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
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"
DESCRIPTION¶
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 cards.
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/device.hints).
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 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 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.
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.⟨X⟩.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.⟨X⟩.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) 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 applications, and deny for
everything else.
- 1
- Always allow PROT_EXEC page mappings.
- hw.snd.default_auto
- Enable to automatically assign default sound unit to the
most recent 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 inteprolator, 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.
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.
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_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,
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.