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_CORE_H
19 #define ANDROID_AUDIO_CORE_H
20
21 #include <stdbool.h>
22 #include <stdint.h>
23 #include <stdio.h>
24 #include <sys/cdefs.h>
25 #include <sys/types.h>
26
27 #include <cutils/bitops.h>
28
29 #include "audio-base.h"
30 #include "audio-base-utils.h"
31
32 __BEGIN_DECLS
33
34 /* The enums were moved here mostly from
35 * frameworks/base/include/media/AudioSystem.h
36 */
37
38 /* represents an invalid uid for tracks; the calling or client uid is often substituted. */
39 #define AUDIO_UID_INVALID ((uid_t)-1)
40
41 /* device address used to refer to the standard remote submix */
42 #define AUDIO_REMOTE_SUBMIX_DEVICE_ADDRESS "0"
43
44 /* AudioFlinger and AudioPolicy services use I/O handles to identify audio sources and sinks */
45 typedef int audio_io_handle_t;
46
47 typedef uint32_t audio_flags_mask_t;
48
49 /* Do not change these values without updating their counterparts
50 * in frameworks/base/media/java/android/media/AudioAttributes.java
51 */
52 enum {
53 AUDIO_FLAG_NONE = 0x0,
54 AUDIO_FLAG_AUDIBILITY_ENFORCED = 0x1,
55 AUDIO_FLAG_SECURE = 0x2,
56 AUDIO_FLAG_SCO = 0x4,
57 AUDIO_FLAG_BEACON = 0x8,
58 AUDIO_FLAG_HW_AV_SYNC = 0x10,
59 AUDIO_FLAG_HW_HOTWORD = 0x20,
60 AUDIO_FLAG_BYPASS_INTERRUPTION_POLICY = 0x40,
61 AUDIO_FLAG_BYPASS_MUTE = 0x80,
62 AUDIO_FLAG_LOW_LATENCY = 0x100,
63 AUDIO_FLAG_DEEP_BUFFER = 0x200,
64 };
65
66 /* Audio attributes */
67 #define AUDIO_ATTRIBUTES_TAGS_MAX_SIZE 256
68 typedef struct {
69 audio_content_type_t content_type;
70 audio_usage_t usage;
71 audio_source_t source;
72 audio_flags_mask_t flags;
73 char tags[AUDIO_ATTRIBUTES_TAGS_MAX_SIZE]; /* UTF8 */
74 } __attribute__((packed)) audio_attributes_t; // sent through Binder;
75
76 /* a unique ID allocated by AudioFlinger for use as an audio_io_handle_t, audio_session_t,
77 * effect ID (int), audio_module_handle_t, and audio_patch_handle_t.
78 * Audio port IDs (audio_port_handle_t) are allocated by AudioPolicy
79 * in a different namespace than AudioFlinger unique IDs.
80 */
81 typedef int audio_unique_id_t;
82
83 /* Possible uses for an audio_unique_id_t */
84 typedef enum {
85 AUDIO_UNIQUE_ID_USE_UNSPECIFIED = 0,
86 AUDIO_UNIQUE_ID_USE_SESSION = 1, // for allocated sessions, not special AUDIO_SESSION_*
87 AUDIO_UNIQUE_ID_USE_MODULE = 2,
88 AUDIO_UNIQUE_ID_USE_EFFECT = 3,
89 AUDIO_UNIQUE_ID_USE_PATCH = 4,
90 AUDIO_UNIQUE_ID_USE_OUTPUT = 5,
91 AUDIO_UNIQUE_ID_USE_INPUT = 6,
92 AUDIO_UNIQUE_ID_USE_PLAYER = 7,
93 AUDIO_UNIQUE_ID_USE_MAX = 8, // must be a power-of-two
94 AUDIO_UNIQUE_ID_USE_MASK = AUDIO_UNIQUE_ID_USE_MAX - 1
95 } audio_unique_id_use_t;
96
97 /* Return the use of an audio_unique_id_t */
audio_unique_id_get_use(audio_unique_id_t id)98 static inline audio_unique_id_use_t audio_unique_id_get_use(audio_unique_id_t id)
99 {
100 return (audio_unique_id_use_t) (id & AUDIO_UNIQUE_ID_USE_MASK);
101 }
102
103 /* Reserved audio_unique_id_t values. FIXME: not a complete list. */
104 #define AUDIO_UNIQUE_ID_ALLOCATE AUDIO_SESSION_ALLOCATE
105
106 /* A channel mask per se only defines the presence or absence of a channel, not the order.
107 * But see AUDIO_INTERLEAVE_* below for the platform convention of order.
108 *
109 * audio_channel_mask_t is an opaque type and its internal layout should not
110 * be assumed as it may change in the future.
111 * Instead, always use the functions declared in this header to examine.
112 *
113 * These are the current representations:
114 *
115 * AUDIO_CHANNEL_REPRESENTATION_POSITION
116 * is a channel mask representation for position assignment.
117 * Each low-order bit corresponds to the spatial position of a transducer (output),
118 * or interpretation of channel (input).
119 * The user of a channel mask needs to know the context of whether it is for output or input.
120 * The constants AUDIO_CHANNEL_OUT_* or AUDIO_CHANNEL_IN_* apply to the bits portion.
121 * It is not permitted for no bits to be set.
122 *
123 * AUDIO_CHANNEL_REPRESENTATION_INDEX
124 * is a channel mask representation for index assignment.
125 * Each low-order bit corresponds to a selected channel.
126 * There is no platform interpretation of the various bits.
127 * There is no concept of output or input.
128 * It is not permitted for no bits to be set.
129 *
130 * All other representations are reserved for future use.
131 *
132 * Warning: current representation distinguishes between input and output, but this will not the be
133 * case in future revisions of the platform. Wherever there is an ambiguity between input and output
134 * that is currently resolved by checking the channel mask, the implementer should look for ways to
135 * fix it with additional information outside of the mask.
136 */
137 typedef uint32_t audio_channel_mask_t;
138
139 /* log(2) of maximum number of representations, not part of public API */
140 #define AUDIO_CHANNEL_REPRESENTATION_LOG2 2
141
142 /* The return value is undefined if the channel mask is invalid. */
audio_channel_mask_get_bits(audio_channel_mask_t channel)143 static inline uint32_t audio_channel_mask_get_bits(audio_channel_mask_t channel)
144 {
145 return channel & ((1 << AUDIO_CHANNEL_COUNT_MAX) - 1);
146 }
147
148 typedef uint32_t audio_channel_representation_t;
149
150 /* The return value is undefined if the channel mask is invalid. */
audio_channel_mask_get_representation(audio_channel_mask_t channel)151 static inline audio_channel_representation_t audio_channel_mask_get_representation(
152 audio_channel_mask_t channel)
153 {
154 // The right shift should be sufficient, but also "and" for safety in case mask is not 32 bits
155 return (audio_channel_representation_t)
156 ((channel >> AUDIO_CHANNEL_COUNT_MAX) & ((1 << AUDIO_CHANNEL_REPRESENTATION_LOG2) - 1));
157 }
158
159 /* Returns true if the channel mask is valid,
160 * or returns false for AUDIO_CHANNEL_NONE, AUDIO_CHANNEL_INVALID, and other invalid values.
161 * This function is unable to determine whether a channel mask for position assignment
162 * is invalid because an output mask has an invalid output bit set,
163 * or because an input mask has an invalid input bit set.
