1 /* 2 * Copyright (C) 2008 The Android Open Source Project 3 * 4 * Licensed under the Apache License, Version 2.0 (the "License"); 5 * you may not use this file except in compliance with the License. 6 * You may obtain a copy of the License at 7 * 8 * http://www.apache.org/licenses/LICENSE-2.0 9 * 10 * Unless required by applicable law or agreed to in writing, software 11 * distributed under the License is distributed on an "AS IS" BASIS, 12 * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. 13 * See the License for the specific language governing permissions and 14 * limitations under the License. 15 */ 16 17 #ifndef ANDROID_AUDIOSYSTEM_H_ 18 #define ANDROID_AUDIOSYSTEM_H_ 19 20 #include <utils/RefBase.h> 21 #include <utils/threads.h> 22 #include <media/IAudioFlinger.h> 23 24 namespace android { 25 26 typedef void (*audio_error_callback)(status_t err); 27 typedef int audio_io_handle_t; 28 29 class IAudioPolicyService; 30 class String8; 31 32 class AudioSystem 33 { 34 public: 35 36 enum stream_type { 37 DEFAULT =-1, 38 VOICE_CALL = 0, 39 SYSTEM = 1, 40 RING = 2, 41 MUSIC = 3, 42 ALARM = 4, 43 NOTIFICATION = 5, 44 BLUETOOTH_SCO = 6, 45 ENFORCED_AUDIBLE = 7, // Sounds that cannot be muted by user and must be routed to speaker 46 DTMF = 8, 47 TTS = 9, 48 NUM_STREAM_TYPES 49 }; 50 51 // Audio sub formats (see AudioSystem::audio_format). 52 enum pcm_sub_format { 53 PCM_SUB_16_BIT = 0x1, // must be 1 for backward compatibility 54 PCM_SUB_8_BIT = 0x2, // must be 2 for backward compatibility 55 }; 56 57 // MP3 sub format field definition : can use 11 LSBs in the same way as MP3 frame header to specify 58 // bit rate, stereo mode, version... 59 enum mp3_sub_format { 60 //TODO 61 }; 62 63 // AMR NB/WB sub format field definition: specify frame block interleaving, bandwidth efficient or octet aligned, 64 // encoding mode for recording... 65 enum amr_sub_format { 66 //TODO 67 }; 68 69 // AAC sub format field definition: specify profile or bitrate for recording... 70 enum aac_sub_format { 71 //TODO 72 }; 73 74 // VORBIS sub format field definition: specify quality for recording... 75 enum vorbis_sub_format { 76 //TODO 77 }; 78 79 // Audio format consists in a main format field (upper 8 bits) and a sub format field (lower 24 bits). 80 // The main format indicates the main codec type. The sub format field indicates options and parameters 81 // for each format. The sub format is mainly used for record to indicate for instance the requested bitrate 82 // or profile. It can also be used for certain formats to give informations not present in the encoded 83 // audio stream (e.g. octet alignement for AMR). 84 enum audio_format { 85 INVALID_FORMAT = -1, 86 FORMAT_DEFAULT = 0, 87 PCM = 0x00000000, // must be 0 for backward compatibility 88 MP3 = 0x01000000, 89 AMR_NB = 0x02000000, 90 AMR_WB = 0x03000000, 91 AAC = 0x04000000, 92 HE_AAC_V1 = 0x05000000, 93 HE_AAC_V2 = 0x06000000, 94 VORBIS = 0x07000000, 95 MAIN_FORMAT_MASK = 0xFF000000, 96 SUB_FORMAT_MASK = 0x00FFFFFF, 97 // Aliases 98 PCM_16_BIT = (PCM|PCM_SUB_16_BIT), 99 PCM_8_BIT = (PCM|PCM_SUB_8_BIT) 100 }; 101 102 103 // Channel mask definitions must be kept in sync with JAVA