1 /* 2 * Copyright (C) 2011 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 18 #ifndef ANDROID_AUDIO_HAL_INTERFACE_H 19 #define ANDROID_AUDIO_HAL_INTERFACE_H 20 21 #include <stdint.h> 22 #include <strings.h> 23 #include <sys/cdefs.h> 24 #include <sys/types.h> 25 26 #include <cutils/bitops.h> 27 28 #include <hardware/hardware.h> 29 #include <system/audio.h> 30 #include <hardware/audio_effect.h> 31 32 __BEGIN_DECLS 33 34 /** 35 * The id of this module 36 */ 37 #define AUDIO_HARDWARE_MODULE_ID "audio" 38 39 /** 40 * Name of the audio devices to open 41 */ 42 #define AUDIO_HARDWARE_INTERFACE "audio_hw_if" 43 44 45 /* Use version 0.1 to be compatible with first generation of audio hw module with version_major 46 * hardcoded to 1. No audio module API change. 47 */ 48 #define AUDIO_MODULE_API_VERSION_0_1 HARDWARE_MODULE_API_VERSION(0, 1) 49 #define AUDIO_MODULE_API_VERSION_CURRENT AUDIO_MODULE_API_VERSION_0_1 50 51 /* First generation of audio devices had version hardcoded to 0. all devices with versions < 1.0 52 * will be considered of first generation API. 53 */ 54 #define AUDIO_DEVICE_API_VERSION_0_0 HARDWARE_DEVICE_API_VERSION(0, 0) 55 #define AUDIO_DEVICE_API_VERSION_1_0 HARDWARE_DEVICE_API_VERSION(1, 0) 56 #define AUDIO_DEVICE_API_VERSION_2_0 HARDWARE_DEVICE_API_VERSION(2, 0) 57 #define AUDIO_DEVICE_API_VERSION_3_0 HARDWARE_DEVICE_API_VERSION(3, 0) 58 #define AUDIO_DEVICE_API_VERSION_CURRENT AUDIO_DEVICE_API_VERSION_3_0 59 /* Minimal audio HAL version supported by the audio framework */ 60 #define AUDIO_DEVICE_API_VERSION_MIN AUDIO_DEVICE_API_VERSION_2_0 61 62 /** 63 * List of known audio HAL modules. This is the base name of the audio HAL 64 * library composed of the "audio." prefix, one of the base names below and 65 * a suffix specific to the device. 66 * e.g: audio.primary.goldfish.so or audio.a2dp.default.so 67 */ 68 69 #define AUDIO_HARDWARE_MODULE_ID_PRIMARY "primary" 70 #define AUDIO_HARDWARE_MODULE_ID_A2DP "a2dp" 71 #define AUDIO_HARDWARE_MODULE_ID_USB "usb" 72 #define AUDIO_HARDWARE_MODULE_ID_REMOTE_SUBMIX "r_submix" 73 #define AUDIO_HARDWARE_MODULE_ID_CODEC_OFFLOAD "codec_offload" 74 75 /**************************************/ 76 77 /** 78 * standard audio parameters that the HAL may need to handle 79 */ 80 81 /** 82 * audio device parameters 83 */ 84 85 /* BT SCO Noise Reduction + Echo Cancellation parameters */ 86 #define AUDIO_PARAMETER_KEY_BT_NREC "bt_headset_nrec" 87 #define AUDIO_PARAMETER_VALUE_ON "on" 88 #define AUDIO_PARAMETER_VALUE_OFF "off" 89 90 /* TTY mode selection */ 91 #define AUDIO_PARAMETER_KEY_TTY_MODE "tty_mode" 92 #define AUDIO_PARAMETER_VALUE_TTY_OFF "tty_off" 93 #define AUDIO_PARAMETER_VALUE_TTY_VCO "tty_vco" 94 #define AUDIO_PARAMETER_VALUE_TTY_HCO "tty_hco" 95 #define AUDIO_PARAMETER_VALUE_TTY_FULL "tty_full" 96 97 /* Hearing Aid Compatibility - Telecoil (HAC-T) mode on/off 98 Strings must be in sync with CallFeaturesSetting.java */ 99 #define AUDIO_PARAMETER_KEY_HAC "HACSetting" 100 #define AUDIO_PARAMETER_VALUE_HAC_ON "ON" 101 #define AUDIO_PARAMETER_VALUE_HAC_OFF "OFF" 102 103 /* A2DP sink address set by framework */ 104 #define AUDIO_PARAMETER_A2DP_SINK_ADDRESS "a2dp_sink_address" 105 106 /* A2DP source address set by framework */ 107 #define AUDIO_PARAMETER_A2DP_SOURCE_ADDRESS "a2dp_source_address" 108 109 /* Screen state */ 110 #define AUDIO_PARAMETER_KEY_SCREEN_STATE "screen_state" 111 112 /* Bluetooth SCO wideband */ 113 #define AUDIO_PARAMETER_KEY_BT_SCO_WB "bt_wbs" 114 115 /* Get a new HW synchronization source identifier. 