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_CBK_EVENT_ERROR, /* stream hit some error, let AF take action */
258 } stream_callback_event_t;
259
260 typedef int (*stream_callback_t)(stream_callback_event_t event, void *param, void *cookie);
261
262 /* type of drain requested to audio_stream_out->drain(). Mutually exclusive */
263 typedef enum {
264 AUDIO_DRAIN_ALL, /* drain() returns when all data has been played */
265 AUDIO_DRAIN_EARLY_NOTIFY /* drain() returns a short time before all data
266 from the current track has been played to
267 give time for gapless track switch */
268 } audio_drain_type_t;
269
270 /**
271 * audio_stream_out is the abstraction interface for the audio output hardware.
272 *
273 * It provides information about various properties of the audio output
274 * hardware driver.
275 */
276
277 struct audio_stream_out {
278 /**
279 * Common methods of the audio stream out. This *must* be the first member of audio_stream_out
280 * as users of this structure will cast a audio_stream to audio_stream_out pointer in contexts
281 * where it's known the audio_stream references an audio_stream_out.
282 */
283 struct audio_stream common;
284
285 /**
286 * Return the audio hardware driver estimated latency in milliseconds.
287 */
288 uint32_t (*get_latency)(const struct audio_stream_out *stream);
289
290 /**
291 * Use this method in situations where audio mixing is done in the
292 * hardware. This method serves as a direct interface with hardware,
293 * allowing you to directly set the volume as apposed to via the framework.
294 * This method might produce multiple PCM outputs or hardware accelerated
295 * codecs, such as MP3 or AAC.
296 */
297 int (*set_volume)(struct audio_stream_out *stream, float left, float right);
298
299 /**
300 * Write audio buffer to driver. Returns number of bytes written, or a
301 * negative status_t. If at least one frame was written successfully prior to the error,
302 * it is suggested that the driver return that successful (short) byte count
303 * and then return an error in the subsequent call.
304 *
305 * If set_callback() has previously been called to enable non-blocking mode
306 * the write() is not allowed to block. It must write only the number of
307 * bytes that currently fit in the driver/hardware buffer and then return
308 * this byte count. If this is less than the requested write size the
309 * callback function must be called when more space is available in the
310 * driver/hardware buffer.
311 */
312 ssize_t (*write)(struct audio_stream_out *stream, const void* buffer,
313 size_t bytes);
314
315 /* return the number of audio frames written by the audio dsp to DAC since
316 * the output has exited standby
317 */
318 int (*get_render_position)(const struct audio_stream_out *stream,
319 uint32_t *dsp_frames);
320
321 /**
322 * get the local time at which the next write to the audio driver will be presented.
323 * The units are microseconds, where the epoch is decided by the local audio HAL.
324 */
325 int (*get_next_write_timestamp)(const struct audio_stream_out *stream,
326 int64_t *timestamp);
327
328 /**
329 * set the callback function for notifying completion of non-blocking
330 * write and drain.
331 * Calling this function implies that all future write() and drain()
332 * must be non-blocking and use the callback to signal completion.
333 */
334 int (*set_callback)(struct audio_stream_out *stream,
335 stream_callback_t callback, void *cookie);
336
337 /**
338 * Notifies to the audio driver to stop playback however the queued buffers are
339 * retained by the hardware. Useful for implementing pause/resume. Empty implementation
340 * if not supported however should be implemented for hardware with non-trivial
341 * latency. In the pause state audio hardware could still be using power. User may
342 * consider calling suspend after a timeout.
343 *
344 * Implementation of this function is mandatory for offloaded playback.
345 */
346 int (*pause)(struct audio_stream_out* stream);
347
348 /**
349 * Notifies to the audio driver to resume playback following a pause.
350 * Returns error if called without matching pause.
351 *
352 * Implementation of this function is mandatory for offloaded playback.
353 */
354 int (*resume)(struct audio_stream_out* stream);
355
356 /**
357 * Requests notification when data buffered by the driver/hardware has
358 * been played. If set_callback() has previously been called to enable
359 * non-blocking mode, the drain() must not block, instead it should return
360 * quickly and completion of the drain is notified through the callback.
361 * If set_callback() has not been called, the drain() must block until
362 * completion.
363 * If type==AUDIO_DRAIN_ALL, the drain completes when all previously written
364 * data has been played.