164 * All other APIs that take a channel mask assume that it is valid.
165 */
audio_channel_mask_is_valid(audio_channel_mask_t channel)166 static inline bool audio_channel_mask_is_valid(audio_channel_mask_t channel)
167 {
168 uint32_t bits = audio_channel_mask_get_bits(channel);
169 audio_channel_representation_t representation = audio_channel_mask_get_representation(channel);
170 switch (representation) {
171 case AUDIO_CHANNEL_REPRESENTATION_POSITION:
172 case AUDIO_CHANNEL_REPRESENTATION_INDEX:
173 break;
174 default:
175 bits = 0;
176 break;
177 }
178 return bits != 0;
179 }
180
181 /* Not part of public API */
audio_channel_mask_from_representation_and_bits(audio_channel_representation_t representation,uint32_t bits)182 static inline audio_channel_mask_t audio_channel_mask_from_representation_and_bits(
183 audio_channel_representation_t representation, uint32_t bits)
184 {
185 return (audio_channel_mask_t) ((representation << AUDIO_CHANNEL_COUNT_MAX) | bits);
186 }
187
188 /**
189 * Expresses the convention when stereo audio samples are stored interleaved
190 * in an array. This should improve readability by allowing code to use
191 * symbolic indices instead of hard-coded [0] and [1].
192 *
193 * For multi-channel beyond stereo, the platform convention is that channels
194 * are interleaved in order from least significant channel mask bit to most
195 * significant channel mask bit, with unused bits skipped. Any exceptions
196 * to this convention will be noted at the appropriate API.
197 */
198 enum {
199 AUDIO_INTERLEAVE_LEFT = 0,
200 AUDIO_INTERLEAVE_RIGHT = 1,
201 };
202
203 /* This enum is deprecated */
204 typedef enum {
205 AUDIO_IN_ACOUSTICS_NONE = 0,
206 AUDIO_IN_ACOUSTICS_AGC_ENABLE = 0x0001,
207 AUDIO_IN_ACOUSTICS_AGC_DISABLE = 0,
208 AUDIO_IN_ACOUSTICS_NS_ENABLE = 0x0002,
209 AUDIO_IN_ACOUSTICS_NS_DISABLE = 0,
210 AUDIO_IN_ACOUSTICS_TX_IIR_ENABLE = 0x0004,
211 AUDIO_IN_ACOUSTICS_TX_DISABLE = 0,
212 } audio_in_acoustics_t;
213
214 typedef uint32_t audio_devices_t;
215 /**
216 * Stub audio output device. Used in policy configuration file on platforms without audio outputs.
217 * This alias value to AUDIO_DEVICE_OUT_DEFAULT is only used in the audio policy context.
218 */
219 #define AUDIO_DEVICE_OUT_STUB AUDIO_DEVICE_OUT_DEFAULT
220 /**
221 * Stub audio input device. Used in policy configuration file on platforms without audio inputs.
222 * This alias value to AUDIO_DEVICE_IN_DEFAULT is only used in the audio policy context.
223 */
224 #define AUDIO_DEVICE_IN_STUB AUDIO_DEVICE_IN_DEFAULT
225
226 /* Additional information about compressed streams offloaded to
227 * hardware playback
228 * The version and size fields must be initialized by the caller by using
229 * one of the constants defined here.
230 * Must be aligned to transmit as raw memory through Binder.
231 */
232 typedef struct {
233 uint16_t version; // version of the info structure
234 uint16_t size; // total size of the structure including version and size
235 uint32_t sample_rate; // sample rate in Hz
236 audio_channel_mask_t channel_mask; // channel mask
237 audio_format_t format; // audio format
238 audio_stream_type_t stream_type; // stream type
239 uint32_t bit_rate; // bit rate in bits per second
240 int64_t duration_us; // duration in microseconds, -1 if unknown
241 bool has_video; // true if stream is tied to a video stream
242 bool is_streaming; // true if streaming, false if local playback
243 uint32_t bit_width;
244 uint32_t offload_buffer_size; // offload fragment size
245 audio_usage_t usage;
246 } __attribute__((aligned(8))) audio_offload_info_t;
247
248 #define AUDIO_MAKE_OFFLOAD_INFO_VERSION(maj,min) \
249 ((((maj) & 0xff) << 8) | ((min) & 0xff))
250
251 #define AUDIO_OFFLOAD_INFO_VERSION_0_1 AUDIO_MAKE_OFFLOAD_INFO_VERSION(0, 1)
252 #define AUDIO_OFFLOAD_INFO_VERSION_CURRENT AUDIO_OFFLOAD_INFO_VERSION_0_1
253
254 static const audio_offload_info_t AUDIO_INFO_INITIALIZER = {
255 /* .version = */ AUDIO_OFFLOAD_INFO_VERSION_CURRENT,
256 /* .size = */ sizeof(audio_offload_info_t),
257 /* .sample_rate = */ 0,
258 /* .channel_mask = */ 0,
259 /* .format = */ AUDIO_FORMAT_DEFAULT,
260 /* .stream_type = */ AUDIO_STREAM_VOICE_CALL,
261 /* .bit_rate = */ 0,
262 /* .duration_us = */ 0,
263 /* .has_video = */ false,
264 /* .is_streaming = */ false,
265 /* .bit_width = */ 16,
266 /* .offload_buffer_size = */ 0,
267 /* .usage = */ AUDIO_USAGE_UNKNOWN
268 };
269
270 /* common audio stream configuration parameters
271 * You should memset() the entire structure to zero before use to
272 * ensure forward compatibility
273 * Must be aligned to transmit as raw memory through Binder.
274 */
275 struct __attribute__((aligned(8))) audio_config {
276 uint32_t sample_rate;
277 audio_channel_mask_t channel_mask;
278 audio_format_t format;
279 audio_offload_info_t offload_info;
280 uint32_t frame_count;
281 };
282 typedef struct audio_config audio_config_t;
283
284 static const audio_config_t AUDIO_CONFIG_INITIALIZER = {
285 /* .sample_rate = */ 0,
286 /* .channel_mask = */ AUDIO_CHANNEL_NONE,
287 /* .format = */ AUDIO_FORMAT_DEFAULT,
288 /* .offload_info = */ {
289 /* .version = */ AUDIO_OFFLOAD_INFO_VERSION_CURRENT,
290 /* .size = */ sizeof(audio_offload_info_t),
291 /* .sample_rate = */ 0,
292 /* .channel_mask = */ 0,
293 /* .format = */ AUDIO_FORMAT_DEFAULT,
294 /* .stream_type = */ AUDIO_STREAM_VOICE_CALL,
295 /* .bit_rate = */ 0,
296 /* .duration_us = */ 0,
297 /* .has_video = */ false,
298 /* .is_streaming = */ false,
299 /* .bit_width = */ 16,
300 /* .offload_buffer_size = */ 0,
301 /* .usage = */ AUDIO_USAGE_UNKNOWN
302 },
303 /* .frame_count = */ 0,
304 };
305
306 struct audio_config_base {
307 uint32_t sample_rate;
308 audio_channel_mask_t channel_mask;
309 audio_format_t format;
310 };
311
312 typedef struct audio_config_base audio_config_base_t;
313
314 static const audio_config_base_t AUDIO_CONFIG_BASE_INITIALIZER = {
315 /* .sample_rate = */ 0,
316 /* .channel_mask = */ AUDIO_CHANNEL_NONE,
317 /* .format = */ AUDIO_FORMAT_DEFAULT
318 };
319
320 /* audio hw module handle functions or structures referencing a module */
321 typedef int audio_module_handle_t;
322
323 /******************************
324 * Volume control
325 *****************************/
326
327 /** 3 dB headroom are allowed on float samples (3db = 10^(3/20) = 1.412538).