116 * Return a valid source (positive integer) or AUDIO_HW_SYNC_INVALID if an error occurs 117 * or no HW sync is available. */ 118 #define AUDIO_PARAMETER_HW_AV_SYNC "hw_av_sync" 119 120 /** 121 * audio stream parameters 122 */ 123 124 #define AUDIO_PARAMETER_STREAM_ROUTING "routing" /* audio_devices_t */ 125 #define AUDIO_PARAMETER_STREAM_FORMAT "format" /* audio_format_t */ 126 #define AUDIO_PARAMETER_STREAM_CHANNELS "channels" /* audio_channel_mask_t */ 127 #define AUDIO_PARAMETER_STREAM_FRAME_COUNT "frame_count" /* size_t */ 128 #define AUDIO_PARAMETER_STREAM_INPUT_SOURCE "input_source" /* audio_source_t */ 129 #define AUDIO_PARAMETER_STREAM_SAMPLING_RATE "sampling_rate" /* uint32_t */ 130 131 #define AUDIO_PARAMETER_DEVICE_CONNECT "connect" /* audio_devices_t */ 132 #define AUDIO_PARAMETER_DEVICE_DISCONNECT "disconnect" /* audio_devices_t */ 133 134 /* Query supported formats. The response is a '|' separated list of strings from 135 * audio_format_t enum e.g: "sup_formats=AUDIO_FORMAT_PCM_16_BIT" */ 136 #define AUDIO_PARAMETER_STREAM_SUP_FORMATS "sup_formats" 137 /* Query supported channel masks. The response is a '|' separated list of strings from 138 * audio_channel_mask_t enum e.g: "sup_channels=AUDIO_CHANNEL_OUT_STEREO|AUDIO_CHANNEL_OUT_MONO" */ 139 #define AUDIO_PARAMETER_STREAM_SUP_CHANNELS "sup_channels" 140 /* Query supported sampling rates. The response is a '|' separated list of integer values e.g: 141 * "sup_sampling_rates=44100|48000" */ 142 #define AUDIO_PARAMETER_STREAM_SUP_SAMPLING_RATES "sup_sampling_rates" 143 144 /* Set the HW synchronization source for an output stream. */ 145 #define AUDIO_PARAMETER_STREAM_HW_AV_SYNC "hw_av_sync" 146 147 /* Enable mono audio playback if 1, else should be 0. */ 148 #define AUDIO_PARAMETER_MONO_OUTPUT "mono_output" 149 150 /** 151 * audio codec parameters 152 */ 153 154 #define AUDIO_OFFLOAD_CODEC_PARAMS "music_offload_codec_param" 155 #define AUDIO_OFFLOAD_CODEC_BIT_PER_SAMPLE "music_offload_bit_per_sample" 156 #define AUDIO_OFFLOAD_CODEC_BIT_RATE "music_offload_bit_rate" 157 #define AUDIO_OFFLOAD_CODEC_AVG_BIT_RATE "music_offload_avg_bit_rate" 158 #define AUDIO_OFFLOAD_CODEC_ID "music_offload_codec_id" 159 #define AUDIO_OFFLOAD_CODEC_BLOCK_ALIGN "music_offload_block_align" 160 #define AUDIO_OFFLOAD_CODEC_SAMPLE_RATE "music_offload_sample_rate" 161 #define AUDIO_OFFLOAD_CODEC_ENCODE_OPTION "music_offload_encode_option" 162 #define AUDIO_OFFLOAD_CODEC_NUM_CHANNEL "music_offload_num_channels" 163 #define AUDIO_OFFLOAD_CODEC_DOWN_SAMPLING "music_offload_down_sampling" 164 #define AUDIO_OFFLOAD_CODEC_DELAY_SAMPLES "delay_samples" 165 #define AUDIO_OFFLOAD_CODEC_PADDING_SAMPLES "padding_samples" 166 167 /**************************************/ 168 169 /* common audio stream parameters and operations */ 170 struct audio_stream { 171 172 /** 173 * Return the sampling rate in Hz - eg. 44100. 174 */ 175 uint32_t (*get_sample_rate)(const struct audio_stream *stream); 176 177 /* currently unused - use set_parameters with key 178 * AUDIO_PARAMETER_STREAM_SAMPLING_RATE 179 */ 180 int (*set_sample_rate)(struct audio_stream *stream, uint32_t rate); 181 182 /** 183 * Return size of input/output buffer in bytes for this stream - eg. 4800. 184 * It should be a multiple of the frame size. See also get_input_buffer_size. 185 */ 186 size_t (*get_buffer_size)(const struct audio_stream *stream); 187 188 /** 189 * Return the channel mask - 190 * e.g. AUDIO_CHANNEL_OUT_STEREO or AUDIO_CHANNEL_IN_STEREO 191 */ 192 audio_channel_mask_t (*get_channels)(const struct audio_stream *stream); 193 194 /** 195 * Return the audio format - e.g. AUDIO_FORMAT_PCM_16_BIT 196 */ 197 audio_format_t (*get_format)(const struct audio_stream *stream); 198 199 /* currently unused - use set_parameters with key 200 * AUDIO_PARAMETER_STREAM_FORMAT 201 */ 202 int (*set_format)(struct audio_stream *stream, audio_format_t format); 203 204 /** 205 * Put the audio hardware input/output into standby mode. 206 * Driver should exit from standby mode at the next I/O operation. 207 * Returns 0 on success and <0 on failure. 