values in /media/java/android/media/AudioFormat.java 104 enum audio_channels { 105 // output channels 106 CHANNEL_OUT_FRONT_LEFT = 0x4, 107 CHANNEL_OUT_FRONT_RIGHT = 0x8, 108 CHANNEL_OUT_FRONT_CENTER = 0x10, 109 CHANNEL_OUT_LOW_FREQUENCY = 0x20, 110 CHANNEL_OUT_BACK_LEFT = 0x40, 111 CHANNEL_OUT_BACK_RIGHT = 0x80, 112 CHANNEL_OUT_FRONT_LEFT_OF_CENTER = 0x100, 113 CHANNEL_OUT_FRONT_RIGHT_OF_CENTER = 0x200, 114 CHANNEL_OUT_BACK_CENTER = 0x400, 115 CHANNEL_OUT_MONO = CHANNEL_OUT_FRONT_LEFT, 116 CHANNEL_OUT_STEREO = (CHANNEL_OUT_FRONT_LEFT | CHANNEL_OUT_FRONT_RIGHT), 117 CHANNEL_OUT_QUAD = (CHANNEL_OUT_FRONT_LEFT | CHANNEL_OUT_FRONT_RIGHT | 118 CHANNEL_OUT_BACK_LEFT | CHANNEL_OUT_BACK_RIGHT), 119 CHANNEL_OUT_SURROUND = (CHANNEL_OUT_FRONT_LEFT | CHANNEL_OUT_FRONT_RIGHT | 120 CHANNEL_OUT_FRONT_CENTER | CHANNEL_OUT_BACK_CENTER), 121 CHANNEL_OUT_5POINT1 = (CHANNEL_OUT_FRONT_LEFT | CHANNEL_OUT_FRONT_RIGHT | 122 CHANNEL_OUT_FRONT_CENTER | CHANNEL_OUT_LOW_FREQUENCY | CHANNEL_OUT_BACK_LEFT | CHANNEL_OUT_BACK_RIGHT), 123 CHANNEL_OUT_7POINT1 = (CHANNEL_OUT_FRONT_LEFT | CHANNEL_OUT_FRONT_RIGHT | 124 CHANNEL_OUT_FRONT_CENTER | CHANNEL_OUT_LOW_FREQUENCY | CHANNEL_OUT_BACK_LEFT | CHANNEL_OUT_BACK_RIGHT | 125 CHANNEL_OUT_FRONT_LEFT_OF_CENTER | CHANNEL_OUT_FRONT_RIGHT_OF_CENTER), 126 CHANNEL_OUT_ALL = (CHANNEL_OUT_FRONT_LEFT | CHANNEL_OUT_FRONT_RIGHT | 127 CHANNEL_OUT_FRONT_CENTER | CHANNEL_OUT_LOW_FREQUENCY | CHANNEL_OUT_BACK_LEFT | CHANNEL_OUT_BACK_RIGHT | 128 CHANNEL_OUT_FRONT_LEFT_OF_CENTER | CHANNEL_OUT_FRONT_RIGHT_OF_CENTER | CHANNEL_OUT_BACK_CENTER), 129 130 // input channels 131 CHANNEL_IN_LEFT = 0x4, 132 CHANNEL_IN_RIGHT = 0x8, 133 CHANNEL_IN_FRONT = 0x10, 134 CHANNEL_IN_BACK = 0x20, 135 CHANNEL_IN_LEFT_PROCESSED = 0x40, 136 CHANNEL_IN_RIGHT_PROCESSED = 0x80, 137 CHANNEL_IN_FRONT_PROCESSED = 0x100, 138 CHANNEL_IN_BACK_PROCESSED = 0x200, 139 CHANNEL_IN_PRESSURE = 0x400, 140 CHANNEL_IN_X_AXIS = 0x800, 141 CHANNEL_IN_Y_AXIS = 0x1000, 142 CHANNEL_IN_Z_AXIS = 0x2000, 143 CHANNEL_IN_VOICE_UPLINK = 0x4000, 144 CHANNEL_IN_VOICE_DNLINK = 0x8000, 145 CHANNEL_IN_MONO = CHANNEL_IN_FRONT, 146 CHANNEL_IN_STEREO = (CHANNEL_IN_LEFT | CHANNEL_IN_RIGHT), 147 CHANNEL_IN_ALL = (CHANNEL_IN_LEFT | CHANNEL_IN_RIGHT | CHANNEL_IN_FRONT | CHANNEL_IN_BACK| 148 CHANNEL_IN_LEFT_PROCESSED | CHANNEL_IN_RIGHT_PROCESSED | CHANNEL_IN_FRONT_PROCESSED | CHANNEL_IN_BACK_PROCESSED| 149 CHANNEL_IN_PRESSURE | CHANNEL_IN_X_AXIS | CHANNEL_IN_Y_AXIS | CHANNEL_IN_Z_AXIS | 150 CHANNEL_IN_VOICE_UPLINK | CHANNEL_IN_VOICE_DNLINK) 151 }; 152 153 enum audio_mode { 154 MODE_INVALID = -2, 155 MODE_CURRENT = -1, 156 MODE_NORMAL = 0, 157 MODE_RINGTONE, 158 MODE_IN_CALL, 159 NUM_MODES // not a valid entry, denotes end-of-list 160 }; 161 162 enum audio_in_acoustics { 163 AGC_ENABLE = 0x0001, 164 AGC_DISABLE = 0, 165 NS_ENABLE = 0x0002, 166 NS_DISABLE = 0, 167 TX_IIR_ENABLE = 0x0004, 168 TX_DISABLE = 0 169 }; 170 171 // special audio session values 172 enum audio_sessions { 173 SESSION_OUTPUT_STAGE = -1, // session for effects attached to a particular output stream 174 // (value must be less than 0) 175 SESSION_OUTPUT_MIX = 0, // session for effects applied to output mix. These effects