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