208 */ 209 int (*standby)(struct audio_stream *stream); 210 211 /** dump the state of the audio input/output device */ 212 int (*dump)(const struct audio_stream *stream, int fd); 213 214 /** Return the set of device(s) which this stream is connected to */ 215 audio_devices_t (*get_device)(const struct audio_stream *stream); 216 217 /** 218 * Currently unused - set_device() corresponds to set_parameters() with key 219 * AUDIO_PARAMETER_STREAM_ROUTING for both input and output. 220 * AUDIO_PARAMETER_STREAM_INPUT_SOURCE is an additional information used by 221 * input streams only. 222 */ 223 int (*set_device)(struct audio_stream *stream, audio_devices_t device); 224 225 /** 226 * set/get audio stream parameters. The function accepts a list of 227 * parameter key value pairs in the form: key1=value1;key2=value2;... 228 * 229 * Some keys are reserved for standard parameters (See AudioParameter class) 230 * 231 * If the implementation does not accept a parameter change while 232 * the output is active but the parameter is acceptable otherwise, it must 233 * return -ENOSYS. 234 * 235 * The audio flinger will put the stream in standby and then change the 236 * parameter value. 237 */ 238 int (*set_parameters)(struct audio_stream *stream, const char *kv_pairs); 239 240 /* 241 * Returns a pointer to a heap allocated string. The caller is responsible 242 * for freeing the memory for it using free(). 243 */ 244 char * (*get_parameters)(const struct audio_stream *stream, 245 const char *keys); 246 int (*add_audio_effect)(const struct audio_stream *stream, 247 effect_handle_t effect); 248 int (*remove_audio_effect)(const struct audio_stream *stream, 249 effect_handle_t effect); 250 }; 251 typedef struct audio_stream audio_stream_t; 252 253 /* type of asynchronous write callback events. Mutually exclusive */ 254 typedef enum { 255 STREAM_CBK_EVENT_WRITE_READY, /* non blocking write completed */ 256 STREAM_CBK_EVENT_DRAIN_READY, /* drain completed */ 257 STREAM_CBK_EVENT_ERROR, /* stream hit some error, let AF take action */ 258 } stream_callback_event_t; 259 260 typedef int (*stream_callback_t)(stream_callback_event_t event, void *param, void *cookie); 261 262 /* type of drain requested to audio_stream_out->drain(). Mutually exclusive */ 263 typedef enum { 264 AUDIO_DRAIN_ALL, /* drain() returns when all data has been played */ 265 AUDIO_DRAIN_EARLY_NOTIFY /* drain() returns a short time before all data 266 from the current track has been played to 267 give time for gapless track switch */ 268 } audio_drain_type_t; 269 270 /** 271 * audio_stream_out is the abstraction interface for the audio output hardware. 272 * 273 * It provides information about various properties of the audio output 274 * hardware driver. 275 */ 276 277 struct audio_stream_out { 278 /** 279 * Common methods of the audio stream out. This *must* be the first member of audio_stream_out 280 * as users of this structure will cast a audio_stream to audio_stream_out pointer in contexts 281 * where it's known the audio_stream references an audio_stream_out. 282 */ 283 struct audio_stream common; 284 285 /** 286 * Return the audio hardware driver estimated latency in milliseconds. 287 */ 288 uint32_t (*get_latency)(const struct audio_stream_out *stream); 289 290 /** 291 * Use this method in situations where audio mixing is done in the 292 * hardware. This method serves as a direct interface with hardware, 293 * allowing you to directly set the volume as apposed to via the framework. 294 * This method might produce multiple PCM outputs or hardware accelerated 295 * codecs, such as MP3 or AAC. 296 */ 297 int (*set_volume)(struct audio_stream_out *stream, float left, float right); 298 299 /** 300 * Write audio buffer to driver. Returns number of bytes written, or a 301 * negative status_t. If at least one frame was written successfully prior to the error, 302 * it is suggested that the driver return that successful (short) byte count 303 * and then return an error in the subsequent call. 304 * 305 * If set_callback() has previously been called to enable non-blocking mode 306 * the write() is not allowed to block. It must write only the number of 307 * bytes that currently fit in the driver/hardware buffer and then return 308 * this byte count. If this is less than the requested write size the 309 * callback function must be called when more space is available in the 310 * driver/hardware buffer. 