIO_CHANNEL_OUT_TOP_FRONT_LEFT) s.append("top-front-left, "); 708 if (mask & AUDIO_CHANNEL_OUT_TOP_FRONT_CENTER) s.append("top-front-center, "); 709 if (mask & AUDIO_CHANNEL_OUT_TOP_FRONT_RIGHT) s.append("top-front-right, "); 710 if (mask & AUDIO_CHANNEL_OUT_TOP_BACK_LEFT) s.append("top-back-left, "); 711 if (mask & AUDIO_CHANNEL_OUT_TOP_BACK_CENTER) s.append("top-back-center, " ); 712 if (mask & AUDIO_CHANNEL_OUT_TOP_BACK_RIGHT) s.append("top-back-right, " ); 713 if (mask & ~AUDIO_CHANNEL_OUT_ALL) s.append("unknown, "); 714 } else { 715 if (mask & AUDIO_CHANNEL_IN_LEFT) s.append("left, "); 716 if (mask & AUDIO_CHANNEL_IN_RIGHT) s.append("right, "); 717 if (mask & AUDIO_CHANNEL_IN_FRONT) s.append("front, "); 718 if (mask & AUDIO_CHANNEL_IN_BACK) s.append("back, "); 719 if (mask & AUDIO_CHANNEL_IN_LEFT_PROCESSED) s.append("left-processed, "); 720 if (mask & AUDIO_CHANNEL_IN_RIGHT_PROCESSED) s.append("right-processed, "); 721 if (mask & AUDIO_CHANNEL_IN_FRONT_PROCESSED) s.append("front-processed, "); 722 if (mask & AUDIO_CHANNEL_IN_BACK_PROCESSED) s.append("back-processed, "); 723 if (mask & AUDIO_CHANNEL_IN_PRESSURE) s.append("pressure, "); 724 if (mask & AUDIO_CHANNEL_IN_X_AXIS) s.append("X, "); 725 if (mask & AUDIO_CHANNEL_IN_Y_AXIS) s.append("Y, "); 726 if (mask & AUDIO_CHANNEL_IN_Z_AXIS) s.append("Z, "); 727 if (mask & AUDIO_CHANNEL_IN_VOICE_UPLINK) s.append("voice-uplink, "); 728 if (mask & AUDIO_CHANNEL_IN_VOICE_DNLINK) s.append("voice-dnlink, "); 729 if (mask & ~AUDIO_CHANNEL_IN_ALL) s.append("unknown, "); 730 } 731 int len = s.length(); 732 if (s.length() > 2) { 733 char *str = s.lockBuffer(len); 734 s.unlockBuffer(len - 2); 735 } 736 return s; 737} 738 739void AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args __unused) 740{ 741 const size_t SIZE = 256; 742 char buffer[SIZE]; 743 String8 result; 744 745 bool locked = AudioFlinger::dumpTryLock(mLock); 746 if (!locked) { 747 dprintf(fd, "thread %p may be deadlocked\n", this); 748 } 749 750 dprintf(fd, " Thread name: %s\n", mThreadName); 751 dprintf(fd, " I/O handle: %d\n", mId); 752 dprintf(fd, " TID: %d\n", getTid()); 753 dprintf(fd, " Standby: %s\n", mStandby ? "yes" : "no"); 754 dprintf(fd, " Sample rate: %u Hz\n", mSampleRate); 755 dprintf(fd, " HAL frame count: %zu\n", mFrameCount); 756 dprintf(fd, " HAL format: 0x%x (%s)\n", mHALFormat, formatToString(mHALFormat)); 757 dprintf(fd, " HAL buffer size: %u bytes\n", mBufferSize); 758 dprintf(fd, " Channel count: %u\n", mChannelCount); 759 dprintf(fd, " Channel mask: 0x%08x (%s)\n", mChannelMask, 760 channelMaskToString(mChannelMask, mType != RECORD).string()); 761 dprintf(fd, " Format: 0x%x (%s)\n", mFormat, formatToString(mFormat)); 762 dprintf(fd, " Frame size: %zu bytes\n", mFrameSize); 763 dprintf(fd, " Pending config events:"); 764 size_t numConfig = mConfigEvents.size(); 765 if (numConfig) { 766 for (size_t i = 0; i < numConfig; i++) { 767 mConfigEvents[i]->dump(buffer, SIZE); 768 dprintf(fd, "\n %s", buffer); 769 } 770 dprintf(fd, "\n"); 771 } else { 772 dprintf(fd, " none\n"); 773 } 774 dprintf(fd, " Output device: %#x (%s)\n", mOutDevice, devicesToString(mOutDevice).string()); 775 dprintf(fd, " Input device: %#x (%s)\n", mInDevice, devicesToString(mInDevice).string()); 776 dprintf(fd, " Audio source: %d (%s)\n", mAudioSource, sourceToString(mAudioSource)); 777 778 if (locked) { 779 mLock.unlock(); 780 } 781} 782 783void AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args) 784{ 785 const size_t SIZE = 256; 786 char buffer[SIZE]; 787 String8 result; 788 789 size_t numEffectChains = mEffectChains.size(); 790 snprintf(buffer, SIZE, " %zu Effect Chains\n", numEffectChains); 791 write(fd, buffer, strlen(buffer)); 792 793 for (size_t i = 0; i < numEffectChains; ++i) { 794 sp<EffectChain> chain = mEffectChains[i]; 795 if (chain != 0) { 796 chain->dump(fd, args); 797 } 798 } 799} 800 801void AudioFlinger::ThreadBase::acquireWakeLock(int uid) 802{ 803 Mutex::Autolock _l(mLock); 804 acquireWakeLock_l(uid); 805} 806 807String16 AudioFlinger::ThreadBase::getWakeLockTag() 808{ 809 switch (mType) { 810 case MIXER: 811 return String16("AudioMix"); 812 case DIRECT: 813 return String16("AudioDirectOut"); 814 case DUPLICATING: 815 return String16("AudioDup"); 816 case RECORD: 817 return String16("AudioIn"); 818 case OFFLOAD: 819 return String16("AudioOffload"); 820 default: 821 ALOG_ASSERT(false); 822 return String16("AudioUnknown"); 823 } 824} 825 826void AudioFlinger::ThreadBase::acquireWakeLock_l(int uid) 827{ 828 getPowerManager_l(); 829 if (mPowerManager != 0) { 830 sp<IBinder> binder = new BBinder(); 831 status_t status; 832 if (uid >= 0) { 833 status = mPowerManager->acquireWakeLockWithUid(POWERMANAGER_PARTIAL_WAKE_LOCK, 834 binder, 835 getWakeLockTag(), 836 String16("media"), 837 uid, 838 true /* FIXME force oneway contrary to .aidl */); 839 } else { 840 status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK, 841 binder, 842 getWakeLockTag(), 843 String16("media"), 844 true /* FIXME force oneway contrary to .aidl */); 845 } 846 if (status == NO_ERROR) { 847 mWakeLockToken = binder; 848 } 849 ALOGV("acquireWakeLock_l() %s status %d", mThreadName, status); 850 } 851} 852 853void AudioFlinger::ThreadBase::releaseWakeLock() 854{ 855 Mutex::Autolock _l(mLock); 856 releaseWakeLock_l(); 857} 858 859void AudioFlinger::ThreadBase::releaseWakeLock_l() 860{ 861 if (mWakeLockToken != 0) { 862 ALOGV("releaseWakeLock_l() %s", mThreadName); 863 if (mPowerManager != 0) { 864 mPowerManager->releaseWakeLock(mWakeLockToken, 0, 865 true /* FIXME force oneway contrary to .aidl */); 866 } 867 mWakeLockToken.clear(); 868 } 869} 870 871void AudioFlinger::ThreadBase::updateWakeLockUids(const SortedVector<int> &uids) { 872 Mutex::Autolock _l(mLock); 873 updateWakeLockUids_l(uids); 874} 875 876void AudioFlinger::ThreadBase::getPowerManager_l() { 877 878 if (mPowerManager == 0) { 879 // use checkService() to avoid blocking if power service is not up yet 880 sp<IBinder> binder = 881 defaultServiceManager()->checkService(String16("power")); 882 if (binder == 0) { 883 ALOGW("Thread %s cannot connect to the power manager service", mThreadName); 884 } else { 885 mPowerManager = interface_cast<IPowerManager>(binder); 886 binder->linkToDeath(mDeathRecipient); 887 } 888 } 889} 890 891void AudioFlinger::ThreadBase::updateWakeLockUids_l(const SortedVector<int> &uids) { 892 893 getPowerManager_l(); 894 if (mWakeLockToken == NULL) { 895 ALOGE("no wake lock to update!"); 896 return; 897 } 898 if (mPowerManager != 0) { 899 sp<IBinder> binder = new BBinder(); 900 status_t status; 901 status = mPowerManager->updateWakeLockUids(mWakeLockToken, uids.size(), uids.array(), 902 true /* FIXME force oneway contrary to .aidl */); 903 ALOGV("acquireWakeLock_l() %s status %d", mThreadName, status); 904 } 905} 906 907void AudioFlinger::ThreadBase::clearPowerManager() 908{ 909 Mutex::Autolock _l(mLock); 910 releaseWakeLock_l(); 911 mPowerManager.clear(); 912} 913 914void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who __unused) 915{ 916 sp<ThreadBase> thread = mThread.promote(); 917 if (thread != 0) { 918 thread->clearPowerManager(); 919 } 920 ALOGW("power manager service died !!!"); 921} 922 923void AudioFlinger::ThreadBase::setEffectSuspended( 924 const effect_uuid_t *type, bool suspend, int sessionId) 925{ 926 Mutex::Autolock _l(mLock); 927 setEffectSuspended_l(type, suspend, sessionId); 928} 929 930void AudioFlinger::ThreadBase::setEffectSuspended_l( 931 const effect_uuid_t *type, bool suspend, int sessionId) 932{ 933 sp<EffectChain> chain = getEffectChain_l(sessionId); 934 if (chain != 0) { 935 if (type != NULL) { 936 chain->setEffectSuspended_l(type, suspend); 937 } else { 938 chain->setEffectSuspendedAll_l(suspend); 939 } 940 } 941 942 updateSuspendedSessions_l(type, suspend, sessionId); 943} 944 945void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain) 946{ 947 ssize_t index = mSuspendedSessions.indexOfKey(chain->sessionId()); 948 if (index < 0) { 949 return; 950 } 951 952 const KeyedVector <int, sp<SuspendedSessionDesc> >& sessionEffects = 953 mSuspendedSessions.valueAt(index); 954 955 for (size_t i = 0; i < sessionEffects.size(); i++) { 956 sp<SuspendedSessionDesc> desc = sessionEffects.valueAt(i); 957 for (int j = 0; j < desc->mRefCount; j++) { 958 if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) { 959 chain->setEffectSuspendedAll_l(true); 960 } else { 961 ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x", 962 desc->mType.timeLow); 963 chain->setEffectSuspended_l(&desc->mType, true); 964 } 965 } 966 } 967} 968 969void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type, 970 bool suspend, 971 int sessionId) 972{ 973 ssize_t index = mSuspendedSessions.indexOfKey(sessionId); 974 975 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects; 976 977 if (suspend) { 978 if (index >= 0) { 979 sessionEffects = mSuspendedSessions.valueAt(index); 980 } else { 981 mSuspendedSessions.add(sessionId, sessionEffects); 982 } 983 } else { 984 if (index < 0) { 985 return; 986 } 987 sessionEffects = mSuspendedSessions.valueAt(index); 988 } 989 990 991 int key = EffectChain::kKeyForSuspendAll; 992 if (type != NULL) { 993 key = type->timeLow; 994 } 995 index = sessionEffects.indexOfKey(key); 996 997 sp<SuspendedSessionDesc> desc; 998 if (suspend) { 999 if (index >= 0) { 1000 desc = sessionEffects.valueAt(index); 1001 } else { 1002 desc = new SuspendedSessionDesc(); 1003 if (type != NULL) { 1004 desc->mType = *type; 1005 } 1006 sessionEffects.add(key, desc); 1007 ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key); 1008 } 1009 desc->mRefCount++; 1010 } else { 1011 if (index < 0) { 1012 return; 1013 } 1014 desc = sessionEffects.valueAt(index); 1015 if (--desc->mRefCount == 0) { 1016 ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key); 1017 sessionEffects.removeItemsAt(index); 1018 if (sessionEffects.isEmpty()) { 1019 ALOGV("updateSuspendedSessions_l() restore removing session %d", 1020 sessionId); 1021 mSuspendedSessions.removeItem(sessionId); 1022 } 1023 } 1024 } 1025 if (!sessionEffects.isEmpty()) { 1026 mSuspendedSessions.replaceValueFor(sessionId, sessionEffects); 1027 } 1028} 1029 1030void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 1031 bool enabled, 1032 int sessionId) 1033{ 1034 Mutex::Autolock _l(mLock); 1035 checkSuspendOnEffectEnabled_l(effect, enabled, sessionId); 1036} 1037 1038void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 1039 bool enabled, 1040 int sessionId) 1041{ 1042 if (mType != RECORD) { 1043 // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on 1044 // another session. This gives the priority to well behaved effect control panels 1045 // and applications not using global effects. 1046 // Enabling post processing in AUDIO_SESSION_OUTPUT_STAGE session does not affect 1047 // global effects 1048 if ((sessionId != AUDIO_SESSION_OUTPUT_MIX) && (sessionId != AUDIO_SESSION_OUTPUT_STAGE)) { 1049 setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX); 1050 } 1051 } 1052 1053 sp<EffectChain> chain = getEffectChain_l(sessionId); 1054 if (chain != 0) { 1055 chain->checkSuspendOnEffectEnabled(effect, enabled); 1056 } 1057} 1058 1059// ThreadBase::createEffect_l() must be called with AudioFlinger::mLock held 1060sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 1061 const sp<AudioFlinger::Client>& client, 1062 const sp<IEffectClient>& effectClient, 1063 int32_t priority, 1064 int sessionId, 1065 effect_descriptor_t *desc, 1066 int *enabled, 1067 status_t *status) 1068{ 1069 sp<EffectModule> effect; 1070 sp<EffectHandle> handle; 1071 status_t lStatus; 1072 sp<EffectChain> chain; 1073 bool chainCreated = false; 1074 bool effectCreated = false; 1075 bool effectRegistered = false; 1076 1077 lStatus = initCheck(); 1078 if (lStatus != NO_ERROR) { 1079 ALOGW("createEffect_l() Audio driver not initialized."); 1080 goto Exit; 1081 } 1082 1083 // Reject any effect on Direct output threads for now, since the format of 1084 // mSinkBuffer is not guaranteed to be compatible with effect processing (PCM 16 stereo). 1085 if (mType == DIRECT) { 1086 ALOGW("createEffect_l() Cannot add effect %s on Direct output type thread %s", 1087 desc->name, mThreadName); 1088 lStatus = BAD_VALUE; 1089 goto Exit; 1090 } 1091 1092 // Reject any effect on mixer or duplicating multichannel sinks. 1093 // TODO: fix both format and multichannel issues with effects. 1094 if ((mType == MIXER || mType == DUPLICATING) && mChannelCount != FCC_2) { 1095 ALOGW("createEffect_l() Cannot add effect %s for multichannel(%d) %s threads", 1096 desc->name, mChannelCount, mType == MIXER ? "MIXER" : "DUPLICATING"); 1097 lStatus = BAD_VALUE; 1098 goto Exit; 1099 } 1100 1101 // Allow global effects only on offloaded and mixer threads 1102 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 1103 switch (mType) { 1104 case MIXER: 1105 case OFFLOAD: 1106 break; 1107 case DIRECT: 1108 case DUPLICATING: 1109 case RECORD: 1110 default: 1111 ALOGW("createEffect_l() Cannot add global effect %s on thread %s", 1112 desc->name, mThreadName); 1113 lStatus = BAD_VALUE; 1114 goto Exit; 1115 } 1116 } 1117 1118 // Only Pre processor effects are allowed on input threads and only on input threads 1119 if ((mType == RECORD) != ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { 1120 ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d", 1121 desc->name, desc->flags, mType); 1122 lStatus = BAD_VALUE; 1123 goto Exit; 1124 } 1125 1126 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 1127 1128 { // scope for mLock 1129 Mutex::Autolock _l(mLock); 1130 1131 // check for existing effect chain with the requested audio session 1132 chain = getEffectChain_l(sessionId); 1133 if (chain == 0) { 1134 // create a new chain for this session 1135 ALOGV("createEffect_l() new effect chain for session %d", sessionId); 1136 chain = new EffectChain(this, sessionId); 1137 addEffectChain_l(chain); 1138 chain->setStrategy(getStrategyForSession_l(sessionId)); 1139 chainCreated = true; 1140 } else { 1141 effect = chain->getEffectFromDesc_l(desc); 1142 } 1143 1144 ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get()); 1145 1146 if (effect == 0) { 1147 int id = mAudioFlinger->nextUniqueId(); 1148 // Check CPU and memory usage 1149 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 1150 if (lStatus != NO_ERROR) { 1151 goto Exit; 1152 } 1153 effectRegistered = true; 1154 // create a new effect module if none present in the chain 1155 effect = new EffectModule(this, chain, desc, id, sessionId); 1156 lStatus = effect->status(); 1157 if (lStatus != NO_ERROR) { 1158 goto Exit; 1159 } 1160 effect->setOffloaded(mType == OFFLOAD, mId); 1161 1162 lStatus = chain->addEffect_l(effect); 1163 if (lStatus != NO_ERROR) { 1164 goto Exit; 1165 } 1166 effectCreated = true; 1167 1168 effect->setDevice(mOutDevice); 1169 effect->setDevice(mInDevice); 1170 effect->setMode(mAudioFlinger->getMode()); 1171 effect->setAudioSource(mAudioSource); 1172 } 1173 // create effect handle and connect it to effect module 1174 handle = new EffectHandle(effect, client, effectClient, priority); 1175 lStatus = handle->initCheck(); 1176 if (lStatus == OK) { 1177 lStatus = effect->addHandle(handle.get()); 1178 } 1179 if (enabled != NULL) { 1180 *enabled = (int)effect->isEnabled(); 1181 } 1182 } 1183 1184Exit: 1185 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 1186 Mutex::Autolock _l(mLock); 1187 if (effectCreated) { 1188 chain->removeEffect_l(effect); 1189 } 1190 if (effectRegistered) { 1191 AudioSystem::unregisterEffect(effect->id()); 1192 } 1193 if (chainCreated) { 1194 removeEffectChain_l(chain); 1195 } 1196 handle.clear(); 1197 } 1198 1199 *status = lStatus; 1200 return handle; 1201} 1202 1203sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect(int sessionId, int effectId) 1204{ 1205 Mutex::Autolock _l(mLock); 1206 return getEffect_l(sessionId, effectId); 1207} 1208 1209sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId) 1210{ 1211 sp<EffectChain> chain = getEffectChain_l(sessionId); 1212 return chain != 0 ? chain->getEffectFromId_l(effectId) : 0; 1213} 1214 1215// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 1216// PlaybackThread::mLock held 1217status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 1218{ 1219 // check for existing effect chain with the requested audio session 1220 int sessionId = effect->sessionId(); 1221 sp<EffectChain> chain = getEffectChain_l(sessionId); 1222 bool chainCreated = false; 1223 1224 ALOGD_IF((mType == OFFLOAD) && !effect->isOffloadable(), 1225 "addEffect_l() on offloaded thread %p: effect %s does not support offload flags %x", 1226 this, effect->desc().name, effect->desc().flags); 1227 1228 if (chain == 0) { 1229 // create a new chain for this session 1230 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 1231 chain = new EffectChain(this, sessionId); 1232 addEffectChain_l(chain); 1233 chain->setStrategy(getStrategyForSession_l(sessionId)); 1234 chainCreated = true; 1235 } 1236 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 1237 1238 if (chain->getEffectFromId_l(effect->id()) != 0) { 1239 ALOGW("addEffect_l() %p effect %s already present in chain %p", 1240 this, effect->desc().name, chain.get()); 1241 return BAD_VALUE; 1242 } 1243 1244 effect->setOffloaded(mType == OFFLOAD, mId); 1245 1246 status_t status = chain->addEffect_l(effect); 1247 if (status != NO_ERROR) { 1248 if (chainCreated) { 1249 removeEffectChain_l(chain); 1250 } 1251 return status; 1252 } 1253 1254 effect->setDevice(mOutDevice); 1255 effect->setDevice(mInDevice); 1256 effect->setMode(mAudioFlinger->getMode()); 1257 effect->setAudioSource(mAudioSource); 1258 return NO_ERROR; 1259} 1260 1261void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 1262 1263 ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 1264 effect_descriptor_t desc = effect->desc(); 1265 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 1266 detachAuxEffect_l(effect->id()); 1267 } 1268 1269 sp<EffectChain> chain = effect->chain().promote(); 1270 if (chain != 0) { 1271 // remove effect chain if removing last effect 1272 if (chain->removeEffect_l(effect) == 0) { 1273 removeEffectChain_l(chain); 1274 } 1275 } else { 1276 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 1277 } 1278} 1279 1280void AudioFlinger::ThreadBase::lockEffectChains_l( 1281 Vector< sp<AudioFlinger::EffectChain> >& effectChains) 1282{ 1283 effectChains = mEffectChains; 1284 for (size_t i = 0; i < mEffectChains.size(); i++) { 1285 mEffectChains[i]->lock(); 1286 } 1287} 1288 1289void AudioFlinger::ThreadBase::unlockEffectChains( 1290 const Vector< sp<AudioFlinger::EffectChain> >& effectChains) 1291{ 1292 for (size_t i = 0; i < effectChains.size(); i++) { 1293 effectChains[i]->unlock(); 1294 } 1295} 1296 1297sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId) 1298{ 1299 Mutex::Autolock _l(mLock); 1300 return getEffectChain_l(sessionId); 1301} 1302 1303sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) const 1304{ 1305 size_t size = mEffectChains.size(); 1306 for (size_t i = 0; i < size; i++) { 1307 if (mEffectChains[i]->sessionId() == sessionId) { 1308 return mEffectChains[i]; 1309 } 1310 } 1311 return 0; 1312} 1313 1314void AudioFlinger::ThreadBase::setMode(audio_mode_t mode) 1315{ 1316 Mutex::Autolock _l(mLock); 1317 size_t size = mEffectChains.size(); 1318 for (size_t i = 0; i < size; i++) { 1319 mEffectChains[i]->setMode_l(mode); 1320 } 1321} 1322 1323void AudioFlinger::ThreadBase::getAudioPortConfig(struct audio_port_config *config) 1324{ 1325 config->type = AUDIO_PORT_TYPE_MIX; 1326 config->ext.mix.handle = mId; 1327 config->sample_rate = mSampleRate; 1328 config->format = mFormat; 1329 config->channel_mask = mChannelMask; 1330 config->config_mask = AUDIO_PORT_CONFIG_SAMPLE_RATE|AUDIO_PORT_CONFIG_CHANNEL_MASK| 1331 AUDIO_PORT_CONFIG_FORMAT; 1332} 1333 1334 1335// ---------------------------------------------------------------------------- 1336// Playback 1337// ---------------------------------------------------------------------------- 1338 1339AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, 1340 AudioStreamOut* output, 1341 audio_io_handle_t id, 1342 audio_devices_t device, 1343 type_t type) 1344 : ThreadBase(audioFlinger, id, device, AUDIO_DEVICE_NONE, type), 1345 mNormalFrameCount(0), mSinkBuffer(NULL), 1346 mMixerBufferEnabled(AudioFlinger::kEnableExtendedPrecision), 1347 mMixerBuffer(NULL), 1348 mMixerBufferSize(0), 1349 mMixerBufferFormat(AUDIO_FORMAT_INVALID), 1350 mMixerBufferValid(false), 1351 mEffectBufferEnabled(AudioFlinger::kEnableExtendedPrecision), 1352 mEffectBuffer(NULL), 1353 mEffectBufferSize(0), 1354 mEffectBufferFormat(AUDIO_FORMAT_INVALID), 1355 mEffectBufferValid(false), 1356 mSuspended(0), mBytesWritten(0), 1357 mActiveTracksGeneration(0), 1358 // mStreamTypes[] initialized in constructor body 1359 mOutput(output), 1360 mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false), 1361 mMixerStatus(MIXER_IDLE), 1362 mMixerStatusIgnoringFastTracks(MIXER_IDLE), 1363 standbyDelay(AudioFlinger::mStandbyTimeInNsecs), 1364 mBytesRemaining(0), 1365 mCurrentWriteLength(0), 1366 mUseAsyncWrite(false), 1367 mWriteAckSequence(0), 1368 mDrainSequence(0), 1369 mSignalPending(false), 1370 mScreenState(AudioFlinger::mScreenState), 1371 // index 0 is reserved for normal mixer's submix 1372 mFastTrackAvailMask(((1 << FastMixerState::kMaxFastTracks) - 1) & ~1), 1373 mHwSupportsPause(false), mHwPaused(false), mFlushPending(false), 1374 // mLatchD, mLatchQ, 1375 mLatchDValid(false), mLatchQValid(false) 1376{ 1377 snprintf(mThreadName, kThreadNameLength, "AudioOut_%X", id); 1378 mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mThreadName); 1379 1380 // Assumes constructor is called by AudioFlinger with it's mLock held, but 1381 // it would be safer to explicitly pass initial masterVolume/masterMute as 1382 // parameter. 1383 // 1384 // If the HAL we are using has support for master volume or master mute, 1385 // then do not attenuate or mute during mixing (just leave the volume at 1.0 1386 // and the mute set to false). 