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_3_1 HARDWARE_DEVICE_API_VERSION(3, 1)
60 #define AUDIO_DEVICE_API_VERSION_3_2 HARDWARE_DEVICE_API_VERSION(3, 2)
61 #define AUDIO_DEVICE_API_VERSION_CURRENT AUDIO_DEVICE_API_VERSION_3_2
62 /* Minimal audio HAL version supported by the audio framework */
63 #define AUDIO_DEVICE_API_VERSION_MIN AUDIO_DEVICE_API_VERSION_2_0
64
65 /**************************************/
66
67 /**
68 * standard audio parameters that the HAL may need to handle
69 */
70
71 /**
72 * audio device parameters
73 */
74
75 /* TTY mode selection */
76 #define AUDIO_PARAMETER_KEY_TTY_MODE "tty_mode"
77 #define AUDIO_PARAMETER_VALUE_TTY_OFF "tty_off"
78 #define AUDIO_PARAMETER_VALUE_TTY_VCO "tty_vco"
79 #define AUDIO_PARAMETER_VALUE_TTY_HCO "tty_hco"
80 #define AUDIO_PARAMETER_VALUE_TTY_FULL "tty_full"
81
82 /* Hearing Aid Compatibility - Telecoil (HAC-T) mode on/off */
83 #define AUDIO_PARAMETER_KEY_HAC "HACSetting"
84 #define AUDIO_PARAMETER_VALUE_HAC_ON "ON"
85 #define AUDIO_PARAMETER_VALUE_HAC_OFF "OFF"
86
87 /* A2DP sink address set by framework */
88 #define AUDIO_PARAMETER_A2DP_SINK_ADDRESS "a2dp_sink_address"
89
90 /* A2DP source address set by framework */
91 #define AUDIO_PARAMETER_A2DP_SOURCE_ADDRESS "a2dp_source_address"
92
93 /* Bluetooth SCO wideband */
94 #define AUDIO_PARAMETER_KEY_BT_SCO_WB "bt_wbs"
95
96 /* BT SCO headset name for debug */
97 #define AUDIO_PARAMETER_KEY_BT_SCO_HEADSET_NAME "bt_headset_name"
98
99 /* BT SCO HFP control */
100 #define AUDIO_PARAMETER_KEY_HFP_ENABLE "hfp_enable"
101 #define AUDIO_PARAMETER_KEY_HFP_SET_SAMPLING_RATE "hfp_set_sampling_rate"
102 #define AUDIO_PARAMETER_KEY_HFP_VOLUME "hfp_volume"
103
104 /* Set screen orientation */
105 #define AUDIO_PARAMETER_KEY_ROTATION "rotation"
106
107 /**
108 * audio stream parameters
109 */
110
111 /* Enable AANC */
112 #define AUDIO_PARAMETER_KEY_AANC "aanc_enabled"
113
114 /**************************************/
115
116 /* common audio stream parameters and operations */
117 struct audio_stream {
118
119 /**
120 * Return the sampling rate in Hz - eg. 44100.
121 */
122 uint32_t (*get_sample_rate)(const struct audio_stream *stream);
123
124 /* currently unused - use set_parameters with key
125 * AUDIO_PARAMETER_STREAM_SAMPLING_RATE
126 */
127 int (*set_sample_rate)(struct audio_stream *stream, uint32_t rate);
128
129 /**
130 * Return size of input/output buffer in bytes for this stream - eg. 4800.
131 * It should be a multiple of the frame size. See also get_input_buffer_size.
132 */
133 size_t (*get_buffer_size)(const struct audio_stream *stream);
134
135 /**
136 * Return the channel mask -
137 * e.g. AUDIO_CHANNEL_OUT_STEREO or AUDIO_CHANNEL_IN_STEREO
138 */
139 audio_channel_mask_t (*get_channels)(const struct audio_stream *stream);
140
141 /**
142 * Return the audio format - e.g. AUDIO_FORMAT_PCM_16_BIT
143 */
144 audio_format_t (*get_format)(const struct audio_stream *stream);
145
146 /* currently unused - use set_parameters with key
147 * AUDIO_PARAMETER_STREAM_FORMAT
148 */
149 int (*set_format)(struct audio_stream *stream, audio_format_t format);
150
151 /**
152 * Put the audio hardware input/output into standby mode.
153 * Driver should exit from standby mode at the next I/O operation.
154 * Returns 0 on success and <0 on failure.
155 */
156 int (*standby)(struct audio_stream *stream);
157
158 /** dump the state of the audio input/output device */
159 int (*dump)(const struct audio_stream *stream, int fd);
160
161 /** Return the set of device(s) which this stream is connected to */
162 audio_devices_t (*get_device)(const struct audio_stream *stream);
163
164 /**
165 * Currently unused - set_device() corresponds to set_parameters() with key
166 * AUDIO_PARAMETER_STREAM_ROUTING for both input and output.
167 * AUDIO_PARAMETER_STREAM_INPUT_SOURCE is an additional information used by
168 * input streams only.
169 */
170 int (*set_device)(struct audio_stream *stream, audio_devices_t device);
171
172 /**
173 * set/get audio stream parameters. The function accepts a list of
174 * parameter key value pairs in the form: key1=value1;key2=value2;...