311 */ 312 ssize_t (*write)(struct audio_stream_out *stream, const void* buffer, 313 size_t bytes); 314 315 /* return the number of audio frames written by the audio dsp to DAC since 316 * the output has exited standby 317 */ 318 int (*get_render_position)(const struct audio_stream_out *stream, 319 uint32_t *dsp_frames); 320 321 /** 322 * get the local time at which the next write to the audio driver will be presented. 323 * The units are microseconds, where the epoch is decided by the local audio HAL. 324 */ 325 int (*get_next_write_timestamp)(const struct audio_stream_out *stream, 326 int64_t *timestamp); 327 328 /** 329 * set the callback function for notifying completion of non-blocking 330 * write and drain. 331 * Calling this function implies that all future write() and drain() 332 * must be non-blocking and use the callback to signal completion. 333 */ 334 int (*set_callback)(struct audio_stream_out *stream, 335 stream_callback_t callback, void *cookie); 336 337 /** 338 * Notifies to the audio driver to stop playback however the queued buffers are 339 * retained by the hardware. Useful for implementing pause/resume. Empty implementation 340 * if not supported however should be implemented for hardware with non-trivial 341 * latency. In the pause state audio hardware could still be using power. User may 342 * consider calling suspend after a timeout. 343 * 344 * Implementation of this function is mandatory for offloaded playback. 345 */ 346 int (*pause)(struct audio_stream_out* stream); 347 348 /** 349 * Notifies to the audio driver to resume playback following a pause. 350 * Returns error if called without matching pause. 351 * 352 * Implementation of this function is mandatory for offloaded playback. 353 */ 354 int (*resume)(struct audio_stream_out* stream); 355 356 /** 357 * Requests notification when data buffered by the driver/hardware has 358 * been played. If set_callback() has previously been called to enable 359 * non-blocking mode, the drain() must not block, instead it should return 360 * quickly and completion of the drain is notified through the callback. 361 * If set_callback() has not been called, the drain() must block until 362 * completion. 363 * If type==AUDIO_DRAIN_ALL, the drain completes when all previously written 364 * data has been played. 365 * If type==AUDIO_DRAIN_EARLY_NOTIFY, the drain completes shortly before all 366 * data for the current track has played to allow time for the framework 367 * to perform a gapless track switch. 368 * 369 * Drain must return immediately on stop() and flush() call 370 * 371 * Implementation of this function is mandatory for offloaded playback. 372 */ 373 int (*drain)(struct audio_stream_out* stream, audio_drain_type_t type ); 374 375 /** 376 * Notifies to the audio driver to flush the queued data. Stream must already 377 * be paused before calling flush(). 378 * 379 * Implementation of this function is mandatory for offloaded playback. 380 */ 381 int (*flush)(struct audio_stream_out* stream); 382 383 /** 384 * Return a recent count of the number of audio frames presented to an external observer. 385 * This excludes frames which have been written but are still in the pipeline. 386 * The count is not reset to zero when output enters standby. 387 * Also returns the value of CLOCK_MONOTONIC as of this presentation count. 388 * The returned count is expected to be 'recent', 389 * but does not need to be the most recent possible value. 390 * However, the associated time should correspond to whatever count is returned. 