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configuration.md

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Configuration

Sources

Audio sources are added to the server as source in the [stream] section of the configuration file /etc/snapserver.conf. Every source must be fed with a fixed sample format, that can be configured per stream (e.g. 48000:16:2).

The following notation is used in this paragraph:

  • <angle brackets>: the whole expression must be replaced with your specific setting
  • [square brackets]: the whole expression is optional and can be left out
  • [key=value]: if you leave this option out, "value" will be the default for "key"

The general format of an audio source is:

TYPE://host/path?name=<name>[&codec=<codec>][&sampleformat=<sampleformat>][&chunk_ms=<chunk ms>]

parameters have the form key=value, they are concatenated with an & character. Parameter name is mandatory for all sources, while codec, sampleformat and chunk_ms are optional and will override the default codec, sampleformat or chunk_ms settings. Non blocking sources support the dryout_ms parameter: when no new data is read from the source, send silence to the clients

Available audio source types are:

pipe

Captures audio from a named pipe

pipe:///<path/to/pipe>?name=<name>[&mode=create][&dryout_ms=2000]

mode can be create or read. Sometimes your audio source might insist in creating the pipe itself. So the pipe creation mode can by changed to "not create, but only read mode", using the mode option set to read

librespot

Launches librespot and reads audio from stdout

librespot:///<path/to/librespot>?name=<name>[&dryout_ms=2000][&username=<my username>&password=<my password>][&devicename=Snapcast][&bitrate=320][&wd_timeout=7800][&volume=100][&onevent=""][&normalize=false][&autoplay=false][&cache=""][&disable_audio_cache=false][&killall=false][&params=extra-params]

Note that you need to have the librespot binary on your machine and the sampleformat will be set to 44100:16:2

Available parameters

Parameters used to configure the librespot binary (see librespot-org options):

  • username: Username to sign in with
  • password: Password
  • devicename: Device name
  • bitrate: Bitrate (96, 160 or 320). Defaults to 320
  • volume: Initial volume in %, once connected [0-100]
  • onevent: The path to a script that gets run when one of librespot's events is triggered
  • normalize: Enables volume normalisation for librespot
  • autoplay: Autoplay similar songs when your music ends
  • cache: Path to a directory where files will be cached
  • disable_audio_cache: Disable caching of the audio data
  • params: Optional string appended to the librespot invocation. This allows for arbitrary flags to be passed to librespot, for instance params=--device-type%20avr. The value has to be properly URL-encoded.

Parameters introduced by Snapclient:

  • killall: Kill all running librespot instances before launching librespot
  • wd_timeout: Restart librespot if it doesn't create log messages for x seconds

airplay

Launches shairport-sync and reads audio from stdout

airplay:///<path/to/shairport-sync>?name=<name>[&dryout_ms=2000][&devicename=Snapcast][&port=5000][&password=<my password>]

Note that you need to have the shairport-sync binary on your machine and the sampleformat will be set to 44100:16:2

Available parameters

Parameters used to configure the shairport-sync binary:

  • devicename: Advertised name
  • port: RTSP listening port
  • password: Password
  • params: Optional string appended to the shairport-sync invocation. This allows for arbitrary flags to be passed to shairport-sync, for instance params=--on-start=start.sh%20--on-stop=stop.sh. The value has to be properly URL-encoded.

file

Reads PCM audio from a file

file:///<path/to/PCM/file>?name=<name>

process

Launches a process and reads audio from stdout

process:///<path/to/process>?name=<name>[&dryout_ms=2000][&wd_timeout=0][&log_stderr=false][&params=<process arguments>]

Available parameters

  • wd_timeout: kill and restart the process if there was no message logged for x seconds to stderr (0 = disabled)
  • log_stderr: Forward stderr log messages to Snapclient logging
  • params: Params to start the process with

tcp server

Receives audio from a TCP socket (acting as server)

tcp://<listen IP, e.g. 127.0.0.1>:<port>?name=<name>[&mode=server]

default for port (if omitted) is 4953, default for mode is server

Mopdiy configuration would look like this (running GStreamer in client mode)

