June 11, 2007 AUDIO 4

### NAME

audio - device-independent audio driver layer

### SYNOPSIS

### DESCRIPTION

The audio driver provides support for various audio peripherals. It provides a uniform programming interface layer above different underlying audio hardware drivers. The audio layer provides full-duplex operation if the underlying hardware configuration supports it.

There are four device files available for audio operation: ```/dev/audio`, ```/dev/sound`, ```/dev/audioctl`, and ```/dev/mixer`.

```/dev/audio` and ```/dev/sound` are used for recording or playback of digital samples.

```/dev/mixer` is used to manipulate volume, recording source, or other audio mixer functions.

```/dev/audioctl` accepts the same ioctl(2) operations as ```/dev/sound`, but no other operations.

In contrast to ```/dev/sound` which has the exclusive open property ```/dev/audioctl` can be opened at any time and can be used to manipulate the audio device while it is in use.

### SAMPLING DEVICES

When ```/dev/audio` is opened, it automatically directs the underlying driver to manipulate monaural 8-bit mu-law samples. In addition, if it is opened read-only (write-only) the device is set to half-duplex record (play) mode with recording (playing) unpaused and playing (recording) paused. When ```/dev/sound` is opened, it maintains the previous audio sample mode and record/playback mode. In all other respects ```/dev/audio` and ```/dev/sound` are identical.

Only one process may hold open a sampling device at a given time (although file descriptors may be shared between processes once the first open completes).

On a half-duplex device, writes while recording is in progress will be immediately discarded. Similarly, reads while playback is in progress will be filled with silence but delayed to return at the current sampling rate. If both playback and recording are requested on a half-duplex device, playback mode takes precedence and recordings will get silence.

On a full-duplex device, reads and writes may operate concurrently without interference. If a full-duplex capable audio device is opened for both reading and writing it will start in half-duplex play mode; full-duplex mode has to be set explicitly.

On either type of device, if the playback mode is paused then silence is played instead of the provided samples, and if recording is paused then the process blocks in read(2) until recording is unpaused.

If a writing process does not call write(2) frequently enough to provide samples at the pace the hardware consumes them silence is inserted. If the ```AUMODE_PLAY_ALL` mode is not set the writing process must provide enough data via subsequent write calls to ``catch up'' in time to the current audio block before any more process-provided samples will be played. If a reading process does not call read(2) frequently enough, it will simply miss samples.

The audio device is normally accessed with read(2) or write(2) calls, but it can also be mapped into user memory with mmap(2) (when supported by the device). Once the device has been mapped it can no longer be accessed by read or write; all access is by reading and writing to the mapped memory. The device appears as a block of memory of size buffersize (as available via ```AUDIO_GETINFO` or ```AUDIO_GETBUFINFO`). The device driver will continuously move data from this buffer from/to the audio hardware, wrapping around at the end of the buffer. To find out where the hardware is currently accessing data in the buffer the ```AUDIO_GETIOFFS` and ```AUDIO_GETOOFFS` calls can be used. The playing and recording buffers are distinct and must be mapped separately if both are to be used. Only encodings that are not emulated (i.e. where ```AUDIO_ENCODINGFLAG_EMULATED` is not set) work properly for a mapped device.

The audio device, like most devices, can be used in select, can be set in non-blocking mode and can be set (with a ```FIOASYNC` ioctl) to send a ```SIGIO` when I/O is possible. The mixer device can be set to generate a ```SIGIO` whenever a mixer value is changed.

The following ioctl(2) commands are supported on the sample devices:

```AUDIO_FLUSH`
  This command stops all playback and recording, clears all queued buffers, resets error counters, and restarts recording and playback as appropriate for the current sampling mode.

```AUDIO_RERROR``(int)`
  This command fetches the count of dropped input samples into its integer argument. There is no information regarding when in the sample stream they were dropped.

```AUDIO_WSEEK``(int)`
  This command fetches the count of samples that are queued ahead of the first sample in the most recent sample block written into its integer argument.

```AUDIO_DRAIN`
  This command suspends the calling process until all queued playback samples have been played by the hardware.

```AUDIO_GETDEV``(audio_device_t)`
  This command fetches the current hardware device information into the audio_device_t argument. `

      typedef struct audio_device {
              char name[MAX_AUDIO_DEV_LEN];
              char version[MAX_AUDIO_DEV_LEN];
              char config[MAX_AUDIO_DEV_LEN];
      } audio_device_t;

  `

```AUDIO_GETFD``(int)`
  The command returns the current setting of the full duplex mode.

