Ubuntu Manpage: sound, pcm, snd — FreeBSD PCM audio device infrastructure

Provided by: freebsd-manpages_11.1-3_all

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_audiocs(4) (enabled by default on sparc64)
• 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, sparc64)
• 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_t4dwave(4) (enabled by default on sparc64)
• snd_uaudio(4) (enabled by default on amd64, i386, powerpc, sparc64)
• 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.

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.