1 /* AudioHardwareALSA.h 2 ** 3 ** Copyright 2008-2010, Wind River Systems 4 ** Copyright (c) 2011-2012, Code Aurora Forum. All rights reserved. 5 ** 6 ** Licensed under the Apache License, Version 2.0 (the "License"); 7 ** you may not use this file except in compliance with the License. 8 ** You may obtain a copy of the License at 9 ** 10 ** http://www.apache.org/licenses/LICENSE-2.0 11 ** 12 ** Unless required by applicable law or agreed to in writing, software 13 ** distributed under the License is distributed on an "AS IS" BASIS, 14 ** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. 15 ** See the License for the specific language governing permissions and 16 ** limitations under the License. 17 */ 18 19 #ifndef ANDROID_AUDIO_HARDWARE_ALSA_H 20 #define ANDROID_AUDIO_HARDWARE_ALSA_H 21 22 #define QCOM_CSDCLIENT_ENABLED 1 23 24 #include <utils/List.h> 25 #include <hardware_legacy/AudioHardwareBase.h> 26 27 #include <hardware_legacy/AudioHardwareInterface.h> 28 #include <hardware_legacy/AudioSystemLegacy.h> 29 #include <system/audio.h> 30 #include <hardware/audio.h> 31 #include <utils/threads.h> 32 #include <dlfcn.h> 33 34 #ifdef QCOM_USBAUDIO_ENABLED 35 #include <AudioUsbALSA.h> 36 #endif 37 38 extern "C" { 39 #include <sound/asound.h> 40 #include "alsa_audio.h" 41 #include "msm8960_use_cases.h" 42 } 43 44 #include <hardware/hardware.h> 45 46 namespace android_audio_legacy 47 { 48 using android::List; 49 using android::Mutex; 50 class AudioHardwareALSA; 51 52 /** 53 * The id of ALSA module 54 */ 55 #define ALSA_HARDWARE_MODULE_ID "alsa" 56 #define ALSA_HARDWARE_NAME "alsa" 57 58 #define DEFAULT_SAMPLING_RATE 48000 59 #define DEFAULT_CHANNEL_MODE 2 60 #define VOICE_SAMPLING_RATE 8000 61 #define VOICE_CHANNEL_MODE 1 62 #define PLAYBACK_LATENCY 170000 63 #define RECORD_LATENCY 96000 64 #define VOICE_LATENCY 85333 65 #define DEFAULT_BUFFER_SIZE 4096 66 //4032 = 336(kernel buffer size) * 2(bytes pcm_16) * 6(number of channels) 67 #define DEFAULT_MULTI_CHANNEL_BUF_SIZE 4032 68 #define DEFAULT_VOICE_BUFFER_SIZE 2048 69 #define PLAYBACK_LOW_LATENCY_BUFFER_SIZE 1024 70 #define PLAYBACK_LOW_LATENCY 22000 71 #define PLAYBACK_LOW_LATENCY_MEASURED 42000 72 #define DEFAULT_IN_BUFFER_SIZE 320 73 #define MIN_CAPTURE_BUFFER_SIZE_PER_CH 320 74 #define MAX_CAPTURE_BUFFER_SIZE_PER_CH 2048 75 #define FM_BUFFER_SIZE 1024 76 77 #define VOIP_SAMPLING_RATE_8K 8000 78 #define VOIP_SAMPLING_RATE_16K 16000 79 #define VOIP_DEFAULT_CHANNEL_MODE 1 80 #define VOIP_BUFFER_SIZE_8K 320 81 #define VOIP_BUFFER_SIZE_16K 640 82 #define VOIP_BUFFER_MAX_SIZE VOIP_BUFFER_SIZE_16K 83 #define VOIP_PLAYBACK_LATENCY 6400 84 #define VOIP_RECORD_LATENCY 6400 85 86 #define MODE_IS127 0x2 87 #define MODE_4GV_NB 0x3 88 #define MODE_4GV_WB 0x4 89 #define MODE_AMR 0x5 90 #define MODE_AMR_WB 0xD 91 #define MODE_PCM 0xC 92 93 #define DUALMIC_KEY "dualmic_enabled" 94 #define FLUENCE_KEY "fluence" 95 #define ANC_KEY "anc_enabled" 96 #define TTY_MODE_KEY "tty_mode" 97 #define BT_SAMPLERATE_KEY "bt_samplerate" 98 #define BTHEADSET_VGS "bt_headset_vgs" 99 #define WIDEVOICE_KEY "wide_voice_enable" 100 #define VOIPRATE_KEY "voip_rate" 101 #define FENS_KEY "fens_enable" 102 #define ST_KEY "st_enable" 103 #define INCALLMUSIC_KEY "incall_music_enabled" 104 105 #define ANC_FLAG 0x00000001 106 #define DMIC_FLAG 0x00000002 107 #define QMIC_FLAG 0x00000004 108 #ifdef QCOM_SSR_ENABLED 109 #define SSRQMIC_FLAG 0x00000008 110 #endif 111 112 #define TTY_OFF 0x00000010 