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	(*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	(*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);
    	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 *);
    	void	(*copy_output)(void *hdl, size_t bytes);
    	void	(*underrun)(void *hdl);
    	int	(*set_blksz)(void *, int,
    		    struct audio_params *, struct audio_params *, int);
    	int	(*set_nblks)(void *, int, int,
    		    struct audio_params *, int);
    };
    struct audio_params {
    	u_long	sample_rate;		/* sample rate */
    	u_int	encoding;		/* mu-law, linear, etc */
    	u_int	precision;		/* bits/sample */
    	u_int	bps;			/* bytes/sample */
    	u_int	msb;			/* data alignment */
    	u_int	channels;		/* mono(1), stereo(2) */
    };

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(const 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 initializes 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 (*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 combination of these flags, but indicates the current mode of the device (i.e., the value corresponding to the flags argument to the open() function). The play and rec structures contain the encoding parameters that will be set. The values of the structures must 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). Except the channel count, the same value is passed in both play and rec.

    The machine independent audio driver does some preliminary parameter 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, otherwise 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, otherwise 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 returns 0 on success, otherwise an error code.

int (*setfd)(void *hdl, int fd)
    This function is optional. If supplied, it is called when the device is opened in full-duplex mode, 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 error 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 error 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.

int (*get_props)(void *hdl)
    This function returns 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 routine). 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.

void (*copy_output)(void *hdl, size_t bytes)
    This function is called whenever the given amount of bytes was appended to the play ring buffer, typically during a write(2) system call. Drivers using bounce buffers for transfers between the audio ring buffer and the device could implement this function to copy the given amount of bytes into their bounce buffers. There's no analogue function for recording as data is produced by the device and could simply be copied upon transfer completion.

void (*underrun)(void *hdl)
    This function is called at interrupt context whenever a play block was skipped by the audio(4) driver. Drivers using bounce buffers for transfers between the audio ring buffer and the device must implement this method to skip one block from the audio ring buffer and transfer the corresponding amount of silence to the device.

int (*set_blksz)(void *hdl, int mode, struct audio_params *play, struct audio_params *rec, int blksz)
    This function is called to set the audio block size. mode is a combination of the AUMODE_RECORD and AUMODE_PLAY flags indicating the current mode set with the open() function. The play and rec structures contain the current encoding set with the set_params() function. blksz is the desired block size in frames. It may be adjusted to match hardware constraints. This function returns the adjusted block size.

int (*set_nblks)(void *hdl, int dir, int blksz, struct audio_params *params, int nblks)
    This function is called to set the number of audio blocks in the ring buffer. dir is either the AUMODE_RECORD or the AUMODE_PLAY flag, indicating which ring buffer size is set. The params structure contains the encoding parameters set by the set_params() method. blksz is the current block size in frames set with the set_params function. The params structure is the current encoding parameters, set with the set_params() function. nblks is the desired number of blocks in the ring buffer. It may be lowered to at least two, to match hardware constraints. This function returns the adjusted number of blocks.

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), open(2), sio_open(3), audio(4), free(9), malloc(9)

HISTORY
This audio interface first appeared in OpenBSD 1.2.