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SOUND(4)		 BSD Kernel Interfaces Manual		      SOUND(4)

     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

     Non-PnP sound cards require the following lines in device.hints(5):


     The sound driver provides support for PCM audio play and capture.	This
     driver also supports various PCI, ISA, WSS/MSS compatible sound cards,
     AC97 mixer and High Definition Audio.  Once the sound driver attaches,
     supported devices provide audio record and playback channels.  The
     FreeBSD sound system provides dynamic mixing “VCHAN” and rate conversion
     “soft formats”.  True full duplex operation is available on most sound

     If the sound card is supported by a bridge driver, the sound driver works
     in conjunction with the bridge driver.

     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.

     The driver does its best to recognize the installed hardware and drive it
     correctly so the user is not required to add several lines in
     /boot/device.hints.  For PCI and ISA PnP cards this is actually easy
     since they identify themselves.  For legacy ISA cards, the driver looks
     for MSS cards at addresses 0x530 and 0x604 (unless overridden in

   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_emu10k1_load  (“NO”) If set to “YES”, only the SoundBlaster 5.1
			     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 chan‐
     nel 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 sup‐
     ported 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 chan‐
     nels used.	 The current multichannel interleaved structure and arrange‐
     ment was implemented by inspecting various popular UNIX applications.
     There were no single standard, so much care has been taken to try to sat‐
     isfy each possible scenario, despite the fact that each application has
     its own conflicting standard.

     The Parametric Software Equlizer (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.⟨X⟩.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 equal‐
	     izer.  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

	     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) compability.	 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 applica‐
		 tions, and deny for everything else.

	     1	 Always allow PROT_EXEC page mappings.

	     Enable to automatically assign default sound unit to the most
	     recent 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 /dev/dsp${hw.snd.default_unit}.

	     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 require‐
	     ment 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 appli‐
	     cable 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 inter‐
	     polation.	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%

	     3	 Continuation of the bandlimited SINC interpolator, with 100dB
		 stopband, 36 taps and 90% bandwidth as quality values.

	     4	 Continuation of the bandlimited SINC inteprolator, 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 tun‐
	     able.  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 aggres‐
	     sive 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 consid‐
	     ered 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 trans‐
	     parently 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 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 cur‐
		 rent format, speed, and pseudo device statistics such as buf‐
		 fer overruns and buffer underruns.

	     3	 File names and versions of the currently loaded sound mod‐

	     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 applica‐
	     tions to use the correct ioctl(): SNDCTL_DSP_GETPLAYVOL,
	     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.

	     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 convert‐
	     ing 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 device.

	     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 formats.

		 Channel mixing is done using fixed format/rate, but advanced
		 operations such as digital passthrough also work.  All chan‐
		 nels will produce sound as usual until a digital format play‐
		 back 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 for‐
		 mat/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) mecha‐
	     nism.  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.

     Channel statistics are only kept while the device is open.	 So with situ‐
     ations 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 applica‐
     tions 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 appli‐
     cations may need to be recompiled with a slightly modified audio module.
     See <sys/soundcard.h> for a complete list of the supported ioctl() func‐

     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 driv‐

     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

     pcm%d:play:%d:dsp%d.p%d: play interrupt timeout, channel dead  The hard‐
     ware 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_als4000(4), snd_atiixp(4), snd_audiocs(4), snd_cmi(4),
     snd_cs4281(4), snd_csa(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_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), 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,

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

     The OSS API, http://www.opensound.com/pguide/oss.pdf.

     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 stan‐

     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.

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

BSD				 July 13, 2009				   BSD

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