AudioFlinger.h revision 93c3d41bdb15e39dac0faea9c5b60f1637cd477c
1/* 2** 3** Copyright 2007, The Android Open Source Project 4** 5** Licensed under the Apache License, Version 2.0 (the "License"); 6** you may not use this file except in compliance with the License. 7** You may obtain a copy of the License at 8** 9** http://www.apache.org/licenses/LICENSE-2.0 10** 11** Unless required by applicable law or agreed to in writing, software 12** distributed under the License is distributed on an "AS IS" BASIS, 13** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. 14** See the License for the specific language governing permissions and 15** limitations under the License. 16*/ 17 18#ifndef ANDROID_AUDIO_FLINGER_H 19#define ANDROID_AUDIO_FLINGER_H 20 21#include "Configuration.h" 22#include <stdint.h> 23#include <sys/types.h> 24#include <limits.h> 25 26#include <common_time/cc_helper.h> 27 28#include <cutils/compiler.h> 29 30#include <media/IAudioFlinger.h> 31#include <media/IAudioFlingerClient.h> 32#include <media/IAudioTrack.h> 33#include <media/IAudioRecord.h> 34#include <media/AudioSystem.h> 35#include <media/AudioTrack.h> 36 37#include <utils/Atomic.h> 38#include <utils/Errors.h> 39#include <utils/threads.h> 40#include <utils/SortedVector.h> 41#include <utils/TypeHelpers.h> 42#include <utils/Vector.h> 43 44#include <binder/BinderService.h> 45#include <binder/MemoryDealer.h> 46 47#include <system/audio.h> 48#include <hardware/audio.h> 49#include <hardware/audio_policy.h> 50 51#include <media/AudioBufferProvider.h> 52#include <media/ExtendedAudioBufferProvider.h> 53 54#include "FastCapture.h" 55#include "FastMixer.h" 56#include <media/nbaio/NBAIO.h> 57#include "AudioWatchdog.h" 58#include "AudioMixer.h" 59 60#include <powermanager/IPowerManager.h> 61 62#include <media/nbaio/NBLog.h> 63#include <private/media/AudioTrackShared.h> 64 65namespace android { 66 67struct audio_track_cblk_t; 68struct effect_param_cblk_t; 69class AudioMixer; 70class AudioBuffer; 71class AudioResampler; 72class FastMixer; 73class ServerProxy; 74 75// ---------------------------------------------------------------------------- 76 77// AudioFlinger has a hard-coded upper limit of 2 channels for capture and playback. 78// There is support for > 2 channel tracks down-mixed to 2 channel output via a down-mix effect. 79// Adding full support for > 2 channel capture or playback would require more than simply changing 80// this #define. There is an independent hard-coded upper limit in AudioMixer; 81// removing that AudioMixer limit would be necessary but insufficient to support > 2 channels. 82// The macro FCC_2 highlights some (but not all) places where there is are 2-channel assumptions. 83// Search also for "2", "left", "right", "[0]", "[1]", ">> 16", "<< 16", etc. 84#define FCC_2 2 // FCC_2 = Fixed Channel Count 2 85 86static const nsecs_t kDefaultStandbyTimeInNsecs = seconds(3); 87 88#define INCLUDING_FROM_AUDIOFLINGER_H 89 90class AudioFlinger : 91 public BinderService<AudioFlinger>, 92 public BnAudioFlinger 93{ 94 friend class BinderService<AudioFlinger>; // for AudioFlinger() 95public: 96 static const char* getServiceName() ANDROID_API { return "media.audio_flinger"; } 97 98 virtual status_t dump(int fd, const Vector<String16>& args); 99 100 // IAudioFlinger interface, in binder opcode order 101 virtual sp<IAudioTrack> createTrack( 102 audio_stream_type_t streamType, 103 uint32_t