175 *
176 * Some keys are reserved for standard parameters (See AudioParameter class)
177 *
178 * If the implementation does not accept a parameter change while
179 * the output is active but the parameter is acceptable otherwise, it must
180 * return -ENOSYS.
181 *
182 * The audio flinger will put the stream in standby and then change the
183 * parameter value.
184 */
185 int (*set_parameters)(struct audio_stream *stream, const char *kv_pairs);
186
187 /*
188 * Returns a pointer to a heap allocated string. The caller is responsible
189 * for freeing the memory for it using free().
190 */
191 char * (*get_parameters)(const struct audio_stream *stream,
192 const char *keys);
193 int (*add_audio_effect)(const struct audio_stream *stream,
194 effect_handle_t effect);
195 int (*remove_audio_effect)(const struct audio_stream *stream,
196 effect_handle_t effect);
197 };
198 typedef struct audio_stream audio_stream_t;
199
200 /* type of asynchronous write callback events. Mutually exclusive */
201 typedef enum {
202 STREAM_CBK_EVENT_WRITE_READY, /* non blocking write completed */
203 STREAM_CBK_EVENT_DRAIN_READY, /* drain completed */
204 STREAM_CBK_EVENT_ERROR, /* stream hit some error, let AF take action */
205 } stream_callback_event_t;
206
207 typedef enum {
208 STREAM_EVENT_CBK_TYPE_CODEC_FORMAT_CHANGED, /* codec format of the stream changed */
209 } stream_event_callback_type_t;
210
211 typedef int (*stream_callback_t)(stream_callback_event_t event, void *param, void *cookie);
212
213 typedef int (*stream_event_callback_t)(stream_event_callback_type_t event,
214 void *param, void *cookie);
215
216 /* type of drain requested to audio_stream_out->drain(). Mutually exclusive */
217 typedef enum {
218 AUDIO_DRAIN_ALL, /* drain() returns when all data has been played */
219 AUDIO_DRAIN_EARLY_NOTIFY /* drain() returns a short time before all data
220 from the current track has been played to
221 give time for gapless track switch */
222 } audio_drain_type_t;
223
224 typedef struct source_metadata {
225 size_t track_count;
226 /** Array of metadata of each track connected to this source. */
227 struct playback_track_metadata* tracks;
228 } source_metadata_t;
229
230 typedef struct sink_metadata {
231 size_t track_count;
232 /** Array of metadata of each track connected to this sink. */
233 struct record_track_metadata* tracks;
234 } sink_metadata_t;
235
236 /* HAL version 3.2 and higher only. */
237 typedef struct source_metadata_v7 {
238 size_t track_count;
239 /** Array of metadata of each track connected to this source. */
240 struct playback_track_metadata_v7* tracks;
241 } source_metadata_v7_t;
242
243 /* HAL version 3.2 and higher only. */
244 typedef struct sink_metadata_v7 {
245 size_t track_count;
246 /** Array of metadata of each track connected to this sink. */
247 struct record_track_metadata_v7* tracks;
248 } sink_metadata_v7_t;
249
250 /** output stream callback method to indicate changes in supported latency modes */
251 typedef void (*stream_latency_mode_callback_t)(
252 audio_latency_mode_t *modes, size_t num_modes, void *cookie);
253
254 /**
255 * audio_stream_out is the abstraction interface for the audio output hardware.
256 *
257 * It provides information about various properties of the audio output
258 * hardware driver.
259 */
260 struct audio_stream_out {
261 /**
262 * Common methods of the audio stream out. This *must* be the first member of audio_stream_out
263 * as users of this structure will cast a audio_stream to audio_stream_out pointer in contexts
264 * where it's known the audio_stream references an audio_stream_out.
265 */
266 struct audio_stream common;
267
268 /**
269 * Return the audio hardware driver estimated latency in milliseconds.
270 */
271 uint32_t (*get_latency)(const struct audio_stream_out *stream);
272
273 /**
274 * Use this method in situations where audio mixing is done in the
275 * hardware. This method serves as a direct interface with hardware,
276 * allowing you to directly set the volume as apposed to via the framework.
277 * This method might produce multiple PCM outputs or hardware accelerated
278 * codecs, such as MP3 or AAC.
279 */
280 int (*set_volume)(struct audio_stream_out *stream, float left, float right);
281
282 /**
283 * Write audio buffer to driver. Returns number of bytes written, or a
284 * negative status_t. If at least one frame was written successfully prior to the error,
285 * it is suggested that the driver return that successful (short) byte count
286 * and then return an error in the subsequent call.
287 *
288 * If set_callback() has previously been called to enable non-blocking mode
289 * the write() is not allowed to block. It must write only the number of
290 * bytes that currently fit in the driver/hardware buffer and then return
291 * this byte count. If this is less than the requested write size the
292 * callback function must be called when more space is available in the
293 * driver/hardware buffer.
294 */
295 ssize_t (*write)(struct audio_stream_out *stream, const void* buffer,
296 size_t bytes);
297
298 /* return the number of audio frames written by the audio dsp to DAC since
299 * the output has exited standby
300 */
301 int (*get_render_position)(const struct audio_stream_out *stream,
302 uint32_t *dsp_frames);
303
304 /**
305 * get the local time at which the next write to the audio driver will be presented.
