AudioFlinger.h revision e4756fe3a387615acb63c6a05788c8db9b5786cb
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 <stdint.h> 22#include <sys/types.h> 23#include <limits.h> 24 25#include <common_time/cc_helper.h> 26 27#include <media/IAudioFlinger.h> 28#include <media/IAudioFlingerClient.h> 29#include <media/IAudioTrack.h> 30#include <media/IAudioRecord.h> 31#include <media/AudioSystem.h> 32#include <media/AudioTrack.h> 33 34#include <utils/Atomic.h> 35#include <utils/Errors.h> 36#include <utils/threads.h> 37#include <utils/SortedVector.h> 38#include <utils/TypeHelpers.h> 39#include <utils/Vector.h> 40 41#include <binder/BinderService.h> 42#include <binder/MemoryDealer.h> 43 44#include <system/audio.h> 45#include <hardware/audio.h> 46#include <hardware/audio_policy.h> 47 48#include <media/AudioBufferProvider.h> 49#include <media/ExtendedAudioBufferProvider.h> 50#include "FastMixer.h" 51#include <media/nbaio/NBAIO.h> 52#include "AudioWatchdog.h" 53 54#include <powermanager/IPowerManager.h> 55 56namespace android { 57 58class audio_track_cblk_t; 59class effect_param_cblk_t; 60class AudioMixer; 61class AudioBuffer; 62class AudioResampler; 63class FastMixer; 64 65// ---------------------------------------------------------------------------- 66 67// AudioFlinger has a hard-coded upper limit of 2 channels for capture and playback. 68// There is support for > 2 channel tracks down-mixed to 2 channel output via a down-mix effect. 69// Adding full support for > 2 channel capture or playback would require more than simply changing 70// this #define. There is an independent hard-coded upper limit in AudioMixer; 71// removing that AudioMixer limit would be necessary but insufficient to support > 2 channels. 72// The macro FCC_2 highlights some (but not all) places where there is are 2-channel assumptions. 73// Search also for "2", "left", "right", "[0]", "[1]", ">> 16", "<< 16", etc. 74#define FCC_2 2 // FCC_2 = Fixed Channel Count 2 75 76static const nsecs_t kDefaultStandbyTimeInNsecs = seconds(3); 77 78#define MAX_GAIN 4096.0f 79#define MAX_GAIN_INT 0x1000 80 81#define INCLUDING_FROM_AUDIOFLINGER_H 82 83class AudioFlinger : 84 public BinderService<AudioFlinger>, 85 public BnAudioFlinger 86{ 87 friend class BinderService<AudioFlinger>; // for AudioFlinger() 88public: 89 static const char* getServiceName() { return "media.audio_flinger"; } 90 91 virtual status_t dump(int fd, const Vector<String16>& args); 92 93 // IAudioFlinger interface, in binder opcode order 94 virtual sp<IAudioTrack> createTrack( 95 pid_t pid, 96 audio_stream_type_t streamType, 97 uint32_t sampleRate, 98 audio_format_t format, 99 audio_channel_mask_t channelMask, 100 size_t frameCount, 101 IAudioFlinger::track_flags_t *flags, 102 const sp<IMemory>& sharedBuffer, 103 audio_io_handle_t output, 104 pid_t tid, 105 int *sessionId, 106 status_t *status); 107 108 virtual sp<IAudioRecord> openRecord( 109 pid_t pid, 110 audio_io_handle_t input, 111 uint32_t sampleRate, 112 audio_format_t format, 113 audio_channel_mask_t channelMask, 114 size_t frameCount, 115 IAudioFlinger::track_flags_t flags, 116 pid_t tid, 117 int *sessionId, 118 status_t *status); 119 120 virtual uint32_t sampleRate(audio_io_handle_t output) const; 121 virtual int channelCount(audio_io_handle_t output) const; 122 virtual audio_format_t format(audio_io_handle_t output) const; 123 virtual size_t frameCount(audio_io_handle_t output) const; 124 virtual uint32_t latency(audio_io_handle_t output) const; 125 126 virtual status_t setMasterVolume(float value); 127 virtual status_t setMasterMute(bool muted); 128 129 virtual float masterVolume() const; 130 virtual bool masterMute() const; 131 132 virtual status_t setStreamVolume(audio_stream_type_t stream, float value, 133 audio_io_handle_t output); 134 virtual status_t setStreamMute(audio_stream_type_t