1387 mMasterVolume = audioFlinger->masterVolume_l(); 1388 mMasterMute = audioFlinger->masterMute_l(); 1389 if (mOutput && mOutput->audioHwDev) { 1390 if (mOutput->audioHwDev->canSetMasterVolume()) { 1391 mMasterVolume = 1.0; 1392 } 1393 1394 if (mOutput->audioHwDev->canSetMasterMute()) { 1395 mMasterMute = false; 1396 } 1397 } 1398 1399 readOutputParameters_l(); 1400 1401 // ++ operator does not compile 1402 for (audio_stream_type_t stream = AUDIO_STREAM_MIN; stream < AUDIO_STREAM_CNT; 1403 stream = (audio_stream_type_t) (stream + 1)) { 1404 mStreamTypes[stream].volume = mAudioFlinger->streamVolume_l(stream); 1405 mStreamTypes[stream].mute = mAudioFlinger->streamMute_l(stream); 1406 } 1407} 1408 1409AudioFlinger::PlaybackThread::~PlaybackThread() 1410{ 1411 mAudioFlinger->unregisterWriter(mNBLogWriter); 1412 free(mSinkBuffer); 1413 free(mMixerBuffer); 1414 free(mEffectBuffer); 1415} 1416 1417void AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) 1418{ 1419 dumpInternals(fd, args); 1420 dumpTracks(fd, args); 1421 dumpEffectChains(fd, args); 1422} 1423 1424void AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args __unused) 1425{ 1426 const size_t SIZE = 256; 1427 char buffer[SIZE]; 1428 String8 result; 1429 1430 result.appendFormat(" Stream volumes in dB: "); 1431 for (int i = 0; i < AUDIO_STREAM_CNT; ++i) { 1432 const stream_type_t *st = &mStreamTypes[i]; 1433 if (i > 0) { 1434 result.appendFormat(", "); 1435 } 1436 result.appendFormat("%d:%.2g", i, 20.0 * log10(st->volume)); 1437 if (st->mute) { 1438 result.append("M"); 1439 } 1440 } 1441 result.append("\n"); 1442 write(fd, result.string(), result.length()); 1443 result.clear(); 1444 1445 // These values are "raw"; they will wrap around. See prepareTracks_l() for a better way. 1446 FastTrackUnderruns underruns = getFastTrackUnderruns(0); 1447 dprintf(fd, " Normal mixer raw underrun counters: partial=%u empty=%u\n", 1448 underruns.mBitFields.mPartial, underruns.mBitFields.mEmpty); 1449 1450 size_t numtracks = mTracks.size(); 1451 size_t numactive = mActiveTracks.size(); 1452 dprintf(fd, " %d Tracks", numtracks); 1453 size_t numactiveseen = 0; 1454 if (numtracks) { 1455 dprintf(fd, " of which %d are active\n", numactive); 1456 Track::appendDumpHeader(result); 1457 for (size_t i = 0; i < numtracks; ++i) { 1458 sp<Track> track = mTracks[i]; 1459 if (track != 0) { 1460 bool active = mActiveTracks.indexOf(track) >= 0; 1461 if (active) { 1462 numactiveseen++; 1463 } 1464 track->dump(buffer, SIZE, active); 1465 result.append(buffer); 1466 } 1467 } 1468 } else { 1469 result.append("\n"); 1470 } 1471 if (numactiveseen != numactive) { 1472 // some tracks in the active list were not in the tracks list 1473 snprintf(buffer, SIZE, " The following tracks are in the active list but" 1474 " not in the track list\n"); 1475 result.append(buffer); 1476 Track::appendDumpHeader(result); 1477 for (size_t i = 0; i < numactive; ++i) { 1478 sp<Track> track = mActiveTracks[i].promote(); 1479 if (track != 0 && mTracks.indexOf(track) < 0) { 1480 track->dump(buffer, SIZE, true); 1481 result.append(buffer); 1482 } 1483 } 1484 } 1485 1486 write(fd, result.string(), result.size()); 1487} 1488 1489void AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 1490{ 1491 dprintf(fd, "\nOutput thread %p type %d (%s):\n", this, type(), threadTypeToString(type())); 1492 1493 dumpBase(fd, args); 1494 1495 dprintf(fd, " Normal frame count: %zu\n", mNormalFrameCount); 1496 dprintf(fd, " Last write occurred (msecs): %llu\n", ns2ms(systemTime() - mLastWriteTime)); 1497 dprintf(fd, " Total writes: %d\n", mNumWrites); 1498 dprintf(fd, " Delayed writes: %d\n", mNumDelayedWrites); 1499 dprintf(fd, " Blocked in write: %s\n", mInWrite ? "yes" : "no"); 1500 dprintf(fd, " Suspend count: %d\n", mSuspended); 1501 dprintf(fd, " Sink buffer : %p\n", mSinkBuffer); 1502 dprintf(fd, " Mixer buffer: %p\n", mMixerBuffer); 1503 dprintf(fd, " Effect buffer: %p\n", mEffectBuffer); 1504 dprintf(fd, " Fast track availMask=%#x\n", mFastTrackAvailMask); 1505 AudioStreamOut *output = mOutput; 1506 audio_output_flags_t flags = output != NULL ? output->flags : AUDIO_OUTPUT_FLAG_NONE; 1507 String8 flagsAsString = outputFlagsToString(flags); 1508 dprintf(fd, " AudioStreamOut: %p flags %#x (%s)\n", output, flags, flagsAsString.string()); 1509} 1510 1511// Thread virtuals 1512 1513void AudioFlinger::PlaybackThread::onFirstRef() 1514{ 1515 run(mThreadName, ANDROID_PRIORITY_URGENT_AUDIO); 1516} 1517 1518// ThreadBase virtuals 1519void AudioFlinger::PlaybackThread::preExit() 1520{ 1521 ALOGV(" preExit()"); 1522 // FIXME this is using hard-coded strings but in the future, this functionality will be 1523 // converted to use audio HAL extensions required to support tunneling 1524 mOutput->stream->common.set_parameters(&mOutput->stream->common, "exiting=1"); 1525} 1526 1527// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 1528sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 1529 const sp<AudioFlinger::Client>& client, 1530 audio_stream_type_t streamType, 1531 uint32_t sampleRate, 1532 audio_format_t format, 1533 audio_channel_mask_t channelMask, 1534 size_t *pFrameCount, 1535 const sp<IMemory>& sharedBuffer, 1536 int sessionId, 1537 IAudioFlinger::track_flags_t *flags, 1538 pid_t tid, 1539 int uid, 1540 status_t *status) 1541{ 1542 size_t frameCount = *pFrameCount; 1543 sp<Track> track; 1544 status_t lStatus; 1545 1546 bool isTimed = (*flags & IAudioFlinger::TRACK_TIMED) != 0; 1547 1548 // client expresses a preference for FAST, but we get the final say 1549 if (*flags & IAudioFlinger::TRACK_FAST) { 1550 if ( 1551 // not timed 1552 (!isTimed) && 1553 // either of these use cases: 1554 ( 1555 // use case 1: shared buffer with any frame count 1556 ( 1557 (sharedBuffer != 0) 1558 ) || 1559 // use case 2: frame count is default or at least as large as HAL 1560 ( 1561 // we formerly checked for a callback handler (non-0 tid), 1562 // but that is no longer required for TRANSFER_OBTAIN mode 1563 ((frameCount == 0) || 1564 (frameCount >= mFrameCount)) 1565 ) 1566 ) && 1567 // PCM data 1568 audio_is_linear_pcm(format) && 1569 // identical channel mask to sink, or mono in and stereo sink 1570 (channelMask == mChannelMask || 1571 (channelMask == AUDIO_CHANNEL_OUT_MONO && 1572 mChannelMask == AUDIO_CHANNEL_OUT_STEREO)) && 1573 // hardware sample rate 1574 (sampleRate == mSampleRate) && 1575 // normal mixer has an associated fast mixer 1576 hasFastMixer() && 1577 // there are sufficient fast track slots available 1578 (mFastTrackAvailMask != 0) 1579 // FIXME test that MixerThread for this fast track has a capable output HAL 1580 // FIXME add a permission test also? 1581 ) { 1582 // if frameCount not specified, then it defaults to fast mixer (HAL) frame count 1583 if (frameCount == 0) { 1584 // read the fast track multiplier property the first time it is needed 1585 int ok = pthread_once(&sFastTrackMultiplierOnce, sFastTrackMultiplierInit); 1586 if (ok != 0) { 1587 ALOGE("%s pthread_once failed: %d", __func__, ok); 1588 } 1589 frameCount = mFrameCount * sFastTrackMultiplier; 1590 } 1591 ALOGV("AUDIO_OUTPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d", 1592 frameCount, mFrameCount); 1593 } else { 1594 ALOGV("AUDIO_OUTPUT_FLAG_FAST denied: isTimed=%d sharedBuffer=%p frameCount=%d " 1595 "mFrameCount=%d format=%#x mFormat=%#x isLinear=%d channelMask=%#x " 1596 "sampleRate=%u mSampleRate=%u " 1597 "hasFastMixer=%d tid=%d fastTrackAvailMask=%#x", 1598 isTimed, sharedBuffer.get(), frameCount, mFrameCount, format, mFormat, 1599 audio_is_linear_pcm(format), 1600 channelMask, sampleRate, mSampleRate, hasFastMixer(), tid, mFastTrackAvailMask); 1601 *flags &= ~IAudioFlinger::TRACK_FAST; 1602 } 1603 } 1604 // For normal PCM streaming tracks, update minimum frame count. 1605 // For compatibility with AudioTrack calculation, buffer depth is forced 1606 // to be at least 2 x the normal mixer frame count and cover audio hardware latency. 1607 // This is probably too conservative, but legacy application code may depend on it. 1608 // If you change this calculation, also review the start threshold which is related. 1609 if (!(*flags & IAudioFlinger::TRACK_FAST) 1610 && audio_is_linear_pcm(format) && sharedBuffer == 0) { 1611 uint32_t latencyMs = mOutput->stream->get_latency(mOutput->stream); 1612 uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate); 1613 if (minBufCount < 2) { 1614 minBufCount = 2; 1615 } 1616 size_t minFrameCount = 1617 minBufCount * sourceFramesNeeded(sampleRate, mNormalFrameCount, mSampleRate); 1618 if (frameCount < minFrameCount) { // including frameCount == 0 1619 frameCount = minFrameCount; 1620 } 1621 } 1622 *pFrameCount = frameCount; 1623 1624 switch (mType) { 1625 1626 case DIRECT: 1627 if (audio_is_linear_pcm(format)) { 1628 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1629 ALOGE("createTrack_l() Bad parameter: sampleRate %u format %#x, channelMask 0x%08x " 1630 "for output %p with format %#x", 1631 sampleRate, format, channelMask, mOutput, mFormat); 1632 lStatus = BAD_VALUE; 1633 goto Exit; 1634 } 1635 } 1636 break; 1637 1638 case OFFLOAD: 1639 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1640 ALOGE("createTrack_l() Bad parameter: sampleRate %d format %#x, channelMask 0x%08x \"" 1641 "for output %p with format %#x", 1642 sampleRate, format, channelMask, mOutput, mFormat); 1643 lStatus = BAD_VALUE; 1644 goto Exit; 1645 } 1646 break; 1647 1648 default: 1649 if (!audio_is_linear_pcm(format)) { 1650 ALOGE("createTrack_l() Bad parameter: format %#x \"" 1651 "for output %p with format %#x", 1652 format, mOutput, mFormat); 1653 lStatus = BAD_VALUE; 1654 goto Exit; 1655 } 1656 if (sampleRate > mSampleRate * AUDIO_RESAMPLER_DOWN_RATIO_MAX) { 1657 ALOGE("Sample rate out of range: %u mSampleRate %u", sampleRate, mSampleRate); 1658 lStatus = BAD_VALUE; 1659 goto Exit; 1660 } 1661 break; 1662 1663 } 1664 1665 lStatus = initCheck(); 1666 if (lStatus != NO_ERROR) { 1667 ALOGE("createTrack_l() audio driver not initialized"); 1668 goto Exit; 1669 } 1670 1671 { // scope for mLock 1672 Mutex::Autolock _l(mLock); 1673 1674 // all tracks in same audio session must share the same routing strategy otherwise 1675 // conflicts will happen when tracks are moved from one output to another by audio policy 1676 // manager 1677 uint32_t strategy = AudioSystem::getStrategyForStream(streamType); 1678 for (size_t i = 0; i < mTracks.size(); ++i) { 1679 sp<Track> t = mTracks[i]; 1680 if (t != 0 && t->isExternalTrack()) { 1681 uint32_t actual = AudioSystem::getStrategyForStream(t->streamType()); 1682 if (sessionId == t->sessionId() && strategy != actual) { 1683 ALOGE("createTrack_l() mismatched strategy; expected %u but found %u", 1684 strategy, actual); 1685 lStatus = BAD_VALUE; 1686 goto Exit; 1687 } 1688 } 1689 } 1690 1691 if (!isTimed) { 1692 track = new Track(this, client, streamType, sampleRate, format, 1693 channelMask, frameCount, NULL, sharedBuffer, 1694 sessionId, uid, *flags, TrackBase::TYPE_DEFAULT); 1695 } else { 1696 track = TimedTrack::create(this, client, streamType, sampleRate, format, 1697 channelMask, frameCount, sharedBuffer, sessionId, uid); 1698 } 1699 1700 // new Track always returns non-NULL, 1701 // but TimedTrack::create() is a factory that could fail by returning NULL 1702 lStatus = track != 0 ? track->initCheck() : (status_t) NO_MEMORY; 1703 if (lStatus != NO_ERROR) { 1704 ALOGE("createTrack_l() initCheck failed %d; no control block?", lStatus); 1705 // track must be cleared from the caller as the caller has the AF lock 1706 goto Exit; 1707 } 1708 mTracks.add(track); 1709 1710 sp<EffectChain> chain = getEffectChain_l(sessionId); 1711 if (chain != 0) { 1712 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1713 track->setMainBuffer(chain->inBuffer()); 1714 chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType())); 1715 chain->incTrackCnt(); 1716 } 1717 1718 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 1719 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 1720 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 1721 // so ask activity manager to do this on our behalf 1722 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 1723 } 1724 } 1725 1726 lStatus = NO_ERROR; 1727 1728Exit: 1729 *status = lStatus; 1730 return track; 1731} 1732 1733uint32_t AudioFlinger::PlaybackThread::correctLatency_l(uint32_t latency) const 1734{ 1735 return latency; 1736} 1737 1738uint32_t AudioFlinger::PlaybackThread::latency() const 1739{ 1740 Mutex::Autolock _l(mLock); 1741 return latency_l(); 1742} 1743uint32_t AudioFlinger::PlaybackThread::latency_l() const 1744{ 1745 if (initCheck() == NO_ERROR) { 1746 return correctLatency_l(mOutput->stream->get_latency(mOutput->stream)); 1747 } else { 1748 return 0; 1749 } 1750} 1751 1752void AudioFlinger::PlaybackThread::setMasterVolume(float value) 1753{ 1754 Mutex::Autolock _l(mLock); 1755 // Don't apply master volume in SW if our HAL can do it for us. 1756 if (mOutput && mOutput->audioHwDev && 1757 mOutput->audioHwDev->canSetMasterVolume()) { 1758 mMasterVolume = 1.0; 1759 } else { 1760 mMasterVolume = value; 1761 } 1762} 1763 1764void AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1765{ 1766 Mutex::Autolock _l(mLock); 1767 // Don't apply master mute in SW if our HAL can do it for us. 1768 if (mOutput && mOutput->audioHwDev && 1769 mOutput->audioHwDev->canSetMasterMute()) { 1770 mMasterMute = false; 1771 } else { 1772 mMasterMute = muted; 1773 } 1774} 1775 1776void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) 1777{ 1778 Mutex::Autolock _l(mLock); 1779 mStreamTypes[stream].volume = value; 1780 broadcast_l(); 1781} 1782 1783void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) 1784{ 1785 Mutex::Autolock _l(mLock); 1786 mStreamTypes[stream].mute = muted; 1787 broadcast_l(); 1788} 1789 1790float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const 1791{ 1792 Mutex::Autolock _l(mLock); 1793 return mStreamTypes[stream].volume; 1794} 1795 1796// addTrack_l() must be called with ThreadBase::mLock held 1797status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 1798{ 1799 status_t status = ALREADY_EXISTS; 1800 1801 // set retry count for buffer fill 1802 track->mRetryCount = kMaxTrackStartupRetries; 1803 if (mActiveTracks.indexOf(track) < 0) { 1804 // the track is newly added, make sure it fills up all its 1805 // buffers before playing. This is to ensure the client will 1806 // effectively get the latency it requested. 1807 if (track->isExternalTrack()) { 1808 TrackBase::track_state state = track->mState; 1809 mLock.unlock(); 1810 status = AudioSystem::startOutput(mId, track->streamType(), 1811 (audio_session_t)track->sessionId()); 1812 mLock.lock(); 1813 // abort track was stopped/paused while we released the lock 1814 if (state != track->mState) { 1815 if (status == NO_ERROR) { 1816 mLock.unlock(); 1817 AudioSystem::stopOutput(mId, track->streamType(), 1818 (audio_session_t)track->sessionId()); 1819 mLock.lock(); 1820 } 1821 return INVALID_OPERATION; 1822 } 1823 // abort if start is rejected by audio policy manager 1824 if (status != NO_ERROR) { 1825 return PERMISSION_DENIED; 1826 } 1827#ifdef ADD_BATTERY_DATA 1828 // to track the speaker usage 1829 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 1830#endif 1831 } 1832 1833 track->mFillingUpStatus = track->sharedBuffer() != 0 ? Track::FS_FILLED : Track::FS_FILLING; 1834 track->mResetDone = false; 1835 track->mPresentationCompleteFrames = 0; 1836 mActiveTracks.add(track); 1837 mWakeLockUids.add(track->uid()); 1838 mActiveTracksGeneration++; 1839 mLatestActiveTrack = track; 1840 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1841 if (chain != 0) { 1842 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), 1843 track->sessionId()); 1844 chain->incActiveTrackCnt(); 1845 } 1846 1847 status = NO_ERROR; 1848 } 1849 1850 onAddNewTrack_l(); 1851 return status; 1852} 1853 1854bool AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 1855{ 1856 track->terminate(); 1857 // active tracks are removed by threadLoop() 1858 bool trackActive = (mActiveTracks.indexOf(track) >= 0); 1859 track->mState = TrackBase::STOPPED; 1860 if (!trackActive) { 1861 removeTrack_l(track); 1862 } else if (track->isFastTrack() || track->isOffloaded() || track->isDirect()) { 1863 track->mState = TrackBase::STOPPING_1; 1864 } 1865 1866 return trackActive; 1867} 1868 1869void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 1870{ 1871 track->triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 1872 mTracks.remove(track); 1873 deleteTrackName_l(track->name()); 1874 // redundant as track is about to be destroyed, for dumpsys only 1875 track->mName = -1; 1876 if (track->isFastTrack()) { 1877 int index = track->mFastIndex; 1878 ALOG_ASSERT(0 < index && index < (int)FastMixerState::kMaxFastTracks); 1879 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << index))); 1880 mFastTrackAvailMask |= 1 << index; 1881 // redundant as track is about to be destroyed, for dumpsys only 1882 track->mFastIndex = -1; 1883 } 1884 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1885 if (chain != 0) { 1886 chain->decTrackCnt(); 1887 } 1888} 1889 1890void AudioFlinger::PlaybackThread::broadcast_l() 1891{ 1892 // Thread could be blocked waiting for async 1893 // so signal it to handle state changes immediately 1894 // If threadLoop is currently unlocked a signal of mWaitWorkCV will 1895 // be lost so we also flag to prevent it blocking on mWaitWorkCV 1896 mSignalPending = true; 1897 mWaitWorkCV.broadcast(); 1898} 1899 1900String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 1901{ 1902 Mutex::Autolock _l(mLock); 1903 if (initCheck() != NO_ERROR) { 1904 return String8(); 1905 } 1906 1907 char *s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 1908 const String8 out_s8(s); 1909 free(s); 1910 return out_s8; 1911} 1912 1913void AudioFlinger::PlaybackThread::audioConfigChanged(int event, int param) { 1914 AudioSystem::OutputDescriptor desc; 1915 void *param2 = NULL; 1916 1917 ALOGV("PlaybackThread::audioConfigChanged, thread %p, event %d, param %d", this, event, 1918 param); 1919 1920 switch (event) { 1921 case AudioSystem::OUTPUT_OPENED: 1922 case AudioSystem::OUTPUT_CONFIG_CHANGED: 1923 desc.channelMask = mChannelMask; 1924 desc.samplingRate = mSampleRate; 1925 desc.format = mFormat; 1926 desc.frameCount = mNormalFrameCount; // FIXME see 1927 // AudioFlinger::frameCount(audio_io_handle_t) 1928 desc.latency = latency_l(); 1929 param2 = &desc; 1930 break; 1931 1932 case AudioSystem::STREAM_CONFIG_CHANGED: 1933 param2 = ¶m; 1934 case AudioSystem::OUTPUT_CLOSED: 1935 default: 1936 break; 1937 } 1938 mAudioFlinger->audioConfigChanged(event, mId, param2); 1939} 1940 1941void AudioFlinger::PlaybackThread::writeCallback() 1942{ 1943 ALOG_ASSERT(mCallbackThread != 0); 1944 mCallbackThread->resetWriteBlocked(); 1945} 1946 1947void AudioFlinger::PlaybackThread::drainCallback() 1948{ 1949 ALOG_ASSERT(mCallbackThread != 0); 1950 mCallbackThread->resetDraining(); 1951} 1952 1953void AudioFlinger::PlaybackThread::resetWriteBlocked(uint32_t sequence) 1954{ 1955 Mutex::Autolock _l(mLock); 1956 // reject out of sequence requests 1957 if ((mWriteAckSequence & 1) && (sequence == mWriteAckSequence)) { 1958 mWriteAckSequence &= ~1; 1959 mWaitWorkCV.signal(); 1960 } 1961} 1962 1963void AudioFlinger::PlaybackThread::resetDraining(uint32_t sequence) 1964{ 1965 Mutex::Autolock _l(mLock); 1966 // reject out of sequence requests 1967 if ((mDrainSequence & 1) && (sequence == mDrainSequence)) { 1968 mDrainSequence &= ~1; 1969 mWaitWorkCV.signal(); 1970 } 1971} 1972 1973// static 1974int AudioFlinger::PlaybackThread::asyncCallback(stream_callback_event_t event, 1975 void *param __unused, 1976 void *cookie) 1977{ 1978 AudioFlinger::PlaybackThread *me = (AudioFlinger::PlaybackThread *)cookie; 1979 ALOGV("asyncCallback() event %d", event); 1980 switch (event) { 1981 case STREAM_CBK_EVENT_WRITE_READY: 1982 me->writeCallback(); 1983 break; 1984 case STREAM_CBK_EVENT_DRAIN_READY: 1985 me->drainCallback(); 1986 break; 1987 default: 1988 ALOGW("asyncCallback() unknown event %d", event); 1989 break; 1990 } 1991 return 0; 1992} 1993 1994void AudioFlinger::PlaybackThread::readOutputParameters_l() 1995{ 1996 // unfortunately we have no way of recovering from errors here, hence the LOG_ALWAYS_FATAL 1997 mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common); 1998 mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common); 1999 if (!audio_is_output_channel(mChannelMask)) { 2000 LOG_ALWAYS_FATAL("HAL channel mask %#x not valid for output", mChannelMask); 2001 } 2002 if ((mType == MIXER || mType == DUPLICATING) 2003 && !isValidPcmSinkChannelMask(mChannelMask)) { 2004 LOG_ALWAYS_FATAL("HAL channel mask %#x not supported for mixed output", 2005 mChannelMask); 2006 } 2007 mChannelCount = audio_channel_count_from_out_mask(mChannelMask); 2008 mHALFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 2009 mFormat = mHALFormat; 2010 if (!audio_is_valid_format(mFormat)) { 2011 LOG_ALWAYS_FATAL("HAL format %#x not valid for output", mFormat); 2012 } 2013 if ((mType == MIXER || mType == DUPLICATING) 2014 && !isValidPcmSinkFormat(mFormat)) { 2015 LOG_FATAL("HAL format %#x not supported for mixed output", 2016 mFormat); 2017 } 2018 mFrameSize = mOutput->getFrameSize(); 2019 mBufferSize = mOutput->stream->common.get_buffer_size(&mOutput->stream->common); 2020 mFrameCount = mBufferSize / mFrameSize; 2021 if (mFrameCount & 15) { 2022 ALOGW("HAL output buffer size is %u frames but AudioMixer requires multiples of 16 frames", 2023 mFrameCount); 2024 } 2025 2026 if ((mOutput->flags & AUDIO_OUTPUT_FLAG_NON_BLOCKING) && 2027 (mOutput->stream->set_callback != NULL)) { 2028 if (mOutput->stream->set_callback(mOutput->stream, 2029 AudioFlinger::PlaybackThread::asyncCallback, this) == 0) { 2030 mUseAsyncWrite = true; 2031 mCallbackThread = new AudioFlinger::AsyncCallbackThread(this); 2032 } 2033 } 2034 2035 mHwSupportsPause = false; 2036 if (mOutput->flags & AUDIO_OUTPUT_FLAG_DIRECT) { 2037 if (mOutput->stream->pause != NULL) { 2038 if (mOutput->stream->resume != NULL) { 2039 mHwSupportsPause = true; 2040 } else { 2041 ALOGW("direct output implements pause but not resume"); 2042 } 2043 } else if (mOutput->stream->resume != NULL) { 2044 ALOGW("direct output implements resume but not pause"); 2045 } 2046 } 2047 2048 if (mType == DUPLICATING && mMixerBufferEnabled && mEffectBufferEnabled) { 2049 // For best precision, we use float instead of the associated output 2050 // device format (typically PCM 16 bit). 2051 2052 mFormat = AUDIO_FORMAT_PCM_FLOAT; 2053 mFrameSize = mChannelCount * audio_bytes_per_sample(mFormat); 2054 mBufferSize = mFrameSize * mFrameCount; 2055 2056 // TODO: We currently use the associated output device channel mask and sample rate. 2057 // (1) Perhaps use the ORed channel mask of all downstream MixerThreads 2058 // (if a valid mask) to avoid premature downmix. 2059 // (2) Perhaps use the maximum sample rate of all downstream MixerThreads 2060 // instead of the output device sample rate to avoid loss of high frequency information. 2061 // This may need to be updated as MixerThread/OutputTracks are added and not here. 2062 } 2063 2064 // Calculate size of normal sink buffer relative to the HAL output buffer size 2065 double multiplier = 1.0; 2066 if (mType == MIXER && (kUseFastMixer == FastMixer_Static || 2067 kUseFastMixer == FastMixer_Dynamic)) { 2068 size_t minNormalFrameCount = (kMinNormalSinkBufferSizeMs * mSampleRate) / 1000; 2069 size_t maxNormalFrameCount = (kMaxNormalSinkBufferSizeMs * mSampleRate) / 1000; 2070 // round up minimum and round down maximum to nearest 16 frames to satisfy AudioMixer 2071 minNormalFrameCount = (minNormalFrameCount + 15) & ~15; 2072 maxNormalFrameCount = maxNormalFrameCount & ~15; 2073 if (maxNormalFrameCount < minNormalFrameCount) { 2074 maxNormalFrameCount = minNormalFrameCount; 2075 } 2076 multiplier = (double) minNormalFrameCount / (double) mFrameCount; 2077 if (multiplier <= 1.0) { 2078 multiplier = 1.0; 2079 } else if (multiplier <= 2.0) { 2080 if (2 * mFrameCount <= maxNormalFrameCount) { 2081 multiplier = 2.0; 2082 } else { 2083 multiplier = (double) maxNormalFrameCount / (double) mFrameCount; 2084 } 2085 } else { 2086 // prefer an even multiplier, for compatibility with doubling of fast tracks due to HAL 2087 // SRC (it would be unusual for the normal sink buffer size to not be a multiple of fast 2088 // track, but we sometimes have to do this to satisfy the maximum frame count 2089 // constraint) 2090 // FIXME this rounding up should not be done if no HAL SRC 2091 uint32_t truncMult = (uint32_t) multiplier; 2092 if ((truncMult & 1)) { 2093 if ((truncMult + 1) * mFrameCount <= maxNormalFrameCount) { 2094 ++truncMult; 2095 } 2096 } 2097 multiplier = (double) truncMult; 2098 } 2099 } 2100 mNormalFrameCount = multiplier * mFrameCount; 2101 // round up to nearest 16 frames to satisfy AudioMixer 2102 if (mType == MIXER || mType == DUPLICATING) { 2103 mNormalFrameCount = (mNormalFrameCount + 15) & ~15; 2104 } 2105 ALOGI("HAL output buffer size %u frames, normal sink buffer size %u frames", mFrameCount, 2106 mNormalFrameCount); 2107 2108 // mSinkBuffer is the sink buffer. Size is always multiple-of-16 frames. 2109 // Originally this was int16_t[] array, need to remove legacy implications. 2110 free(mSinkBuffer); 2111 mSinkBuffer = NULL; 2112 // For sink buffer size, we use the frame size from the downstream sink to avoid problems 2113 // with non PCM formats for compressed music, e.g. AAC, and Offload threads. 2114 const size_t sinkBufferSize = mNormalFrameCount * mFrameSize; 2115 (void)posix_memalign(&mSinkBuffer, 32, sinkBufferSize); 2116 2117 // We resize the mMixerBuffer according to the requirements of the sink buffer which 2118 // drives the output. 2119 free(mMixerBuffer); 2120 mMixerBuffer = NULL; 2121 if (mMixerBufferEnabled) { 2122 mMixerBufferFormat = AUDIO_FORMAT_PCM_FLOAT; // also valid: AUDIO_FORMAT_PCM_16_BIT. 2123 mMixerBufferSize = mNormalFrameCount * mChannelCount 2124 * audio_bytes_per_sample(mMixerBufferFormat); 2125 (void)posix_memalign(&mMixerBuffer, 32, mMixerBufferSize); 2126 } 2127 free(mEffectBuffer); 2128 mEffectBuffer = NULL; 2129 if (mEffectBufferEnabled) { 2130 mEffectBufferFormat = AUDIO_FORMAT_PCM_16_BIT; // Note: Effects support 16b only 2131 mEffectBufferSize = mNormalFrameCount * mChannelCount 2132 * audio_bytes_per_sample(mEffectBufferFormat); 2133 (void)posix_memalign(&mEffectBuffer, 32, mEffectBufferSize); 2134 } 2135 2136 // force reconfiguration of effect chains and engines to take new buffer size and audio 2137 // parameters into account 2138 // Note that mLock is not held when readOutputParameters_l() is called from the constructor 2139 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 2140 // matter. 2141 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 2142 Vector< sp<EffectChain> > effectChains = mEffectChains; 2143 for (size_t i = 0; i < effectChains.size(); i ++) { 2144 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 2145 } 2146} 2147 2148 2149status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames) 2150{ 2151 if (halFrames == NULL || dspFrames == NULL) { 2152 return BAD_VALUE; 2153 } 2154 Mutex::Autolock _l(mLock); 2155 if (initCheck() != NO_ERROR) { 2156 return INVALID_OPERATION; 2157 } 2158 size_t framesWritten = mBytesWritten / mFrameSize; 2159 *halFrames = framesWritten; 2160 2161 if (isSuspended()) { 2162 // return an estimation of rendered frames when the output is suspended 2163 size_t latencyFrames = (latency_l() * mSampleRate) / 1000; 2164 *dspFrames = framesWritten >= latencyFrames ? framesWritten - latencyFrames : 0; 2165 return NO_ERROR; 2166 } else { 2167 status_t status; 2168 uint32_t frames; 2169 status = mOutput->getRenderPosition(&frames); 2170 *dspFrames = (size_t)frames; 2171 return status; 2172 } 2173} 2174 2175uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) const 2176{ 2177 Mutex::Autolock _l(mLock); 2178 uint32_t result = 0; 2179 if (getEffectChain_l(sessionId) != 0) { 2180 result = EFFECT_SESSION; 2181 } 2182 2183 for (size_t i = 0; i < mTracks.size(); ++i) { 2184 sp<Track> track = mTracks[i]; 2185 if (sessionId == track->sessionId() && !track->isInvalid()) { 2186 result |= TRACK_SESSION; 2187 break; 2188 } 2189 } 2190 2191 return result; 2192} 2193 2194uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 2195{ 2196 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 2197 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 2198 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 2199 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 2200 } 2201 for (size_t i = 0; i < mTracks.size(); i++) { 2202 sp<Track> track = mTracks[i]; 2203 if (sessionId == track->sessionId() && !track->isInvalid()) { 2204 return AudioSystem::getStrategyForStream(track->streamType()); 2205 } 2206 } 2207 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 2208} 2209 2210 2211AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 2212{ 2213 Mutex::Autolock _l(mLock); 2214 return mOutput; 2215} 2216 2217AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 2218{ 2219 Mutex::Autolock _l(mLock); 2220 AudioStreamOut *output = mOutput; 2221 mOutput = NULL; 2222 // FIXME FastMixer might also have a raw ptr to mOutputSink; 2223 // must push a NULL and wait for ack 2224 mOutputSink.clear(); 2225 mPipeSink.clear(); 2226 mNormalSink.clear(); 2227 return output; 2228} 2229 2230// this method must always be called either with ThreadBase mLock held or inside the thread loop 2231audio_stream_t* AudioFlinger::PlaybackThread::stream() const 2232{ 2233 if (mOutput == NULL) { 2234 return NULL; 2235 } 2236 return &mOutput->stream->common; 2237} 2238 2239uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() const 2240{ 2241 return (uint32_t)((uint32_t)((mNormalFrameCount * 1000) / mSampleRate) * 1000); 2242} 2243 2244status_t AudioFlinger::PlaybackThread::setSyncEvent(const sp<SyncEvent>& event) 2245{ 2246 if (!isValidSyncEvent(event)) { 2247 return BAD_VALUE; 2248 } 2249 2250 Mutex::Autolock _l(mLock); 2251 2252 for (size_t i = 0; i < mTracks.size(); ++i) { 2253 sp<Track> track = mTracks[i]; 2254 if (event->triggerSession() == track->sessionId()) { 2255 (void) track->setSyncEvent(event); 2256 return NO_ERROR; 2257 } 2258 } 2259 2260 return NAME_NOT_FOUND; 2261} 2262 2263bool AudioFlinger::PlaybackThread::isValidSyncEvent(const sp<SyncEvent>& event) const 2264{ 2265 return event->type() == AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE; 2266} 2267 2268void AudioFlinger::PlaybackThread::threadLoop_removeTracks( 2269 const Vector< sp<Track> >& tracksToRemove) 2270{ 2271 size_t count = tracksToRemove.size(); 2272 if (count > 0) { 2273 for (size_t i = 0 ; i < count ; i++) { 2274 const sp<Track>& track = tracksToRemove.itemAt(i); 2275 if (track->isExternalTrack()) { 2276 AudioSystem::stopOutput(mId, track->streamType(), 2277 (audio_session_t)track->sessionId()); 2278#ifdef ADD_BATTERY_DATA 2279 // to track the speaker usage 2280 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 2281#endif 2282 if (track->isTerminated()) { 2283 AudioSystem::releaseOutput(mId, track->streamType(), 2284 (audio_session_t)track->sessionId()); 2285 } 2286 } 2287 } 2288 } 2289} 2290 2291void AudioFlinger::PlaybackThread::checkSilentMode_l() 2292{ 2293 if (!mMasterMute) { 2294 char value[PROPERTY_VALUE_MAX]; 2295 if (property_get("ro.audio.silent", value, "0") > 0) { 2296 char *endptr; 2297 unsigned long ul = strtoul(value, &endptr, 0); 2298 if (*endptr == '\0' && ul != 0) { 2299 ALOGD("Silence is golden"); 2300 // The setprop command will not allow a property to be changed after 2301 // the first time it is set, so we don't have to worry about un-muting. 