306 * The units are microseconds, where the epoch is decided by the local audio HAL.
307 */
308 int (*get_next_write_timestamp)(const struct audio_stream_out *stream,
309 int64_t *timestamp);
310
311 /**
312 * set the callback function for notifying completion of non-blocking
313 * write and drain.
314 * Calling this function implies that all future write() and drain()
315 * must be non-blocking and use the callback to signal completion.
316 */
317 int (*set_callback)(struct audio_stream_out *stream,
318 stream_callback_t callback, void *cookie);
319
320 /**
321 * Notifies to the audio driver to stop playback however the queued buffers are
322 * retained by the hardware. Useful for implementing pause/resume. Empty implementation
323 * if not supported however should be implemented for hardware with non-trivial
324 * latency. In the pause state audio hardware could still be using power. User may
325 * consider calling suspend after a timeout.
326 *
327 * Implementation of this function is mandatory for offloaded playback.
328 */
329 int (*pause)(struct audio_stream_out* stream);
330
331 /**
332 * Notifies to the audio driver to resume playback following a pause.
333 * Returns error if called without matching pause.
334 *
335 * Implementation of this function is mandatory for offloaded playback.
336 */
337 int (*resume)(struct audio_stream_out* stream);
338
339 /**
340 * Requests notification when data buffered by the driver/hardware has
341 * been played. If set_callback() has previously been called to enable
342 * non-blocking mode, the drain() must not block, instead it should return
343 * quickly and completion of the drain is notified through the callback.
344 * If set_callback() has not been called, the drain() must block until
345 * completion.
346 * If type==AUDIO_DRAIN_ALL, the drain completes when all previously written
347 * data has been played.
348 * If type==AUDIO_DRAIN_EARLY_NOTIFY, the drain completes shortly before all
349 * data for the current track has played to allow time for the framework
350 * to perform a gapless track switch.
351 *
352 * Drain must return immediately on stop() and flush() call
353 *
354 * Implementation of this function is mandatory for offloaded playback.
355 */
356 int (*drain)(struct audio_stream_out* stream, audio_drain_type_t type );
357
358 /**
359 * Notifies to the audio driver to flush the queued data. Stream must already
360 * be paused before calling flush().
361 *
362 * Implementation of this function is mandatory for offloaded playback.
363 */
364 int (*flush)(struct audio_stream_out* stream);
365
366 /**
367 * Return a recent count of the number of audio frames presented to an external observer.
368 * This excludes frames which have been written but are still in the pipeline.
369 * The count is not reset to zero when output enters standby.
370 * Also returns the value of CLOCK_MONOTONIC as of this presentation count.
371 * The returned count is expected to be 'recent',
372 * but does not need to be the most recent possible value.
373 * However, the associated time should correspond to whatever count is returned.
374 * Example: assume that N+M frames have been presented, where M is a 'small' number.
375 * Then it is permissible to return N instead of N+M,
376 * and the timestamp should correspond to N rather than N+M.
377 * The terms 'recent' and 'small' are not defined.
378 * They reflect the quality of the implementation.
379 *
380 * 3.0 and higher only.
381 */
382 int (*get_presentation_position)(const struct audio_stream_out *stream,
383 uint64_t *frames, struct timespec *timestamp);
384
385 /**
386 * Called by the framework to start a stream operating in mmap mode.
387 * create_mmap_buffer must be called before calling start()
388 *
389 * \note Function only implemented by streams operating in mmap mode.
390 *
391 * \param[in] stream the stream object.
392 * \return 0 in case of success.
393 * -ENOSYS if called out of sequence or on non mmap stream
394 */
395 int (*start)(const struct audio_stream_out* stream);
396
397 /**
398 * Called by the framework to stop a stream operating in mmap mode.
399 * Must be called after start()
400 *
401 * \note Function only implemented by streams operating in mmap mode.
402 *
403 * \param[in] stream the stream object.
404 * \return 0 in case of success.
405 * -ENOSYS if called out of sequence or on non mmap stream
406 */
407 int (*stop)(const struct audio_stream_out* stream);
408
409 /**
410 * Called by the framework to retrieve information on the mmap buffer used for audio
411 * samples transfer.
412 *
413 * \note Function only implemented by streams operating in mmap mode.
414 *
415 * \param[in] stream the stream object.
416 * \param[in] min_size_frames minimum buffer size requested. The actual buffer
417 * size returned in struct audio_mmap_buffer_info can be larger.
418 * \param[out] info address at which the mmap buffer information should be returned.
419 *
420 * \return 0 if the buffer was allocated.
421 * -ENODEV in case of initialization error
422 * -EINVAL if the requested buffer size is too large
423 * -ENOSYS if called out of sequence (e.g. buffer already allocated)
424 */
425 int (*create_mmap_buffer)(const struct audio_stream_out *stream,
426 int32_t min_size_frames,
427 struct audio_mmap_buffer_info *info);
428
429 /**
430 * Called by the framework to read current read/write position in the mmap buffer
431 * with associated time stamp.
432 *
433 * \note Function only implemented by streams operating in mmap mode.
434 *
435 * \param[in] stream the stream object.
