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Version: 361827 (ubuntu - 24/10/10)

Section: 4 (Pilotes et protocoles réseau)

BSD mandoc


sound pcm snd - Fx PCM audio device infrastructure


To compile this driver into the kernel, place the following line in your kernel configuration file:
device sound

Non-PnP sound cards require the following lines in device.hints5:



The driver provides support for PCM audio play and capture. This driver also supports various PCI ISA WSS/MSS compatible sound cards, AC97 mixer and High Definition Audio. Once the driver attaches, supported devices provide audio record and playback channels. The Fx sound system provides dynamic mixing ``VCHAN'' and rate conversion ``soft formats'' True full duplex operation is available on most sound cards.

If the sound card is supported by a bridge driver, the driver works in conjunction with the bridge driver.

Apart from the usual parameters, the flags field is used to specify the secondary DMA channel (generally used for capture in full duplex cards). Flags are set to 0 for cards not using a secondary DMA channel, or to 0x10 + C to specify channel C.

The driver does its best to recognize the installed hardware and drive it correctly so the user is not required to add several lines in /boot/device.hints For PCI and ISA PnP cards this is actually easy since they identify themselves. For legacy ISA cards, the driver looks for MSS cards at addresses 0x530 and 0x604 (unless overridden in /boot/device.hints )

Boot Variables

In general, the module snd_foo corresponds to device snd_foo and can be loaded by the boot loader(8) via loader.conf5 or from the command line using the kldload(8) utility. Options which can be specified in /boot/loader.conf include:
(``NO '' ) If set to ``YES '' this option loads all available drivers.
(``NO '' ) If set to ``YES '' only the SoundBlaster 5.1 driver and dependent modules will be loaded.
(``NO '' ) If set to ``YES '' load driver for card/chipset foo.

To define default values for the different mixer channels, set the channel to the preferred value using hints, e.g.: hint.pcm.0.line = Qq 0 This will mute the input channel per default.

Multichannel Audio

Multichannel audio, popularly referred to as ``surround sound'' is supported and enabled by default. The FreeBSD multichannel matrix processor supports up to 18 interleaved channels, but the limit is currently set to 8 channels (as commonly used for 7.1 surround sound). The internal matrix mapping can handle reduction, expansion or re-routing of channels. This provides a base interface for related multichannel Fn ioctl support. Multichannel audio works both with and without VCHA Most bridge device drivers are still missing multichannel matrixing support , but in most cases this should be trivial to implement. Use the dev.pcm.%d.[play|rec].vchanformat sysctl(8) to adjust the number of channels used. The current multichannel interleaved structure and arrangement was implemented by inspecting various popular UNIX applications. There were no single standard, so much care has been taken to try to satisfy each possible scenario, despite the fact that each application has its own conflicting standard.


The Parametric Software Equlizer (EQ) enables the use of ``tone'' controls (bass and treble). Commonly used for ear-candy or frequency compensation due to the vast difference in hardware quality. EQ is disabled by default, but can be enabled with the hint.pcm. Ao X Ac .eq tunable.

VCHAEach device can optionally support more playback and recording channels

than physical hardware provides by using ``virtual channels'' or VCHA VCHAN options can be configured via the sysctl(8) interface but can only be manipulated while the device is inactive.


FreeBSD supports independent and individual volume controls for each active application, without touching the master volume. This is sometimes referred to as Volume Per Channel (VPC). The VPC feature is enabled by default.

Loader Tunables

The following loader tunables are used to set driver configuration at the loader(8) prompt before booting the kernel, or they can be stored in /boot/loader.conf in order to automatically set them before booting the kernel. It is also possible to use kenv(1) to change these tunables before loading the driver. The following tunables can not be changed during runtime using sysctl(8).
hint.pcm. Ao X Ac .eq
Set to 1 or 0 to explicitly enable (1) or disable (0) the equalizer. Requires a driver reload if changed. Enabling this will make bass and treble controls appear in mixer applications. This tunable is undefined by default. Equalizing is disabled by default.
hint.pcm. Ao X Ac .vpc
Set to 1 or 0 to explicitly enable (1) or disable (0) the VPC feature. This tunable is undefined by default. VPC is however enabled by default.

