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 #include <time.h> 26 27 #include <cutils/bitops.h> 28 29 #include <hardware/hardware.h> 30 #include <system/audio.h> 31 #include <hardware/audio_effect.h> 32 33 __BEGIN_DECLS 34 35 /** 36 * The id of this module 37 */ 38 #define AUDIO_HARDWARE_MODULE_ID "audio" 39 40 /** 41 * Name of the audio devices to open 42 */ 43 #define AUDIO_HARDWARE_INTERFACE "audio_hw_if" 44 45 46 /* Use version 0.1 to be compatible with first generation of audio hw module with version_major 47 * hardcoded to 1. No audio module API change. 48 */ 49 #define AUDIO_MODULE_API_VERSION_0_1 HARDWARE_MODULE_API_VERSION(0, 1) 50 #define AUDIO_MODULE_API_VERSION_CURRENT AUDIO_MODULE_API_VERSION_0_1 51 52 /* First generation of audio devices had version hardcoded to 0. all devices with versions < 1.0 53 * will be considered of first generation API. 54 */ 55 #define AUDIO_DEVICE_API_VERSION_0_0 HARDWARE_DEVICE_API_VERSION(0, 0) 56 #define AUDIO_DEVICE_API_VERSION_1_0 HARDWARE_DEVICE_API_VERSION(1, 0) 57 #define AUDIO_DEVICE_API_VERSION_2_0 HARDWARE_DEVICE_API_VERSION(2, 0) 58 #define AUDIO_DEVICE_API_VERSION_3_0 HARDWARE_DEVICE_API_VERSION(3, 0) 59 #define AUDIO_DEVICE_API_VERSION_CURRENT AUDIO_DEVICE_API_VERSION_3_0 60 /* Minimal audio HAL version supported by the audio framework */ 61 #define AUDIO_DEVICE_API_VERSION_MIN AUDIO_DEVICE_API_VERSION_2_0 62 63 /**************************************/ 64 65 /** 66 * standard audio parameters that the HAL may need to handle 67 */ 68 69 /** 70 * audio device parameters 71 */ 72 73 /* TTY mode selection */ 74 #define AUDIO_PARAMETER_KEY_TTY_MODE "tty_mode" 75 #define AUDIO_PARAMETER_VALUE_TTY_OFF "tty_off" 76 #define AUDIO_PARAMETER_VALUE_TTY_VCO "tty_vco" 77 #define AUDIO_PARAMETER_VALUE_TTY_HCO "tty_hco" 78 #define AUDIO_PARAMETER_VALUE_TTY_FULL "tty_full" 79 80 /* Hearing Aid Compatibility - Telecoil (HAC-T) mode on/off */ 81 #define AUDIO_PARAMETER_KEY_HAC "HACSetting" 82 #define AUDIO_PARAMETER_VALUE_HAC_ON "ON" 83 #define AUDIO_PARAMETER_VALUE_HAC_OFF "OFF" 84 85 /* A2DP sink address set by framework */ 86 #define AUDIO_PARAMETER_A2DP_SINK_ADDRESS "a2dp_sink_address" 87 88 /* A2DP source address set by framework */ 89 #define AUDIO_PARAMETER_A2DP_SOURCE_ADDRESS "a2dp_source_address" 90 91 /* Bluetooth SCO wideband */ 92 #define AUDIO_PARAMETER_KEY_BT_SCO_WB "bt_wbs" 93 94 /* BT SCO headset name for debug */ 95 #define AUDIO_PARAMETER_KEY_BT_SCO_HEADSET_NAME "bt_headset_name" 96 97 /* BT SCO HFP control */ 98 #define AUDIO_PARAMETER_KEY_HFP_ENABLE "hfp_enable" 99 #define AUDIO_PARAMETER_KEY_HFP_SET_SAMPLING_RATE "hfp_set_sampling_rate" 100 #define AUDIO_PARAMETER_KEY_HFP_VOLUME "hfp_volume" 101 102 /* Set screen orientation */ 103 #define AUDIO_PARAMETER_KEY_ROTATION "rotation" 104 105 /** 106 * audio stream parameters 107 */ 108 109 /* Enable AANC */ 110 #define AUDIO_PARAMETER_KEY_AANC "aanc_enabled" 111 112 /**************************************/ 113 114 /* common audio stream parameters and operations */ 115 struct audio_stream { 116 117 /** 118 * Return the sampling rate in Hz - eg. 44100. 119 */ 120 uint32_t (*get_sample_rate)(const struct audio_stream *stream); 121 122 /* currently unused - use set_parameters with key 123 * AUDIO_PARAMETER_STREAM_SAMPLING_RATE 124 */ 125 int (*set_sample_rate)(struct audio_stream *stream, uint32_t rate); 126 127 /** 128 * Return size of input/output buffer in bytes for this stream - eg. 4800. 129 * It should be a multiple of the frame size. See also get_input_buffer_size. 130 */ 131 size_t (*get_buffer_size)(const struct audio_stream *stream); 132 133 /** 134 * Return the channel mask - 135 * e.g. AUDIO_CHANNEL_OUT_STEREO or AUDIO_CHANNEL_IN_STEREO 136 */ 137 audio_channel_mask_t (*get_channels)(const struct audio_stream *stream); 138 139 /** 140 * Return the audio format - e.g. AUDIO_FORMAT_PCM_16_BIT 141 */ 142 audio_format_t (*get_format)(const struct audio_stream *stream); 143 144 /* currently unused - use set_parameters with key 145 * AUDIO_PARAMETER_STREAM_FORMAT 146 */ 147 int (*set_format)(struct audio_stream *stream, audio_format_t format); 148 149 /** 150 * Put the audio hardware input/output into standby mode. 151 * Driver should exit from standby mode at the next I/O operation. 