audio - interface between low and high level audio drivers
The audio device driver is divided into a high level, hardware independent
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,
struct audio_params *, struct audio_params *);
int (*round_blocksize)(void *, int);
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, int, int);
void (*freem)(void *, void *, int);
size_t (*round_buffersize)(void *, int, size_t);
int (*mappage)(void *, void *, int, int);
int (*get_props)(void *);
int (*trigger_output)(void *, void *, void *, int,
void (*)(void *), void *, struct audio_params *);
int (*trigger_input)(void *, void *, void *, int,
void (*)(void *), void *, struct audio_params *);
int (*dev_ioctl)(void *, u_long, caddr_t, int, struct proc *);
};
struct audio_params {
u_long sample_rate; /* sample rate */
u_int encoding; /* ulaw, linear, etc */
u_int precision; /* bits/sample */
u_int channels; /* mono(1), stereo(2) */
/* Software en/decode functions, set if SW coding required by HW */
void (*sw_code)(void *, u_char *, int);
int factor; /* coding space change */
int factor_denom; /* denominator of factor */
/*
* The following four members represent what format is used in a
* hardware. If hw_sample_rate != sample_rate || hw_channels !=
* channels, the audio framework converts data. Encoding and
* precision are converted in sw_code().
* set_params() should set correct values to them if no conversion is
* needed.
*/
u_long hw_sample_rate;
u_int hw_encoding;
u_int hw_precision;
u_int hw_channels;
};
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 processes
in these. The data is presented to the lower level in smaller
chunks, called blocks. If there, during playback, 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)
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)
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,
struct audio_params *play, struct audio_params *rec)
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.
If the hardware requires software assistance with some encoding
(e.g., it might be lacking mulaw support) it should fill the
sw_code, factor and factor_denom fields of these structures with
translation information, and set to hw_* fields what format the
hardware actually uses. For example, if play requests [8000Hz,
mulaw, 8bit, 1ch] and the hardware supports not 8bit mulaw but
16bit slinear_le, the driver should set mulaw_to_slinear16_le to
sw_code, 2 to factor, 1 to factor_denom,
AUDIO_ENCODING_SLINEAR_LE to hw_encoding and 16 to hw_precision.
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).
The hardware driver can also request sampling rate conversion and
mono-stereo conversion. If set_params sets to hw_sampling_rate a
value which is different than sampling_rate or sets to
hw_channels a value which is different than channels, and set
AUDIO_ENCODING_SLINEAR_LE to hw_encoding, the hardware independent
driver performs sampling rate and/or mono-stereo conversion.
If such conversion is not needed, set_params must keep
sampling_rate and channels are the same as hw_sampling_rate and
hw_channels respectively.
Note: The order of conversion is sw_code followed by sampling
rate and mono-stereo in playing, and sampling rate and monostereo
followed by sw_code in recording.
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)
optional, is called with the block size, bs, that has been computed
by the upper layer. It should return a block size, possibly
changed according to the needs of the hardware 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 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)
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, otherwise
an error code.
int set_port(void *hdl, mixer_ctl_t *mc)
is called in when AUDIO_MIXER_WRITE is used. It should take data
from the mixer_ctl_t struct at set the corresponding mixer values.
Return 0 on success, otherwise an error code.
int get_port(void *hdl, mixer_ctl_t *mc)
is called in when AUDIO_MIXER_READ is used. It should fill the
mixer_ctl_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, int 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, int 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.
int mappage(void *hdl, void *addr, int 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,
struct audio_params *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,
struct audio_params *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, caddr_t addr,
int flag, struct proc *p)
optional, is called when an ioctl(2) is not recognized by the
generic audio driver. Return 0 on success, otherwise an error
code.
The query_devinfo method should define certain mixer controls for
AUDIO_SETINFO to be able to change the port and gain.
If the audio hardware 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. For each of the named sources there should be a
control in the AudioCinputs class of type AUDIO_MIXER_VALUE if recording
level of the source can be set. If the overall recording level can be
changed (i.e. regardless of the input source) then this control should be
named AudioNrecord and be of class AudioCinputs.
If the audio hardware is capable of output to more than one destination
it should define AudioNoutput in class AudioCmonitor. 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 there should
be a control in the AudioCoutputs class of type AUDIO_MIXER_VALUE if output
level of the destination can be set. If the overall output level can
be changed (i.e. regardless of the destination) then this control should
be named AudioNmaster and be of class AudioCoutputs.
audio(4)
This audio interface first appeared in NetBSD 1.3.
BSD March 11, 2002 BSD
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