can 176 // be moved by audio policy manager to another output stream 177 // (value must be 0) 178 }; 179 180 /* These are static methods to control the system-wide AudioFlinger 181 * only privileged processes can have access to them 182 */ 183 184 // mute/unmute microphone 185 static status_t muteMicrophone(bool state); 186 static status_t isMicrophoneMuted(bool *state); 187 188 // set/get master volume 189 static status_t setMasterVolume(float value); 190 static status_t getMasterVolume(float* volume); 191 // mute/unmute audio outputs 192 static status_t setMasterMute(bool mute); 193 static status_t getMasterMute(bool* mute); 194 195 // set/get stream volume on specified output 196 static status_t setStreamVolume(int stream, float value, int output); 197 static status_t getStreamVolume(int stream, float* volume, int output); 198 199 // mute/unmute stream 200 static status_t setStreamMute(int stream, bool mute); 201 static status_t getStreamMute(int stream, bool* mute); 202 203 // set audio mode in audio hardware (see AudioSystem::audio_mode) 204 static status_t setMode(int mode); 205 206 // returns true in *state if tracks are active on the specified stream 207 static status_t isStreamActive(int stream, bool *state); 208 209 // set/get audio hardware parameters. The function accepts a list of parameters 210 // key value pairs in the form: key1=value1;key2=value2;... 211 // Some keys are reserved for standard parameters (See AudioParameter class). 212 static status_t setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs); 213 static String8 getParameters(audio_io_handle_t ioHandle, const String8& keys); 214 215 static void setErrorCallback(audio_error_callback cb); 216 217 // helper function to obtain AudioFlinger service handle 218 static const sp<IAudioFlinger>& get_audio_flinger(); 219 220 static float linearToLog(int volume); 221 static int logToLinear(float volume); 222 223 static status_t getOutputSamplingRate(int* samplingRate, int stream = DEFAULT); 224 static status_t getOutputFrameCount(int* frameCount, int stream = DEFAULT); 225 static status_t getOutputLatency(uint32_t* latency, int stream = DEFAULT); 226 227 static bool routedToA2dpOutput(int streamType); 228 229 static status_t getInputBufferSize(uint32_t sampleRate, int format, int channelCount, 230 size_t* buffSize); 231 232 static status_t setVoiceVolume(float volume); 233 234 // return the number of audio frames written by AudioFlinger to audio HAL and 235 // audio dsp to DAC since the output on which the specificed stream is playing 236 // has exited standby. 237 // returned status (from utils/Errors.h) can be: 238 // - NO_ERROR: successful operation, halFrames and dspFrames point to valid data 239 // - INVALID_OPERATION: Not supported on current hardware platform 240 // - BAD_VALUE: invalid parameter 241 // NOTE: this feature is not supported on all hardware platforms and it is 242 // necessary to check returned status before using the returned values. 