2302 setMasterMute_l(true); 2303 } 2304 } 2305 } 2306} 2307 2308// shared by MIXER and DIRECT, overridden by DUPLICATING 2309ssize_t AudioFlinger::PlaybackThread::threadLoop_write() 2310{ 2311 // FIXME rewrite to reduce number of system calls 2312 mLastWriteTime = systemTime(); 2313 mInWrite = true; 2314 ssize_t bytesWritten; 2315 const size_t offset = mCurrentWriteLength - mBytesRemaining; 2316 2317 // If an NBAIO sink is present, use it to write the normal mixer's submix 2318 if (mNormalSink != 0) { 2319 2320 const size_t count = mBytesRemaining / mFrameSize; 2321 2322 ATRACE_BEGIN("write"); 2323 // update the setpoint when AudioFlinger::mScreenState changes 2324 uint32_t screenState = AudioFlinger::mScreenState; 2325 if (screenState != mScreenState) { 2326 mScreenState = screenState; 2327 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 2328 if (pipe != NULL) { 2329 pipe->setAvgFrames((mScreenState & 1) ? 2330 (pipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2331 } 2332 } 2333 ssize_t framesWritten = mNormalSink->write((char *)mSinkBuffer + offset, count); 2334 ATRACE_END(); 2335 if (framesWritten > 0) { 2336 bytesWritten = framesWritten * mFrameSize; 2337 } else { 2338 bytesWritten = framesWritten; 2339 } 2340 mLatchDValid = false; 2341 status_t status = mNormalSink->getTimestamp(mLatchD.mTimestamp); 2342 if (status == NO_ERROR) { 2343 size_t totalFramesWritten = mNormalSink->framesWritten(); 2344 if (totalFramesWritten >= mLatchD.mTimestamp.mPosition) { 2345 mLatchD.mUnpresentedFrames = totalFramesWritten - mLatchD.mTimestamp.mPosition; 2346 // mLatchD.mFramesReleased is set immediately before D is clocked into Q 2347 mLatchDValid = true; 2348 } 2349 } 2350 // otherwise use the HAL / AudioStreamOut directly 2351 } else { 2352 // Direct output and offload threads 2353 2354 if (mUseAsyncWrite) { 2355 ALOGW_IF(mWriteAckSequence & 1, "threadLoop_write(): out of sequence write request"); 2356 mWriteAckSequence += 2; 2357 mWriteAckSequence |= 1; 2358 ALOG_ASSERT(mCallbackThread != 0); 2359 mCallbackThread->setWriteBlocked(mWriteAckSequence); 2360 } 2361 // FIXME We should have an implementation of timestamps for direct output threads. 2362 // They are used e.g for multichannel PCM playback over HDMI. 2363 bytesWritten = mOutput->write((char *)mSinkBuffer + offset, mBytesRemaining); 2364 if (mUseAsyncWrite && 2365 ((bytesWritten < 0) || (bytesWritten == (ssize_t)mBytesRemaining))) { 2366 // do not wait for async callback in case of error of full write 2367 mWriteAckSequence &= ~1; 2368 ALOG_ASSERT(mCallbackThread != 0); 2369 mCallbackThread->setWriteBlocked(mWriteAckSequence); 2370 } 2371 } 2372 2373 mNumWrites++; 2374 mInWrite = false; 2375 mStandby = false; 2376 return bytesWritten; 2377} 2378 2379void AudioFlinger::PlaybackThread::threadLoop_drain() 2380{ 2381 if (mOutput->stream->drain) { 2382 ALOGV("draining %s", (mMixerStatus == MIXER_DRAIN_TRACK) ? "early" : "full"); 2383 if (mUseAsyncWrite) { 2384 ALOGW_IF(mDrainSequence & 1, "threadLoop_drain(): out of sequence drain request"); 2385 mDrainSequence |= 1; 2386 ALOG_ASSERT(mCallbackThread != 0); 2387 mCallbackThread->setDraining(mDrainSequence); 2388 } 2389 mOutput->stream->drain(mOutput->stream, 2390 (mMixerStatus == MIXER_DRAIN_TRACK) ? AUDIO_DRAIN_EARLY_NOTIFY 2391 : AUDIO_DRAIN_ALL); 2392 } 2393} 2394 2395void AudioFlinger::PlaybackThread::threadLoop_exit() 2396{ 2397 { 2398 Mutex::Autolock _l(mLock); 2399 for (size_t i = 0; i < mTracks.size(); i++) { 2400 sp<Track> track = mTracks[i]; 2401 track->invalidate(); 2402 } 2403 } 2404} 2405 2406/* 2407The derived values that are cached: 2408 - mSinkBufferSize from frame count * frame size 2409 - activeSleepTime from activeSleepTimeUs() 2410 - idleSleepTime from idleSleepTimeUs() 2411 - standbyDelay from mActiveSleepTimeUs (DIRECT only) 2412 - maxPeriod from frame count and sample rate (MIXER only) 2413 2414The parameters that affect these derived values are: 2415 - frame count 2416 - frame size 2417 - sample rate 2418 - device type: A2DP or not 2419 - device latency 2420 - format: PCM or not 2421 - active sleep time 2422 - idle sleep time 2423*/ 2424 2425void AudioFlinger::PlaybackThread::cacheParameters_l() 2426{ 2427 mSinkBufferSize = mNormalFrameCount * mFrameSize; 2428 activeSleepTime = activeSleepTimeUs(); 2429 idleSleepTime = idleSleepTimeUs(); 2430} 2431 2432void AudioFlinger::PlaybackThread::invalidateTracks(audio_stream_type_t streamType) 2433{ 2434 ALOGV("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 2435 this, streamType, mTracks.size()); 2436 Mutex::Autolock _l(mLock); 2437 2438 size_t size = mTracks.size(); 2439 for (size_t i = 0; i < size; i++) { 2440 sp<Track> t = mTracks[i]; 2441 if (t->streamType() == streamType) { 2442 t->invalidate(); 2443 } 2444 } 2445} 2446 2447status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 2448{ 2449 int session = chain->sessionId(); 2450 int16_t* buffer = reinterpret_cast<int16_t*>(mEffectBufferEnabled 2451 ? mEffectBuffer : mSinkBuffer); 2452 bool ownsBuffer = false; 2453 2454 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 2455 if (session > 0) { 2456 // Only one effect chain can be present in direct output thread and it uses 2457 // the sink buffer as input 2458 if (mType != DIRECT) { 2459 size_t numSamples = mNormalFrameCount * mChannelCount; 2460 buffer = new int16_t[numSamples]; 2461 memset(buffer, 0, numSamples * sizeof(int16_t)); 2462 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 2463 ownsBuffer = true; 2464 } 2465 2466 // Attach all tracks with same session ID to this chain. 2467 for (size_t i = 0; i < mTracks.size(); ++i) { 2468 sp<Track> track = mTracks[i]; 2469 if (session == track->sessionId()) { 2470 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), 2471 buffer); 2472 track->setMainBuffer(buffer); 2473 chain->incTrackCnt(); 2474 } 2475 } 2476 2477 // indicate all active tracks in the chain 2478 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 2479 sp<Track> track = mActiveTracks[i].promote(); 2480 if (track == 0) { 2481 continue; 2482 } 2483 if (session == track->sessionId()) { 2484 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 2485 chain->incActiveTrackCnt(); 2486 } 2487 } 2488 } 2489 chain->setThread(this); 2490 chain->setInBuffer(buffer, ownsBuffer); 2491 chain->setOutBuffer(reinterpret_cast<int16_t*>(mEffectBufferEnabled 2492 ? mEffectBuffer : mSinkBuffer)); 2493 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 2494 // chains list in order to be processed last as it contains output stage effects 2495 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 2496 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 2497 // after track specific effects and before output stage 2498 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 2499 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 2500 // Effect chain for other sessions are inserted at beginning of effect 2501 // chains list to be processed before output mix effects. Relative order between other 2502 // sessions is not important 2503 size_t size = mEffectChains.size(); 2504 size_t i = 0; 2505 for (i = 0; i < size; i++) { 2506 if (mEffectChains[i]->sessionId() < session) { 2507 break; 2508 } 2509 } 2510 mEffectChains.insertAt(chain, i); 2511 checkSuspendOnAddEffectChain_l(chain); 2512 2513 return NO_ERROR; 2514} 2515 2516size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 2517{ 2518 int session = chain->sessionId(); 2519 2520 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 2521 2522 for (size_t i = 0; i < mEffectChains.size(); i++) { 2523 if (chain == mEffectChains[i]) { 2524 mEffectChains.removeAt(i); 2525 // detach all active tracks from the chain 2526 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 2527 sp<Track> track = mActiveTracks[i].promote(); 2528 if (track == 0) { 2529 continue; 2530 } 2531 if (session == track->sessionId()) { 2532 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 2533 chain.get(), session); 2534 chain->decActiveTrackCnt(); 2535 } 2536 } 2537 2538 // detach all tracks with same session ID from this chain 2539 for (size_t i = 0; i < mTracks.size(); ++i) { 2540 sp<Track> track = mTracks[i]; 2541 if (session == track->sessionId()) { 2542 track->setMainBuffer(reinterpret_cast<int16_t*>(mSinkBuffer)); 2543 chain->decTrackCnt(); 2544 } 2545 } 2546 break; 2547 } 2548 } 2549 return mEffectChains.size(); 2550} 2551 2552status_t AudioFlinger::PlaybackThread::attachAuxEffect( 2553 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2554{ 2555 Mutex::Autolock _l(mLock); 2556 return attachAuxEffect_l(track, EffectId); 2557} 2558 2559status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 2560 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2561{ 2562 status_t status = NO_ERROR; 2563 2564 if (EffectId == 0) { 2565 track->setAuxBuffer(0, NULL); 2566 } else { 2567 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 2568 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 2569 if (effect != 0) { 2570 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 2571 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 2572 } else { 2573 status = INVALID_OPERATION; 2574 } 2575 } else { 2576 status = BAD_VALUE; 2577 } 2578 } 2579 return status; 2580} 2581 2582void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 2583{ 2584 for (size_t i = 0; i < mTracks.size(); ++i) { 2585 sp<Track> track = mTracks[i]; 2586 if (track->auxEffectId() == effectId) { 2587 attachAuxEffect_l(track, 0); 2588 } 2589 } 2590} 2591 2592bool AudioFlinger::PlaybackThread::threadLoop() 2593{ 2594 Vector< sp<Track> > tracksToRemove; 2595 2596 standbyTime = systemTime(); 2597 2598 // MIXER 2599 nsecs_t lastWarning = 0; 2600 2601 // DUPLICATING 2602 // FIXME could this be made local to while loop? 2603 writeFrames = 0; 2604 2605 int lastGeneration = 0; 2606 2607 cacheParameters_l(); 2608 sleepTime = idleSleepTime; 2609 2610 if (mType == MIXER) { 2611 sleepTimeShift = 0; 2612 } 2613 2614 CpuStats cpuStats; 2615 const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid())); 2616 2617 acquireWakeLock(); 2618 2619 // mNBLogWriter->log can only be called while thread mutex mLock is held. 2620 // So if you need to log when mutex is unlocked, set logString to a non-NULL string, 2621 // and then that string will be logged at the next convenient opportunity. 2622 const char *logString = NULL; 2623 2624 checkSilentMode_l(); 2625 2626 while (!exitPending()) 2627 { 2628 cpuStats.sample(myName); 2629 2630 Vector< sp<EffectChain> > effectChains; 2631 2632 { // scope for mLock 2633 2634 Mutex::Autolock _l(mLock); 2635 2636 processConfigEvents_l(); 2637 2638 if (logString != NULL) { 2639 mNBLogWriter->logTimestamp(); 2640 mNBLogWriter->log(logString); 2641 logString = NULL; 2642 } 2643 2644 // Gather the framesReleased counters for all active tracks, 2645 // and latch them atomically with the timestamp. 2646 // FIXME We're using raw pointers as indices. A unique track ID would be a better index. 2647 mLatchD.mFramesReleased.clear(); 2648 size_t size = mActiveTracks.size(); 2649 for (size_t i = 0; i < size; i++) { 2650 sp<Track> t = mActiveTracks[i].promote(); 2651 if (t != 0) { 2652 mLatchD.mFramesReleased.add(t.get(), 2653 t->mAudioTrackServerProxy->framesReleased()); 2654 } 2655 } 2656 if (mLatchDValid) { 2657 mLatchQ = mLatchD; 2658 mLatchDValid = false; 2659 mLatchQValid = true; 2660 } 2661 2662 saveOutputTracks(); 2663 if (mSignalPending) { 2664 // A signal was raised while we were unlocked 2665 mSignalPending = false; 2666 } else if (waitingAsyncCallback_l()) { 2667 if (exitPending()) { 2668 break; 2669 } 2670 releaseWakeLock_l(); 2671 mWakeLockUids.clear(); 2672 mActiveTracksGeneration++; 2673 ALOGV("wait async completion"); 2674 mWaitWorkCV.wait(mLock); 2675 ALOGV("async completion/wake"); 2676 acquireWakeLock_l(); 2677 standbyTime = systemTime() + standbyDelay; 2678 sleepTime = 0; 2679 2680 continue; 2681 } 2682 if ((!mActiveTracks.size() && systemTime() > standbyTime) || 2683 isSuspended()) { 2684 // put audio hardware into standby after short delay 2685 if (shouldStandby_l()) { 2686 2687 threadLoop_standby(); 2688 2689 mStandby = true; 2690 } 2691 2692 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2693 // we're about to wait, flush the binder command buffer 2694 IPCThreadState::self()->flushCommands(); 2695 2696 clearOutputTracks(); 2697 2698 if (exitPending()) { 2699 break; 2700 } 2701 2702 releaseWakeLock_l(); 2703 mWakeLockUids.clear(); 2704 mActiveTracksGeneration++; 2705 // wait until we have something to do... 2706 ALOGV("%s going to sleep", myName.string()); 2707 mWaitWorkCV.wait(mLock); 2708 ALOGV("%s waking up", myName.string()); 2709 acquireWakeLock_l(); 2710 2711 mMixerStatus = MIXER_IDLE; 2712 mMixerStatusIgnoringFastTracks = MIXER_IDLE; 2713 mBytesWritten = 0; 2714 mBytesRemaining = 0; 2715 checkSilentMode_l(); 2716 2717 standbyTime = systemTime() + standbyDelay; 2718 sleepTime = idleSleepTime; 2719 if (mType == MIXER) { 2720 sleepTimeShift = 0; 2721 } 2722 2723 continue; 2724 } 2725 } 2726 // mMixerStatusIgnoringFastTracks is also updated internally 2727 mMixerStatus = prepareTracks_l(&tracksToRemove); 2728 2729 // compare with previously applied list 2730 if (lastGeneration != mActiveTracksGeneration) { 2731 // update wakelock 2732 updateWakeLockUids_l(mWakeLockUids); 2733 lastGeneration = mActiveTracksGeneration; 2734 } 2735 2736 // prevent any changes in effect chain list and in each effect chain 2737 // during mixing and effect process as the audio buffers could be deleted 2738 // or modified if an effect is created or deleted 2739 lockEffectChains_l(effectChains); 2740 } // mLock scope ends 2741 2742 if (mBytesRemaining == 0) { 2743 mCurrentWriteLength = 0; 2744 if (mMixerStatus == MIXER_TRACKS_READY) { 2745 // threadLoop_mix() sets mCurrentWriteLength 2746 threadLoop_mix(); 2747 } else if ((mMixerStatus != MIXER_DRAIN_TRACK) 2748 && (mMixerStatus != MIXER_DRAIN_ALL)) { 2749 // threadLoop_sleepTime sets sleepTime to 0 if data 2750 // must be written to HAL 2751 threadLoop_sleepTime(); 2752 if (sleepTime == 0) { 2753 mCurrentWriteLength = mSinkBufferSize; 2754 } 2755 } 2756 // Either threadLoop_mix() or threadLoop_sleepTime() should have set 2757 // mMixerBuffer with data if mMixerBufferValid is true and sleepTime == 0. 2758 // Merge mMixerBuffer data into mEffectBuffer (if any effects are valid) 2759 // or mSinkBuffer (if there are no effects). 2760 // 2761 // This is done pre-effects computation; if effects change to 2762 // support higher precision, this needs to move. 2763 // 2764 // mMixerBufferValid is only set true by MixerThread::prepareTracks_l(). 2765 // TODO use sleepTime == 0 as an additional condition. 2766 if (mMixerBufferValid) { 2767 void *buffer = mEffectBufferValid ? mEffectBuffer : mSinkBuffer; 2768 audio_format_t format = mEffectBufferValid ? mEffectBufferFormat : mFormat; 2769 2770 memcpy_by_audio_format(buffer, format, mMixerBuffer, mMixerBufferFormat, 2771 mNormalFrameCount * mChannelCount); 2772 } 2773 2774 mBytesRemaining = mCurrentWriteLength; 2775 if (isSuspended()) { 2776 sleepTime = suspendSleepTimeUs(); 2777 // simulate write to HAL when suspended 2778 mBytesWritten += mSinkBufferSize; 2779 mBytesRemaining = 0; 2780 } 2781 2782 // only process effects if we're going to write 2783 if (sleepTime == 0 && mType != OFFLOAD) { 2784 for (size_t i = 0; i < effectChains.size(); i ++) { 2785 effectChains[i]->process_l(); 2786 } 2787 } 2788 } 2789 // Process effect chains for offloaded thread even if no audio 2790 // was read from audio track: process only updates effect state 2791 // and thus does have to be synchronized with audio writes but may have 2792 // to be called while waiting for async write callback 2793 if (mType == OFFLOAD) { 2794 for (size_t i = 0; i < effectChains.size(); i ++) { 2795 effectChains[i]->process_l(); 2796 } 2797 } 2798 2799 // Only if the Effects buffer is enabled and there is data in the 2800 // Effects buffer (buffer valid), we need to 2801 // copy into the sink buffer. 2802 // TODO use sleepTime == 0 as an additional condition. 2803 if (mEffectBufferValid) { 2804 //ALOGV("writing effect buffer to sink buffer format %#x", mFormat); 2805 memcpy_by_audio_format(mSinkBuffer, mFormat, mEffectBuffer, mEffectBufferFormat, 2806 mNormalFrameCount * mChannelCount); 2807 } 2808 2809 // enable changes in effect chain 2810 unlockEffectChains(effectChains); 2811 2812 if (!waitingAsyncCallback()) { 2813 // sleepTime == 0 means we must write to audio hardware 2814 if (sleepTime == 0) { 2815 if (mBytesRemaining) { 2816 ssize_t ret = threadLoop_write(); 2817 if (ret < 0) { 2818 mBytesRemaining = 0; 2819 } else { 2820 mBytesWritten += ret; 2821 mBytesRemaining -= ret; 2822 } 2823 } else if ((mMixerStatus == MIXER_DRAIN_TRACK) || 2824 (mMixerStatus == MIXER_DRAIN_ALL)) { 2825 threadLoop_drain(); 2826 } 2827 if (mType == MIXER) { 2828 // write blocked detection 2829 nsecs_t now = systemTime(); 2830 nsecs_t delta = now - mLastWriteTime; 2831 if (!mStandby && delta > maxPeriod) { 2832 mNumDelayedWrites++; 2833 if ((now - lastWarning) > kWarningThrottleNs) { 2834 ATRACE_NAME("underrun"); 2835 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2836 ns2ms(delta), mNumDelayedWrites, this); 2837 lastWarning = now; 2838 } 2839 } 2840 } 2841 2842 } else { 2843 ATRACE_BEGIN("sleep"); 2844 usleep(sleepTime); 2845 ATRACE_END(); 2846 } 2847 } 2848 2849 // Finally let go of removed track(s), without the lock held 2850 // since we can't guarantee the destructors won't acquire that 2851 // same lock. This will also mutate and push a new fast mixer state. 2852 threadLoop_removeTracks(tracksToRemove); 2853 tracksToRemove.clear(); 2854 2855 // FIXME I don't understand the need for this here; 2856 // it was in the original code but maybe the 2857 // assignment in saveOutputTracks() makes this unnecessary? 2858 clearOutputTracks(); 2859 2860 // Effect chains will be actually deleted here if they were removed from 2861 // mEffectChains list during mixing or effects processing 2862 effectChains.clear(); 2863 2864 // FIXME Note that the above .clear() is no longer necessary since effectChains 2865 // is now local to this block, but will keep it for now (at least until merge done). 2866 } 2867 2868 threadLoop_exit(); 2869 2870 if (!mStandby) { 2871 threadLoop_standby(); 2872 mStandby = true; 2873 } 2874 2875 releaseWakeLock(); 2876 mWakeLockUids.clear(); 2877 mActiveTracksGeneration++; 2878 2879 ALOGV("Thread %p type %d exiting", this, mType); 2880 return false; 2881} 2882 2883// removeTracks_l() must be called with ThreadBase::mLock held 2884void AudioFlinger::PlaybackThread::removeTracks_l(const Vector< sp<Track> >& tracksToRemove) 2885{ 2886 size_t count = tracksToRemove.size(); 2887 if (count > 0) { 2888 for (size_t i=0 ; i<count ; i++) { 2889 const sp<Track>& track = tracksToRemove.itemAt(i); 2890 mActiveTracks.remove(track); 2891 mWakeLockUids.remove(track->uid()); 2892 mActiveTracksGeneration++; 2893 ALOGV("removeTracks_l removing track on session %d", track->sessionId()); 2894 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 2895 if (chain != 0) { 2896 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), 2897 track->sessionId()); 2898 chain->decActiveTrackCnt(); 2899 } 2900 if (track->isTerminated()) { 2901 removeTrack_l(track); 2902 } 2903 } 2904 } 2905 2906} 2907 2908status_t AudioFlinger::PlaybackThread::getTimestamp_l(AudioTimestamp& timestamp) 2909{ 2910 if (mNormalSink != 0) { 2911 return mNormalSink->getTimestamp(timestamp); 2912 } 2913 if ((mType == OFFLOAD || mType == DIRECT) 2914 && mOutput != NULL && mOutput->stream->get_presentation_position) { 2915 uint64_t position64; 2916 int ret = mOutput->getPresentationPosition(&position64, ×tamp.mTime); 2917 if (ret == 0) { 2918 timestamp.mPosition = (uint32_t)position64; 2919 return NO_ERROR; 2920 } 2921 } 2922 return INVALID_OPERATION; 2923} 2924 2925status_t AudioFlinger::PlaybackThread::createAudioPatch_l(const struct audio_patch *patch, 2926 audio_patch_handle_t *handle) 2927{ 2928 status_t status = NO_ERROR; 2929 if (mOutput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 2930 // store new device and send to effects 2931 audio_devices_t type = AUDIO_DEVICE_NONE; 2932 for (unsigned int i = 0; i < patch->num_sinks; i++) { 2933 type |= patch->sinks[i].ext.device.type; 2934 } 2935 mOutDevice = type; 2936 for (size_t i = 0; i < mEffectChains.size(); i++) { 2937 mEffectChains[i]->setDevice_l(mOutDevice); 2938 } 2939 2940 audio_hw_device_t *hwDevice = mOutput->audioHwDev->hwDevice(); 2941 status = hwDevice->create_audio_patch(hwDevice, 2942 patch->num_sources, 2943 patch->sources, 2944 patch->num_sinks, 2945 patch->sinks, 2946 handle); 2947 } else { 2948 ALOG_ASSERT(false, "createAudioPatch_l() called on a pre 3.0 HAL"); 2949 } 2950 return status; 2951} 2952 2953status_t AudioFlinger::PlaybackThread::releaseAudioPatch_l(const audio_patch_handle_t handle) 2954{ 2955 status_t status = NO_ERROR; 2956 if (mOutput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 2957 audio_hw_device_t *hwDevice = mOutput->audioHwDev->hwDevice(); 2958 status = hwDevice->release_audio_patch(hwDevice, handle); 2959 } else { 2960 ALOG_ASSERT(false, "releaseAudioPatch_l() called on a pre 3.0 HAL"); 2961 } 2962 return status; 2963} 2964 2965void AudioFlinger::PlaybackThread::addPatchTrack(const sp<PatchTrack>& track) 2966{ 2967 Mutex::Autolock _l(mLock); 2968 mTracks.add(track); 2969} 2970 2971void AudioFlinger::PlaybackThread::deletePatchTrack(const sp<PatchTrack>& track) 2972{ 2973 Mutex::Autolock _l(mLock); 2974 destroyTrack_l(track); 2975} 2976 2977void AudioFlinger::PlaybackThread::getAudioPortConfig(struct audio_port_config *config) 2978{ 2979 ThreadBase::getAudioPortConfig(config); 2980 config->role = AUDIO_PORT_ROLE_SOURCE; 2981 config->ext.mix.hw_module = mOutput->audioHwDev->handle(); 2982 config->ext.mix.usecase.stream = AUDIO_STREAM_DEFAULT; 2983} 2984 2985// ---------------------------------------------------------------------------- 2986 2987AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 2988 audio_io_handle_t id, audio_devices_t device, type_t type) 2989 : PlaybackThread(audioFlinger, output, id, device, type), 2990 // mAudioMixer below 2991 // mFastMixer below 2992 mFastMixerFutex(0) 2993 // mOutputSink below 2994 // mPipeSink below 2995 // mNormalSink below 2996{ 2997 ALOGV("MixerThread() id=%d device=%#x type=%d", id, device, type); 2998 ALOGV("mSampleRate=%u, mChannelMask=%#x, mChannelCount=%u, mFormat=%d, mFrameSize=%u, " 2999 "mFrameCount=%d, mNormalFrameCount=%d", 3000 mSampleRate, mChannelMask, mChannelCount, mFormat, mFrameSize, mFrameCount, 3001 mNormalFrameCount); 3002 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 3003 3004 if (type == DUPLICATING) { 3005 // The Duplicating thread uses the AudioMixer and delivers data to OutputTracks 3006 // (downstream MixerThreads) in DuplicatingThread::threadLoop_write(). 3007 // Do not create or use mFastMixer, mOutputSink, mPipeSink, or mNormalSink. 