113 #define TTY_FULL 0x00000020 114 #define TTY_VCO 0x00000040 115 #define TTY_HCO 0x00000080 116 #define TTY_CLEAR 0xFFFFFF0F 117 118 #define LPA_SESSION_ID 1 119 #define TUNNEL_SESSION_ID 2 120 #ifdef QCOM_USBAUDIO_ENABLED 121 static int USBPLAYBACKBIT_MUSIC = (1 << 0); 122 static int USBPLAYBACKBIT_VOICECALL = (1 << 1); 123 static int USBPLAYBACKBIT_VOIPCALL = (1 << 2); 124 static int USBPLAYBACKBIT_FM = (1 << 3); 125 static int USBPLAYBACKBIT_LPA = (1 << 4); 126 127 static int USBRECBIT_REC = (1 << 0); 128 static int USBRECBIT_VOICECALL = (1 << 1); 129 static int USBRECBIT_VOIPCALL = (1 << 2); 130 static int USBRECBIT_FM = (1 << 3); 131 #endif 132 133 #define DEVICE_SPEAKER_HEADSET "Speaker Headset" 134 #define DEVICE_HEADSET "Headset" 135 #define DEVICE_HEADPHONES "Headphones" 136 137 #ifdef QCOM_SSR_ENABLED 138 #define COEFF_ARRAY_SIZE 4 139 #define FILT_SIZE ((512+1)* 6) /* # ((FFT bins)/2+1)*numOutputs */ 140 #define SSR_FRAME_SIZE 512 141 #define SSR_INPUT_FRAME_SIZE (SSR_FRAME_SIZE * 4) 142 #define SSR_OUTPUT_FRAME_SIZE (SSR_FRAME_SIZE * 6) 143 #endif 144 145 #define MODE_CALL_KEY "CALL_KEY" 146 147 struct alsa_device_t; 148 static uint32_t FLUENCE_MODE_ENDFIRE = 0; 149 static uint32_t FLUENCE_MODE_BROADSIDE = 1; 150 151 enum { 152 INCALL_REC_MONO, 153 INCALL_REC_STEREO, 154 }; 155 156 enum audio_call_mode { 157 CS_INACTIVE = 0x0, 158 CS_ACTIVE = 0x1, 159 CS_HOLD = 0x2, 160 IMS_INACTIVE = 0x0, 161 IMS_ACTIVE = 0x10, 162 IMS_HOLD = 0x20 163 }; 164 165 166 struct alsa_handle_t { 167 alsa_device_t * module; 168 uint32_t devices; 169 char useCase[MAX_STR_LEN]; 170 struct pcm * handle; 171 snd_pcm_format_t format; 172 uint32_t channels; 173 audio_channel_mask_t channelMask; 174 uint32_t sampleRate; 175 unsigned int latency; // Delay in usec 176 unsigned int bufferSize; // Size of sample buffer 177 unsigned int periodSize; 178 bool isDeepbufferOutput; 179 struct pcm * rxHandle; 180 snd_use_case_mgr_t *ucMgr; 181 }; 182 183 typedef List < alsa_handle_t > ALSAHandleList; 184 185 struct use_case_t { 186 char useCase[MAX_STR_LEN]; 187 }; 188 189 typedef List < use_case_t > ALSAUseCaseList; 190 191 struct alsa_device_t { 192 hw_device_t common; 193 194 status_t (*init)(alsa_device_t *, ALSAHandleList &); 195 status_t (*open)(alsa_handle_t *); 196 status_t (*close)(alsa_handle_t *); 197 status_t (*standby)(alsa_handle_t *); 198 status_t (*route)(alsa_handle_t *, uint32_t, int); 199 status_t (*startVoiceCall)(alsa_handle_t *); 200 status_t (*startVoipCall)(alsa_handle_t *); 201 status_t (*startFm)(alsa_handle_t *); 202 void (*setVoiceVolume)(int); 203 void (*setVoipVolume)(int); 204 void (*setMicMute)(int); 205 void (*setVoipMicMute)(int); 206 void (*setVoipConfig)(int, int); 207 status_t (*setFmVolume)(int); 208 void (*setBtscoRate)(int); 209 status_t (*setLpaVolume)(int); 210 void (*enableWideVoice)(bool); 211 void (*enableFENS)(bool); 212 void (*setFlags)(uint32_t); 213 status_t (*setCompressedVolume)(int); 214 void (*enableSlowTalk)(bool); 215 void (*setVocRecMode)(uint8_t); 216 void (*setVoLTEMicMute)(int); 217 void (*setVoLTEVolume)(int); 218 #ifdef SEPERATED_AUDIO_INPUT 219 void (*setInput)(int); 220 #endif 221 #ifdef QCOM_CSDCLIENT_ENABLED 222 void (*setCsdHandle)(void*); 223 #endif 224 }; 225 226 // ---------------------------------------------------------------------------- 227 228 class ALSAMixer 229 { 230 public: 231 ALSAMixer(); 232 virtual ~ALSAMixer(); 233 234 bool isValid() { return 1;} 