daemon is an intermediate layer between
audio or MIDI programs and the hardware. It performs the necessary audio
processing to allow any program to work on any supported hardware. By default,
accepts connections from programs running
on the same system only; it initializes only when programs are using its
services, allowing sndiod
to consume a negligible
amount of system resources the rest of the time. Systems with no audio
hardware can use sndiod
to keep hot-pluggable
devices usable by default at virtually no cost.
operates as follows: it exposes at least one
that any number of audio programs can
connect to and use as if it was audio hardware. During playback,
receives audio data concurrently from all
programs, mixes it and sends the result to the hardware device. Similarly,
during recording it duplicates audio data recorded from the device and sends
it to all programs. Since audio data flows through the
process, it has the opportunity to process
audio data on the fly:
- Change the sound encoding to overcome incompatibilities
between software and hardware.
- Route the sound from one channel to another, join stereo
or split mono.
- Control the per-application playback volume as well as
the master volume.
- Monitor the sound being played, allowing one program to
record what other programs play.
Processing is configured on a per sub-device basis, meaning that the sound of
all programs connected to the same sub-device will be processed according to
the same configuration. Multiple sub-devices can be defined, allowing multiple
configurations to coexist. The user selects the configuration a given program
will use by selecting the sub-device the program uses.
exposes MIDI thru boxes (hubs), allowing
programs to send MIDI messages to each other or to hardware MIDI ports in a
exposes a control MIDI port usable
- Volume control.
- Common clock source for audio and MIDI programs.
- Start, stop and relocate groups of audio programs.
The options are as follows:
- Control whether sndiod opens
the audio device or the MIDI port only when needed or keeps it open all
the time. If the flag is on then the
audio device or MIDI port is kept open all the time, ensuring no other
program can steal it. If the flag is off,
then it's automatically closed, allowing other programs to have direct
access to the audio device, or the device to be disconnected. The default
- The buffer size of the audio device in frames. A frame
consists of one sample for each channel in the stream. This is the number
of frames that will be buffered before being played and thus controls the
playback latency. The default is 7680 or twice the block size
(-z), if the block size is set.
- The range of channel numbers for recording and playback
directions, respectively any client is allowed to use. This is a subset of
the audio device channels. The default is 0:1, i.e. stereo.
- Enable debugging to standard error, and do not disassociate
from the controlling terminal. Can be specified multiple times to further
increase log verbosity.
- Attempt to configure the device to use this encoding. The
default is s16. Encoding names use the
following scheme: signedness (s or
u) followed by the precision in bits, the
byte-order (le or
be), the number of bytes per sample, and
the alignment (msb or
lsb). Only the signedness and the
precision are mandatory. Examples: u8,
- Add this
device to devices used for playing and/or recording. Preceding per-device
options (-aberwz) apply to this device.
Sub-devices (-s) that are applied after will
be attached to this device. Device mode and parameters are determined from
sub-devices attached to it.
- Control whether program channels are joined or expanded if
the number of channels requested by a program is not equal to the device
number of channels. If the flag is off
then client channels are routed to the corresponding device channel,
possibly discarding channels not present in the device. If the flag is
on, then a single client channel may be
sent on multiple device channels, or multiple client channels may be sent
to a single device channel. For instance, this feature could be used for
mono to stereo conversions. The default is
- Specify a local network address
sndiod should listen on;
sndiod will listen on TCP port 11025+n, where
n is the unit number specified with -U.
Without this option, sndiod listens on the
UNIX-domain socket only, and is not reachable from
any network. If the option argument is ‘-’ then
sndiod will accept connections from any
address. As the communication is not secure, this option is only suitable
for local networks where all hosts and users are trusted.
- Set the sub-device mode. Valid modes are
mon, corresponding to playback, recording
and monitoring. A monitoring stream is a fake recording stream
corresponding to the mix of all playback streams. Multiple modes can be
specified, separated by commas, but the same sub-device cannot be used for
both recording and monitoring. The default is
- Expose the given MIDI port. This allows multiple programs
to share the port.
- Attempt to force the device to use this sample rate in
Hertz. The default is 48000.
- Add name to the list of
sub-devices to expose. This allows clients to use
sndiod instead of the physical audio device
for audio input and output in order to share the physical device with
other clients. Defining multiple sub-devices allows splitting a physical
audio device into sub-devices having different properties (e.g. channel
ranges). The given name corresponds to
the “option” part of the
- Select the way clients are controlled by MIDI Machine
Control (MMC) messages received by sndiod. If
the mode is off (the default), then
programs are not affected by MMC messages. If the mode is
slave, then programs are started
synchronously by MMC start messages; additionally, the server clock is
exposed as MIDI Time Code (MTC) messages allowing MTC-capable software or
hardware to be synchronized to audio programs.
- Unit number. Each sndiod
server instance has an unique unit number, used in
names. The default is 0.