Runtime Configuration

There are a number of sysctl(8) variables available which can be modified during runtime. These values can also be stored in /etc/sysctl.conf in order to automatically set them during the boot process. hw.snd.* are global settings and dev.pcm.* are device specific.
Linux mmap(2) compability. The following values are supported (default is 0):
Force disabling/denying PROT_EXEC mmap(2) requests.
Auto detect proc/ABI type, allow mmap(2) for Linux applications, and deny for everything else.
Always allow PROT_EXEC page mappings.
Enable to automatically assign default sound unit to the most recent attached device.
Default sound card for systems with multiple sound cards. When using devfs(5), the default device for /dev/dsp Equivalent to a symlink from /dev/dsp to /dev/dsp ${hw.snd.default_unit}
Only certain rates are allowed for precise processing. The default behavior is however to allow sloppy processing for all rates, even the unsupported ones. Enable to toggle this requirement and only allow processing for supported rates.
Maximum allowable sample rate.
Minimum allowable sample rate.
Adjust to set the maximum number of allowed polyphase entries during the process of building resampling filters. Disabling polyphase resampling has the benefit of reducing memory usage, at the expense of slower and lower quality conversion. Only applicable when the SINC interpolator is used. Default value is 183040. Set to 0 to disable polyphase resampling.
Sample rate converter quality. Default value is 1, linear interpolation. Available options include:
Zero Order Hold, ZOH. Very fast, but with poor quality.
Linear interpolation. Fast, quality is subject to personal preference. Technically the quality is poor however, due to the lack of anti-aliasing filtering.
Bandlimited SINC interpolator. Implements polyphase banking to boost the conversion speed, at the cost of memory usage, with multiple high quality polynomial interpolators to improve the conversion accuracy. 100% fixed point, 64bit accumulator with 32bit coefficients and high precision sample buffering. Quality values are 100dB stopband, 8 taps and 85% bandwidth.
Continuation of the bandlimited SINC interpolator, with 100dB stopband, 36 taps and 90% bandwidth as quality values.
Continuation of the bandlimited SINC inteprolator, with 100dB stopband, 164 taps and 97% bandwidth as quality values.
Sample rate rounding threshold, to avoid large prime division at the cost of accuracy. All requested sample rates will be rounded to the nearest threshold value. Possible values range between 0 (disabled) and 500. Default is 25.
Configure the buffering latency. Only affects applications that do not explicitly request blocksize / fragments. This tunable provides finer granularity than the hw.snd.latency_profile tunable. Possible values range between 0 (lowest latency) and 10 (highest latency).
Define sets of buffering latency conversion tables for the hw.snd.latency tunable. A value of 0 will use a low and aggressive latency profile which can result in possible underruns if the application cannot keep up with a rapid irq rate, especially during high workload. The default value is 1, which is considered a moderate/safe latency profile.
Global VCHAN setting that only affects devices with at least one playback or recording channel available. The sound system will dynamically create up to this many VCHA Set to ``0'' if no VCHANS are desired. Maximum value is 256.
Controls the internal format conversion if it is available transparently to the application software. When disabled or not available, the application will only be able to select formats the device natively supports.
Enable seamless channel matrixing even if the hardware does not support it. Makes it possible to play multichannel streams even with a simple stereo sound card.
Level of verbosity for the /dev/sndstat device. Higher values include more output and the highest level, four, should be used when reporting problems. Other options include:
Installed devices and their allocated bus resources.
The number of playback, record, virtual channels, and flags per device.
Channel information per device including the channel's current format, speed, and pseudo device statistics such as buffer overruns and buffer underruns.
File names and versions of the currently loaded sound modules.
Various messages intended for debugging.
Default value for volume. Increase to give more room for attenuation control. Decrease for more amplification, with the possible cost of sound clipping.
When a channel is closed the channel volume will be reset to 0db. This means that any changes to the volume will be lost. Enabling this will preserve the volume, at the cost of possible confusion when applications tries to re-open the same device.
The recommended way to use the VPC feature is to teach applications to use the correct Fn ioctl : SNDCTL_DSP_GETPLAYVOL, SNDCTL_DSP_SETPLAYVOL, SNDCTL_DSP_SETRECVOL, SNDCTL_DSP_SETRECVOL. This is however not always possible. Enable this to allow applications to use their own existing mixer logic to control their own channel volume.
Enable to restore all channel volumes back to the default value of 0db.
Enable or disable bitperfect mode. When enabled, channels will skip all dsp processing, such as channel matrixing, rate converting and equalizing. The pure stream will be fed directly to the hardware. If VCHAare enabled, the bitperfect mode will use the VCHAN format/rate as the definitive format/rate target. The recommended way to use bitperfect mode is to disable VCHAand enable this sysctl. Default is disabled.
The current number of VCHAallocated per device. This can be set to preallocate a certain number of VCHA Setting this value to ``0'' will disable VCHAfor this device.
Format for VCHAN mixing. All playback paths will be converted to this format before the mixing process begins. By default only 2 channels are enabled. Available options include:
Stereo, 2 channels (left, right).
3 channels (left, right, LFE).
3 channels (left, right, rear center).
Quadraphonic, 4 channels (front/rear left and right).
5 channels (4.0 + LFE).
5 channels (4.0 + center).
6 channels (4.0 + center + LFE).
6 channels (4.0 + front/rear center).
7 channels (6.0 + LFE).
8 channels (4.0 + center + LFE + left and right side).
VCHAN format/rate selection. Available options include:
Channel mixing is done using fixed format/rate. Advanced operations such as digital passthrough will not work. Can be considered as a ``legacy'' mode. This is the default mode for hardware channels which lack support for digital formats.
Channel mixing is done using fixed format/rate, but advanced operations such as digital passthrough also work. All channels will produce sound as usual until a digital format playback is requested. When this happens all other channels will be muted and the latest incoming digital format will be allowed to pass through undisturbed. Multiple concurrent digital streams are supported, but the latest stream will take precedence and mute all other streams.
Works like the ``passthrough'' mode, but is a bit smarter, especially for multiple channels with different format/rate. When a new channel is about to start, the entire list of virtual channels will be scanned, and the channel with the best format/rate (usually the highest/biggest) will be selected. This ensures that mixing quality depends on the best channel. The downside is that the hardware DMA mode needs to be restarted, which may cause annoying pops or clicks.
Sample rate speed for VCHAN mixing. All playback paths will be converted to this sample rate before the mixing process begins.
Experimental polling mode support where the driver operates by querying the device state on each tick using a callout(9) mechanism. Disabled by default and currently only available for a few device drivers.

