MirOS Manual: audio(9)

AUDIO(9)                      BSD Kernel 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,
                         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 *, struct mixer_ctrl *);
             int     (*get_port)(void *, struct mixer_ctrl *);

             int     (*query_devinfo)(void *, struct mixer_devinfo *);

             void    *(*allocm)(void *, int, size_t, int, int);
             void    (*freem)(void *, void *, int);
             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 *, struct audio_params *);
             int     (*trigger_input)(void *, void *, void *, int,
                         void (*)(void *), void *, struct audio_params *);
     };

     struct audio_params {
             u_long  sample_rate;            /* sample rate */
             u_int   encoding;               /* mu-law, 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 */
     };

     The high level audio driver attaches to the low level driver when the
     latter calls audio_attach_mi(). This call is:

           struct device *
           audio_attach_mi(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 ahwp 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. During playback, if there is no data available
     from the user process when the hardware requests another block, a block
     of silence will be used instead. Similarly, 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)
             This function is called when the audio device is opened, with
             flags the kernel representation of flags passed to the open(2)
             system call (see OFLAGS and FFLAGS in <sys/fcntl.h>). It initial-
             izes the hardware for I/O. Every successful call to open() is
             matched by a call to close(). This function returns 0 on success,
             otherwise an error code.

     void (*close)(void *hdl)
             This function is called when the audio device is closed.

     int (*drain)(void *hdl)
             This function is optional. If supplied, it is called before the
             device is closed or when the AUDIO_DRAIN ioctl(2) is called. It
             makes sure that no samples remain to be played that could be lost
             when close() is called. This function returns 0 on success, oth-
             erwise an error code.

     int (*query_encoding)(void *hdl, struct audio_encoding *ae)
             This function is used when the AUDIO_GETENC ioctl(2) is called.
             It fills ae and returns 0 or, if there is no encoding with the
             given number, returns EINVAL.

     int (*set_params)(void *hdl, int setmode, int usemode, struct
             audio_params *play, struct audio_params *rec)
             This function is 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 combina-
             tion 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 will be set.
             If the hardware requires software assistance with some encoding
             (e.g., it might be lacking mu-law support), it will fill the
             sw_code and factor fields of these structures. See
             /usr/src/sys/dev/auconv.h for available software support. The
             values of the structures may also be modified if the hardware
             cannot be set to exactly the requested mode (e.g., if the re-
             quested sampling rate is not supported, but one close enough is).
             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 are copied into rec after the call to
             set_params().

             The machine independent audio driver does some preliminary param-
             eter checking; it verifies that the precision is compatible with
             the encoding, and it translates AUDIO_ENCODING_[US]LINEAR to
             AUDIO_ENCODING_[US]LINEAR_{LE,BE}.

             This function returns 0 on success, otherwise an error code.

     int (*round_blocksize)(void *hdl, int bs)
             This function is optional. If supplied, it is called with the
             block size, bs, which has been computed by the upper layer. It
             returns a block size, possibly changed according to the needs of
             the hardware driver.

     int (*commit_settings)(void *hdl)
             This function is optional. If supplied, it 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. This function returns 0 on success, otherwise an
             error code.

     int (*init_output)(void *hdl, void *buffer, int size)
             This function is optional. If supplied, it is called before any
             output starts, but only after the total size of the output buffer
             has been determined. It can be used to initialize looping DMA for
             hardware that needs it. This function returns 0 on success, oth-
             erwise an error code.

     int (*init_input)(void *hdl, void *buffer, int size)
             This function is optional. If supplied, it is called before any
             input starts, but only after the total size of the input buffer
             has been determined. It can be used to initialize looping DMA for
             hardware that needs it. This function returns 0 on success, oth-
             erwise an error code.

     int (*start_output)(void *hdl, void *block, int bsize, void (*intr)(void
             *), void *intrarg)
             This function is called to start the transfer of bsize bytes from
             block to the audio hardware. The call returns when the data
             transfer has been initiated (normally with DMA). When the
             hardware is ready to accept more samples the function intr will
             be called with the argument intrarg. Calling intr will normally
             initiate another call to start_output(). This function returns 0
             on success, otherwise an error code.