stream, bool muted); 135 136 virtual float streamVolume(audio_stream_type_t stream, 137 audio_io_handle_t output) const; 138 virtual bool streamMute(audio_stream_type_t stream) const; 139 140 virtual status_t setMode(audio_mode_t mode); 141 142 virtual status_t setMicMute(bool state); 143 virtual bool getMicMute() const; 144 145 virtual status_t setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs); 146 virtual String8 getParameters(audio_io_handle_t ioHandle, const String8& keys) const; 147 148 virtual void registerClient(const sp<IAudioFlingerClient>& client); 149 150 virtual size_t getInputBufferSize(uint32_t sampleRate, audio_format_t format, 151 audio_channel_mask_t channelMask) const; 152 153 virtual audio_io_handle_t openOutput(audio_module_handle_t module, 154 audio_devices_t *pDevices, 155 uint32_t *pSamplingRate, 156 audio_format_t *pFormat, 157 audio_channel_mask_t *pChannelMask, 158 uint32_t *pLatencyMs, 159 audio_output_flags_t flags); 160 161 virtual audio_io_handle_t openDuplicateOutput(audio_io_handle_t output1, 162 audio_io_handle_t output2); 163 164 virtual status_t closeOutput(audio_io_handle_t output); 165 166 virtual status_t suspendOutput(audio_io_handle_t output); 167 168 virtual status_t restoreOutput(audio_io_handle_t output); 169 170 virtual audio_io_handle_t openInput(audio_module_handle_t module, 171 audio_devices_t *pDevices, 172 uint32_t *pSamplingRate, 173 audio_format_t *pFormat, 174 audio_channel_mask_t *pChannelMask); 175 176 virtual status_t closeInput(audio_io_handle_t input); 177 178 virtual status_t setStreamOutput(audio_stream_type_t stream, audio_io_handle_t output); 179 180 virtual status_t setVoiceVolume(float volume); 181 182 virtual status_t getRenderPosition(size_t *halFrames, size_t *dspFrames, 183 audio_io_handle_t output) const; 184 185 virtual unsigned int getInputFramesLost(audio_io_handle_t ioHandle) const; 186 187 virtual int newAudioSessionId(); 188 189 virtual void acquireAudioSessionId(int audioSession); 190 191 virtual void releaseAudioSessionId(int audioSession); 192 193 virtual status_t queryNumberEffects(uint32_t *numEffects) const; 194 195 virtual status_t queryEffect(uint32_t index, effect_descriptor_t *descriptor) const; 196 197 virtual status_t getEffectDescriptor(const effect_uuid_t *pUuid, 198 effect_descriptor_t *descriptor) const; 199 200 virtual sp<IEffect> createEffect(pid_t pid, 201 effect_descriptor_t *pDesc, 202 const sp<IEffectClient>& effectClient, 203 int32_t priority, 204 audio_io_handle_t io, 205 int sessionId, 206 status_t *status, 207 int *id, 208 int *enabled); 209 210 virtual status_t moveEffects(int sessionId, audio_io_handle_t srcOutput, 211 audio_io_handle_t dstOutput); 212 213 virtual audio_module_handle_t loadHwModule(const char *name); 214 215 virtual uint32_t getPrimaryOutputSamplingRate(); 216 virtual size_t getPrimaryOutputFrameCount(); 217 218 virtual status_t onTransact( 219 uint32_t code, 220 const Parcel& data, 221 Parcel* reply, 222 uint32_t flags); 223 224 // end of IAudioFlinger interface 225 226 class SyncEvent; 227 228 typedef void (*sync_event_callback_t)(const wp<SyncEvent>& event) ; 229 230 class SyncEvent : public RefBase { 231 public: 232 SyncEvent(AudioSystem::sync_event_t type, 233 int triggerSession, 234 int listenerSession, 235 sync_event_callback_t callBack, 236 void *cookie) 237 : mType(type), mTriggerSession(triggerSession), mListenerSession(listenerSession), 238 mCallback(callBack), mCookie(cookie) 239 {} 240 241 virtual ~SyncEvent() {} 242 243 void trigger() { Mutex::Autolock _l(mLock); if (mCallback) mCallback(this); } 244 bool isCancelled() const { Mutex::Autolock _l(mLock); return (mCallback == NULL); } 245 void cancel() { Mutex::Autolock _l(mLock); mCallback = NULL; } 246 AudioSystem::sync_event_t type() const { return mType; } 247 int triggerSession() const { return mTriggerSession; } 248 int listenerSession() const { return mListenerSession; } 249 void *cookie() const { return mCookie; } 250 251 private: 252 const AudioSystem::sync_event_t mType; 253 const int mTriggerSession; 254 const int mListenerSession; 255 sync_event_callback_t mCallback; 256 void * const mCookie; 257 mutable Mutex mLock; 258 }; 259 260 sp<SyncEvent> createSyncEvent(AudioSystem::sync_event_t type, 261 int triggerSession, 262 int listenerSession, 263 sync_event_callback_t callBack, 264 void *cookie); 265 266private: 267 class AudioHwDevice; // fwd declaration for findSuitableHwDev_l 268 269 audio_mode_t getMode() const { return mMode; } 270 271 bool btNrecIsOff() const { return mBtNrecIsOff; } 272 273 AudioFlinger(); 274 virtual ~AudioFlinger(); 275 276 // call in any IAudioFlinger method that accesses mPrimaryHardwareDev 277 status_t initCheck() const { return mPrimaryHardwareDev == NULL ? 278 NO_INIT : NO_ERROR; } 279 280 // RefBase 281 virtual void onFirstRef(); 282 283 AudioHwDevice* findSuitableHwDev_l(audio_module_handle_t module, 284 audio_devices_t devices); 285 void purgeStaleEffects_l(); 286 287 // standby delay for MIXER and DUPLICATING playback threads is read from property 288 // ro.audio.flinger_standbytime_ms or defaults to kDefaultStandbyTimeInNsecs 289 static nsecs_t mStandbyTimeInNsecs; 290 291 // incremented by 2 when screen state changes, bit 0 == 1 means "off" 292 // AudioFlinger::setParameters() updates, other threads read w/o lock 293 static uint32_t mScreenState; 294 295 // Internal dump utilities. 296 static const int kDumpLockRetries = 50; 297 static const int kDumpLockSleepUs = 20000; 298 static bool dumpTryLock(Mutex& mutex); 299 void dumpPermissionDenial(int fd, const Vector<String16>& args); 300 void dumpClients(int fd, const Vector<String16>& args); 301 void dumpInternals(int fd, const Vector<String16>& args); 302 303 // --- Client --- 304 class Client : public RefBase { 305 public: 306 Client(const sp<AudioFlinger>& audioFlinger, pid_t pid); 307 virtual ~Client(); 308 sp<MemoryDealer> heap() const; 309 pid_t pid() const { return mPid; } 310 sp<AudioFlinger> audioFlinger() const { return mAudioFlinger; } 311 312 bool reserveTimedTrack(); 313 void releaseTimedTrack(); 314 315 private: 316 Client(const Client&); 317 Client& operator = (const Client&); 318 const sp<AudioFlinger> mAudioFlinger; 319 const sp<MemoryDealer> mMemoryDealer; 320 const pid_t mPid; 321 322 Mutex mTimedTrackLock; 323 int mTimedTrackCount; 324 }; 325 326 // --- Notification Client --- 327 class NotificationClient : public IBinder::DeathRecipient { 328 public: 329 NotificationClient(const sp<AudioFlinger>& audioFlinger, 330 const sp<IAudioFlingerClient>& client, 331 pid_t pid); 332 virtual ~NotificationClient(); 333 334 sp<IAudioFlingerClient> audioFlingerClient() const { return mAudioFlingerClient; } 335 336 // IBinder::DeathRecipient 337 virtual void binderDied(const wp<IBinder>& who); 338 339 private: 340 NotificationClient(const NotificationClient&); 341 NotificationClient& operator = (const NotificationClient&); 342 343 const sp<AudioFlinger> mAudioFlinger; 344 const pid_t mPid; 345 const sp<IAudioFlingerClient> mAudioFlingerClient; 346 }; 347 348 class TrackHandle; 349 class RecordHandle; 350 class RecordThread; 351 class PlaybackThread; 352 class MixerThread; 353 class DirectOutputThread; 354 class DuplicatingThread; 355 class Track; 356 class RecordTrack; 357 class EffectModule; 358 class EffectHandle; 359 class EffectChain; 360 struct AudioStreamOut; 361 struct AudioStreamIn; 362 363 struct stream_type_t { 364 stream_type_t() 365 : volume(1.0f), 366 mute(false) 367 { 368 } 369 float volume; 370 bool mute; 371 }; 372 373 // --- PlaybackThread --- 374 375#include "Threads.h" 376 377#include "Effects.h" 378 379 // server side of the client's