shairport-sync.conf - keep it simple

shairport-sync.conf

// hifi-pi.local. - shairport-sync config
// general settings
general = {
name = "%h"; 
// This means "Hostname" -- see below. This is the name the service will advertise to iTunes.
// The default is "Hostname" -- i.e. the machine's hostname with the first
// letter capitalised (ASCII only.) You can use the following substitutions:
//  %h for the hostname,
//  %H for the Hostname (i.e. with first letter capitalised (ASCII only)),
//  %v for the version number, e.g. 3.0 and
//  %V for the full version string, e.g. 3.0-OpenSSL-Avahi-ALSA-soxr-metadata-sysconfdir:/etc
// Overall length can not exceed 50 characters. Example: "Shairport Sync %v on %H".

//password = "secret"; 
// leave this commented out if you don't want to require a password

interpolation = "basic"; 
// aka "stuffing". Default is "basic", alternative is "soxr". Use "soxr" only
// if you have a reasonably fast processor.

output_backend = "alsa"; 
// Run "shairport-sync -h" to get a list of all output_backends, e.g. "alsa",
// "pipe", "stdout". The default is the first one.

mdns_backend = "avahi"; 
// Run "shairport-sync -h" to get a list of all mdns_backends. The default is
// the first one.  
//port = 5000; // Listen for service requests on this port
//udp_port_base = 6001; // start allocating UDP ports from this port number when needed
//udp_port_range = 100; 
//  look for free ports in this number of places, starting at the UDP port base.
//  Allow at least 10, though only three are needed in a steady state.
//drift_tolerance_in_seconds = 0.002; 
//  allow a timing error of this number of seconds of drift away from exact
//  synchronisation before attempting to correct it
//resync_threshold_in_seconds = 0.050; 
//  a synchronisation error greater than this number of seconds will cause
//  resynchronisation; 0 disables it

// setting this to "yes" since i'd like to be able to control the
// volume for this exclusively through the amplifier.  
ignore_volume_control = "yes"; 
//  set this to "yes" if you want the volume to be at 100% no matter what the
//  source's volume control is set to.
//volume_range_db = 60 ; 
//  use this advanced setting to set the range, in dB, you want between the
//  maximum volume and the minimum volume. Range is 30 to 150 dB. Leave it
//  commented out to use mixer's native range.
//volume_max_db = 0.0 ; 
//  use this advanced setting, which must have a decimal point in it, to set the
//  maximum volume, in dB, you wish to use.
//  The setting is for the hardware mixer, if chosen, or the software mixer
//  otherwise. The value must be in the mixer's range (0.0 to -96.2 for the
//  software mixer).
//  Leave it commented out to use mixer's maximum volume.
//volume_control_profile = "standard" ; 
//  use this advanced setting to specify how the airplay volume is transferred
//  to the mixer volume.  "standard" makes the volume change more quickly at
//  lower volumes and slower at higher volumes.  "flat" makes the volume change
//  at the same rate at all volumes.
//run_this_when_volume_is_set = "/full/path/to/application/and/args"; 
//  Run the specified application whenever the volume control is set or
//  changed. The desired AirPlay volume is appended to the end of the command
//  line – leave a space if you want it treated as an extra argument.  AirPlay
//  volume goes from 0 to -30 and -144 means "mute".
//regtype = "_raop._tcp"; 
//  Use this advanced setting to set the service type and transport to be
//  advertised by Zeroconf/Bonjour. Default is "_raop._tcp".
//playback_mode = "stereo"; 
//  This can be "stereo", "mono", "reverse stereo", "both left" or "both
//  right". Default is "stereo".
// alac_decoder = "hammerton"; 
//  This can be "hammerton" or "apple". This advanced setting allows you to choose
//  the original Shairport decoder by David Hammerton or the Apple Lossless
//  Audio Codec (ALAC) decoder written by Apple.
//interface = "name"; 
//  Use this advanced setting to specify the interface on which Shairport Sync
//  should provide its service. Leave it commented out to get the default, which
//  is to select the interface(s) automatically.  
//audio_backend_latency_offset_in_seconds = 0.0; 
//  Set this offset to compensate for a fixed delay in the audio back end. E.g.
//  if the output device delays by 100 ms, set this to -0.1.
//audio_backend_buffer_desired_length_in_seconds = 0.15; 
//  If set too small, buffer underflow occurs on low-powered machines. Too long
//  and the response time to volume changes becomes annoying. Default is 0.15
//  seconds in the alsa backend, 0.35 seconds in the pa backend and 1.0 seconds
//  otherwise.
//audio_backend_silent_lead_in_time = 2.0; 
//  This optional advanced setting, from 0.0 and 4.0 seconds, sets the length
//  of the period of silence that precedes the start of the audio. The default
//  is the latency, usually 2.0 seconds. Values greater than the latency are
//  ignored. Values that are too low will affect initial synchronisation.
//dbus_service_bus = "system"; 
//  The Shairport Sync dbus interface, if selected at compilation, will appear
//  as "org.gnome.ShairportSync" on the whichever bus you specify here:
//  "system" (default) or "session".
//mpris_service_bus = "system"; 
//  The Shairport Sync mpris interface, if selected at compilation, will appear
//  as "org.gnome.ShairportSync" on the whichever bus you specify here:
//  "system" (default) or "session".