Recording Channels

On devices that have more than one recording source (ie: mic and line), there is a corresponding /dev/dsp%d.r%d device.


Channel statistics are only kept while the device is open. So with situations involving overruns and underruns, consider the output while the errant application is open and running.

IOCTL Support

The driver supports most of the OSS Fn ioctl functions, and most applications work unmodified. A few differences exist, while memory mapped playback is supported natively and in Linux emulation, memory mapped recording is not due to VM system design. As a consequence, some applications may need to be recompiled with a slightly modified audio module. See In sys/soundcard.h for a complete list of the supported Fn ioctl functions.


The drivers may create the following device nodes:
Sparc-compatible audio device.
Digitized voice device.
Like /dev/dsp but 16 bits per sample.
Playback channel.
Record channel.
Virtual playback channel.
Virtual recording channel.
Current status, including all channels and drivers.

The first number in the device node represents the unit number of the device. All devices are listed in /dev/sndstat Additional messages are sometimes recorded when the device is probed and attached, these messages can be viewed with the dmesg(8) utility.

The above device nodes are only created on demand through the dynamic devfs(5) clone handler. Users are strongly discouraged to access them directly. For specific sound card access, please instead use /dev/dsp or /dev/dsp%d


pcm%d:play:%d:dsp%d.p%d: play interrupt timeout, channel dead
The hardware does not generate interrupts to serve incoming (play) or outgoing (record) data.
unsupported subdevice XX
A device node is not created properly.


snd_ad18164, snd_als40004, snd_atiixp4, snd_audiocs4, snd_cmi4, snd_cs42814, snd_csa4, snd_ds14, snd_emu10k14, snd_emu10kx4, snd_envy244, snd_envy24ht4, snd_es137x4, snd_ess4, snd_fm8014, snd_gusc4, snd_hda4, snd_ich4, snd_maestro4, snd_maestro34, snd_mss4, snd_neomagic4, snd_sbc4, snd_solo4, snd_spicds4, snd_t4dwave4, snd_uaudio4, snd_via82334, snd_via82c6864, snd_vibes4, devfs(5), device.hints5, loader.conf5, dmesg(8), kldload(8), sysctl(8)
"Cookbook formulae for audio EQ biquad filter coefficients, by Robert Bristow-Johnson" "http://www.musicdsp.org/files/Audio-EQ-Cookbook.txt"
"Julius O'Smith's Digital Audio Resampling" "http://ccrma.stanford.edu/~jos/resample/"
"Polynomial Interpolators for High-Quality Resampling of Oversampled Audio, by Olli Niemitalo" "http://www.student.oulu.fi/~oniemita/dsp/deip.pdf"
"The OSS API" "http://www.opensound.com/pguide/oss.pdf"


The device driver first appeared in Fx 2.2.6 as pcm written by An Luigi Rizzo . It was later rewritten in Fx 4.0 by An Cameron Grant . The API evolved from the VOXWARE standard which later became OSS standard.


An -nosplit An Luigi Rizzo Aq luigi@iet.unipi.it initially wrote the pcm device driver and this manual page. An Cameron Grant Aq gandalf@vilnya.demon.co.uk later revised the device driver for Fx 4.0 . An Seigo Tanimura Aq tanimura@r.dl.itc.u-tokyo.ac.jp revised this manual page. It was then rewritten for Fx 5.2 .


Some features of your sound card (e.g., global volume control) might not be supported on all devices.