[audio]
output = audioresample ! audioconvert ! audio/x-raw,rate=48000,channels=2,format=S16LE ! wavenc ! tcpclientsink

tcp client

Receives audio from a TCP socket (acting as client)

tcp://<server IP, e.g. 127.0.0.1>:<port>?name=<name>&mode=client

Mopdiy configuration would look like this (running GStreamer in server mode):

[audio]
output = audioresample ! audioconvert ! audio/x-raw,rate=48000,channels=2,format=S16LE ! wavenc ! tcpserversink

alsa

Captures audio from an alsa device

alsa://?name=<name>&device=<alsa device>[&send_silence=false][&idle_threshold=100]

Available parameters

  • device: alsa device name or identifier, e.g. default or hw:0,0 or hw:0,0,0
  • idle_threshold: switch stream state from playing to idle after receiving idle_threshold milliseconds of silence
  • send_silence: forward silence to clients when stream state is idle

The output of any audio player that uses alsa can be redirected to Snapcast by using an alsa loopback device:

  1. Setup the alsa loopback device by loading the kernel module:

    sudo modprobe snd-aloop

    The loopback device can be created during boot by adding snd-aloop to /etc/modules

  2. The loopback device should show up in aplay -l

    aplay -l
**** List of PLAYBACK Hardware Devices ****
card 0: Loopback [Loopback], device 0: Loopback PCM [Loopback PCM]
Subdevices: 8/8
Subdevice #0: subdevice #0
Subdevice #1: subdevice #1
Subdevice #2: subdevice #2
Subdevice #3: subdevice #3
Subdevice #4: subdevice #4
Subdevice #5: subdevice #5
Subdevice #6: subdevice #6
Subdevice #7: subdevice #7
card 0: Loopback [Loopback], device 1: Loopback PCM [Loopback PCM]
Subdevices: 8/8
Subdevice #0: subdevice #0
Subdevice #1: subdevice #1
Subdevice #2: subdevice #2
Subdevice #3: subdevice #3
Subdevice #4: subdevice #4
Subdevice #5: subdevice #5
Subdevice #6: subdevice #6
Subdevice #7: subdevice #7
card 1: Intel [HDA Intel], device 0: CX20561 Analog [CX20561 Analog]
Subdevices: 1/1
Subdevice #0: subdevice #0
card 1: Intel [HDA Intel], device 1: CX20561 Digital [CX20561 Digital]
Subdevices: 1/1
Subdevice #0: subdevice #0
card 2: CODEC [USB Audio CODEC], device 0: USB Audio [USB Audio]
Subdevices: 0/1
Subdevice #0: subdevice #0

    In this example the loopback device is card 0 with devices 0 and 1, each having 8 subdevices.
    The devices are addressed with hw:<card idx>,<device idx>,<subdevice num>, e.g. hw:0,0,0.
    If a process plays audio using hw:0,0,x, then the audio will be looped back to hw:0,1,x

  3. Configure your player to use a loopback device

    For mopidy (gstreamer) in mopidy.conf:

    output = audioresample ! audioconvert ! audio/x-raw,rate=48000,channels=2,format=S16LE ! alsasink device=hw:0,0,0

    For mpd: in mpd.conf

    audio_output_format            "48000:16:2"
audio_output {
        type            "alsa"
        name            "My ALSA Device"
        device          "hw:0,0,0"      # optional
#       auto_resample   "no"
#       mixer_type      "hardware"      # optional
#       mixer_device    "default"       # optional
#       mixer_control   "PCM"           # optional
#       mixer_index     "0"             # optional
}

  4. Configure Snapserver to capture the loopback device:

    [stream]
source = alsa://?name=SomeName&sampleformat=48000:16:2&device=hw:0,1,0

meta

Read and mix audio from other stream sources

meta:///<name of source#1>/<name of source#2>/.../<name of source#N>?name=<name>

Plays audio from the active source with the highest priority, with source#1 having the highest priority and source#N the lowest.
Use codec=null for stream sources that should only serve as input for meta streams