235 status_t setMasterVolume(float volume); 236 status_t setMasterGain(float gain); 237 238 status_t setVolume(uint32_t device, float left, float right); 239 status_t setGain(uint32_t device, float gain); 240 241 status_t setCaptureMuteState(uint32_t device, bool state); 242 status_t getCaptureMuteState(uint32_t device, bool *state); 243 status_t setPlaybackMuteState(uint32_t device, bool state); 244 status_t getPlaybackMuteState(uint32_t device, bool *state); 245 246 }; 247 248 class ALSAControl 249 { 250 public: 251 ALSAControl(const char *device = "/dev/snd/controlC0"); 252 virtual ~ALSAControl(); 253 254 status_t get(const char *name, unsigned int &value, int index = 0); 255 status_t set(const char *name, unsigned int value, int index = -1); 256 status_t set(const char *name, const char *); 257 status_t setext(const char *name, int count, char **setValues); 258 259 private: 260 struct mixer* mHandle; 261 }; 262 263 class ALSAStreamOps 264 { 265 public: 266 ALSAStreamOps(AudioHardwareALSA *parent, alsa_handle_t *handle); 267 virtual ~ALSAStreamOps(); 268 269 status_t set(int *format, uint32_t *channels, uint32_t *rate, uint32_t device); 270 271 status_t setParameters(const String8& keyValuePairs); 272 String8 getParameters(const String8& keys); 273 274 uint32_t sampleRate() const; 275 size_t bufferSize() const; 276 int format() const; 277 uint32_t channels() const; 278 279 status_t open(int mode); 280 void close(); 281 282 protected: 283 friend class AudioHardwareALSA; 284 285 AudioHardwareALSA * mParent; 286 alsa_handle_t * mHandle; 287 uint32_t mDevices; 288 }; 289 290 // ---------------------------------------------------------------------------- 291 292 class AudioStreamOutALSA : public AudioStreamOut, public ALSAStreamOps 293 { 294 public: 295 AudioStreamOutALSA(AudioHardwareALSA *parent, alsa_handle_t *handle); 296 virtual ~AudioStreamOutALSA(); 297 298 virtual uint32_t sampleRate() const 299 { 300 return ALSAStreamOps::sampleRate(); 301 } 302 303 virtual size_t bufferSize() const 304 { 305 return ALSAStreamOps::bufferSize(); 306 } 307 308 virtual uint32_t channels() const; 309 310 virtual int format() const 311 { 312 return ALSAStreamOps::format(); 313 } 314 315 virtual uint32_t latency() const; 316 317 virtual ssize_t write(const void *buffer, size_t bytes); 318 virtual status_t dump(int fd, const Vector<String16>& args); 319 320 status_t setVolume(float left, float right); 321 322 virtual status_t standby(); 323 324 virtual status_t setParameters(const String8& keyValuePairs) { 325 return ALSAStreamOps::setParameters(keyValuePairs); 326 } 327 328 virtual String8 getParameters(const String8& keys) { 329 return ALSAStreamOps::getParameters(keys); 330 } 331 332 // return the number of audio frames written by the audio dsp to DAC since 333 // the output has exited standby 334 virtual status_t getRenderPosition(uint32_t *dspFrames); 335 336 status_t open(int mode); 337 status_t close(); 338 339 private: 340 uint32_t mFrameCount; 341 342 protected: 343 AudioHardwareALSA * mParent; 344 }; 345 346 class AudioStreamInALSA : public AudioStreamIn, public ALSAStreamOps 347 { 348 public: 349 AudioStreamInALSA(AudioHardwareALSA *parent, 350 alsa_handle_t *handle, 351 AudioSystem::audio_in_acoustics audio_acoustics); 352 virtual ~AudioStreamInALSA(); 353 354 virtual uint32_t sampleRate() const 355 { 356 return ALSAStreamOps::sampleRate(); 357 } 358 359 virtual size_t bufferSize() const 360 { 361 return