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

Provided by: freebsd-manpages_12.2-2_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.

Debian                                           March 22, 2012                                         SOUND(4)