```AUDIO_GETENC``(audio_encoding_t)`
  This command is used iteratively to fetch sample encoding names and format_ids into the input/output audio_encoding_t argument. `

      typedef struct audio_encoding {
              int index;      /* input: nth encoding */
              char name[MAX_AUDIO_DEV_LEN]; /* name of encoding */
              int encoding;   /* value for encoding parameter */
              int precision;  /* value for precision parameter */
              int flags;
      #define AUDIO_ENCODINGFLAG_EMULATED 1 /* software emulation mode */
      } audio_encoding_t;

  `

  To query all the supported encodings, start with an index field of 0 and continue with successive encodings (1, 2, ...) until the command returns an error.

```AUDIO_SETFD``(int)`
  This command sets the device into full-duplex operation if its integer argument has a non-zero value, or into half-duplex operation if it contains a zero value. If the device does not support full-duplex operation, attempting to set full-duplex mode returns an error.

```AUDIO_GETPROPS``(int)`
  This command gets a bit set of hardware properties. If the hardware has a certain property the corresponding bit is set, otherwise it is not. The properties can have the following values:

  ```AUDIO_PROP_FULLDUPLEX`
    the device admits full duplex operation.
  ```AUDIO_PROP_MMAP`
    the device can be used with mmap(2).
  ```AUDIO_PROP_INDEPENDENT`
    the device can set the playing and recording encoding parameters independently.

```AUDIO_GETIOFFS``(audio_offset_t)`

```AUDIO_GETOOFFS``(audio_offset_t)`
  This command fetches the current offset in the input(output) buffer where the audio hardware's DMA engine will be putting(getting) data. It mostly useful when the device buffer is available in user space via the mmap(2) call. The information is returned in the audio_offset structure. `

      typedef struct audio_offset {
              u_int   samples;   /* Total number of bytes transferred */
              u_int   deltablks; /* Blocks transferred since last checked */
              u_int   offset;    /* Physical transfer offset in buffer */
      } audio_offset_t;

  `

```AUDIO_GETINFO``(audio_info_t)`

```AUDIO_GETBUFINFO``(audio_info_t)`

```AUDIO_SETINFO``(audio_info_t)`
  Get or set audio information as encoded in the audio_info structure. `

      typedef struct audio_info {
              struct  audio_prinfo play;   /* info for play (output) side */
              struct  audio_prinfo record; /* info for record (input) side */
              u_int   monitor_gain;                   /* input to output mix */
              /* BSD extensions */
              u_int   blocksize;      /* H/W read/write block size */
              u_int   hiwat;          /* output high water mark */
              u_int   lowat;          /* output low water mark */
              u_int   _ispare1;
              u_int   mode;           /* current device mode */
      #define AUMODE_PLAY     0x01
      #define AUMODE_RECORD   0x02
      #define AUMODE_PLAY_ALL 0x04    /* do not do real-time correction */
      } audio_info_t;

  `

  When setting the current state with ```AUDIO_SETINFO`, the audio_info structure should first be initialized with ```AUDIO_INITINFO` `&info(.blm Pp`) and then the particular values to be changed should be set. This allows the audio driver to only set those things that you wish to change and eliminates the need to query the device with ```AUDIO_GETINFO` or ```AUDIO_GETBUFINFO` first.

  The mode field should be set to ```AUMODE_PLAY`, ```AUMODE_RECORD`, ```AUMODE_PLAY_ALL`, or a bitwise OR combination of the three. Only full-duplex audio devices support simultaneous record and playback.

  hiwat and lowat are used to control write behavior. Writes to the audio devices will queue up blocks until the high-water mark is reached, at which point any more write calls will block until the queue is drained to the low-water mark. hiwat and lowat set those high- and low-water marks (in audio blocks). The default for hiwat is the maximum value and for lowat 75 % of hiwat.

  blocksize sets the current audio blocksize. The generic audio driver layer and the hardware driver have the opportunity to adjust this block size to get it within implementation-required limits. Upon return from an ```AUDIO_SETINFO` call, the actual blocksize set is returned in this field. Normally the blocksize is calculated to correspond to 50ms of sound and it is recalculated when the encoding parameter changes, but if the blocksize is set explicitly this value becomes sticky, i.e., it remains even when the encoding is changed. The stickiness can be cleared by reopening the device or setting the blocksize to 0. ``

