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_CURRENT AUDIO_DEVICE_API_VERSION_2_0
58
59 /**
60 * List of known audio HAL modules. This is the base name of the audio HAL
61 * library composed of the "audio." prefix, one of the base names below and
62 * a suffix specific to the device.
63 * e.g: audio.primary.goldfish.so or audio.a2dp.default.so
64 */
65
66 #define AUDIO_HARDWARE_MODULE_ID_PRIMARY "primary"
67 #define AUDIO_HARDWARE_MODULE_ID_A2DP "a2dp"
68 #define AUDIO_HARDWARE_MODULE_ID_USB "usb"
69 #define AUDIO_HARDWARE_MODULE_ID_REMOTE_SUBMIX "r_submix"
70 #define AUDIO_HARDWARE_MODULE_ID_CODEC_OFFLOAD "codec_offload"
71
72 /**************************************/
73
74 /**
75 * standard audio parameters that the HAL may need to handle
76 */
77
78 /**
79 * audio device parameters
80 */
81
82 /* BT SCO Noise Reduction + Echo Cancellation parameters */
83 #define AUDIO_PARAMETER_KEY_BT_NREC "bt_headset_nrec"
84 #define AUDIO_PARAMETER_VALUE_ON "on"
85 #define AUDIO_PARAMETER_VALUE_OFF "off"
86
87 /* TTY mode selection */
88 #define AUDIO_PARAMETER_KEY_TTY_MODE "tty_mode"
89 #define AUDIO_PARAMETER_VALUE_TTY_OFF "tty_off"
90 #define AUDIO_PARAMETER_VALUE_TTY_VCO "tty_vco"
91 #define AUDIO_PARAMETER_VALUE_TTY_HCO "tty_hco"
92 #define AUDIO_PARAMETER_VALUE_TTY_FULL "tty_full"
93
94 /* A2DP sink address set by framework */
95 #define AUDIO_PARAMETER_A2DP_SINK_ADDRESS "a2dp_sink_address"
96
97 /* Screen state */
98 #define AUDIO_PARAMETER_KEY_SCREEN_STATE "screen_state"
99
100 /**
101 * audio stream parameters
102 */
103
104 #define AUDIO_PARAMETER_STREAM_ROUTING "routing" // audio_devices_t
105 #define AUDIO_PARAMETER_STREAM_FORMAT "format" // audio_format_t
106 #define AUDIO_PARAMETER_STREAM_CHANNELS "channels" // audio_channel_mask_t
107 #define AUDIO_PARAMETER_STREAM_FRAME_COUNT "frame_count" // size_t
108 #define AUDIO_PARAMETER_STREAM_INPUT_SOURCE "input_source" // audio_source_t
109 #define AUDIO_PARAMETER_STREAM_SAMPLING_RATE "sampling_rate" // uint32_t
110
111 /* Query supported formats. The response is a '|' separated list of strings from
112 * audio_format_t enum e.g: "sup_formats=AUDIO_FORMAT_PCM_16_BIT" */
113 #define AUDIO_PARAMETER_STREAM_SUP_FORMATS "sup_formats"
114 /* Query supported channel masks. The response is a '|' separated list of strings from
115 * audio_channel_mask_t enum e.g: "sup_channels=AUDIO_CHANNEL_OUT_STEREO|AUDIO_CHANNEL_OUT_MONO" */
116 #define AUDIO_PARAMETER_STREAM_SUP_CHANNELS "sup_channels"
117 /* Query supported sampling rates. The response is a '|' separated list of integer values e.g:
118 * "sup_sampling_rates=44100|48000" */
119 #define AUDIO_PARAMETER_STREAM_SUP_SAMPLING_RATES "sup_sampling_rates"
120
121 /**
122 * audio codec parameters
123 */
124
125 #define AUDIO_OFFLOAD_CODEC_PARAMS "music_offload_codec_param"
126 #define AUDIO_OFFLOAD_CODEC_BIT_PER_SAMPLE "music_offload_bit_per_sample"
127 #define AUDIO_OFFLOAD_CODEC_BIT_RATE "music_offload_bit_rate"
128 #define AUDIO_OFFLOAD_CODEC_AVG_BIT_RATE "music_offload_avg_bit_rate"
129 #define AUDIO_OFFLOAD_CODEC_ID "music_offload_codec_id"
130 #define AUDIO_OFFLOAD_CODEC_BLOCK_ALIGN "music_offload_block_align"
131 #define AUDIO_OFFLOAD_CODEC_SAMPLE_RATE "music_offload_sample_rate"
132 #define AUDIO_OFFLOAD_CODEC_ENCODE_OPTION "music_offload_encode_option"
133 #define AUDIO_OFFLOAD_CODEC_NUM_CHANNEL "music_offload_num_channels"
134 #define AUDIO_OFFLOAD_CODEC_DOWN_SAMPLING "music_offload_down_sampling"
135 #define AUDIO_OFFLOAD_CODEC_DELAY_SAMPLES "delay_samples"
136 #define AUDIO_OFFLOAD_CODEC_PADDING_SAMPLES "padding_samples"
137
138 /**************************************/
139
140 /* common audio stream configuration parameters
141 * You should memset() the entire structure to zero before use to
142 * ensure forward compatibility
143 */
144 struct audio_config {
145 uint32_t sample_rate;
146 audio_channel_mask_t channel_mask;
147 audio_format_t format;
148 audio_offload_info_t offload_info;
149 };
150 typedef struct audio_config audio_config_t;
151
152 /* common audio stream parameters and operations */
153 struct audio_stream {
154
155 /**
156 * Return the sampling rate in Hz - eg. 44100.
