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_callback_event_t;
258
259 typedef int (*stream_callback_t)(stream_callback_event_t event, void *param, void *cookie);
260
261 /* type of drain requested to audio_stream_out->drain(). Mutually exclusive */
262 typedef enum {
263 AUDIO_DRAIN_ALL, /* drain() returns when all data has been played */
264 AUDIO_DRAIN_EARLY_NOTIFY /* drain() returns a short time before all data
265 from the current track has been played to
266 give time for gapless track switch */
267 } audio_drain_type_t;
268
269 /**
270 * audio_stream_out is the abstraction interface for the audio output hardware.
271 *
272 * It provides information about various properties of the audio output
273 * hardware driver.
274 */
275
276 struct audio_stream_out {
277 /**
278 * Common methods of the audio stream out. This *must* be the first member of audio_stream_out
279 * as users of this structure will cast a audio_stream to audio_stream_out pointer in contexts
280 * where it's known the audio_stream references an audio_stream_out.
281 */
282 struct audio_stream common;
283
284 /**
285 * Return the audio hardware driver estimated latency in milliseconds.
286 */
287 uint32_t (*get_latency)(const struct audio_stream_out *stream);
288
289 /**
290 * Use this method in situations where audio mixing is done in the
291 * hardware. This method serves as a direct interface with hardware,
292 * allowing you to directly set the volume as apposed to via the framework.
293 * This method might produce multiple PCM outputs or hardware accelerated
294 * codecs, such as MP3 or AAC.
295 */
296 int (*set_volume)(struct audio_stream_out *stream, float left, float right);
297
298 /**
299 * Write audio buffer to driver. Returns number of bytes written, or a
300 * negative status_t. If at least one frame was written successfully prior to the error,
301 * it is suggested that the driver return that successful (short) byte count
302 * and then return an error in the subsequent call.
303 *
304 * If set_callback() has previously been called to enable non-blocking mode
305 * the write() is not allowed to block. It must write only the number of
306 * bytes that currently fit in the driver/hardware buffer and then return
307 * this byte count. If this is less than the requested write size the
308 * callback function must be called when more space is available in the
309 * driver/hardware buffer.
310 */
311 ssize_t (*write)(struct audio_stream_out *stream, const void* buffer,
312 size_t bytes);
313
314 /* return the number of audio frames written by the audio dsp to DAC since
315 * the output has exited standby
316 */
317 int (*get_render_position)(const struct audio_stream_out *stream,
318 uint32_t *dsp_frames);
319
320 /**
321 * get the local time at which the next write to the audio driver will be presented.
322 * The units are microseconds, where the epoch is decided by the local audio HAL.
323 */
324 int (*get_next_write_timestamp)(const struct audio_stream_out *stream,
325 int64_t *timestamp);
326
327 /**
328 * set the callback function for notifying completion of non-blocking
329 * write and drain.
330 * Calling this function implies that all future write() and drain()
331 * must be non-blocking and use the callback to signal completion.
332 */
333 int (*set_callback)(struct audio_stream_out *stream,
334 stream_callback_t callback, void *cookie);
335
336 /**
337 * Notifies to the audio driver to stop playback however the queued buffers are
338 * retained by the hardware. Useful for implementing pause/resume. Empty implementation
339 * if not supported however should be implemented for hardware with non-trivial
340 * latency. In the pause state audio hardware could still be using power. User may
341 * consider calling suspend after a timeout.
342 *
343 * Implementation of this function is mandatory for offloaded playback.
344 */
345 int (*pause)(struct audio_stream_out* stream);
346
347 /**
348 * Notifies to the audio driver to resume playback following a pause.
349 * Returns error if called without matching pause.
350 *
351 * Implementation of this function is mandatory for offloaded playback.
352 */
353 int (*resume)(struct audio_stream_out* stream);
354
355 /**
356 * Requests notification when data buffered by the driver/hardware has
357 * been played. If set_callback() has previously been called to enable
358 * non-blocking mode, the drain() must not block, instead it should return
359 * quickly and completion of the drain is notified through the callback.
360 * If set_callback() has not been called, the drain() must block until
361 * completion.
362 * If type==AUDIO_DRAIN_ALL, the drain completes when all previously written
363 * data has been played.
364 * If type==AUDIO_DRAIN_EARLY_NOTIFY, the drain completes shortly before all
365 * data for the current track has played to allow time for the framework
366 * to perform a gapless track switch.