243 static status_t getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, int stream = DEFAULT); 244 245 static unsigned int getInputFramesLost(audio_io_handle_t ioHandle); 246 247 static int newAudioSessionId(); 248 // 249 // AudioPolicyService interface 250 // 251 252 enum audio_devices { 253 // output devices 254 DEVICE_OUT_EARPIECE = 0x1, 255 DEVICE_OUT_SPEAKER = 0x2, 256 DEVICE_OUT_WIRED_HEADSET = 0x4, 257 DEVICE_OUT_WIRED_HEADPHONE = 0x8, 258 DEVICE_OUT_BLUETOOTH_SCO = 0x10, 259 DEVICE_OUT_BLUETOOTH_SCO_HEADSET = 0x20, 260 DEVICE_OUT_BLUETOOTH_SCO_CARKIT = 0x40, 261 DEVICE_OUT_BLUETOOTH_A2DP = 0x80, 262 DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES = 0x100, 263 DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER = 0x200, 264 DEVICE_OUT_AUX_DIGITAL = 0x400, 265 DEVICE_OUT_DEFAULT = 0x8000, 266 DEVICE_OUT_ALL = (DEVICE_OUT_EARPIECE | DEVICE_OUT_SPEAKER | DEVICE_OUT_WIRED_HEADSET | 267 DEVICE_OUT_WIRED_HEADPHONE | DEVICE_OUT_BLUETOOTH_SCO | DEVICE_OUT_BLUETOOTH_SCO_HEADSET | 268 DEVICE_OUT_BLUETOOTH_SCO_CARKIT | DEVICE_OUT_BLUETOOTH_A2DP | DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES | 269 DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER | DEVICE_OUT_AUX_DIGITAL | DEVICE_OUT_DEFAULT), 270 DEVICE_OUT_ALL_A2DP = (DEVICE_OUT_BLUETOOTH_A2DP | DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES | 271 DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER), 272 273 // input devices 274 DEVICE_IN_COMMUNICATION = 0x10000, 275 DEVICE_IN_AMBIENT = 0x20000, 276 DEVICE_IN_BUILTIN_MIC = 0x40000, 277 DEVICE_IN_BLUETOOTH_SCO_HEADSET = 0x80000, 278 DEVICE_IN_WIRED_HEADSET = 0x100000, 279 DEVICE_IN_AUX_DIGITAL = 0x200000, 280 DEVICE_IN_VOICE_CALL = 0x400000, 281 DEVICE_IN_BACK_MIC = 0x800000, 282 DEVICE_IN_DEFAULT = 0x80000000, 283 284 DEVICE_IN_ALL = (DEVICE_IN_COMMUNICATION | DEVICE_IN_AMBIENT | DEVICE_IN_BUILTIN_MIC | 285 DEVICE_IN_BLUETOOTH_SCO_HEADSET | DEVICE_IN_WIRED_HEADSET | DEVICE_IN_AUX_DIGITAL | 286 DEVICE_IN_VOICE_CALL | DEVICE_IN_BACK_MIC | DEVICE_IN_DEFAULT) 287 }; 288 289 // device connection states used for setDeviceConnectionState() 290 enum device_connection_state { 291 DEVICE_STATE_UNAVAILABLE, 292 DEVICE_STATE_AVAILABLE, 293 NUM_DEVICE_STATES 294 }; 295 296 // request to open a direct output with getOutput() (by opposition to sharing an output with other AudioTracks) 297 enum output_flags { 298 OUTPUT_FLAG_INDIRECT = 0x0, 299 OUTPUT_FLAG_DIRECT = 0x1 300 }; 301 302 // device categories used for setForceUse() 303 enum forced_config { 304 FORCE_NONE, 305 FORCE_SPEAKER, 306 FORCE_HEADPHONES, 307 FORCE_BT_SCO, 308 FORCE_BT_A2DP, 309 FORCE_WIRED_ACCESSORY, 310 FORCE_BT_CAR_DOCK, 311 FORCE_BT_DESK_DOCK, 312 NUM_FORCE_CONFIG, 313 FORCE_DEFAULT = FORCE_NONE 314 }; 315 316 // usages used for setForceUse() 317 enum force_use { 318 FOR_COMMUNICATION, 319 FOR_MEDIA, 320 FOR_RECORD, 321 FOR_DOCK, 322 NUM_FORCE_USE 323 }; 324 325 // types of io configuration change events received with ioConfigChanged() 326 enum io_config_event { 327 OUTPUT_OPENED, 328 OUTPUT_CLOSED, 329 OUTPUT_CONFIG_CHANGED, 330 INPUT_OPENED, 331 INPUT_CLOSED, 332 INPUT_CONFIG_CHANGED, 333 STREAM_CONFIG_CHANGED, 334 NUM_CONFIG_EVENTS 335 }; 336 337 // audio output descritor used to cache output configurations in client process to avoid frequent calls 338 // through IAudioFlinger 339 class OutputDescriptor { 340 public: 341 OutputDescriptor() 342 : samplingRate(0), format(0), channels(0), frameCount(0), latency(0) {} 343 344 uint32_t samplingRate; 345 int32_t format; 346 int32_t channels; 347 size_t frameCount; 348 uint32_t latency; 349 }; 350 351 // 352 // IAudioPolicyService interface (see AudioPolicyInterface for method descriptions) 353 // 354 static status_t setDeviceConnectionState(audio_devices device, device_connection_state state, const char *device_address); 355 static device_connection_state getDeviceConnectionState(audio_devices device, const char *device_address); 356 static status_t setPhoneState(int state); 357 static status_t setRingerMode(uint32_t mode, uint32_t mask); 358 static status_t setForceUse(force_use usage, forced_config config); 359 static forced_config getForceUse(force_use usage); 360 static audio_io_handle_t getOutput(stream_type stream, 361 uint32_t samplingRate = 0, 362 uint32_t format = FORMAT_DEFAULT, 363 uint32_t channels = CHANNEL_OUT_STEREO, 364 output_flags flags = OUTPUT_FLAG_INDIRECT); 365 static status_t startOutput(audio_io_handle_t output, 366 AudioSystem::stream_type stream, 367 int session = 0); 368 static status_t stopOutput(audio_io_handle_t output, 369 AudioSystem::stream_type stream, 370 int session = 0); 371 static void releaseOutput(audio_io_handle_t output); 372 static audio_io_handle_t getInput(int inputSource, 373 uint32_t samplingRate = 0, 374 uint32_t format = FORMAT_DEFAULT, 375 uint32_t channels = CHANNEL_IN_MONO, 376 audio_in_acoustics acoustics = (audio_in_acoustics)0); 377 static status_t startInput(audio_io_handle_t input); 378 static status_t stopInput(audio_io_handle_t input); 379 static void releaseInput(audio_io_handle_t input); 380 static status_t initStreamVolume(stream_type stream, 381 int indexMin, 382 int indexMax); 383 static status_t setStreamVolumeIndex(stream_type stream, int index); 384 static status_t getStreamVolumeIndex(stream_type stream, int *index); 385 386 static uint32_t getStrategyForStream(stream_type stream); 387 388 static audio_io_handle_t getOutputForEffect(effect_descriptor_t *desc); 389 static status_t registerEffect(effect_descriptor_t *desc, 390 audio_io_handle_t output, 391 uint32_t strategy, 392 int session, 393 int id); 394 static status_t unregisterEffect(int id); 395 396 static const sp<IAudioPolicyService>& get_audio_policy_service(); 397 398 // ---------------------------------------------------------------------------- 399 400 static uint32_t popCount(uint32_t u); 401 static bool isOutputDevice(audio_devices device); 402 static bool isInputDevice(audio_devices device); 403 static bool isA2dpDevice(audio_devices device); 404 static bool isBluetoothScoDevice(audio_devices device); 405 static bool isLowVisibility(stream_type stream); 406 static bool isOutputChannel(uint32_t channel); 407 static bool isInputChannel(uint32_t channel); 408 static bool isValidFormat(uint32_t format); 409 static bool isLinearPCM(uint32_t format); 410 411 private: 412 413 class AudioFlingerClient: public IBinder::DeathRecipient, public BnAudioFlingerClient 414 { 415 public: 416 AudioFlingerClient() { 417 } 418 419 // DeathRecipient 420 