152 * Returns 0 on success and <0 on failure. 153 */ 154 int (*standby)(struct audio_stream *stream); 155 156 /** dump the state of the audio input/output device */ 157 int (*dump)(const struct audio_stream *stream, int fd); 158 159 /** Return the set of device(s) which this stream is connected to */ 160 audio_devices_t (*get_device)(const struct audio_stream *stream); 161 162 /** 163 * Currently unused - set_device() corresponds to set_parameters() with key 164 * AUDIO_PARAMETER_STREAM_ROUTING for both input and output. 165 * AUDIO_PARAMETER_STREAM_INPUT_SOURCE is an additional information used by 166 * input streams only. 167 */ 168 int (*set_device)(struct audio_stream *stream, audio_devices_t device); 169 170 /** 171 * set/get audio stream parameters. The function accepts a list of 172 * parameter key value pairs in the form: key1=value1;key2=value2;... 173 * 174 * Some keys are reserved for standard parameters (See AudioParameter class) 175 * 176 * If the implementation does not accept a parameter change while 177 * the output is active but the parameter is acceptable otherwise, it must 178 * return -ENOSYS. 179 * 180 * The audio flinger will put the stream in standby and then change the 181 * parameter value. 182 */ 183 int (*set_parameters)(struct audio_stream *stream, const char *kv_pairs); 184 185 /* 186 * Returns a pointer to a heap allocated string. The caller is responsible 187 * for freeing the memory for it using free(). 188 */ 189 char * (*get_parameters)(const struct audio_stream *stream, 190 const char *keys); 191 int (*add_audio_effect)(const struct audio_stream *stream, 192 effect_handle_t effect); 193 int (*remove_audio_effect)(const struct audio_stream *stream, 194 effect_handle_t effect); 195 }; 196 typedef struct audio_stream audio_stream_t; 197 198 /* type of asynchronous write callback events. Mutually exclusive */ 199 typedef enum { 200 STREAM_CBK_EVENT_WRITE_READY, /* non blocking write completed */ 201 STREAM_CBK_EVENT_DRAIN_READY, /* drain completed */ 202 STREAM_CBK_EVENT_ERROR, /* stream hit some error, let AF take action */ 203 } stream_callback_event_t; 204 205 typedef int (*stream_callback_t)(stream_callback_event_t event, void *param, void *cookie); 206 207 /* type of drain requested to audio_stream_out->drain(). Mutually exclusive */ 208 typedef enum { 209 AUDIO_DRAIN_ALL, /* drain() returns when all data has been played */ 210 AUDIO_DRAIN_EARLY_NOTIFY /* drain() returns a short time before all data 211 from the current track has been played to 212 give time for gapless track switch */ 213 } audio_drain_type_t; 214 215 typedef struct source_metadata { 216 size_t track_count; 217 /** Array of metadata of each track connected to this source. */ 218 struct playback_track_metadata* tracks; 219 } source_metadata_t; 220 221 typedef struct sink_metadata { 222 size_t track_count; 223 /** Array of metadata of each track connected to this sink. */ 224 struct record_track_metadata* tracks; 225 } sink_metadata_t; 226 227 /** 228 * audio_stream_out is the abstraction interface for the audio output hardware. 229 * 230 * It provides information about various properties of the audio output 231 * hardware driver. 232 */ 233 struct audio_stream_out { 234 /** 235 * Common methods of the audio stream out. This *must* be the first member of audio_stream_out 236 * as users of this structure will cast a audio_stream to audio_stream_out pointer in contexts 237 * where it's known the audio_stream references an audio_stream_out. 238 */ 239 struct audio_stream common; 240 241 /** 242 * Return the audio hardware driver estimated latency in milliseconds. 