328 * See: https://developer.android.com/reference/android/media/AudioTrack.html#write(float[], int, int, int)
329 */
330 #define FLOAT_NOMINAL_RANGE_HEADROOM 1.412538
331
332 /* If the audio hardware supports gain control on some audio paths,
333 * the platform can expose them in the audio_policy.conf file. The audio HAL
334 * will then implement gain control functions that will use the following data
335 * structures. */
336
337 typedef uint32_t audio_gain_mode_t;
338
339
340 /* An audio_gain struct is a representation of a gain stage.
341 * A gain stage is always attached to an audio port. */
342 struct audio_gain {
343 audio_gain_mode_t mode; /* e.g. AUDIO_GAIN_MODE_JOINT */
344 audio_channel_mask_t channel_mask; /* channels which gain an be controlled.
345 N/A if AUDIO_GAIN_MODE_CHANNELS is not supported */
346 int min_value; /* minimum gain value in millibels */
347 int max_value; /* maximum gain value in millibels */
348 int default_value; /* default gain value in millibels */
349 unsigned int step_value; /* gain step in millibels */
350 unsigned int min_ramp_ms; /* minimum ramp duration in ms */
351 unsigned int max_ramp_ms; /* maximum ramp duration in ms */
352 };
353
354 /* The gain configuration structure is used to get or set the gain values of a
355 * given port */
356 struct audio_gain_config {
357 int index; /* index of the corresponding audio_gain in the
358 audio_port gains[] table */
359 audio_gain_mode_t mode; /* mode requested for this command */
360 audio_channel_mask_t channel_mask; /* channels which gain value follows.
361 N/A in joint mode */
362
363 // note this "8" is not FCC_8, so it won't need to be changed for > 8 channels
364 int values[sizeof(audio_channel_mask_t) * 8]; /* gain values in millibels
365 for each channel ordered from LSb to MSb in
366 channel mask. The number of values is 1 in joint
367 mode or popcount(channel_mask) */
368 unsigned int ramp_duration_ms; /* ramp duration in ms */
369 };
370
371 /******************************
372 * Routing control
373 *****************************/
374
375 /* Types defined here are used to describe an audio source or sink at internal
376 * framework interfaces (audio policy, patch panel) or at the audio HAL.
377 * Sink and sources are grouped in a concept of “audio port” representing an
378 * audio end point at the edge of the system managed by the module exposing
379 * the interface. */
380
381 /* Each port has a unique ID or handle allocated by policy manager */
382 typedef int audio_port_handle_t;
383
384 /* the maximum length for the human-readable device name */
385 #define AUDIO_PORT_MAX_NAME_LEN 128
386
387 /* maximum audio device address length */
388 #define AUDIO_DEVICE_MAX_ADDRESS_LEN 32
389
390 /* extension for audio port configuration structure when the audio port is a
391 * hardware device */
392 struct audio_port_config_device_ext {
393 audio_module_handle_t hw_module; /* module the device is attached to */
394 audio_devices_t type; /* device type (e.g AUDIO_DEVICE_OUT_SPEAKER) */
395 char address[AUDIO_DEVICE_MAX_ADDRESS_LEN]; /* device address. "" if N/A */
396 };
397
398 /* extension for audio port configuration structure when the audio port is a
399 * sub mix */
400 struct audio_port_config_mix_ext {
401 audio_module_handle_t hw_module; /* module the stream is attached to */
402 audio_io_handle_t handle; /* I/O handle of the input/output stream */
403 union {
404 //TODO: change use case for output streams: use strategy and mixer attributes
405 audio_stream_type_t stream;
406 audio_source_t source;
407 } usecase;
408 };
409
410 /* extension for audio port configuration structure when the audio port is an
411 * audio session */
412 struct audio_port_config_session_ext {
413 audio_session_t session; /* audio session */
414 };
415
416 /* audio port configuration structure used to specify a particular configuration of
417 * an audio port */
418 struct audio_port_config {
419 audio_port_handle_t id; /* port unique ID */
420 audio_port_role_t role; /* sink or source */
421 audio_port_type_t type; /* device, mix ... */
422 unsigned int config_mask; /* e.g AUDIO_PORT_CONFIG_ALL */
423 unsigned int sample_rate; /* sampling rate in Hz */
424 audio_channel_mask_t channel_mask; /* channel mask if applicable */
425 audio_format_t format; /* format if applicable */
426 struct audio_gain_config gain; /* gain to apply if applicable */
427 union {
428 struct audio_port_config_device_ext device; /* device specific info */
429 struct audio_port_config_mix_ext mix; /* mix specific info */
430 struct audio_port_config_session_ext session; /* session specific info */
431 } ext;
432 };
433
434
435 /* max number of sampling rates in audio port */
436 #define AUDIO_PORT_MAX_SAMPLING_RATES 32
437 /* max number of channel masks in audio port */
438 #define AUDIO_PORT_MAX_CHANNEL_MASKS 32
439 /* max number of audio formats in audio port */
440 #define AUDIO_PORT_MAX_FORMATS 32
441 /* max number of gain controls in audio port */
442 #define AUDIO_PORT_MAX_GAINS 16
443
444 /* extension for audio port structure when the audio port is a hardware device */
445 struct audio_port_device_ext {
446 audio_module_handle_t hw_module; /* module the device is attached to */
447 audio_devices_t type; /* device type (e.g AUDIO_DEVICE_OUT_SPEAKER) */
448 char address[AUDIO_DEVICE_MAX_ADDRESS_LEN];
449 };
450
451 /* extension for audio port structure when the audio port is a sub mix */
452 struct audio_port_mix_ext {
453 audio_module_handle_t hw_module; /* module the stream is attached to */
454 audio_io_handle_t handle; /* I/O handle of the input.output stream */
455 audio_mix_latency_class_t latency_class; /* latency class */
456 // other attributes: routing strategies
457 };
458
459 /* extension for audio port structure when the audio port is an audio session */
460 struct audio_port_session_ext {
461 audio_session_t session; /* audio session */
462 };
463
464 struct audio_port {
465 audio_port_handle_t id; /* port unique ID */
466 audio_port_role_t role; /* sink or source */
467 audio_port_type_t type; /* device, mix ... */
468 char name[AUDIO_PORT_MAX_NAME_LEN];
469 unsigned int num_sample_rates; /* number of sampling rates in following array */
470 unsigned int sample_rates[AUDIO_PORT_MAX_SAMPLING_RATES];
471 unsigned int num_channel_masks; /* number of channel masks in following array */
472 audio_channel_mask_t channel_masks[AUDIO_PORT_MAX_CHANNEL_MASKS];
473 unsigned int num_formats; /* number of formats in following array */
474 audio_format_t formats[AUDIO_PORT_MAX_FORMATS];
475 unsigned int num_gains; /* number of gains in following array */
476 struct audio_gain gains[AUDIO_PORT_MAX_GAINS];
477 struct audio_port_config active_config; /* current audio port configuration */
478 union {
479 struct audio_port_device_ext device;
480 struct audio_port_mix_ext mix;
481 struct audio_port_session_ext session;
482 } ext;
483 };
484
485 /* An audio patch represents a connection between one or more source ports and
486 * one or more sink ports. Patches are connected and disconnected by audio policy manager or by
487 * applications via framework APIs.