sampleRate, 104 audio_format_t format, 105 audio_channel_mask_t channelMask, 106 size_t *pFrameCount, 107 IAudioFlinger::track_flags_t *flags, 108 const sp<IMemory>& sharedBuffer, 109 audio_io_handle_t output, 110 pid_t tid, 111 int *sessionId, 112 int clientUid, 113 status_t *status /*non-NULL*/); 114 115 virtual sp<IAudioRecord> openRecord( 116 audio_io_handle_t input, 117 uint32_t sampleRate, 118 audio_format_t format, 119 audio_channel_mask_t channelMask, 120 size_t *pFrameCount, 121 IAudioFlinger::track_flags_t *flags, 122 pid_t tid, 123 int *sessionId, 124 size_t *notificationFrames, 125 sp<IMemory>& cblk, 126 sp<IMemory>& buffers, 127 status_t *status /*non-NULL*/); 128 129 virtual uint32_t sampleRate(audio_io_handle_t output) const; 130 virtual audio_format_t format(audio_io_handle_t output) const; 131 virtual size_t frameCount(audio_io_handle_t output) const; 132 virtual uint32_t latency(audio_io_handle_t output) const; 133 134 virtual status_t setMasterVolume(float value); 135 virtual status_t setMasterMute(bool muted); 136 137 virtual float masterVolume() const; 138 virtual bool masterMute() const; 139 140 virtual status_t setStreamVolume(audio_stream_type_t stream, float value, 141 audio_io_handle_t output); 142 virtual status_t setStreamMute(audio_stream_type_t stream, bool muted); 143 144 virtual float streamVolume(audio_stream_type_t stream, 145 audio_io_handle_t output) const; 146 virtual bool streamMute(audio_stream_type_t stream) const; 147 148 virtual status_t setMode(audio_mode_t mode); 149 150 virtual status_t setMicMute(bool state); 151 virtual bool getMicMute() const; 152 153 virtual status_t setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs); 154 virtual String8 getParameters(audio_io_handle_t ioHandle, const String8& keys) const; 155 156 virtual void registerClient(const sp<IAudioFlingerClient>& client); 157 158 virtual size_t getInputBufferSize(uint32_t sampleRate, audio_format_t format, 159 audio_channel_mask_t channelMask) const; 160 161 virtual status_t openOutput(audio_module_handle_t module, 162 audio_io_handle_t *output, 163 audio_config_t *config, 164 audio_devices_t *devices, 165 const String8& address, 166 uint32_t *latencyMs, 167 audio_output_flags_t flags); 168 169 virtual audio_io_handle_t openDuplicateOutput(audio_io_handle_t output1, 170 audio_io_handle_t output2); 171 172 virtual status_t closeOutput(audio_io_handle_t output); 173 174 virtual status_t suspendOutput(audio_io_handle_t output); 175 176 virtual status_t restoreOutput(audio_io_handle_t output); 177 178 virtual status_t openInput(audio_module_handle_t module, 179 audio_io_handle_t *input, 180 audio_config_t *config, 181 audio_devices_t *device, 182 const String8& address, 183 audio_source_t source, 184 audio_input_flags_t flags); 185 186 virtual status_t closeInput(audio_io_handle_t input); 187 188 virtual status_t invalidateStream(audio_stream_type_t stream); 189 190 virtual status_t setVoiceVolume(float volume); 191 192 virtual status_t getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, 193 audio_io_handle_t output) const; 194 195 virtual uint32_t getInputFramesLost(audio_io_handle_t ioHandle) const; 196 197 virtual audio_unique_id_t newAudioUniqueId(); 198 199 virtual void acquireAudioSessionId(int audioSession, pid_t pid); 200 201 virtual void releaseAudioSessionId(int audioSession, pid_t pid); 202 203 virtual status_t