--- name: "pcm" section: "4" platform: "freebsd" package: "base" version: "16.0-CURRENT-20261027-61c70dc16a79" release: "FreeBSD 16.0-CURRENT (2026-10-27)" platforms: ["linux", "freebsd"] url: "https://new.carta.tech/pcm" --- # pcm(4) FreeBSD PCM audio device infrastructure ## 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)](/kldload/8.md) or during boot via [loader.conf(5)](/loader.conf/5.md). The following bridge device drivers are available: - [snd\_ai2s(4)](/snd_ai2s/4.md) (enabled by default on powerpc) - [snd\_als4000(4)](/snd_als4000/4.md) - [snd\_atiixp(4)](/snd_atiixp/4.md) - [snd\_cmi(4)](/snd_cmi/4.md) (enabled by default on amd64, i386) - [snd\_cs4281(4)](/snd_cs4281/4.md) - [snd\_csa(4)](/snd_csa/4.md) (enabled by default on amd64, i386) - [snd\_davbus(4)](/snd_davbus/4.md) (enabled by default on powerpc) - [snd\_emu10k1(4)](/snd_emu10k1/4.md) - [snd\_emu10kx(4)](/snd_emu10kx/4.md) (enabled by default on amd64, i386) - [snd\_envy24(4)](/snd_envy24/4.md) - [snd\_envy24ht(4)](/snd_envy24ht/4.md) - [snd\_es137x(4)](/snd_es137x/4.md) (enabled by default on amd64, i386) - [snd\_fm801(4)](/snd_fm801/4.md) - [snd\_hda(4)](/snd_hda/4.md) (enabled by default on amd64, i386) - [snd\_hdsp(4)](/snd_hdsp/4.md) - [snd\_hdspe(4)](/snd_hdspe/4.md) - [snd\_ich(4)](/snd_ich/4.md) (enabled by default on amd64, i386) - [snd\_maestro3(4)](/snd_maestro3/4.md) - [snd\_neomagic(4)](/snd_neomagic/4.md) - [snd\_solo(4)](/snd_solo/4.md) - [snd\_spicds(4)](/snd_spicds/4.md) - [snd\_uaudio(4)](/snd_uaudio/4.md) (auto-loaded on device plug) - [snd\_via8233(4)](/snd_via8233/4.md) (enabled by default on amd64, i386) - [snd\_via82c686(4)](/snd_via82c686/4.md) - [snd\_vibes(4)](/snd_vibes/4.md) Refer to the manual page for each bridge device driver for driver specific settings and information. ### Boot Variables In general, the module *snd\_foo* corresponds to **device snd\_foo** and can be loaded by the boot [loader(8)](/loader/8.md) via [loader.conf(5)](/loader.conf/5.md) or from the command line using the [kldload(8)](/kldload/8.md) 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)](/sysctl/8.md) 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. ### 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)](/sysctl/8.md) 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). ### Runtime Configuration There are a number of [sysctl(8)](/sysctl/8.md) 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* This [sysctl(8)](/sysctl/8.md) variable is available only when Linux application support is compiled into the kernel, and affects Linux binaries only. It is a hack to get around the fact that, for i386 emulation, [linux(4)](/linux/4.md) historically set PROT\_EXEC automatically when PROT\_READ or PROT\_WRITE was set during an [mmap(2)](/mmap/2.md) call, which FreeBSD does not. The following values are supported (default is -1): - \-1 Force-disable PROT\_EXEC [mmap(2)](/mmap/2.md) requests, even for Linux applications. - 0 Allow PROT\_EXEC [mmap(2)](/mmap/2.md) requests for Linux applications only. - *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(4)](/devfs/4.md), 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.vchans\_enable* Global VCHAN setting to enable (1) or disable (0) VCHANs. This setting can be overridden for an individual device by using the *dev.pcm.%d.\[play\|rec\].vchans* tunables. Default is enabled. - *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\_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.eq* Set to 1 or 0 to enable (1) or disable (0) the equalizer. Default is disabled. When enabled, the mixer bass and treble controls can be used, as well as the *dev.pcm.%d.eq\_preamp* tunable. - *dev.pcm.%d.eq\_preamp* Equalizer preamp (in dB). *dev.pcm.%d.eq* has to be enabled, in order for this tunable to take effect. Read the tunable's description to see the available range of values. - *dev.pcm.%d.