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NetBSD 6.1.5 - man page for audio (netbsd section 9)

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AUDIO(9)			  BSD 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     (*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 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 0 if not needed.

     int open(void *hdl, int flags)
	     optional, is called when the audio device is opened.  It should initialize the hard-
	     ware 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 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_slinear16, and audio_params_t repre-
	     senting [8000Hz, slinear_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 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)
	     optional, is called after all calls to set_params, and set_port, are done.  A hard-
	     ware 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, other-
	     wise 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_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 values.  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 allo-
	     cation 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 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, parame-
	     terized 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 inter-
	     rupt 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)
	     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 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 con-
     trol.  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 play-
     back 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 destina-
     tions 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.

BSD					November 23, 2011				      BSD
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