436 * \param[out] position address at which the mmap read/write position should be returned.
437 *
438 * \return 0 if the position is successfully returned.
439 * -ENODATA if the position cannot be retrieved
440 * -ENOSYS if called before create_mmap_buffer()
441 */
442 int (*get_mmap_position)(const struct audio_stream_out *stream,
443 struct audio_mmap_position *position);
444
445 /**
446 * Called when the metadata of the stream's source has been changed.
447 * @param source_metadata Description of the audio that is played by the clients.
448 */
449 void (*update_source_metadata)(struct audio_stream_out *stream,
450 const struct source_metadata* source_metadata);
451
452 /**
453 * Set the callback function for notifying events for an output stream.
454 */
455 int (*set_event_callback)(struct audio_stream_out *stream,
456 stream_event_callback_t callback,
457 void *cookie);
458
459 /**
460 * Called when the metadata of the stream's source has been changed.
461 * HAL version 3.2 and higher only.
462 * @param source_metadata Description of the audio that is played by the clients.
463 */
464 void (*update_source_metadata_v7)(struct audio_stream_out *stream,
465 const struct source_metadata_v7* source_metadata);
466
467 /**
468 * Returns the Dual Mono mode presentation setting.
469 *
470 * \param[in] stream the stream object.
471 * \param[out] mode current setting of Dual Mono mode.
472 *
473 * \return 0 if the position is successfully returned.
474 * -EINVAL if the arguments are invalid
475 * -ENOSYS if the function is not available
476 */
477 int (*get_dual_mono_mode)(struct audio_stream_out *stream, audio_dual_mono_mode_t *mode);
478
479 /**
480 * Sets the Dual Mono mode presentation on the output device.
481 *
482 * \param[in] stream the stream object.
483 * \param[in] mode selected Dual Mono mode.
484 *
485 * \return 0 in case of success.
486 * -EINVAL if the arguments are invalid
487 * -ENOSYS if the function is not available
488 */
489 int (*set_dual_mono_mode)(struct audio_stream_out *stream, const audio_dual_mono_mode_t mode);
490
491 /**
492 * Returns the Audio Description Mix level in dB.
493 *
494 * \param[in] stream the stream object.
495 * \param[out] leveldB the current Audio Description Mix Level in dB.
496 *
497 * \return 0 in case of success.
498 * -EINVAL if the arguments are invalid
499 * -ENOSYS if the function is not available
500 */
501 int (*get_audio_description_mix_level)(struct audio_stream_out *stream, float *leveldB);
502
503 /**
504 * Sets the Audio Description Mix level in dB.
505 *
506 * \param[in] stream the stream object.
507 * \param[in] leveldB Audio Description Mix Level in dB.
508 *
509 * \return 0 in case of success.
510 * -EINVAL if the arguments are invalid
511 * -ENOSYS if the function is not available
512 */
513 int (*set_audio_description_mix_level)(struct audio_stream_out *stream, const float leveldB);
514
515 /**
516 * Retrieves current playback rate parameters.
517 *
518 * \param[in] stream the stream object.
519 * \param[out] playbackRate current playback parameters.
520 *
521 * \return 0 in case of success.
522 * -EINVAL if the arguments are invalid
523 * -ENOSYS if the function is not available
524 */
525 int (*get_playback_rate_parameters)(struct audio_stream_out *stream,
526 audio_playback_rate_t *playbackRate);
527
528 /**
529 * Sets the playback rate parameters that control playback behavior.
530 *
531 * \param[in] stream the stream object.
532 * \param[in] playbackRate playback parameters.
533 *
534 * \return 0 in case of success.
535 * -EINVAL if the arguments are invalid
536 * -ENOSYS if the function is not available
537 */
538 int (*set_playback_rate_parameters)(struct audio_stream_out *stream,
539 const audio_playback_rate_t *playbackRate);
540
541 /**
542 * Indicates the requested latency mode for this output stream.
543 *
544 * The requested mode can be one of the modes returned by
545 * get_recommended_latency_modes().
546 *
547 * Support for this method is optional but mandated on specific spatial audio
548 * streams indicated by AUDIO_OUTPUT_FLAG_SPATIALIZER flag if they can be routed
549 * to a BT classic sink.
550 *
551 * \param[in] stream the stream object.
552 * \param[in] mode the requested latency mode.
553 * \return 0 in case of success.
554 * -EINVAL if the arguments are invalid
555 * -ENOSYS if the function is not available
556 */
557 int (*set_latency_mode)(struct audio_stream_out *stream, audio_latency_mode_t mode);
558
559 /**
560 * Indicates which latency modes are currently supported on this output stream.
561 * If the transport protocol (e.g Bluetooth A2DP) used by this output stream to reach
562 * the output device supports variable latency modes, the HAL indicates which
563 * modes are currently supported.
564 * The framework can then call setLatencyMode() with one of the supported modes to select
565 * the desired operation mode.
566 *
567 * Support for this method is optional but mandated on specific spatial audio
568 * streams indicated by AUDIO_OUTPUT_FLAG_SPATIALIZER flag if they can be routed
569 * to a BT classic sink.
570 *
571 * \return 0 in case of success.