\[play\|rec\].vchans* Enable (1) or disable (0) VCHANs. Default is enabled. - *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)](/callout/9.md) mechanism. Disabled by default and currently only available for a few device drivers. ### 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. ### kqueue Support The *kevent* structure's fields are defined as follows: - *ident* The file descriptor for the device. - *filter* The kernel filter used to process the event. This is either EVFILT\_READ for recording or EVFILT\_WRITE for playback. - *data* The number of bytes available to be read or written. - *ext* - *ext\[0\]* For devices that use [mmap(2)](/mmap/2.md), it contains the *count\_info.ptr* field reported by *SNDCTL\_DSP\_GETIPTR* for recording or *SNDCTL\_DSP\_GETOPTR* for playback, whereas for devices that do not use [mmap(2)](/mmap/2.md), it contains the *oss\_count\_t.fifo\_samples* field reported by *SNDCTL\_DSP\_CURRENT\_IPTR* for recording or *SNDCTL\_DSP\_CURRENT\_OPTR* for playback. - *ext\[1\]* contains the *audio\_errinfo.rec\_overruns* field for recording events, or the *audio\_errinfo.play\_underruns* field for playback events, as reported by *SNDCTL\_DSP\_GETERROR*. This allows a single *kevent* to report that data is ready, how many bytes can be read or written, the current buffer position, and the current error count. This is especially useful for scenarios that use [mmap(2)](/mmap/2.md), because without [kqueue(2)](/kqueue/2.md) the typical processing loop consists of: - Calculating the wait time and calling [clock\_nanosleep(2)](/clock_nanosleep/2.md). - Calling [ioctl(2)](/ioctl/2.md) for *SNDCTL\_DSP\_GETIPTR* or *SNDCTL\_DSP\_GETOPTR* to get the pointer into the memory-mapped buffer where new data starts. - Calling [ioctl(2)](/ioctl/2.md) for *SNDCTL\_DSP\_GETERROR* to get over/underruns and handle them. ## FILES The **sound** drivers may create the following device nodes: - */dev/dsp%d* Audio device. The number represents the unit number of the device. - */dev/dsp* Alias of */dev/dsp${hw.snd.default\_unit}*. Available only if *hw.snd.basename\_clone* is set. - */dev/sndstat* Current **sound** status, including all channels and drivers. 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)](/dmesg/8.md) utility. Users have to be part of the audio group in order to access any device node created by **sound**, as well as the device drivers, and [virtual\_oss(8)](/virtual_oss/8.md). ## 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: ``` mixer -d pcm1 ``` ## SEE ALSO [devfs(4)](/devfs/4.md), [snd\_ai2s(4)](/snd_ai2s/4.md), [snd\_als4000(4)](/snd_als4000/4.md), [snd\_atiixp(4)](/snd_atiixp/4.md), [snd\_cmi(4)](/snd_cmi/4.md), [snd\_cs4281(4)](/snd_cs4281/4.md), [snd\_csa(4)](/snd_csa/4.md), [snd\_davbus(4)](/snd_davbus/4.md), [snd\_emu10k1(4)](/snd_emu10k1/4.md), [snd\_emu10kx(4)](/snd_emu10kx/4.md), [snd\_envy24(4)](/snd_envy24/4.md), [snd\_envy24ht(4)](/snd_envy24ht/4.md), [snd\_es137x(4)](/snd_es137x/4.md), [snd\_fm801(4)](/snd_fm801/4.md), [snd\_hda(4)](/snd_hda/4.md), [snd\_hdsp(4)](/snd_hdsp/4.md), [snd\_hdspe(4)](/snd_hdspe/4.md), [snd\_ich(4)](/snd_ich/4.md), [snd\_maestro3(4)](/snd_maestro3/4.md), [snd\_neomagic(4)](/snd_neomagic/4.md), [snd\_solo(4)](/snd_solo/4.md), [snd\_spicds(4)](/snd_spicds/4.md), [snd\_t4dwave(4)](/snd_t4dwave/4.md), [snd\_uaudio(4)](/snd_uaudio/4.md), [snd\_via8233(4)](/snd_via8233/4.md), [snd\_via82c686(4)](/snd_via82c686/4.md), [snd\_vibes(4)](/snd_vibes/4.md), [device.hints(5)](/device.hints/5.md), [loader.conf(5)](/loader.conf/5.md), [dmesg(8)](/dmesg/8.md), [kldload(8)](/kldload/8.md), [mixer(8)](/mixer/8.md), [sysctl(8)](/sysctl/8.md), [virtual\_oss(8)](/virtual_oss/8.md) *Cookbook formulae for audio EQ biquad filter coefficients (Audio-EQ-Cookbook.txt), by Robert Bristow-Johnson*, https://www.musicdsp.org/en/latest/Filters/197-rbj-audio-eq-cookbook.html. *Julius O'Smith's Digital Audio Resampling*, https://ccrma.stanford.edu/~jos/resample/. *Polynomial Interpolators for High-Quality Resampling of Oversampled Audio, by Olli Niemitalo*, https://yehar.com/blog/wp-content/uploads/2009/08/deip.pdf. *The OSS API*, https://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. Some audio devices might refuse to work properly unless the sample rate is configured the same for both recording and playback, even if only simplex is used. See the *dev.pcm.%d.\[play\|rec\].vchanrate* sysctls.