488 * Each patch is identified by a handle at the interface used to create that patch. For instance,
489 * when a patch is created by the audio HAL, the HAL allocates and returns a handle.
490 * This handle is unique to a given audio HAL hardware module.
491 * But the same patch receives another system wide unique handle allocated by the framework.
492 * This unique handle is used for all transactions inside the framework.
493 */
494 typedef int audio_patch_handle_t;
495
496 #define AUDIO_PATCH_PORTS_MAX 16
497
498 struct audio_patch {
499 audio_patch_handle_t id; /* patch unique ID */
500 unsigned int num_sources; /* number of sources in following array */
501 struct audio_port_config sources[AUDIO_PATCH_PORTS_MAX];
502 unsigned int num_sinks; /* number of sinks in following array */
503 struct audio_port_config sinks[AUDIO_PATCH_PORTS_MAX];
504 };
505
506
507
508 /* a HW synchronization source returned by the audio HAL */
509 typedef uint32_t audio_hw_sync_t;
510
511 /* an invalid HW synchronization source indicating an error */
512 #define AUDIO_HW_SYNC_INVALID 0
513
514 /**
515 * Mmap buffer descriptor returned by audio_stream->create_mmap_buffer().
516 * note\ Used by streams opened in mmap mode.
517 */
518 struct audio_mmap_buffer_info {
519 void* shared_memory_address; /**< base address of mmap memory buffer.
520 For use by local process only */
521 int32_t shared_memory_fd; /**< FD for mmap memory buffer */
522 int32_t buffer_size_frames; /**< total buffer size in frames */
523 int32_t burst_size_frames; /**< transfer size granularity in frames */
524 };
525
526 /**
527 * Mmap buffer read/write position returned by audio_stream->get_mmap_position().
528 * note\ Used by streams opened in mmap mode.
529 */
530 struct audio_mmap_position {
531 int64_t time_nanoseconds; /**< timestamp in ns, CLOCK_MONOTONIC */
532 int32_t position_frames; /**< increasing 32 bit frame count reset when stream->stop()
533 is called */
534 };
535
536 /** Metadata of a record track for an in stream. */
537 typedef struct playback_track_metadata {
538 audio_usage_t usage;
539 audio_content_type_t content_type;
540 float gain; // Normalized linear volume. 0=silence, 1=0dbfs...
541 } playback_track_metadata_t;
542
543 /** Metadata of a playback track for an out stream. */
544 typedef struct record_track_metadata {
545 audio_source_t source;
546 float gain; // Normalized linear volume. 0=silence, 1=0dbfs...
547 } record_track_metadata_t;
548
549
550 /******************************
551 * Helper functions
552 *****************************/
553
audio_is_output_device(audio_devices_t device)554 static inline bool audio_is_output_device(audio_devices_t device)
555 {
556 if (((device & AUDIO_DEVICE_BIT_IN) == 0) &&
557 (popcount(device) == 1) && ((device & ~AUDIO_DEVICE_OUT_ALL) == 0))
558 return true;
559 else
560 return false;
561 }
562
audio_is_input_device(audio_devices_t device)563 static inline bool audio_is_input_device(audio_devices_t device)
564 {
565 if ((device & AUDIO_DEVICE_BIT_IN) != 0) {
566 device &= ~AUDIO_DEVICE_BIT_IN;
567 if ((popcount(device) == 1) && ((device & ~AUDIO_DEVICE_IN_ALL) == 0))
568 return true;
569 }
570 return false;
571 }
572
audio_is_output_devices(audio_devices_t device)573 static inline bool audio_is_output_devices(audio_devices_t device)
574 {
575 return (device & AUDIO_DEVICE_BIT_IN) == 0;
576 }
577
audio_is_a2dp_in_device(audio_devices_t device)578 static inline bool audio_is_a2dp_in_device(audio_devices_t device)
579 {
580 if ((device & AUDIO_DEVICE_BIT_IN) != 0) {
581 device &= ~AUDIO_DEVICE_BIT_IN;
582 if ((popcount(device) == 1) && (device & AUDIO_DEVICE_IN_BLUETOOTH_A2DP))
583 return true;
584 }
585 return false;
586 }
587
audio_is_a2dp_out_device(audio_devices_t device)588 static inline bool audio_is_a2dp_out_device(audio_devices_t device)
589 {
590 if ((popcount(device) == 1) && (device & AUDIO_DEVICE_OUT_ALL_A2DP))
591 return true;
592 else
593 return false;
594 }
595
596 // Deprecated - use audio_is_a2dp_out_device() instead
audio_is_a2dp_device(audio_devices_t device)597 static inline bool audio_is_a2dp_device(audio_devices_t device)
598 {
599 return audio_is_a2dp_out_device(device);
600 }
601
audio_is_bluetooth_sco_device(audio_devices_t device)602 static inline bool audio_is_bluetooth_sco_device(audio_devices_t device)
603 {
604 if ((device & AUDIO_DEVICE_BIT_IN) == 0) {
605 if ((popcount(device) == 1) && ((device & ~AUDIO_DEVICE_OUT_ALL_SCO) == 0))
606 return true;
607 } else {
608 device &= ~AUDIO_DEVICE_BIT_IN;
609 if ((popcount(device) == 1) && ((device & ~AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET) == 0))
610 return true;
611 }
612
613 return false;
614 }
615
audio_is_hearing_aid_out_device(audio_devices_t device)616 static inline bool audio_is_hearing_aid_out_device(audio_devices_t device)
617 {
618 return device == AUDIO_DEVICE_OUT_HEARING_AID;
619 }
620
audio_is_usb_out_device(audio_devices_t device)621 static inline bool audio_is_usb_out_device(audio_devices_t device)
622 {
623 return ((popcount(device) == 1) && (device & AUDIO_DEVICE_OUT_ALL_USB));
624 }
625
audio_is_usb_in_device(audio_devices_t device)626 static inline bool audio_is_usb_in_device(audio_devices_t device)
627 {
628 if ((device & AUDIO_DEVICE_BIT_IN) != 0) {
629 device &= ~AUDIO_DEVICE_BIT_IN;
630 if (popcount(device) == 1 && (device & AUDIO_DEVICE_IN_ALL_USB) != 0)
631 return true;
632 }
633 return false;
634 }
635
636 /* OBSOLETE - use audio_is_usb_out_device() instead. */
audio_is_usb_device(audio_devices_t device)637 static inline bool audio_is_usb_device(audio_devices_t device)
638 {
639 return audio_is_usb_out_device(device);
640 }
641
audio_is_remote_submix_device(audio_devices_t device)642 static inline bool audio_is_remote_submix_device(audio_devices_t device)
643 {
644 if ((audio_is_output_devices(device) &&
645 (device & AUDIO_DEVICE_OUT_REMOTE_SUBMIX) == AUDIO_DEVICE_OUT_REMOTE_SUBMIX)
646 || (!audio_is_output_devices(device) &&
647 (device & AUDIO_DEVICE_IN_REMOTE_SUBMIX) == AUDIO_DEVICE_IN_REMOTE_SUBMIX))
648 return true;
649 else
650 return false;
651 }
652
653 /* Returns true if:
654 * representation is valid, and
655 * there is at least one channel bit set which _could_ correspond to an input channel, and
656 * there are no channel bits set which could _not_ correspond to an input channel.