391 * Example: assume that N+M frames have been presented, where M is a 'small' number. 392 * Then it is permissible to return N instead of N+M, 393 * and the timestamp should correspond to N rather than N+M. 394 * The terms 'recent' and 'small' are not defined. 395 * They reflect the quality of the implementation. 396 * 397 * 3.0 and higher only. 398 */ 399 int (*get_presentation_position)(const struct audio_stream_out *stream, 400 uint64_t *frames, struct timespec *timestamp); 401 402 }; 403 typedef struct audio_stream_out audio_stream_out_t; 404 405 struct audio_stream_in { 406 /** 407 * Common methods of the audio stream in. This *must* be the first member of audio_stream_in 408 * as users of this structure will cast a audio_stream to audio_stream_in pointer in contexts 409 * where it's known the audio_stream references an audio_stream_in. 410 */ 411 struct audio_stream common; 412 413 /** set the input gain for the audio driver. This method is for 414 * for future use */ 415 int (*set_gain)(struct audio_stream_in *stream, float gain); 416 417 /** Read audio buffer in from audio driver. Returns number of bytes read, or a 418 * negative status_t. If at least one frame was read prior to the error, 419 * read should return that byte count and then return an error in the subsequent call. 420 */ 421 ssize_t (*read)(struct audio_stream_in *stream, void* buffer, 422 size_t bytes); 423 424 /** 425 * Return the amount of input frames lost in the audio driver since the 426 * last call of this function. 427 * Audio driver is expected to reset the value to 0 and restart counting 428 * upon returning the current value by this function call. 429 * Such loss typically occurs when the user space process is blocked 430 * longer than the capacity of audio driver buffers. 431 * 432 * Unit: the number of input audio frames 433 */ 434 uint32_t (*get_input_frames_lost)(struct audio_stream_in *stream); 435 436 /** 437 * Return a recent count of the number of audio frames received and 438 * the clock time associated with that frame count. 439 * 440 * frames is the total frame count received. This should be as early in 441 * the capture pipeline as possible. In general, 442 * frames should be non-negative and should not go "backwards". 443 * 444 * time is the clock MONOTONIC time when frames was measured. In general, 445 * time should be a positive quantity and should not go "backwards". 446 * 447 * The status returned is 0 on success, -ENOSYS if the device is not 448 * ready/available, or -EINVAL if the arguments are null or otherwise invalid. 449 */ 450 int (*get_capture_position)(const struct audio_stream_in *stream, 451 int64_t *frames, int64_t *time); 452 }; 453 typedef struct audio_stream_in audio_stream_in_t; 454 455 /** 456 * return the frame size (number of bytes per sample). 457 * 458 * Deprecated: use audio_stream_out_frame_size() or audio_stream_in_frame_size() instead. 459 */ 460 __attribute__((__deprecated__)) 461 static inline size_t audio_stream_frame_size(const struct audio_stream *s) 462 { 463 size_t chan_samp_sz; 464 audio_format_t format = s->get_format(s); 465 466 if (audio_has_proportional_frames(format)) { 467 chan_samp_sz = audio_bytes_per_sample(format); 468 return popcount(s->get_channels(s)) * chan_samp_sz; 469 } 470 471 return sizeof(int8_t); 472 } 473 474 /** 475 * return the frame size (number of bytes per sample) of an output stream. 476 */ 477 static inline size_t audio_stream_out_frame_size(const struct audio_stream_out *s) 478 { 479 size_t chan_samp_sz; 480 audio_format_t format = s->common.get_format(&s->common); 481 482 if (audio_has_proportional_frames(format)) { 483 chan_samp_sz = audio_bytes_per_sample(format); 484 return audio_channel_count_from_out_mask(s->common.get_channels(&s->common)) * chan_samp_sz; 485 } 486 487 return sizeof(int8_t); 488 } 489 490 /** 491 * return the frame size (number of bytes per sample) of an input stream. 