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_CURRENT AUDIO_DEVICE_API_VERSION_2_0 58 59/** 60 * List of known audio HAL modules. This is the base name of the audio HAL 61 * library composed of the "audio." prefix, one of the base names below and 62 * a suffix specific to the device. 63 * e.g: audio.primary.goldfish.so or audio.a2dp.default.so 64 */ 65 66#define AUDIO_HARDWARE_MODULE_ID_PRIMARY "primary" 67#define AUDIO_HARDWARE_MODULE_ID_A2DP "a2dp" 68#define AUDIO_HARDWARE_MODULE_ID_USB "usb" 69#define AUDIO_HARDWARE_MODULE_ID_REMOTE_SUBMIX "r_submix" 70#define AUDIO_HARDWARE_MODULE_ID_CODEC_OFFLOAD "codec_offload" 71 72/**************************************/ 73 74/** 75 * standard audio parameters that the HAL may need to handle 76 */ 77 78/** 79 * audio device parameters 80 */ 81 82/* BT SCO Noise Reduction + Echo Cancellation parameters */ 83#define AUDIO_PARAMETER_KEY_BT_NREC "bt_headset_nrec" 84#define AUDIO_PARAMETER_VALUE_ON "on" 85#define AUDIO_PARAMETER_VALUE_OFF "off" 86 87/* TTY mode selection */ 88#define AUDIO_PARAMETER_KEY_TTY_MODE "tty_mode" 89#define AUDIO_PARAMETER_VALUE_TTY_OFF "tty_off" 90#define AUDIO_PARAMETER_VALUE_TTY_VCO "tty_vco" 91#define AUDIO_PARAMETER_VALUE_TTY_HCO "tty_hco" 92#define AUDIO_PARAMETER_VALUE_TTY_FULL "tty_full" 93 94/* A2DP sink address set by framework */ 95#define AUDIO_PARAMETER_A2DP_SINK_ADDRESS "a2dp_sink_address" 96 97/* Screen state */ 98#define AUDIO_PARAMETER_KEY_SCREEN_STATE "screen_state" 99 100/** 101 * audio stream parameters 102 */ 103 104#define AUDIO_PARAMETER_STREAM_ROUTING "routing" // audio_devices_t 105#define AUDIO_PARAMETER_STREAM_FORMAT "format" // audio_format_t 106#define AUDIO_PARAMETER_STREAM_CHANNELS "channels" // audio_channel_mask_t 107#define AUDIO_PARAMETER_STREAM_FRAME_COUNT "frame_count" // size_t 108#define AUDIO_PARAMETER_STREAM_INPUT_SOURCE "input_source" // audio_source_t 109#define AUDIO_PARAMETER_STREAM_SAMPLING_RATE "sampling_rate" // uint32_t 110 111/* Query supported formats. The response is a '|' separated list of strings from 112 * audio_format_t enum e.g: "sup_formats=AUDIO_FORMAT_PCM_16_BIT" */ 113#define AUDIO_PARAMETER_STREAM_SUP_FORMATS "sup_formats" 114/* Query supported channel masks. The response is a '|' separated list of strings from 115 * audio_channel_mask_t enum e.g: "sup_channels=AUDIO_CHANNEL_OUT_STEREO|AUDIO_CHANNEL_OUT_MONO" */ 116#define AUDIO_PARAMETER_STREAM_SUP_CHANNELS "sup_channels" 117/* Query supported sampling rates. The response is a '|' separated list of integer values e.g: 118 * "sup_sampling_rates=44100|48000" */ 119#define AUDIO_PARAMETER_STREAM_SUP_SAMPLING_RATES "sup_sampling_rates" 120 121/** 122 * audio codec parameters 123 */ 124 125#define AUDIO_OFFLOAD_CODEC_PARAMS "music_offload_codec_param" 126#define AUDIO_OFFLOAD_CODEC_BIT_PER_SAMPLE "music_offload_bit_per_sample" 127#define AUDIO_OFFLOAD_CODEC_BIT_RATE "music_offload_bit_rate" 128#define AUDIO_OFFLOAD_CODEC_AVG_BIT_RATE "music_offload_avg_bit_rate" 129#define AUDIO_OFFLOAD_CODEC_ID "music_offload_codec_id" 130#define AUDIO_OFFLOAD_CODEC_BLOCK_ALIGN "music_offload_block_align" 131#define AUDIO_OFFLOAD_CODEC_SAMPLE_RATE "music_offload_sample_rate" 132#define AUDIO_OFFLOAD_CODEC_ENCODE_OPTION "music_offload_encode_option" 133#define AUDIO_OFFLOAD_CODEC_NUM_CHANNEL "music_offload_num_channels" 134#define AUDIO_OFFLOAD_CODEC_DOWN_SAMPLING "music_offload_down_sampling" 135#define AUDIO_OFFLOAD_CODEC_DELAY_SAMPLES "delay_samples" 136#define AUDIO_OFFLOAD_CODEC_PADDING_SAMPLES "padding_samples" 137 138/**************************************/ 139 140/* common audio stream configuration parameters 141 * You should memset() the entire structure to zero before use to 142 * ensure forward compatibility 143 */ 144struct audio_config { 145 uint32_t sample_rate; 146 audio_channel_mask_t channel_mask; 147 audio_format_t format; 148 audio_offload_info_t offload_info; 149}; 150typedef struct audio_config audio_config_t; 151 152/* common audio stream parameters and operations */ 153struct audio_stream { 154 155 /** 156 * Return the sampling rate in Hz - eg. 44100. 