657 * Otherwise returns false.
658 */
audio_is_input_channel(audio_channel_mask_t channel)659 static inline bool audio_is_input_channel(audio_channel_mask_t channel)
660 {
661 uint32_t bits = audio_channel_mask_get_bits(channel);
662 switch (audio_channel_mask_get_representation(channel)) {
663 case AUDIO_CHANNEL_REPRESENTATION_POSITION:
664 if (bits & ~AUDIO_CHANNEL_IN_ALL) {
665 bits = 0;
666 }
667 // fall through
668 case AUDIO_CHANNEL_REPRESENTATION_INDEX:
669 return bits != 0;
670 default:
671 return false;
672 }
673 }
674
675 /* Returns true if:
676 * representation is valid, and
677 * there is at least one channel bit set which _could_ correspond to an output channel, and
678 * there are no channel bits set which could _not_ correspond to an output channel.
679 * Otherwise returns false.
680 */
audio_is_output_channel(audio_channel_mask_t channel)681 static inline bool audio_is_output_channel(audio_channel_mask_t channel)
682 {
683 uint32_t bits = audio_channel_mask_get_bits(channel);
684 switch (audio_channel_mask_get_representation(channel)) {
685 case AUDIO_CHANNEL_REPRESENTATION_POSITION:
686 if (bits & ~AUDIO_CHANNEL_OUT_ALL) {
687 bits = 0;
688 }
689 // fall through
690 case AUDIO_CHANNEL_REPRESENTATION_INDEX:
691 return bits != 0;
692 default:
693 return false;
694 }
695 }
696
697 /* Returns the number of channels from an input channel mask,
698 * used in the context of audio input or recording.
699 * If a channel bit is set which could _not_ correspond to an input channel,
700 * it is excluded from the count.
701 * Returns zero if the representation is invalid.
702 */
audio_channel_count_from_in_mask(audio_channel_mask_t channel)703 static inline uint32_t audio_channel_count_from_in_mask(audio_channel_mask_t channel)
704 {
705 uint32_t bits = audio_channel_mask_get_bits(channel);
706 switch (audio_channel_mask_get_representation(channel)) {
707 case AUDIO_CHANNEL_REPRESENTATION_POSITION:
708 // TODO: We can now merge with from_out_mask and remove anding
709 bits &= AUDIO_CHANNEL_IN_ALL;
710 // fall through
711 case AUDIO_CHANNEL_REPRESENTATION_INDEX:
712 return popcount(bits);
713 default:
714 return 0;
715 }
716 }
717
718 /* Returns the number of channels from an output channel mask,
719 * used in the context of audio output or playback.
720 * If a channel bit is set which could _not_ correspond to an output channel,
721 * it is excluded from the count.
722 * Returns zero if the representation is invalid.
723 */
audio_channel_count_from_out_mask(audio_channel_mask_t channel)724 static inline uint32_t audio_channel_count_from_out_mask(audio_channel_mask_t channel)
725 {
726 uint32_t bits = audio_channel_mask_get_bits(channel);
727 switch (audio_channel_mask_get_representation(channel)) {
728 case AUDIO_CHANNEL_REPRESENTATION_POSITION:
729 // TODO: We can now merge with from_in_mask and remove anding
730 bits &= AUDIO_CHANNEL_OUT_ALL;
731 // fall through
732 case AUDIO_CHANNEL_REPRESENTATION_INDEX:
733 return popcount(bits);
734 default:
735 return 0;
736 }
737 }
738
739 /* Derive a channel mask for index assignment from a channel count.
740 * Returns the matching channel mask,
741 * or AUDIO_CHANNEL_NONE if the channel count is zero,
742 * or AUDIO_CHANNEL_INVALID if the channel count exceeds AUDIO_CHANNEL_COUNT_MAX.
743 */
audio_channel_mask_for_index_assignment_from_count(uint32_t channel_count)744 static inline audio_channel_mask_t audio_channel_mask_for_index_assignment_from_count(
745 uint32_t channel_count)
746 {
747 if (channel_count == 0) {
748 return AUDIO_CHANNEL_NONE;
749 }
750 if (channel_count > AUDIO_CHANNEL_COUNT_MAX) {
751 return AUDIO_CHANNEL_INVALID;
752 }
753 uint32_t bits = (1 << channel_count) - 1;
754 return audio_channel_mask_from_representation_and_bits(
755 AUDIO_CHANNEL_REPRESENTATION_INDEX, bits);
756 }
757
758 /* Derive an output channel mask for position assignment from a channel count.
759 * This is to be used when the content channel mask is unknown. The 1, 2, 4, 5, 6, 7 and 8 channel
760 * cases are mapped to the standard game/home-theater layouts, but note that 4 is mapped to quad,
761 * and not stereo + FC + mono surround. A channel count of 3 is arbitrarily mapped to stereo + FC
762 * for continuity with stereo.
763 * Returns the matching channel mask,
764 * or AUDIO_CHANNEL_NONE if the channel count is zero,
765 * or AUDIO_CHANNEL_INVALID if the channel count exceeds that of the
766 * configurations for which a default output channel mask is defined.
767 */
audio_channel_out_mask_from_count(uint32_t channel_count)768 static inline audio_channel_mask_t audio_channel_out_mask_from_count(uint32_t channel_count)
769 {
770 uint32_t bits;
771 switch (channel_count) {
772 case 0:
773 return AUDIO_CHANNEL_NONE;
774 case 1:
775 bits = AUDIO_CHANNEL_OUT_MONO;
776 break;
777 case 2:
778 bits = AUDIO_CHANNEL_OUT_STEREO;
779 break;
780 case 3:
781 bits = AUDIO_CHANNEL_OUT_STEREO | AUDIO_CHANNEL_OUT_FRONT_CENTER;
782 break;
783 case 4: // 4.0
784 bits = AUDIO_CHANNEL_OUT_QUAD;
785 break;
786 case 5: // 5.0
787 bits = AUDIO_CHANNEL_OUT_QUAD | AUDIO_CHANNEL_OUT_FRONT_CENTER;
788 break;
789 case 6: // 5.1
790 bits = AUDIO_CHANNEL_OUT_5POINT1;
791 break;
792 case 7: // 6.1
793 bits = AUDIO_CHANNEL_OUT_5POINT1 | AUDIO_CHANNEL_OUT_BACK_CENTER;
794 break;
795 case 8:
796 bits = AUDIO_CHANNEL_OUT_7POINT1;
797 break;
798 // FIXME FCC_8
799 default:
800 return AUDIO_CHANNEL_INVALID;
801 }
802 return audio_channel_mask_from_representation_and_bits(
803 AUDIO_CHANNEL_REPRESENTATION_POSITION, bits);
804 }
805
806 /* Derive a default input channel mask from a channel count.
807 * Assumes a position mask for mono and stereo, or an index mask for channel counts > 2.
808 * Returns the matching channel mask,
809 * or AUDIO_CHANNEL_NONE if the channel count is zero,
810 * or AUDIO_CHANNEL_INVALID if the channel count exceeds that of the
811 * configurations for which a default input channel mask is defined.
