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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:

     +o   snd_ad1816(4)
     +o   snd_ai2s(4) (enabled by default on powerpc)
     +o   snd_als4000(4)
     +o   snd_atiixp(4)
     +o   snd_audiocs(4) (enabled by default on sparc64)
     +o   snd_cmi(4) (enabled by default on amd64, i386)
     +o   snd_cs4281(4)
     +o   snd_csa(4) (enabled by default on amd64, i386)
     +o   snd_davbus(4) (enabled by default on powerpc)
     +o   snd_ds1(4)
     +o   snd_emu10k1(4)
     +o   snd_emu10kx(4) (enabled by default on amd64, i386)
     +o   snd_envy24(4)
     +o   snd_envy24ht(4)
     +o   snd_es137x(4) (enabled by default on amd64, i386, sparc64)
     +o   snd_ess(4)
     +o   snd_fm801(4)
     +o   snd_gusc(4)
     +o   snd_hda(4) (enabled by default on amd64, i386)
     +o   snd_hdspe(4)
     +o   snd_ich(4) (enabled by default on amd64, i386)
     +o   snd_maestro(4)
     +o   snd_maestro3(4)
     +o   snd_mss(4)
     +o   snd_neomagic(4)
     +o   snd_sb16
     +o   snd_sb8
     +o   snd_sbc(4)
     +o   snd_solo(4)
     +o   snd_spicds(4)
     +o   snd_t4dwave(4) (enabled by default on sparc64)
     +o   snd_uaudio(4) (enabled by default on amd64, i386, powerpc, sparc64)
     +o   snd_via8233(4) (enabled by default on amd64, i386)
     +o   snd_via82c686(4)
     +o   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_audiocs(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, 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.

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 11.0-PRERELEASE         March 22, 2012         FreeBSD 11.0-PRERELEASE

NAME | SYNOPSIS | DESCRIPTION | FILES | EXAMPLES | DIAGNOSTICS | SEE ALSO | HISTORY | AUTHORS | BUGS

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