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__))
audio_stream_frame_size(const struct audio_stream * s)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 */
audio_stream_out_frame_size(const struct audio_stream_out * s)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 */
audio_stream_in_frame_size(const struct audio_stream_in * s)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
audio_hw_device_open(const struct hw_module_t * module,struct audio_hw_device ** device)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
audio_hw_device_close(struct audio_hw_device * device)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