492 */ 493 static inline size_t audio_stream_in_frame_size(const struct audio_stream_in *s) 494 { 495 size_t chan_samp_sz; 496 audio_format_t format = s->common.get_format(&s->common); 497 498 if (audio_has_proportional_frames(format)) { 499 chan_samp_sz = audio_bytes_per_sample(format); 500 return audio_channel_count_from_in_mask(s->common.get_channels(&s->common)) * chan_samp_sz; 501 } 502 503 return sizeof(int8_t); 504 } 505 506 /**********************************************************************/ 507 508 /** 509 * Every hardware module must have a data structure named HAL_MODULE_INFO_SYM 510 * and the fields of this data structure must begin with hw_module_t 511 * followed by module specific information. 512 */ 513 struct audio_module { 514 struct hw_module_t common; 515 }; 516 517 struct audio_hw_device { 518 /** 519 * Common methods of the audio device. This *must* be the first member of audio_hw_device 520 * as users of this structure will cast a hw_device_t to audio_hw_device pointer in contexts 521 * where it's known the hw_device_t references an audio_hw_device. 522 */ 523 struct hw_device_t common; 524 525 /** 526 * used by audio flinger to enumerate what devices are supported by 527 * each audio_hw_device implementation. 528 * 529 * Return value is a bitmask of 1 or more values of audio_devices_t 530 * 531 * NOTE: audio HAL implementations starting with 532 * AUDIO_DEVICE_API_VERSION_2_0 do not implement this function. 533 * All supported devices should be listed in audio_policy.conf 534 * file and the audio policy manager must choose the appropriate 535 * audio module based on information in this file. 536 */ 537 uint32_t (*get_supported_devices)(const struct audio_hw_device *dev); 538 539 /** 540 * check to see if the audio hardware interface has been initialized. 541 * returns 0 on success, -ENODEV on failure. 542 */ 543 int (*init_check)(const struct audio_hw_device *dev); 544 545 /** set the audio volume of a voice call. Range is between 0.0 and 1.0 */ 546 int (*set_voice_volume)(struct audio_hw_device *dev, float volume); 547 548 /** 549 * set the audio volume for all audio activities other than voice call. 550 * Range between 0.0 and 1.0. If any value other than 0 is returned, 551 * the software mixer will emulate this capability. 552 */ 553 int (*set_master_volume)(struct audio_hw_device *dev, float volume); 554 555 /** 556 * Get the current master volume value for the HAL, if the HAL supports 557 * master volume control. AudioFlinger will query this value from the 558 * primary audio HAL when the service starts and use the value for setting 559 * the initial master volume across all HALs. HALs which do not support 560 * this method may leave it set to NULL. 561 */ 562 int (*get_master_volume)(struct audio_hw_device *dev, float *volume); 563 564 /** 565 * set_mode is called when the audio mode changes. AUDIO_MODE_NORMAL mode 566 * is for standard audio playback, AUDIO_MODE_RINGTONE when a ringtone is 567 * playing, and AUDIO_MODE_IN_CALL when a call is in progress. 568 */ 569 int (*set_mode)(struct audio_hw_device *dev, audio_mode_t mode); 570 571 /* mic mute */ 572 int (*set_mic_mute)(struct audio_hw_device *dev, bool state); 573 int (*get_mic_mute)(const struct audio_hw_device *dev, bool *state); 574 575 /* set/get global audio parameters */ 576 int (*set_parameters)(struct audio_hw_device *dev, const char *kv_pairs); 577 578 /* 579 * Returns a pointer to a heap allocated string. The caller is responsible 580 * for freeing the memory for it using free(). 581 */ 582 char * (*get_parameters)(const struct audio_hw_device *dev, 583 const char *keys); 584 585 /* Returns audio input buffer size according to parameters passed or 586 * 0 if one of the parameters is not supported. 587 * See also get_buffer_size which is for a particular stream. 