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