3008 return; 3009 } 3010 // create an NBAIO sink for the HAL output stream, and negotiate 3011 mOutputSink = new AudioStreamOutSink(output->stream); 3012 size_t numCounterOffers = 0; 3013 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount, mFormat)}; 3014 ssize_t index = mOutputSink->negotiate(offers, 1, NULL, numCounterOffers); 3015 ALOG_ASSERT(index == 0); 3016 3017 // initialize fast mixer depending on configuration 3018 bool initFastMixer; 3019 switch (kUseFastMixer) { 3020 case FastMixer_Never: 3021 initFastMixer = false; 3022 break; 3023 case FastMixer_Always: 3024 initFastMixer = true; 3025 break; 3026 case FastMixer_Static: 3027 case FastMixer_Dynamic: 3028 initFastMixer = mFrameCount < mNormalFrameCount; 3029 break; 3030 } 3031 if (initFastMixer) { 3032 audio_format_t fastMixerFormat; 3033 if (mMixerBufferEnabled && mEffectBufferEnabled) { 3034 fastMixerFormat = AUDIO_FORMAT_PCM_FLOAT; 3035 } else { 3036 fastMixerFormat = AUDIO_FORMAT_PCM_16_BIT; 3037 } 3038 if (mFormat != fastMixerFormat) { 3039 // change our Sink format to accept our intermediate precision 3040 mFormat = fastMixerFormat; 3041 free(mSinkBuffer); 3042 mFrameSize = mChannelCount * audio_bytes_per_sample(mFormat); 3043 const size_t sinkBufferSize = mNormalFrameCount * mFrameSize; 3044 (void)posix_memalign(&mSinkBuffer, 32, sinkBufferSize); 3045 } 3046 3047 // create a MonoPipe to connect our submix to FastMixer 3048 NBAIO_Format format = mOutputSink->format(); 3049 NBAIO_Format origformat = format; 3050 // adjust format to match that of the Fast Mixer 3051 ALOGV("format changed from %d to %d", format.mFormat, fastMixerFormat); 3052 format.mFormat = fastMixerFormat; 3053 format.mFrameSize = audio_bytes_per_sample(format.mFormat) * format.mChannelCount; 3054 3055 // This pipe depth compensates for scheduling latency of the normal mixer thread. 3056 // When it wakes up after a maximum latency, it runs a few cycles quickly before 3057 // finally blocking. Note the pipe implementation rounds up the request to a power of 2. 3058 MonoPipe *monoPipe = new MonoPipe(mNormalFrameCount * 4, format, true /*writeCanBlock*/); 3059 const NBAIO_Format offers[1] = {format}; 3060 size_t numCounterOffers = 0; 3061 ssize_t index = monoPipe->negotiate(offers, 1, NULL, numCounterOffers); 3062 ALOG_ASSERT(index == 0); 3063 monoPipe->setAvgFrames((mScreenState & 1) ? 3064 (monoPipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 3065 mPipeSink = monoPipe; 3066 3067#ifdef TEE_SINK 3068 if (mTeeSinkOutputEnabled) { 3069 // create a Pipe to archive a copy of FastMixer's output for dumpsys 3070 Pipe *teeSink = new Pipe(mTeeSinkOutputFrames, origformat); 3071 const NBAIO_Format offers2[1] = {origformat}; 3072 numCounterOffers = 0; 3073 index = teeSink->negotiate(offers2, 1, NULL, numCounterOffers); 3074 ALOG_ASSERT(index == 0); 3075 mTeeSink = teeSink; 3076 PipeReader *teeSource = new PipeReader(*teeSink); 3077 numCounterOffers = 0; 3078 index = teeSource->negotiate(offers2, 1, NULL, numCounterOffers); 3079 ALOG_ASSERT(index == 0); 3080 mTeeSource = teeSource; 3081 } 3082#endif 3083 3084 // create fast mixer and configure it initially with just one fast track for our submix 3085 mFastMixer = new FastMixer(); 3086 FastMixerStateQueue *sq = mFastMixer->sq(); 3087#ifdef STATE_QUEUE_DUMP 3088 sq->setObserverDump(&mStateQueueObserverDump); 3089 sq->setMutatorDump(&mStateQueueMutatorDump); 3090#endif 3091 FastMixerState *state = sq->begin(); 3092 FastTrack *fastTrack = &state->mFastTracks[0]; 3093 // wrap the source side of the MonoPipe to make it an AudioBufferProvider 3094 fastTrack->mBufferProvider = new SourceAudioBufferProvider(new MonoPipeReader(monoPipe)); 3095 fastTrack->mVolumeProvider = NULL; 3096 fastTrack->mChannelMask = mChannelMask; // mPipeSink channel mask for audio to FastMixer 3097 fastTrack->mFormat = mFormat; // mPipeSink format for audio to FastMixer 3098 fastTrack->mGeneration++; 3099 state->mFastTracksGen++; 3100 state->mTrackMask = 1; 3101 // fast mixer will use the HAL output sink 3102 state->mOutputSink = mOutputSink.get(); 3103 state->mOutputSinkGen++; 3104 state->mFrameCount = mFrameCount; 3105 state->mCommand = FastMixerState::COLD_IDLE; 3106 // already done in constructor initialization list 3107 //mFastMixerFutex = 0; 3108 state->mColdFutexAddr = &mFastMixerFutex; 3109 state->mColdGen++; 3110 state->mDumpState = &mFastMixerDumpState; 3111#ifdef TEE_SINK 3112 state->mTeeSink = mTeeSink.get(); 3113#endif 3114 mFastMixerNBLogWriter = audioFlinger->newWriter_l(kFastMixerLogSize, "FastMixer"); 3115 state->mNBLogWriter = mFastMixerNBLogWriter.get(); 3116 sq->end(); 3117 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3118 3119 // start the fast mixer 3120 mFastMixer->run("FastMixer", PRIORITY_URGENT_AUDIO); 3121 pid_t tid = mFastMixer->getTid(); 3122 int err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 3123 if (err != 0) { 3124 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 3125 kPriorityFastMixer, getpid_cached, tid, err); 3126 } 3127 3128#ifdef AUDIO_WATCHDOG 3129 // create and start the watchdog 3130 mAudioWatchdog = new AudioWatchdog(); 3131 mAudioWatchdog->setDump(&mAudioWatchdogDump); 3132 mAudioWatchdog->run("AudioWatchdog", PRIORITY_URGENT_AUDIO); 3133 tid = mAudioWatchdog->getTid(); 3134 err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 3135 if (err != 0) { 3136 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 3137 kPriorityFastMixer, getpid_cached, tid, err); 3138 } 3139#endif 3140 3141 } 3142 3143 switch (kUseFastMixer) { 3144 case FastMixer_Never: 3145 case FastMixer_Dynamic: 3146 mNormalSink = mOutputSink; 3147 break; 3148 case FastMixer_Always: 3149 mNormalSink = mPipeSink; 3150 break; 3151 case FastMixer_Static: 3152 mNormalSink = initFastMixer ? mPipeSink : mOutputSink; 3153 break; 3154 } 3155} 3156 3157AudioFlinger::MixerThread::~MixerThread() 3158{ 3159 if (mFastMixer != 0) { 3160 FastMixerStateQueue *sq = mFastMixer->sq(); 3161 FastMixerState *state = sq->begin(); 3162 if (state->mCommand == FastMixerState::COLD_IDLE) { 3163 int32_t old = android_atomic_inc(&mFastMixerFutex); 3164 if (old == -1) { 3165 (void) syscall(__NR_futex, &mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 3166 } 3167 } 3168 state->mCommand = FastMixerState::EXIT; 3169 sq->end(); 3170 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3171 mFastMixer->join(); 3172 // Though the fast mixer thread has exited, it's state queue is still valid. 3173 // We'll use that extract the final state which contains one remaining fast track 3174 // corresponding to our sub-mix. 3175 state = sq->begin(); 3176 ALOG_ASSERT(state->mTrackMask == 1); 3177 FastTrack *fastTrack = &state->mFastTracks[0]; 3178 ALOG_ASSERT(fastTrack->mBufferProvider != NULL); 3179 delete fastTrack->mBufferProvider; 3180 sq->end(false /*didModify*/); 3181 mFastMixer.clear(); 3182#ifdef AUDIO_WATCHDOG 3183 if (mAudioWatchdog != 0) { 3184 mAudioWatchdog->requestExit(); 3185 mAudioWatchdog->requestExitAndWait(); 3186 mAudioWatchdog.clear(); 3187 } 3188#endif 3189 } 3190 mAudioFlinger->unregisterWriter(mFastMixerNBLogWriter); 3191 delete mAudioMixer; 3192} 3193 3194 3195uint32_t AudioFlinger::MixerThread::correctLatency_l(uint32_t latency) const 3196{ 3197 if (mFastMixer != 0) { 3198 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 3199 latency += (pipe->getAvgFrames() * 1000) / mSampleRate; 3200 } 3201 return latency; 3202} 3203 3204 3205void AudioFlinger::MixerThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) 3206{ 3207 PlaybackThread::threadLoop_removeTracks(tracksToRemove); 3208} 3209 3210ssize_t AudioFlinger::MixerThread::threadLoop_write() 3211{ 3212 // FIXME we should only do one push per cycle; confirm this is true 3213 // Start the fast mixer if it's not already running 3214 if (mFastMixer != 0) { 3215 FastMixerStateQueue *sq = mFastMixer->sq(); 3216 FastMixerState *state = sq->begin(); 3217 if (state->mCommand != FastMixerState::MIX_WRITE && 3218 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)) { 3219 if (state->mCommand == FastMixerState::COLD_IDLE) { 3220 int32_t old = android_atomic_inc(&mFastMixerFutex); 3221 if (old == -1) { 3222 (void) syscall(__NR_futex, &mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 3223 } 3224#ifdef AUDIO_WATCHDOG 3225 if (mAudioWatchdog != 0) { 3226 mAudioWatchdog->resume(); 3227 } 3228#endif 3229 } 3230 state->mCommand = FastMixerState::MIX_WRITE; 3231#ifdef FAST_THREAD_STATISTICS 3232 mFastMixerDumpState.increaseSamplingN(mAudioFlinger->isLowRamDevice() ? 3233 FastThreadDumpState::kSamplingNforLowRamDevice : FastThreadDumpState::kSamplingN); 3234#endif 3235 sq->end(); 3236 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3237 if (kUseFastMixer == FastMixer_Dynamic) { 3238 mNormalSink = mPipeSink; 3239 } 3240 } else { 3241 sq->end(false /*didModify*/); 3242 } 3243 } 3244 return PlaybackThread::threadLoop_write(); 3245} 3246 3247void AudioFlinger::MixerThread::threadLoop_standby() 3248{ 3249 // Idle the fast mixer if it's currently running 3250 if (mFastMixer != 0) { 3251 FastMixerStateQueue *sq = mFastMixer->sq(); 3252 FastMixerState *state = sq->begin(); 3253 if (!(state->mCommand & FastMixerState::IDLE)) { 3254 state->mCommand = FastMixerState::COLD_IDLE; 3255 state->mColdFutexAddr = &mFastMixerFutex; 3256 state->mColdGen++; 3257 mFastMixerFutex = 0; 3258 sq->end(); 3259 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 3260 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3261 if (kUseFastMixer == FastMixer_Dynamic) { 3262 mNormalSink = mOutputSink; 3263 } 3264#ifdef AUDIO_WATCHDOG 3265 if (mAudioWatchdog != 0) { 3266 mAudioWatchdog->pause(); 3267 } 3268#endif 3269 } else { 3270 sq->end(false /*didModify*/); 3271 } 3272 } 3273 PlaybackThread::threadLoop_standby(); 3274} 3275 3276bool AudioFlinger::PlaybackThread::waitingAsyncCallback_l() 3277{ 3278 return false; 3279} 3280 3281bool AudioFlinger::PlaybackThread::shouldStandby_l() 3282{ 3283 return !mStandby; 3284} 3285 3286bool AudioFlinger::PlaybackThread::waitingAsyncCallback() 3287{ 3288 Mutex::Autolock _l(mLock); 3289 return waitingAsyncCallback_l(); 3290} 3291 3292// shared by MIXER and DIRECT, overridden by DUPLICATING 3293void AudioFlinger::PlaybackThread::threadLoop_standby() 3294{ 3295 ALOGV("Audio hardware entering standby, mixer %p, suspend count %d", this, mSuspended); 3296 mOutput->standby(); 3297 if (mUseAsyncWrite != 0) { 3298 // discard any pending drain or write ack by incrementing sequence 3299 mWriteAckSequence = (mWriteAckSequence + 2) & ~1; 3300 mDrainSequence = (mDrainSequence + 2) & ~1; 3301 ALOG_ASSERT(mCallbackThread != 0); 3302 mCallbackThread->setWriteBlocked(mWriteAckSequence); 3303 mCallbackThread->setDraining(mDrainSequence); 3304 } 3305 mHwPaused = false; 3306} 3307 3308void AudioFlinger::PlaybackThread::onAddNewTrack_l() 3309{ 3310 ALOGV("signal playback thread"); 3311 broadcast_l(); 3312} 3313 3314void AudioFlinger::MixerThread::threadLoop_mix() 3315{ 3316 // obtain the presentation timestamp of the next output buffer 3317 int64_t pts; 3318 status_t status = INVALID_OPERATION; 3319 3320 if (mNormalSink != 0) { 3321 status = mNormalSink->getNextWriteTimestamp(&pts); 3322 } else { 3323 status = mOutputSink->getNextWriteTimestamp(&pts); 3324 } 3325 3326 if (status != NO_ERROR) { 3327 pts = AudioBufferProvider::kInvalidPTS; 3328 } 3329 3330 // mix buffers... 3331 mAudioMixer->process(pts); 3332 mCurrentWriteLength = mSinkBufferSize; 3333 // increase sleep time progressively when application underrun condition clears. 3334 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 3335 // that a steady state of alternating ready/not ready conditions keeps the sleep time 3336 // such that we would underrun the audio HAL. 3337 if ((sleepTime == 0) && (sleepTimeShift > 0)) { 3338 sleepTimeShift--; 3339 } 3340 sleepTime = 0; 3341 standbyTime = systemTime() + standbyDelay; 3342 //TODO: delay standby when effects have a tail 3343 3344} 3345 3346void AudioFlinger::MixerThread::threadLoop_sleepTime() 3347{ 3348 // If no tracks are ready, sleep once for the duration of an output 3349 // buffer size, then write 0s to the output 3350 if (sleepTime == 0) { 3351 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3352 sleepTime = activeSleepTime >> sleepTimeShift; 3353 if (sleepTime < kMinThreadSleepTimeUs) { 3354 sleepTime = kMinThreadSleepTimeUs; 3355 } 3356 // reduce sleep time in case of consecutive application underruns to avoid 3357 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 3358 // duration we would end up writing less data than needed by the audio HAL if 3359 // the condition persists. 3360 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 3361 sleepTimeShift++; 3362 } 3363 } else { 3364 sleepTime = idleSleepTime; 3365 } 3366 } else if (mBytesWritten != 0 || (mMixerStatus == MIXER_TRACKS_ENABLED)) { 3367 // clear out mMixerBuffer or mSinkBuffer, to ensure buffers are cleared 3368 // before effects processing or output. 3369 if (mMixerBufferValid) { 3370 memset(mMixerBuffer, 0, mMixerBufferSize); 3371 } else { 3372 memset(mSinkBuffer, 0, mSinkBufferSize); 3373 } 3374 sleepTime = 0; 3375 ALOGV_IF(mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED), 3376 "anticipated start"); 3377 } 3378 // TODO add standby time extension fct of effect tail 3379} 3380 3381// prepareTracks_l() must be called with ThreadBase::mLock held 3382AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 3383 Vector< sp<Track> > *tracksToRemove) 3384{ 3385 3386 mixer_state mixerStatus = MIXER_IDLE; 3387 // find out which tracks need to be processed 3388 size_t count = mActiveTracks.size(); 3389 size_t mixedTracks = 0; 3390 size_t tracksWithEffect = 0; 3391 // counts only _active_ fast tracks 3392 size_t fastTracks = 0; 3393 uint32_t resetMask = 0; // bit mask of fast tracks that need to be reset 3394 3395 float masterVolume = mMasterVolume; 3396 bool masterMute = mMasterMute; 3397 3398 if (masterMute) { 3399 masterVolume = 0; 3400 } 3401 // Delegate master volume control to effect in output mix effect chain if needed 3402 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 3403 if (chain != 0) { 3404 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 3405 chain->setVolume_l(&v, &v); 3406 masterVolume = (float)((v + (1 << 23)) >> 24); 3407 chain.clear(); 3408 } 3409 3410 // prepare a new state to push 3411 FastMixerStateQueue *sq = NULL; 3412 FastMixerState *state = NULL; 3413 bool didModify = false; 3414 FastMixerStateQueue::block_t block = FastMixerStateQueue::BLOCK_UNTIL_PUSHED; 3415 if (mFastMixer != 0) { 3416 sq = mFastMixer->sq(); 3417 state = sq->begin(); 3418 } 3419 3420 mMixerBufferValid = false; // mMixerBuffer has no valid data until appropriate tracks found. 3421 mEffectBufferValid = false; // mEffectBuffer has no valid data until tracks found. 3422 3423 for (size_t i=0 ; i<count ; i++) { 3424 const sp<Track> t = mActiveTracks[i].promote(); 3425 if (t == 0) { 3426 continue; 3427 } 3428 3429 // this const just means the local variable doesn't change 3430 Track* const track = t.get(); 3431 3432 // process fast tracks 3433 if (track->isFastTrack()) { 3434 3435 // It's theoretically possible (though unlikely) for a fast track to be created 3436 // and then removed within the same normal mix cycle. This is not a problem, as 3437 // the track never becomes active so it's fast mixer slot is never touched. 3438 // The converse, of removing an (active) track and then creating a new track 3439 // at the identical fast mixer slot within the same normal mix cycle, 3440 // is impossible because the slot isn't marked available until the end of each cycle. 3441 int j = track->mFastIndex; 3442 ALOG_ASSERT(0 < j && j < (int)FastMixerState::kMaxFastTracks); 3443 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << j))); 3444 FastTrack *fastTrack = &state->mFastTracks[j]; 3445 3446 // Determine whether the track is currently in underrun condition, 3447 // and whether it had a recent underrun. 3448 FastTrackDump *ftDump = &mFastMixerDumpState.mTracks[j]; 3449 FastTrackUnderruns underruns = ftDump->mUnderruns; 3450 uint32_t recentFull = (underruns.mBitFields.mFull - 3451 track->mObservedUnderruns.mBitFields.mFull) & UNDERRUN_MASK; 3452 uint32_t recentPartial = (underruns.mBitFields.mPartial - 3453 track->mObservedUnderruns.mBitFields.mPartial) & UNDERRUN_MASK; 3454 uint32_t recentEmpty = (underruns.mBitFields.mEmpty - 3455 track->mObservedUnderruns.mBitFields.mEmpty) & UNDERRUN_MASK; 3456 uint32_t recentUnderruns = recentPartial + recentEmpty; 3457 track->mObservedUnderruns = underruns; 3458 // don't count underruns that occur while stopping or pausing 3459 // or stopped which can occur when flush() is called while active 3460 if (!(track->isStopping() || track->isPausing() || track->isStopped()) && 3461 recentUnderruns > 0) { 3462 // FIXME fast mixer will pull & mix partial buffers, but we count as a full underrun 3463 track->mAudioTrackServerProxy->tallyUnderrunFrames(recentUnderruns * mFrameCount); 3464 } 3465 3466 // This is similar to the state machine for normal tracks, 3467 // with a few modifications for fast tracks. 3468 bool isActive = true; 3469 switch (track->mState) { 3470 case TrackBase::STOPPING_1: 3471 // track stays active in STOPPING_1 state until first underrun 3472 if (recentUnderruns > 0 || track->isTerminated()) { 3473 track->mState = TrackBase::STOPPING_2; 3474 } 3475 break; 3476 case TrackBase::PAUSING: 3477 // ramp down is not yet implemented 3478 track->setPaused(); 3479 break; 3480 case TrackBase::RESUMING: 3481 // ramp up is not yet implemented 3482 track->mState = TrackBase::ACTIVE; 3483 break; 3484 case TrackBase::ACTIVE: 3485 if (recentFull > 0 || recentPartial > 0) { 3486 // track has provided at least some frames recently: reset retry count 3487 track->mRetryCount = kMaxTrackRetries; 3488 } 3489 if (recentUnderruns == 0) { 3490 // no recent underruns: stay active 3491 break; 3492 } 3493 // there has recently been an underrun of some kind 3494 if (track->sharedBuffer() == 0) { 3495 // were any of the recent underruns "empty" (no frames available)? 3496 if (recentEmpty == 0) { 3497 // no, then ignore the partial underruns as they are allowed indefinitely 3498 break; 3499 } 3500 // there has recently been an "empty" underrun: decrement the retry counter 3501 if (--(track->mRetryCount) > 0) { 3502 break; 3503 } 3504 // indicate to client process that the track was disabled because of underrun; 3505 // it will then automatically call start() when data is available 3506 android_atomic_or(CBLK_DISABLED, &track->mCblk->mFlags); 3507 // remove from active list, but state remains ACTIVE [confusing but true] 3508 isActive = false; 3509 break; 3510 } 3511 // fall through 3512 case TrackBase::STOPPING_2: 3513 case TrackBase::PAUSED: 3514 case TrackBase::STOPPED: 3515 case TrackBase::FLUSHED: // flush() while active 3516 // Check for presentation complete if track is inactive 3517 // We have consumed all the buffers of this track. 3518 // This would be incomplete if we auto-paused on underrun 3519 { 3520 size_t audioHALFrames = 3521 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 3522 size_t framesWritten = mBytesWritten / mFrameSize; 3523 if (!(mStandby || track->presentationComplete(framesWritten, audioHALFrames))) { 3524 // track stays in active list until presentation is complete 3525 break; 3526 } 3527 } 3528 if (track->isStopping_2()) { 3529 track->mState = TrackBase::STOPPED; 3530 } 3531 if (track->isStopped()) { 3532 // Can't reset directly, as fast mixer is still polling this track 3533 // track->reset(); 3534 // So instead mark this track as needing to be reset after push with ack 3535 resetMask |= 1 << i; 3536 } 3537 isActive = false; 3538 break; 3539 case TrackBase::IDLE: 3540 default: 3541 LOG_ALWAYS_FATAL("unexpected track state %d", track->mState); 3542 } 3543 3544 if (isActive) { 3545 // was it previously inactive? 3546 if (!(state->mTrackMask & (1 << j))) { 3547 ExtendedAudioBufferProvider *eabp = track; 3548 VolumeProvider *vp = track; 3549 fastTrack->mBufferProvider = eabp; 3550 fastTrack->mVolumeProvider = vp; 3551 fastTrack->mChannelMask = track->mChannelMask; 3552 fastTrack->mFormat = track->mFormat; 3553 fastTrack->mGeneration++; 3554 state->mTrackMask |= 1 << j; 3555 didModify = true; 3556 // no acknowledgement required for newly active tracks 3557 } 3558 // cache the combined master volume and stream type volume for fast mixer; this 3559 // lacks any synchronization or barrier so VolumeProvider may read a stale value 3560 track->mCachedVolume = masterVolume * mStreamTypes[track->streamType()].volume; 3561 ++fastTracks; 3562 } else { 3563 // was it previously active? 3564 if (state->mTrackMask & (1 << j)) { 3565 fastTrack->mBufferProvider = NULL; 3566 fastTrack->mGeneration++; 3567 state->mTrackMask &= ~(1 << j); 3568 didModify = true; 3569 // If any fast tracks were removed, we must wait for acknowledgement 3570 // because we're about to decrement the last sp<> on those tracks. 3571 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3572 } else { 3573 LOG_ALWAYS_FATAL("fast track %d should have been active", j); 3574 } 3575 tracksToRemove->add(track); 3576 // Avoids a misleading display in dumpsys 3577 track->mObservedUnderruns.mBitFields.mMostRecent = UNDERRUN_FULL; 3578 } 3579 continue; 3580 } 3581 3582 { // local variable scope to avoid goto warning 3583 3584 audio_track_cblk_t* cblk = track->cblk(); 3585 3586 // The first time a track is added we wait 3587 // for all its buffers to be filled before processing it 3588 int name = track->name(); 3589 // make sure that we have enough frames to mix one full buffer. 3590 // enforce this condition only once to enable draining the buffer in case the client 3591 // app does not call stop() and relies on underrun to stop: 3592 // hence the test on (mMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 3593 // during last round 3594 size_t desiredFrames; 3595 uint32_t sr = track->sampleRate(); 3596 if (sr == mSampleRate) { 3597 desiredFrames = mNormalFrameCount; 3598 } else { 3599 desiredFrames = sourceFramesNeeded(sr, mNormalFrameCount, mSampleRate); 3600 // add frames already consumed but not yet released by the resampler 3601 // because mAudioTrackServerProxy->framesReady() will include these frames 3602 desiredFrames += mAudioMixer->getUnreleasedFrames(track->name()); 3603#if 0 3604 // the minimum track buffer size is normally twice the number of frames necessary 3605 // to fill one buffer and the resampler should not leave more than one buffer worth 3606 // of unreleased frames after each pass, but just in case... 3607 ALOG_ASSERT(desiredFrames <= cblk->frameCount_); 3608#endif 3609 } 3610 uint32_t minFrames = 1; 3611 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() && 3612 (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) { 3613 minFrames = desiredFrames; 3614 } 3615 3616 size_t framesReady = track->framesReady(); 3617 if (ATRACE_ENABLED()) { 3618 // I wish we had formatted trace names 3619 char traceName[16]; 3620 strcpy(traceName, "nRdy"); 3621 int name = track->name(); 3622 if (AudioMixer::TRACK0 <= name && 3623 name < (int) (AudioMixer::TRACK0 + AudioMixer::MAX_NUM_TRACKS)) { 3624 name -= AudioMixer::TRACK0; 3625 traceName[4] = (name / 10) + '0'; 3626 traceName[5] = (name % 10) + '0'; 3627 } else { 3628 traceName[4] = '?'; 3629 traceName[5] = '?'; 3630 } 3631 traceName[6] = '\0'; 3632 ATRACE_INT(traceName, framesReady); 3633 } 3634 if ((framesReady >= minFrames) && track->isReady() && 3635 !track->isPaused() && !track->isTerminated()) 3636 { 3637 ALOGVV("track %d s=%08x [OK] on thread %p", name, cblk->mServer, this); 3638 3639 mixedTracks++; 3640 3641 // track->mainBuffer() != mSinkBuffer or mMixerBuffer means 3642 // there is an effect chain connected to the track 3643 chain.clear(); 3644 if (track->mainBuffer() != mSinkBuffer && 3645 track->mainBuffer() != mMixerBuffer) { 3646 if (mEffectBufferEnabled) { 3647 mEffectBufferValid = true; // Later can set directly. 