157 */ 158 uint32_t (*get_sample_rate)(const struct audio_stream *stream); 159 160 /* currently unused - use set_parameters with key 161 * AUDIO_PARAMETER_STREAM_SAMPLING_RATE 162 */ 163 int (*set_sample_rate)(struct audio_stream *stream, uint32_t rate); 164 165 /** 166 * Return size of input/output buffer in bytes for this stream - eg. 4800. 167 * It should be a multiple of the frame size. See also get_input_buffer_size. 168 */ 169 size_t (*get_buffer_size)(const struct audio_stream *stream); 170 171 /** 172 * Return the channel mask - 173 * e.g. AUDIO_CHANNEL_OUT_STEREO or AUDIO_CHANNEL_IN_STEREO 174 */ 175 audio_channel_mask_t (*get_channels)(const struct audio_stream *stream); 176 177 /** 178 * Return the audio format - e.g. AUDIO_FORMAT_PCM_16_BIT 179 */ 180 audio_format_t (*get_format)(const struct audio_stream *stream); 181 182 /* currently unused - use set_parameters with key 183 * AUDIO_PARAMETER_STREAM_FORMAT 184 */ 185 int (*set_format)(struct audio_stream *stream, audio_format_t format); 186 187 /** 188 * Put the audio hardware input/output into standby mode. 189 * Driver should exit from standby mode at the next I/O operation. 190 * Returns 0 on success and <0 on failure. 191 */ 192 int (*standby)(struct audio_stream *stream); 193 194 /** dump the state of the audio input/output device */ 195 int (*dump)(const struct audio_stream *stream, int fd); 196 197 /** Return the set of device(s) which this stream is connected to */ 198 audio_devices_t (*get_device)(const struct audio_stream *stream); 199 200 /** 201 * Currently unused - set_device() corresponds to set_parameters() with key 202 * AUDIO_PARAMETER_STREAM_ROUTING for both input and output. 203 * AUDIO_PARAMETER_STREAM_INPUT_SOURCE is an additional information used by 204 * input streams only. 205 */ 206 int (*set_device)(struct audio_stream *stream, audio_devices_t device); 207 208 /** 209 * set/get audio stream parameters. The function accepts a list of 210 * parameter key value pairs in the form: key1=value1;key2=value2;... 211 * 212 * Some keys are reserved for standard parameters (See AudioParameter class) 213 * 214 * If the implementation does not accept a parameter change while 215 * the output is active but the parameter is acceptable otherwise, it must 216 * return -ENOSYS. 217 * 218 * The audio flinger will put the stream in standby and then change the 219 * parameter value. 220 */ 221 int (*set_parameters)(struct audio_stream *stream, const char *kv_pairs); 222 223 /* 224 * Returns a pointer to a heap allocated string. The caller is responsible 225 * for freeing the memory for it using free(). 226 */ 227 char * (*get_parameters)(const struct audio_stream *stream, 228 const char *keys); 229 int (*add_audio_effect)(const struct audio_stream *stream, 230 effect_handle_t effect); 231 int (*remove_audio_effect)(const struct audio_stream *stream, 232 effect_handle_t effect); 233}; 234typedef struct audio_stream audio_stream_t; 235 236/* type of asynchronous write callback events. Mutually exclusive */ 237typedef enum { 238 STREAM_CBK_EVENT_WRITE_READY, /* non blocking write completed */ 239 STREAM_CBK_EVENT_DRAIN_READY /* drain completed */ 240} stream_callback_event_t; 241 242typedef int (*stream_callback_t)(stream_callback_event_t event, void *param, void *cookie); 243 244/* type of drain requested to audio_stream_out->drain(). Mutually exclusive */ 245typedef enum { 246 AUDIO_DRAIN_ALL, /* drain() returns when all data has been played */ 247 AUDIO_DRAIN_EARLY_NOTIFY /* drain() returns a short time before all data 248 from the current track has been played to 249 give time for gapless track switch */ 250} audio_drain_type_t; 251 252/** 253 * audio_stream_out is the abstraction interface for the audio output hardware. 254 * 255 * It provides information about various properties of the audio output 256 * hardware driver. 257 */ 258 259struct audio_stream_out { 260 struct audio_stream common; 261 262 /** 263 * Return the audio hardware driver estimated latency in milliseconds. 