path: root/documentation/content/en/books/arch-handbook/sound/_index.adoc

---
title: Chapter 15. Sound Subsystem
prev: books/arch-handbook/newbus
next: books/arch-handbook/pccard
description: FreeBSD Sound Subsystem
tags: ["Sound", "OSS", "pcm", "mixer"]
showBookMenu: true
weight: 17
path: "/books/arch-handbook/"
---

[[oss]]
= Sound Subsystem
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[[oss-intro]]
== 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 man: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 [.filename]#pcm# code.

In this chapter, the term [.filename]#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[ https://people.FreeBSD.org/~cg/template.c]

[[oss-files]]
== Files

All the relevant code lives in [.filename]#/usr/src/sys/dev/sound/#, except for the public ioctl interface definitions, found in [.filename]#/usr/src/sys/sys/soundcard.h#

Under [.filename]#/usr/src/sys/dev/sound/#, the [.filename]#pcm/# directory holds the central code, while the [.filename]#pci/#, [.filename]#isa/# and [.filename]#usb/# directories have the drivers for PCI and ISA boards, and for USB audio devices.

[[pcm-probe-and-attach]]
== 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 crossref:isa-driver[isa-driver,ISA] or crossref:pci[pci,PCI] specific sections of the handbook for more information.

However, sound drivers differ in some ways:

* They declare themselves as [.filename]#pcm# class devices, with a `struct snddev_info` device private structure:
+
[.programlisting]
....
          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 [.filename]#pcm# by the calls to `pcm_register()` and `mixer_init()`. [.filename]#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 [.filename]#pcm# by calling `mixer_init()`. For a MIXER interface, this causes in turn a call to <<xxxmixer-init,`xxxmixer_init()`>>.
* The sound driver attach routine declares its general CHANNEL configuration to [.filename]#pcm# by calling `pcm_register(dev, sc, nplay, nrec)`, where `sc` is the address for the device data structure, used in further calls from [.filename]#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 [.filename]#pcm# and causes in turn a call to <<xxxchannel-init,`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: [.filename]#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: [.filename]#sound/isa/mss.c#).

[.filename]#pcm# drivers should implement `device_suspend`, `device_resume` and `device_shutdown` routines, so that power management and module unloading function correctly.

[[oss-interfaces]]
== Interfaces

The interface between the [.filename]#pcm# core and the sound drivers is defined in terms of <<kernel-objects,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 [.filename]#pcm#.

=== The CHANNEL Interface

==== 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 [.filename]#pcm#).

==== Overview of Data Transfer Operations

For sound data transfers, the [.filename]#pcm# core and the sound drivers communicate through a shared memory area, described by a `struct snd_dbuf`.

`struct snd_dbuf` is private to [.filename]#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):

* [.filename]#pcm# initially fills up the buffer, then calls the sound driver's <<channel-trigger,`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()`[.filename]#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]]
==== 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 <<pcm-probe-and-attach,probe and attach section>>).

[.programlisting]
....
          static void *
          xxxchannel_init(kobj_t obj, void *data,
             struct snd_dbuf *b, struct pcm_channel *c, int dir) <.>
          {
              struct xxx_info *sc = data;
              struct xxx_chinfo *ch;
               ...
              return ch; <.>
           }
....

<.> `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 [.filename]#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 [.filename]#pcm# (ie: `chn_intr(c)`).`dir` indicates the channel direction (`PCMDIR_PLAY` or `PCMDIR_REC`).

<.> 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.

==== channel_setformat

`xxxchannel_setformat()` should set up the hardware for the specified channel for the specified sound format.

[.programlisting]
....
          static int
          xxxchannel_setformat(kobj_t obj, void *data, u_int32_t format) <.>
          {
              struct xxx_chinfo *ch = data;
               ...
              return 0;
           }
....

<.> `format` is specified as an `AFMT_XXX value` ([.filename]#soundcard.h#).

==== channel_setspeed

`xxxchannel_setspeed()` sets up the channel hardware for the specified sampling speed, and returns the possibly adjusted speed.

[.programlisting]
....
          static int
          xxxchannel_setspeed(kobj_t obj, void *data, u_int32_t speed)
          {
              struct xxx_chinfo *ch = data;
               ...
              return speed;
           }
....

==== channel_setblocksize

`xxxchannel_setblocksize()` sets the block size, which is the size of unit transactions between [.filename]#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 [.filename]#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.

[.programlisting]
....
          static int
          xxxchannel_setblocksize(kobj_t obj, void *data, u_int32_t blocksize)
          {
              struct xxx_chinfo *ch = data;
                ...
              return blocksize; <.>
           }
....

<.> 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.

[[channel-trigger]]
==== channel_trigger

`xxxchannel_trigger()` is called by [.filename]#pcm# to control data transfer operations in the driver.

[.programlisting]
....
          static int
          xxxchannel_trigger(kobj_t obj, void *data, int go) <.>
          {
              struct xxx_chinfo *ch = data;
               ...
              return 0;
           }
....

<.> `go` defines the action for the current call. The possible values are:

[NOTE]
====
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.
====

==== channel_getptr

`xxxchannel_getptr()` returns the current offset in the transfer buffer. This will typically be called by `chn_intr()`, and this is how [.filename]#pcm# knows where it can transfer new data.

==== 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.

==== channel_getcaps

[.programlisting]
....
          struct pcmchan_caps *
          xxxchannel_getcaps(kobj_t obj, void *data)
          {
              return &xxx_caps; <.>
           }
....

<.> The routine returns a pointer to a (usually statically-defined) `pcmchan_caps` structure (defined in [.filename]#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.

==== 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}.

`channel_setdir()` is deprecated.

=== The MIXER Interface

[[xxxmixer-init]]
==== mixer_init

`xxxmixer_init()` initializes the hardware and tells [.filename]#pcm# what mixer devices are available for playing and recording

[.programlisting]
....
          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] <.>
              mix_setdevs(m, v);
              [Set appropriate bits in v for record mixers]
              mix_setrecdevs(m, v)

              return 0;
          }
....

<.> Set bits in an integer value and call `mix_setdevs()` and `mix_setrecdevs()` to tell [.filename]#pcm# what devices exist.

Mixer bits definitions can be found in [.filename]#soundcard.h# (`SOUND_MASK_XXX` values and `SOUND_MIXER_XXX` bit shifts).

==== mixer_set

`xxxmixer_set()` sets the volume level for one mixer device.

[.programlisting]
....
          static int
          xxxmixer_set(struct snd_mixer *m, unsigned dev,
                           unsigned left, unsigned right) <.>
          {
              struct sc_info *sc = mix_getdevinfo(m);
              [set volume level]
              return left | (right << 8); <.>
          }
....

<.> 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.
<.> 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.

==== mixer_setrecsrc

`xxxmixer_setrecsrc()` sets the recording source device.

[.programlisting]
....
          static int
          xxxmixer_setrecsrc(struct snd_mixer *m, u_int32_t src) <.>
          {
              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; <.>
           }
....

<.> The desired recording devices are specified as a bit field
<.> 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.

==== 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.

=== 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 [.filename]#pcm# to perform higher level operations. Look at [.filename]#sound/pci/maestro3.c# or many others under [.filename]#sound/pci/# for an example.