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AUDIO(9) 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
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 (*query_format)(void *, audio_format_query_t *);
int (*set_format)(void *, int,
const audio_params_t *, const audio_params_t *,
audio_filter_reg_t *, audio_filter_reg_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 (*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);
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 phase that these functions are called is classified into three.
Attach phase, Closed phase and Opened phase. Attach phase is during
device attach and it transits to the Closed phase when the attach
succeeded. Closed phase is when no sampling device is opened and it
transits to the Opened phase when open succeeded. Opened phase is when
any sampling device is opened and it transits to the Closed phase when
close succeeded.
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)
optional, is called when the first device combining playback and
recording is opened. On a full duplex hardware, (FREAD | FWRITE)
is passed to flags. On a half duplex hardware, FWRITE is passed
for playback, or FREAD for recording. Every successful call to
open is matched by a call to close. Return 0 on success,
otherwise an error code. It is called in the Closed phase.
void close(void *hdl)
optional, is called when the last audio device combining playback
and recording is closed. Before call to this, halt_input and
halt_output are called if necessary. It is called in the Opened
phase.
int query_format(void *hdl, audio_format_query_t *afp)
is called to enumerate formats supported by the hardware. It
should fill the audio_format_t structure according to given
number afp->index. If there is no format with the given number,
return EINVAL. It can be called at any time.
typedef struct audio_format_query {
u_int index;
struct audio_format fmt;
} audio_format_query_t;
It is also used by the upper layer to determine the default
format, as follows:
1. Higher priority is preferred (normally 0, the highest is 3,
the lowest is 0).
2. AUDIO_ENCODING_SLINEAR_NE:16 is preferred if exists.
3. AUDIO_ENCODING_SLINEAR_OE:16 is preferred if exists.
4. The format with more channels is preferred.
If the driver supports SLINEAR_NE:16 and the upper layer chooses
it, the driver does not need to provide a conversion function in
set_format. Similarly, if the driver supports SLINEAR_OE:16 and
the upper layer chooses it, the driver does not need to provide a
conversion function, because the upper layer supports conversion
between SLINEAR_NE:16 and SLINEAR_OE:16 for convenience. If the
upper layer chooses another format, the driver needs to provide a
conversion function in set_format. See also set_format. If the
driver can not provide the conversion from/to SLINEAR_NE:16, set
priority to -1. It means that the hardware supports this format
but the driver does not (e.g. AC3), and it will never be chosen.
int set_format(void *hdl, int setmode,
const audio_params_t *play, const audio_params_t *rec,
audio_filter_reg_t *pfil, audio_filter_reg_t *rfil)
is called to set specified format to the hardware, when the
device is attached or the hardware format is changed. setmode is
a combination of the AUMODE_RECORD and AUMODE_PLAY flags to
indicate which modes are to be set.
The play and rec structures contain the encoding parameters that
should be set to the hardware. All of these parameters are
chosen from formats returned by query_format. Therefore play
and/or rec are always settable. If the hardware does not support
AUDIO_ENCODING_SLINEAR_{NE,OE}:16, conversion information should
be filled the pfil for playing or rfil for recording. The
definition of audio_filter_reg_t and a related structure follow:
typedef struct {
const void *src;
const audio_format2_t *srcfmt;
void *dst;
const audio_format2_t *dstfmt;
int count;
void *context;
} audio_filter_arg_t;
typedef void(*audio_filter_t)(audio_filter_arg_t *arg);
typedef struct {
audio_filter_t codec;
void *context;
} audio_filter_reg_t;
codec is a conversion function and context is an optional opaque
pointer passed to codec.