IAudioTrack 380 class TrackHandle : public android::BnAudioTrack { 381 public: 382 TrackHandle(const sp<PlaybackThread::Track>& track); 383 virtual ~TrackHandle(); 384 virtual sp<IMemory> getCblk() const; 385 virtual status_t start(); 386 virtual void stop(); 387 virtual void flush(); 388 virtual void pause(); 389 virtual status_t attachAuxEffect(int effectId); 390 virtual status_t allocateTimedBuffer(size_t size, 391 sp<IMemory>* buffer); 392 virtual status_t queueTimedBuffer(const sp<IMemory>& buffer, 393 int64_t pts); 394 virtual status_t setMediaTimeTransform(const LinearTransform& xform, 395 int target); 396 virtual status_t onTransact( 397 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags); 398 private: 399 const sp<PlaybackThread::Track> mTrack; 400 }; 401 402 // server side of the client's IAudioRecord 403 class RecordHandle : public android::BnAudioRecord { 404 public: 405 RecordHandle(const sp<RecordThread::RecordTrack>& recordTrack); 406 virtual ~RecordHandle(); 407 virtual sp<IMemory> getCblk() const; 408 virtual status_t start(int /*AudioSystem::sync_event_t*/ event, int triggerSession); 409 virtual void stop(); 410 virtual status_t onTransact( 411 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags); 412 private: 413 const sp<RecordThread::RecordTrack> mRecordTrack; 414 415 // for use from destructor 416 void stop_nonvirtual(); 417 }; 418 419 PlaybackThread *checkPlaybackThread_l(audio_io_handle_t output) const; 420 MixerThread *checkMixerThread_l(audio_io_handle_t output) const; 421 RecordThread *checkRecordThread_l(audio_io_handle_t input) const; 422 // no range check, AudioFlinger::mLock held 423 bool streamMute_l(audio_stream_type_t stream) const 424 { return mStreamTypes[stream].mute; } 425 // no range check, doesn't check per-thread stream volume, AudioFlinger::mLock held 426 float streamVolume_l(audio_stream_type_t stream) const 427 { return mStreamTypes[stream].volume; } 428 void audioConfigChanged_l(int event, audio_io_handle_t ioHandle, const void *param2); 429 430 // allocate an audio_io_handle_t, session ID, or effect ID 431 uint32_t nextUniqueId(); 432 433 status_t moveEffectChain_l(int sessionId, 434 PlaybackThread *srcThread, 435 PlaybackThread *dstThread, 436 bool reRegister); 437 // return thread associated with primary hardware device, or NULL 438 PlaybackThread *primaryPlaybackThread_l() const; 439 audio_devices_t primaryOutputDevice_l() const; 440 441 sp<PlaybackThread> getEffectThread_l(int sessionId, int EffectId); 442 443 444 void removeClient_l(pid_t pid); 445 void removeNotificationClient(pid_t pid); 446 447 class AudioHwDevice { 448 public: 449 enum Flags { 450 AHWD_CAN_SET_MASTER_VOLUME = 0x1, 451 AHWD_CAN_SET_MASTER_MUTE = 0x2, 452 }; 453 454 AudioHwDevice(const char *moduleName, 455 audio_hw_device_t *hwDevice, 456 Flags flags) 457 : mModuleName(strdup(moduleName)) 458 , mHwDevice(hwDevice) 459 , mFlags(flags) { } 460 /*virtual*/ ~AudioHwDevice() { free((void *)mModuleName); } 461 462 bool canSetMasterVolume() const { 463 return (0 != (mFlags & AHWD_CAN_SET_MASTER_VOLUME)); 464 } 465 466 bool canSetMasterMute() const { 467 return (0 != (mFlags & AHWD_CAN_SET_MASTER_MUTE)); 468 } 469 470 const char *moduleName() const { return mModuleName; } 471 audio_hw_device_t *hwDevice() const { return mHwDevice; } 472 private: 473 const char * const mModuleName; 474 audio_hw_device_t * const mHwDevice; 475 Flags mFlags; 476 }; 477 478 // AudioStreamOut and AudioStreamIn are immutable, so their fields are const. 479 // For emphasis, we could also make all pointers to them be "const *", 480 // but that would clutter the code unnecessarily. 