812 */
audio_channel_in_mask_from_count(uint32_t channel_count)813 static inline audio_channel_mask_t audio_channel_in_mask_from_count(uint32_t channel_count)
814 {
815 uint32_t bits;
816 switch (channel_count) {
817 case 0:
818 return AUDIO_CHANNEL_NONE;
819 case 1:
820 bits = AUDIO_CHANNEL_IN_MONO;
821 break;
822 case 2:
823 bits = AUDIO_CHANNEL_IN_STEREO;
824 break;
825 case 3:
826 case 4:
827 case 5:
828 case 6:
829 case 7:
830 case 8:
831 // FIXME FCC_8
832 return audio_channel_mask_for_index_assignment_from_count(channel_count);
833 default:
834 return AUDIO_CHANNEL_INVALID;
835 }
836 return audio_channel_mask_from_representation_and_bits(
837 AUDIO_CHANNEL_REPRESENTATION_POSITION, bits);
838 }
839
audio_channel_mask_in_to_out(audio_channel_mask_t in)840 static inline audio_channel_mask_t audio_channel_mask_in_to_out(audio_channel_mask_t in)
841 {
842 switch (in) {
843 case AUDIO_CHANNEL_IN_MONO:
844 return AUDIO_CHANNEL_OUT_MONO;
845 case AUDIO_CHANNEL_IN_STEREO:
846 return AUDIO_CHANNEL_OUT_STEREO;
847 case AUDIO_CHANNEL_IN_5POINT1:
848 return AUDIO_CHANNEL_OUT_5POINT1;
849 case AUDIO_CHANNEL_IN_3POINT1POINT2:
850 return AUDIO_CHANNEL_OUT_3POINT1POINT2;
851 case AUDIO_CHANNEL_IN_3POINT0POINT2:
852 return AUDIO_CHANNEL_OUT_3POINT0POINT2;
853 case AUDIO_CHANNEL_IN_2POINT1POINT2:
854 return AUDIO_CHANNEL_OUT_2POINT1POINT2;
855 case AUDIO_CHANNEL_IN_2POINT0POINT2:
856 return AUDIO_CHANNEL_OUT_2POINT0POINT2;
857 default:
858 return AUDIO_CHANNEL_INVALID;
859 }
860 }
861
audio_is_valid_format(audio_format_t format)862 static inline bool audio_is_valid_format(audio_format_t format)
863 {
864 switch (format & AUDIO_FORMAT_MAIN_MASK) {
865 case AUDIO_FORMAT_PCM:
866 switch (format) {
867 case AUDIO_FORMAT_PCM_16_BIT:
868 case AUDIO_FORMAT_PCM_8_BIT:
869 case AUDIO_FORMAT_PCM_32_BIT:
870 case AUDIO_FORMAT_PCM_8_24_BIT:
871 case AUDIO_FORMAT_PCM_FLOAT:
872 case AUDIO_FORMAT_PCM_24_BIT_PACKED:
873 return true;
874 default:
875 return false;
876 }
877 /* not reached */
878 case AUDIO_FORMAT_MP3:
879 case AUDIO_FORMAT_AMR_NB:
880 case AUDIO_FORMAT_AMR_WB:
881 case AUDIO_FORMAT_AAC:
882 case AUDIO_FORMAT_AAC_ADTS:
883 case AUDIO_FORMAT_HE_AAC_V1:
884 case AUDIO_FORMAT_HE_AAC_V2:
885 case AUDIO_FORMAT_AAC_ELD:
886 case AUDIO_FORMAT_AAC_XHE:
887 case AUDIO_FORMAT_VORBIS:
888 case AUDIO_FORMAT_OPUS:
889 case AUDIO_FORMAT_AC3:
890 case AUDIO_FORMAT_E_AC3:
891 case AUDIO_FORMAT_DTS:
892 case AUDIO_FORMAT_DTS_HD:
893 case AUDIO_FORMAT_IEC61937:
894 case AUDIO_FORMAT_DOLBY_TRUEHD:
895 case AUDIO_FORMAT_QCELP:
896 case AUDIO_FORMAT_EVRC:
897 case AUDIO_FORMAT_EVRCB:
898 case AUDIO_FORMAT_EVRCWB:
899 case AUDIO_FORMAT_AAC_ADIF:
900 case AUDIO_FORMAT_AMR_WB_PLUS:
901 case AUDIO_FORMAT_MP2:
902 case AUDIO_FORMAT_EVRCNW:
903 case AUDIO_FORMAT_FLAC:
904 case AUDIO_FORMAT_ALAC:
905 case AUDIO_FORMAT_APE:
906 case AUDIO_FORMAT_WMA:
907 case AUDIO_FORMAT_WMA_PRO:
908 case AUDIO_FORMAT_DSD:
909 case AUDIO_FORMAT_AC4:
910 case AUDIO_FORMAT_LDAC:
911 case AUDIO_FORMAT_E_AC3_JOC:
912 case AUDIO_FORMAT_MAT_1_0:
913 case AUDIO_FORMAT_MAT_2_0:
914 case AUDIO_FORMAT_MAT_2_1:
915 return true;
916 default:
917 return false;
918 }
919 }
920
921 /**
922 * Extract the primary format, eg. PCM, AC3, etc.
923 */
audio_get_main_format(audio_format_t format)924 static inline audio_format_t audio_get_main_format(audio_format_t format)
925 {
926 return (audio_format_t)(format & AUDIO_FORMAT_MAIN_MASK);
927 }
928
929 /**
930 * Is the data plain PCM samples that can be scaled and mixed?
931 */
audio_is_linear_pcm(audio_format_t format)932 static inline bool audio_is_linear_pcm(audio_format_t format)
933 {
934 return (audio_get_main_format(format) == AUDIO_FORMAT_PCM);
935 }
936
937 /**
938 * For this format, is the number of PCM audio frames directly proportional
939 * to the number of data bytes?
940 *
941 * In other words, is the format transported as PCM audio samples,
942 * but not necessarily scalable or mixable.
943 * This returns true for real PCM, but also for AUDIO_FORMAT_IEC61937,
944 * which is transported as 16 bit PCM audio, but where the encoded data
945 * cannot be mixed or scaled.