queryNumberEffects(uint32_t *numEffects) const; 204 205 virtual status_t queryEffect(uint32_t index, effect_descriptor_t *descriptor) const; 206 207 virtual status_t getEffectDescriptor(const effect_uuid_t *pUuid, 208 effect_descriptor_t *descriptor) const; 209 210 virtual sp<IEffect> createEffect( 211 effect_descriptor_t *pDesc, 212 const sp<IEffectClient>& effectClient, 213 int32_t priority, 214 audio_io_handle_t io, 215 int sessionId, 216 status_t *status /*non-NULL*/, 217 int *id, 218 int *enabled); 219 220 virtual status_t moveEffects(int sessionId, audio_io_handle_t srcOutput, 221 audio_io_handle_t dstOutput); 222 223 virtual audio_module_handle_t loadHwModule(const char *name); 224 225 virtual uint32_t getPrimaryOutputSamplingRate(); 226 virtual size_t getPrimaryOutputFrameCount(); 227 228 virtual status_t setLowRamDevice(bool isLowRamDevice); 229 230 /* List available audio ports and their attributes */ 231 virtual status_t listAudioPorts(unsigned int *num_ports, 232 struct audio_port *ports); 233 234 /* Get attributes for a given audio port */ 235 virtual status_t getAudioPort(struct audio_port *port); 236 237 /* Create an audio patch between several source and sink ports */ 238 virtual status_t createAudioPatch(const struct audio_patch *patch, 239 audio_patch_handle_t *handle); 240 241 /* Release an audio patch */ 242 virtual status_t releaseAudioPatch(audio_patch_handle_t handle); 243 244 /* List existing audio patches */ 245 virtual status_t listAudioPatches(unsigned int *num_patches, 246 struct audio_patch *patches); 247 248 /* Set audio port configuration */ 249 virtual status_t setAudioPortConfig(const struct audio_port_config *config); 250 251 /* Get the HW synchronization source used for an audio session */ 252 virtual audio_hw_sync_t getAudioHwSyncForSession(audio_session_t sessionId); 253 254 virtual status_t onTransact( 255 uint32_t code, 256 const Parcel& data, 257 Parcel* reply, 258 uint32_t flags); 259 260 // end of IAudioFlinger interface 261 262 sp<NBLog::Writer> newWriter_l(size_t size, const char *name); 263 void unregisterWriter(const sp<NBLog::Writer>& writer); 264private: 265 static const size_t kLogMemorySize = 40 * 1024; 266 sp<MemoryDealer> mLogMemoryDealer; // == 0 when NBLog is disabled 267 // When a log writer is unregistered, it is done lazily so that media.log can continue to see it 268 // for as long as possible. The memory is only freed when it is needed for another log writer. 269 Vector< sp<NBLog::Writer> > mUnregisteredWriters; 270 Mutex mUnregisteredWritersLock; 271public: 272 273 class SyncEvent; 274 275 typedef void (*sync_event_callback_t)(const wp<SyncEvent>& event) ; 276 277 class SyncEvent : public RefBase { 278 public: 279 SyncEvent(AudioSystem::sync_event_t type, 280 int triggerSession, 281 int listenerSession, 282 sync_event_callback_t callBack, 283 wp<RefBase> cookie) 284 : mType(type), mTriggerSession(triggerSession), mListenerSession(listenerSession), 285 mCallback(callBack), mCookie(cookie) 286 {} 287 288 virtual ~SyncEvent() {} 289 290 void trigger() { Mutex::Autolock _l(mLock); if (mCallback) mCallback(this); } 291 bool isCancelled() const { Mutex::Autolock _l(mLock); return (mCallback == NULL); } 292 void cancel() { Mutex::Autolock _l(mLock); mCallback = NULL; } 293 AudioSystem::sync_event_t type() const { return mType; } 294 int triggerSession() const { return mTriggerSession; } 295 int