367 *
368 * Drain must return immediately on stop() and flush() call
369 *
370 * Implementation of this function is mandatory for offloaded playback.
371 */
372 int (*drain)(struct audio_stream_out* stream, audio_drain_type_t type );
373
374 /**
375 * Notifies to the audio driver to flush the queued data. Stream must already
376 * be paused before calling flush().
377 *
378 * Implementation of this function is mandatory for offloaded playback.
379 */
380 int (*flush)(struct audio_stream_out* stream);
381
382 /**
383 * Return a recent count of the number of audio frames presented to an external observer.
384 * This excludes frames which have been written but are still in the pipeline.
385 * The count is not reset to zero when output enters standby.
386 * Also returns the value of CLOCK_MONOTONIC as of this presentation count.
387 * The returned count is expected to be 'recent',
388 * but does not need to be the most recent possible value.
389 * However, the associated time should correspond to whatever count is returned.
390 * Example: assume that N+M frames have been presented, where M is a 'small' number.
391 * Then it is permissible to return N instead of N+M,
392 * and the timestamp should correspond to N rather than N+M.
393 * The terms 'recent' and 'small' are not defined.
394 * They reflect the quality of the implementation.
395 *
396 * 3.0 and higher only.
397 */
398 int (*get_presentation_position)(const struct audio_stream_out *stream,
399 uint64_t *frames, struct timespec *timestamp);
400
401 };
402 typedef struct audio_stream_out audio_stream_out_t;
403
404 struct audio_stream_in {
405 /**
406 * Common methods of the audio stream in. This *must* be the first member of audio_stream_in
407 * as users of this structure will cast a audio_stream to audio_stream_in pointer in contexts
408 * where it's known the audio_stream references an audio_stream_in.
409 */
410 struct audio_stream common;
411
412 /** set the input gain for the audio driver. This method is for
413 * for future use */
414 int (*set_gain)(struct audio_stream_in *stream, float gain);
415
416 /** Read audio buffer in from audio driver. Returns number of bytes read, or a
417 * negative status_t. If at least one frame was read prior to the error,
418 * read should return that byte count and then return an error in the subsequent call.
419 */
420 ssize_t (*read)(struct audio_stream_in *stream, void* buffer,
421 size_t bytes);
422
423 /**
424 * Return the amount of input frames lost in the audio driver since the
425 * last call of this function.
426 * Audio driver is expected to reset the value to 0 and restart counting
427 * upon returning the current value by this function call.
428 * Such loss typically occurs when the user space process is blocked
429 * longer than the capacity of audio driver buffers.
430 *
431 * Unit: the number of input audio frames
432 */
433 uint32_t (*get_input_frames_lost)(struct audio_stream_in *stream);
434
435 /**
436 * Return a recent count of the number of audio frames received and
437 * the clock time associated with that frame count.
438 *
439 * frames is the total frame count received. This should be as early in
440 * the capture pipeline as possible. In general,
441 * frames should be non-negative and should not go "backwards".
442 *
443 * time is the clock MONOTONIC time when frames was measured. In general,
444 * time should be a positive quantity and should not go "backwards".
445 *
446 * The status returned is 0 on success, -ENOSYS if the device is not
447 * ready/available, or -EINVAL if the arguments are null or otherwise invalid.
448 */
449 int (*get_capture_position)(const struct audio_stream_in *stream,
450 int64_t *frames, int64_t *time);
451 };
452 typedef struct audio_stream_in audio_stream_in_t;
453
454 /**
455 * return the frame size (number of bytes per sample).
456 *
457 * Deprecated: use audio_stream_out_frame_size() or audio_stream_in_frame_size() instead.
458 */
459 __attribute__((__deprecated__))
audio_stream_frame_size(const struct audio_stream * s)460 static inline size_t audio_stream_frame_size(const struct audio_stream *s)
461 {
462 size_t chan_samp_sz;
463 audio_format_t format = s->get_format(s);
464
465 if (audio_has_proportional_frames(format)) {
466 chan_samp_sz = audio_bytes_per_sample(format);
467 return popcount(s->get_channels(s)) * chan_samp_sz;
468 }
469
470 return sizeof(int8_t);
471 }
472
473 /**
474 * return the frame size (number of bytes per sample) of an output stream.