572 * -EINVAL if the arguments are invalid
573 * -ENOSYS if the function is not available
574 * \param[in] stream the stream object.
575 * \param[out] modes the supported latency modes.
576 * \param[in/out] num_modes as input the maximum number of modes to return,
577 * as output the actual number of modes returned.
578 */
579 int (*get_recommended_latency_modes)(struct audio_stream_out *stream,
580 audio_latency_mode_t *modes, size_t *num_modes);
581
582 /**
583 * Set the callback interface for notifying changes in supported latency modes.
584 *
585 * Calling this method with a null pointer will result in clearing a previously set callback.
586 *
587 * Support for this method is optional but mandated on specific spatial audio
588 * streams indicated by AUDIO_OUTPUT_FLAG_SPATIALIZER flag if they can be routed
589 * to a BT classic sink.
590 *
591 * \param[in] stream the stream object.
592 * \param[in] callback the registered callback or null to unregister.
593 * \param[in] cookie the context to pass when calling the callback.
594 * \return 0 in case of success.
595 * -EINVAL if the arguments are invalid
596 * -ENOSYS if the function is not available
597 */
598 int (*set_latency_mode_callback)(struct audio_stream_out *stream,
599 stream_latency_mode_callback_t callback, void *cookie);
600 };
601
602 typedef struct audio_stream_out audio_stream_out_t;
603
604 struct audio_stream_in {
605 /**
606 * Common methods of the audio stream in. This *must* be the first member of audio_stream_in
607 * as users of this structure will cast a audio_stream to audio_stream_in pointer in contexts
608 * where it's known the audio_stream references an audio_stream_in.
609 */
610 struct audio_stream common;
611
612 /** set the input gain for the audio driver. This method is for
613 * for future use */
614 int (*set_gain)(struct audio_stream_in *stream, float gain);
615
616 /** Read audio buffer in from audio driver. Returns number of bytes read, or a
617 * negative status_t. If at least one frame was read prior to the error,
618 * read should return that byte count and then return an error in the subsequent call.
619 */
620 ssize_t (*read)(struct audio_stream_in *stream, void* buffer,
621 size_t bytes);
622
623 /**
624 * Return the amount of input frames lost in the audio driver since the
625 * last call of this function.
626 * Audio driver is expected to reset the value to 0 and restart counting
627 * upon returning the current value by this function call.
628 * Such loss typically occurs when the user space process is blocked
629 * longer than the capacity of audio driver buffers.
630 *
631 * Unit: the number of input audio frames
632 */
633 uint32_t (*get_input_frames_lost)(struct audio_stream_in *stream);
634
635 /**
636 * Return a recent count of the number of audio frames received and
637 * the clock time associated with that frame count.
638 *
639 * frames is the total frame count received. This should be as early in
640 * the capture pipeline as possible. In general,
641 * frames should be non-negative and should not go "backwards".
642 *
643 * time is the clock MONOTONIC time when frames was measured. In general,
644 * time should be a positive quantity and should not go "backwards".
645 *
646 * The status returned is 0 on success, -ENOSYS if the device is not
647 * ready/available, or -EINVAL if the arguments are null or otherwise invalid.
648 */
649 int (*get_capture_position)(const struct audio_stream_in *stream,
650 int64_t *frames, int64_t *time);
651
652 /**
653 * Called by the framework to start a stream operating in mmap mode.
654 * create_mmap_buffer must be called before calling start()
655 *
656 * \note Function only implemented by streams operating in mmap mode.
657 *
658 * \param[in] stream the stream object.
659 * \return 0 in case off success.
660 * -ENOSYS if called out of sequence or on non mmap stream
661 */
662 int (*start)(const struct audio_stream_in* stream);
663
664 /**
665 * Called by the framework to stop a stream operating in mmap mode.
666 *
667 * \note Function only implemented by streams operating in mmap mode.
668 *
669 * \param[in] stream the stream object.
670 * \return 0 in case of success.
671 * -ENOSYS if called out of sequence or on non mmap stream
672 */
673 int (*stop)(const struct audio_stream_in* stream);
674
675 /**
676 * Called by the framework to retrieve information on the mmap buffer used for audio
677 * samples transfer.
678 *
679 * \note Function only implemented by streams operating in mmap mode.
680 *
681 * \param[in] stream the stream object.
682 * \param[in] min_size_frames minimum buffer size requested. The actual buffer
683 * size returned in struct audio_mmap_buffer_info can be larger.
684 * \param[out] info address at which the mmap buffer information should be returned.
685 *
686 * \return 0 if the buffer was allocated.
687 * -ENODEV in case of initialization error
688 * -EINVAL if the requested buffer size is too large
689 * -ENOSYS if called out of sequence (e.g. buffer already allocated)
690 */
691 int (*create_mmap_buffer)(const struct audio_stream_in *stream,
692 int32_t min_size_frames,
693 struct audio_mmap_buffer_info *info);
694
695 /**
696 * Called by the framework to read current read/write position in the mmap buffer
697 * with associated time stamp.
698 *
699 * \note Function only implemented by streams operating in mmap mode.
700 *
701 * \param[in] stream the stream object.
702 * \param[out] position address at which the mmap read/write position should be returned.
703 *
704 * \return 0 if the position is successfully returned.