listenerSession() const { return mListenerSession; } 296 wp<RefBase> cookie() const { return mCookie; } 297 298 private: 299 const AudioSystem::sync_event_t mType; 300 const int mTriggerSession; 301 const int mListenerSession; 302 sync_event_callback_t mCallback; 303 const wp<RefBase> mCookie; 304 mutable Mutex mLock; 305 }; 306 307 sp<SyncEvent> createSyncEvent(AudioSystem::sync_event_t type, 308 int triggerSession, 309 int listenerSession, 310 sync_event_callback_t callBack, 311 wp<RefBase> cookie); 312 313private: 314 class AudioHwDevice; // fwd declaration for findSuitableHwDev_l 315 316 audio_mode_t getMode() const { return mMode; } 317 318 bool btNrecIsOff() const { return mBtNrecIsOff; } 319 320 AudioFlinger() ANDROID_API; 321 virtual ~AudioFlinger(); 322 323 // call in any IAudioFlinger method that accesses mPrimaryHardwareDev 324 status_t initCheck() const { return mPrimaryHardwareDev == NULL ? 325 NO_INIT : NO_ERROR; } 326 327 // RefBase 328 virtual void onFirstRef(); 329 330 AudioHwDevice* findSuitableHwDev_l(audio_module_handle_t module, 331 audio_devices_t devices); 332 void purgeStaleEffects_l(); 333 334 // Set kEnableExtendedChannels to true to enable greater than stereo output 335 // for the MixerThread and device sink. Number of channels allowed is 336 // FCC_2 <= channels <= AudioMixer::MAX_NUM_CHANNELS. 337 static const bool kEnableExtendedChannels = true; 338 339 // Returns true if channel mask is permitted for the PCM sink in the MixerThread 340 static inline bool isValidPcmSinkChannelMask(audio_channel_mask_t channelMask) { 341 switch (audio_channel_mask_get_representation(channelMask)) { 342 case AUDIO_CHANNEL_REPRESENTATION_POSITION: { 343 uint32_t channelCount = FCC_2; // stereo is default 344 if (kEnableExtendedChannels) { 345 channelCount = audio_channel_count_from_out_mask(channelMask); 346 if (channelCount > AudioMixer::MAX_NUM_CHANNELS) { 347 return false; 348 } 349 } 350 // check that channelMask is the "canonical" one we expect for the channelCount. 351 return channelMask == audio_channel_out_mask_from_count(channelCount); 352 } 353 default: 354 return false; 355 } 356 } 357 358 // Set kEnableExtendedPrecision to true to use extended precision in MixerThread 359 static const bool kEnableExtendedPrecision = true; 360 361 // Returns true if format is permitted for the PCM sink in the MixerThread 362 static inline bool isValidPcmSinkFormat(audio_format_t format) { 363 switch (format) { 364 case AUDIO_FORMAT_PCM_16_BIT: 365 return true; 366 case AUDIO_FORMAT_PCM_FLOAT: 367 case AUDIO_FORMAT_PCM_24_BIT_PACKED: 368 case AUDIO_FORMAT_PCM_32_BIT: 369 case AUDIO_FORMAT_PCM_8_24_BIT: 370 return kEnableExtendedPrecision; 371 default: 372 return false; 373 } 374 } 375 376 // standby delay for MIXER and DUPLICATING playback threads is read from property 377 // ro.audio.flinger_standbytime_ms or defaults to kDefaultStandbyTimeInNsecs 378 static nsecs_t mStandbyTimeInNsecs; 379 380 // incremented by 2 when screen state changes, bit 0 == 1 means "off" 381 // AudioFlinger::setParameters() updates, other threads read w/o lock 382 static uint32_t mScreenState; 383 384 // Internal dump utilities. 