audio(9)
- NetBSD Manual Pages
AUDIO(9) NetBSD Kernel Developer's Manual AUDIO(9)
NAME
audio -- interface between low and high level audio drivers
DESCRIPTION
The audio device driver is divided into a high level, hardware indepen-
dent layer, and a low level hardware dependent layer. The interface
between these is the audio_hw_if structure.
struct audio_hw_if {
int (*open)(void *, int);
void (*close)(void *);
int (*drain)(void *);
int (*query_encoding)(void *, struct audio_encoding *);
int (*set_params)(void *, int, int,
audio_params_t *, audio_params_t *,
stream_filter_list_t *, stream_filter_list_t *);
int (*round_blocksize)(void *, int, int, const audio_params_t *);
int (*commit_settings)(void *);
int (*init_output)(void *, void *, int);
int (*init_input)(void *, void *, int);
int (*start_output)(void *, void *, int, void (*)(void *),
void *);
int (*start_input)(void *, void *, int, void (*)(void *),
void *);
int (*halt_output)(void *);
int (*halt_input)(void *);
int (*speaker_ctl)(void *, int);
#define SPKR_ON 1
#define SPKR_OFF 0
int (*getdev)(void *, struct audio_device *);
int (*setfd)(void *, int);
int (*set_port)(void *, mixer_ctrl_t *);
int (*get_port)(void *, mixer_ctrl_t *);
int (*query_devinfo)(void *, mixer_devinfo_t *);
void *(*allocm)(void *, int, size_t, struct malloc_type *, int);
void (*freem)(void *, void *, struct malloc_type *);
size_t (*round_buffersize)(void *, int, size_t);
paddr_t (*mappage)(void *, void *, off_t, int);
int (*get_props)(void *);
int (*trigger_output)(void *, void *, void *, int,
void (*)(void *), void *, const audio_params_t *);
int (*trigger_input)(void *, void *, void *, int,
void (*)(void *), void *, const audio_params_t *);
int (*dev_ioctl)(void *, u_long, void *, int, struct lwp *);
void (*get_locks)(void *, kmutex_t **, kmutex_t **);
};
typedef struct audio_params {
u_int sample_rate; /* sample rate */
u_int encoding; /* e.g. mu-law, linear, etc */
u_int precision; /* bits/subframe */
u_int validbits; /* valid bits in a subframe */
u_int channels; /* mono(1), stereo(2) */
} audio_params_t;
The high level audio driver attaches to the low level driver when the
latter calls audio_attach_mi. This call should be
void
audio_attach_mi(ahwp, hdl, dev)
struct audio_hw_if *ahwp;
void *hdl;
struct device *dev;
The audio_hw_if struct is as shown above. The hdl argument is a handle
to some low level data structure. It is sent as the first argument to
all the functions in audio_hw_if when the high level driver calls them.
dev is the device struct for the hardware device.
The upper layer of the audio driver allocates one buffer for playing and
one for recording. It handles the buffering of data from the user pro-
cesses in these. The data is presented to the lower level in smaller
chunks, called blocks. If, during playback, there is no data available
from the user process when the hardware request another block a block of
silence will be used instead. Furthermore, if the user process does not
read data quickly enough during recording data will be thrown away.
The fields of audio_hw_if are described in some more detail below. Some
fields are optional and can be set to 0 if not needed.
int open(void *hdl, int flags)
optional, is called when the audio device is opened. It should
initialize the hardware for I/O. Every successful call to open
is matched by a call to close. Return 0 on success, otherwise an
error code.
void close(void *hdl)
optional, is called when the audio device is closed.
int drain(void *hdl)
optional, is called before the device is closed or when
AUDIO_DRAIN is called. It should make sure that no samples
remain in to be played that could be lost when close is called.
Return 0 on success, otherwise an error code.
int query_encoding(void *hdl, struct audio_encoding *ae)
is used when AUDIO_GETENC is called. It should fill the
audio_encoding structure and return 0 or, if there is no encoding
with the given number, return EINVAL.
int set_params(void *hdl, int setmode, int usemode,
audio_params_t *play, audio_params_t *rec,
stream_filter_list_t *pfil, stream_filter_list_t *rfil)
Called to set the audio encoding mode. setmode is a combination
of the AUMODE_RECORD and AUMODE_PLAY flags to indicate which
mode(s) are to be set. usemode is also a combination of these
flags, but indicates the current mode of the device (i.e., the
value of mode in the audio_info struct).
The play and rec structures contain the encoding parameters that
should be set. The values of the structures may also be modified
if the hardware cannot be set to exactly the requested mode
(e.g., if the requested sampling rate is not supported, but one
close enough is).