243 */ 244 uint32_t (*get_latency)(const struct audio_stream_out *stream); 245 246 /** 247 * Use this method in situations where audio mixing is done in the 248 * hardware. This method serves as a direct interface with hardware, 249 * allowing you to directly set the volume as apposed to via the framework. 250 * This method might produce multiple PCM outputs or hardware accelerated 251 * codecs, such as MP3 or AAC. 252 */ 253 int (*set_volume)(struct audio_stream_out *stream, float left, float right); 254 255 /** 256 * Write audio buffer to driver. Returns number of bytes written, or a 257 * negative status_t. If at least one frame was written successfully prior to the error, 258 * it is suggested that the driver return that successful (short) byte count 259 * and then return an error in the subsequent call. 260 * 261 * If set_callback() has previously been called to enable non-blocking mode 262 * the write() is not allowed to block. It must write only the number of 263 * bytes that currently fit in the driver/hardware buffer and then return 264 * this byte count. If this is less than the requested write size the 265 * callback function must be called when more space is available in the 266 * driver/hardware buffer. 267 */ 268 ssize_t (*write)(struct audio_stream_out *stream, const void* buffer, 269 size_t bytes); 270 271 /* return the number of audio frames written by the audio dsp to DAC since 272 * the output has exited standby 273 */ 274 int (*get_render_position)(const struct audio_stream_out *stream, 275 uint32_t *dsp_frames); 276 277 /** 278 * get the local time at which the next write to the audio driver will be presented. 279 * The units are microseconds, where the epoch is decided by the local audio HAL. 280 */ 281 int (*get_next_write_timestamp)(const struct audio_stream_out *stream, 282 int64_t *timestamp); 283 284 /** 285 * set the callback function for notifying completion of non-blocking 286 * write and drain. 287 * Calling this function implies that all future write() and drain() 288 * must be non-blocking and use the callback to signal completion. 289 */ 290 int (*set_callback)(struct audio_stream_out *stream, 291 stream_callback_t callback, void *cookie); 292 293 /** 294 * Notifies to the audio driver to stop playback however the queued buffers are 295 * retained by the hardware. Useful for implementing pause/resume. Empty implementation 296 * if not supported however should be implemented for hardware with non-trivial 297 * latency. In the pause state audio hardware could still be using power. User may 298 * consider calling suspend after a timeout. 299 * 300 * Implementation of this function is mandatory for offloaded playback. 301 */ 302 int (*pause)(struct audio_stream_out* stream); 303 304 /** 305 * Notifies to the audio driver to resume playback following a pause. 306 * Returns error if called without matching pause. 307 * 308 * Implementation of this function is mandatory for offloaded playback. 309 */ 310 int (*resume)(struct audio_stream_out* stream); 311 312 /** 313 * Requests notification when data buffered by the driver/hardware has 314 * been played. If set_callback() has previously been called to enable 315 * non-blocking mode, the drain() must not block, instead it should return 316 * quickly and completion of the drain is notified through the callback. 317 * If set_callback() has not been called, the drain() must block until 318 * completion. 319 * If type==AUDIO_DRAIN_ALL, the drain completes when all previously written 320 * data has been played. 321 * If type==AUDIO_DRAIN_EARLY_NOTIFY, the drain completes shortly before all 322 * data for the current track has played to allow time for the framework 323 * to perform a gapless track switch. 