365 * If type==AUDIO_DRAIN_EARLY_NOTIFY, the drain completes shortly before all
366 * data for the current track has played to allow time for the framework
367 * to perform a gapless track switch.
368 *
369 * Drain must return immediately on stop() and flush() call
370 *
371 * Implementation of this function is mandatory for offloaded playback.
372 */
373 int (*drain)(struct audio_stream_out* stream, audio_drain_type_t type );
374
375 /**
376 * Notifies to the audio driver to flush the queued data. Stream must already
377 * be paused before calling flush().
378 *
379 * Implementation of this function is mandatory for offloaded playback.
380 */
381 int (*flush)(struct audio_stream_out* stream);
382
383 /**
384 * Return a recent count of the number of audio frames presented to an external observer.
385 * This excludes frames which have been written but are still in the pipeline.
386 * The count is not reset to zero when output enters standby.
387 * Also returns the value of CLOCK_MONOTONIC as of this presentation count.
388 * The returned count is expected to be 'recent',
389 * but does not need to be the most recent possible value.
390 * However, the associated time should correspond to whatever count is returned.
391 * Example: assume that N+M frames have been presented, where M is a 'small' number.
392 * Then it is permissible to return N instead of N+M,
393 * and the timestamp should correspond to N rather than N+M.
394 * The terms 'recent' and 'small' are not defined.
395 * They reflect the quality of the implementation.
396 *
397 * 3.0 and higher only.
398 */
399 int (*get_presentation_position)(const struct audio_stream_out *stream,
400 uint64_t *frames, struct timespec *timestamp);
401
402 };
403 typedef struct audio_stream_out audio_stream_out_t;
404
405 struct audio_stream_in {
406 /**
407 * Common methods of the audio stream in. This *must* be the first member of audio_stream_in
408 * as users of this structure will cast a audio_stream to audio_stream_in pointer in contexts
409 * where it's known the audio_stream references an audio_stream_in.
410 */
411 struct audio_stream common;
412
413 /** set the input gain for the audio driver. This method is for
414 * for future use */
415 int (*set_gain)(struct audio_stream_in *stream, float gain);
416
417 /** Read audio buffer in from audio driver. Returns number of bytes read, or a
418 * negative status_t. If at least one frame was read prior to the error,
419 * read should return that byte count and then return an error in the subsequent call.
420 */
421 ssize_t (*read)(struct audio_stream_in *stream, void* buffer,
422 size_t bytes);
423
424 /**
425 * Return the amount of input frames lost in the audio driver since the
426 * last call of this function.
427 * Audio driver is expected to reset the value to 0 and restart counting
428 * upon returning the current value by this function call.
429 * Such loss typically occurs when the user space process is blocked
430 * longer than the capacity of audio driver buffers.
431 *
432 * Unit: the number of input audio frames
433 */
434 uint32_t (*get_input_frames_lost)(struct audio_stream_in *stream);
435
436 /**
437 * Return a recent count of the number of audio frames received and
438 * the clock time associated with that frame count.
439 *
440 * frames is the total frame count received. This should be as early in
441 * the capture pipeline as possible. In general,
442 * frames should be non-negative and should not go "backwards".
443 *
444 * time is the clock MONOTONIC time when frames was measured. In general,
445 * time should be a positive quantity and should not go "backwards".
446 *
447 * The status returned is 0 on success, -ENOSYS if the device is not
448 * ready/available, or -EINVAL if the arguments are null or otherwise invalid.
449 */
450 int (*get_capture_position)(const struct audio_stream_in *stream,
451 int64_t *frames, int64_t *time);
452 };
453 typedef struct audio_stream_in audio_stream_in_t;
454
455 /**
456 * return the frame size (number of bytes per sample).
457 *
458 * Deprecated: use audio_stream_out_frame_size() or audio_stream_in_frame_size() instead.
459 */
460 __attribute__((__deprecated__))
audio_stream_frame_size(const struct audio_stream * s)461 static inline size_t audio_stream_frame_size(const struct audio_stream *s)
462 {
463 size_t chan_samp_sz;
464 audio_format_t format = s->get_format(s);
465
466 if (audio_has_proportional_frames(format)) {
467 chan_samp_sz = audio_bytes_per_sample(format);
468 return popcount(s->get_channels(s)) * chan_samp_sz;
469 }
470
471 return sizeof(int8_t);
472 }
473
474 /**
475 * return the frame size (number of bytes per sample) of an output stream.