946 */
audio_has_proportional_frames(audio_format_t format)947 static inline bool audio_has_proportional_frames(audio_format_t format)
948 {
949 audio_format_t mainFormat = audio_get_main_format(format);
950 return (mainFormat == AUDIO_FORMAT_PCM
951 || mainFormat == AUDIO_FORMAT_IEC61937);
952 }
953
audio_bytes_per_sample(audio_format_t format)954 static inline size_t audio_bytes_per_sample(audio_format_t format)
955 {
956 size_t size = 0;
957
958 switch (format) {
959 case AUDIO_FORMAT_PCM_32_BIT:
960 case AUDIO_FORMAT_PCM_8_24_BIT:
961 size = sizeof(int32_t);
962 break;
963 case AUDIO_FORMAT_PCM_24_BIT_PACKED:
964 size = sizeof(uint8_t) * 3;
965 break;
966 case AUDIO_FORMAT_PCM_16_BIT:
967 case AUDIO_FORMAT_IEC61937:
968 size = sizeof(int16_t);
969 break;
970 case AUDIO_FORMAT_PCM_8_BIT:
971 size = sizeof(uint8_t);
972 break;
973 case AUDIO_FORMAT_PCM_FLOAT:
974 size = sizeof(float);
975 break;
976 default:
977 break;
978 }
979 return size;
980 }
981
audio_bytes_per_frame(uint32_t channel_count,audio_format_t format)982 static inline size_t audio_bytes_per_frame(uint32_t channel_count, audio_format_t format)
983 {
984 // cannot overflow for reasonable channel_count
985 return channel_count * audio_bytes_per_sample(format);
986 }
987
988 /* converts device address to string sent to audio HAL via set_parameters */
audio_device_address_to_parameter(audio_devices_t device,const char * address)989 static inline char *audio_device_address_to_parameter(audio_devices_t device, const char *address)
990 {
991 const size_t kSize = AUDIO_DEVICE_MAX_ADDRESS_LEN + sizeof("a2dp_sink_address=");
992 char param[kSize];
993
994 if (device & AUDIO_DEVICE_OUT_ALL_A2DP)
995 snprintf(param, kSize, "%s=%s", "a2dp_sink_address", address);
996 else if (device & AUDIO_DEVICE_OUT_REMOTE_SUBMIX)
997 snprintf(param, kSize, "%s=%s", "mix", address);
998 else
999 snprintf(param, kSize, "%s", address);
1000
1001 return strdup(param);
1002 }
1003
audio_device_is_digital(audio_devices_t device)1004 static inline bool audio_device_is_digital(audio_devices_t device) {
1005 if ((device & AUDIO_DEVICE_BIT_IN) != 0) {
1006 // input
1007 return (~AUDIO_DEVICE_BIT_IN & device & (AUDIO_DEVICE_IN_ALL_USB |
1008 AUDIO_DEVICE_IN_HDMI |
1009 AUDIO_DEVICE_IN_SPDIF |
1010 AUDIO_DEVICE_IN_IP |
1011 AUDIO_DEVICE_IN_BUS)) != 0;
1012 } else {
1013 // output
1014 return (device & (AUDIO_DEVICE_OUT_ALL_USB |
1015 AUDIO_DEVICE_OUT_HDMI |
1016 AUDIO_DEVICE_OUT_HDMI_ARC |
1017 AUDIO_DEVICE_OUT_SPDIF |
1018 AUDIO_DEVICE_OUT_IP |
1019 AUDIO_DEVICE_OUT_BUS)) != 0;
1020 }
1021 }
1022
1023 // Unique effect ID (can be generated from the following site:
1024 // http://www.itu.int/ITU-T/asn1/uuid.html)
1025 // This struct is used for effects identification and in soundtrigger.
1026 typedef struct audio_uuid_s {
1027 uint32_t timeLow;
1028 uint16_t timeMid;
1029 uint16_t timeHiAndVersion;
1030 uint16_t clockSeq;
1031 uint8_t node[6];
1032 } audio_uuid_t;
1033
1034 //TODO: audio_microphone_location_t need to move to HAL v4.0
1035 typedef enum {
1036 AUDIO_MICROPHONE_LOCATION_UNKNOWN = 0,
1037 AUDIO_MICROPHONE_LOCATION_MAINBODY = 1,
1038 AUDIO_MICROPHONE_LOCATION_MAINBODY_MOVABLE = 2,
1039 AUDIO_MICROPHONE_LOCATION_PERIPHERAL = 3,
1040 AUDIO_MICROPHONE_LOCATION_CNT = 4,
1041 } audio_microphone_location_t;
1042
1043 //TODO: audio_microphone_directionality_t need to move to HAL v4.0
1044 typedef enum {
1045 AUDIO_MICROPHONE_DIRECTIONALITY_UNKNOWN = 0,
1046 AUDIO_MICROPHONE_DIRECTIONALITY_OMNI = 1,
1047 AUDIO_MICROPHONE_DIRECTIONALITY_BI_DIRECTIONAL = 2,
1048 AUDIO_MICROPHONE_DIRECTIONALITY_CARDIOID = 3,
1049 AUDIO_MICROPHONE_DIRECTIONALITY_HYPER_CARDIOID = 4,
1050 AUDIO_MICROPHONE_DIRECTIONALITY_SUPER_CARDIOID = 5,
1051 AUDIO_MICROPHONE_DIRECTIONALITY_CNT = 6,
1052 } audio_microphone_directionality_t;
1053
1054 /* A 3D point which could be used to represent geometric location
1055 * or orientation of a microphone.
1056 */
1057 struct audio_microphone_coordinate {
1058 float x;
1059 float y;
1060 float z;
1061 };
1062
1063 /* An number to indicate which group the microphone locate. Main body is
1064 * usually group 0. Developer could use this value to group the microphones
1065 * that locate on the same peripheral or attachments.
1066 */
1067 typedef int audio_microphone_group_t;
1068
1069 typedef enum {
1070 AUDIO_MICROPHONE_CHANNEL_MAPPING_UNUSED = 0,
1071 AUDIO_MICROPHONE_CHANNEL_MAPPING_DIRECT = 1,
1072 AUDIO_MICROPHONE_CHANNEL_MAPPING_PROCESSED = 2,
1073 AUDIO_MICROPHONE_CHANNEL_MAPPING_CNT = 3,
1074 } audio_microphone_channel_mapping_t;
1075
1076 /* the maximum length for the microphone id */
1077 #define AUDIO_MICROPHONE_ID_MAX_LEN 32
1078 /* max number of frequency responses in a frequency response table */
1079 #define AUDIO_MICROPHONE_MAX_FREQUENCY_RESPONSES 256
1080 /* max number of microphone */
1081 #define AUDIO_MICROPHONE_MAX_COUNT 32
1082 /* the value of unknown spl */
1083 #define AUDIO_MICROPHONE_SPL_UNKNOWN -FLT_MAX
1084 /* the value of unknown sensitivity */
1085 #define AUDIO_MICROPHONE_SENSITIVITY_UNKNOWN -FLT_MAX
1086 /* the value of unknown coordinate */
1087 #define AUDIO_MICROPHONE_COORDINATE_UNKNOWN -FLT_MAX
1088 /* the value used as address when the address of bottom microphone is empty */
1089 #define AUDIO_BOTTOM_MICROPHONE_ADDRESS "bottom"
1090 /* the value used as address when the address of back microphone is empty */
1091 #define AUDIO_BACK_MICROPHONE_ADDRESS "back"
1092
1093 struct audio_microphone_characteristic_t {
1094 char device_id[AUDIO_MICROPHONE_ID_MAX_LEN];
1095 audio_port_handle_t id;
1096 audio_devices_t device;
1097 char address[AUDIO_DEVICE_MAX_ADDRESS_LEN];
1098 audio_microphone_channel_mapping_t channel_mapping[AUDIO_CHANNEL_COUNT_MAX];
1099 audio_microphone_location_t location;
1100 audio_microphone_group_t group;
1101 unsigned int index_in_the_group;
1102 float sensitivity;
1103 float max_spl;
1104 float min_spl;
1105 audio_microphone_directionality_t directionality;
1106 unsigned int num_frequency_responses;
1107 float frequency_responses[2][AUDIO_MICROPHONE_MAX_FREQUENCY_RESPONSES];
1108 struct audio_microphone_coordinate geometric_location;
1109 struct audio_microphone_coordinate orientation;
1110 };
1111
1112 __END_DECLS
1113
1114 /**
1115 * List of known audio HAL modules. This is the base name of the audio HAL
1116 * library composed of the "audio." prefix, one of the base names below and
1117 * a suffix specific to the device.