705 * -ENODATA if the position cannot be retreived
706 * -ENOSYS if called before mmap_read_position()
707 */
708 int (*get_mmap_position)(const struct audio_stream_in *stream,
709 struct audio_mmap_position *position);
710
711 /**
712 * Called by the framework to read active microphones
713 *
714 * \param[in] stream the stream object.
715 * \param[out] mic_array Pointer to first element on array with microphone info
716 * \param[out] mic_count When called, this holds the value of the max number of elements
717 * allowed in the mic_array. The actual number of elements written
718 * is returned here.
719 * if mic_count is passed as zero, mic_array will not be populated,
720 * and mic_count will return the actual number of active microphones.
721 *
722 * \return 0 if the microphone array is successfully filled.
723 * -ENOSYS if there is an error filling the data
724 */
725 int (*get_active_microphones)(const struct audio_stream_in *stream,
726 struct audio_microphone_characteristic_t *mic_array,
727 size_t *mic_count);
728
729 /**
730 * Called by the framework to instruct the HAL to optimize the capture stream in the
731 * specified direction.
732 *
733 * \param[in] stream the stream object.
734 * \param[in] direction The direction constant (from audio-base.h)
735 * MIC_DIRECTION_UNSPECIFIED Don't do any directionality processing of the
736 * activated microphone(s).
737 * MIC_DIRECTION_FRONT Optimize capture for audio coming from the screen-side
738 * of the device.
739 * MIC_DIRECTION_BACK Optimize capture for audio coming from the side of the
740 * device opposite the screen.
741 * MIC_DIRECTION_EXTERNAL Optimize capture for audio coming from an off-device
742 * microphone.
743 * \return OK if the call is successful, an error code otherwise.
744 */
745 int (*set_microphone_direction)(const struct audio_stream_in *stream,
746 audio_microphone_direction_t direction);
747
748 /**
749 * Called by the framework to specify to the HAL the desired zoom factor for the selected
750 * microphone(s).
751 *
752 * \param[in] stream the stream object.
753 * \param[in] zoom the zoom factor.
754 * \return OK if the call is successful, an error code otherwise.
755 */
756 int (*set_microphone_field_dimension)(const struct audio_stream_in *stream,
757 float zoom);
758
759 /**
760 * Called when the metadata of the stream's sink has been changed.
761 * @param sink_metadata Description of the audio that is recorded by the clients.
762 */
763 void (*update_sink_metadata)(struct audio_stream_in *stream,
764 const struct sink_metadata* sink_metadata);
765
766 /**
767 * Called when the metadata of the stream's sink has been changed.
768 * HAL version 3.2 and higher only.
769 * @param sink_metadata Description of the audio that is recorded by the clients.
770 */
771 void (*update_sink_metadata_v7)(struct audio_stream_in *stream,
772 const struct sink_metadata_v7* sink_metadata);
773 };
774 typedef struct audio_stream_in audio_stream_in_t;
775
776 /**
777 * return the frame size (number of bytes per sample).
778 *
779 * Deprecated: use audio_stream_out_frame_size() or audio_stream_in_frame_size() instead.
780 */
781 __attribute__((__deprecated__))
audio_stream_frame_size(const struct audio_stream * s)782 static inline size_t audio_stream_frame_size(const struct audio_stream *s)
783 {
784 size_t chan_samp_sz;
785 audio_format_t format = s->get_format(s);
786
787 if (audio_has_proportional_frames(format)) {
788 chan_samp_sz = audio_bytes_per_sample(format);
789 return popcount(s->get_channels(s)) * chan_samp_sz;
790 }
791
792 return sizeof(int8_t);
793 }
794
795 /**
796 * return the frame size (number of bytes per sample) of an output stream.
797 */
audio_stream_out_frame_size(const struct audio_stream_out * s)798 static inline size_t audio_stream_out_frame_size(const struct audio_stream_out *s)
799 {
800 size_t chan_samp_sz;
801 audio_format_t format = s->common.get_format(&s->common);
802
803 if (audio_has_proportional_frames(format)) {
804 chan_samp_sz = audio_bytes_per_sample(format);
805 return audio_channel_count_from_out_mask(s->common.get_channels(&s->common)) * chan_samp_sz;
806 }
807
808 return sizeof(int8_t);
809 }
810
811 /**
812 * return the frame size (number of bytes per sample) of an input stream.
813 */
audio_stream_in_frame_size(const struct audio_stream_in * s)814 static inline size_t audio_stream_in_frame_size(const struct audio_stream_in *s)
815 {
816 size_t chan_samp_sz;
817 audio_format_t format = s->common.get_format(&s->common);
818
819 if (audio_has_proportional_frames(format)) {
820 chan_samp_sz = audio_bytes_per_sample(format);
821 return audio_channel_count_from_in_mask(s->common.get_channels(&s->common)) * chan_samp_sz;
822 }
823
824 return sizeof(int8_t);
825 }
826
827 /**********************************************************************/
828
829 /**
830 * Every hardware module must have a data structure named HAL_MODULE_INFO_SYM
831 * and the fields of this data structure must begin with hw_module_t
832 * followed by module specific information.