476 */
audio_stream_out_frame_size(const struct audio_stream_out * s)477 static inline size_t audio_stream_out_frame_size(const struct audio_stream_out *s)
478 {
479 size_t chan_samp_sz;
480 audio_format_t format = s->common.get_format(&s->common);
481
482 if (audio_has_proportional_frames(format)) {
483 chan_samp_sz = audio_bytes_per_sample(format);
484 return audio_channel_count_from_out_mask(s->common.get_channels(&s->common)) * chan_samp_sz;
485 }
486
487 return sizeof(int8_t);
488 }
489
490 /**
491 * return the frame size (number of bytes per sample) of an input stream.
492 */
audio_stream_in_frame_size(const struct audio_stream_in * s)493 static inline size_t audio_stream_in_frame_size(const struct audio_stream_in *s)
494 {
495 size_t chan_samp_sz;
496 audio_format_t format = s->common.get_format(&s->common);
497
498 if (audio_has_proportional_frames(format)) {
499 chan_samp_sz = audio_bytes_per_sample(format);
500 return audio_channel_count_from_in_mask(s->common.get_channels(&s->common)) * chan_samp_sz;
501 }
502
503 return sizeof(int8_t);
504 }
505
506 /**********************************************************************/
507
508 /**
509 * Every hardware module must have a data structure named HAL_MODULE_INFO_SYM
510 * and the fields of this data structure must begin with hw_module_t
511 * followed by module specific information.
512 */
513 struct audio_module {
514 struct hw_module_t common;
515 };
516
517 struct audio_hw_device {
518 /**
519 * Common methods of the audio device. This *must* be the first member of audio_hw_device
520 * as users of this structure will cast a hw_device_t to audio_hw_device pointer in contexts
521 * where it's known the hw_device_t references an audio_hw_device.
522 */
523 struct hw_device_t common;
524
525 /**
526 * used by audio flinger to enumerate what devices are supported by
527 * each audio_hw_device implementation.
528 *
529 * Return value is a bitmask of 1 or more values of audio_devices_t
530 *
531 * NOTE: audio HAL implementations starting with
532 * AUDIO_DEVICE_API_VERSION_2_0 do not implement this function.
533 * All supported devices should be listed in audio_policy.conf
534 * file and the audio policy manager must choose the appropriate
535 * audio module based on information in this file.
536 */
537 uint32_t (*get_supported_devices)(const struct audio_hw_device *dev);
538
539 /**
540 * check to see if the audio hardware interface has been initialized.
541 * returns 0 on success, -ENODEV on failure.
542 */
543 int (*init_check)(const struct audio_hw_device *dev);
544
545 /** set the audio volume of a voice call. Range is between 0.0 and 1.0 */
546 int (*set_voice_volume)(struct audio_hw_device *dev, float volume);
547
548 /**
549 * set the audio volume for all audio activities other than voice call.
550 * Range between 0.0 and 1.0. If any value other than 0 is returned,
551 * the software mixer will emulate this capability.
552 */
553 int (*set_master_volume)(struct audio_hw_device *dev, float volume);
554
555 /**
556 * Get the current master volume value for the HAL, if the HAL supports
557 * master volume control. AudioFlinger will query this value from the
558 * primary audio HAL when the service starts and use the value for setting
559 * the initial master volume across all HALs. HALs which do not support
560 * this method may leave it set to NULL.
561 */
562 int (*get_master_volume)(struct audio_hw_device *dev, float *volume);
563
564 /**
565 * set_mode is called when the audio mode changes. AUDIO_MODE_NORMAL mode
566 * is for standard audio playback, AUDIO_MODE_RINGTONE when a ringtone is
567 * playing, and AUDIO_MODE_IN_CALL when a call is in progress.
568 */
569 int (*set_mode)(struct audio_hw_device *dev, audio_mode_t mode);
570
571 /* mic mute */
572 int (*set_mic_mute)(struct audio_hw_device *dev, bool state);
573 int (*get_mic_mute)(const struct audio_hw_device *dev, bool *state);
574
575 /* set/get global audio parameters */
576 int (*set_parameters)(struct audio_hw_device *dev, const char *kv_pairs);
577
578 /*
579 * Returns a pointer to a heap allocated string. The caller is responsible
580 * for freeing the memory for it using free().