If the hardware requires software assistance with some encoding
(e.g., it might be lacking mu-law support) it should fill the
pfil for playing or rfil for recording with conversion informa-
tion. For example, if play requests [8000Hz, mu-law, 8/8bit,
1ch] and the hardware does not support 8bit mu-law, but 16bit
slinear_le, the driver should call pfil->append() with pfil,
mulaw_to_linear16, and audio_params_t representing [8000Hz, slin-
ear_le, 16/16bit, 2ch]. If the driver needs multiple conver-
sions, a conversion nearest to the hardware should be set to the
head of pfil or rfil. The definition of stream_filter_list_t
follows:
typedef struct stream_filter_list {
void (*append)(struct stream_filter_list *,
stream_filter_factory_t,
const audio_params_t *);
void (*prepend)(struct stream_filter_list *,
stream_filter_factory_t,
const audio_params_t *);
void (*set)(struct stream_filter_list *, int,
stream_filter_factory_t,
const audio_params_t *);
int req_size;
struct stream_filter_req {
stream_filter_factory_t *factory;
audio_params_t param; /* from-param for recording,
to-param for playing */
} filters[AUDIO_MAX_FILTERS];
} stream_filter_list_t;
For playing, pfil constructs conversions as follows:
(play) == write(2) input
| pfil->filters[pfil->req_size-1].factory
(pfil->filters[pfil->req_size-1].param)
| pfil->filters[pfil->req_size-2].factory
:
| pfil->filters[1].factory
(pfil->filters[1].param)
| pfil->filters[0].factory
(pfil->filters[0].param) == hardware input
For recording, rfil constructs conversions as follows:
(rfil->filters[0].param) == hardware output
| rfil->filters[0].factory
(rfil->filters[1].param)
| rfil->filters[1].factory
:
| rfil->filters[rfil->req_size-2].factory
(rfil->filters[rfil->req_size-1].param)
| rfil->filters[rfil->req_size-1].factory
(rec) == read(2) output
If the device does not have the AUDIO_PROP_INDEPENDENT property
the same value is passed in both play and rec and the encoding
parameters from play is copied into rec after the call to
set_params. Return 0 on success, otherwise an error code.
int round_blocksize(void *hdl, int bs, int mode,
const audio_params_t *param)
optional, is called with the block size, bs, that has been com-
puted by the upper layer, mode, AUMODE_PLAY or AUMODE_RECORD, and
param, encoding parameters for the hardware. It should return a
block size, possibly changed according to the needs of the hard-
ware driver.
int commit_settings(void *hdl)
optional, is called after all calls to set_params, and set_port,
are done. A hardware driver that needs to get the hardware in
and out of command mode for each change can save all the changes
during previous calls and do them all here. Return 0 on success,
otherwise an error code.
int init_output(void *hdl, void *buffer, int size)
optional, is called before any output starts, but when the total
size of the output buffer has been determined. It can be used to
initialize looping DMA for hardware that needs that. Return 0 on
success, otherwise an error code.
int init_input(void *hdl, void *buffer, int size)
optional, is called before any input starts, but when the total
size of the input buffer has been determined. It can be used to
initialize looping DMA for hardware that needs that. Return 0 on
success, otherwise an error code.
int start_output(void *hdl, void *block, int blksize,
void (*intr)(void*), void *intrarg)
is called to start the transfer of blksize bytes from block to
the audio hardware. The call should return when the data trans-
fer has been initiated (normally with DMA). When the hardware is
ready to accept more samples the function intr should be called
with the argument intrarg. Calling intr will normally initiate
another call to start_output. Return 0 on success, otherwise an
error code.
int start_input(void *hdl, void *block, int blksize,
void (*intr)(void*), void *intrarg)
is called to start the transfer of blksize bytes to block from
the audio hardware. The call should return when the data trans-
fer has been initiated (normally with DMA). When the hardware is
ready to deliver more samples the function intr should be called
with the argument intrarg. Calling intr will normally initiate
another call to start_input. Return 0 on success, otherwise an
error code.
int halt_output(void *hdl)
is called to abort the output transfer (started by start_output)
in progress. Return 0 on success, otherwise an error code.
int halt_input(void *hdl)
is called to abort the input transfer (started by start_input) in
progress. Return 0 on success, otherwise an error code.
int speaker_ctl(void *hdl, int on)
optional, is called when a half duplex device changes between
playing and recording. It can, e.g., be used to turn on and off
the speaker. Return 0 on success, otherwise an error code.
int getdev(void *hdl, struct audio_device *ret)
Should fill the audio_device struct with relevant information
about the driver. Return 0 on success, otherwise an error code.
int setfd(void *hdl, int fd)
optional, is called when AUDIO_SETFD is used, but only if the
device has AUDIO_PROP_FULLDUPLEX set. Return 0 on success, oth-
erwise an error code.
int set_port(void *hdl, mixer_ctrl_t *mc)
is called in when AUDIO_MIXER_WRITE is used. It should take data
from the mixer_ctrl_t struct at set the corresponding mixer val-
ues. Return 0 on success, otherwise an error code.
int get_port(void *hdl, mixer_ctrl_t *mc)
is called in when AUDIO_MIXER_READ is used. It should fill the
mixer_ctrl_t struct. Return 0 on success, otherwise an error
code.