157 */
158 uint32_t (*get_sample_rate)(const struct audio_stream *stream);
159
160 /* currently unused - use set_parameters with key
161 * AUDIO_PARAMETER_STREAM_SAMPLING_RATE
162 */
163 int (*set_sample_rate)(struct audio_stream *stream, uint32_t rate);
164
165 /**
166 * Return size of input/output buffer in bytes for this stream - eg. 4800.
167 * It should be a multiple of the frame size. See also get_input_buffer_size.
168 */
169 size_t (*get_buffer_size)(const struct audio_stream *stream);
170
171 /**
172 * Return the channel mask -
173 * e.g. AUDIO_CHANNEL_OUT_STEREO or AUDIO_CHANNEL_IN_STEREO
174 */
175 audio_channel_mask_t (*get_channels)(const struct audio_stream *stream);
176
177 /**
178 * Return the audio format - e.g. AUDIO_FORMAT_PCM_16_BIT
179 */
180 audio_format_t (*get_format)(const struct audio_stream *stream);
181
182 /* currently unused - use set_parameters with key
183 * AUDIO_PARAMETER_STREAM_FORMAT
184 */
185 int (*set_format)(struct audio_stream *stream, audio_format_t format);
186
187 /**
188 * Put the audio hardware input/output into standby mode.
189 * Driver should exit from standby mode at the next I/O operation.
190 * Returns 0 on success and <0 on failure.
191 */
192 int (*standby)(struct audio_stream *stream);
193
194 /** dump the state of the audio input/output device */
195 int (*dump)(const struct audio_stream *stream, int fd);
196
197 /** Return the set of device(s) which this stream is connected to */
198 audio_devices_t (*get_device)(const struct audio_stream *stream);
199
200 /**
201 * Currently unused - set_device() corresponds to set_parameters() with key
202 * AUDIO_PARAMETER_STREAM_ROUTING for both input and output.
203 * AUDIO_PARAMETER_STREAM_INPUT_SOURCE is an additional information used by
204 * input streams only.
205 */
206 int (*set_device)(struct audio_stream *stream, audio_devices_t device);
207
208 /**
209 * set/get audio stream parameters. The function accepts a list of
210 * parameter key value pairs in the form: key1=value1;key2=value2;...
211 *
212 * Some keys are reserved for standard parameters (See AudioParameter class)
213 *
214 * If the implementation does not accept a parameter change while
215 * the output is active but the parameter is acceptable otherwise, it must
216 * return -ENOSYS.
217 *
218 * The audio flinger will put the stream in standby and then change the
219 * parameter value.
220 */
221 int (*set_parameters)(struct audio_stream *stream, const char *kv_pairs);
222
223 /*
224 * Returns a pointer to a heap allocated string. The caller is responsible
225 * for freeing the memory for it using free().