};

// Advanced parameters for controlling how Shairport Sync runs a play session
sessioncontrol =
{
//run_this_before_play_begins = "/full/path/to/application and args"; 
//  make sure the application has executable permission. If it's a script,
//  include the shebang (#!/bin/...) on the first line
//run_this_after_play_ends = "/full/path/to/application and args"; 
//  make sure the application has executable permission. If it's a script,
//  include the shebang (#!/bin/...) on the first line
//wait_for_completion = "no"; 
//  set to "yes" to get Shairport Sync to wait until the "run_this..."
//  applications have terminated before continuing
//allow_session_interruption = "no"; 
//  set to "yes" to allow another device to interrupt Shairport Sync while it's
//  playing from an existing audio source
//session_timeout = 120; 
//  wait for this number of seconds after a source disappears before
//  terminating the session and becoming available again.
};

// Back End Settings

// These are parameters for the "alsa" audio back end.
alsa = {
//output_device = "snd_rpi_hifiberry_dacplus"; 
 output_device = "default"; 
//  the name of the alsa output device. Use "alsamixer" or "aplay" to find out
//  the names of devices, mixers, etc.
//mixer_control_name = "PCM"; 
//  the name of the mixer to use to adjust output volume. If not specified,
//  volume in adjusted in software.
mixer_device = "default"
//  the mixer_device default is whatever the output_device is. Normally you
//  wouldn't have to use this.
//output_rate = 44100; 
//  can be 44100, 88200, 176400 or 352800, but the device must have the
//  capability.
//output_format = "S16"; 
//  can be "U8", "S8", "S16", "S24", "S24_3LE", "S24_3BE" or "S32", but the
//  device must have the capability. Except where stated using (*LE or *BE),
//  endianness matches that of the processor.
//disable_synchronization = "no"; 
//  Set to "yes" to disable synchronization. Default is "no".
//period_size = <number>; 
//  Use this optional advanced setting to set the alsa period size near to this
//  value
//buffer_size = <number>; 
//  Use this optional advanced setting to set the alsa buffer size near to this
//  value
//use_mmap_if_available = "yes"; 
//  Use this optional advanced setting to control whether MMAP-based output is
//  used to communicate  with the DAC. Default is "yes"
//use_hardware_mute_if_available = "no"; 
//  Use this optional advanced setting to control whether the hardware in the
//  DAC is used for muting. Default is "no", for compatibility with other audio
//  players.
};

