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     sound, pcm, snd — FreeBSD PCM audio device infrastructure


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

           device sound


     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_ai2s(4) (enabled by default on powerpc)
        snd_audiocs(4) (enabled by default on sparc64)
        snd_cmi(4) (enabled by default on amd64, i386)
        snd_csa(4) (enabled by default on amd64, i386)
        snd_davbus(4) (enabled by default on powerpc)
        snd_emu10kx(4) (enabled by default on amd64, i386)
        snd_es137x(4) (enabled by default on amd64, i386, sparc64)
        snd_hda(4) (enabled by default on amd64, i386)
        snd_ich(4) (enabled by default on amd64, i386)
        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)

     Refer to the manual page for each bridge device driver for driver specific settings and

   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):


     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.

     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.

     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.

     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).

             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.

             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) 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.

             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.

             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

             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:

             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

             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.

             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 VCHANs.  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

             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.

             Default value for sound 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
             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

             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
             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.

             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

             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

                 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
                 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

             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.  The mixer(8) utility can be used to start and stop
     recording from an specific 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

   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.


     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.


     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


     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_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,

     Julius O'Smith's Digital Audio Resampling,

     Polynomial Interpolators for High-Quality Resampling of Oversampled Audio, by Olli

     The OSS API,


     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.


     Luigi Rizzo <> initially wrote the pcm device driver and this manual page.
     Cameron Grant <> later revised the device driver for FreeBSD 4.0.
     Seigo Tanimura <> revised this manual page.  It was then
     rewritten for FreeBSD 5.2.


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