When codec is called, all parameters required by codec are
contained in arg. src points to the input buffer block, srcfmt
contains the input encoding parameters, dst points to the output
buffer block and dstfmt contains the output encoding parameters.
count represents the number of frames to process on this call.
src and dst are guaranteed to be able to consecutively access
number of frames specified by count. codec must fill the entirety
of dst. For example, let count = 100, srcfmt is { precision =
16, channels = 3 }, dstfmt is { precision = 8, channels = 4 }, in
this case, src block length = 2(bytes) * 3(channels) *
100(frames) = 600 bytes, The length to be written to dst block is
1(byte) * 4(channels) * 100(frames) = 400 bytes. codec cannot
abort the conversion halfway and there is no error reporting
mechanism. context is a opaque pointer that can be used by codec
if necessary.
If the device does not have the AUDIO_PROP_INDEPENDENT property
the same value is passed in both play and rec. Returns 0 on
success, otherwise an error code. It is called in the Attach or
Closed phases.
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. bs passed is
always non-zero and a multiple of the frame size represented by
param->channels * param->precision / 8. It should return a block
size, possibly changed according to the needs of the hardware
driver. The return value also must be non-zero and a multiple of
the frame size. It is called in the Attach or Closed phases.
int commit_settings(void *hdl)
optional, is called after all calls to set_format, 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. Returns 0 on
success, otherwise an error code. It is called in the Attach or
Closed phases.
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. It is called in the Attach or
Closed phases.
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. Returns 0
on success, otherwise an error code. It is called in the Attach
or Closed phases.
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. Returns 0 on success,
otherwise an error code. This field is optional only if the
driver doesn't support playback. It is called in the Opened
phase.
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. Returns 0 on
success, otherwise an error code. This field is optional only if
the driver doesn't support recording. It is called in the Opened
phase.
int halt_output(void *hdl)
is called to abort the output transfer (started by start_output)
in progress. Returns 0 on success, otherwise an error code.
This field is optional only if the driver doesn't support
playback. It is called in the Opened phase.
int halt_input(void *hdl)
is called to abort the input transfer (started by start_input) in
progress. Returns 0 on success, otherwise an error code. This
field is optional only if the driver doesn't support recording,
It is called in the Opened phase.
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. Returns 0 on success, otherwise an error code. It
is called in the Opened phase.
int getdev(void *hdl, struct audio_device *ret)
Should fill the audio_device struct with relevant information
about the driver. Returns 0 on success, otherwise an error code.
It is called in the Opened phase.
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 and set the corresponding mixer
values. Returns 0 on success, otherwise an error code. It is
called in the Opened or Closed phases.
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. Returns 0 on success, otherwise an error
code. It is called in the Opened or Closed phases.
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. It is called at any time.
void *allocm(void *hdl, int direction, size_t size)
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 NULL on
failure. It is called in the Attached or Closed phases.
void freem(void *hdl, void *addr, size_t size)
optional, is called to free memory allocated by allocm. If not
supplied free(9) is used. It is called in the Attached or Closed
phases.
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. It is called in the
Attached or Closed phases.
int get_props(void *hdl)
Should return the device properties in a combination of following
flags:
AUDIO_PROP_PLAYBACK the device is capable of audio playback.
AUDIO_PROP_CAPTURE the device is capable of audio capture.
AUDIO_PROP_FULLDUPLEX the device admits full duplex operation.
Don't set it if the device is
unidirectional.
AUDIO_PROP_INDEPENDENT the device can set the playing and
recording encoding parameters
independently. Don't set it if the
device is unidirectional.
AUDIO_PROP_MMAP is handled in the upper layer, so new
drivers should not return this property.
It is called in the Attach phase.
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. It is called in
the Opened phase.
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. It is called in
the Opened phase.
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. It is called in the Opened phase.
void get_locks(void *hdl, kmutex_t **intr, kmutex_t **thread)
Returns the interrupt and thread locks to the common audio layer.
It is called in the Attach phase.
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.
SEE ALSO
audio(4)
HISTORY
This audio interface first appeared in NetBSD 1.3.
NetBSD 10.99 February 2, 2021 NetBSD 10.99