475 */
audio_stream_out_frame_size(const struct audio_stream_out * s)476 static inline size_t audio_stream_out_frame_size(const struct audio_stream_out *s)
477 {
478 size_t chan_samp_sz;
479 audio_format_t format = s->common.get_format(&s->common);
480
481 if (audio_has_proportional_frames(format)) {
482 chan_samp_sz = audio_bytes_per_sample(format);
483 return audio_channel_count_from_out_mask(s->common.get_channels(&s->common)) * chan_samp_sz;
484 }
485
486 return sizeof(int8_t);
487 }
488
489 /**
490 * return the frame size (number of bytes per sample) of an input stream.
491 */
audio_stream_in_frame_size(const struct audio_stream_in * s)492 static inline size_t audio_stream_in_frame_size(const struct audio_stream_in *s)
493 {
494 size_t chan_samp_sz;
495 audio_format_t format = s->common.get_format(&s->common);
496
497 if (audio_has_proportional_frames(format)) {
498 chan_samp_sz = audio_bytes_per_sample(format);
499 return audio_channel_count_from_in_mask(s->common.get_channels(&s->common)) * chan_samp_sz;
500 }
501
502 return sizeof(int8_t);
503 }
504
505 /**********************************************************************/
506
507 /**
508 * Every hardware module must have a data structure named HAL_MODULE_INFO_SYM
509 * and the fields of this data structure must begin with hw_module_t
510 * followed by module specific information.
511 */
512 struct audio_module {
513 struct hw_module_t common;
514 };
515
516 struct audio_hw_device {
517 /**
518 * Common methods of the audio device. This *must* be the first member of audio_hw_device
519 * as users of this structure will cast a hw_device_t to audio_hw_device pointer in contexts
520 * where it's known the hw_device_t references an audio_hw_device.
521 */
522 struct hw_device_t common;
523
524 /**
525 * used by audio flinger to enumerate what devices are supported by
526 * each audio_hw_device implementation.
527 *
528 * Return value is a bitmask of 1 or more values of audio_devices_t
529 *
530 * NOTE: audio HAL implementations starting with
531 * AUDIO_DEVICE_API_VERSION_2_0 do not implement this function.
532 * All supported devices should be listed in audio_policy.conf
533 * file and the audio policy manager must choose the appropriate
534 * audio module based on information in this file.
535 */
536 uint32_t (*get_supported_devices)(const struct audio_hw_device *dev);
537
538 /**
539 * check to see if the audio hardware interface has been initialized.
540 * returns 0 on success, -ENODEV on failure.
541 */
542 int (*init_check)(const struct audio_hw_device *dev);
543
544 /** set the audio volume of a voice call. Range is between 0.0 and 1.0 */
545 int (*set_voice_volume)(struct audio_hw_device *dev, float volume);
546
547 /**
548 * set the audio volume for all audio activities other than voice call.
549 * Range between 0.0 and 1.0. If any value other than 0 is returned,
550 * the software mixer will emulate this capability.
551 */
552 int (*set_master_volume)(struct audio_hw_device *dev, float volume);
553
554 /**
555 * Get the current master volume value for the HAL, if the HAL supports
556 * master volume control. AudioFlinger will query this value from the
557 * primary audio HAL when the service starts and use the value for setting
558 * the initial master volume across all HALs. HALs which do not support
559 * this method may leave it set to NULL.
560 */
561 int (*get_master_volume)(struct audio_hw_device *dev, float *volume);
562
563 /**
564 * set_mode is called when the audio mode changes. AUDIO_MODE_NORMAL mode
565 * is for standard audio playback, AUDIO_MODE_RINGTONE when a ringtone is
566 * playing, and AUDIO_MODE_IN_CALL when a call is in progress.
567 */
568 int (*set_mode)(struct audio_hw_device *dev, audio_mode_t mode);
569
570 /* mic mute */
571 int (*set_mic_mute)(struct audio_hw_device *dev, bool state);
572 int (*get_mic_mute)(const struct audio_hw_device *dev, bool *state);
573
574 /* set/get global audio parameters */
575 int (*set_parameters)(struct audio_hw_device *dev, const char *kv_pairs);
576
577 /*
578 * Returns a pointer to a heap allocated string. The caller is responsible
579 * for freeing the memory for it using free().