264 */ 265 uint32_t (*get_latency)(const struct audio_stream_out *stream); 266 267 /** 268 * Use this method in situations where audio mixing is done in the 269 * hardware. This method serves as a direct interface with hardware, 270 * allowing you to directly set the volume as apposed to via the framework. 271 * This method might produce multiple PCM outputs or hardware accelerated 272 * codecs, such as MP3 or AAC. 273 */ 274 int (*set_volume)(struct audio_stream_out *stream, float left, float right); 275 276 /** 277 * Write audio buffer to driver. Returns number of bytes written, or a 278 * negative status_t. If at least one frame was written successfully prior to the error, 279 * it is suggested that the driver return that successful (short) byte count 280 * and then return an error in the subsequent call. 281 * 282 * If set_callback() has previously been called to enable non-blocking mode 283 * the write() is not allowed to block. It must write only the number of 284 * bytes that currently fit in the driver/hardware buffer and then return 285 * this byte count. If this is less than the requested write size the 286 * callback function must be called when more space is available in the 287 * driver/hardware buffer. 288 */ 289 ssize_t (*write)(struct audio_stream_out *stream, const void* buffer, 290 size_t bytes); 291 292 /* return the number of audio frames written by the audio dsp to DAC since 293 * the output has exited standby 294 */ 295 int (*get_render_position)(const struct audio_stream_out *stream, 296 uint32_t *dsp_frames); 297 298 /** 299 * get the local time at which the next write to the audio driver will be presented. 300 * The units are microseconds, where the epoch is decided by the local audio HAL. 301 */ 302 int (*get_next_write_timestamp)(const struct audio_stream_out *stream, 303 int64_t *timestamp); 304 305 /** 306 * set the callback function for notifying completion of non-blocking 307 * write and drain. 308 * Calling this function implies that all future write() and drain() 309 * must be non-blocking and use the callback to signal completion. 310 */ 311 int (*set_callback)(struct audio_stream_out *stream, 312 stream_callback_t callback, void *cookie); 313 314 /** 315 * Notifies to the audio driver to stop playback however the queued buffers are 316 * retained by the hardware. Useful for implementing pause/resume. Empty implementation 317 * if not supported however should be implemented for hardware with non-trivial 318 * latency. In the pause state audio hardware could still be using power. User may 319 * consider calling suspend after a timeout. 320 * 321 * Implementation of this function is mandatory for offloaded playback. 322 */ 323 int (*pause)(struct audio_stream_out* stream); 324 325 /** 326 * Notifies to the audio driver to resume playback following a pause. 327 * Returns error if called without matching pause. 328 * 329 * Implementation of this function is mandatory for offloaded playback. 330 */ 331 int (*resume)(struct audio_stream_out* stream); 332 333 /** 334 * Requests notification when data buffered by the driver/hardware has 335 * been played. If set_callback() has previously been called to enable 336 * non-blocking mode, the drain() must not block, instead it should return 337 * quickly and completion of the drain is notified through the callback. 338 * If set_callback() has not been called, the drain() must block until 339 * completion. 340 * If type==AUDIO_DRAIN_ALL, the drain completes when all previously written 341 * data has been played. 342 * If type==AUDIO_DRAIN_EARLY_NOTIFY, the drain completes shortly before all 343 * data for the current track has played to allow time for the framework 344 * to perform a gapless track switch. 345 * 346 * Drain must return immediately on stop() and flush() call 347 * 348 * Implementation of this function is mandatory for offloaded playback. 349 */ 350 int (*drain)(struct audio_stream_out* stream, audio_drain_type_t type ); 351 352 /** 353 * Notifies to the audio driver to flush the queued data. Stream must already 354 * be paused before calling flush(). 