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 MODE_IN_COMMUNICATION, 160 NUM_MODES // not a valid entry, denotes end-of-list 161 }; 162 163 enum audio_in_acoustics { 164 AGC_ENABLE = 0x0001, 165 AGC_DISABLE = 0, 166 NS_ENABLE = 0x0002, 167 NS_DISABLE = 0, 168 TX_IIR_ENABLE = 0x0004, 169 TX_DISABLE = 0 170 }; 171 172 // special audio session values 173 enum audio_sessions { 174 SESSION_OUTPUT_STAGE = -1, // session for effects attached to a particular output stream 175 // (value must be less than 0) 176 SESSION_OUTPUT_MIX = 0, // session for effects applied to output mix. These effects can 177 // be moved by audio policy manager to another output stream 178 // (value must be 0) 179 }; 180 181 /* These are static methods to control the system-wide AudioFlinger 182 * only privileged processes can have access to them 183 */ 184 185 // mute/unmute microphone 186 static status_t muteMicrophone(bool state); 187 static status_t isMicrophoneMuted(bool *state); 188 189 // set/get master volume 190 static status_t setMasterVolume(float value); 191 static status_t getMasterVolume(float* volume); 192 // mute/unmute audio outputs 193 static status_t setMasterMute(bool mute); 194 static status_t getMasterMute(bool* mute); 195 196 // set/get stream volume on specified output 197 static status_t setStreamVolume(int stream, float value, int output); 198 static status_t getStreamVolume(int stream, float* volume, int output); 199 200 // mute/unmute stream 201 static status_t setStreamMute(int stream, bool mute); 202 static status_t getStreamMute(int stream, bool* mute); 203 204 // set audio mode in audio hardware (see AudioSystem::audio_mode) 205 static status_t setMode(int mode); 206 207 // returns true in *state if tracks are active on the specified stream 208 static status_t isStreamActive(int stream, bool *state); 209 210 // set/get audio hardware parameters. The function accepts a list of parameters 211 // key value pairs in the form: key1=value1;key2=value2;... 212 // Some keys are reserved for standard parameters (See AudioParameter class). 213 static status_t setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs); 214 static String8 getParameters(audio_io_handle_t ioHandle, const String8& keys); 215 216 static void setErrorCallback(audio_error_callback cb); 217 218 // helper function to obtain AudioFlinger service handle 219 static const sp<IAudioFlinger>& get_audio_flinger(); 220 221 static float linearToLog(int volume); 222 static int logToLinear(float volume); 223 224 static status_t getOutputSamplingRate(int* samplingRate, int stream = DEFAULT); 225 static status_t getOutputFrameCount(int* frameCount, int stream = DEFAULT); 226 static status_t getOutputLatency(uint32_t* latency, int stream = DEFAULT); 227 228 static bool routedToA2dpOutput(int streamType); 229 230 static status_t getInputBufferSize(uint32_t sampleRate, int format, int channelCount, 231 size_t* buffSize); 232 233 static status_t setVoiceVolume(float volume); 234 235 // return the number of audio frames written by AudioFlinger to audio HAL and 236 // audio dsp to DAC since the output on which the specificed stream is playing 237 // has exited standby. 238 // returned status (from utils/Errors.h) can be: 239 // - NO_ERROR: successful operation, halFrames and dspFrames point to valid data 240 // - INVALID_OPERATION: Not supported on current hardware platform 241 // - BAD_VALUE: invalid parameter 242 // NOTE: this feature is not supported on all hardware platforms and it is 243 // necessary to check returned status before using the returned values. 244 static status_t getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, int stream = DEFAULT); 245 246 static unsigned int getInputFramesLost(audio_io_handle_t ioHandle); 247 248 static int newAudioSessionId(); 249 // 250 // AudioPolicyService interface 251 // 252 253 enum audio_devices { 254 // output devices 255 DEVICE_OUT_EARPIECE = 0x1, 256 DEVICE_OUT_SPEAKER = 0x2, 257 DEVICE_OUT_WIRED_HEADSET = 0x4, 258 DEVICE_OUT_WIRED_HEADPHONE = 0x8, 259 DEVICE_OUT_BLUETOOTH_SCO = 0x10, 260 DEVICE_OUT_BLUETOOTH_SCO_HEADSET = 0x20, 261 DEVICE_OUT_BLUETOOTH_SCO_CARKIT = 0x40, 262 DEVICE_OUT_BLUETOOTH_A2DP = 0x80, 263 DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES = 0x100, 264 DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER = 0x200, 265 DEVICE_OUT_AUX_DIGITAL = 0x400, 266 DEVICE_OUT_DEFAULT = 0x8000, 267 DEVICE_OUT_ALL = (DEVICE_OUT_EARPIECE | DEVICE_OUT_SPEAKER | DEVICE_OUT_WIRED_HEADSET | 268 DEVICE_OUT_WIRED_HEADPHONE | DEVICE_OUT_BLUETOOTH_SCO | DEVICE_OUT_BLUETOOTH_SCO_HEADSET | 269 DEVICE_OUT_BLUETOOTH_SCO_CARKIT | DEVICE_OUT_BLUETOOTH_A2DP | DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES | 270 DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER | DEVICE_OUT_AUX_DIGITAL | DEVICE_OUT_DEFAULT), 271 DEVICE_OUT_ALL_A2DP = (DEVICE_OUT_BLUETOOTH_A2DP | DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES | 272 DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER), 273 274 // input devices 275 DEVICE_IN_COMMUNICATION = 0x10000, 276 DEVICE_IN_AMBIENT = 0x20000, 277 DEVICE_IN_BUILTIN_MIC = 0x40000, 278 DEVICE_IN_BLUETOOTH_SCO_HEADSET = 0x80000, 279 DEVICE_IN_WIRED_HEADSET = 0x100000, 280 DEVICE_IN_AUX_DIGITAL = 0x200000, 281 DEVICE_IN_VOICE_CALL = 0x400000, 282 DEVICE_IN_BACK_MIC = 0x800000, 283 DEVICE_IN_DEFAULT = 0x80000000, 284 285 DEVICE_IN_ALL = (DEVICE_IN_COMMUNICATION | DEVICE_IN_AMBIENT | DEVICE_IN_BUILTIN_MIC | 286 DEVICE_IN_BLUETOOTH_SCO_HEADSET | DEVICE_IN_WIRED_HEADSET | DEVICE_IN_AUX_DIGITAL | 287 DEVICE_IN_VOICE_CALL | DEVICE_IN_BACK_MIC | DEVICE_IN_DEFAULT) 288 }; 289 290 // device connection states used for setDeviceConnectionState() 291 enum device_connection_state { 292 DEVICE_STATE_UNAVAILABLE, 293 DEVICE_STATE_AVAILABLE, 294 NUM_DEVICE_STATES 295 }; 296 297 // request to open a direct output with getOutput() (by opposition to sharing an output with other AudioTracks) 298 enum output_flags { 299 OUTPUT_FLAG_INDIRECT = 0x0, 300 OUTPUT_FLAG_DIRECT = 0x1 301 }; 302 303 // device categories used for setForceUse() 304 enum forced_config { 305 FORCE_NONE, 306 FORCE_SPEAKER, 307 FORCE_HEADPHONES, 308 FORCE_BT_SCO, 309 FORCE_BT_A2DP, 310 FORCE_WIRED_ACCESSORY, 311 FORCE_BT_CAR_DOCK, 312 FORCE_BT_DESK_DOCK, 313 NUM_FORCE_CONFIG, 314 FORCE_DEFAULT = FORCE_NONE 315 }; 316 317 // usages used for setForceUse() 318 enum force_use { 319 FOR_COMMUNICATION, 320 FOR_MEDIA, 321 FOR_RECORD, 322 FOR_DOCK, 323 NUM_FORCE_USE 324 }; 325 326 // types of io configuration change events received with ioConfigChanged() 327 enum io_config_event { 328 OUTPUT_OPENED, 329 OUTPUT_CLOSED, 330 OUTPUT_CONFIG_CHANGED, 331 INPUT_OPENED, 332 INPUT_CLOSED, 333 INPUT_CONFIG_CHANGED, 334 STREAM_CONFIG_CHANGED, 335 NUM_CONFIG_EVENTS 336 }; 337 338 // audio output descritor used to cache output configurations in client process to avoid frequent calls 339 // through IAudioFlinger 340 class OutputDescriptor { 341 public: 342 OutputDescriptor() 343 : samplingRate(0), format(0), channels(0), frameCount(0), latency(0) {} 344 345 uint32_t samplingRate; 346 int32_t format; 347 int32_t channels; 348 size_t frameCount; 349 uint32_t latency; 350 }; 351 352 // 353 // IAudioPolicyService interface (see AudioPolicyInterface for method descriptions) 354 // 355 static status_t setDeviceConnectionState(audio_devices device, device_connection_state state, const char *device_address); 356 static device_connection_state getDeviceConnectionState(audio_devices device, const char *device_address); 357 static status_t setPhoneState(int state); 358 static status_t setRingerMode(uint32_t mode, uint32_t mask); 359 static status_t setForceUse(force_use usage, forced_config config); 360 static forced_config getForceUse(force_use usage); 361 static audio_io_handle_t getOutput(stream_type stream, 362 uint32_t samplingRate = 0, 363 uint32_t format = FORMAT_DEFAULT, 364 uint32_t channels = CHANNEL_OUT_STEREO, 365 output_flags flags = OUTPUT_FLAG_INDIRECT); 366 static status_t startOutput(audio_io_handle_t output, 367 AudioSystem::stream_type stream, 368 int session = 0); 369 static status_t stopOutput(audio_io_handle_t output, 370 AudioSystem::stream_type stream, 371 int session = 0); 372 static void releaseOutput(audio_io_handle_t output); 373 static audio_io_handle_t getInput(int inputSource, 374 uint32_t samplingRate = 0, 375 uint32_t format = FORMAT_DEFAULT, 376 uint32_t channels = CHANNEL_IN_MONO, 377 audio_in_acoustics acoustics = (audio_in_acoustics)0); 378 static status_t startInput(audio_io_handle_t input); 379 static status_t stopInput(audio_io_handle_t input); 380 static void releaseInput(audio_io_handle_t input); 381 static status_t initStreamVolume(stream_type stream, 382 int indexMin, 383 int indexMax); 384 static status_t setStreamVolumeIndex(stream_type stream, int index); 385 static status_t getStreamVolumeIndex(stream_type stream, int *index); 386 387 static uint32_t getStrategyForStream(stream_type