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