385 static const int kDumpLockRetries = 50; 386 static const int kDumpLockSleepUs = 20000; 387 static bool dumpTryLock(Mutex& mutex); 388 void dumpPermissionDenial(int fd, const Vector<String16>& args); 389 void dumpClients(int fd, const Vector<String16>& args); 390 void dumpInternals(int fd, const Vector<String16>& args); 391 392 // --- Client --- 393 class Client : public RefBase { 394 public: 395 Client(const sp<AudioFlinger>& audioFlinger, pid_t pid); 396 virtual ~Client(); 397 sp<MemoryDealer> heap() const; 398 pid_t pid() const { return mPid; } 399 sp<AudioFlinger> audioFlinger() const { return mAudioFlinger; } 400 401 bool reserveTimedTrack(); 402 void releaseTimedTrack(); 403 404 private: 405 Client(const Client&); 406 Client& operator = (const Client&); 407 const sp<AudioFlinger> mAudioFlinger; 408 const sp<MemoryDealer> mMemoryDealer; 409 const pid_t mPid; 410 411 Mutex mTimedTrackLock; 412 int mTimedTrackCount; 413 }; 414 415 // --- Notification Client --- 416 class NotificationClient : public IBinder::DeathRecipient { 417 public: 418 NotificationClient(const sp<AudioFlinger>& audioFlinger, 419 const sp<IAudioFlingerClient>& client, 420 pid_t pid); 421 virtual ~NotificationClient(); 422 423 sp<IAudioFlingerClient> audioFlingerClient() const { return mAudioFlingerClient; } 424 425 // IBinder::DeathRecipient 426 virtual void binderDied(const wp<IBinder>& who); 427 428 private: 429 NotificationClient(const NotificationClient&); 430 NotificationClient& operator = (const NotificationClient&); 431 432 const sp<AudioFlinger> mAudioFlinger; 433 const pid_t mPid; 434 const sp<IAudioFlingerClient> mAudioFlingerClient; 435 }; 436 437 class TrackHandle; 438 class RecordHandle; 439 class RecordThread; 440 class PlaybackThread; 441 class MixerThread; 442 class DirectOutputThread; 443 class OffloadThread; 444 class DuplicatingThread; 445 class AsyncCallbackThread; 446 class Track; 447 class RecordTrack; 448 class EffectModule; 449 class EffectHandle; 450 class EffectChain; 451 struct AudioStreamOut; 452 struct AudioStreamIn; 453 454 struct stream_type_t { 455 stream_type_t() 456 : volume(1.0f), 457 mute(false) 458 { 459 } 460 float volume; 461 bool mute; 462 }; 463 464 // --- PlaybackThread --- 465 466#include "Threads.h" 467 468#include "Effects.h" 469 470#include "PatchPanel.h" 471 472 // server side of the client's IAudioTrack 473 class TrackHandle : public android::BnAudioTrack { 474 public: 475 TrackHandle(const sp<PlaybackThread::Track>& track); 476 virtual ~TrackHandle(); 477 virtual sp<IMemory> getCblk() const; 478 virtual status_t start(); 479 virtual void stop(); 480 virtual void flush(); 481 virtual void pause(); 482 virtual status_t attachAuxEffect(int effectId); 483 virtual status_t allocateTimedBuffer(size_t size, 484 sp<IMemory>* buffer); 485 virtual status_t queueTimedBuffer(const sp<IMemory>& buffer, 486 int64_t pts); 487 virtual status_t setMediaTimeTransform(const LinearTransform& xform, 488 int target); 489 virtual status_t setParameters(const String8& keyValuePairs); 490 virtual status_t getTimestamp(AudioTimestamp& timestamp); 491 virtual void signal(); // signal playback thread for a change in control block 492 493 virtual status_t onTransact( 494 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags); 495 496 private: 497 const sp<PlaybackThread::Track> mTrack; 498 }; 499 500 // server side of the client's IAudioRecord 501 class RecordHandle : public android::BnAudioRecord { 502 public: 503 RecordHandle(const sp<RecordThread::RecordTrack>& recordTrack); 504 virtual ~RecordHandle(); 505 virtual status_t start(int /*AudioSystem::sync_event_t*/ event, int triggerSession); 506 