226 */
227 char * (*get_parameters)(const struct audio_stream *stream,
228 const char *keys);
229 int (*add_audio_effect)(const struct audio_stream *stream,
230 effect_handle_t effect);
231 int (*remove_audio_effect)(const struct audio_stream *stream,
232 effect_handle_t effect);
233 };
234 typedef struct audio_stream audio_stream_t;
235
236 /* type of asynchronous write callback events. Mutually exclusive */
237 typedef enum {
238 STREAM_CBK_EVENT_WRITE_READY, /* non blocking write completed */
239 STREAM_CBK_EVENT_DRAIN_READY /* drain completed */
240 } stream_callback_event_t;
241
242 typedef int (*stream_callback_t)(stream_callback_event_t event, void *param, void *cookie);
243
244 /* type of drain requested to audio_stream_out->drain(). Mutually exclusive */
245 typedef enum {
246 AUDIO_DRAIN_ALL, /* drain() returns when all data has been played */
247 AUDIO_DRAIN_EARLY_NOTIFY /* drain() returns a short time before all data
248 from the current track has been played to
249 give time for gapless track switch */
250 } audio_drain_type_t;
251
252 /**
253 * audio_stream_out is the abstraction interface for the audio output hardware.
254 *
255 * It provides information about various properties of the audio output
256 * hardware driver.
257 */
258
259 struct audio_stream_out {
260 struct audio_stream common;
261
262 /**
263 * Return the audio hardware driver estimated latency in milliseconds.
264 */
265 uint32_t (*get_latency)(const struct audio_stream_out *stream);
266
267 /**
268 * Use this method in situations where audio mixing is done in the
269 * hardware. This method serves as a direct interface with hardware,
270 * allowing you to directly set the volume as apposed to via the framework.
271 * This method might produce multiple PCM outputs or hardware accelerated
272 * codecs, such as MP3 or AAC.
273 */
274 int (*set_volume)(struct audio_stream_out *stream, float left, float right);
275
276 /**
277 * Write audio buffer to driver. Returns number of bytes written, or a
278 * negative status_t. If at least one frame was written successfully prior to the error,
279 * it is suggested that the driver return that successful (short) byte count
280 * and then return an error in the subsequent call.
281 *
282 * If set_callback() has previously been called to enable non-blocking mode
283 * the write() is not allowed to block. It must write only the number of
284 * bytes that currently fit in the driver/hardware buffer and then return
285 * this byte count. If this is less than the requested write size the
286 * callback function must be called when more space is available in the
287 * driver/hardware buffer.
288 */
289 ssize_t (*write)(struct audio_stream_out *stream, const void* buffer,
290 size_t bytes);
291
292 /* return the number of audio frames written by the audio dsp to DAC since
293 * the output has exited standby
294 */
295 int (*get_render_position)(const struct audio_stream_out *stream,
296 uint32_t *dsp_frames);
297
298 /**
299 * get the local time at which the next write to the audio driver will be presented.
300 * The units are microseconds, where the epoch is decided by the local audio HAL.
301 */
302 int (*get_next_write_timestamp)(const struct audio_stream_out *stream,
303 int64_t *timestamp);
304
305 /**
306 * set the callback function for notifying completion of non-blocking
307 * write and drain.
308 * Calling this function implies that all future write() and drain()
309 * must be non-blocking and use the callback to signal completion.
310 */
311 int (*set_callback)(struct audio_stream_out *stream,
312 stream_callback_t callback, void *cookie);
313
314 /**
315 * Notifies to the audio driver to stop playback however the queued buffers are
316 * retained by the hardware. Useful for implementing pause/resume. Empty implementation
317 * if not supported however should be implemented for hardware with non-trivial
318 * latency. In the pause state audio hardware could still be using power. User may
319 * consider calling suspend after a timeout.
320 *
321 * Implementation of this function is mandatory for offloaded playback.
322 */
323 int (*pause)(struct audio_stream_out* stream);
324
325 /**
326 * Notifies to the audio driver to resume playback following a pause.
327 * Returns error if called without matching pause.
328 *
329 * Implementation of this function is mandatory for offloaded playback.
330 */
331 int (*resume)(struct audio_stream_out* stream);
332
333 /**
334 * Requests notification when data buffered by the driver/hardware has
335 * been played. If set_callback() has previously been called to enable
336 * non-blocking mode, the drain() must not block, instead it should return
337 * quickly and completion of the drain is notified through the callback.
338 * If set_callback() has not been called, the drain() must block until
339 * completion.
340 * If type==AUDIO_DRAIN_ALL, the drain completes when all previously written
341 * data has been played.