- Software volume attenuation of playback. The value must be
between 1 and 127, corresponding to -42dB and -0dB attenuation in 1/3dB
steps. Clients inherit this parameter. Reducing the volume in advance
allows a client's volume to stay independent from the number of clients as
long as their number is small enough. 18 volume units (i.e. -6dB
attenuation) allows the number of playback programs to be doubled. The
default is 118 i.e. -3dB.
- Control sndiod behaviour when
the maximum volume of the hardware is reached and a new program starts
playing. This happens only when volumes are not properly set using the
-v option. If the flag is
on, then the master volume is
automatically adjusted to avoid clipping. Using
off makes sense in the rare situation
where all programs lower their volumes. The default is
- The audio device block size in frames. This is the number
of frames between audio clock ticks, i.e. the clock resolution. If a
sub-device is created with the -t option, and
MTC is used for synchronization, the clock resolution must be 96, 100 or
120 ticks per second for maximum accuracy. For instance, 100 ticks per
second at 48000Hz corresponds to a 480 frame block size. The default is
960 or half of the buffer size (-b), if the
buffer size is set.
On the command line, per-device parameters
) must precede the device definition
), and per-sub-device parameters
) must precede the sub-device definition
). Sub-device definitions
) must follow the definition of the device
) to which they are attached.
If no audio devices (-f
) are specified, settings
are applied as if the default device is specified. If no sub-devices
) are specified for a device, a default
sub-device is created attached to it. If a device
) is defined twice, both definitions are
merged: parameters of the first one are used but sub-devices
) of both definitions are created. The default
and the default sub-device exposed by sndiod
, it terminates.
By default, when the program cannot accept recorded data fast enough or cannot
provide data to play fast enough, the program is paused, i.e. samples that
cannot be written are discarded and samples that cannot be read are replaced
by silence. If a sub-device is created with the
option, then recorded samples are discarded,
but the same amount of silence will be written once the program is unblocked,
in order to reach the right position in time. Similarly silence is played, but
the same amount of samples will be discarded once the program is unblocked.
This ensures proper synchronization between programs.
creates a MIDI port with the same name as
the exposed audio sub-device to which MIDI programs can connect.
exposes the audio device clock and allows
audio device properties to be controlled through MIDI.
A MIDI channel is assigned to each stream, and the volume is changed using the
standard volume controller (number 7). Similarly, when the audio client
changes its volume, the same MIDI controller message is sent out; it can be
used for instance for monitoring or as feedback for motorized faders.
The master volume can be changed using the standard master volume system
Streams created with the -t
option are controlled
by the following MMC messages:
- This message is ignored by audio
sndiod clients, but the given time position
is sent to MIDI ports as an MTC “full frame” message forcing
all MTC-slaves to relocate to the given position (see below).
- Put all streams in starting mode. In this mode,
sndiod waits for all streams to become ready
to start, and then starts them synchronously. Once started, new streams
can be created (sndiod) but they will be
blocked until the next stop-to-start transition.
- Put all streams in stopped mode (the default). In this
mode, any stream attempting to start playback or recording is paused.
Client streams that are already started are not affected until they stop
and try to start again.
Streams created with the -t
option export the
device clock using MTC, allowing non-audio
software or hardware to be synchronized to the audio stream. Maximum accuracy
is achieved when the number of blocks per second is equal to one of the
standard MTC clock rates (96, 100 and 120Hz). The following sample rates
) and block sizes
) are recommended:
- 44100Hz, 441 frames (MTC rate is 100Hz)
- 48000Hz, 400 frames (MTC rate is 120Hz)
- 48000Hz, 480 frames (MTC rate is 100Hz)
- 48000Hz, 500 frames (MTC rate is 96Hz)
For instance, the following command will create two devices: the default
and a MIDI-controlled
$ sndiod -r 48000 -z 400 -s default -t slave -s mmc
Streams connected to snd/0
while streams connected to snd/0.mmc
the MMC start signal and start synchronously. Regardless of which device a
stream is connected to, its playback volume knob is exposed.
Start server using default parameters, creating an additional sub-device for
output to channels 2:3 only (rear speakers on most cards), exposing the
$ sndiod -s default -c 2:3 -s rear
Start server creating the default sub-device with low volume and an additional
sub-device for high volume output, exposing the
$ sndiod -v 65 -s default -v 127 -s max
Start server configuring the audio device to use a 48kHz sample frequency,
240-frame block size, and 2-block buffers. The corresponding latency is 10ms,
which is the time it takes the sound to propagate 3.5 meters.
$ sndiod -r 48000 -b 480 -z 240
Resampling is low quality; down-sampling especially should be avoided when
Processing is done using 16-bit arithmetic, thus samples with more than 16 bits
are rounded. 16 bits (i.e. 97dB dynamic) are largely enough for most
applications though. Processing precision can be increased to 24-bit at
compilation time though.
If -a off
creates sub-devices to expose first and
then opens the audio hardware on demand. Technically, this allows
to attempt to use one of the sub-devices
it exposes as an audio device, creating a deadlock. There's nothing to prevent
the user from shooting himself in the foot by creating such a deadlock.