      struct audio_prinfo {
              u_int   sample_rate;    /* sample rate in samples/s */
              u_int   channels;       /* number of channels, usually 1 or 2 */
              u_int   precision;      /* number of bits/sample */
              u_int   encoding;       /* data encoding (AUDIO_ENCODING_* below) */
              u_int   gain;           /* volume level */
              u_int   port;           /* selected I/O port */
              u_long  seek;           /* BSD extension */
              u_int   avail_ports;    /* available I/O ports */
              u_int   buffer_size;    /* total size audio buffer */
              u_int   _ispare[1];
              /* Current state of device: */
              u_int   samples;        /* number of samples */
              u_int   eof;            /* End Of File (zero-size writes) counter */
              u_char  pause;          /* non-zero if paused, zero to resume */
              u_char  error;          /* non-zero if underflow/overflow occurred */
              u_char  waiting;        /* non-zero if another process hangs in open */
              u_char  balance;        /* stereo channel balance */
              u_char  cspare[2];
              u_char  open;           /* non-zero if currently open */
              u_char  active;         /* non-zero if I/O is currently active */
      };

  Note: many hardware audio drivers require identical playback and recording sample rates, sample encodings, and channel counts. The playing information is always set last and will prevail on such hardware. If the hardware can handle different settings the ```AUDIO_PROP_INDEPENDENT` property is set.

  The encoding parameter can have the following values:

  ```AUDIO_ENCODING_ULAW`
    mu-law encoding, 8 bits/sample
  ```AUDIO_ENCODING_ALAW`
    A-law encoding, 8 bits/sample
  ```AUDIO_ENCODING_SLINEAR`
    two's complement signed linear encoding with the platform byte order
  ```AUDIO_ENCODING_ULINEAR`
    unsigned linear encoding with the platform byte order
  ```AUDIO_ENCODING_ADPCM`
    ADPCM encoding, 8 bits/sample
  ```AUDIO_ENCODING_SLINEAR_LE`
    two's complement signed linear encoding with little endian byte order
  ```AUDIO_ENCODING_SLINEAR_BE`
    two's complement signed linear encoding with big endian byte order
  ```AUDIO_ENCODING_ULINEAR_LE`
    unsigned linear encoding with little endian byte order
  ```AUDIO_ENCODING_ULINEAR_BE`
    unsigned linear encoding with big endian byte order

  The gain, port and balance settings provide simple shortcuts to the richer mixer interface described below and are not obtained by ```AUDIO_GETBUFINFO`. The gain should be in the range [```AUDIO_MIN_GAIN```,`AUDIO_MAX_GAIN`] and the balance in the range [```AUDIO_LEFT_BALANCE```,`AUDIO_RIGHT_BALANCE`] with the normal setting at ```AUDIO_MID_BALANCE`.

  The input port should be a combination of:

  ```AUDIO_MICROPHONE`
    to select microphone input.
  ```AUDIO_LINE_IN`
    to select line input.
  ```AUDIO_CD`
    to select CD input.

  The output port should be a combination of:

  ```AUDIO_SPEAKER`
    to select speaker output.
  ```AUDIO_HEADPHONE`
    to select headphone output.
  ```AUDIO_LINE_OUT`
    to select line output.

  The available ports can be found in avail_ports ``(`AUDIO_GETBUFINFO` only).

  buffer_size is the total size of the audio buffer. The buffer size divided by the blocksize gives the maximum value for hiwat. Currently the buffer_size can only be read and not set.

  The seek and samples fields are only used by ```AUDIO_GETINFO` and ```AUDIO_GETBUFINFO`. seek represents the count of samples pending; samples represents the total number of bytes recorded or played, less those that were dropped due to inadequate consumption/production rates.

  pause returns the current pause/unpause state for recording or playback. For ```AUDIO_SETINFO`, if the pause value is specified it will either pause or unpause the particular direction.

### MIXER DEVICE

The mixer device, ```/dev/mixer`, may be manipulated with ioctl(2) but does not support read(2) or write(2). It supports the following ioctl(2) commands:

```AUDIO_GETDEV``(audio_device_t)`
  This command is the same as described above for the sampling devices.

```AUDIO_MIXER_READ``(mixer_ctrl_t)`

```AUDIO_MIXER_WRITE``(mixer_ctrl_t)`
  These commands read the current mixer state or set new mixer state for the specified device dev. type identifies which type of value is supplied in the mixer_ctrl_t argument. `

      #define AUDIO_MIXER_CLASS  0
      #define AUDIO_MIXER_ENUM   1
      #define AUDIO_MIXER_SET    2
      #define AUDIO_MIXER_VALUE  3
      typedef struct mixer_ctrl {
              int dev;                        /* input: nth device */
              int type;
              union {
                      int ord;                /* enum */
                      int mask;               /* set */
                      mixer_level_t value;    /* value */
              } un;
      } mixer_ctrl_t;
      #define AUDIO_MIN_GAIN  0
      #define AUDIO_MAX_GAIN  255
      typedef struct mixer_level {
              int num_channels;
              u_char level[8];               /* [num_channels] */
      } mixer_level_t;
      #define AUDIO_MIXER_LEVEL_MONO  0
      #define AUDIO_MIXER_LEVEL_LEFT  0
      #define AUDIO_MIXER_LEVEL_RIGHT 1

