The FreeBSD sound subsystem cleanly separates generic sound handling issues from device-specific ones. This makes it easier to add support for new hardware.
The pcm(4) framework is the central piece of the sound subsystem. It mainly implements the following elements:
A system call interface (read, write, ioctls) to digitized sound and mixer functions. The ioctl command set is compatible with the legacy OSS or Voxware interface, allowing common multimedia applications to be ported without modification.
Common code for processing sound data (format conversions, virtual channels).
A uniform software interface to hardware-specific audio interface modules.
Additional support for some common hardware interfaces (ac97), or shared hardware-specific code (ex: ISA DMA routines).
The support for specific sound cards is implemented by hardware-specific drivers, which provide channel and mixer interfaces to plug into the generic pcm code.
In this chapter, the term pcm will refer to the central, common part of the sound driver, as opposed to the hardware-specific modules.
The prospective driver writer will of course want to start from an existing module and use the code as the ultimate reference. But, while the sound code is nice and clean, it is also mostly devoid of comments. This document tries to give an overview of the framework interface and answer some questions that may arise while adapting the existing code.
As an alternative, or in addition to starting from a working example, you can find a commented driver template at https://people.FreeBSD.org/~cg/template.c
All the relevant code lives in /usr/src/sys/dev/sound/ , except for the public ioctl interface definitions, found in /usr/src/sys/sys/soundcard.h
Under /usr/src/sys/dev/sound/ , the pcm/ directory holds the central code, while the pci/ , isa/ and usb/ directories have the drivers for PCI and ISA boards, and for USB audio devices.
Sound drivers probe and attach in almost the same way as any hardware driver module. You might want to look at the ISA or PCI specific sections of the handbook for more information.
However, sound drivers differ in some ways:
They declare themselves as
pcm
class devices, with a
struct snddev_info
device private structure:
static driver_t xxx_driver = {
"pcm",
xxx_methods,
sizeof(struct snddev_info)
};
DRIVER_MODULE(snd_xxxpci, pci, xxx_driver, pcm_devclass, 0, 0);
MODULE_DEPEND(snd_xxxpci, snd_pcm, PCM_MINVER, PCM_PREFVER,PCM_MAXVER);
Most sound drivers need to store additional private information about their device. A private data structure is usually allocated in the attach routine. Its address is passed to
pcm
by the calls to
pcm_register()
and
mixer_init()
.
pcm
later passes back this address as a parameter in calls to the sound driver interfaces.
The sound driver attach routine should declare its MIXER or AC97 interface to
pcm
by calling
mixer_init()
. For a MIXER interface, this causes
in turn a call to
xxxmixer_init()
.
The sound driver attach routine declares its general CHANNEL configuration to
pcm
by calling
pcm_register(dev, sc, nplay, nrec)
, where
sc
is the address for the device data structure, used in further calls from
pcm
, and
nplay
and
nrec
are the number of play and record channels.
The sound driver attach routine declares each of its channel objects by calls
to
pcm_addchan()
. This sets up the channel glue in
pcm
and
causes in turn a call to
xxxchannel_init()
.
The sound driver detach routine should call
pcm_unregister()
before releasing its resources.
There are two possible methods to handle non-PnP devices:
Use a
device_identify()
method (example:
sound/isa/es1888.c
). The
device_identify()
method probes for the hardware at known addresses and, if it finds a supported device, creates a new pcm device which is then passed to probe/attach.
Use a custom kernel configuration with appropriate hints for pcm devices (example: sound/isa/mss.c ).
pcm
drivers should implement
device_suspend
,
device_resume
and
device_shutdown
routines, so that power management and module unloading function correctly.
The interface between the pcm core and the sound drivers is defined in terms of kernel objects .
There are two main interfaces that a sound driver will usually provide: CHANNEL and either MIXER or AC97 .
The AC97 interface is a very small hardware access (register read/write) interface, implemented by drivers for hardware with an AC97 codec. In this case, the actual MIXER interface is provided by the shared AC97 code in pcm .
Sound drivers usually have a private data structure to describe their device, and one structure for each play and record data channel that it supports.
For all CHANNEL interface functions, the first parameter is an opaque pointer.
The second parameter is a pointer to the private channel data structure, except for
channel_init()
which has a pointer to the private device structure (and returns the channel pointer for further use by
pcm
).
For sound data transfers, the
pcm
core and the sound drivers communicate through a shared memory area, described by a
struct snd_dbuf
.
struct snd_dbuf
is private to
pcm
, and sound drivers obtain values of interest by calls to accessor functions (
sndbuf_getxxx()
).
The shared memory area has a size of
sndbuf_getsize()
and is divided into fixed size blocks of
sndbuf_getblksz()
bytes.
When playing, the general transfer mechanism is as follows (reverse the idea for recording):
pcm
initially fills up the buffer, then calls the sound driver’s
xxxchannel_trigger()
function with a parameter of PCMTRIG_START.
The sound driver then arranges to repeatedly transfer the whole memory area (
sndbuf_getbuf()
,
sndbuf_getsize()
) to the device, in blocks of
sndbuf_getblksz()
bytes. It calls back the
chn_intr()
pcm
function for each transferred block (this will typically happen at interrupt time).
chn_intr()
arranges to copy new data to the area that was transferred to the device (now free), and make appropriate updates to the
snd_dbuf
structure.
xxxchannel_init()
is called to initialize each of the play or record channels.