3648 } 3649 chain = getEffectChain_l(track->sessionId()); 3650 // Delegate volume control to effect in track effect chain if needed 3651 if (chain != 0) { 3652 tracksWithEffect++; 3653 } else { 3654 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on " 3655 "session %d", 3656 name, track->sessionId()); 3657 } 3658 } 3659 3660 3661 int param = AudioMixer::VOLUME; 3662 if (track->mFillingUpStatus == Track::FS_FILLED) { 3663 // no ramp for the first volume setting 3664 track->mFillingUpStatus = Track::FS_ACTIVE; 3665 if (track->mState == TrackBase::RESUMING) { 3666 track->mState = TrackBase::ACTIVE; 3667 param = AudioMixer::RAMP_VOLUME; 3668 } 3669 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 3670 // FIXME should not make a decision based on mServer 3671 } else if (cblk->mServer != 0) { 3672 // If the track is stopped before the first frame was mixed, 3673 // do not apply ramp 3674 param = AudioMixer::RAMP_VOLUME; 3675 } 3676 3677 // compute volume for this track 3678 uint32_t vl, vr; // in U8.24 integer format 3679 float vlf, vrf, vaf; // in [0.0, 1.0] float format 3680 if (track->isPausing() || mStreamTypes[track->streamType()].mute) { 3681 vl = vr = 0; 3682 vlf = vrf = vaf = 0.; 3683 if (track->isPausing()) { 3684 track->setPaused(); 3685 } 3686 } else { 3687 3688 // read original volumes with volume control 3689 float typeVolume = mStreamTypes[track->streamType()].volume; 3690 float v = masterVolume * typeVolume; 3691 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 3692 gain_minifloat_packed_t vlr = proxy->getVolumeLR(); 3693 vlf = float_from_gain(gain_minifloat_unpack_left(vlr)); 3694 vrf = float_from_gain(gain_minifloat_unpack_right(vlr)); 3695 // track volumes come from shared memory, so can't be trusted and must be clamped 3696 if (vlf > GAIN_FLOAT_UNITY) { 3697 ALOGV("Track left volume out of range: %.3g", vlf); 3698 vlf = GAIN_FLOAT_UNITY; 3699 } 3700 if (vrf > GAIN_FLOAT_UNITY) { 3701 ALOGV("Track right volume out of range: %.3g", vrf); 3702 vrf = GAIN_FLOAT_UNITY; 3703 } 3704 // now apply the master volume and stream type volume 3705 vlf *= v; 3706 vrf *= v; 3707 // assuming master volume and stream type volume each go up to 1.0, 3708 // then derive vl and vr as U8.24 versions for the effect chain 3709 const float scaleto8_24 = MAX_GAIN_INT * MAX_GAIN_INT; 3710 vl = (uint32_t) (scaleto8_24 * vlf); 3711 vr = (uint32_t) (scaleto8_24 * vrf); 3712 // vl and vr are now in U8.24 format 3713 uint16_t sendLevel = proxy->getSendLevel_U4_12(); 3714 // send level comes from shared memory and so may be corrupt 3715 if (sendLevel > MAX_GAIN_INT) { 3716 ALOGV("Track send level out of range: %04X", sendLevel); 3717 sendLevel = MAX_GAIN_INT; 3718 } 3719 // vaf is represented as [0.0, 1.0] float by rescaling sendLevel 3720 vaf = v * sendLevel * (1. / MAX_GAIN_INT); 3721 } 3722 3723 // Delegate volume control to effect in track effect chain if needed 3724 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 3725 // Do not ramp volume if volume is controlled by effect 3726 param = AudioMixer::VOLUME; 3727 // Update remaining floating point volume levels 3728 vlf = (float)vl / (1 << 24); 3729 vrf = (float)vr / (1 << 24); 3730 track->mHasVolumeController = true; 3731 } else { 3732 // force no volume ramp when volume controller was just disabled or removed 3733 // from effect chain to avoid volume spike 3734 if (track->mHasVolumeController) { 3735 param = AudioMixer::VOLUME; 3736 } 3737 track->mHasVolumeController = false; 3738 } 3739 3740 // XXX: these things DON'T need to be done each time 3741 mAudioMixer->setBufferProvider(name, track); 3742 mAudioMixer->enable(name); 3743 3744 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, &vlf); 3745 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, &vrf); 3746 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, &vaf); 3747 mAudioMixer->setParameter( 3748 name, 3749 AudioMixer::TRACK, 3750 AudioMixer::FORMAT, (void *)track->format()); 3751 mAudioMixer->setParameter( 3752 name, 3753 AudioMixer::TRACK, 3754 AudioMixer::CHANNEL_MASK, (void *)(uintptr_t)track->channelMask()); 3755 mAudioMixer->setParameter( 3756 name, 3757 AudioMixer::TRACK, 3758 AudioMixer::MIXER_CHANNEL_MASK, (void *)(uintptr_t)mChannelMask); 3759 // limit track sample rate to 2 x output sample rate, which changes at re-configuration 3760 uint32_t maxSampleRate = mSampleRate * AUDIO_RESAMPLER_DOWN_RATIO_MAX; 3761 uint32_t reqSampleRate = track->mAudioTrackServerProxy->getSampleRate(); 3762 if (reqSampleRate == 0) { 3763 reqSampleRate = mSampleRate; 3764 } else if (reqSampleRate > maxSampleRate) { 3765 reqSampleRate = maxSampleRate; 3766 } 3767 mAudioMixer->setParameter( 3768 name, 3769 AudioMixer::RESAMPLE, 3770 AudioMixer::SAMPLE_RATE, 3771 (void *)(uintptr_t)reqSampleRate); 3772 /* 3773 * Select the appropriate output buffer for the track. 3774 * 3775 * Tracks with effects go into their own effects chain buffer 3776 * and from there into either mEffectBuffer or mSinkBuffer. 3777 * 3778 * Other tracks can use mMixerBuffer for higher precision 3779 * channel accumulation. If this buffer is enabled 3780 * (mMixerBufferEnabled true), then selected tracks will accumulate 3781 * into it. 3782 * 3783 */ 3784 if (mMixerBufferEnabled 3785 && (track->mainBuffer() == mSinkBuffer 3786 || track->mainBuffer() == mMixerBuffer)) { 3787 mAudioMixer->setParameter( 3788 name, 3789 AudioMixer::TRACK, 3790 AudioMixer::MIXER_FORMAT, (void *)mMixerBufferFormat); 3791 mAudioMixer->setParameter( 3792 name, 3793 AudioMixer::TRACK, 3794 AudioMixer::MAIN_BUFFER, (void *)mMixerBuffer); 3795 // TODO: override track->mainBuffer()? 3796 mMixerBufferValid = true; 3797 } else { 3798 mAudioMixer->setParameter( 3799 name, 3800 AudioMixer::TRACK, 3801 AudioMixer::MIXER_FORMAT, (void *)AUDIO_FORMAT_PCM_16_BIT); 3802 mAudioMixer->setParameter( 3803 name, 3804 AudioMixer::TRACK, 3805 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 3806 } 3807 mAudioMixer->setParameter( 3808 name, 3809 AudioMixer::TRACK, 3810 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 3811 3812 // reset retry count 3813 track->mRetryCount = kMaxTrackRetries; 3814 3815 // If one track is ready, set the mixer ready if: 3816 // - the mixer was not ready during previous round OR 3817 // - no other track is not ready 3818 if (mMixerStatusIgnoringFastTracks != MIXER_TRACKS_READY || 3819 mixerStatus != MIXER_TRACKS_ENABLED) { 3820 mixerStatus = MIXER_TRACKS_READY; 3821 } 3822 } else { 3823 if (framesReady < desiredFrames && !track->isStopped() && !track->isPaused()) { 3824 track->mAudioTrackServerProxy->tallyUnderrunFrames(desiredFrames); 3825 } 3826 // clear effect chain input buffer if an active track underruns to avoid sending 3827 // previous audio buffer again to effects 3828 chain = getEffectChain_l(track->sessionId()); 3829 if (chain != 0) { 3830 chain->clearInputBuffer(); 3831 } 3832 3833 ALOGVV("track %d s=%08x [NOT READY] on thread %p", name, cblk->mServer, this); 3834 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3835 track->isStopped() || track->isPaused()) { 3836 // We have consumed all the buffers of this track. 3837 // Remove it from the list of active tracks. 3838 // TODO: use actual buffer filling status instead of latency when available from 3839 // audio HAL 3840 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 3841 size_t framesWritten = mBytesWritten / mFrameSize; 3842 if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) { 3843 if (track->isStopped()) { 3844 track->reset(); 3845 } 3846 tracksToRemove->add(track); 3847 } 3848 } else { 3849 // No buffers for this track. Give it a few chances to 3850 // fill a buffer, then remove it from active list. 3851 if (--(track->mRetryCount) <= 0) { 3852 ALOGI("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 3853 tracksToRemove->add(track); 3854 // indicate to client process that the track was disabled because of underrun; 3855 // it will then automatically call start() when data is available 3856 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 3857 // If one track is not ready, mark the mixer also not ready if: 3858 // - the mixer was ready during previous round OR 3859 // - no other track is ready 3860 } else if (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY || 3861 mixerStatus != MIXER_TRACKS_READY) { 3862 mixerStatus = MIXER_TRACKS_ENABLED; 3863 } 3864 } 3865 mAudioMixer->disable(name); 3866 } 3867 3868 } // local variable scope to avoid goto warning 3869track_is_ready: ; 3870 3871 } 3872 3873 // Push the new FastMixer state if necessary 3874 bool pauseAudioWatchdog = false; 3875 if (didModify) { 3876 state->mFastTracksGen++; 3877 // if the fast mixer was active, but now there are no fast tracks, then put it in cold idle 3878 if (kUseFastMixer == FastMixer_Dynamic && 3879 state->mCommand == FastMixerState::MIX_WRITE && state->mTrackMask <= 1) { 3880 state->mCommand = FastMixerState::COLD_IDLE; 3881 state->mColdFutexAddr = &mFastMixerFutex; 3882 state->mColdGen++; 3883 mFastMixerFutex = 0; 3884 if (kUseFastMixer == FastMixer_Dynamic) { 3885 mNormalSink = mOutputSink; 3886 } 3887 // If we go into cold idle, need to wait for acknowledgement 3888 // so that fast mixer stops doing I/O. 3889 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3890 pauseAudioWatchdog = true; 3891 } 3892 } 3893 if (sq != NULL) { 3894 sq->end(didModify); 3895 sq->push(block); 3896 } 3897#ifdef AUDIO_WATCHDOG 3898 if (pauseAudioWatchdog && mAudioWatchdog != 0) { 3899 mAudioWatchdog->pause(); 3900 } 3901#endif 3902 3903 // Now perform the deferred reset on fast tracks that have stopped 3904 while (resetMask != 0) { 3905 size_t i = __builtin_ctz(resetMask); 3906 ALOG_ASSERT(i < count); 3907 resetMask &= ~(1 << i); 3908 sp<Track> t = mActiveTracks[i].promote(); 3909 if (t == 0) { 3910 continue; 3911 } 3912 Track* track = t.get(); 3913 ALOG_ASSERT(track->isFastTrack() && track->isStopped()); 3914 track->reset(); 3915 } 3916 3917 // remove all the tracks that need to be... 3918 removeTracks_l(*tracksToRemove); 3919 3920 if (getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX) != 0) { 3921 mEffectBufferValid = true; 3922 } 3923 3924 if (mEffectBufferValid) { 3925 // as long as there are effects we should clear the effects buffer, to avoid 3926 // passing a non-clean buffer to the effect chain 3927 memset(mEffectBuffer, 0, mEffectBufferSize); 3928 } 3929 // sink or mix buffer must be cleared if all tracks are connected to an 3930 // effect chain as in this case the mixer will not write to the sink or mix buffer 3931 // and track effects will accumulate into it 3932 if ((mBytesRemaining == 0) && ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 3933 (mixedTracks == 0 && fastTracks > 0))) { 3934 // FIXME as a performance optimization, should remember previous zero status 3935 if (mMixerBufferValid) { 3936 memset(mMixerBuffer, 0, mMixerBufferSize); 3937 // TODO: In testing, mSinkBuffer below need not be cleared because 3938 // the PlaybackThread::threadLoop() copies mMixerBuffer into mSinkBuffer 3939 // after mixing. 3940 // 3941 // To enforce this guarantee: 3942 // ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 3943 // (mixedTracks == 0 && fastTracks > 0)) 3944 // must imply MIXER_TRACKS_READY. 3945 // Later, we may clear buffers regardless, and skip much of this logic. 3946 } 3947 // FIXME as a performance optimization, should remember previous zero status 3948 memset(mSinkBuffer, 0, mNormalFrameCount * mFrameSize); 3949 } 3950 3951 // if any fast tracks, then status is ready 3952 mMixerStatusIgnoringFastTracks = mixerStatus; 3953 if (fastTracks > 0) { 3954 mixerStatus = MIXER_TRACKS_READY; 3955 } 3956 return mixerStatus; 3957} 3958 3959// getTrackName_l() must be called with ThreadBase::mLock held 3960int AudioFlinger::MixerThread::getTrackName_l(audio_channel_mask_t channelMask, 3961 audio_format_t format, int sessionId) 3962{ 3963 return mAudioMixer->getTrackName(channelMask, format, sessionId); 3964} 3965 3966// deleteTrackName_l() must be called with ThreadBase::mLock held 3967void AudioFlinger::MixerThread::deleteTrackName_l(int name) 3968{ 3969 ALOGV("remove track (%d) and delete from mixer", name); 3970 mAudioMixer->deleteTrackName(name); 3971} 3972 3973// checkForNewParameter_l() must be called with ThreadBase::mLock held 3974bool AudioFlinger::MixerThread::checkForNewParameter_l(const String8& keyValuePair, 3975 status_t& status) 3976{ 3977 bool reconfig = false; 3978 3979 status = NO_ERROR; 3980 3981 // if !&IDLE, holds the FastMixer state to restore after new parameters processed 3982 FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE; 3983 if (mFastMixer != 0) { 3984 FastMixerStateQueue *sq = mFastMixer->sq(); 3985 FastMixerState *state = sq->begin(); 3986 if (!(state->mCommand & FastMixerState::IDLE)) { 3987 previousCommand = state->mCommand; 3988 state->mCommand = FastMixerState::HOT_IDLE; 3989 sq->end(); 3990 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3991 } else { 3992 sq->end(false /*didModify*/); 3993 } 3994 } 3995 3996 AudioParameter param = AudioParameter(keyValuePair); 3997 int value; 3998 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 3999 reconfig = true; 4000 } 4001 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 4002 if (!isValidPcmSinkFormat((audio_format_t) value)) { 4003 status = BAD_VALUE; 4004 } else { 4005 // no need to save value, since it's constant 4006 reconfig = true; 4007 } 4008 } 4009 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 4010 if (!isValidPcmSinkChannelMask((audio_channel_mask_t) value)) { 4011 status = BAD_VALUE; 4012 } else { 4013 // no need to save value, since it's constant 4014 reconfig = true; 4015 } 4016 } 4017 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 4018 // do not accept frame count changes if tracks are open as the track buffer 4019 // size depends on frame count and correct behavior would not be guaranteed 4020 // if frame count is changed after track creation 4021 if (!mTracks.isEmpty()) { 4022 status = INVALID_OPERATION; 4023 } else { 4024 reconfig = true; 4025 } 4026 } 4027 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 4028#ifdef ADD_BATTERY_DATA 4029 // when changing the audio output device, call addBatteryData to notify 4030 // the change 4031 if (mOutDevice != value) { 4032 uint32_t params = 0; 4033 // check whether speaker is on 4034 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 4035 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 4036 } 4037 4038 audio_devices_t deviceWithoutSpeaker 4039 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 4040 // check if any other device (except speaker) is on 4041 if (value & deviceWithoutSpeaker ) { 4042 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 4043 } 4044 4045 if (params != 0) { 4046 addBatteryData(params); 4047 } 4048 } 4049#endif 4050 4051 // forward device change to effects that have requested to be 4052 // aware of attached audio device. 4053 if (value != AUDIO_DEVICE_NONE) { 4054 mOutDevice = value; 4055 for (size_t i = 0; i < mEffectChains.size(); i++) { 4056 mEffectChains[i]->setDevice_l(mOutDevice); 4057 } 4058 } 4059 } 4060 4061 if (status == NO_ERROR) { 4062 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 4063 keyValuePair.string()); 4064 if (!mStandby && status == INVALID_OPERATION) { 4065 mOutput->standby(); 4066 mStandby = true; 4067 mBytesWritten = 0; 4068 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 4069 keyValuePair.string()); 4070 } 4071 if (status == NO_ERROR && reconfig) { 4072 readOutputParameters_l(); 4073 delete mAudioMixer; 4074 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 4075 for (size_t i = 0; i < mTracks.size() ; i++) { 4076 int name = getTrackName_l(mTracks[i]->mChannelMask, 4077 mTracks[i]->mFormat, mTracks[i]->mSessionId); 4078 if (name < 0) { 4079 break; 4080 } 4081 mTracks[i]->mName = name; 4082 } 4083 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 4084 } 4085 } 4086 4087 if (!(previousCommand & FastMixerState::IDLE)) { 4088 ALOG_ASSERT(mFastMixer != 0); 4089 FastMixerStateQueue *sq = mFastMixer->sq(); 4090 FastMixerState *state = sq->begin(); 4091 ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE); 4092 state->mCommand = previousCommand; 4093 sq->end(); 4094 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 4095 } 4096 4097 return reconfig; 4098} 4099 4100 4101void AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 4102{ 4103 const size_t SIZE = 256; 4104 char buffer[SIZE]; 4105 String8 result; 4106 4107 PlaybackThread::dumpInternals(fd, args); 4108 4109 dprintf(fd, " AudioMixer tracks: 0x%08x\n", mAudioMixer->trackNames()); 4110 4111 // Make a non-atomic copy of fast mixer dump state so it won't change underneath us 4112 const FastMixerDumpState copy(mFastMixerDumpState); 4113 copy.dump(fd); 4114 4115#ifdef STATE_QUEUE_DUMP 4116 // Similar for state queue 4117 StateQueueObserverDump observerCopy = mStateQueueObserverDump; 4118 observerCopy.dump(fd); 4119 StateQueueMutatorDump mutatorCopy = mStateQueueMutatorDump; 4120 mutatorCopy.dump(fd); 4121#endif 4122 4123#ifdef TEE_SINK 4124 // Write the tee output to a .wav file 4125 dumpTee(fd, mTeeSource, mId); 4126#endif 4127 4128#ifdef AUDIO_WATCHDOG 4129 if (mAudioWatchdog != 0) { 4130 // Make a non-atomic copy of audio watchdog dump so it won't change underneath us 4131 AudioWatchdogDump wdCopy = mAudioWatchdogDump; 4132 wdCopy.dump(fd); 4133 } 4134#endif 4135} 4136 4137uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const 4138{ 4139 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000) / 2; 4140} 4141 4142uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() const 4143{ 4144 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000); 4145} 4146 4147void AudioFlinger::MixerThread::cacheParameters_l() 4148{ 4149 PlaybackThread::cacheParameters_l(); 4150 4151 // FIXME: Relaxed timing because of a certain device that can't meet latency 4152 // Should be reduced to 2x after the vendor fixes the driver issue 4153 // increase threshold again due to low power audio mode. The way this warning 4154 // threshold is calculated and its usefulness should be reconsidered anyway. 4155 maxPeriod = seconds(mNormalFrameCount) / mSampleRate * 15; 4156} 4157 4158// ---------------------------------------------------------------------------- 4159 4160AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 4161 AudioStreamOut* output, audio_io_handle_t id, audio_devices_t device) 4162 : PlaybackThread(audioFlinger, output, id, device, DIRECT) 4163 // mLeftVolFloat, mRightVolFloat 4164{ 4165} 4166 4167AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 4168 AudioStreamOut* output, audio_io_handle_t id, uint32_t device, 4169 ThreadBase::type_t type) 4170 : PlaybackThread(audioFlinger, output, id, device, type) 4171 // mLeftVolFloat, mRightVolFloat 4172{ 4173} 4174 4175AudioFlinger::DirectOutputThread::~DirectOutputThread() 4176{ 4177} 4178 4179void AudioFlinger::DirectOutputThread::processVolume_l(Track *track, bool lastTrack) 4180{ 4181 audio_track_cblk_t* cblk = track->cblk(); 4182 float left, right; 4183 4184 if (mMasterMute || mStreamTypes[track->streamType()].mute) { 4185 left = right = 0; 4186 } else { 4187 float typeVolume = mStreamTypes[track->streamType()].volume; 4188 float v = mMasterVolume * typeVolume; 4189 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 4190 gain_minifloat_packed_t vlr = proxy->getVolumeLR(); 4191 left = float_from_gain(gain_minifloat_unpack_left(vlr)); 4192 if (left > GAIN_FLOAT_UNITY) { 4193 left = GAIN_FLOAT_UNITY; 4194 } 4195 left *= v; 4196 right = float_from_gain(gain_minifloat_unpack_right(vlr)); 4197 if (right > GAIN_FLOAT_UNITY) { 4198 right = GAIN_FLOAT_UNITY; 4199 } 4200 right *= v; 4201 } 4202 4203 if (lastTrack) { 4204 if (left != mLeftVolFloat || right != mRightVolFloat) { 4205 mLeftVolFloat = left; 4206 mRightVolFloat = right; 4207 4208 // Convert volumes from float to 8.24 4209 uint32_t vl = (uint32_t)(left * (1 << 24)); 4210 uint32_t vr = (uint32_t)(right * (1 << 24)); 4211 4212 // Delegate volume control to effect in track effect chain if needed 4213 // only one effect chain can be present on DirectOutputThread, so if 4214 // there is one, the track is connected to it 4215 if (!mEffectChains.isEmpty()) { 4216 mEffectChains[0]->setVolume_l(&vl, &vr); 4217 left = (float)vl / (1 << 24); 4218 right = (float)vr / (1 << 24); 4219 } 4220 if (mOutput->stream->set_volume) { 4221 mOutput->stream->set_volume(mOutput->stream, left, right); 4222 } 4223 } 4224 } 4225} 4226 4227 4228AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l( 4229 Vector< sp<Track> > *tracksToRemove 4230) 4231{ 4232 size_t count = mActiveTracks.size(); 4233 mixer_state mixerStatus = MIXER_IDLE; 4234 bool doHwPause = false; 4235 bool doHwResume = false; 4236 bool flushPending = false; 4237 4238 // find out which tracks need to be processed 4239 for (size_t i = 0; i < count; i++) { 4240 sp<Track> t = mActiveTracks[i].promote(); 4241 // The track died recently 4242 if (t == 0) { 4243 continue; 4244 } 4245 4246 Track* const track = t.get(); 4247 audio_track_cblk_t* cblk = track->cblk(); 4248 // Only consider last track started for volume and mixer state control. 4249 // In theory an older track could underrun and restart after the new one starts 4250 // but as we only care about the transition phase between two tracks on a 4251 // direct output, it is not a problem to ignore the underrun case. 4252 sp<Track> l = mLatestActiveTrack.promote(); 4253 bool last = l.get() == track; 4254 4255 if (mHwSupportsPause && track->isPausing()) { 4256 track->setPaused(); 4257 if (last && !mHwPaused) { 4258 doHwPause = true; 4259 mHwPaused = true; 4260 } 4261 tracksToRemove->add(track); 4262 } else if (track->isFlushPending()) { 4263 track->flushAck(); 4264 if (last) { 4265 flushPending = true; 4266 } 4267 } else if (mHwSupportsPause && track->isResumePending()){ 4268 track->resumeAck(); 4269 if (last) { 4270 if (mHwPaused) { 4271 doHwResume = true; 4272 mHwPaused = false; 4273 } 4274 } 4275 } 4276 4277 // The first time a track is added we wait 4278 // for all its buffers to be filled before processing it. 4279 // Allow draining the buffer in case the client 4280 // app does not call stop() and relies on underrun to stop: 4281 // hence the test on (track->mRetryCount > 1). 4282 // If retryCount<=1 then track is about to underrun and be removed. 4283 uint32_t minFrames; 4284 if ((track->sharedBuffer() == 0) && !track->isStopping_1() && !track->isPausing() 4285 && (track->mRetryCount > 1)) { 4286 minFrames = mNormalFrameCount; 4287 } else { 4288 minFrames = 1; 4289 } 4290 4291 if ((track->framesReady() >= minFrames) && track->isReady() && !track->isPaused() && 4292 !track->isStopping_2() && !track->isStopped()) 4293 { 4294 ALOGVV("track %d s=%08x [OK]", track->name(), cblk->mServer); 4295 4296 if (track->mFillingUpStatus == Track::FS_FILLED) { 4297 track->mFillingUpStatus = Track::FS_ACTIVE; 4298 // make sure processVolume_l() will apply new volume even if 0 4299 mLeftVolFloat = mRightVolFloat = -1.0; 4300 if (!mHwSupportsPause) { 4301 track->resumeAck(); 4302 } 4303 } 4304 4305 // compute volume for this track 4306 processVolume_l(track, last); 4307 if (last) { 4308 // reset retry count 4309 track->mRetryCount = kMaxTrackRetriesDirect; 4310 mActiveTrack = t; 4311 mixerStatus = MIXER_TRACKS_READY; 4312 if (usesHwAvSync() && mHwPaused) { 4313 doHwResume = true; 4314 mHwPaused = false; 4315 } 4316 } 4317 } else { 4318 // clear effect chain input buffer if the last active track started underruns 4319 // to avoid sending previous audio buffer again to effects 4320 if (!mEffectChains.isEmpty() && last) { 4321 mEffectChains[0]->clearInputBuffer(); 4322 } 4323 if (track->isStopping_1()) { 4324 track->mState = TrackBase::STOPPING_2; 4325 if (last && mHwPaused) { 4326 doHwResume = true; 4327 mHwPaused = false; 4328 } 4329 } 4330 if ((track->sharedBuffer() != 0) || track->isStopped() || 4331 track->isStopping_2() || track->isPaused()) { 4332 // We have consumed all the buffers of this track. 4333 // Remove it from the list of active tracks. 4334 size_t audioHALFrames; 4335 if (audio_is_linear_pcm(mFormat)) { 4336 audioHALFrames = (latency_l() * mSampleRate) / 1000; 4337 } else { 4338 audioHALFrames = 0; 4339 } 4340 4341 size_t framesWritten = mBytesWritten / mFrameSize; 4342 if (mStandby || !last || 4343 track->presentationComplete(framesWritten, audioHALFrames)) { 4344 if (track->isStopping_2()) { 4345 track->mState = TrackBase::STOPPED; 4346 } 4347 if (track->isStopped()) { 4348 track->reset(); 4349 } 4350 tracksToRemove->add(track); 4351 } 4352 } else { 4353 // No buffers for this track. Give it a few chances to 4354 // fill a buffer, then remove it from active list. 4355 // Only consider last track started for mixer state control 4356 if (--(track->mRetryCount) <= 0) { 4357 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 4358 tracksToRemove->add(track); 4359 // indicate to client process that the track was disabled because of underrun; 4360 // it will then automatically call start() when data is available 4361 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 4362 } else if (last) { 4363 mixerStatus = MIXER_TRACKS_ENABLED; 4364 if (usesHwAvSync() && !mHwPaused && !mStandby) { 4365 doHwPause = true; 4366 mHwPaused = true; 4367 } 4368 } 4369 } 4370 } 4371 } 4372 4373 // if an active track did not command a flush, check for pending flush on stopped tracks 4374 if (!flushPending) { 4375 for (size_t i = 0; i < mTracks.size(); i++) { 4376 if (mTracks[i]->isFlushPending()) { 4377 mTracks[i]->flushAck(); 4378 flushPending = true; 4379 } 4380 } 4381 } 4382 4383 // make sure the pause/flush/resume sequence is executed in the right order. 4384 // If a flush is pending and a track is active but the HW is not paused, force a HW pause 4385 // before flush and then resume HW. This can happen in case of pause/flush/resume 4386 // if resume is received before pause is executed. 4387 if (mHwSupportsPause && !mStandby && 4388 (doHwPause || (flushPending && !mHwPaused && (count != 0)))) { 4389 mOutput->stream->pause(mOutput->stream); 4390 } 4391 if (flushPending) { 4392 flushHw_l(); 4393 } 4394 if (mHwSupportsPause && !mStandby && doHwResume) { 4395 mOutput->stream->resume(mOutput->stream); 4396 } 4397 // remove all the tracks that need to be... 4398 removeTracks_l(*tracksToRemove); 4399 4400 return mixerStatus; 4401} 4402 4403void AudioFlinger::DirectOutputThread::threadLoop_mix() 4404{ 4405 size_t frameCount = mFrameCount; 4406 int8_t *curBuf = (int8_t *)mSinkBuffer; 4407 // output audio to hardware 4408 while (frameCount) { 4409 AudioBufferProvider::Buffer buffer; 4410 buffer.frameCount = frameCount; 4411 status_t status = mActiveTrack->getNextBuffer(&buffer); 4412 if (status != NO_ERROR || buffer.raw == NULL) { 4413 memset(curBuf, 0, frameCount * mFrameSize); 4414 break; 4415 } 4416 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 4417 frameCount -= buffer.frameCount; 4418 curBuf += buffer.frameCount * mFrameSize; 4419 mActiveTrack->releaseBuffer(&buffer); 4420 } 4421 mCurrentWriteLength = curBuf - (int8_t *)mSinkBuffer; 4422 sleepTime = 0; 4423 standbyTime = systemTime() + standbyDelay; 4424 mActiveTrack.clear(); 4425} 4426 4427void AudioFlinger::DirectOutputThread::threadLoop_sleepTime() 4428{ 4429 // do not write to HAL when paused 4430 if (mHwPaused || (usesHwAvSync() && mStandby)) { 4431 sleepTime = idleSleepTime; 4432 return; 4433 } 4434 if (sleepTime == 0) { 4435 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4436 sleepTime = activeSleepTime; 4437 } else { 4438 sleepTime = idleSleepTime; 4439 } 4440 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 4441 memset(mSinkBuffer, 0, mFrameCount * mFrameSize); 4442 sleepTime = 0; 4443 } 4444} 4445 4446void AudioFlinger::DirectOutputThread::threadLoop_exit() 4447{ 4448 { 4449 Mutex::Autolock _l(mLock); 4450 bool flushPending = false; 4451 for (size_t i = 0; i < mTracks.size(); i++) { 4452 if (mTracks[i]->isFlushPending()) { 4453 mTracks[i]->flushAck(); 4454 flushPending = true; 4455 } 4456 } 4457 if (flushPending) { 4458 flushHw_l(); 4459 } 4460 } 4461 PlaybackThread::threadLoop_exit(); 4462} 4463 4464// must be called with thread mutex locked 4465bool AudioFlinger::DirectOutputThread::shouldStandby_l() 4466{ 4467 bool trackPaused = false; 4468 bool trackStopped = false; 4469 4470 // do not put the HAL in standby when paused. AwesomePlayer clear the offloaded AudioTrack 4471 // after a timeout and we will enter standby then. 4472 if (mTracks.size() > 0) { 4473 trackPaused = mTracks[mTracks.size() - 1]->isPaused(); 4474 trackStopped = mTracks[mTracks.size() - 1]->isStopped() || 4475 mTracks[mTracks.size() - 1]->mState == TrackBase::IDLE; 4476 } 4477 4478 return !mStandby && !