324 * 325 * Drain must return immediately on stop() and flush() call 326 * 327 * Implementation of this function is mandatory for offloaded playback. 328 */ 329 int (*drain)(struct audio_stream_out* stream, audio_drain_type_t type ); 330 331 /** 332 * Notifies to the audio driver to flush the queued data. Stream must already 333 * be paused before calling flush(). 334 * 335 * Implementation of this function is mandatory for offloaded playback. 336 */ 337 int (*flush)(struct audio_stream_out* stream); 338 339 /** 340 * Return a recent count of the number of audio frames presented to an external observer. 341 * This excludes frames which have been written but are still in the pipeline. 342 * The count is not reset to zero when output enters standby. 343 * Also returns the value of CLOCK_MONOTONIC as of this presentation count. 344 * The returned count is expected to be 'recent', 345 * but does not need to be the most recent possible value. 346 * However, the associated time should correspond to whatever count is returned. 347 * Example: assume that N+M frames have been presented, where M is a 'small' number. 348 * Then it is permissible to return N instead of N+M, 349 * and the timestamp should correspond to N rather than N+M. 350 * The terms 'recent' and 'small' are not defined. 351 * They reflect the quality of the implementation. 352 * 353 * 3.0 and higher only. 354 */ 355 int (*get_presentation_position)(const struct audio_stream_out *stream, 356 uint64_t *frames, struct timespec *timestamp); 357 358 /** 359 * Called by the framework to start a stream operating in mmap mode. 360 * create_mmap_buffer must be called before calling start() 361 * 362 * \note Function only implemented by streams operating in mmap mode. 363 * 364 * \param[in] stream the stream object. 365 * \return 0 in case of success. 366 * -ENOSYS if called out of sequence or on non mmap stream 367 */ 368 int (*start)(const struct audio_stream_out* stream); 369 370 /** 371 * Called by the framework to stop a stream operating in mmap mode. 372 * Must be called after start() 373 * 374 * \note Function only implemented by streams operating in mmap mode. 375 * 376 * \param[in] stream the stream object. 377 * \return 0 in case of success. 378 * -ENOSYS if called out of sequence or on non mmap stream 379 */ 380 int (*stop)(const struct audio_stream_out* stream); 381 382 /** 383 * Called by the framework to retrieve information on the mmap buffer used for audio 384 * samples transfer. 385 * 386 * \note Function only implemented by streams operating in mmap mode. 387 * 388 * \param[in] stream the stream object. 389 * \param[in] min_size_frames minimum buffer size requested. The actual buffer 390 * size returned in struct audio_mmap_buffer_info can be larger. 391 * \param[out] info address at which the mmap buffer information should be returned. 392 * 393 * \return 0 if the buffer was allocated. 394 * -ENODEV in case of initialization error 395 * -EINVAL if the requested buffer size is too large 396 * -ENOSYS if called out of sequence (e.g. buffer already allocated) 397 */ 398 int (*create_mmap_buffer)(const struct audio_stream_out *stream, 399 int32_t min_size_frames, 400 struct audio_mmap_buffer_info *info); 401 402 /** 403 * Called by the framework to read current read/write position in the mmap buffer 404 * with associated time stamp. 405 * 406 * \note Function only implemented by streams operating in mmap mode. 407 * 408 * \param[in] stream the stream object. 409 * \param[out] position address at which the mmap read/write position should be returned. 410 * 411 * \return 0 if the position is successfully returned. 412 * -ENODATA if the position cannot be retrieved 413 * -ENOSYS if called before create_mmap_buffer() 414 */ 415 int (*get_mmap_position)(const struct audio_stream_out *stream, 416 struct audio_mmap_position *position); 417 418 /** 419 * Called when the metadata of the stream's source has been changed. 