833 */
834 struct audio_module {
835 struct hw_module_t common;
836 };
837
838 struct audio_hw_device {
839 /**
840 * Common methods of the audio device. This *must* be the first member of audio_hw_device
841 * as users of this structure will cast a hw_device_t to audio_hw_device pointer in contexts
842 * where it's known the hw_device_t references an audio_hw_device.
843 */
844 struct hw_device_t common;
845
846 /**
847 * used by audio flinger to enumerate what devices are supported by
848 * each audio_hw_device implementation.
849 *
850 * Return value is a bitmask of 1 or more values of audio_devices_t
851 *
852 * NOTE: audio HAL implementations starting with
853 * AUDIO_DEVICE_API_VERSION_2_0 do not implement this function.
854 * All supported devices should be listed in audio_policy.conf
855 * file and the audio policy manager must choose the appropriate
856 * audio module based on information in this file.
857 */
858 uint32_t (*get_supported_devices)(const struct audio_hw_device *dev);
859
860 /**
861 * check to see if the audio hardware interface has been initialized.
862 * returns 0 on success, -ENODEV on failure.
863 */
864 int (*init_check)(const struct audio_hw_device *dev);
865
866 /** set the audio volume of a voice call. Range is between 0.0 and 1.0 */
867 int (*set_voice_volume)(struct audio_hw_device *dev, float volume);
868
869 /**
870 * set the audio volume for all audio activities other than voice call.
871 * Range between 0.0 and 1.0. If any value other than 0 is returned,
872 * the software mixer will emulate this capability.
873 */
874 int (*set_master_volume)(struct audio_hw_device *dev, float volume);
875
876 /**
877 * Get the current master volume value for the HAL, if the HAL supports
878 * master volume control. AudioFlinger will query this value from the
879 * primary audio HAL when the service starts and use the value for setting
880 * the initial master volume across all HALs. HALs which do not support
881 * this method may leave it set to NULL.
882 */
883 int (*get_master_volume)(struct audio_hw_device *dev, float *volume);
884
885 /**
886 * set_mode is called when the audio mode changes. AUDIO_MODE_NORMAL mode
887 * is for standard audio playback, AUDIO_MODE_RINGTONE when a ringtone is
888 * playing, and AUDIO_MODE_IN_CALL when a call is in progress.
889 */
890 int (*set_mode)(struct audio_hw_device *dev, audio_mode_t mode);
891
892 /* mic mute */
893 int (*set_mic_mute)(struct audio_hw_device *dev, bool state);
894 int (*get_mic_mute)(const struct audio_hw_device *dev, bool *state);
895
896 /* set/get global audio parameters */
897 int (*set_parameters)(struct audio_hw_device *dev, const char *kv_pairs);
898
899 /*
900 * Returns a pointer to a heap allocated string. The caller is responsible
901 * for freeing the memory for it using free().
902 */
903 char * (*get_parameters)(const struct audio_hw_device *dev,
904 const char *keys);
905
906 /* Returns audio input buffer size according to parameters passed or
907 * 0 if one of the parameters is not supported.
908 * See also get_buffer_size which is for a particular stream.
909 */
910 size_t (*get_input_buffer_size)(const struct audio_hw_device *dev,
911 const struct audio_config *config);
912
913 /** This method creates and opens the audio hardware output stream.
914 * The "address" parameter qualifies the "devices" audio device type if needed.
915 * The format format depends on the device type:
916 * - Bluetooth devices use the MAC address of the device in the form "00:11:22:AA:BB:CC"
917 * - USB devices use the ALSA card and device numbers in the form "card=X;device=Y"
918 * - Other devices may use a number or any other string.
919 */
920
921 int (*open_output_stream)(struct audio_hw_device *dev,
922 audio_io_handle_t handle,
923 audio_devices_t devices,
924 audio_output_flags_t flags,
925 struct audio_config *config,
926 struct audio_stream_out **stream_out,
927 const char *address);
928
929 void (*close_output_stream)(struct audio_hw_device *dev,
930 struct audio_stream_out* stream_out);
931
932 /** This method creates and opens the audio hardware input stream */
933 int (*open_input_stream)(struct audio_hw_device *dev,
934 audio_io_handle_t handle,
935 audio_devices_t devices,
936 struct audio_config *config,
937 struct audio_stream_in **stream_in,
938 audio_input_flags_t flags,
939 const char *address,
940 audio_source_t source);
941
942 void (*close_input_stream)(struct audio_hw_device *dev,
943 struct audio_stream_in *stream_in);
944
945 /**
946 * Called by the framework to read available microphones characteristics.
947 *
948 * \param[in] dev the hw_device object.
949 * \param[out] mic_array Pointer to first element on array with microphone info
950 * \param[out] mic_count When called, this holds the value of the max number of elements
951 * allowed in the mic_array. The actual number of elements written
952 * is returned here.
953 * if mic_count is passed as zero, mic_array will not be populated,
954 * and mic_count will return the actual number of microphones in the
955 * system.
956 *
957 * \return 0 if the microphone array is successfully filled.