The calls are initiated from the sound driver attach routine. (See the crossref:sound[pcm-probe-and-attach,probe and attach section).
static void *
xxxchannel_init(kobj_t obj, void *data,
struct snd_dbuf *b, struct pcm_channel *c, int dir) (1)
{
struct xxx_info *sc = data;
struct xxx_chinfo *ch;
...
return ch; (2)
}
| 1 |
b
is the address for the channel
struct snd_dbuf
. It should be initialized in the function by calling
sndbuf_alloc()
. The buffer size to use is normally a small multiple of the 'typical' unit transfer size for your device.
c
is the
pcm
channel control structure pointer. This is an opaque object. The function should store it in the local channel structure, to be used in later calls to
pcm
(ie:
chn_intr(c)
).
dir
indicates the channel direction (
PCMDIR_PLAY
or
PCMDIR_REC
).
|
| 2 | The function should return a pointer to the private area used to control this channel. This will be passed as a parameter to other channel interface calls. |
xxxchannel_setformat()
should set up the hardware for the specified channel for the specified sound format.
static int
xxxchannel_setformat(kobj_t obj, void *data, u_int32_t format) (1)
{
struct xxx_chinfo *ch = data;
...
return 0;
}
| 1 |
format
is specified as an
AFMT_XXX value
(
soundcard.h
).
|
xxxchannel_setspeed()
sets up the channel hardware for the specified sampling speed, and returns the possibly adjusted speed.
static int
xxxchannel_setspeed(kobj_t obj, void *data, u_int32_t speed)
{
struct xxx_chinfo *ch = data;
...
return speed;
}
xxxchannel_setblocksize()
sets the block size, which is the size of unit transactions between
pcm
and the sound driver, and between the sound driver and the device. Typically, this would be the number of bytes transferred before an interrupt occurs. During a transfer, the sound driver should call
pcm
's
chn_intr()
every time this size has been transferred.
Most sound drivers only take note of the block size here, to be used when an actual transfer will be started.
static int
xxxchannel_setblocksize(kobj_t obj, void *data, u_int32_t blocksize)
{
struct xxx_chinfo *ch = data;
...
return blocksize; (1)
}
| 1 |
The function returns the possibly adjusted block size. In case the block size is indeed changed,
sndbuf_resize()
should be called to adjust the buffer.
|
xxxchannel_trigger()
is called by
pcm
to control data transfer operations in the driver.
static int
xxxchannel_trigger(kobj_t obj, void *data, int go) (1)
{
struct xxx_chinfo *ch = data;
...
return 0;
}
| 1 |
go
defines the action for the current call. The possible values are:
|
|
If the driver uses ISA DMA,
|
xxxchannel_getptr()
returns the current offset in the transfer buffer. This will typically be called by
chn_intr()
, and this is how
pcm
knows where it can transfer new data.
xxxchannel_free()
is called to free up channel resources, for example when the driver is unloaded, and should be implemented if the channel data structures are dynamically allocated or if
sndbuf_alloc()
was not used for buffer allocation.
struct pcmchan_caps *
xxxchannel_getcaps(kobj_t obj, void *data)
{
return &xxx_caps; (1)
}
| 1 |
The routine returns a pointer to a (usually statically-defined)
pcmchan_caps
structure (defined in
sound/pcm/channel.h
. The structure holds the minimum and maximum sampling frequencies, and the accepted sound formats. Look at any sound driver for an example.
|
channel_reset()
,
channel_resetdone()
, and
channel_notify()
are for special purposes and should not be implemented in a driver without discussing it on the
FreeBSD multimedia mailing list
.
channel_setdir()
is deprecated.
xxxmixer_init()
initializes the hardware and tells
pcm
what mixer devices are available for playing and recording
static int
xxxmixer_init(struct snd_mixer *m)
{
struct xxx_info *sc = mix_getdevinfo(m);
u_int32_t v;
[Initialize hardware]
[Set appropriate bits in v for play mixers] (1)
mix_setdevs(m, v);
[Set appropriate bits in v for record mixers]
mix_setrecdevs(m, v)
return 0;
}
| 1 |
Set bits in an integer value and call
mix_setdevs()
and
mix_setrecdevs()
to tell
pcm
what devices exist.
|
Mixer bits definitions can be found in
soundcard.h
(
SOUND_MASK_XXX
values and
SOUND_MIXER_XXX
bit shifts).
xxxmixer_set()
sets the volume level for one mixer device.
static int
xxxmixer_set(struct snd_mixer *m, unsigned dev,
unsigned left, unsigned right) (1)
{
struct sc_info *sc = mix_getdevinfo(m);
[set volume level]
return left | (right << 8); (2)
}
| 1 |
The device is specified as a
SOUND_MIXER_XXX
value. The volume values are specified in range [0-100]. A value of zero should mute the device.
|
| 2 | As the hardware levels probably will not match the input scale, and some rounding will occur, the routine returns the actual level values (in range 0-100) as shown. |
xxxmixer_setrecsrc()
sets the recording source device.
static int
xxxmixer_setrecsrc(struct snd_mixer *m, u_int32_t src) (1)
{
struct xxx_info *sc = mix_getdevinfo(m);
[look for non zero bit(s) in src, set up hardware]
[update src to reflect actual action]
return src; (2)
}
| 1 | The desired recording devices are specified as a bit field |
| 2 | The actual devices set for recording are returned. Some drivers can only set one device for recording. The function should return -1 if an error occurs. |
The AC97 interface is implemented by drivers with an AC97 codec. It only has three methods:
xxxac97_init()
returns the number of ac97 codecs found.
ac97_read()
and
ac97_write()
read or write a specified register.
The AC97 interface is used by the AC97 code in pcm to perform higher level operations. Look at sound/pci/maestro3.c or many others under sound/pci/ for an example.
Last modified on : February 18, 2025 by Fernando Apesteguía