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)

HISTORY
     This audio interface first appeared in NetBSD 1.3.

NetBSD 6.0                       July 13, 2014                      NetBSD 6.0

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©1996-2014 Modified for NetBSD by Kimmo Suominen