420 * @param source_metadata Description of the audio that is played by the clients. 421 */ 422 void (*update_source_metadata)(struct audio_stream_out *stream, 423 const struct source_metadata* source_metadata); 424 }; 425 typedef struct audio_stream_out audio_stream_out_t; 426 427 struct audio_stream_in { 428 /** 429 * Common methods of the audio stream in. This *must* be the first member of audio_stream_in 430 * as users of this structure will cast a audio_stream to audio_stream_in pointer in contexts 431 * where it's known the audio_stream references an audio_stream_in. 432 */ 433 struct audio_stream common; 434 435 /** set the input gain for the audio driver. This method is for 436 * for future use */ 437 int (*set_gain)(struct audio_stream_in *stream, float gain); 438 439 /** Read audio buffer in from audio driver. Returns number of bytes read, or a 440 * negative status_t. If at least one frame was read prior to the error, 441 * read should return that byte count and then return an error in the subsequent call. 442 */ 443 ssize_t (*read)(struct audio_stream_in *stream, void* buffer, 444 size_t bytes); 445 446 /** 447 * Return the amount of input frames lost in the audio driver since the 448 * last call of this function. 449 * Audio driver is expected to reset the value to 0 and restart counting 450 * upon returning the current value by this function call. 451 * Such loss typically occurs when the user space process is blocked 452 * longer than the capacity of audio driver buffers. 453 * 454 * Unit: the number of input audio frames 455 */ 456 uint32_t (*get_input_frames_lost)(struct audio_stream_in *stream); 457 458 /** 459 * Return a recent count of the number of audio frames received and 460 * the clock time associated with that frame count. 461 * 462 * frames is the total frame count received. This should be as early in 463 * the capture pipeline as possible. In general, 464 * frames should be non-negative and should not go "backwards". 465 * 466 * time is the clock MONOTONIC time when frames was measured. In general, 467 * time should be a positive quantity and should not go "backwards". 468 * 469 * The status returned is 0 on success, -ENOSYS if the device is not 470 * ready/available, or -EINVAL if the arguments are null or otherwise invalid. 471 */ 472 int (*get_capture_position)(const struct audio_stream_in *stream, 473 int64_t *frames, int64_t *time); 474 475 /** 476 * Called by the framework to start a stream operating in mmap mode. 477 * create_mmap_buffer must be called before calling start() 478 * 479 * \note Function only implemented by streams operating in mmap mode. 480 * 481 * \param[in] stream the stream object. 482 * \return 0 in case off success. 483 * -ENOSYS if called out of sequence or on non mmap stream 484 */ 485 int (*start)(const struct audio_stream_in* stream); 486 487 /** 488 * Called by the framework to stop a stream operating in mmap mode. 489 * 490 * \note Function only implemented by streams operating in mmap mode. 491 * 492 * \param[in] stream the stream object. 493 * \return 0 in case of success. 494 * -ENOSYS if called out of sequence or on non mmap stream 495 */ 496 int (*stop)(const struct audio_stream_in* stream); 497 498 /** 499 * Called by the framework to retrieve information on the mmap buffer used for audio 500 * samples transfer. 501 * 502 * \note Function only implemented by streams operating in mmap mode. 503 * 504 * \param[in] stream the stream object. 505 * \param[in] min_size_frames minimum buffer size requested. The actual buffer 506 * size returned in struct audio_mmap_buffer_info can be larger. 507 * \param[out] info address at which the mmap buffer information should be returned. 508 * 509 * \return 0 if the buffer was allocated. 