1118 * e.g: audio.primary.goldfish.so or audio.a2dp.default.so
1119 *
1120 * The same module names are used in audio policy configuration files.
1121 */
1122
1123 #define AUDIO_HARDWARE_MODULE_ID_PRIMARY "primary"
1124 #define AUDIO_HARDWARE_MODULE_ID_A2DP "a2dp"
1125 #define AUDIO_HARDWARE_MODULE_ID_USB "usb"
1126 #define AUDIO_HARDWARE_MODULE_ID_REMOTE_SUBMIX "r_submix"
1127 #define AUDIO_HARDWARE_MODULE_ID_CODEC_OFFLOAD "codec_offload"
1128 #define AUDIO_HARDWARE_MODULE_ID_STUB "stub"
1129 #define AUDIO_HARDWARE_MODULE_ID_HEARING_AID "hearing_aid"
1130
1131 /**
1132 * Multi-Stream Decoder (MSD) HAL service name. MSD HAL is used to mix
1133 * encoded streams together with PCM streams, producing re-encoded
1134 * streams or PCM streams.
1135 *
1136 * The service must register itself using this name, and audioserver
1137 * tries to instantiate a device factory using this name as well.
1138 * Note that the HIDL implementation library file name *must* have the
1139 * suffix "msd" in order to be picked up by HIDL that is:
1140 *
1141 * android.hardware.audio@x.x-implmsd.so
1142 */
1143 #define AUDIO_HAL_SERVICE_NAME_MSD "msd"
1144
1145 /**
1146 * Parameter definitions.
1147 * Note that in the framework code it's recommended to use AudioParameter.h
1148 * instead of these preprocessor defines, and for sure avoid just copying
1149 * the constant values.
1150 */
1151
1152 #define AUDIO_PARAMETER_VALUE_ON "on"
1153 #define AUDIO_PARAMETER_VALUE_OFF "off"
1154
1155 /**
1156 * audio device parameters
1157 */
1158
1159 /* BT SCO Noise Reduction + Echo Cancellation parameters */
1160 #define AUDIO_PARAMETER_KEY_BT_NREC "bt_headset_nrec"
1161
1162 /* Get a new HW synchronization source identifier.
1163 * Return a valid source (positive integer) or AUDIO_HW_SYNC_INVALID if an error occurs
1164 * or no HW sync is available. */
1165 #define AUDIO_PARAMETER_HW_AV_SYNC "hw_av_sync"
1166
1167 /* Screen state */
1168 #define AUDIO_PARAMETER_KEY_SCREEN_STATE "screen_state"
1169
1170 /**
1171 * audio stream parameters
1172 */
1173
1174 #define AUDIO_PARAMETER_STREAM_ROUTING "routing" /* audio_devices_t */
1175 #define AUDIO_PARAMETER_STREAM_FORMAT "format" /* audio_format_t */
1176 #define AUDIO_PARAMETER_STREAM_CHANNELS "channels" /* audio_channel_mask_t */
1177 #define AUDIO_PARAMETER_STREAM_FRAME_COUNT "frame_count" /* size_t */
1178 #define AUDIO_PARAMETER_STREAM_INPUT_SOURCE "input_source" /* audio_source_t */
1179 #define AUDIO_PARAMETER_STREAM_SAMPLING_RATE "sampling_rate" /* uint32_t */
1180
1181 /* Request the presentation id to be decoded by a next gen audio decoder */
1182 #define AUDIO_PARAMETER_STREAM_PRESENTATION_ID "presentation_id" /* int32_t */
1183
1184 /* Request the program id to be decoded by a next gen audio decoder */
1185 #define AUDIO_PARAMETER_STREAM_PROGRAM_ID "program_id" /* int32_t */
1186
1187 #define AUDIO_PARAMETER_DEVICE_CONNECT "connect" /* audio_devices_t */
1188 #define AUDIO_PARAMETER_DEVICE_DISCONNECT "disconnect" /* audio_devices_t */
1189
1190 /* Enable mono audio playback if 1, else should be 0. */
1191 #define AUDIO_PARAMETER_MONO_OUTPUT "mono_output"
1192
1193 /* Set the HW synchronization source for an output stream. */
1194 #define AUDIO_PARAMETER_STREAM_HW_AV_SYNC "hw_av_sync"
1195
1196 /* Query supported formats. The response is a '|' separated list of strings from
1197 * audio_format_t enum e.g: "sup_formats=AUDIO_FORMAT_PCM_16_BIT" */
1198 #define AUDIO_PARAMETER_STREAM_SUP_FORMATS "sup_formats"
1199 /* Query supported channel masks. The response is a '|' separated list of strings from
1200 * audio_channel_mask_t enum e.g: "sup_channels=AUDIO_CHANNEL_OUT_STEREO|AUDIO_CHANNEL_OUT_MONO" */
1201 #define AUDIO_PARAMETER_STREAM_SUP_CHANNELS "sup_channels"
1202 /* Query supported sampling rates. The response is a '|' separated list of integer values e.g:
1203 * "sup_sampling_rates=44100|48000" */
1204 #define AUDIO_PARAMETER_STREAM_SUP_SAMPLING_RATES "sup_sampling_rates"
1205
1206 #define AUDIO_PARAMETER_VALUE_LIST_SEPARATOR "|"
1207
1208 /* Reconfigure offloaded A2DP codec */
1209 #define AUDIO_PARAMETER_RECONFIG_A2DP "reconfigA2dp"
1210 /* Query if HwModule supports reconfiguration of offloaded A2DP codec */
1211 #define AUDIO_PARAMETER_A2DP_RECONFIG_SUPPORTED "isReconfigA2dpSupported"
1212
1213 /**
1214 * audio codec parameters
1215 */
1216
1217 #define AUDIO_OFFLOAD_CODEC_PARAMS "music_offload_codec_param"
1218 #define AUDIO_OFFLOAD_CODEC_BIT_PER_SAMPLE "music_offload_bit_per_sample"
1219 #define AUDIO_OFFLOAD_CODEC_BIT_RATE "music_offload_bit_rate"
1220 #define AUDIO_OFFLOAD_CODEC_AVG_BIT_RATE "music_offload_avg_bit_rate"
1221 #define AUDIO_OFFLOAD_CODEC_ID "music_offload_codec_id"
1222 #define AUDIO_OFFLOAD_CODEC_BLOCK_ALIGN "music_offload_block_align"
1223 #define AUDIO_OFFLOAD_CODEC_SAMPLE_RATE "music_offload_sample_rate"
1224 #define AUDIO_OFFLOAD_CODEC_ENCODE_OPTION "music_offload_encode_option"
1225 #define AUDIO_OFFLOAD_CODEC_NUM_CHANNEL "music_offload_num_channels"
1226 #define AUDIO_OFFLOAD_CODEC_DOWN_SAMPLING "music_offload_down_sampling"
1227 #define AUDIO_OFFLOAD_CODEC_DELAY_SAMPLES "delay_samples"
1228 #define AUDIO_OFFLOAD_CODEC_PADDING_SAMPLES "padding_samples"
1229
1230 #endif // ANDROID_AUDIO_CORE_H
1231