     int (*start_input)(void *hdl, void *block, int bsize, void (*intr)(void
             *), void *intrarg)
             This function is called to start the transfer of bsize bytes to
             block from the audio hardware. The call returns when the data
             transfer has been initiated (normally with DMA). When the
             hardware is ready to deliver more samples the function intr will
             be called with the argument intrarg. Calling intr will normally
             initiate another call to start_input(). This function returns 0
             on success, otherwise an error code.

     int (*halt_output)(void *hdl)
             This function is called to abort the output transfer (started by
             start_output()) in progress. This function returns 0 on success,
             otherwise an error code.

     int (*halt_input)(void *hdl)
             This function is called to abort the input transfer (started by
             start_input()) in progress. This function returns 0 on success,
             otherwise an error code.

     int (*speaker_ctl)(void *hdl, int on)
             This function is optional. If supplied, it is called when a half
             duplex device changes between playing and recording. It can,
             e.g., be used to turn the speaker on and off. This function re-
             turns 0 on success, otherwise an error code.

     int (*getdev)(void *hdl, struct audio_device *ret)
             This function fills ret with relevant information about the
             driver and returns 0 on success, or it returns an error code on
             failure.

     int (*setfd)(void *hdl, int fd)
             This function is optional. If supplied, it is called when the
             AUDIO_SETFD ioctl(2) is used, but only if the device has
             AUDIO_PROP_FULLDUPLEX set. This function returns 0 on success,
             otherwise an error code.

     int (*set_port)(void *hdl, struct mixer_ctrl *mc)
             This function is called when the AUDIO_MIXER_WRITE ioctl(2) is
             used. It takes data from mc and sets the corresponding mixer
             values. This function returns 0 on success, otherwise an error
             code.

     int (*get_port)(void *hdl, struct mixer_ctrl *mc)
             This function is called when the AUDIO_MIXER_READ ioctl(2) is
             used. It fills mc and returns 0 on success, or it returns an er-
             ror code on failure.

     int (*query_devinfo)(void *hdl, struct mixer_devinfo *di)
             This function is called when the AUDIO_MIXER_DEVINFO ioctl(2) is
             used. It fills di and returns 0 on success, or it returns an er-
             ror code on failure.

     void *(*allocm)(void *hdl, int direction, size_t size, int type, int
             flags)
             This function is optional. If supplied, it is called to allocate
             the device buffers. If not supplied, 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. direction is AUMODE_PLAY
             or AUMODE_RECORD. This function returns the address of the buffer
             on success, or 0 on failure.

     void (*freem)(void *hdl, void *addr, int type)
             This function is optional. If supplied, it is called to free
             memory allocated by allocm(). If not supplied, free(9) is used
             instead.

     size_t (*round_buffersize)(void *hdl, int direction, size_t bufsize)
             This function is optional. If supplied, it 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. Note that the buffer size is always a multiple of the
             block size, so round_blocksize() and round_buffersize() must be
             consistent.

     paddr_t (*mappage)(void *hdl, void *addr, off_t offs, int prot)
             This function is optional. If supplied, it is called for mmap(2).
             It returns the map value for the page at offset offs from address
             addr mapped with protection prot. This function returns -1 on
             failure, or a machine dependent opaque value on success.

     int (*get_props)(void *hdl)
             This function returns the device properties, as per audio(4)
             AUDIO_GETPROPS ioctl(2), i.e., a combination of AUDIO_PROP_xxx
             properties.

     int (*trigger_output)(void *hdl, void *start, void *end, int blksize,
             void (*intr)(void *), void *intrarg, struct audio_params *param)
             This function is optional. If supplied, it 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
             returns when the data transfer has been initiated (normally with
             DMA). When the hardware is finished transferring each blksize
             sized block, the function intr will be called with the argument
             intrarg (typically from the audio hardware interrupt service rou-
             tine). Once started, the transfer may be stopped using
             halt_output(). This function returns 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)
             This function is optional. If supplied, it 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 returns when the data transfer has been initiated (normally
             with DMA). When the hardware is finished transferring each
             blksize sized block, the function intr will be called with the
             argument intrarg (typically from the audio hardware interrupt
             service routine). Once started, the transfer may be stopped using
             halt_input(). This function returns 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.

SEE ALSO

     ioctl(2), mmap(2), open(2), audio(4), free(9), malloc(9)

HISTORY

     This audio interface first appeared in OpenBSD 1.2.

MirOS BSD #10-current         February 11, 2000                              4

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