510 * -ENODEV in case of initialization error 511 * -EINVAL if the requested buffer size is too large 512 * -ENOSYS if called out of sequence (e.g. buffer already allocated) 513 */ 514 int (*create_mmap_buffer)(const struct audio_stream_in *stream, 515 int32_t min_size_frames, 516 struct audio_mmap_buffer_info *info); 517 518 /** 519 * Called by the framework to read current read/write position in the mmap buffer 520 * with associated time stamp. 521 * 522 * \note Function only implemented by streams operating in mmap mode. 523 * 524 * \param[in] stream the stream object. 525 * \param[out] position address at which the mmap read/write position should be returned. 526 * 527 * \return 0 if the position is successfully returned. 528 * -ENODATA if the position cannot be retreived 529 * -ENOSYS if called before mmap_read_position() 530 */ 531 int (*get_mmap_position)(const struct audio_stream_in *stream, 532 struct audio_mmap_position *position); 533 534 /** 535 * Called by the framework to read active microphones 536 * 537 * \param[in] stream the stream object. 538 * \param[out] mic_array Pointer to first element on array with microphone info 539 * \param[out] mic_count When called, this holds the value of the max number of elements 540 * allowed in the mic_array. The actual number of elements written 541 * is returned here. 542 * if mic_count is passed as zero, mic_array will not be populated, 543 * and mic_count will return the actual number of active microphones. 544 * 545 * \return 0 if the microphone array is successfully filled. 546 * -ENOSYS if there is an error filling the data 547 */ 548 int (*get_active_microphones)(const struct audio_stream_in *stream, 549 struct audio_microphone_characteristic_t *mic_array, 550 size_t *mic_count); 551 552 /** 553 * Called when the metadata of the stream's sink has been changed. 554 * @param sink_metadata Description of the audio that is recorded by the clients. 555 */ 556 void (*update_sink_metadata)(struct audio_stream_in *stream, 557 const struct sink_metadata* sink_metadata); 558 }; 559 typedef struct audio_stream_in audio_stream_in_t; 560 561 /** 562 * return the frame size (number of bytes per sample). 563 * 564 * Deprecated: use audio_stream_out_frame_size() or audio_stream_in_frame_size() instead. 565 */ 566 __attribute__((__deprecated__)) 567 static inline size_t audio_stream_frame_size(const struct audio_stream *s) 568 { 569 size_t chan_samp_sz; 570 audio_format_t format = s->get_format(s); 571 572 if (audio_has_proportional_frames(format)) { 573 chan_samp_sz = audio_bytes_per_sample(format); 574 return popcount(s->get_channels(s)) * chan_samp_sz; 575 } 576 577 return sizeof(int8_t); 578 } 579 580 /** 581 * return the frame size (number of bytes per sample) of an output stream. 582 */ 583 static inline size_t audio_stream_out_frame_size(const struct audio_stream_out *s) 584 { 585 size_t chan_samp_sz; 586 audio_format_t format = s->common.get_format(&s->common); 587 588 if (audio_has_proportional_frames(format)) { 589 chan_samp_sz = audio_bytes_per_sample(format); 590 return audio_channel_count_from_out_mask(s->common.get_channels(&s->common)) * chan_samp_sz; 591 } 592 593 return sizeof(int8_t); 594 } 595 596 /** 597 * return the frame size (number of bytes per sample) of an input stream. 598 */ 599 static inline size_t audio_stream_in_frame_size(const struct audio_stream_in *s) 600 { 601 size_t chan_samp_sz; 602 audio_format_t format = s->common.get_format(&s->common); 603 604 if (audio_has_proportional_frames(format)) { 605 chan_samp_sz = audio_bytes_per_sample(format); 606 return audio_channel_count_from_in_mask(s->common.get_channels(&s->common)) * chan_samp_sz; 607 } 608 609 return sizeof(int8_t); 610 } 611 612 /**********************************************************************/ 613 614 /** 615 * Every hardware module must have a data structure named HAL_MODULE_INFO_SYM 616 * and the fields of this data structure must begin with hw_module_t 617 * followed by module specific information. 