(trackPaused || (usesHwAvSync() && mHwPaused && !trackStopped)); 4479} 4480 4481// getTrackName_l() must be called with ThreadBase::mLock held 4482int AudioFlinger::DirectOutputThread::getTrackName_l(audio_channel_mask_t channelMask __unused, 4483 audio_format_t format __unused, int sessionId __unused) 4484{ 4485 return 0; 4486} 4487 4488// deleteTrackName_l() must be called with ThreadBase::mLock held 4489void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name __unused) 4490{ 4491} 4492 4493// checkForNewParameter_l() must be called with ThreadBase::mLock held 4494bool AudioFlinger::DirectOutputThread::checkForNewParameter_l(const String8& keyValuePair, 4495 status_t& status) 4496{ 4497 bool reconfig = false; 4498 4499 status = NO_ERROR; 4500 4501 AudioParameter param = AudioParameter(keyValuePair); 4502 int value; 4503 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 4504 // forward device change to effects that have requested to be 4505 // aware of attached audio device. 4506 if (value != AUDIO_DEVICE_NONE) { 4507 mOutDevice = value; 4508 for (size_t i = 0; i < mEffectChains.size(); i++) { 4509 mEffectChains[i]->setDevice_l(mOutDevice); 4510 } 4511 } 4512 } 4513 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 4514 // do not accept frame count changes if tracks are open as the track buffer 4515 // size depends on frame count and correct behavior would not be garantied 4516 // if frame count is changed after track creation 4517 if (!mTracks.isEmpty()) { 4518 status = INVALID_OPERATION; 4519 } else { 4520 reconfig = true; 4521 } 4522 } 4523 if (status == NO_ERROR) { 4524 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 4525 keyValuePair.string()); 4526 if (!mStandby && status == INVALID_OPERATION) { 4527 mOutput->standby(); 4528 mStandby = true; 4529 mBytesWritten = 0; 4530 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 4531 keyValuePair.string()); 4532 } 4533 if (status == NO_ERROR && reconfig) { 4534 readOutputParameters_l(); 4535 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 4536 } 4537 } 4538 4539 return reconfig; 4540} 4541 4542uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() const 4543{ 4544 uint32_t time; 4545 if (audio_is_linear_pcm(mFormat)) { 4546 time = PlaybackThread::activeSleepTimeUs(); 4547 } else { 4548 time = 10000; 4549 } 4550 return time; 4551} 4552 4553uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() const 4554{ 4555 uint32_t time; 4556 if (audio_is_linear_pcm(mFormat)) { 4557 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 4558 } else { 4559 time = 10000; 4560 } 4561 return time; 4562} 4563 4564uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() const 4565{ 4566 uint32_t time; 4567 if (audio_is_linear_pcm(mFormat)) { 4568 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 4569 } else { 4570 time = 10000; 4571 } 4572 return time; 4573} 4574 4575void AudioFlinger::DirectOutputThread::cacheParameters_l() 4576{ 4577 PlaybackThread::cacheParameters_l(); 4578 4579 // use shorter standby delay as on normal output to release 4580 // hardware resources as soon as possible 4581 // no delay on outputs with HW A/V sync 4582 if (usesHwAvSync()) { 4583 standbyDelay = 0; 4584 } else if (audio_is_linear_pcm(mFormat)) { 4585 standbyDelay = microseconds(activeSleepTime*2); 4586 } else { 4587 standbyDelay = kOffloadStandbyDelayNs; 4588 } 4589} 4590 4591void AudioFlinger::DirectOutputThread::flushHw_l() 4592{ 4593 mOutput->flush(); 4594 mHwPaused = false; 4595} 4596 4597// ---------------------------------------------------------------------------- 4598 4599AudioFlinger::AsyncCallbackThread::AsyncCallbackThread( 4600 const wp<AudioFlinger::PlaybackThread>& playbackThread) 4601 : Thread(false /*canCallJava*/), 4602 mPlaybackThread(playbackThread), 4603 mWriteAckSequence(0), 4604 mDrainSequence(0) 4605{ 4606} 4607 4608AudioFlinger::AsyncCallbackThread::~AsyncCallbackThread() 4609{ 4610} 4611 4612void AudioFlinger::AsyncCallbackThread::onFirstRef() 4613{ 4614 run("Offload Cbk", ANDROID_PRIORITY_URGENT_AUDIO); 4615} 4616 4617bool AudioFlinger::AsyncCallbackThread::threadLoop() 4618{ 4619 while (!exitPending()) { 4620 uint32_t writeAckSequence; 4621 uint32_t drainSequence; 4622 4623 { 4624 Mutex::Autolock _l(mLock); 4625 while (!((mWriteAckSequence & 1) || 4626 (mDrainSequence & 1) || 4627 exitPending())) { 4628 mWaitWorkCV.wait(mLock); 4629 } 4630 4631 if (exitPending()) { 4632 break; 4633 } 4634 ALOGV("AsyncCallbackThread mWriteAckSequence %d mDrainSequence %d", 4635 mWriteAckSequence, mDrainSequence); 4636 writeAckSequence = mWriteAckSequence; 4637 mWriteAckSequence &= ~1; 4638 drainSequence = mDrainSequence; 4639 mDrainSequence &= ~1; 4640 } 4641 { 4642 sp<AudioFlinger::PlaybackThread> playbackThread = mPlaybackThread.promote(); 4643 if (playbackThread != 0) { 4644 if (writeAckSequence & 1) { 4645 playbackThread->resetWriteBlocked(writeAckSequence >> 1); 4646 } 4647 if (drainSequence & 1) { 4648 playbackThread->resetDraining(drainSequence >> 1); 4649 } 4650 } 4651 } 4652 } 4653 return false; 4654} 4655 4656void AudioFlinger::AsyncCallbackThread::exit() 4657{ 4658 ALOGV("AsyncCallbackThread::exit"); 4659 Mutex::Autolock _l(mLock); 4660 requestExit(); 4661 mWaitWorkCV.broadcast(); 4662} 4663 4664void AudioFlinger::AsyncCallbackThread::setWriteBlocked(uint32_t sequence) 4665{ 4666 Mutex::Autolock _l(mLock); 4667 // bit 0 is cleared 4668 mWriteAckSequence = sequence << 1; 4669} 4670 4671void AudioFlinger::AsyncCallbackThread::resetWriteBlocked() 4672{ 4673 Mutex::Autolock _l(mLock); 4674 // ignore unexpected callbacks 4675 if (mWriteAckSequence & 2) { 4676 mWriteAckSequence |= 1; 4677 mWaitWorkCV.signal(); 4678 } 4679} 4680 4681void AudioFlinger::AsyncCallbackThread::setDraining(uint32_t sequence) 4682{ 4683 Mutex::Autolock _l(mLock); 4684 // bit 0 is cleared 4685 mDrainSequence = sequence << 1; 4686} 4687 4688void AudioFlinger::AsyncCallbackThread::resetDraining() 4689{ 4690 Mutex::Autolock _l(mLock); 4691 // ignore unexpected callbacks 4692 if (mDrainSequence & 2) { 4693 mDrainSequence |= 1; 4694 mWaitWorkCV.signal(); 4695 } 4696} 4697 4698 4699// ---------------------------------------------------------------------------- 4700AudioFlinger::OffloadThread::OffloadThread(const sp<AudioFlinger>& audioFlinger, 4701 AudioStreamOut* output, audio_io_handle_t id, uint32_t device) 4702 : DirectOutputThread(audioFlinger, output, id, device, OFFLOAD), 4703 mPausedBytesRemaining(0) 4704{ 4705 //FIXME: mStandby should be set to true by ThreadBase constructor 4706 mStandby = true; 4707} 4708 4709void AudioFlinger::OffloadThread::threadLoop_exit() 4710{ 4711 if (mFlushPending || mHwPaused) { 4712 // If a flush is pending or track was paused, just discard buffered data 4713 flushHw_l(); 4714 } else { 4715 mMixerStatus = MIXER_DRAIN_ALL; 4716 threadLoop_drain(); 4717 } 4718 if (mUseAsyncWrite) { 4719 ALOG_ASSERT(mCallbackThread != 0); 4720 mCallbackThread->exit(); 4721 } 4722 PlaybackThread::threadLoop_exit(); 4723} 4724 4725AudioFlinger::PlaybackThread::mixer_state AudioFlinger::OffloadThread::prepareTracks_l( 4726 Vector< sp<Track> > *tracksToRemove 4727) 4728{ 4729 size_t count = mActiveTracks.size(); 4730 4731 mixer_state mixerStatus = MIXER_IDLE; 4732 bool doHwPause = false; 4733 bool doHwResume = false; 4734 4735 ALOGV("OffloadThread::prepareTracks_l active tracks %d", count); 4736 4737 // find out which tracks need to be processed 4738 for (size_t i = 0; i < count; i++) { 4739 sp<Track> t = mActiveTracks[i].promote(); 4740 // The track died recently 4741 if (t == 0) { 4742 continue; 4743 } 4744 Track* const track = t.get(); 4745 audio_track_cblk_t* cblk = track->cblk(); 4746 // Only consider last track started for volume and mixer state control. 4747 // In theory an older track could underrun and restart after the new one starts 4748 // but as we only care about the transition phase between two tracks on a 4749 // direct output, it is not a problem to ignore the underrun case. 4750 sp<Track> l = mLatestActiveTrack.promote(); 4751 bool last = l.get() == track; 4752 4753 if (track->isInvalid()) { 4754 ALOGW("An invalidated track shouldn't be in active list"); 4755 tracksToRemove->add(track); 4756 continue; 4757 } 4758 4759 if (track->mState == TrackBase::IDLE) { 4760 ALOGW("An idle track shouldn't be in active list"); 4761 continue; 4762 } 4763 4764 if (track->isPausing()) { 4765 track->setPaused(); 4766 if (last) { 4767 if (!mHwPaused) { 4768 doHwPause = true; 4769 mHwPaused = true; 4770 } 4771 // If we were part way through writing the mixbuffer to 4772 // the HAL we must save this until we resume 4773 // BUG - this will be wrong if a different track is made active, 4774 // in that case we want to discard the pending data in the 4775 // mixbuffer and tell the client to present it again when the 4776 // track is resumed 4777 mPausedWriteLength = mCurrentWriteLength; 4778 mPausedBytesRemaining = mBytesRemaining; 4779 mBytesRemaining = 0; // stop writing 4780 } 4781 tracksToRemove->add(track); 4782 } else if (track->isFlushPending()) { 4783 track->flushAck(); 4784 if (last) { 4785 mFlushPending = true; 4786 } 4787 } else if (track->isResumePending()){ 4788 track->resumeAck(); 4789 if (last) { 4790 if (mPausedBytesRemaining) { 4791 // Need to continue write that was interrupted 4792 mCurrentWriteLength = mPausedWriteLength; 4793 mBytesRemaining = mPausedBytesRemaining; 4794 mPausedBytesRemaining = 0; 4795 } 4796 if (mHwPaused) { 4797 doHwResume = true; 4798 mHwPaused = false; 4799 // threadLoop_mix() will handle the case that we need to 4800 // resume an interrupted write 4801 } 4802 // enable write to audio HAL 4803 sleepTime = 0; 4804 4805 // Do not handle new data in this iteration even if track->framesReady() 4806 mixerStatus = MIXER_TRACKS_ENABLED; 4807 } 4808 } else if (track->framesReady() && track->isReady() && 4809 !track->isPaused() && !track->isTerminated() && !track->isStopping_2()) { 4810 ALOGVV("OffloadThread: track %d s=%08x [OK]", track->name(), cblk->mServer); 4811 if (track->mFillingUpStatus == Track::FS_FILLED) { 4812 track->mFillingUpStatus = Track::FS_ACTIVE; 4813 // make sure processVolume_l() will apply new volume even if 0 4814 mLeftVolFloat = mRightVolFloat = -1.0; 4815 } 4816 4817 if (last) { 4818 sp<Track> previousTrack = mPreviousTrack.promote(); 4819 if (previousTrack != 0) { 4820 if (track != previousTrack.get()) { 4821 // Flush any data still being written from last track 4822 mBytesRemaining = 0; 4823 if (mPausedBytesRemaining) { 4824 // Last track was paused so we also need to flush saved 4825 // mixbuffer state and invalidate track so that it will 4826 // re-submit that unwritten data when it is next resumed 4827 mPausedBytesRemaining = 0; 4828 // Invalidate is a bit drastic - would be more efficient 4829 // to have a flag to tell client that some of the 4830 // previously written data was lost 4831 previousTrack->invalidate(); 4832 } 4833 // flush data already sent to the DSP if changing audio session as audio 4834 // comes from a different source. Also invalidate previous track to force a 4835 // seek when resuming. 4836 if (previousTrack->sessionId() != track->sessionId()) { 4837 previousTrack->invalidate(); 4838 } 4839 } 4840 } 4841 mPreviousTrack = track; 4842 // reset retry count 4843 track->mRetryCount = kMaxTrackRetriesOffload; 4844 mActiveTrack = t; 4845 mixerStatus = MIXER_TRACKS_READY; 4846 } 4847 } else { 4848 ALOGVV("OffloadThread: track %d s=%08x [NOT READY]", track->name(), cblk->mServer); 4849 if (track->isStopping_1()) { 4850 // Hardware buffer can hold a large amount of audio so we must 4851 // wait for all current track's data to drain before we say 4852 // that the track is stopped. 4853 if (mBytesRemaining == 0) { 4854 // Only start draining when all data in mixbuffer 4855 // has been written 4856 ALOGV("OffloadThread: underrun and STOPPING_1 -> draining, STOPPING_2"); 4857 track->mState = TrackBase::STOPPING_2; // so presentation completes after drain 4858 // do not drain if no data was ever sent to HAL (mStandby == true) 4859 if (last && !mStandby) { 4860 // do not modify drain sequence if we are already draining. This happens 4861 // when resuming from pause after drain. 4862 if ((mDrainSequence & 1) == 0) { 4863 sleepTime = 0; 4864 standbyTime = systemTime() + standbyDelay; 4865 mixerStatus = MIXER_DRAIN_TRACK; 4866 mDrainSequence += 2; 4867 } 4868 if (mHwPaused) { 4869 // It is possible to move from PAUSED to STOPPING_1 without 4870 // a resume so we must ensure hardware is running 4871 doHwResume = true; 4872 mHwPaused = false; 4873 } 4874 } 4875 } 4876 } else if (track->isStopping_2()) { 4877 // Drain has completed or we are in standby, signal presentation complete 4878 if (!(mDrainSequence & 1) || !last || mStandby) { 4879 track->mState = TrackBase::STOPPED; 4880 size_t audioHALFrames = 4881 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 4882 size_t framesWritten = 4883 mBytesWritten / mOutput->getFrameSize(); 4884 track->presentationComplete(framesWritten, audioHALFrames); 4885 track->reset(); 4886 tracksToRemove->add(track); 4887 } 4888 } else { 4889 // No buffers for this track. Give it a few chances to 4890 // fill a buffer, then remove it from active list. 4891 if (--(track->mRetryCount) <= 0) { 4892 ALOGV("OffloadThread: BUFFER TIMEOUT: remove(%d) from active list", 4893 track->name()); 4894 tracksToRemove->add(track); 4895 // indicate to client process that the track was disabled because of underrun; 4896 // it will then automatically call start() when data is available 4897 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 4898 } else if (last){ 4899 mixerStatus = MIXER_TRACKS_ENABLED; 4900 } 4901 } 4902 } 4903 // compute volume for this track 4904 processVolume_l(track, last); 4905 } 4906 4907 // make sure the pause/flush/resume sequence is executed in the right order. 4908 // If a flush is pending and a track is active but the HW is not paused, force a HW pause 4909 // before flush and then resume HW. This can happen in case of pause/flush/resume 4910 // if resume is received before pause is executed. 4911 if (!mStandby && (doHwPause || (mFlushPending && !mHwPaused && (count != 0)))) { 4912 mOutput->stream->pause(mOutput->stream); 4913 } 4914 if (mFlushPending) { 4915 flushHw_l(); 4916 mFlushPending = false; 4917 } 4918 if (!mStandby && doHwResume) { 4919 mOutput->stream->resume(mOutput->stream); 4920 } 4921 4922 // remove all the tracks that need to be... 4923 removeTracks_l(*tracksToRemove); 4924 4925 return mixerStatus; 4926} 4927 4928// must be called with thread mutex locked 4929bool AudioFlinger::OffloadThread::waitingAsyncCallback_l() 4930{ 4931 ALOGVV("waitingAsyncCallback_l mWriteAckSequence %d mDrainSequence %d", 4932 mWriteAckSequence, mDrainSequence); 4933 if (mUseAsyncWrite && ((mWriteAckSequence & 1) || (mDrainSequence & 1))) { 4934 return true; 4935 } 4936 return false; 4937} 4938 4939bool AudioFlinger::OffloadThread::waitingAsyncCallback() 4940{ 4941 Mutex::Autolock _l(mLock); 4942 return waitingAsyncCallback_l(); 4943} 4944 4945void AudioFlinger::OffloadThread::flushHw_l() 4946{ 4947 DirectOutputThread::flushHw_l(); 4948 // Flush anything still waiting in the mixbuffer 4949 mCurrentWriteLength = 0; 4950 mBytesRemaining = 0; 4951 mPausedWriteLength = 0; 4952 mPausedBytesRemaining = 0; 4953 4954 if (mUseAsyncWrite) { 4955 // discard any pending drain or write ack by incrementing sequence 4956 mWriteAckSequence = (mWriteAckSequence + 2) & ~1; 4957 mDrainSequence = (mDrainSequence + 2) & ~1; 4958 ALOG_ASSERT(mCallbackThread != 0); 4959 mCallbackThread->setWriteBlocked(mWriteAckSequence); 4960 mCallbackThread->setDraining(mDrainSequence); 4961 } 4962} 4963 4964void AudioFlinger::OffloadThread::onAddNewTrack_l() 4965{ 4966 sp<Track> previousTrack = mPreviousTrack.promote(); 4967 sp<Track> latestTrack = mLatestActiveTrack.promote(); 4968 4969 if (previousTrack != 0 && latestTrack != 0 && 4970 (previousTrack->sessionId() != latestTrack->sessionId())) { 4971 mFlushPending = true; 4972 } 4973 PlaybackThread::onAddNewTrack_l(); 4974} 4975 4976// ---------------------------------------------------------------------------- 4977 4978AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 4979 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id) 4980 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->outDevice(), 4981 DUPLICATING), 4982 mWaitTimeMs(UINT_MAX) 4983{ 4984 addOutputTrack(mainThread); 4985} 4986 4987AudioFlinger::DuplicatingThread::~DuplicatingThread() 4988{ 4989 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4990 mOutputTracks[i]->destroy(); 4991 } 4992} 4993 4994void AudioFlinger::DuplicatingThread::threadLoop_mix() 4995{ 4996 // mix buffers... 4997 if (outputsReady(outputTracks)) { 4998 mAudioMixer->process(AudioBufferProvider::kInvalidPTS); 4999 } else { 5000 if (mMixerBufferValid) { 5001 memset(mMixerBuffer, 0, mMixerBufferSize); 5002 } else { 5003 memset(mSinkBuffer, 0, mSinkBufferSize); 5004 } 5005 } 5006 sleepTime = 0; 5007 writeFrames = mNormalFrameCount; 5008 mCurrentWriteLength = mSinkBufferSize; 5009 standbyTime = systemTime() + standbyDelay; 5010} 5011 5012void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() 5013{ 5014 if (sleepTime == 0) { 5015 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 5016 sleepTime = activeSleepTime; 5017 } else { 5018 sleepTime = idleSleepTime; 5019 } 5020 } else if (mBytesWritten != 0) { 5021 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 5022 writeFrames = mNormalFrameCount; 5023 memset(mSinkBuffer, 0, mSinkBufferSize); 5024 } else { 5025 // flush remaining overflow buffers in output tracks 5026 writeFrames = 0; 5027 } 5028 sleepTime = 0; 5029 } 5030} 5031 5032ssize_t AudioFlinger::DuplicatingThread::threadLoop_write() 5033{ 5034 for (size_t i = 0; i < outputTracks.size(); i++) { 5035 outputTracks[i]->write(mSinkBuffer, writeFrames); 5036 } 5037 mStandby = false; 5038 return (ssize_t)mSinkBufferSize; 5039} 5040 5041void AudioFlinger::DuplicatingThread::threadLoop_standby() 5042{ 5043 // DuplicatingThread implements standby by stopping all tracks 5044 for (size_t i = 0; i < outputTracks.size(); i++) { 5045 outputTracks[i]->stop(); 5046 } 5047} 5048 5049void AudioFlinger::DuplicatingThread::saveOutputTracks() 5050{ 5051 outputTracks = mOutputTracks; 5052} 5053 5054void AudioFlinger::DuplicatingThread::clearOutputTracks() 5055{ 5056 outputTracks.clear(); 5057} 5058 5059void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 5060{ 5061 Mutex::Autolock _l(mLock); 5062 // The downstream MixerThread consumes thread->frameCount() amount of frames per mix pass. 5063 // Adjust for thread->sampleRate() to determine minimum buffer frame count. 5064 // Then triple buffer because Threads do not run synchronously and may not be clock locked. 5065 const size_t frameCount = 5066 3 * sourceFramesNeeded(mSampleRate, thread->frameCount(), thread->sampleRate()); 5067 // TODO: Consider asynchronous sample rate conversion to handle clock disparity 5068 // from different OutputTracks and their associated MixerThreads (e.g. one may 5069 // nearly empty and the other may be dropping data). 5070 5071 sp<OutputTrack> outputTrack = new OutputTrack(thread, 5072 this, 5073 mSampleRate, 5074 mFormat, 5075 mChannelMask, 5076 frameCount, 5077 IPCThreadState::self()->getCallingUid()); 5078 if (outputTrack->cblk() != NULL) { 5079 thread->setStreamVolume(AUDIO_STREAM_PATCH, 1.0f); 5080 mOutputTracks.add(outputTrack); 5081 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack.get(), thread); 5082 updateWaitTime_l(); 5083 } 5084} 5085 5086void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 5087{ 5088 Mutex::Autolock _l(mLock); 5089 for (size_t i = 0; i < mOutputTracks.size(); i++) { 5090 if (mOutputTracks[i]->thread() == thread) { 5091 mOutputTracks[i]->destroy(); 5092 mOutputTracks.removeAt(i); 5093 updateWaitTime_l(); 5094 return; 5095 } 5096 } 5097 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 5098} 5099 5100// caller must hold mLock 5101void AudioFlinger::DuplicatingThread::updateWaitTime_l() 5102{ 5103 mWaitTimeMs = UINT_MAX; 5104 for (size_t i = 0; i < mOutputTracks.size(); i++) { 5105 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 5106 if (strong != 0) { 5107 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 5108 if (waitTimeMs < mWaitTimeMs) { 5109 mWaitTimeMs = waitTimeMs; 5110 } 5111 } 5112 } 5113} 5114 5115 5116bool AudioFlinger::DuplicatingThread::outputsReady( 5117 const SortedVector< sp<OutputTrack> > &outputTracks) 5118{ 5119 for (size_t i = 0; i < outputTracks.size(); i++) { 5120 sp<ThreadBase> thread = outputTracks[i]->thread().promote(); 5121 if (thread == 0) { 5122 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", 5123 outputTracks[i].get()); 5124 return false; 5125 } 5126 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 5127 // see note at standby() declaration 5128 if (playbackThread->standby() && !playbackThread->isSuspended()) { 5129 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), 5130 thread.get()); 5131 return false; 5132 } 5133 } 5134 return true; 5135} 5136 5137uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() const 5138{ 5139 return (mWaitTimeMs * 1000) / 2; 5140} 5141 5142void AudioFlinger::DuplicatingThread::cacheParameters_l() 5143{ 5144 // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first 5145 updateWaitTime_l(); 5146 5147 MixerThread::cacheParameters_l(); 5148} 5149 5150// ---------------------------------------------------------------------------- 5151// Record 5152// ---------------------------------------------------------------------------- 5153 5154AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 5155 AudioStreamIn *input, 5156 audio_io_handle_t id, 5157 audio_devices_t outDevice, 5158 audio_devices_t inDevice 5159#ifdef TEE_SINK 5160 , const sp<NBAIO_Sink>& teeSink 5161#endif 5162 ) : 5163 ThreadBase(audioFlinger, id, outDevice, inDevice, RECORD), 5164 mInput(input), mActiveTracksGen(0), mRsmpInBuffer(NULL), 5165 // mRsmpInFrames and mRsmpInFramesP2 are set by readInputParameters_l() 5166 mRsmpInRear(0) 5167#ifdef TEE_SINK 5168 , mTeeSink(teeSink) 5169#endif 5170 , mReadOnlyHeap(new MemoryDealer(kRecordThreadReadOnlyHeapSize, 5171 "RecordThreadRO", MemoryHeapBase::READ_ONLY)) 5172 // mFastCapture below 5173 , mFastCaptureFutex(0) 5174 // mInputSource 5175 // mPipeSink 5176 // mPipeSource 5177 , mPipeFramesP2(0) 5178 // mPipeMemory 5179 // mFastCaptureNBLogWriter 5180 , mFastTrackAvail(false) 5181{ 5182 snprintf(mThreadName, kThreadNameLength, "AudioIn_%X", id); 5183 mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mThreadName); 5184 5185 readInputParameters_l(); 5186 5187 // create an NBAIO source for the HAL input stream, and negotiate 5188 mInputSource = new AudioStreamInSource(input->stream); 5189 size_t numCounterOffers = 0; 5190 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount, mFormat)}; 5191 ssize_t index = mInputSource->negotiate(offers, 1, NULL, numCounterOffers); 5192 ALOG_ASSERT(index == 0); 5193 5194 // initialize fast capture depending on configuration 5195 bool initFastCapture; 5196 switch (kUseFastCapture) { 5197 case FastCapture_Never: 5198 initFastCapture = false; 5199 break; 5200 case FastCapture_Always: 5201 initFastCapture = true; 5202 break; 5203 case FastCapture_Static: 5204 uint32_t primaryOutputSampleRate; 5205 { 5206 AutoMutex _l(audioFlinger->mHardwareLock); 5207 primaryOutputSampleRate = audioFlinger->mPrimaryOutputSampleRate; 5208 } 5209 initFastCapture = 5210 // either capture sample rate is same as (a reasonable) primary output sample rate 5211 (((primaryOutputSampleRate == 44100 || primaryOutputSampleRate == 48000) && 5212 (mSampleRate == primaryOutputSampleRate)) || 5213 // or primary output sample rate is unknown, and capture sample rate is reasonable 5214 ((primaryOutputSampleRate == 0) && 5215 ((mSampleRate == 44100 || mSampleRate == 48000)))) && 5216 // and the buffer size is < 12 ms 5217 (mFrameCount * 1000) / mSampleRate < 12; 5218 break; 5219 // case FastCapture_Dynamic: 5220 } 5221 5222 if (initFastCapture) { 5223 // create a Pipe for FastCapture to write to, and for us and fast tracks to read from 5224 NBAIO_Format format = mInputSource->format(); 5225 size_t pipeFramesP2 = roundup(mSampleRate / 25); // double-buffering of 20 ms each 5226 size_t pipeSize = pipeFramesP2 * Format_frameSize(format); 5227 void *pipeBuffer; 5228 const sp<MemoryDealer> roHeap(readOnlyHeap()); 5229 sp<IMemory> pipeMemory; 5230 if ((roHeap == 0) || 5231 (pipeMemory = roHeap->allocate(pipeSize)) == 0 || 5232 (pipeBuffer = pipeMemory->pointer()) == NULL) { 5233 ALOGE("not enough memory for pipe buffer size=%zu", pipeSize); 5234 goto failed; 5235 } 5236 // pipe will be shared directly with fast clients, so clear to avoid leaking old information 5237 memset(pipeBuffer, 0, pipeSize); 5238 Pipe *pipe = new Pipe(pipeFramesP2, format, pipeBuffer); 5239 const NBAIO_Format offers[1] = {format}; 5240 size_t numCounterOffers = 0; 5241 ssize_t index = pipe->negotiate(offers, 1, NULL, numCounterOffers); 5242 ALOG_ASSERT(index == 0); 