Android Hardware Abstraction Layer
audio.h
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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 };
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 };
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 */
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 */
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 
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 };
381 
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 };
409 
410 /**
411  * return the frame size (number of bytes per sample).
412  */
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 {
436 };
437 
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 */
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 
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 };
548 
549 /** convenience API for opening and closing a supported device */
550 
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 
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