342 * If type==AUDIO_DRAIN_EARLY_NOTIFY, the drain completes shortly before all
343 * data for the current track has played to allow time for the framework
344 * to perform a gapless track switch.
345 *
346 * Drain must return immediately on stop() and flush() call
347 *
348 * Implementation of this function is mandatory for offloaded playback.
349 */
350 int (*drain)(struct audio_stream_out* stream, audio_drain_type_t type );
351
352 /**
353 * Notifies to the audio driver to flush the queued data. Stream must already
354 * be paused before calling flush().
355 *
356 * Implementation of this function is mandatory for offloaded playback.
357 */
358 int (*flush)(struct audio_stream_out* stream);
359
360 /**
361 * Return a recent count of the number of audio frames presented to an external observer.
362 * This excludes frames which have been written but are still in the pipeline.
363 * The count is not reset to zero when output enters standby.
364 * Also returns the value of CLOCK_MONOTONIC as of this presentation count.
365 * The returned count is expected to be 'recent',
366 * but does not need to be the most recent possible value.
367 * However, the associated time should correspond to whatever count is returned.
368 * Example: assume that N+M frames have been presented, where M is a 'small' number.
369 * Then it is permissible to return N instead of N+M,
370 * and the timestamp should correspond to N rather than N+M.
371 * The terms 'recent' and 'small' are not defined.
372 * They reflect the quality of the implementation.
373 *
374 * 3.0 and higher only.
375 */
376 int (*get_presentation_position)(const struct audio_stream_out *stream,
377 uint64_t *frames, struct timespec *timestamp);
378
379 };
380 typedef struct audio_stream_out audio_stream_out_t;
381
382 struct audio_stream_in {
383 struct audio_stream common;
384
385 /** set the input gain for the audio driver. This method is for
386 * for future use */
387 int (*set_gain)(struct audio_stream_in *stream, float gain);
388
389 /** Read audio buffer in from audio driver. Returns number of bytes read, or a
390 * negative status_t. If at least one frame was read prior to the error,
391 * read should return that byte count and then return an error in the subsequent call.
392 */
393 ssize_t (*read)(struct audio_stream_in *stream, void* buffer,
394 size_t bytes);
395
396 /**
397 * Return the amount of input frames lost in the audio driver since the
398 * last call of this function.
399 * Audio driver is expected to reset the value to 0 and restart counting
400 * upon returning the current value by this function call.
401 * Such loss typically occurs when the user space process is blocked
402 * longer than the capacity of audio driver buffers.
403 *
404 * Unit: the number of input audio frames
405 */
406 uint32_t (*get_input_frames_lost)(struct audio_stream_in *stream);
407 };
408 typedef struct audio_stream_in audio_stream_in_t;
409
410 /**
411 * return the frame size (number of bytes per sample).
412 */
audio_stream_frame_size(const struct audio_stream * s)413 static inline size_t audio_stream_frame_size(const struct audio_stream *s)
414 {
415 size_t chan_samp_sz;
416 audio_format_t format = s->get_format(s);
417
418 if (audio_is_linear_pcm(format)) {
419 chan_samp_sz = audio_bytes_per_sample(format);
420 return popcount(s->get_channels(s)) * chan_samp_sz;
421 }
422
423 return sizeof(int8_t);
424 }
425
426
427 /**********************************************************************/
428
429 /**
430 * Every hardware module must have a data structure named HAL_MODULE_INFO_SYM
431 * and the fields of this data structure must begin with hw_module_t
432 * followed by module specific information.
433 */
434 struct audio_module {
435 struct hw_module_t common;
436 };
437
438 struct audio_hw_device {
439 struct hw_device_t common;
440
441 /**
442 * used by audio flinger to enumerate what devices are supported by
443 * each audio_hw_device implementation.
444 *
445 * Return value is a bitmask of 1 or more values of audio_devices_t
446 *
447 * NOTE: audio HAL implementations starting with
448 * AUDIO_DEVICE_API_VERSION_2_0 do not implement this function.
449 * All supported devices should be listed in audio_policy.conf
450 * file and the audio policy manager must choose the appropriate
451 * audio module based on information in this file.
452 */
453 uint32_t (*get_supported_devices)(const struct audio_hw_device *dev);
454
455 /**
456 * check to see if the audio hardware interface has been initialized.
457 * returns 0 on success, -ENODEV on failure.
