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