virtual void stop(); 507 virtual status_t onTransact( 508 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags); 509 private: 510 const sp<RecordThread::RecordTrack> mRecordTrack; 511 512 // for use from destructor 513 void stop_nonvirtual(); 514 }; 515 516 517 PlaybackThread *checkPlaybackThread_l(audio_io_handle_t output) const; 518 MixerThread *checkMixerThread_l(audio_io_handle_t output) const; 519 RecordThread *checkRecordThread_l(audio_io_handle_t input) const; 520 sp<RecordThread> openInput_l(audio_module_handle_t module, 521 audio_io_handle_t *input, 522 audio_config_t *config, 523 audio_devices_t device, 524 const String8& address, 525 audio_source_t source, 526 audio_input_flags_t flags); 527 sp<PlaybackThread> openOutput_l(audio_module_handle_t module, 528 audio_io_handle_t *output, 529 audio_config_t *config, 530 audio_devices_t devices, 531 const String8& address, 532 audio_output_flags_t flags); 533 534 void closeOutputFinish(sp<PlaybackThread> thread); 535 void closeInputFinish(sp<RecordThread> thread); 536 537 // no range check, AudioFlinger::mLock held 538 bool streamMute_l(audio_stream_type_t stream) const 539 { return mStreamTypes[stream].mute; } 540 // no range check, doesn't check per-thread stream volume, AudioFlinger::mLock held 541 float streamVolume_l(audio_stream_type_t stream) const 542 { return mStreamTypes[stream].volume; } 543 void audioConfigChanged(int event, audio_io_handle_t ioHandle, const void *param2); 544 545 // Allocate an audio_io_handle_t, session ID, effect ID, or audio_module_handle_t. 546 // They all share the same ID space, but the namespaces are actually independent 547 // because there are separate KeyedVectors for each kind of ID. 548 // The return value is uint32_t, but is cast to signed for some IDs. 549 // FIXME This API does not handle rollover to zero (for unsigned IDs), 550 // or from positive to negative (for signed IDs). 551 // Thus it may fail by returning an ID of the wrong sign, 552 // or by returning a non-unique ID. 553 uint32_t nextUniqueId(); 554 555 status_t moveEffectChain_l(int sessionId, 556 PlaybackThread *srcThread, 557 PlaybackThread *dstThread, 558 bool reRegister); 559 // return thread associated with primary hardware device, or NULL 560 PlaybackThread *primaryPlaybackThread_l() const; 561 audio_devices_t primaryOutputDevice_l() const; 562 563 sp<PlaybackThread> getEffectThread_l(int sessionId, int EffectId); 564 565 566 void removeClient_l(pid_t pid); 567 void removeNotificationClient(pid_t pid); 568 bool isNonOffloadableGlobalEffectEnabled_l(); 569 void onNonOffloadableGlobalEffectEnable(); 570 571 class AudioHwDevice { 572 public: 573 enum Flags { 574 AHWD_CAN_SET_MASTER_VOLUME = 0x1, 575 AHWD_CAN_SET_MASTER_MUTE = 0x2, 576 }; 577 578 AudioHwDevice(audio_module_handle_t handle, 579 const char *moduleName, 580 audio_hw_device_t *hwDevice, 581 Flags flags) 582 : mHandle(handle), mModuleName(strdup(moduleName)) 583 , mHwDevice(hwDevice) 584 , mFlags(flags) { } 585 /*virtual*/ ~AudioHwDevice() { free((void *)mModuleName); } 586 587 bool canSetMasterVolume() const { 588 return (0 != (mFlags & AHWD_CAN_SET_MASTER_VOLUME)); 589 } 590 591 bool canSetMasterMute() const { 592 return (0 != (mFlags & AHWD_CAN_SET_MASTER_MUTE)); 593 } 594 595 audio_module_handle_t handle() const { return mHandle; } 596 const char *moduleName() const { return mModuleName; } 597 audio_hw_device_t *hwDevice() const { return mHwDevice; } 598 uint32_t version() const { return mHwDevice->common.version; } 599 600 private: 601 const audio_module_handle_t mHandle; 602 const char * const mModuleName; 603 audio_hw_device_t * const mHwDevice; 604 const Flags mFlags; 605 }; 606 607 // AudioStreamOut and AudioStreamIn are immutable, so their fields are const. 