580 */
581 char * (*get_parameters)(const struct audio_hw_device *dev,
582 const char *keys);
583
584 /* Returns audio input buffer size according to parameters passed or
585 * 0 if one of the parameters is not supported.
586 * See also get_buffer_size which is for a particular stream.
587 */
588 size_t (*get_input_buffer_size)(const struct audio_hw_device *dev,
589 const struct audio_config *config);
590
591 /** This method creates and opens the audio hardware output stream.
592 * The "address" parameter qualifies the "devices" audio device type if needed.
593 * The format format depends on the device type:
594 * - Bluetooth devices use the MAC address of the device in the form "00:11:22:AA:BB:CC"
595 * - USB devices use the ALSA card and device numbers in the form "card=X;device=Y"
596 * - Other devices may use a number or any other string.
597 */
598
599 int (*open_output_stream)(struct audio_hw_device *dev,
600 audio_io_handle_t handle,
601 audio_devices_t devices,
602 audio_output_flags_t flags,
603 struct audio_config *config,
604 struct audio_stream_out **stream_out,
605 const char *address);
606
607 void (*close_output_stream)(struct audio_hw_device *dev,
608 struct audio_stream_out* stream_out);
609
610 /** This method creates and opens the audio hardware input stream */
611 int (*open_input_stream)(struct audio_hw_device *dev,
612 audio_io_handle_t handle,
613 audio_devices_t devices,
614 struct audio_config *config,
615 struct audio_stream_in **stream_in,
616 audio_input_flags_t flags,
617 const char *address,
618 audio_source_t source);
619
620 void (*close_input_stream)(struct audio_hw_device *dev,
621 struct audio_stream_in *stream_in);
622
623 /** This method dumps the state of the audio hardware */
624 int (*dump)(const struct audio_hw_device *dev, int fd);
625
626 /**
627 * set the audio mute status for all audio activities. If any value other
628 * than 0 is returned, the software mixer will emulate this capability.
629 */
630 int (*set_master_mute)(struct audio_hw_device *dev, bool mute);
631
632 /**
633 * Get the current master mute status for the HAL, if the HAL supports
634 * master mute control. AudioFlinger will query this value from the primary
635 * audio HAL when the service starts and use the value for setting the
636 * initial master mute across all HALs. HALs which do not support this
637 * method may leave it set to NULL.
638 */
639 int (*get_master_mute)(struct audio_hw_device *dev, bool *mute);
640
641 /**
642 * Routing control
643 */
644
645 /* Creates an audio patch between several source and sink ports.
646 * The handle is allocated by the HAL and should be unique for this
647 * audio HAL module. */
648 int (*create_audio_patch)(struct audio_hw_device *dev,
649 unsigned int num_sources,
650 const struct audio_port_config *sources,
651 unsigned int num_sinks,
652 const struct audio_port_config *sinks,
653 audio_patch_handle_t *handle);
654
655 /* Release an audio patch */
656 int (*release_audio_patch)(struct audio_hw_device *dev,
657 audio_patch_handle_t handle);
658
659 /* Fills the list of supported attributes for a given audio port.
660 * As input, "port" contains the information (type, role, address etc...)
661 * needed by the HAL to identify the port.
662 * As output, "port" contains possible attributes (sampling rates, formats,
663 * channel masks, gain controllers...) for this port.
664 */
665 int (*get_audio_port)(struct audio_hw_device *dev,
666 struct audio_port *port);
667
668 /* Set audio port configuration */
669 int (*set_audio_port_config)(struct audio_hw_device *dev,
670 const struct audio_port_config *config);
671
672 };
673 typedef struct audio_hw_device audio_hw_device_t;
674
675 /** convenience API for opening and closing a supported device */
676
audio_hw_device_open(const struct hw_module_t * module,struct audio_hw_device ** device)677 static inline int audio_hw_device_open(const struct hw_module_t* module,
678 struct audio_hw_device** device)
679 {
680 return module->methods->open(module, AUDIO_HARDWARE_INTERFACE,
681 (struct hw_device_t**)device);
682 }
683
audio_hw_device_close(struct audio_hw_device * device)684 static inline int audio_hw_device_close(struct audio_hw_device* device)
685 {
686 return device->common.close(&device->common);
687 }
688
689
690 __END_DECLS
691
692 #endif // ANDROID_AUDIO_INTERFACE_H
693