ALSAStreamOps::bufferSize(); 362 } 363 364 virtual uint32_t channels() const 365 { 366 return ALSAStreamOps::channels(); 367 } 368 369 virtual int format() const 370 { 371 return ALSAStreamOps::format(); 372 } 373 374 virtual ssize_t read(void* buffer, ssize_t bytes); 375 virtual status_t dump(int fd, const Vector<String16>& args); 376 377 virtual status_t setGain(float gain); 378 379 virtual status_t standby(); 380 381 virtual status_t setParameters(const String8& keyValuePairs) 382 { 383 return ALSAStreamOps::setParameters(keyValuePairs); 384 } 385 386 virtual String8 getParameters(const String8& keys) 387 { 388 return ALSAStreamOps::getParameters(keys); 389 } 390 391 // Return the amount of input frames lost in the audio driver since the last call of this function. 392 // Audio driver is expected to reset the value to 0 and restart counting upon returning the current value by this function call. 393 // Such loss typically occurs when the user space process is blocked longer than the capacity of audio driver buffers. 394 // Unit: the number of input audio frames 395 virtual unsigned int getInputFramesLost() const; 396 397 virtual status_t addAudioEffect(effect_handle_t effect) 398 { 399 return BAD_VALUE; 400 } 401 402 virtual status_t removeAudioEffect(effect_handle_t effect) 403 { 404 return BAD_VALUE; 405 } 406 status_t setAcousticParams(void* params); 407 408 status_t open(int mode); 409 status_t close(); 410 #ifdef QCOM_SSR_ENABLED 411 // Helper function to initialize the Surround Sound library. 412 status_t initSurroundSoundLibrary(unsigned long buffersize); 413 #endif 414 415 private: 416 void resetFramesLost(); 417 418 #ifdef QCOM_CSDCLIENT_ENABLED 419 int start_csd_record(int); 420 int stop_csd_record(void); 421 #endif 422 423 unsigned int mFramesLost; 424 AudioSystem::audio_in_acoustics mAcoustics; 425 426 #ifdef QCOM_SSR_ENABLED 427 // Function to read coefficients from files. 428 status_t readCoeffsFromFile(); 429 430 FILE *mFp_4ch; 431 FILE *mFp_6ch; 432 int16_t **mRealCoeffs; 433 int16_t **mImagCoeffs; 434 void *mSurroundObj; 435 436 int16_t *mSurroundInputBuffer; 437 int16_t *mSurroundOutputBuffer; 438 int mSurroundInputBufferIdx; 439 int mSurroundOutputBufferIdx; 440 #endif 441 442 protected: 443 AudioHardwareALSA * mParent; 444 }; 445 446 class AudioHardwareALSA : public AudioHardwareBase 447 { 448 public: 449 AudioHardwareALSA(); 450 virtual ~AudioHardwareALSA(); 451 452 /** 453 * check to see if the audio hardware interface has been initialized. 454 * return status based on values defined in include/utils/Errors.h 455 */ 456 virtual status_t initCheck(); 457 458 /** set the audio volume of a voice call. Range is between 0.0 and 1.0 */ 459 virtual status_t setVoiceVolume(float volume); 460 461 /** 462 * set the audio volume for all audio activities other than voice call. 463 * Range between 0.0 and 1.0. If any value other than NO_ERROR is returned, 464 * the software mixer will emulate this capability. 465 */ 466 virtual status_t setMasterVolume(float volume); 467 #ifdef QCOM_FM_ENABLED 468 virtual status_t setFmVolume(float volume); 469 #endif 470 /** 471 * setMode is called when the audio mode changes. NORMAL mode is for 472 * standard audio playback, RINGTONE when a ringtone is playing, and IN_CALL 473 * when a call is in progress. 