Chapter 15. Sound Subsystem

15.1. Introduction

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

15.2. Files

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.

15.3. Probing, Attaching, etc.

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.

15.4. Interfaces

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.

15.4.1. The CHANNEL Interface

15.4.1.1. Common Notes for Function Parameters

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).

15.4.1.2. Overview of Data Transfer Operations

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.

15.4.1.3. channel_init

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.

15.4.1.4. channel_setformat

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).

15.4.1.5. channel_setspeed

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;
           }

15.4.1.6. channel_setblocksize

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.

15.4.1.7. channel_trigger

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, sndbuf_isadma() should be called before performing actions on the device, and will take care of the DMA chip side of things.

15.4.1.8. channel_getptr

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.

15.4.1.9. channel_free

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.

15.4.1.10. channel_getcaps

          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.

15.4.1.11. More Functions

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.

15.4.2. The MIXER Interface

15.4.2.1. mixer_init

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).

15.4.2.2. mixer_set

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.

15.4.2.3. mixer_setrecsrc

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.

15.4.2.4. mixer_uninit, mixer_reinit

xxxmixer_uninit() should ensure that all sound is muted and if possible mixer hardware should be powered down.

xxxmixer_reinit() should ensure that the mixer hardware is powered up and any settings not controlled by mixer_set() or mixer_setrecsrc() are restored.

15.4.3. The AC97 Interface

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: August 11, 2024 by Fernando Apesteguía

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