int query_devinfo(void *hdl, mixer_devinfo_t *di)
is called in when AUDIO_MIXER_DEVINFO is used. It should fill
the mixer_devinfo_t struct. Return 0 on success, otherwise an
error code.
void *allocm(void *hdl, int direction, size_t size, struct malloc_type
*type, int flags)
optional, is called to allocate the device buffers. If not
present malloc(9) is used instead (with the same arguments but
the first two). The reason for using a device dependent routine
instead of malloc(9) is that some buses need special allocation
to do DMA. Returns the address of the buffer, or 0 on failure.
void freem(void *hdl, void *addr, struct malloc_type *type)
optional, is called to free memory allocated by alloc. If not
supplied free(9) is used.
size_t round_buffersize(void *hdl, int direction, size_t bufsize)
optional, is called at startup to determine the audio buffer
size. The upper layer supplies the suggested size in bufsize,
which the hardware driver can then change if needed. E.g., DMA
on the ISA bus cannot exceed 65536 bytes.
paddr_t mappage(void *hdl, void *addr, off_t offs, int prot)
optional, is called for mmap(2). Should return the map value for
the page at offset offs from address addr mapped with protection
prot. Returns -1 on failure, or a machine dependent opaque value
on success.
int get_props(void *hdl)
Should return the device properties; i.e., a combination of
AUDIO_PROP_xxx.
int trigger_output(void *hdl, void *start, void *end,
int blksize, void (*intr)(void*), void *intrarg,
const audio_params_t *param)
optional, is called to start the transfer of data from the circu-
lar buffer delimited by start and end to the audio hardware,
parameterized as in param. The call should return when the data
transfer has been initiated (normally with DMA). When the hard-
ware is finished transferring each blksize sized block, the func-
tion intr should be called with the argument intrarg (typically
from the audio hardware interrupt service routine). Once started
the transfer may be stopped using halt_output. Return 0 on suc-
cess, otherwise an error code.
int trigger_input(void *hdl, void *start, void *end,
int blksize, void (*intr)(void*), void *intrarg,
const audio_params_t *param)
optional, is called to start the transfer of data from the audio
hardware, parameterized as in param, to the circular buffer
delimited by start and end. The call should return when the data
transfer has been initiated (normally with DMA). When the hard-
ware is finished transferring each blksize sized block, the func-
tion intr should be called with the argument intrarg (typically
from the audio hardware interrupt service routine). Once started
the transfer may be stopped using halt_input. Return 0 on suc-
cess, otherwise an error code.
int dev_ioctl(void *hdl, u_long cmd, void *addr,
int flag, struct lwp *l)
optional, is called when an ioctl(2) is not recognized by the
generic audio driver. Return 0 on success, otherwise an error
code.
void get_locks(void *hdl, kmutex_t **intr, kmutex_t **thread)
Returns the interrupt and thread locks to the common audio layer.
The query_devinfo method should define certain mixer controls for
AUDIO_SETINFO to be able to change the port and gain, and AUDIO_GETINFO
to read them, as follows.
If the record mixer is capable of input from more than one source, it
should define AudioNsource in class AudioCrecord. This mixer control
should be of type AUDIO_MIXER_ENUM or AUDIO_MIXER_SET and enumerate the
possible input sources. Each of the named sources for which the record-
ing level can be set should have a control in the AudioCrecord class of
type AUDIO_MIXER_VALUE, except the "mixerout" source is special, and will
never have its own control. Its selection signifies, rather, that vari-
ous sources in class AudioCrecord will be combined and presented to the
single recording output in the same fashion that the sources of class
AudioCinputs are combined and presented to the playback output(s). If
the overall recording level can be changed, regardless of the input
source, then this control should be named AudioNmaster and be of class
AudioCrecord.
Controls for various sources that affect only the playback output, as
opposed to recording, should be in the AudioCinputs class, as of course
should any controls that affect both playback and recording.
If the play mixer is capable of output to more than one destination, it
should define AudioNselect in class AudioCoutputs. This mixer control
should be of type AUDIO_MIXER_ENUM or AUDIO_MIXER_SET and enumerate the
possible destinations. For each of the named destinations for which the
output level can be set, there should be a control in the AudioCoutputs
class of type AUDIO_MIXER_VALUE. If the overall output level can be
changed, which is invariably the case, then this control should be named
AudioNmaster and be of class AudioCoutputs.
There's one additional source recognized specially by AUDIO_SETINFO and
AUDIO_GETINFO, to be presented as monitor_gain, and that is a control
named AudioNmonitor, of class AudioCmonitor.
SEE ALSO
audio(4), audio_system(9)
HISTORY
This audio interface first appeared in NetBSD 1.3.
NetBSD 8.0 May 15, 2018 NetBSD 8.0
Powered by man-cgi (2024-03-20).
Maintained for NetBSD
by Kimmo Suominen.
Based on man-cgi by Panagiotis Christias.