474 */ 475 virtual status_t setMode(int mode); 476 477 // mic mute 478 virtual status_t setMicMute(bool state); 479 virtual status_t getMicMute(bool* state); 480 481 // set/get global audio parameters 482 virtual status_t setParameters(const String8& keyValuePairs); 483 virtual String8 getParameters(const String8& keys); 484 485 // Returns audio input buffer size according to parameters passed or 0 if one of the 486 // parameters is not supported 487 virtual size_t getInputBufferSize(uint32_t sampleRate, int format, int channels); 488 489 #ifdef QCOM_TUNNEL_LPA_ENABLED 490 /** This method creates and opens the audio hardware output 491 * session for LPA */ 492 virtual AudioStreamOut* openOutputSession( 493 uint32_t devices, 494 int *format, 495 status_t *status, 496 int sessionId, 497 uint32_t samplingRate=0, 498 uint32_t channels=0); 499 virtual void closeOutputSession(AudioStreamOut* out); 500 #endif 501 502 /** This method creates and opens the audio hardware output stream */ 503 virtual AudioStreamOut* openOutputStream( 504 uint32_t devices, 505 int *format=0, 506 uint32_t *channels=0, 507 uint32_t *sampleRate=0, 508 status_t *status=0); 509 virtual void closeOutputStream(AudioStreamOut* out); 510 511 /** This method creates and opens the audio hardware input stream */ 512 virtual AudioStreamIn* openInputStream( 513 uint32_t devices, 514 int *format, 515 uint32_t *channels, 516 uint32_t *sampleRate, 517 status_t *status, 518 AudioSystem::audio_in_acoustics acoustics); 519 virtual void closeInputStream(AudioStreamIn* in); 520 521 /**This method dumps the state of the audio hardware */ 522 //virtual status_t dumpState(int fd, const Vector<String16>& args); 523 524 static AudioHardwareInterface* create(); 525 526 int mode() 527 { 528 return mMode; 529 } 530 531 protected: 532 virtual status_t dump(int fd, const Vector<String16>& args); 533 virtual uint32_t getVoipMode(int format); 534 void doRouting(int device); 535 #ifdef QCOM_FM_ENABLED 536 void handleFm(int device); 537 #endif 538 #ifdef QCOM_USBAUDIO_ENABLED 539 void closeUSBPlayback(); 540 void closeUSBRecording(); 541 void closeUsbRecordingIfNothingActive(); 542 void closeUsbPlaybackIfNothingActive(); 543 void startUsbPlaybackIfNotStarted(); 544 void startUsbRecordingIfNotStarted(); 545 #endif 546 547 void disableVoiceCall(char* verb, char* modifier, int mode, int device); 548 void enableVoiceCall(char* verb, char* modifier, int mode, int device); 549 bool routeVoiceCall(int device, int newMode); 550 bool routeVoLTECall(int device, int newMode); 551 friend class AudioStreamOutALSA; 552 friend class AudioStreamInALSA; 553 friend class ALSAStreamOps; 554 555 alsa_device_t * mALSADevice; 556 557 ALSAHandleList mDeviceList; 558 559 #ifdef QCOM_USBAUDIO_ENABLED 560 AudioUsbALSA *mAudioUsbALSA; 561 #endif 562 563 Mutex mLock; 564 565 snd_use_case_mgr_t *mUcMgr; 566 567 uint32_t mCurDevice; 568 /* The flag holds all the audio related device settings from 569 * Settings and Qualcomm Settings applications */ 570 uint32_t mDevSettingsFlag; 571 uint32_t mVoipStreamCount; 572 uint32_t mVoipBitRate; 573 uint32_t mIncallMode; 574 575 bool mMicMute; 576 int mCSCallActive; 577 int mVolteCallActive; 578 int mCallState; 579 int mIsFmActive; 580 bool mBluetoothVGS; 581 bool mFusion3Platform; 582 #ifdef QCOM_USBAUDIO_ENABLED 583 int musbPlaybackState; 584 int musbRecordingState; 585 #endif 586 void *mAcdbHandle; 587 void *mCsdHandle; 588 }; 589 590 // ---------------------------------------------------------------------------- 591 592 }; // namespace android_audio_legacy 593 #endif // ANDROID_AUDIO_HARDWARE_ALSA_H 594