581 */
582 char * (*get_parameters)(const struct audio_hw_device *dev,
583 const char *keys);
584
585 /* Returns audio input buffer size according to parameters passed or
586 * 0 if one of the parameters is not supported.
587 * See also get_buffer_size which is for a particular stream.
588 */
589 size_t (*get_input_buffer_size)(const struct audio_hw_device *dev,
590 const struct audio_config *config);
591
592 /** This method creates and opens the audio hardware output stream.
593 * The "address" parameter qualifies the "devices" audio device type if needed.
594 * The format format depends on the device type:
595 * - Bluetooth devices use the MAC address of the device in the form "00:11:22:AA:BB:CC"
596 * - USB devices use the ALSA card and device numbers in the form "card=X;device=Y"
597 * - Other devices may use a number or any other string.
598 */
599
600 int (*open_output_stream)(struct audio_hw_device *dev,
601 audio_io_handle_t handle,
602 audio_devices_t devices,
603 audio_output_flags_t flags,
604 struct audio_config *config,
605 struct audio_stream_out **stream_out,
606 const char *address);
607
608 void (*close_output_stream)(struct audio_hw_device *dev,
609 struct audio_stream_out* stream_out);
610
611 /** This method creates and opens the audio hardware input stream */
612 int (*open_input_stream)(struct audio_hw_device *dev,
613 audio_io_handle_t handle,
614 audio_devices_t devices,
615 struct audio_config *config,
616 struct audio_stream_in **stream_in,
617 audio_input_flags_t flags,
618 const char *address,
619 audio_source_t source);
620
621 void (*close_input_stream)(struct audio_hw_device *dev,
622 struct audio_stream_in *stream_in);
623
624 /** This method dumps the state of the audio hardware */
625 int (*dump)(const struct audio_hw_device *dev, int fd);
626
627 /**
628 * set the audio mute status for all audio activities. If any value other
629 * than 0 is returned, the software mixer will emulate this capability.
630 */
631 int (*set_master_mute)(struct audio_hw_device *dev, bool mute);
632
633 /**
634 * Get the current master mute status for the HAL, if the HAL supports
635 * master mute control. AudioFlinger will query this value from the primary
636 * audio HAL when the service starts and use the value for setting the
637 * initial master mute across all HALs. HALs which do not support this
638 * method may leave it set to NULL.
639 */
640 int (*get_master_mute)(struct audio_hw_device *dev, bool *mute);
641
642 /**
643 * Routing control
644 */
645
646 /* Creates an audio patch between several source and sink ports.
647 * The handle is allocated by the HAL and should be unique for this
648 * audio HAL module. */
649 int (*create_audio_patch)(struct audio_hw_device *dev,
650 unsigned int num_sources,
651 const struct audio_port_config *sources,
652 unsigned int num_sinks,
653 const struct audio_port_config *sinks,
654 audio_patch_handle_t *handle);
655
656 /* Release an audio patch */
657 int (*release_audio_patch)(struct audio_hw_device *dev,
658 audio_patch_handle_t handle);
659
660 /* Fills the list of supported attributes for a given audio port.
661 * As input, "port" contains the information (type, role, address etc...)
662 * needed by the HAL to identify the port.
663 * As output, "port" contains possible attributes (sampling rates, formats,
664 * channel masks, gain controllers...) for this port.
665 */
666 int (*get_audio_port)(struct audio_hw_device *dev,
667 struct audio_port *port);
668
669 /* Set audio port configuration */
670 int (*set_audio_port_config)(struct audio_hw_device *dev,
671 const struct audio_port_config *config);
672
673 };
674 typedef struct audio_hw_device audio_hw_device_t;
675
676 /** convenience API for opening and closing a supported device */
677
audio_hw_device_open(const struct hw_module_t * module,struct audio_hw_device ** device)678 static inline int audio_hw_device_open(const struct hw_module_t* module,
679 struct audio_hw_device** device)
680 {
681 return module->methods->open(module, AUDIO_HARDWARE_INTERFACE,
682 (struct hw_device_t**)device);
683 }
684
audio_hw_device_close(struct audio_hw_device * device)685 static inline int audio_hw_device_close(struct audio_hw_device* device)
686 {
687 return device->common.close(&device->common);
688 }
689
690
691 __END_DECLS
692
693 #endif // ANDROID_AUDIO_INTERFACE_H
694