618 */ 619 struct audio_module { 620 struct hw_module_t common; 621 }; 622 623 struct audio_hw_device { 624 /** 625 * Common methods of the audio device. This *must* be the first member of audio_hw_device 626 * as users of this structure will cast a hw_device_t to audio_hw_device pointer in contexts 627 * where it's known the hw_device_t references an audio_hw_device. 628 */ 629 struct hw_device_t common; 630 631 /** 632 * used by audio flinger to enumerate what devices are supported by 633 * each audio_hw_device implementation. 634 * 635 * Return value is a bitmask of 1 or more values of audio_devices_t 636 * 637 * NOTE: audio HAL implementations starting with 638 * AUDIO_DEVICE_API_VERSION_2_0 do not implement this function. 639 * All supported devices should be listed in audio_policy.conf 640 * file and the audio policy manager must choose the appropriate 641 * audio module based on information in this file. 642 */ 643 uint32_t (*get_supported_devices)(const struct audio_hw_device *dev); 644 645 /** 646 * check to see if the audio hardware interface has been initialized. 647 * returns 0 on success, -ENODEV on failure. 648 */ 649 int (*init_check)(const struct audio_hw_device *dev); 650 651 /** set the audio volume of a voice call. Range is between 0.0 and 1.0 */ 652 int (*set_voice_volume)(struct audio_hw_device *dev, float volume); 653 654 /** 655 * set the audio volume for all audio activities other than voice call. 656 * Range between 0.0 and 1.0. If any value other than 0 is returned, 657 * the software mixer will emulate this capability. 658 */ 659 int (*set_master_volume)(struct audio_hw_device *dev, float volume); 660 661 /** 662 * Get the current master volume value for the HAL, if the HAL supports 663 * master volume control. AudioFlinger will query this value from the 664 * primary audio HAL when the service starts and use the value for setting 665 * the initial master volume across all HALs. HALs which do not support 666 * this method may leave it set to NULL. 667 */ 668 int (*get_master_volume)(struct audio_hw_device *dev, float *volume); 669 670 /** 671 * set_mode is called when the audio mode changes. AUDIO_MODE_NORMAL mode 672 * is for standard audio playback, AUDIO_MODE_RINGTONE when a ringtone is 673 * playing, and AUDIO_MODE_IN_CALL when a call is in progress. 674 */ 675 int (*set_mode)(struct audio_hw_device *dev, audio_mode_t mode); 676 677 /* mic mute */ 678 int (*set_mic_mute)(struct audio_hw_device *dev, bool state); 679 int (*get_mic_mute)(const struct audio_hw_device *dev, bool *state); 680 681 /* set/get global audio parameters */ 682 int (*set_parameters)(struct audio_hw_device *dev, const char *kv_pairs); 683 684 /* 685 * Returns a pointer to a heap allocated string. The caller is responsible 686 * for freeing the memory for it using free(). 687 */ 688 char * (*get_parameters)(const struct audio_hw_device *dev, 689 const char *keys); 690 691 /* Returns audio input buffer size according to parameters passed or 692 * 0 if one of the parameters is not supported. 693 * See also get_buffer_size which is for a particular stream. 694 */ 695 size_t (*get_input_buffer_size)(const struct audio_hw_device *dev, 696 const struct audio_config *config); 697 698 /** This method creates and opens the audio hardware output stream. 699 * The "address" parameter qualifies the "devices" audio device type if needed. 700 * The format format depends on the device type: 701 * - Bluetooth devices use the MAC address of the device in the form "00:11:22:AA:BB:CC" 702 * - USB devices use the ALSA card and device numbers in the form "card=X;device=Y" 703 * - Other devices may use a number or any other string. 