355 * 356 * Implementation of this function is mandatory for offloaded playback. 357 */ 358 int (*flush)(struct audio_stream_out* stream); 359 360 /** 361 * Return a recent count of the number of audio frames presented to an external observer. 362 * This excludes frames which have been written but are still in the pipeline. 363 * The count is not reset to zero when output enters standby. 364 * Also returns the value of CLOCK_MONOTONIC as of this presentation count. 365 * The returned count is expected to be 'recent', 366 * but does not need to be the most recent possible value. 367 * However, the associated time should correspond to whatever count is returned. 368 * Example: assume that N+M frames have been presented, where M is a 'small' number. 369 * Then it is permissible to return N instead of N+M, 370 * and the timestamp should correspond to N rather than N+M. 371 * The terms 'recent' and 'small' are not defined. 372 * They reflect the quality of the implementation. 373 * 374 * 3.0 and higher only. 375 */ 376 int (*get_presentation_position)(const struct audio_stream_out *stream, 377 uint64_t *frames, struct timespec *timestamp); 378 379}; 380typedef struct audio_stream_out audio_stream_out_t; 381 382struct audio_stream_in { 383 struct audio_stream common; 384 385 /** set the input gain for the audio driver. This method is for 386 * for future use */ 387 int (*set_gain)(struct audio_stream_in *stream, float gain); 388 389 /** Read audio buffer in from audio driver. Returns number of bytes read, or a 390 * negative status_t. If at least one frame was read prior to the error, 391 * read should return that byte count and then return an error in the subsequent call. 392 */ 393 ssize_t (*read)(struct audio_stream_in *stream, void* buffer, 394 size_t bytes); 395 396 /** 397 * Return the amount of input frames lost in the audio driver since the 398 * last call of this function. 399 * Audio driver is expected to reset the value to 0 and restart counting 400 * upon returning the current value by this function call. 401 * Such loss typically occurs when the user space process is blocked 402 * longer than the capacity of audio driver buffers. 403 * 404 * Unit: the number of input audio frames 405 */ 406 uint32_t (*get_input_frames_lost)(struct audio_stream_in *stream); 407}; 408typedef struct audio_stream_in audio_stream_in_t; 409 410/** 411 * return the frame size (number of bytes per sample). 412 */ 413static inline size_t audio_stream_frame_size(const struct audio_stream *s) 414{ 415 size_t chan_samp_sz; 416 audio_format_t format = s->get_format(s); 417 418 if (audio_is_linear_pcm(format)) { 419 chan_samp_sz = audio_bytes_per_sample(format); 420 return popcount(s->get_channels(s)) * chan_samp_sz; 421 } 422 423 return sizeof(int8_t); 424} 425 426 427/**********************************************************************/ 428 429/** 430 * Every hardware module must have a data structure named HAL_MODULE_INFO_SYM 431 * and the fields of this data structure must begin with hw_module_t 432 * followed by module specific information. 433 */ 434struct audio_module { 435 struct hw_module_t common; 436}; 437 438struct audio_hw_device { 439 struct hw_device_t common; 440 441 /** 442 * used by audio flinger to enumerate what devices are supported by 443 * each audio_hw_device implementation. 444 * 445 * Return value is a bitmask of 1 or more values of audio_devices_t 446 * 447 * NOTE: audio HAL implementations starting with 448 * AUDIO_DEVICE_API_VERSION_2_0 do not implement this function. 449 * All supported devices should be listed in audio_policy.conf 450 * file and the audio policy manager must choose the appropriate 451 * audio module based on information in this file. 452 */ 453 uint32_t (*get_supported_devices)(const struct audio_hw_device *dev); 454 455 /** 456 * check to see if the audio hardware interface has been initialized. 457 * returns 0 on success, -ENODEV on failure. 