stream); 388 389 static audio_io_handle_t getOutputForEffect(effect_descriptor_t *desc); 390 static status_t registerEffect(effect_descriptor_t *desc, 391 audio_io_handle_t output, 392 uint32_t strategy, 393 int session, 394 int id); 395 static status_t unregisterEffect(int id); 396 397 static const sp<IAudioPolicyService>& get_audio_policy_service(); 398 399 // ---------------------------------------------------------------------------- 400 401 static uint32_t popCount(uint32_t u); 402 static bool isOutputDevice(audio_devices device); 403 static bool isInputDevice(audio_devices device); 404 static bool isA2dpDevice(audio_devices device); 405 static bool isBluetoothScoDevice(audio_devices device); 406 static bool isLowVisibility(stream_type stream); 407 static bool isOutputChannel(uint32_t channel); 408 static bool isInputChannel(uint32_t channel); 409 static bool isValidFormat(uint32_t format); 410 static bool isLinearPCM(uint32_t format); 411 412 private: 413 414 class AudioFlingerClient: public IBinder::DeathRecipient, public BnAudioFlingerClient 415 { 416 public: 417 AudioFlingerClient() { 418 } 419 420 // DeathRecipient 421 virtual void binderDied(const wp<IBinder>& who); 422 423 // IAudioFlingerClient 424 425 // indicate a change in the configuration of an output or input: keeps the cached 426 // values for output/input parameters upto date in client process 427 virtual void ioConfigChanged(int event, int ioHandle, void *param2); 428 }; 429 430 class AudioPolicyServiceClient: public IBinder::DeathRecipient 431 { 432 public: 433 AudioPolicyServiceClient() { 434 } 435 436 // DeathRecipient 437 virtual void binderDied(const wp<IBinder>& who); 438 }; 439 440 static sp<AudioFlingerClient> gAudioFlingerClient; 441 static sp<AudioPolicyServiceClient> gAudioPolicyServiceClient; 442 friend class AudioFlingerClient; 443 friend class AudioPolicyServiceClient; 444 445 static Mutex gLock; 446 static sp<IAudioFlinger> gAudioFlinger; 447 static audio_error_callback gAudioErrorCallback; 448 449 static size_t gInBuffSize; 450 // previous parameters for recording buffer size queries 451 static uint32_t gPrevInSamplingRate; 452 static int gPrevInFormat; 453 static int gPrevInChannelCount; 454 455 static sp<IAudioPolicyService> gAudioPolicyService; 456 457 // mapping between stream types and outputs 458 static DefaultKeyedVector<int, audio_io_handle_t> gStreamOutputMap; 459 // list of output descritor containing cached parameters (sampling rate, framecount, channel count...) 460 static DefaultKeyedVector<audio_io_handle_t, OutputDescriptor *> gOutputs; 461 }; 462 463 class AudioParameter { 464 465 public: 466 AudioParameter() {} 467 AudioParameter(const String8& keyValuePairs); 468 virtual ~AudioParameter(); 469 470 // reserved parameter keys for changing standard parameters with setParameters() function. 471 // Using these keys is mandatory for AudioFlinger to properly monitor audio output/input 472 // configuration changes and act accordingly. 473 // keyRouting: to change audio routing, value is an int in AudioSystem::audio_devices 474 // keySamplingRate: to change sampling rate routing, value is an int 475 // keyFormat: to change audio format, value is an int in AudioSystem::audio_format 476 // keyChannels: to change audio channel configuration, value is an int in AudioSystem::audio_channels 477 // keyFrameCount: to change audio output frame count, value is an int 478 // keyInputSource: to change audio input source, value is an int in audio_source 479 // (defined in media/mediarecorder.h) 480 static const char *keyRouting; 481 static const char *keySamplingRate; 482 static const char *keyFormat; 483 static const char *keyChannels; 484 static const char *keyFrameCount; 485 static const char *keyInputSource; 486 487 String8 toString(); 488 489 status_t add(const String8& key, const String8& value); 490 status_t addInt(const String8& key, const int value); 491 status_t addFloat(const String8& key, const float value); 492 493 status_t remove(const String8& key); 494 495 status_t get(const String8& key, String8& value); 496 status_t getInt(const String8& key, int& value); 497 status_t getFloat(const String8& key, float& value); 498 status_t getAt(size_t index, String8& key, String8& value); 499 500 size_t size() { return mParameters.size(); } 501 502 private: 503 String8 mKeyValuePairs; 504 KeyedVector <String8, String8> mParameters; 505 }; 506 507 }; // namespace android 508 509 #endif /*ANDROID_AUDIOSYSTEM_H_*/ 510
