AUDIO(9) | Kernel Developer's Manual | AUDIO(9) |
audio
—
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); void (*freem)(void *, void *, size_t); 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
device_t audio_attach_mi(const struct audio_hw_if *ahwp, void *hdl, device_t 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 processes 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
NULL
if not needed.
int
open(void *hdl, int flags)
void
close(void *hdl)
int
drain(void *hdl)
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)
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 information. 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, slinear_le, 16/16bit, 2ch]. If the
driver needs multiple conversions, 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 computed 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 hardware driver.
int
commit_settings(void *hdl)
int
init_output(void *hdl, void *buffer, int size)
int
init_input(void *hdl, void *buffer, int size)
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 transfer 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 transfer 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)
int
halt_input(void *hdl)
int
speaker_ctl(void *hdl, int on)
int
getdev(void *hdl, struct audio_device *ret)
int
setfd(void *hdl, int fd)
AUDIO_SETFD
is used, but
only if the device has AUDIO_PROP_FULLDUPLEX set. Return 0 on success,
otherwise an error code.int
set_port(void *hdl, mixer_ctrl_t *mc)
AUDIO_MIXER_WRITE
is used. It
should take data from the mixer_ctrl_t struct at set
the corresponding mixer values. Return 0 on success, otherwise an error
code.int
get_port(void *hdl, mixer_ctrl_t *mc)
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)
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)
NULL
on failure.void
freem(void *hdl, void *addr, size_t size)
size_t
round_buffersize(void *hdl, int direction, size_t bufsize)
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)
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 circular 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 hardware is finished transferring each blksize sized block, the function 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 success, 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 hardware is finished transferring each blksize sized block, the function 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 success, 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)
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 recording 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 various 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
.
audio
interface first appeared in
NetBSD 1.3.
May 15, 2018 | NetBSD 8.99 |