608 // For emphasis, we could also make all pointers to them be "const *", 609 // but that would clutter the code unnecessarily. 610 611 struct AudioStreamOut { 612 AudioHwDevice* const audioHwDev; 613 audio_stream_out_t* const stream; 614 const audio_output_flags_t flags; 615 616 audio_hw_device_t* hwDev() const { return audioHwDev->hwDevice(); } 617 618 AudioStreamOut(AudioHwDevice *dev, audio_stream_out_t *out, audio_output_flags_t flags) : 619 audioHwDev(dev), stream(out), flags(flags) {} 620 }; 621 622 struct AudioStreamIn { 623 AudioHwDevice* const audioHwDev; 624 audio_stream_in_t* const stream; 625 626 audio_hw_device_t* hwDev() const { return audioHwDev->hwDevice(); } 627 628 AudioStreamIn(AudioHwDevice *dev, audio_stream_in_t *in) : 629 audioHwDev(dev), stream(in) {} 630 }; 631 632 // for mAudioSessionRefs only 633 struct AudioSessionRef { 634 AudioSessionRef(int sessionid, pid_t pid) : 635 mSessionid(sessionid), mPid(pid), mCnt(1) {} 636 const int mSessionid; 637 const pid_t mPid; 638 int mCnt; 639 }; 640 641 mutable Mutex mLock; 642 // protects mClients and mNotificationClients. 643 // must be locked after mLock and ThreadBase::mLock if both must be locked 644 // avoids acquiring AudioFlinger::mLock from inside thread loop. 645 mutable Mutex mClientLock; 646 // protected by mClientLock 647 DefaultKeyedVector< pid_t, wp<Client> > mClients; // see ~Client() 648 649 mutable Mutex mHardwareLock; 650 // NOTE: If both mLock and mHardwareLock mutexes must be held, 651 // always take mLock before mHardwareLock 652 653 // These two fields are immutable after onFirstRef(), so no lock needed to access 654 AudioHwDevice* mPrimaryHardwareDev; // mAudioHwDevs[0] or NULL 655 DefaultKeyedVector<audio_module_handle_t, AudioHwDevice*> mAudioHwDevs; 656 657 // for dump, indicates which hardware operation is currently in progress (but not stream ops) 658 enum hardware_call_state { 659 AUDIO_HW_IDLE = 0, // no operation in progress 660 AUDIO_HW_INIT, // init_check 661 AUDIO_HW_OUTPUT_OPEN, // open_output_stream 662 AUDIO_HW_OUTPUT_CLOSE, // unused 663 AUDIO_HW_INPUT_OPEN, // unused 664 AUDIO_HW_INPUT_CLOSE, // unused 665 AUDIO_HW_STANDBY, // unused 666 AUDIO_HW_SET_MASTER_VOLUME, // set_master_volume 667 AUDIO_HW_GET_ROUTING, // unused 668 AUDIO_HW_SET_ROUTING, // unused 669 AUDIO_HW_GET_MODE, // unused 670 AUDIO_HW_SET_MODE, // set_mode 671 AUDIO_HW_GET_MIC_MUTE, // get_mic_mute 672 AUDIO_HW_SET_MIC_MUTE, // set_mic_mute 673 AUDIO_HW_SET_VOICE_VOLUME, // set_voice_volume 674 AUDIO_HW_SET_PARAMETER, // set_parameters 675 AUDIO_HW_GET_INPUT_BUFFER_SIZE, // get_input_buffer_size 676 AUDIO_HW_GET_MASTER_VOLUME, // get_master_volume 677 AUDIO_HW_GET_PARAMETER, // get_parameters 678 AUDIO_HW_SET_MASTER_MUTE, // set_master_mute 679 AUDIO_HW_GET_MASTER_MUTE, // get_master_mute 680 }; 681 682 mutable hardware_call_state mHardwareStatus; // for dump only 683 684 685 DefaultKeyedVector< audio_io_handle_t, sp<PlaybackThread> > mPlaybackThreads; 686 stream_type_t mStreamTypes[AUDIO_STREAM_CNT]; 687 688 // member variables below