virtual void binderDied(const wp<IBinder>& who); 421 422 // IAudioFlingerClient 423 424 // indicate a change in the configuration of an output or input: keeps the cached 425 // values for output/input parameters upto date in client process 426 virtual void ioConfigChanged(int event, int ioHandle, void *param2); 427 }; 428 429 class AudioPolicyServiceClient: public IBinder::DeathRecipient 430 { 431 public: 432 AudioPolicyServiceClient() { 433 } 434 435 // DeathRecipient 436 virtual void binderDied(const wp<IBinder>& who); 437 }; 438 439 static sp<AudioFlingerClient> gAudioFlingerClient; 440 static sp<AudioPolicyServiceClient> gAudioPolicyServiceClient; 441 friend class AudioFlingerClient; 442 friend class AudioPolicyServiceClient; 443 444 static Mutex gLock; 445 static sp<IAudioFlinger> gAudioFlinger; 446 static audio_error_callback gAudioErrorCallback; 447 448 static size_t gInBuffSize; 449 // previous parameters for recording buffer size queries 450 static uint32_t gPrevInSamplingRate; 451 static int gPrevInFormat; 452 static int gPrevInChannelCount; 453 454 static sp<IAudioPolicyService> gAudioPolicyService; 455 456 // mapping between stream types and outputs 457 static DefaultKeyedVector<int, audio_io_handle_t> gStreamOutputMap; 458 // list of output descritor containing cached parameters (sampling rate, framecount, channel count...) 459 static DefaultKeyedVector<audio_io_handle_t, OutputDescriptor *> gOutputs; 460 }; 461 462 class AudioParameter { 463 464 public: 465 AudioParameter() {} 466 AudioParameter(const String8& keyValuePairs); 467 virtual ~AudioParameter(); 468 469 // reserved parameter keys for changeing standard parameters with setParameters() function. 470 // Using these keys is mandatory for AudioFlinger to properly monitor audio output/input 471 // configuration changes and act accordingly. 472 // keyRouting: to change audio routing, value is an int in AudioSystem::audio_devices 473 // keySamplingRate: to change sampling rate routing, value is an int 474 // keyFormat: to change audio format, value is an int in AudioSystem::audio_format 475 // keyChannels: to change audio channel configuration, value is an int in AudioSystem::audio_channels 476 // keyFrameCount: to change audio output frame count, value is an int 477 static const char *keyRouting; 478 static const char *keySamplingRate; 479 static const char *keyFormat; 480 static const char *keyChannels; 481 static const char *keyFrameCount; 482 483 String8 toString(); 484 485 status_t add(const String8& key, const String8& value); 486 status_t addInt(const String8& key, const int value); 487 status_t addFloat(const String8& key, const float value); 488 489 status_t remove(const String8& key); 490 491 status_t get(const String8& key, String8& value); 492 status_t getInt(const String8& key, int& value); 493 status_t getFloat(const String8& key, float& value); 494 status_t getAt(size_t index, String8& key, String8& value); 495 496 size_t size() { return mParameters.size(); } 497 498 private: 499 String8 mKeyValuePairs; 500 KeyedVector <String8, String8> mParameters; 501 }; 502 503 }; // namespace android 504 505 #endif /*ANDROID_AUDIOSYSTEM_H_*/ 506
