SOUND(4) FreeBSD Kernel Interfaces Manual SOUND(4)
NAME
sound, pcm, snd - FreeBSD PCM audio device infrastructure
SYNOPSIS
To compile this driver into the kernel, place the following line in your
kernel configuration file:
device sound
DESCRIPTION
The sound driver is the main component of the FreeBSD sound system. It
works in conjunction with a bridge device driver on supported devices and
provides PCM audio record and playback once it attaches. Each bridge
device driver supports a specific set of audio chipsets and needs to be
enabled together with the sound driver. PCI and ISA PnP audio devices
identify themselves so users are usually not required to add anything to
/boot/device.hints.
Some of the main features of the sound driver are: multichannel audio,
per-application volume control, dynamic mixing through virtual sound
channels, true full duplex operation, bit perfect audio, rate conversion
and low latency modes.
The sound driver is enabled by default, along with several bridge device
drivers. Those not enabled by default can be loaded during runtime with
kldload(8) or during boot via loader.conf(5). The following bridge
device drivers are available:
• snd_ad1816(4)
• snd_ai2s(4) (enabled by default on powerpc)
• snd_als4000(4)
• snd_atiixp(4)
• snd_cmi(4) (enabled by default on amd64, i386)
• snd_cs4281(4)
• snd_csa(4) (enabled by default on amd64, i386)
• snd_davbus(4) (enabled by default on powerpc)
• snd_ds1(4)
• snd_emu10k1(4)
• snd_emu10kx(4) (enabled by default on amd64, i386)
• snd_envy24(4)
• snd_envy24ht(4)
• snd_es137x(4) (enabled by default on amd64, i386)
• snd_ess(4)
• snd_fm801(4)
• snd_gusc(4)
• snd_hda(4) (enabled by default on amd64, i386)
• snd_hdspe(4)
• snd_ich(4) (enabled by default on amd64, i386)
• snd_maestro(4)
• snd_maestro3(4)
• snd_mss(4)
• snd_neomagic(4)
• snd_sb16
• snd_sb8
• snd_sbc(4)
• snd_solo(4)
• snd_spicds(4)
• snd_uaudio(4) (enabled by default on amd64, i386, powerpc)
• snd_via8233(4) (enabled by default on amd64, i386)
• snd_via82c686(4)
• snd_vibes(4)
Refer to the manual page for each bridge device driver for driver
specific settings and information.
Legacy Hardware
For old legacy ISA cards, the driver looks for MSS cards at addresses
0x530 and 0x604. These values can be overridden in /boot/device.hints.
Non-PnP sound cards require the following lines in device.hints(5):
hint.pcm.0.at="isa"
hint.pcm.0.irq="5"
hint.pcm.0.drq="1"
hint.pcm.0.flags="0x0"
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.
Boot Variables
In general, the module snd_foo corresponds to device snd_foo and can be
loaded by the boot loader(8) via loader.conf(5) or from the command line
using the kldload(8) utility. Options which can be specified in
/boot/loader.conf include:
snd_driver_load ("NO") If set to "YES", this option loads all
available drivers.
snd_hda_load ("NO") If set to "YES", only the Intel High
Definition Audio bridge device driver and
dependent modules will be loaded.
snd_foo_load ("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="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
ioctl() support. Multichannel audio works both with and without VCHANs.
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.
EQ
The Parametric Software Equalizer (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.%d.eq tunable.
VCHANs
Each device can optionally support more playback and recording channels
than physical hardware provides by using "virtual channels" or VCHANs.
VCHAN options can be configured via the sysctl(8) interface but can only
be manipulated while the device is inactive.
VPC
FreeBSD supports independent and individual volume controls for each
active application, without touching the master sound 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 sound driver. The following tunables can not be
changed during runtime using sysctl(8).
hint.pcm.%d.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.%d.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.
hw.snd.compat_linux_mmap
Linux mmap(2) compatibility. The following values are supported
(default is 0):
-1 Force disabling/denying PROT_EXEC mmap(2) requests.
0 Auto detect proc/ABI type, allow mmap(2) for Linux
applications, and deny for everything else.
1 Always allow PROT_EXEC page mappings.
hw.snd.default_auto
Automatically assign the default sound unit. The following
values are supported (default is 1):
0 Do not assign the default sound unit automatically.
1 Use the best available sound device based on playing and
recording capabilities of the device.
2 Use the most recently attached device.
hw.snd.default_unit
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}.
hw.snd.feeder_eq_exact_rate
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.
hw.snd.feeder_rate_max
Maximum allowable sample rate.
hw.snd.feeder_rate_min
Minimum allowable sample rate.
hw.snd.feeder_rate_polyphase_max
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.
hw.snd.feeder_rate_quality
Sample rate converter quality. Default value is 1, linear
interpolation. Available options include:
0 Zero Order Hold, ZOH. Very fast, but with poor quality.
1 Linear interpolation. Fast, quality is subject to personal
preference. Technically the quality is poor however, due to
the lack of anti-aliasing filtering.
2 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.
3 Continuation of the bandlimited SINC interpolator, with 100dB
stopband, 36 taps and 90% bandwidth as quality values.
4 Continuation of the bandlimited SINC interprolator, with
100dB stopband, 164 taps and 97% bandwidth as quality values.
hw.snd.feeder_rate_round
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.
hw.snd.latency
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).
hw.snd.latency_profile
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.
hw.snd.maxautovchans
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 VCHANs. Set to "0" if no
VCHANs are desired. Maximum value is 256.
hw.snd.report_soft_formats
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.
hw.snd.report_soft_matrix
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.
hw.snd.verbose
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:
0 Installed devices and their allocated bus resources.
1 The number of playback, record, virtual channels, and flags
per device.