958 * -ENOSYS if there is an error filling the data
959 */
960 int (*get_microphones)(const struct audio_hw_device *dev,
961 struct audio_microphone_characteristic_t *mic_array,
962 size_t *mic_count);
963
964 /** This method dumps the state of the audio hardware */
965 int (*dump)(const struct audio_hw_device *dev, int fd);
966
967 /**
968 * set the audio mute status for all audio activities. If any value other
969 * than 0 is returned, the software mixer will emulate this capability.
970 */
971 int (*set_master_mute)(struct audio_hw_device *dev, bool mute);
972
973 /**
974 * Get the current master mute status for the HAL, if the HAL supports
975 * master mute control. AudioFlinger will query this value from the primary
976 * audio HAL when the service starts and use the value for setting the
977 * initial master mute across all HALs. HALs which do not support this
978 * method may leave it set to NULL.
979 */
980 int (*get_master_mute)(struct audio_hw_device *dev, bool *mute);
981
982 /**
983 * Routing control
984 */
985
986 /* Creates an audio patch between several source and sink ports.
987 * The handle is allocated by the HAL and should be unique for this
988 * audio HAL module. */
989 int (*create_audio_patch)(struct audio_hw_device *dev,
990 unsigned int num_sources,
991 const struct audio_port_config *sources,
992 unsigned int num_sinks,
993 const struct audio_port_config *sinks,
994 audio_patch_handle_t *handle);
995
996 /* Release an audio patch */
997 int (*release_audio_patch)(struct audio_hw_device *dev,
998 audio_patch_handle_t handle);
999
1000 /* Fills the list of supported attributes for a given audio port.
1001 * As input, "port" contains the information (type, role, address etc...)
1002 * needed by the HAL to identify the port.
1003 * As output, "port" contains possible attributes (sampling rates, formats,
1004 * channel masks, gain controllers...) for this port.
1005 */
1006 int (*get_audio_port)(struct audio_hw_device *dev,
1007 struct audio_port *port);
1008
1009 /* Set audio port configuration */
1010 int (*set_audio_port_config)(struct audio_hw_device *dev,
1011 const struct audio_port_config *config);
1012
1013 /**
1014 * Applies an audio effect to an audio device.
1015 *
1016 * @param dev the audio HAL device context.
1017 * @param device identifies the sink or source device the effect must be applied to.
1018 * "device" is the audio_port_handle_t indicated for the device when
1019 * the audio patch connecting that device was created.
1020 * @param effect effect interface handle corresponding to the effect being added.
1021 * @return retval operation completion status.
1022 */
1023 int (*add_device_effect)(struct audio_hw_device *dev,
1024 audio_port_handle_t device, effect_handle_t effect);
1025
1026 /**
1027 * Stops applying an audio effect to an audio device.
1028 *
1029 * @param dev the audio HAL device context.
1030 * @param device identifies the sink or source device this effect was applied to.
1031 * "device" is the audio_port_handle_t indicated for the device when
1032 * the audio patch is created.
1033 * @param effect effect interface handle corresponding to the effect being removed.
1034 * @return retval operation completion status.
1035 */
1036 int (*remove_device_effect)(struct audio_hw_device *dev,
1037 audio_port_handle_t device, effect_handle_t effect);
1038
1039 /**
1040 * Fills the list of supported attributes for a given audio port.
1041 * As input, "port" contains the information (type, role, address etc...)
1042 * needed by the HAL to identify the port.
1043 * As output, "port" contains possible attributes (sampling rates, formats,
1044 * channel masks, gain controllers...) for this port. The possible attributes
1045 * are saved as audio profiles, which contains audio format and the supported
1046 * sampling rates and channel masks.
1047 */
1048 int (*get_audio_port_v7)(struct audio_hw_device *dev,
1049 struct audio_port_v7 *port);
1050
1051 /**
1052 * Called when the state of the connection of an external device has been changed.
1053 * The "port" parameter is only used as input and besides identifying the device
1054 * port, also may contain additional information such as extra audio descriptors.
1055 *
1056 * HAL version 3.2 and higher only. If the HAL does not implement this method,
1057 * it must leave the function entry as null, or return -ENOSYS. In this case
1058 * the framework will use 'set_parameters', which can only pass the device address.
1059 *
1060 * @param dev the audio HAL device context.
1061 * @param port device port identification and extra information.
1062 * @param connected whether the external device is connected.
1063 * @return retval operation completion status.
1064 */
1065 int (*set_device_connected_state_v7)(struct audio_hw_device *dev,
1066 struct audio_port_v7 *port,
1067 bool connected);
1068 };
1069 typedef struct audio_hw_device audio_hw_device_t;
1070
1071 /** convenience API for opening and closing a supported device */
1072
audio_hw_device_open(const struct hw_module_t * module,struct audio_hw_device ** device)1073 static inline int audio_hw_device_open(const struct hw_module_t* module,
1074 struct audio_hw_device** device)
1075 {
1076 return module->methods->open(module, AUDIO_HARDWARE_INTERFACE,
1077 TO_HW_DEVICE_T_OPEN(device));
1078 }
1079
audio_hw_device_close(struct audio_hw_device * device)1080 static inline int audio_hw_device_close(struct audio_hw_device* device)
1081 {
1082 return device->common.close(&device->common);
1083 }
1084
1085
1086 __END_DECLS
1087
1088 #endif // ANDROID_AUDIO_INTERFACE_H
1089