458 */
459 int (*init_check)(const struct audio_hw_device *dev);
460
461 /** set the audio volume of a voice call. Range is between 0.0 and 1.0 */
462 int (*set_voice_volume)(struct audio_hw_device *dev, float volume);
463
464 /**
465 * set the audio volume for all audio activities other than voice call.
466 * Range between 0.0 and 1.0. If any value other than 0 is returned,
467 * the software mixer will emulate this capability.
468 */
469 int (*set_master_volume)(struct audio_hw_device *dev, float volume);
470
471 /**
472 * Get the current master volume value for the HAL, if the HAL supports
473 * master volume control. AudioFlinger will query this value from the
474 * primary audio HAL when the service starts and use the value for setting
475 * the initial master volume across all HALs. HALs which do not support
476 * this method may leave it set to NULL.
477 */
478 int (*get_master_volume)(struct audio_hw_device *dev, float *volume);
479
480 /**
481 * set_mode is called when the audio mode changes. AUDIO_MODE_NORMAL mode
482 * is for standard audio playback, AUDIO_MODE_RINGTONE when a ringtone is
483 * playing, and AUDIO_MODE_IN_CALL when a call is in progress.
484 */
485 int (*set_mode)(struct audio_hw_device *dev, audio_mode_t mode);
486
487 /* mic mute */
488 int (*set_mic_mute)(struct audio_hw_device *dev, bool state);
489 int (*get_mic_mute)(const struct audio_hw_device *dev, bool *state);
490
491 /* set/get global audio parameters */
492 int (*set_parameters)(struct audio_hw_device *dev, const char *kv_pairs);
493
494 /*
495 * Returns a pointer to a heap allocated string. The caller is responsible
496 * for freeing the memory for it using free().
497 */
498 char * (*get_parameters)(const struct audio_hw_device *dev,
499 const char *keys);
500
501 /* Returns audio input buffer size according to parameters passed or
502 * 0 if one of the parameters is not supported.
503 * See also get_buffer_size which is for a particular stream.
504 */
505 size_t (*get_input_buffer_size)(const struct audio_hw_device *dev,
506 const struct audio_config *config);
507
508 /** This method creates and opens the audio hardware output stream */
509 int (*open_output_stream)(struct audio_hw_device *dev,
510 audio_io_handle_t handle,
511 audio_devices_t devices,
512 audio_output_flags_t flags,
513 struct audio_config *config,
514 struct audio_stream_out **stream_out);
515
516 void (*close_output_stream)(struct audio_hw_device *dev,
517 struct audio_stream_out* stream_out);
518
519 /** This method creates and opens the audio hardware input stream */
520 int (*open_input_stream)(struct audio_hw_device *dev,
521 audio_io_handle_t handle,
522 audio_devices_t devices,
523 struct audio_config *config,
524 struct audio_stream_in **stream_in);
525
526 void (*close_input_stream)(struct audio_hw_device *dev,
527 struct audio_stream_in *stream_in);
528
529 /** This method dumps the state of the audio hardware */
530 int (*dump)(const struct audio_hw_device *dev, int fd);
531
532 /**
533 * set the audio mute status for all audio activities. If any value other
534 * than 0 is returned, the software mixer will emulate this capability.
535 */
536 int (*set_master_mute)(struct audio_hw_device *dev, bool mute);
537
538 /**
539 * Get the current master mute status for the HAL, if the HAL supports
540 * master mute control. AudioFlinger will query this value from the primary
541 * audio HAL when the service starts and use the value for setting the
542 * initial master mute across all HALs. HALs which do not support this
543 * method may leave it set to NULL.
544 */
545 int (*get_master_mute)(struct audio_hw_device *dev, bool *mute);
546 };
547 typedef struct audio_hw_device audio_hw_device_t;
548
549 /** convenience API for opening and closing a supported device */
550
audio_hw_device_open(const struct hw_module_t * module,struct audio_hw_device ** device)551 static inline int audio_hw_device_open(const struct hw_module_t* module,
552 struct audio_hw_device** device)
553 {
554 return module->methods->open(module, AUDIO_HARDWARE_INTERFACE,
555 (struct hw_device_t**)device);
556 }
557
audio_hw_device_close(struct audio_hw_device * device)558 static inline int audio_hw_device_close(struct audio_hw_device* device)
559 {
560 return device->common.close(&device->common);
561 }
562
563
564 __END_DECLS
565
566 #endif // ANDROID_AUDIO_INTERFACE_H
567