2 Channel information per device including the channel's
current format, speed, and pseudo device statistics such as
buffer overruns and buffer underruns.
3 File names and versions of the currently loaded sound
modules.
4 Various messages intended for debugging.
hw.snd.vpc_0db
Default value for sound volume. Increase to give more room for
attenuation control. Decrease for more amplification, with the
possible cost of sound clipping.
hw.snd.vpc_autoreset
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.
hw.snd.vpc_mixer_bypass
The recommended way to use the VPC feature is to teach
applications to use the correct 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.
hw.snd.vpc_reset
Enable to restore all channel volumes back to the default value
of 0db.
dev.pcm.%d.bitperfect
Enable or disable bitperfect mode. When enabled, channels will
skip all dsp processing, such as channel matrixing, rate
converting and equalizing. The pure sound stream will be fed
directly to the hardware. If VCHANs are 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
VCHANs and enable this sysctl. Default is disabled.
dev.pcm.%d.[play|rec].vchans
The current number of VCHANs allocated per device. This can be
set to preallocate a certain number of VCHANs. Setting this
value to "0" will disable VCHANs for this device.
dev.pcm.%d.[play|rec].vchanformat
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:
s16le:1.0
Mono.
s16le:2.0
Stereo, 2 channels (left, right).
s16le:2.1
3 channels (left, right, LFE).
s16le:3.0
3 channels (left, right, rear center).
s16le:4.0
Quadraphonic, 4 channels (front/rear left and right).
s16le:4.1
5 channels (4.0 + LFE).
s16le:5.0
5 channels (4.0 + center).
s16le:5.1
6 channels (4.0 + center + LFE).
s16le:6.0
6 channels (4.0 + front/rear center).
s16le:6.1
7 channels (6.0 + LFE).
s16le:7.1
8 channels (4.0 + center + LFE + left and right side).
dev.pcm.%d.[play|rec].vchanmode
VCHAN format/rate selection. Available options include:
fixed
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.
passthrough
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.
adaptive
Works like the "passthrough" mode, but is a bit smarter,
especially for multiple sound 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.
dev.pcm.%d.[play|rec].vchanrate
Sample rate speed for VCHAN mixing. All playback paths will be
converted to this sample rate before the mixing process begins.
dev.pcm.%d.polling
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. The mixer(8) utility can
be used to start and stop recording from an specific device.
Statistics
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 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 <sys/soundcard.h> for a complete list of the supported
ioctl() functions.
FILES
The sound drivers may create the following device nodes:
/dev/audio%d.%d Sparc-compatible audio device.
/dev/dsp%d.%d Digitized voice device.
/dev/dspW%d.%d Like /dev/dsp, but 16 bits per sample.
/dev/dsp%d.p%d Playback channel.
/dev/dsp%d.r%d Record channel.
/dev/dsp%d.vp%d Virtual playback channel.
/dev/dsp%d.vr%d Virtual recording channel.
/dev/sndstat Current sound status, including all channels and
drivers.
The first number in the device node represents the unit number of the
sound device. All sound 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.
EXAMPLES
Use the sound metadriver to load all sound bridge device drivers at once
(for example if it is unclear which the correct driver to use is):
kldload snd_driver
Load a specific bridge device driver, in this case the Intel High
Definition Audio driver:
kldload snd_hda
Check the status of all detected sound devices:
cat /dev/sndstat
Change the default sound device, in this case to the second device. This
is handy if there are multiple sound devices available:
sysctl hw.snd.default_unit=1
DIAGNOSTICS
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.
SEE ALSO
snd_ad1816(4), snd_ai2s(4), snd_als4000(4), snd_atiixp(4), snd_cmi(4),
snd_cs4281(4), snd_csa(4), snd_davbus(4), snd_ds1(4), snd_emu10k1(4),
snd_emu10kx(4), snd_envy24(4), snd_envy24ht(4), snd_es137x(4),
snd_ess(4), snd_fm801(4), snd_gusc(4), snd_hda(4), snd_hdspe(4),
snd_ich(4), snd_maestro(4), snd_maestro3(4), snd_mss(4), snd_neomagic(4),
snd_sbc(4), snd_solo(4), snd_spicds(4), snd_t4dwave(4), snd_uaudio(4),
snd_via8233(4), snd_via82c686(4), snd_vibes(4), devfs(5),
device.hints(5), loader.conf(5), dmesg(8), kldload(8), mixer(8),
sysctl(8)
Cookbook formulae for audio EQ biquad filter coefficients (Audio-EQ-
Cookbook.txt), by Robert Bristow-Johnson,
https://www.musicdsp.org/en/latest/Filters/197-rbj-audio-eq-cookbook.html.
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://yehar.com/blog/wp-content/uploads/2009/08/deip.pdf.
The OSS API, http://www.opensound.com/pguide/oss.pdf.
HISTORY
The sound device driver first appeared in FreeBSD 2.2.6 as pcm, written
by Luigi Rizzo. It was later rewritten in FreeBSD 4.0 by Cameron Grant.
The API evolved from the VOXWARE standard which later became OSS
standard.
AUTHORS
Luigi Rizzo <luigi@iet.unipi.it> initially wrote the pcm device driver
and this manual page. Cameron Grant <gandalf@vilnya.demon.co.uk> later
revised the device driver for FreeBSD 4.0. Seigo Tanimura
<tanimura@r.dl.itc.u-tokyo.ac.jp> revised this manual page. It was then
rewritten for FreeBSD 5.2.
BUGS
Some features of your sound card (e.g., global volume control) might not
be supported on all devices.
FreeBSD 13.1-RELEASE-p6 December 26, 2020 FreeBSD 13.1-RELEASE-p6
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