704 */ 705 706 int (*open_output_stream)(struct audio_hw_device *dev, 707 audio_io_handle_t handle, 708 audio_devices_t devices, 709 audio_output_flags_t flags, 710 struct audio_config *config, 711 struct audio_stream_out **stream_out, 712 const char *address); 713 714 void (*close_output_stream)(struct audio_hw_device *dev, 715 struct audio_stream_out* stream_out); 716 717 /** This method creates and opens the audio hardware input stream */ 718 int (*open_input_stream)(struct audio_hw_device *dev, 719 audio_io_handle_t handle, 720 audio_devices_t devices, 721 struct audio_config *config, 722 struct audio_stream_in **stream_in, 723 audio_input_flags_t flags, 724 const char *address, 725 audio_source_t source); 726 727 void (*close_input_stream)(struct audio_hw_device *dev, 728 struct audio_stream_in *stream_in); 729 730 /** 731 * Called by the framework to read available microphones characteristics. 732 * 733 * \param[in] dev the hw_device object. 734 * \param[out] mic_array Pointer to first element on array with microphone info 735 * \param[out] mic_count When called, this holds the value of the max number of elements 736 * allowed in the mic_array. The actual number of elements written 737 * is returned here. 738 * if mic_count is passed as zero, mic_array will not be populated, 739 * and mic_count will return the actual number of microphones in the 740 * system. 741 * 742 * \return 0 if the microphone array is successfully filled. 743 * -ENOSYS if there is an error filling the data 744 */ 745 int (*get_microphones)(const struct audio_hw_device *dev, 746 struct audio_microphone_characteristic_t *mic_array, 747 size_t *mic_count); 748 749 /** This method dumps the state of the audio hardware */ 750 int (*dump)(const struct audio_hw_device *dev, int fd); 751 752 /** 753 * set the audio mute status for all audio activities. If any value other 754 * than 0 is returned, the software mixer will emulate this capability. 755 */ 756 int (*set_master_mute)(struct audio_hw_device *dev, bool mute); 757 758 /** 759 * Get the current master mute status for the HAL, if the HAL supports 760 * master mute control. AudioFlinger will query this value from the primary 761 * audio HAL when the service starts and use the value for setting the 762 * initial master mute across all HALs. HALs which do not support this 763 * method may leave it set to NULL. 764 */ 765 int (*get_master_mute)(struct audio_hw_device *dev, bool *mute); 766 767 /** 768 * Routing control 769 */ 770 771 /* Creates an audio patch between several source and sink ports. 772 * The handle is allocated by the HAL and should be unique for this 773 * audio HAL module. */ 774 int (*create_audio_patch)(struct audio_hw_device *dev, 775 unsigned int num_sources, 776 const struct audio_port_config *sources, 777 unsigned int num_sinks, 778 const struct audio_port_config *sinks, 779 audio_patch_handle_t *handle); 780 781 /* Release an audio patch */ 782 int (*release_audio_patch)(struct audio_hw_device *dev, 783 audio_patch_handle_t handle); 784 785 /* Fills the list of supported attributes for a given audio port. 786 * As input, "port" contains the information (type, role, address etc...) 787 * needed by the HAL to identify the port. 788 * As output, "port" contains possible attributes (sampling rates, formats, 789 * channel masks, gain controllers...) for this port. 790 */ 791 int (*get_audio_port)(struct audio_hw_device *dev, 792 struct audio_port *port); 793 794 /* Set audio port configuration */ 795 int (*set_audio_port_config)(struct audio_hw_device *dev, 796 const struct audio_port_config *config); 797 798 }; 799 typedef struct audio_hw_device audio_hw_device_t; 800 801 /** convenience API for opening and closing a supported device */ 802 803 static inline int audio_hw_device_open(const struct hw_module_t* module, 804 struct audio_hw_device** device) 805 { 806 return module->methods->open(module, AUDIO_HARDWARE_INTERFACE, 807 TO_HW_DEVICE_T_OPEN(device)); 808 } 809 810 static inline int audio_hw_device_close(struct audio_hw_device* device) 811 { 812 return device->common.close(&device->common); 813 } 814 815 816 __END_DECLS 817 818 #endif // ANDROID_AUDIO_INTERFACE_H 819