  `

  For a mixer value, the value field specifies both the number of channels and the values for each channel. If the channel count does not match the current channel count, the attempt to change the setting may fail (depending on the hardware device driver implementation). For an enumeration value, the ord field should be set to one of the possible values as returned by a prior ```AUDIO_MIXER_DEVINFO` command. The type ```AUDIO_MIXER_CLASS` is only used for classifying particular mixer device types and is not used for ```AUDIO_MIXER_READ` or ```AUDIO_MIXER_WRITE`.

```AUDIO_MIXER_DEVINFO``(mixer_devinfo_t)`
  This command is used iteratively to fetch audio mixer device information into the input/output mixer_devinfo_t argument. To query all the supported devices, start with an index field of 0 and continue with successive devices (1, 2, ...) until the command returns an error. `

      typedef struct mixer_devinfo {
              int index;              /* input: nth mixer device */
              audio_mixer_name_t label;
              int type;
              int mixer_class;
              int next, prev;
      #define AUDIO_MIXER_LAST        -1
              union {
                      struct audio_mixer_enum {
                              int num_mem;
                              struct {
                                      audio_mixer_name_t label;
                                      int ord;
                              } member[32];
                      } e;
                      struct audio_mixer_set {
                              int num_mem;
                              struct {
                                      audio_mixer_name_t label;
                                      int mask;
                              } member[32];
                      } s;
                      struct audio_mixer_value {
                              audio_mixer_name_t units;
                              int num_channels;
                              int delta;
                      } v;
              } un;
      } mixer_devinfo_t;

  `

  The label field identifies the name of this particular mixer control. The index field may be used as the dev field in ```AUDIO_MIXER_READ` and ```AUDIO_MIXER_WRITE` commands. The type field identifies the type of this mixer control. Enumeration types are typically used for on/off style controls (e.g. a mute control) or for input/output device selection (e.g. select recording input source from CD, line in, or microphone). Set types are similar to enumeration types but any combination of the mask bits can be used.

  The mixer_class field identifies what class of control this is. The (arbitrary) value set by the hardware driver may be determined by examining the mixer_class field of the class itself, a mixer of type ```AUDIO_MIXER_CLASS`. For example, a mixer controlling the input gain on the line in circuit would have a mixer_class that matches an input class device with the name ``inputs'' ``(`AudioCinputs`), and would have a label of ``line'' ``(`AudioNline`). Mixer controls which control audio circuitry for a particular audio source (e.g. line-in, CD in, DAC output) are collected under the input class, while those which control all audio sources (e.g. master volume, equalization controls) are under the output class. Hardware devices capable of recording typically also have a record class, for controls that only affect recording, and also a monitor class.

  The next and prev may be used by the hardware device driver to provide hints for the next and previous devices in a related set (for example, the line in level control would have the line in mute as its ``next'' value). If there is no relevant next or previous value, ```AUDIO_MIXER_LAST` is specified.

  For ```AUDIO_MIXER_ENUM` mixer control types, the enumeration values and their corresponding names are filled in. For example, a mute control would return appropriate values paired with ```AudioNon` and ```AudioNoff`. For ```AUDIO_MIXER_VALUE` and ```AUDIO_MIXER_SET` mixer control types, the channel count is returned; the units name specifies what the level controls (typical values are ```AudioNvolume`, ```AudioNtreble`, ```AudioNbass`).

By convention, all the mixer devices can be distinguished from other mixer controls because they use a name from one of the ```AudioC*` string values.

### FILES

```/dev/audio`
```/dev/audioctl`
```/dev/sound`
```/dev/mixer`

### SEE ALSO

audioctl(1), mixerctl(1), ioctl(2), ossaudio(3), midi(4), radio(4)

#### ISA bus

aria(4), ess(4), gus(4), guspnp(4), pas(4), sb(4), wss(4), ym(4)

#### PCI bus

auacer(4), auich(4), auixp(4), autri(4), auvia(4), azalia(4), clcs(4), clct(4), cmpci(4), eap(4), emuxki(4), esa(4), esm(4), eso(4), fms(4), neo(4), sv(4), yds(4)

#### PCMCIA

esl(4)

#### TURBOchannel

bba(4)

#### USB

uaudio(4)

### BUGS

If the device is used in mmap(2) it is currently always mapped for writing (playing) due to VM system weirdness.