458 */ 459 int (*init_check)(const struct audio_hw_device *dev); 460 461 /** set the audio volume of a voice call. Range is between 0.0 and 1.0 */ 462 int (*set_voice_volume)(struct audio_hw_device *dev, float volume); 463 464 /** 465 * set the audio volume for all audio activities other than voice call. 466 * Range between 0.0 and 1.0. If any value other than 0 is returned, 467 * the software mixer will emulate this capability. 468 */ 469 int (*set_master_volume)(struct audio_hw_device *dev, float volume); 470 471 /** 472 * Get the current master volume value for the HAL, if the HAL supports 473 * master volume control. AudioFlinger will query this value from the 474 * primary audio HAL when the service starts and use the value for setting 475 * the initial master volume across all HALs. HALs which do not support 476 * this method may leave it set to NULL. 477 */ 478 int (*get_master_volume)(struct audio_hw_device *dev, float *volume); 479 480 /** 481 * set_mode is called when the audio mode changes. AUDIO_MODE_NORMAL mode 482 * is for standard audio playback, AUDIO_MODE_RINGTONE when a ringtone is 483 * playing, and AUDIO_MODE_IN_CALL when a call is in progress. 484 */ 485 int (*set_mode)(struct audio_hw_device *dev, audio_mode_t mode); 486 487 /* mic mute */ 488 int (*set_mic_mute)(struct audio_hw_device *dev, bool state); 489 int (*get_mic_mute)(const struct audio_hw_device *dev, bool *state); 490 491 /* set/get global audio parameters */ 492 int (*set_parameters)(struct audio_hw_device *dev, const char *kv_pairs); 493 494 /* 495 * Returns a pointer to a heap allocated string. The caller is responsible 496 * for freeing the memory for it using free(). 497 */ 498 char * (*get_parameters)(const struct audio_hw_device *dev, 499 const char *keys); 500 501 /* Returns audio input buffer size according to parameters passed or 502 * 0 if one of the parameters is not supported. 503 * See also get_buffer_size which is for a particular stream. 504 */ 505 size_t (*get_input_buffer_size)(const struct audio_hw_device *dev, 506 const struct audio_config *config); 507 508 /** This method creates and opens the audio hardware output stream */ 509 int (*open_output_stream)(struct audio_hw_device *dev, 510 audio_io_handle_t handle, 511 audio_devices_t devices, 512 audio_output_flags_t flags, 513 struct audio_config *config, 514 struct audio_stream_out **stream_out); 515 516 void (*close_output_stream)(struct audio_hw_device *dev, 517 struct audio_stream_out* stream_out); 518 519 /** This method creates and opens the audio hardware input stream */ 520 int (*open_input_stream)(struct audio_hw_device *dev, 521 audio_io_handle_t handle, 522 audio_devices_t devices, 523 struct audio_config *config, 524 struct audio_stream_in **stream_in); 525 526 void (*close_input_stream)(struct audio_hw_device *dev, 527 struct audio_stream_in *stream_in); 528 529 /** This method dumps the state of the audio hardware */ 530 int (*dump)(const struct audio_hw_device *dev, int fd); 531 532 /** 533 * set the audio mute status for all audio activities. If any value other 534 * than 0 is returned, the software mixer will emulate this capability. 535 */ 536 int (*set_master_mute)(struct audio_hw_device *dev, bool mute); 537 538 /** 539 * Get the current master mute status for the HAL, if the HAL supports 540 * master mute control. AudioFlinger will query this value from the primary 541 * audio HAL when the service starts and use the value for setting the 542 * initial master mute across all HALs. HALs which do not support this 543 * method may leave it set to NULL. 544 */ 545 int (*get_master_mute)(struct audio_hw_device *dev, bool *mute); 546}; 547typedef struct audio_hw_device audio_hw_device_t; 548 549/** convenience API for opening and closing a supported device */ 550 551static inline int audio_hw_device_open(const struct hw_module_t* module, 552 struct audio_hw_device** device) 553{ 554 return module->methods->open(module, AUDIO_HARDWARE_INTERFACE, 555 (struct hw_device_t**)device); 556} 557 558static inline int audio_hw_device_close(struct audio_hw_device* device) 559{ 560 return device->common.close(&device->common); 561} 562 563 564__END_DECLS 565 566#endif // ANDROID_AUDIO_INTERFACE_H 567