481 482 struct AudioStreamOut { 483 AudioHwDevice* const audioHwDev; 484 audio_stream_out_t* const stream; 485 486 audio_hw_device_t* hwDev() const { return audioHwDev->hwDevice(); } 487 488 AudioStreamOut(AudioHwDevice *dev, audio_stream_out_t *out) : 489 audioHwDev(dev), stream(out) {} 490 }; 491 492 struct AudioStreamIn { 493 AudioHwDevice* const audioHwDev; 494 audio_stream_in_t* const stream; 495 496 audio_hw_device_t* hwDev() const { return audioHwDev->hwDevice(); } 497 498 AudioStreamIn(AudioHwDevice *dev, audio_stream_in_t *in) : 499 audioHwDev(dev), stream(in) {} 500 }; 501 502 // for mAudioSessionRefs only 503 struct AudioSessionRef { 504 AudioSessionRef(int sessionid, pid_t pid) : 505 mSessionid(sessionid), mPid(pid), mCnt(1) {} 506 const int mSessionid; 507 const pid_t mPid; 508 int mCnt; 509 }; 510 511 mutable Mutex mLock; 512 513 DefaultKeyedVector< pid_t, wp<Client> > mClients; // see ~Client() 514 515 mutable Mutex mHardwareLock; 516 // NOTE: If both mLock and mHardwareLock mutexes must be held, 517 // always take mLock before mHardwareLock 518 519 // These two fields are immutable after onFirstRef(), so no lock needed to access 520 AudioHwDevice* mPrimaryHardwareDev; // mAudioHwDevs[0] or NULL 521 DefaultKeyedVector<audio_module_handle_t, AudioHwDevice*> mAudioHwDevs; 522 523 // for dump, indicates which hardware operation is currently in progress (but not stream ops) 524 enum hardware_call_state { 525 AUDIO_HW_IDLE = 0, // no operation in progress 526 AUDIO_HW_INIT, // init_check 527 AUDIO_HW_OUTPUT_OPEN, // open_output_stream 528 AUDIO_HW_OUTPUT_CLOSE, // unused 529 AUDIO_HW_INPUT_OPEN, // unused 530 AUDIO_HW_INPUT_CLOSE, // unused 531 AUDIO_HW_STANDBY, // unused 532 AUDIO_HW_SET_MASTER_VOLUME, // set_master_volume 533 AUDIO_HW_GET_ROUTING, // unused 534 AUDIO_HW_SET_ROUTING, // unused 535 AUDIO_HW_GET_MODE, // unused 536 AUDIO_HW_SET_MODE, // set_mode 537 AUDIO_HW_GET_MIC_MUTE, // get_mic_mute 538 AUDIO_HW_SET_MIC_MUTE, // set_mic_mute 539 AUDIO_HW_SET_VOICE_VOLUME, // set_voice_volume 540 AUDIO_HW_SET_PARAMETER, // set_parameters 541 AUDIO_HW_GET_INPUT_BUFFER_SIZE, // get_input_buffer_size 542 AUDIO_HW_GET_MASTER_VOLUME, // get_master_volume 543 AUDIO_HW_GET_PARAMETER, // get_parameters 544 AUDIO_HW_SET_MASTER_MUTE, // set_master_mute 545 AUDIO_HW_GET_MASTER_MUTE, // get_master_mute 546 }; 547 548 mutable hardware_call_state mHardwareStatus; // for dump only 549 550 551 DefaultKeyedVector< audio_io_handle_t, sp<PlaybackThread> > mPlaybackThreads; 552 stream_type_t mStreamTypes[AUDIO_STREAM_CNT]; 553 554 // member variables below are protected by mLock 555 float mMasterVolume; 556 bool mMasterMute; 557 // end of variables protected by mLock 558 559 DefaultKeyedVector< audio_io_handle_t, sp<RecordThread> > mRecordThreads; 560 561 DefaultKeyedVector< pid_t, sp<NotificationClient> > mNotificationClients; 562 volatile int32_t mNextUniqueId; // updated by android_atomic_inc 563 audio_mode_t mMode; 564 bool mBtNrecIsOff; 565 566 // protected by mLock 567 Vector<AudioSessionRef*> mAudioSessionRefs; 568 569 float masterVolume_l() const; 570 bool masterMute_l() const; 571 audio_module_handle_t loadHwModule_l(const char *name); 572 573 Vector < sp<SyncEvent> > mPendingSyncEvents; // sync events awaiting for a session 574 // to be created 575 576private: 577 sp<Client> registerPid_l(pid_t pid); // always returns non-0 578 579 // for use from destructor 580 status_t closeOutput_nonvirtual(audio_io_handle_t output); 581 status_t closeInput_nonvirtual(audio_io_handle_t input); 582 583 // all record threads serially share a common tee sink, which is re-created on format change 584 sp<NBAIO_Sink> mRecordTeeSink; 585 sp<NBAIO_Source> mRecordTeeSource; 586 587public: 588 static void dumpTee(int fd, const sp<NBAIO_Source>& source, audio_io_handle_t id = 0); 589}; 590 591#undef INCLUDING_FROM_AUDIOFLINGER_H 592 593// ---------------------------------------------------------------------------- 594 595}; // namespace android 596 597#endif // ANDROID_AUDIO_FLINGER_H 598