588 */ 589 size_t (*get_input_buffer_size)(const struct audio_hw_device *dev, 590 const struct audio_config *config); 591 592 /** This method creates and opens the audio hardware output stream. 593 * The "address" parameter qualifies the "devices" audio device type if needed. 594 * The format format depends on the device type: 595 * - Bluetooth devices use the MAC address of the device in the form "00:11:22:AA:BB:CC" 596 * - USB devices use the ALSA card and device numbers in the form "card=X;device=Y" 597 * - Other devices may use a number or any other string. 598 */ 599 600 int (*open_output_stream)(struct audio_hw_device *dev, 601 audio_io_handle_t handle, 602 audio_devices_t devices, 603 audio_output_flags_t flags, 604 struct audio_config *config, 605 struct audio_stream_out **stream_out, 606 const char *address); 607 608 void (*close_output_stream)(struct audio_hw_device *dev, 609 struct audio_stream_out* stream_out); 610 611 /** This method creates and opens the audio hardware input stream */ 612 int (*open_input_stream)(struct audio_hw_device *dev, 613 audio_io_handle_t handle, 614 audio_devices_t devices, 615 struct audio_config *config, 616 struct audio_stream_in **stream_in, 617 audio_input_flags_t flags, 618 const char *address, 619 audio_source_t source); 620 621 void (*close_input_stream)(struct audio_hw_device *dev, 622 struct audio_stream_in *stream_in); 623 624 /** This method dumps the state of the audio hardware */ 625 int (*dump)(const struct audio_hw_device *dev, int fd); 626 627 /** 628 * set the audio mute status for all audio activities. If any value other 629 * than 0 is returned, the software mixer will emulate this capability. 630 */ 631 int (*set_master_mute)(struct audio_hw_device *dev, bool mute); 632 633 /** 634 * Get the current master mute status for the HAL, if the HAL supports 635 * master mute control. AudioFlinger will query this value from the primary 636 * audio HAL when the service starts and use the value for setting the 637 * initial master mute across all HALs. HALs which do not support this 638 * method may leave it set to NULL. 639 */ 640 int (*get_master_mute)(struct audio_hw_device *dev, bool *mute); 641 642 /** 643 * Routing control 644 */ 645 646 /* Creates an audio patch between several source and sink ports. 647 * The handle is allocated by the HAL and should be unique for this 648 * audio HAL module. */ 649 int (*create_audio_patch)(struct audio_hw_device *dev, 650 unsigned int num_sources, 651 const struct audio_port_config *sources, 652 unsigned int num_sinks, 653 const struct audio_port_config *sinks, 654 audio_patch_handle_t *handle); 655 656 /* Release an audio patch */ 657 int (*release_audio_patch)(struct audio_hw_device *dev, 658 audio_patch_handle_t handle); 659 660 /* Fills the list of supported attributes for a given audio port. 661 * As input, "port" contains the information (type, role, address etc...) 662 * needed by the HAL to identify the port. 663 * As output, "port" contains possible attributes (sampling rates, formats, 664 * channel masks, gain controllers...) for this port. 665 */ 666 int (*get_audio_port)(struct audio_hw_device *dev, 667 struct audio_port *port); 668 669 /* Set audio port configuration */ 670 int (*set_audio_port_config)(struct audio_hw_device *dev, 671 const struct audio_port_config *config); 672 673 }; 674 typedef struct audio_hw_device audio_hw_device_t; 675 676 /** convenience API for opening and closing a supported device */ 677 678 static inline int audio_hw_device_open(const struct hw_module_t* module, 679 struct audio_hw_device** device) 680 { 681 return module->methods->open(module, AUDIO_HARDWARE_INTERFACE, 682 (struct hw_device_t**)device); 683 } 684 685 static inline int audio_hw_device_close(struct audio_hw_device* device) 686 { 687 return device->common.close(&device->common); 688 } 689 690 691 __END_DECLS 692 693 #endif // ANDROID_AUDIO_INTERFACE_H 694