5243 mPipeSink = pipe; 5244 PipeReader *pipeReader = new PipeReader(*pipe); 5245 numCounterOffers = 0; 5246 index = pipeReader->negotiate(offers, 1, NULL, numCounterOffers); 5247 ALOG_ASSERT(index == 0); 5248 mPipeSource = pipeReader; 5249 mPipeFramesP2 = pipeFramesP2; 5250 mPipeMemory = pipeMemory; 5251 5252 // create fast capture 5253 mFastCapture = new FastCapture(); 5254 FastCaptureStateQueue *sq = mFastCapture->sq(); 5255#ifdef STATE_QUEUE_DUMP 5256 // FIXME 5257#endif 5258 FastCaptureState *state = sq->begin(); 5259 state->mCblk = NULL; 5260 state->mInputSource = mInputSource.get(); 5261 state->mInputSourceGen++; 5262 state->mPipeSink = pipe; 5263 state->mPipeSinkGen++; 5264 state->mFrameCount = mFrameCount; 5265 state->mCommand = FastCaptureState::COLD_IDLE; 5266 // already done in constructor initialization list 5267 //mFastCaptureFutex = 0; 5268 state->mColdFutexAddr = &mFastCaptureFutex; 5269 state->mColdGen++; 5270 state->mDumpState = &mFastCaptureDumpState; 5271#ifdef TEE_SINK 5272 // FIXME 5273#endif 5274 mFastCaptureNBLogWriter = audioFlinger->newWriter_l(kFastCaptureLogSize, "FastCapture"); 5275 state->mNBLogWriter = mFastCaptureNBLogWriter.get(); 5276 sq->end(); 5277 sq->push(FastCaptureStateQueue::BLOCK_UNTIL_PUSHED); 5278 5279 // start the fast capture 5280 mFastCapture->run("FastCapture", ANDROID_PRIORITY_URGENT_AUDIO); 5281 pid_t tid = mFastCapture->getTid(); 5282 int err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 5283 if (err != 0) { 5284 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 5285 kPriorityFastCapture, getpid_cached, tid, err); 5286 } 5287 5288#ifdef AUDIO_WATCHDOG 5289 // FIXME 5290#endif 5291 5292 mFastTrackAvail = true; 5293 } 5294failed: ; 5295 5296 // FIXME mNormalSource 5297} 5298 5299AudioFlinger::RecordThread::~RecordThread() 5300{ 5301 if (mFastCapture != 0) { 5302 FastCaptureStateQueue *sq = mFastCapture->sq(); 5303 FastCaptureState *state = sq->begin(); 5304 if (state->mCommand == FastCaptureState::COLD_IDLE) { 5305 int32_t old = android_atomic_inc(&mFastCaptureFutex); 5306 if (old == -1) { 5307 (void) syscall(__NR_futex, &mFastCaptureFutex, FUTEX_WAKE_PRIVATE, 1); 5308 } 5309 } 5310 state->mCommand = FastCaptureState::EXIT; 5311 sq->end(); 5312 sq->push(FastCaptureStateQueue::BLOCK_UNTIL_PUSHED); 5313 mFastCapture->join(); 5314 mFastCapture.clear(); 5315 } 5316 mAudioFlinger->unregisterWriter(mFastCaptureNBLogWriter); 5317 mAudioFlinger->unregisterWriter(mNBLogWriter); 5318 delete[] mRsmpInBuffer; 5319} 5320 5321void AudioFlinger::RecordThread::onFirstRef() 5322{ 5323 run(mThreadName, PRIORITY_URGENT_AUDIO); 5324} 5325 5326bool AudioFlinger::RecordThread::threadLoop() 5327{ 5328 nsecs_t lastWarning = 0; 5329 5330 inputStandBy(); 5331 5332reacquire_wakelock: 5333 sp<RecordTrack> activeTrack; 5334 int activeTracksGen; 5335 { 5336 Mutex::Autolock _l(mLock); 5337 size_t size = mActiveTracks.size(); 5338 activeTracksGen = mActiveTracksGen; 5339 if (size > 0) { 5340 // FIXME an arbitrary choice 5341 activeTrack = mActiveTracks[0]; 5342 acquireWakeLock_l(activeTrack->uid()); 5343 if (size > 1) { 5344 SortedVector<int> tmp; 5345 for (size_t i = 0; i < size; i++) { 5346 tmp.add(mActiveTracks[i]->uid()); 5347 } 5348 updateWakeLockUids_l(tmp); 5349 } 5350 } else { 5351 acquireWakeLock_l(-1); 5352 } 5353 } 5354 5355 // used to request a deferred sleep, to be executed later while mutex is unlocked 5356 uint32_t sleepUs = 0; 5357 5358 // loop while there is work to do 5359 for (;;) { 5360 Vector< sp<EffectChain> > effectChains; 5361 5362 // sleep with mutex unlocked 5363 if (sleepUs > 0) { 5364 ATRACE_BEGIN("sleep"); 5365 usleep(sleepUs); 5366 ATRACE_END(); 5367 sleepUs = 0; 5368 } 5369 5370 // activeTracks accumulates a copy of a subset of mActiveTracks 5371 Vector< sp<RecordTrack> > activeTracks; 5372 5373 // reference to the (first and only) active fast track 5374 sp<RecordTrack> fastTrack; 5375 5376 // reference to a fast track which is about to be removed 5377 sp<RecordTrack> fastTrackToRemove; 5378 5379 { // scope for mLock 5380 Mutex::Autolock _l(mLock); 5381 5382 processConfigEvents_l(); 5383 5384 // check exitPending here because checkForNewParameters_l() and 5385 // checkForNewParameters_l() can temporarily release mLock 5386 if (exitPending()) { 5387 break; 5388 } 5389 5390 // if no active track(s), then standby and release wakelock 5391 size_t size = mActiveTracks.size(); 5392 if (size == 0) { 5393 standbyIfNotAlreadyInStandby(); 5394 // exitPending() can't become true here 5395 releaseWakeLock_l(); 5396 ALOGV("RecordThread: loop stopping"); 5397 // go to sleep 5398 mWaitWorkCV.wait(mLock); 5399 ALOGV("RecordThread: loop starting"); 5400 goto reacquire_wakelock; 5401 } 5402 5403 if (mActiveTracksGen != activeTracksGen) { 5404 activeTracksGen = mActiveTracksGen; 5405 SortedVector<int> tmp; 5406 for (size_t i = 0; i < size; i++) { 5407 tmp.add(mActiveTracks[i]->uid()); 5408 } 5409 updateWakeLockUids_l(tmp); 5410 } 5411 5412 bool doBroadcast = false; 5413 for (size_t i = 0; i < size; ) { 5414 5415 activeTrack = mActiveTracks[i]; 5416 if (activeTrack->isTerminated()) { 5417 if (activeTrack->isFastTrack()) { 5418 ALOG_ASSERT(fastTrackToRemove == 0); 5419 fastTrackToRemove = activeTrack; 5420 } 5421 removeTrack_l(activeTrack); 5422 mActiveTracks.remove(activeTrack); 5423 mActiveTracksGen++; 5424 size--; 5425 continue; 5426 } 5427 5428 TrackBase::track_state activeTrackState = activeTrack->mState; 5429 switch (activeTrackState) { 5430 5431 case TrackBase::PAUSING: 5432 mActiveTracks.remove(activeTrack); 5433 mActiveTracksGen++; 5434 doBroadcast = true; 5435 size--; 5436 continue; 5437 5438 case TrackBase::STARTING_1: 5439 sleepUs = 10000; 5440 i++; 5441 continue; 5442 5443 case TrackBase::STARTING_2: 5444 doBroadcast = true; 5445 mStandby = false; 5446 activeTrack->mState = TrackBase::ACTIVE; 5447 break; 5448 5449 case TrackBase::ACTIVE: 5450 break; 5451 5452 case TrackBase::IDLE: 5453 i++; 5454 continue; 5455 5456 default: 5457 LOG_ALWAYS_FATAL("Unexpected activeTrackState %d", activeTrackState); 5458 } 5459 5460 activeTracks.add(activeTrack); 5461 i++; 5462 5463 if (activeTrack->isFastTrack()) { 5464 ALOG_ASSERT(!mFastTrackAvail); 5465 ALOG_ASSERT(fastTrack == 0); 5466 fastTrack = activeTrack; 5467 } 5468 } 5469 if (doBroadcast) { 5470 mStartStopCond.broadcast(); 5471 } 5472 5473 // sleep if there are no active tracks to process 5474 if (activeTracks.size() == 0) { 5475 if (sleepUs == 0) { 5476 sleepUs = kRecordThreadSleepUs; 5477 } 5478 continue; 5479 } 5480 sleepUs = 0; 5481 5482 lockEffectChains_l(effectChains); 5483 } 5484 5485 // thread mutex is now unlocked, mActiveTracks unknown, activeTracks.size() > 0 5486 5487 size_t size = effectChains.size(); 5488 for (size_t i = 0; i < size; i++) { 5489 // thread mutex is not locked, but effect chain is locked 5490 effectChains[i]->process_l(); 5491 } 5492 5493 // Push a new fast capture state if fast capture is not already running, or cblk change 5494 if (mFastCapture != 0) { 5495 FastCaptureStateQueue *sq = mFastCapture->sq(); 5496 FastCaptureState *state = sq->begin(); 5497 bool didModify = false; 5498 FastCaptureStateQueue::block_t block = FastCaptureStateQueue::BLOCK_UNTIL_PUSHED; 5499 if (state->mCommand != FastCaptureState::READ_WRITE /* FIXME && 5500 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)*/) { 5501 if (state->mCommand == FastCaptureState::COLD_IDLE) { 5502 int32_t old = android_atomic_inc(&mFastCaptureFutex); 5503 if (old == -1) { 5504 (void) syscall(__NR_futex, &mFastCaptureFutex, FUTEX_WAKE_PRIVATE, 1); 5505 } 5506 } 5507 state->mCommand = FastCaptureState::READ_WRITE; 5508#if 0 // FIXME 5509 mFastCaptureDumpState.increaseSamplingN(mAudioFlinger->isLowRamDevice() ? 5510 FastThreadDumpState::kSamplingNforLowRamDevice : 5511 FastThreadDumpState::kSamplingN); 5512#endif 5513 didModify = true; 5514 } 5515 audio_track_cblk_t *cblkOld = state->mCblk; 5516 audio_track_cblk_t *cblkNew = fastTrack != 0 ? fastTrack->cblk() : NULL; 5517 if (cblkNew != cblkOld) { 5518 state->mCblk = cblkNew; 5519 // block until acked if removing a fast track 5520 if (cblkOld != NULL) { 5521 block = FastCaptureStateQueue::BLOCK_UNTIL_ACKED; 5522 } 5523 didModify = true; 5524 } 5525 sq->end(didModify); 5526 if (didModify) { 5527 sq->push(block); 5528#if 0 5529 if (kUseFastCapture == FastCapture_Dynamic) { 5530 mNormalSource = mPipeSource; 5531 } 5532#endif 5533 } 5534 } 5535 5536 // now run the fast track destructor with thread mutex unlocked 5537 fastTrackToRemove.clear(); 5538 5539 // Read from HAL to keep up with fastest client if multiple active tracks, not slowest one. 5540 // Only the client(s) that are too slow will overrun. But if even the fastest client is too 5541 // slow, then this RecordThread will overrun by not calling HAL read often enough. 5542 // If destination is non-contiguous, first read past the nominal end of buffer, then 5543 // copy to the right place. Permitted because mRsmpInBuffer was over-allocated. 5544 5545 int32_t rear = mRsmpInRear & (mRsmpInFramesP2 - 1); 5546 ssize_t framesRead; 5547 5548 // If an NBAIO source is present, use it to read the normal capture's data 5549 if (mPipeSource != 0) { 5550 size_t framesToRead = mBufferSize / mFrameSize; 5551 framesRead = mPipeSource->read(&mRsmpInBuffer[rear * mChannelCount], 5552 framesToRead, AudioBufferProvider::kInvalidPTS); 5553 if (framesRead == 0) { 5554 // since pipe is non-blocking, simulate blocking input 5555 sleepUs = (framesToRead * 1000000LL) / mSampleRate; 5556 } 5557 // otherwise use the HAL / AudioStreamIn directly 5558 } else { 5559 ssize_t bytesRead = mInput->stream->read(mInput->stream, 5560 &mRsmpInBuffer[rear * mChannelCount], mBufferSize); 5561 if (bytesRead < 0) { 5562 framesRead = bytesRead; 5563 } else { 5564 framesRead = bytesRead / mFrameSize; 5565 } 5566 } 5567 5568 if (framesRead < 0 || (framesRead == 0 && mPipeSource == 0)) { 5569 ALOGE("read failed: framesRead=%d", framesRead); 5570 // Force input into standby so that it tries to recover at next read attempt 5571 inputStandBy(); 5572 sleepUs = kRecordThreadSleepUs; 5573 } 5574 if (framesRead <= 0) { 5575 goto unlock; 5576 } 5577 ALOG_ASSERT(framesRead > 0); 5578 5579 if (mTeeSink != 0) { 5580 (void) mTeeSink->write(&mRsmpInBuffer[rear * mChannelCount], framesRead); 5581 } 5582 // If destination is non-contiguous, we now correct for reading past end of buffer. 5583 { 5584 size_t part1 = mRsmpInFramesP2 - rear; 5585 if ((size_t) framesRead > part1) { 5586 memcpy(mRsmpInBuffer, &mRsmpInBuffer[mRsmpInFramesP2 * mChannelCount], 5587 (framesRead - part1) * mFrameSize); 5588 } 5589 } 5590 rear = mRsmpInRear += framesRead; 5591 5592 size = activeTracks.size(); 5593 // loop over each active track 5594 for (size_t i = 0; i < size; i++) { 5595 activeTrack = activeTracks[i]; 5596 5597 // skip fast tracks, as those are handled directly by FastCapture 5598 if (activeTrack->isFastTrack()) { 5599 continue; 5600 } 5601 5602 // TODO: This code probably should be moved to RecordTrack. 5603 // TODO: Update the activeTrack buffer converter in case of reconfigure. 5604 5605 enum { 5606 OVERRUN_UNKNOWN, 5607 OVERRUN_TRUE, 5608 OVERRUN_FALSE 5609 } overrun = OVERRUN_UNKNOWN; 5610 5611 // loop over getNextBuffer to handle circular sink 5612 for (;;) { 5613 5614 activeTrack->mSink.frameCount = ~0; 5615 status_t status = activeTrack->getNextBuffer(&activeTrack->mSink); 5616 size_t framesOut = activeTrack->mSink.frameCount; 5617 LOG_ALWAYS_FATAL_IF((status == OK) != (framesOut > 0)); 5618 5619 // check available frames and handle overrun conditions 5620 // if the record track isn't draining fast enough. 5621 bool hasOverrun; 5622 size_t framesIn; 5623 activeTrack->mResamplerBufferProvider->sync(&framesIn, &hasOverrun); 5624 if (hasOverrun) { 5625 overrun = OVERRUN_TRUE; 5626 } 5627 if (framesOut == 0 || framesIn == 0) { 5628 break; 5629 } 5630 5631 // Don't allow framesOut to be larger than what is possible with resampling 5632 // from framesIn. 5633 // This isn't strictly necessary but helps limit buffer resizing in 5634 // RecordBufferConverter. TODO: remove when no longer needed. 5635 framesOut = min(framesOut, 5636 destinationFramesPossible( 5637 framesIn, mSampleRate, activeTrack->mSampleRate)); 5638 // process frames from the RecordThread buffer provider to the RecordTrack buffer 5639 framesOut = activeTrack->mRecordBufferConverter->convert( 5640 activeTrack->mSink.raw, activeTrack->mResamplerBufferProvider, framesOut); 5641 5642 if (framesOut > 0 && (overrun == OVERRUN_UNKNOWN)) { 5643 overrun = OVERRUN_FALSE; 5644 } 5645 5646 if (activeTrack->mFramesToDrop == 0) { 5647 if (framesOut > 0) { 5648 activeTrack->mSink.frameCount = framesOut; 5649 activeTrack->releaseBuffer(&activeTrack->mSink); 5650 } 5651 } else { 5652 // FIXME could do a partial drop of framesOut 5653 if (activeTrack->mFramesToDrop > 0) { 5654 activeTrack->mFramesToDrop -= framesOut; 5655 if (activeTrack->mFramesToDrop <= 0) { 5656 activeTrack->clearSyncStartEvent(); 5657 } 5658 } else { 5659 activeTrack->mFramesToDrop += framesOut; 5660 if (activeTrack->mFramesToDrop >= 0 || activeTrack->mSyncStartEvent == 0 || 5661 activeTrack->mSyncStartEvent->isCancelled()) { 5662 ALOGW("Synced record %s, session %d, trigger session %d", 5663 (activeTrack->mFramesToDrop >= 0) ? "timed out" : "cancelled", 5664 activeTrack->sessionId(), 5665 (activeTrack->mSyncStartEvent != 0) ? 5666 activeTrack->mSyncStartEvent->triggerSession() : 0); 5667 activeTrack->clearSyncStartEvent(); 5668 } 5669 } 5670 } 5671 5672 if (framesOut == 0) { 5673 break; 5674 } 5675 } 5676 5677 switch (overrun) { 5678 case OVERRUN_TRUE: 5679 // client isn't retrieving buffers fast enough 5680 if (!activeTrack->setOverflow()) { 5681 nsecs_t now = systemTime(); 5682 // FIXME should lastWarning per track? 5683 if ((now - lastWarning) > kWarningThrottleNs) { 5684 ALOGW("RecordThread: buffer overflow"); 5685 lastWarning = now; 5686 } 5687 } 5688 break; 5689 case OVERRUN_FALSE: 5690 activeTrack->clearOverflow(); 5691 break; 5692 case OVERRUN_UNKNOWN: 5693 break; 5694 } 5695 5696 } 5697 5698unlock: 5699 // enable changes in effect chain 5700 unlockEffectChains(effectChains); 5701 // effectChains doesn't need to be cleared, since it is cleared by destructor at scope end 5702 } 5703 5704 standbyIfNotAlreadyInStandby(); 5705 5706 { 5707 Mutex::Autolock _l(mLock); 5708 for (size_t i = 0; i < mTracks.size(); i++) { 5709 sp<RecordTrack> track = mTracks[i]; 5710 track->invalidate(); 5711 } 5712 mActiveTracks.clear(); 5713 mActiveTracksGen++; 5714 mStartStopCond.broadcast(); 5715 } 5716 5717 releaseWakeLock(); 5718 5719 ALOGV("RecordThread %p exiting", this); 5720 return false; 5721} 5722 5723void AudioFlinger::RecordThread::standbyIfNotAlreadyInStandby() 5724{ 5725 if (!mStandby) { 5726 inputStandBy(); 5727 mStandby = true; 5728 } 5729} 5730 5731void AudioFlinger::RecordThread::inputStandBy() 5732{ 5733 // Idle the fast capture if it's currently running 5734 if (mFastCapture != 0) { 5735 FastCaptureStateQueue *sq = mFastCapture->sq(); 5736 FastCaptureState *state = sq->begin(); 5737 if (!(state->mCommand & FastCaptureState::IDLE)) { 5738 state->mCommand = FastCaptureState::COLD_IDLE; 5739 state->mColdFutexAddr = &mFastCaptureFutex; 5740 state->mColdGen++; 5741 mFastCaptureFutex = 0; 5742 sq->end(); 5743 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 5744 sq->push(FastCaptureStateQueue::BLOCK_UNTIL_ACKED); 5745#if 0 5746 if (kUseFastCapture == FastCapture_Dynamic) { 5747 // FIXME 5748 } 5749#endif 5750#ifdef AUDIO_WATCHDOG 5751 // FIXME 5752#endif 5753 } else { 5754 sq->end(false /*didModify*/); 5755 } 5756 } 5757 mInput->stream->common.standby(&mInput->stream->common); 5758} 5759 5760// RecordThread::createRecordTrack_l() must be called with AudioFlinger::mLock held 5761sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 5762 const sp<AudioFlinger::Client>& client, 5763 uint32_t sampleRate, 5764 audio_format_t format, 5765 audio_channel_mask_t channelMask, 5766 size_t *pFrameCount, 5767 int sessionId, 5768 size_t *notificationFrames, 5769 int uid, 5770 IAudioFlinger::track_flags_t *flags, 5771 pid_t tid, 5772 status_t *status) 5773{ 5774 size_t frameCount = *pFrameCount; 5775 sp<RecordTrack> track; 5776 status_t lStatus; 5777 5778 // client expresses a preference for FAST, but we get the final say 5779 if (*flags & IAudioFlinger::TRACK_FAST) { 5780 if ( 5781 // we formerly checked for a callback handler (non-0 tid), 5782 // but that is no longer required for TRANSFER_OBTAIN mode 5783 // 5784 // frame count is not specified, or is exactly the pipe depth 5785 ((frameCount == 0) || (frameCount == mPipeFramesP2)) && 5786 // PCM data 5787 audio_is_linear_pcm(format) && 5788 // native format 5789 (format == mFormat) && 5790 // native channel mask 5791 (channelMask == mChannelMask) && 5792 // native hardware sample rate 5793 (sampleRate == mSampleRate) && 5794 // record thread has an associated fast capture 5795 hasFastCapture() && 5796 // there are sufficient fast track slots available 5797 mFastTrackAvail 5798 ) { 5799 ALOGV("AUDIO_INPUT_FLAG_FAST accepted: frameCount=%u mFrameCount=%u", 5800 frameCount, mFrameCount); 5801 } else { 5802 ALOGV("AUDIO_INPUT_FLAG_FAST denied: frameCount=%u mFrameCount=%u mPipeFramesP2=%u " 5803 "format=%#x isLinear=%d channelMask=%#x sampleRate=%u mSampleRate=%u " 5804 "hasFastCapture=%d tid=%d mFastTrackAvail=%d", 5805 frameCount, mFrameCount, mPipeFramesP2, 5806 format, audio_is_linear_pcm(format), channelMask, sampleRate, mSampleRate, 5807 hasFastCapture(), tid, mFastTrackAvail); 5808 *flags &= ~IAudioFlinger::TRACK_FAST; 5809 } 5810 } 5811 5812 // compute track buffer size in frames, and suggest the notification frame count 5813 if (*flags & IAudioFlinger::TRACK_FAST) { 5814 // fast track: frame count is exactly the pipe depth 5815 frameCount = mPipeFramesP2; 5816 // ignore requested notificationFrames, and always notify exactly once every HAL buffer 5817 *notificationFrames = mFrameCount; 5818 } else { 5819 // not fast track: max notification period is resampled equivalent of one HAL buffer time 5820 // or 20 ms if there is a fast capture 5821 // TODO This could be a roundupRatio inline, and const 5822 size_t maxNotificationFrames = ((int64_t) (hasFastCapture() ? mSampleRate/50 : mFrameCount) 5823 * sampleRate + mSampleRate - 1) / mSampleRate; 5824 // minimum number of notification periods is at least kMinNotifications, 5825 // and at least kMinMs rounded up to a whole notification period (minNotificationsByMs) 5826 static const size_t kMinNotifications = 3; 5827 static const uint32_t kMinMs = 30; 5828 // TODO This could be a roundupRatio inline 5829 const size_t minFramesByMs = (sampleRate * kMinMs + 1000 - 1) / 1000; 5830 // TODO This could be a roundupRatio inline 5831 const size_t minNotificationsByMs = (minFramesByMs + maxNotificationFrames - 1) / 5832 maxNotificationFrames; 5833 const size_t minFrameCount = maxNotificationFrames * 5834 max(kMinNotifications, minNotificationsByMs); 5835 frameCount = max(frameCount, minFrameCount); 5836 if (*notificationFrames == 0 || *notificationFrames > maxNotificationFrames) { 5837 *notificationFrames = maxNotificationFrames; 5838 } 5839 } 5840 *pFrameCount = frameCount; 5841 5842 lStatus = initCheck(); 5843 if (lStatus != NO_ERROR) { 5844 ALOGE("createRecordTrack_l() audio driver not initialized"); 5845 goto Exit; 5846 } 5847 5848 { // scope for mLock 5849 Mutex::Autolock _l(mLock); 5850 5851 track = new RecordTrack(this, client, sampleRate, 5852 format, channelMask, frameCount, NULL, sessionId, uid, 5853 *flags, TrackBase::TYPE_DEFAULT); 5854 5855 lStatus = track->initCheck(); 5856 if (lStatus != NO_ERROR) { 5857 ALOGE("createRecordTrack_l() initCheck failed %d; no control block?", lStatus); 5858 // track must be cleared from the caller as the caller has the AF lock 5859 goto Exit; 5860 } 5861 mTracks.add(track); 5862 5863 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 5864 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 5865 mAudioFlinger->btNrecIsOff(); 5866 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 5867 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 5868 5869 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 5870 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 5871 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 5872 // so ask activity manager to do this on our behalf 5873 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 5874 } 5875 } 5876 5877 lStatus = NO_ERROR; 5878 5879Exit: 5880 *status = lStatus; 5881 return track; 5882} 5883 5884status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, 5885 AudioSystem::sync_event_t event, 5886 int triggerSession) 5887{ 5888 ALOGV("RecordThread::start event %d, triggerSession %d", event, triggerSession); 5889 sp<ThreadBase> strongMe = this; 5890 status_t status = NO_ERROR; 5891 5892 if (event == AudioSystem::SYNC_EVENT_NONE) { 5893 recordTrack->clearSyncStartEvent(); 5894 } else if (event != AudioSystem::SYNC_EVENT_SAME) { 5895 recordTrack->mSyncStartEvent = mAudioFlinger->createSyncEvent(event, 5896 triggerSession, 5897 recordTrack->sessionId(), 5898 syncStartEventCallback, 5899 recordTrack); 5900 // Sync event can be cancelled by the trigger session if the track is not in a 5901 // compatible state in which case we start record immediately 5902 if (recordTrack->mSyncStartEvent->isCancelled()) { 5903 recordTrack->clearSyncStartEvent(); 5904 } else { 5905 // do not wait for the event for more than AudioSystem::kSyncRecordStartTimeOutMs 5906 recordTrack->mFramesToDrop = - 5907 ((AudioSystem::kSyncRecordStartTimeOutMs * recordTrack->mSampleRate) / 1000); 5908 } 5909 } 5910 5911 { 5912 // This section is a rendezvous between binder thread executing start() and RecordThread 5913 AutoMutex lock(mLock); 5914 if (mActiveTracks.indexOf(recordTrack) >= 0) { 5915 if (recordTrack->mState == TrackBase::PAUSING) { 5916 ALOGV("active record track PAUSING -> ACTIVE"); 5917 recordTrack->mState = TrackBase::ACTIVE; 5918 } else { 5919 ALOGV("active record track state %d", recordTrack->mState); 5920 } 5921 return status; 5922 } 5923 5924 // TODO consider other ways of handling this, such as changing the state to :STARTING and 5925 // adding the track to mActiveTracks after returning from AudioSystem::startInput(), 5926 // or using a separate command thread 5927 recordTrack->mState = TrackBase::STARTING_1; 5928 mActiveTracks.add(recordTrack); 5929 mActiveTracksGen++; 5930 status_t status = NO_ERROR; 5931 if (recordTrack->isExternalTrack()) { 5932 mLock.unlock(); 5933 status = AudioSystem::startInput(mId, (audio_session_t)recordTrack->sessionId()); 5934 mLock.lock(); 5935 // FIXME should verify that recordTrack is still in mActiveTracks 5936 if (status != NO_ERROR) { 5937 mActiveTracks.remove(recordTrack); 5938 mActiveTracksGen++; 5939 recordTrack->clearSyncStartEvent(); 5940 ALOGV("RecordThread::start error %d", status); 5941 return status; 5942 } 5943 } 5944 // Catch up with current buffer indices if thread is already running. 5945 // This is what makes a new client discard all buffered data. If the track's mRsmpInFront 5946 // was initialized to some value closer to the thread's mRsmpInFront, then the track could 5947 // see previously buffered data before it called start(), but with greater risk of overrun. 