are protected by mLock 689 float mMasterVolume; 690 bool mMasterMute; 691 // end of variables protected by mLock 692 693 DefaultKeyedVector< audio_io_handle_t, sp<RecordThread> > mRecordThreads; 694 695 // protected by mClientLock 696 DefaultKeyedVector< pid_t, sp<NotificationClient> > mNotificationClients; 697 698 volatile int32_t mNextUniqueId; // updated by android_atomic_inc 699 // nextUniqueId() returns uint32_t, but this is declared int32_t 700 // because the atomic operations require an int32_t 701 702 audio_mode_t mMode; 703 bool mBtNrecIsOff; 704 705 // protected by mLock 706 Vector<AudioSessionRef*> mAudioSessionRefs; 707 708 float masterVolume_l() const; 709 bool masterMute_l() const; 710 audio_module_handle_t loadHwModule_l(const char *name); 711 712 Vector < sp<SyncEvent> > mPendingSyncEvents; // sync events awaiting for a session 713 // to be created 714 715private: 716 sp<Client> registerPid(pid_t pid); // always returns non-0 717 718 // for use from destructor 719 status_t closeOutput_nonvirtual(audio_io_handle_t output); 720 void closeOutputInternal_l(sp<PlaybackThread> thread); 721 status_t closeInput_nonvirtual(audio_io_handle_t input); 722 void closeInputInternal_l(sp<RecordThread> thread); 723 724#ifdef TEE_SINK 725 // all record threads serially share a common tee sink, which is re-created on format change 726 sp<NBAIO_Sink> mRecordTeeSink; 727 sp<NBAIO_Source> mRecordTeeSource; 728#endif 729 730public: 731 732#ifdef TEE_SINK 733 // tee sink, if enabled by property, allows dumpsys to write most recent audio to .wav file 734 static void dumpTee(int fd, const sp<NBAIO_Source>& source, audio_io_handle_t id = 0); 735 736 // whether tee sink is enabled by property 737 static bool mTeeSinkInputEnabled; 738 static bool mTeeSinkOutputEnabled; 739 static bool mTeeSinkTrackEnabled; 740 741 // runtime configured size of each tee sink pipe, in frames 742 static size_t mTeeSinkInputFrames; 743 static size_t mTeeSinkOutputFrames; 744 static size_t mTeeSinkTrackFrames; 745 746 // compile-time default size of tee sink pipes, in frames 747 // 0x200000 stereo 16-bit PCM frames = 47.5 seconds at 44.1 kHz, 8 megabytes 748 static const size_t kTeeSinkInputFramesDefault = 0x200000; 749 static const size_t kTeeSinkOutputFramesDefault = 0x200000; 750 static const size_t kTeeSinkTrackFramesDefault = 0x200000; 751#endif 752 753 // This method reads from a variable without mLock, but the variable is updated under mLock. So 754 // we might read a stale value, or a value that's inconsistent with respect to other variables. 755 // In this case, it's safe because the return value isn't used for making an important decision. 756 // The reason we don't want to take mLock is because it could block the caller for a long time. 757 bool isLowRamDevice() const { return mIsLowRamDevice; } 758 759private: 760 bool mIsLowRamDevice; 761 bool mIsDeviceTypeKnown; 762 nsecs_t mGlobalEffectEnableTime; // when a global effect was last enabled 763 764 sp<PatchPanel> mPatchPanel; 765 766 uint32_t mPrimaryOutputSampleRate; // sample rate of the primary output, or zero if none 767 // protected by mHardwareLock 768}; 769 770#undef INCLUDING_FROM_AUDIOFLINGER_H 771 772const char *formatToString(audio_format_t format); 773 774// ---------------------------------------------------------------------------- 775 776}; // namespace android 777 778#endif // ANDROID_AUDIO_FLINGER_H 779