5948 5949 recordTrack->mResamplerBufferProvider->reset(); 5950 // clear any converter state as new data will be discontinuous 5951 recordTrack->mRecordBufferConverter->reset(); 5952 recordTrack->mState = TrackBase::STARTING_2; 5953 // signal thread to start 5954 mWaitWorkCV.broadcast(); 5955 if (mActiveTracks.indexOf(recordTrack) < 0) { 5956 ALOGV("Record failed to start"); 5957 status = BAD_VALUE; 5958 goto startError; 5959 } 5960 return status; 5961 } 5962 5963startError: 5964 if (recordTrack->isExternalTrack()) { 5965 AudioSystem::stopInput(mId, (audio_session_t)recordTrack->sessionId()); 5966 } 5967 recordTrack->clearSyncStartEvent(); 5968 // FIXME I wonder why we do not reset the state here? 5969 return status; 5970} 5971 5972void AudioFlinger::RecordThread::syncStartEventCallback(const wp<SyncEvent>& event) 5973{ 5974 sp<SyncEvent> strongEvent = event.promote(); 5975 5976 if (strongEvent != 0) { 5977 sp<RefBase> ptr = strongEvent->cookie().promote(); 5978 if (ptr != 0) { 5979 RecordTrack *recordTrack = (RecordTrack *)ptr.get(); 5980 recordTrack->handleSyncStartEvent(strongEvent); 5981 } 5982 } 5983} 5984 5985bool AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 5986 ALOGV("RecordThread::stop"); 5987 AutoMutex _l(mLock); 5988 if (mActiveTracks.indexOf(recordTrack) != 0 || recordTrack->mState == TrackBase::PAUSING) { 5989 return false; 5990 } 5991 // note that threadLoop may still be processing the track at this point [without lock] 5992 recordTrack->mState = TrackBase::PAUSING; 5993 // do not wait for mStartStopCond if exiting 5994 if (exitPending()) { 5995 return true; 5996 } 5997 // FIXME incorrect usage of wait: no explicit predicate or loop 5998 mStartStopCond.wait(mLock); 5999 // if we have been restarted, recordTrack is in mActiveTracks here 6000 if (exitPending() || mActiveTracks.indexOf(recordTrack) != 0) { 6001 ALOGV("Record stopped OK"); 6002 return true; 6003 } 6004 return false; 6005} 6006 6007bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event __unused) const 6008{ 6009 return false; 6010} 6011 6012status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event __unused) 6013{ 6014#if 0 // This branch is currently dead code, but is preserved in case it will be needed in future 6015 if (!isValidSyncEvent(event)) { 6016 return BAD_VALUE; 6017 } 6018 6019 int eventSession = event->triggerSession(); 6020 status_t ret = NAME_NOT_FOUND; 6021 6022 Mutex::Autolock _l(mLock); 6023 6024 for (size_t i = 0; i < mTracks.size(); i++) { 6025 sp<RecordTrack> track = mTracks[i]; 6026 if (eventSession == track->sessionId()) { 6027 (void) track->setSyncEvent(event); 6028 ret = NO_ERROR; 6029 } 6030 } 6031 return ret; 6032#else 6033 return BAD_VALUE; 6034#endif 6035} 6036 6037// destroyTrack_l() must be called with ThreadBase::mLock held 6038void AudioFlinger::RecordThread::destroyTrack_l(const sp<RecordTrack>& track) 6039{ 6040 track->terminate(); 6041 track->mState = TrackBase::STOPPED; 6042 // active tracks are removed by threadLoop() 6043 if (mActiveTracks.indexOf(track) < 0) { 6044 removeTrack_l(track); 6045 } 6046} 6047 6048void AudioFlinger::RecordThread::removeTrack_l(const sp<RecordTrack>& track) 6049{ 6050 mTracks.remove(track); 6051 // need anything related to effects here? 6052 if (track->isFastTrack()) { 6053 ALOG_ASSERT(!mFastTrackAvail); 6054 mFastTrackAvail = true; 6055 } 6056} 6057 6058void AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 6059{ 6060 dumpInternals(fd, args); 6061 dumpTracks(fd, args); 6062 dumpEffectChains(fd, args); 6063} 6064 6065void AudioFlinger::RecordThread::dumpInternals(int fd, const Vector<String16>& args) 6066{ 6067 dprintf(fd, "\nInput thread %p:\n", this); 6068 6069 dumpBase(fd, args); 6070 6071 if (mActiveTracks.size() == 0) { 6072 dprintf(fd, " No active record clients\n"); 6073 } 6074 dprintf(fd, " Fast capture thread: %s\n", hasFastCapture() ? "yes" : "no"); 6075 dprintf(fd, " Fast track available: %s\n", mFastTrackAvail ? "yes" : "no"); 6076 6077 // Make a non-atomic copy of fast capture dump state so it won't change underneath us 6078 const FastCaptureDumpState copy(mFastCaptureDumpState); 6079 copy.dump(fd); 6080} 6081 6082void AudioFlinger::RecordThread::dumpTracks(int fd, const Vector<String16>& args __unused) 6083{ 6084 const size_t SIZE = 256; 6085 char buffer[SIZE]; 6086 String8 result; 6087 6088 size_t numtracks = mTracks.size(); 6089 size_t numactive = mActiveTracks.size(); 6090 size_t numactiveseen = 0; 6091 dprintf(fd, " %d Tracks", numtracks); 6092 if (numtracks) { 6093 dprintf(fd, " of which %d are active\n", numactive); 6094 RecordTrack::appendDumpHeader(result); 6095 for (size_t i = 0; i < numtracks ; ++i) { 6096 sp<RecordTrack> track = mTracks[i]; 6097 if (track != 0) { 6098 bool active = mActiveTracks.indexOf(track) >= 0; 6099 if (active) { 6100 numactiveseen++; 6101 } 6102 track->dump(buffer, SIZE, active); 6103 result.append(buffer); 6104 } 6105 } 6106 } else { 6107 dprintf(fd, "\n"); 6108 } 6109 6110 if (numactiveseen != numactive) { 6111 snprintf(buffer, SIZE, " The following tracks are in the active list but" 6112 " not in the track list\n"); 6113 result.append(buffer); 6114 RecordTrack::appendDumpHeader(result); 6115 for (size_t i = 0; i < numactive; ++i) { 6116 sp<RecordTrack> track = mActiveTracks[i]; 6117 if (mTracks.indexOf(track) < 0) { 6118 track->dump(buffer, SIZE, true); 6119 result.append(buffer); 6120 } 6121 } 6122 6123 } 6124 write(fd, result.string(), result.size()); 6125} 6126 6127 6128void AudioFlinger::RecordThread::ResamplerBufferProvider::reset() 6129{ 6130 sp<ThreadBase> threadBase = mRecordTrack->mThread.promote(); 6131 RecordThread *recordThread = (RecordThread *) threadBase.get(); 6132 mRsmpInFront = recordThread->mRsmpInRear; 6133 mRsmpInUnrel = 0; 6134} 6135 6136void AudioFlinger::RecordThread::ResamplerBufferProvider::sync( 6137 size_t *framesAvailable, bool *hasOverrun) 6138{ 6139 sp<ThreadBase> threadBase = mRecordTrack->mThread.promote(); 6140 RecordThread *recordThread = (RecordThread *) threadBase.get(); 6141 const int32_t rear = recordThread->mRsmpInRear; 6142 const int32_t front = mRsmpInFront; 6143 const ssize_t filled = rear - front; 6144 6145 size_t framesIn; 6146 bool overrun = false; 6147 if (filled < 0) { 6148 // should not happen, but treat like a massive overrun and re-sync 6149 framesIn = 0; 6150 mRsmpInFront = rear; 6151 overrun = true; 6152 } else if ((size_t) filled <= recordThread->mRsmpInFrames) { 6153 framesIn = (size_t) filled; 6154 } else { 6155 // client is not keeping up with server, but give it latest data 6156 framesIn = recordThread->mRsmpInFrames; 6157 mRsmpInFront = /* front = */ rear - framesIn; 6158 overrun = true; 6159 } 6160 if (framesAvailable != NULL) { 6161 *framesAvailable = framesIn; 6162 } 6163 if (hasOverrun != NULL) { 6164 *hasOverrun = overrun; 6165 } 6166} 6167 6168// AudioBufferProvider interface 6169status_t AudioFlinger::RecordThread::ResamplerBufferProvider::getNextBuffer( 6170 AudioBufferProvider::Buffer* buffer, int64_t pts __unused) 6171{ 6172 sp<ThreadBase> threadBase = mRecordTrack->mThread.promote(); 6173 if (threadBase == 0) { 6174 buffer->frameCount = 0; 6175 buffer->raw = NULL; 6176 return NOT_ENOUGH_DATA; 6177 } 6178 RecordThread *recordThread = (RecordThread *) threadBase.get(); 6179 int32_t rear = recordThread->mRsmpInRear; 6180 int32_t front = mRsmpInFront; 6181 ssize_t filled = rear - front; 6182 // FIXME should not be P2 (don't want to increase latency) 6183 // FIXME if client not keeping up, discard 6184 LOG_ALWAYS_FATAL_IF(!(0 <= filled && (size_t) filled <= recordThread->mRsmpInFrames)); 6185 // 'filled' may be non-contiguous, so return only the first contiguous chunk 6186 front &= recordThread->mRsmpInFramesP2 - 1; 6187 size_t part1 = recordThread->mRsmpInFramesP2 - front; 6188 if (part1 > (size_t) filled) { 6189 part1 = filled; 6190 } 6191 size_t ask = buffer->frameCount; 6192 ALOG_ASSERT(ask > 0); 6193 if (part1 > ask) { 6194 part1 = ask; 6195 } 6196 if (part1 == 0) { 6197 // out of data is fine since the resampler will return a short-count. 6198 buffer->raw = NULL; 6199 buffer->frameCount = 0; 6200 mRsmpInUnrel = 0; 6201 return NOT_ENOUGH_DATA; 6202 } 6203 6204 buffer->raw = recordThread->mRsmpInBuffer + front * recordThread->mChannelCount; 6205 buffer->frameCount = part1; 6206 mRsmpInUnrel = part1; 6207 return NO_ERROR; 6208} 6209 6210// AudioBufferProvider interface 6211void AudioFlinger::RecordThread::ResamplerBufferProvider::releaseBuffer( 6212 AudioBufferProvider::Buffer* buffer) 6213{ 6214 size_t stepCount = buffer->frameCount; 6215 if (stepCount == 0) { 6216 return; 6217 } 6218 ALOG_ASSERT(stepCount <= mRsmpInUnrel); 6219 mRsmpInUnrel -= stepCount; 6220 mRsmpInFront += stepCount; 6221 buffer->raw = NULL; 6222 buffer->frameCount = 0; 6223} 6224 6225AudioFlinger::RecordThread::RecordBufferConverter::RecordBufferConverter( 6226 audio_channel_mask_t srcChannelMask, audio_format_t srcFormat, 6227 uint32_t srcSampleRate, 6228 audio_channel_mask_t dstChannelMask, audio_format_t dstFormat, 6229 uint32_t dstSampleRate) : 6230 mSrcChannelMask(AUDIO_CHANNEL_INVALID), // updateParameters will set following vars 6231 // mSrcFormat 6232 // mSrcSampleRate 6233 // mDstChannelMask 6234 // mDstFormat 6235 // mDstSampleRate 6236 // mSrcChannelCount 6237 // mDstChannelCount 6238 // mDstFrameSize 6239 mBuf(NULL), mBufFrames(0), mBufFrameSize(0), 6240 mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpOutFrameCount(0) 6241{ 6242 (void)updateParameters(srcChannelMask, srcFormat, srcSampleRate, 6243 dstChannelMask, dstFormat, dstSampleRate); 6244} 6245 6246AudioFlinger::RecordThread::RecordBufferConverter::~RecordBufferConverter() { 6247 free(mBuf); 6248 delete mResampler; 6249 free(mRsmpOutBuffer); 6250} 6251 6252size_t AudioFlinger::RecordThread::RecordBufferConverter::convert(void *dst, 6253 AudioBufferProvider *provider, size_t frames) 6254{ 6255 if (mSrcSampleRate == mDstSampleRate) { 6256 ALOGVV("NO RESAMPLING sampleRate:%u mSrcFormat:%#x mDstFormat:%#x", 6257 mSrcSampleRate, mSrcFormat, mDstFormat); 6258 6259 AudioBufferProvider::Buffer buffer; 6260 for (size_t i = frames; i > 0; ) { 6261 buffer.frameCount = i; 6262 status_t status = provider->getNextBuffer(&buffer, 0); 6263 if (status != OK || buffer.frameCount == 0) { 6264 frames -= i; // cannot fill request. 6265 break; 6266 } 6267 // convert to destination buffer 6268 convert(dst, buffer.raw, buffer.frameCount); 6269 6270 dst = (int8_t*)dst + buffer.frameCount * mDstFrameSize; 6271 i -= buffer.frameCount; 6272 provider->releaseBuffer(&buffer); 6273 } 6274 } else { 6275 ALOGVV("RESAMPLING mSrcSampleRate:%u mDstSampleRate:%u mSrcFormat:%#x mDstFormat:%#x", 6276 mSrcSampleRate, mDstSampleRate, mSrcFormat, mDstFormat); 6277 6278 // reallocate mRsmpOutBuffer as needed; we will grow but never shrink 6279 if (mRsmpOutFrameCount < frames) { 6280 // FIXME why does each track need it's own mRsmpOutBuffer? can't they share? 6281 free(mRsmpOutBuffer); 6282 // resampler always outputs stereo (FOR NOW) 6283 (void)posix_memalign(&mRsmpOutBuffer, 32, frames * FCC_2 * sizeof(int32_t) /*Q4.27*/); 6284 mRsmpOutFrameCount = frames; 6285 } 6286 // resampler accumulates, but we only have one source track 6287 memset(mRsmpOutBuffer, 0, frames * FCC_2 * sizeof(int32_t)); 6288 frames = mResampler->resample((int32_t*)mRsmpOutBuffer, frames, provider); 6289 6290 // convert to destination buffer 6291 convert(dst, mRsmpOutBuffer, frames); 6292 } 6293 return frames; 6294} 6295 6296status_t AudioFlinger::RecordThread::RecordBufferConverter::updateParameters( 6297 audio_channel_mask_t srcChannelMask, audio_format_t srcFormat, 6298 uint32_t srcSampleRate, 6299 audio_channel_mask_t dstChannelMask, audio_format_t dstFormat, 6300 uint32_t dstSampleRate) 6301{ 6302 // quick evaluation if there is any change. 6303 if (mSrcFormat == srcFormat 6304 && mSrcChannelMask == srcChannelMask 6305 && mSrcSampleRate == srcSampleRate 6306 && mDstFormat == dstFormat 6307 && mDstChannelMask == dstChannelMask 6308 && mDstSampleRate == dstSampleRate) { 6309 return NO_ERROR; 6310 } 6311 6312 const bool valid = 6313 audio_is_input_channel(srcChannelMask) 6314 && audio_is_input_channel(dstChannelMask) 6315 && audio_is_valid_format(srcFormat) && audio_is_linear_pcm(srcFormat) 6316 && audio_is_valid_format(dstFormat) && audio_is_linear_pcm(dstFormat) 6317 && (srcSampleRate <= dstSampleRate * AUDIO_RESAMPLER_DOWN_RATIO_MAX) 6318 ; // no upsampling checks for now 6319 if (!valid) { 6320 return BAD_VALUE; 6321 } 6322 6323 mSrcFormat = srcFormat; 6324 mSrcChannelMask = srcChannelMask; 6325 mSrcSampleRate = srcSampleRate; 6326 mDstFormat = dstFormat; 6327 mDstChannelMask = dstChannelMask; 6328 mDstSampleRate = dstSampleRate; 6329 6330 // compute derived parameters 6331 mSrcChannelCount = audio_channel_count_from_in_mask(srcChannelMask); 6332 mDstChannelCount = audio_channel_count_from_in_mask(dstChannelMask); 6333 mDstFrameSize = mDstChannelCount * audio_bytes_per_sample(mDstFormat); 6334 6335 // do we need a format buffer? 6336 if (mSrcFormat != mDstFormat && mDstChannelCount != mSrcChannelCount) { 6337 mBufFrameSize = mDstChannelCount * audio_bytes_per_sample(mSrcFormat); 6338 } else { 6339 mBufFrameSize = 0; 6340 } 6341 mBufFrames = 0; // force the buffer to be resized. 6342 6343 // do we need to resample? 6344 if (mSrcSampleRate != mDstSampleRate) { 6345 if (mResampler != NULL) { 6346 delete mResampler; 6347 } 6348 mResampler = AudioResampler::create(AUDIO_FORMAT_PCM_16_BIT, 6349 mSrcChannelCount, mDstSampleRate); // may seem confusing... 6350 mResampler->setSampleRate(mSrcSampleRate); 6351 mResampler->setVolume(AudioMixer::UNITY_GAIN_FLOAT, AudioMixer::UNITY_GAIN_FLOAT); 6352 } 6353 return NO_ERROR; 6354} 6355 6356void AudioFlinger::RecordThread::RecordBufferConverter::convert( 6357 void *dst, /*const*/ void *src, size_t frames) 6358{ 6359 // check if a memcpy will do 6360 if (mResampler == NULL 6361 && mSrcChannelCount == mDstChannelCount 6362 && mSrcFormat == mDstFormat) { 6363 memcpy(dst, src, 6364 frames * mDstChannelCount * audio_bytes_per_sample(mDstFormat)); 6365 return; 6366 } 6367 // reallocate buffer if needed 6368 if (mBufFrameSize != 0 && mBufFrames < frames) { 6369 free(mBuf); 6370 mBufFrames = frames; 6371 (void)posix_memalign(&mBuf, 32, mBufFrames * mBufFrameSize); 6372 } 6373 // do processing 6374 if (mResampler != NULL) { 6375 // src channel count is always >= 2. 6376 void *dstBuf = mBuf != NULL ? mBuf : dst; 6377 // ditherAndClamp() works as long as all buffers returned by 6378 // activeTrack->getNextBuffer() are 32 bit aligned which should be always true. 6379 if (mDstChannelCount == 1) { 6380 // the resampler always outputs stereo samples. 6381 // FIXME: this rewrites back into src 6382 ditherAndClamp((int32_t *)src, (const int32_t *)src, frames); 6383 downmix_to_mono_i16_from_stereo_i16((int16_t *)dstBuf, 6384 (const int16_t *)src, frames); 6385 } else { 6386 ditherAndClamp((int32_t *)dstBuf, (const int32_t *)src, frames); 6387 } 6388 } else if (mSrcChannelCount != mDstChannelCount) { 6389 void *dstBuf = mBuf != NULL ? mBuf : dst; 6390 if (mSrcChannelCount == 1) { 6391 upmix_to_stereo_i16_from_mono_i16((int16_t *)dstBuf, (const int16_t *)src, 6392 frames); 6393 } else { 6394 downmix_to_mono_i16_from_stereo_i16((int16_t *)dstBuf, 6395 (const int16_t *)src, frames); 6396 } 6397 } 6398 if (mSrcFormat != mDstFormat) { 6399 void *srcBuf = mBuf != NULL ? mBuf : src; 6400 memcpy_by_audio_format(dst, mDstFormat, srcBuf, mSrcFormat, 6401 frames * mDstChannelCount); 6402 } 6403} 6404 6405bool AudioFlinger::RecordThread::checkForNewParameter_l(const String8& keyValuePair, 6406 status_t& status) 6407{ 6408 bool reconfig = false; 6409 6410 status = NO_ERROR; 6411 6412 audio_format_t reqFormat = mFormat; 6413 uint32_t samplingRate = mSampleRate; 6414 audio_channel_mask_t channelMask = audio_channel_in_mask_from_count(mChannelCount); 6415 6416 AudioParameter param = AudioParameter(keyValuePair); 6417 int value; 6418 // TODO Investigate when this code runs. Check with audio policy when a sample rate and 6419 // channel count change can be requested. Do we mandate the first client defines the 6420 // HAL sampling rate and channel count or do we allow changes on the fly? 6421 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 6422 samplingRate = value; 6423 reconfig = true; 6424 } 6425 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 6426 if (!audio_is_linear_pcm((audio_format_t) value)) { 6427 status = BAD_VALUE; 6428 } else { 6429 reqFormat = (audio_format_t) value; 6430 reconfig = true; 6431 } 6432 } 6433 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 6434 audio_channel_mask_t mask = (audio_channel_mask_t) value; 6435 if (mask != AUDIO_CHANNEL_IN_MONO && mask != AUDIO_CHANNEL_IN_STEREO) { 6436 status = BAD_VALUE; 6437 } else { 6438 channelMask = mask; 6439 reconfig = true; 6440 } 6441 } 6442 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 6443 // do not accept frame count changes if tracks are open as the track buffer 6444 // size depends on frame count and correct behavior would not be guaranteed 6445 // if frame count is changed after track creation 6446 if (mActiveTracks.size() > 0) { 6447 status = INVALID_OPERATION; 6448 } else { 6449 reconfig = true; 6450 } 6451 } 6452 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 6453 // forward device change to effects that have requested to be 6454 // aware of attached audio device. 6455 for (size_t i = 0; i < mEffectChains.size(); i++) { 6456 mEffectChains[i]->setDevice_l(value); 6457 } 6458 6459 // store input device and output device but do not forward output device to audio HAL. 6460 // Note that status is ignored by the caller for output device 6461 // (see AudioFlinger::setParameters() 6462 if (audio_is_output_devices(value)) { 6463 mOutDevice = value; 6464 status = BAD_VALUE; 6465 } else { 6466 mInDevice = value; 6467 // disable AEC and NS if the device is a BT SCO headset supporting those 6468 // pre processings 6469 if (mTracks.size() > 0) { 6470 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 6471 mAudioFlinger->btNrecIsOff(); 6472 for (size_t i = 0; i < mTracks.size(); i++) { 6473 sp<RecordTrack> track = mTracks[i]; 6474 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId()); 6475 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId()); 6476 } 6477 } 6478 } 6479 } 6480 if (param.getInt(String8(AudioParameter::keyInputSource), value) == NO_ERROR && 6481 mAudioSource != (audio_source_t)value) { 6482 // forward device change to effects that have requested to be 6483 // aware of attached audio device. 6484 for (size_t i = 0; i < mEffectChains.size(); i++) { 6485 mEffectChains[i]->setAudioSource_l((audio_source_t)value); 6486 } 6487 mAudioSource = (audio_source_t)value; 6488 } 6489 6490 if (status == NO_ERROR) { 6491 status = mInput->stream->common.set_parameters(&mInput->stream->common, 6492 keyValuePair.string()); 6493 if (status == INVALID_OPERATION) { 6494 inputStandBy(); 6495 status = mInput->stream->common.set_parameters(&mInput->stream->common, 6496 keyValuePair.string()); 6497 } 6498 if (reconfig) { 6499 if (status == BAD_VALUE && 6500 audio_is_linear_pcm(mInput->stream->common.get_format(&mInput->stream->common)) && 6501 audio_is_linear_pcm(reqFormat) && 6502 (mInput->stream->common.get_sample_rate(&mInput->stream->common) 6503 <= (AUDIO_RESAMPLER_DOWN_RATIO_MAX * samplingRate)) && 6504 audio_channel_count_from_in_mask( 6505 mInput->stream->common.get_channels(&mInput->stream->common)) <= FCC_2 && 6506 (channelMask == AUDIO_CHANNEL_IN_MONO || 6507 channelMask == AUDIO_CHANNEL_IN_STEREO)) { 6508 status = NO_ERROR; 6509 } 6510 if (status == NO_ERROR) { 6511 readInputParameters_l(); 6512 sendIoConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 6513 } 6514 } 6515 } 6516 6517 return reconfig; 6518} 6519 6520String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 6521{ 6522 Mutex::Autolock _l(mLock); 6523 if (initCheck() != NO_ERROR) { 6524 return String8(); 6525 } 6526 6527 char *s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 6528 const String8 out_s8(s); 6529 free(s); 6530 return out_s8; 6531} 6532 6533void AudioFlinger::RecordThread::audioConfigChanged(int event, int param __unused) { 6534 AudioSystem::OutputDescriptor desc; 6535 const void *param2 = NULL; 6536 6537 switch (event) { 6538 case AudioSystem::INPUT_OPENED: 6539 case AudioSystem::INPUT_CONFIG_CHANGED: 6540 desc.channelMask = mChannelMask; 6541 desc.samplingRate = mSampleRate; 6542 desc.format = mFormat; 6543 desc.frameCount = mFrameCount; 6544 desc.latency = 0; 6545 param2 = &desc; 6546 break; 6547 6548 case AudioSystem::INPUT_CLOSED: 6549 default: 6550 break; 6551 } 6552 mAudioFlinger->audioConfigChanged(event, mId, param2); 6553} 6554 6555void AudioFlinger::RecordThread::readInputParameters_l() 6556{ 6557 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 6558 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 6559 mChannelCount = audio_channel_count_from_in_mask(mChannelMask); 6560 mHALFormat = mInput->stream->common.get_format(&mInput->stream->common); 6561 mFormat = mHALFormat; 6562 if (mFormat != AUDIO_FORMAT_PCM_16_BIT) { 6563 ALOGE("HAL format %#x not supported; must be AUDIO_FORMAT_PCM_16_BIT", mFormat); 6564 } 6565 mFrameSize = audio_stream_in_frame_size(mInput->stream); 6566 mBufferSize = mInput->stream->common.get_buffer_size(&mInput->stream->common); 6567 mFrameCount = mBufferSize / mFrameSize; 6568 // This is the formula for calculating the temporary buffer size. 6569 // With 7 HAL buffers, we can guarantee ability to down-sample the input by ratio of 6:1 to 6570 // 1 full output buffer, regardless of the alignment of the available input. 6571 // The value is somewhat arbitrary, and could probably be even larger. 6572 // A larger value should allow more old data to be read after a track calls start(), 6573 // without increasing latency. 6574 // 6575 // Note this is independent of the maximum downsampling ratio permitted for capture. 6576 mRsmpInFrames = mFrameCount * 7; 6577 mRsmpInFramesP2 = roundup(mRsmpInFrames); 6578 delete[] mRsmpInBuffer; 6579 6580 // TODO optimize audio capture buffer sizes ... 6581 // Here we calculate the size of the sliding buffer used as a source 6582 // for resampling. mRsmpInFramesP2 is currently roundup(mFrameCount * 7). 6583 // For current HAL frame counts, this is usually 2048 = 40 ms. It would 6584 // be better to have it derived from the pipe depth in the long term. 6585 // The current value is higher than necessary. However it should not add to latency. 6586 6587 // Over-allocate beyond mRsmpInFramesP2 to permit a HAL read past end of buffer 6588 mRsmpInBuffer = new int16_t[(mRsmpInFramesP2 + mFrameCount - 1) * mChannelCount]; 6589 6590 // AudioRecord mSampleRate and mChannelCount are constant due to AudioRecord API constraints. 6591 // But if thread's mSampleRate or mChannelCount changes, how will that affect active tracks? 6592} 6593 6594uint32_t AudioFlinger::RecordThread::getInputFramesLost() 6595{ 6596 Mutex::Autolock _l(mLock); 6597 if (initCheck() != NO_ERROR) { 6598 return 0; 6599 } 6600 6601 return mInput->stream->get_input_frames_lost(mInput->stream); 6602} 6603 6604uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) const 6605{ 6606 Mutex::Autolock _l(mLock); 6607 uint32_t result = 0; 6608 if (getEffectChain_l(sessionId) != 0) { 6609 result = EFFECT_SESSION; 6610 } 6611 6612 for (size_t i = 0; i < mTracks.size(); ++i) { 6613 if (sessionId == mTracks[i]->sessionId()) { 6614 result |= TRACK_SESSION; 6615 break; 6616 } 6617 } 6618 6619 return result; 6620} 6621 6622KeyedVector<int, bool> AudioFlinger::RecordThread::sessionIds() const 6623{ 6624 KeyedVector<int, bool> ids; 6625 Mutex::Autolock _l(mLock); 6626 for (size_t j = 0; j < mTracks.size(); ++j) { 6627 sp<RecordThread::RecordTrack> track = mTracks[j]; 6628 int sessionId = track->sessionId(); 6629 if (ids.indexOfKey(sessionId) < 0) { 6630 ids.add(sessionId, true); 6631 } 6632 } 6633 return ids; 6634} 6635 6636AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 6637{ 6638 Mutex::Autolock _l(mLock); 6639 AudioStreamIn *input = mInput; 6640 mInput = NULL; 6641 return input; 6642} 6643 6644// this method must always be called either with ThreadBase mLock held or inside the thread loop 6645audio_stream_t* AudioFlinger::RecordThread::stream() const 6646{ 6647 if (mInput == NULL) { 6648 return NULL; 6649 } 6650 return &mInput->stream->common; 6651} 6652 6653status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 6654{ 6655 // only one chain per input thread 6656 if (mEffectChains.size() != 0) { 6657 ALOGW("addEffectChain_l() already one chain %p on thread %p", chain.get(), this); 6658 return INVALID_OPERATION; 6659 } 6660 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 6661 chain->setThread(this); 6662 chain->setInBuffer(NULL); 6663 chain->setOutBuffer(NULL); 6664 6665 checkSuspendOnAddEffectChain_l(chain); 6666 6667 // make sure enabled pre processing effects state is communicated to the HAL as we 6668 // just moved them to a new input stream. 6669 chain->syncHalEffectsState(); 6670 6671 mEffectChains.add(chain); 6672 6673 return NO_ERROR; 6674} 6675 6676size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 6677{ 6678 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 6679 ALOGW_IF(mEffectChains.size() != 1, 6680 "removeEffectChain_l() %p invalid chain size %d on thread %p", 6681 chain.get(), mEffectChains.size(), this); 6682 if (mEffectChains.size() == 1) { 6683 mEffectChains.removeAt(0); 6684 } 6685 return 0; 6686} 6687 6688status_t AudioFlinger::RecordThread::createAudioPatch_l(const struct audio_patch *patch, 6689 audio_patch_handle_t *handle) 6690{ 6691 status_t status = NO_ERROR; 6692 if (mInput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 6693 // store new device and send to effects 6694 mInDevice = patch->sources[0].ext.device.type; 6695 for (size_t i = 0; i < mEffectChains.size(); i++) { 6696 mEffectChains[i]->setDevice_l(mInDevice); 6697 } 6698 6699 // disable AEC and NS if the device is a BT SCO headset supporting those 6700 // pre processings 6701 if (mTracks.size() > 0) { 6702 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 6703 mAudioFlinger->btNrecIsOff(); 6704 for (size_t i = 0; i < mTracks.size(); i++) { 6705 sp<RecordTrack> track = mTracks[i]; 6706 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId()); 6707 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId()); 6708 } 6709 } 6710 6711 // store new source and send to effects 6712 if (mAudioSource != patch->sinks[0].ext.mix.usecase.source) { 6713 mAudioSource = patch->sinks[0].ext.mix.usecase.source; 6714 for (size_t i = 0; i < mEffectChains.size(); i++) { 6715 mEffectChains[i]->setAudioSource_l(mAudioSource); 6716 } 6717 } 6718 6719 audio_hw_device_t *hwDevice = mInput->audioHwDev->hwDevice(); 6720 status = hwDevice->create_audio_patch(hwDevice, 6721 patch->num_sources, 6722 patch->sources, 6723 patch->num_sinks, 6724 patch->sinks, 6725 handle); 6726 } else { 6727 ALOG_ASSERT(false, "createAudioPatch_l() called on a pre 3.0 HAL"); 6728 } 6729 return status; 6730} 6731 6732status_t AudioFlinger::RecordThread::releaseAudioPatch_l(const audio_patch_handle_t handle) 6733{ 6734 status_t status = NO_ERROR; 6735 if (mInput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 6736 audio_hw_device_t *hwDevice = mInput->audioHwDev->hwDevice(); 6737 status = hwDevice->release_audio_patch(hwDevice, handle); 6738 } else { 6739 ALOG_ASSERT(false, "releaseAudioPatch_l() called on a pre 3.0 HAL"); 6740 } 6741 return status; 6742} 6743 6744void AudioFlinger::RecordThread::addPatchRecord(const sp<PatchRecord>& record) 6745{ 6746 Mutex::Autolock _l(mLock); 6747 mTracks.add(record); 6748} 6749 6750void AudioFlinger::RecordThread::deletePatchRecord(const sp<PatchRecord>& record) 6751{ 6752 Mutex::Autolock _l(mLock); 6753 destroyTrack_l(record); 6754} 6755 6756void AudioFlinger::RecordThread::getAudioPortConfig(struct audio_port_config *config) 6757{ 6758 ThreadBase::getAudioPortConfig(config); 6759 config->role = AUDIO_PORT_ROLE_SINK; 6760 config->ext.mix.hw_module = mInput->audioHwDev->handle(); 6761 config->ext.mix.usecase.source = mAudioSource; 6762} 6763 6764} // namespace android 6765