// Parameters for the "sndio" audio back end. All are optional.
sndio = {
//device = "snd/0"; 
//  optional setting to set the name of the output device. Default is the sndio
//  system default.
//rate = 44100; 
//  optional setting  which can be 44100, 88200, 176400 or 352800, but the
//  device must have the capability. Default is 44100.
//format = "S16"; 
//  optional setting  which can be "U8", "S8", "S16", "S24", "S24_3LE",
//  "S24_3BE" or "S32", but the device must have the capability. Except where
//  stated using (*LE or *BE), endianness matches that of the processor.
//round = <number>; 
//  advanced optional setting to set the period size near to this value
//bufsz = <number>; 
//  advanced optional setting to set the buffer size near to this value
};

// Parameters for the "pa" PulseAudio  backend.
pa = {
//application_name = "Shairport Sync"; 
//  Set this to the name that should appear in the Sounds "Applications" tab
//  when Shairport Sync is active.
};

// Parameters for the "pipe" audio back end, a back end that directs raw
// CD-style audio output to a pipe. No interpolation is done.
pipe = {
//name = "/path/to/pipe"; 
//  there is no default pipe name for the output
};

// These are no configuration file parameters for the "stdout" audio back end.
// No interpolation is done.  These are no configuration file  parameters for
// the "ao" audio back end.  No interpolation is done.  Static latency settings
// are deprecated and the settings have been removed.  

dsp = {

//////////////////////////////////////////
// This convolution filter can be used to apply almost any correction to the
// audio signal, like frequency and phase correction.  For example you could
// measure (with a good microphone and a sweep-sine) the frequency response of
// your speakers + room, and apply a correction to get a flat response curve.
//////////////////////////////////////////
//
//convolution = "yes";                  // Activate the convolution filter.
//convolution_ir_file = "impulse.wav";  // Impulse Response file to be convolved to the audio stream
//convolution_gain = -4.0;              // Static gain applied to prevent clipping during the convolution process
//convolution_max_length = 44100;       // Truncate the input file to this length in order to save CPU.


//////////////////////////////////////////
// This loudness filter is used to compensate for human ear non linearity.
// When the volume decreases, our ears loose more sentisitivity in the low
// range frequencies than in the mid range ones.  This filter aims at
// compensating for this loss, applying a variable gain to low frequencies
// depending on the volume.  More info can be found here:
// https://en.wikipedia.org/wiki/Equal-loudness_contour For this filter to work
// properly, you should disable (or set to a fix value) all other volume
// control and only let shairport-sync control your volume.  The setting
// "loudness_reference_volume_db" should be set at the volume reported by
// shairport-sync when listening to music at a normal listening volume.
//////////////////////////////////////////
//
//loudness = "yes";                     
// Activate the filter
//loudness_reference_volume_db = -20.0; 
//  Above this level the filter will have no effect anymore. Below this level it
//  will gradually boost the low frequencies.

};

// How to deal with metadata, including artwork
metadata = {
//enabled = "no"; 
//  set this to yes to get Shairport Sync to solicit metadata from the source
//  and to pass it on via a pipe
//include_cover_art = "no"; 
//  set to "yes" to get Shairport Sync to solicit cover art from the source and
//  pass it via the pipe. You must also set "enabled" to "yes".
//pipe_name = "/tmp/shairport-sync-metadata";
//pipe_timeout = 5000; 
//  wait for this number of milliseconds for a blocked pipe to unblock before
//  giving up
//socket_address = "226.0.0.1"; 
//  if set to a host name or IP address, UDP packets containing metadata will
//  be sent to this address. May be a multicast address. "socket-port" must be
//  non-zero and "enabled" must be set to yes"
//socket_port = 5555; 
//  if socket_address is set, the port to send UDP packets to
//socket_msglength = 65000; 
//  the maximum packet size for any UDP metadata. This will be clipped to be
//  between 500 or 65000. The default is 500.
};

// Diagnostic settings. These are for diagnostic and debugging only. Normally you sould leave them commented out
diagnostics = {
//disable_resend_requests = "no"; 
//  set this to yes to stop Shairport Sync from requesting the retransmission
//  of missing packets. Default is "no".
//statistics = "no"; 
//  set to "yes" to print statistics in the log
//log_verbosity = 0; 
//  "0" means no debug verbosity, "3" is most verbose.
//log_show_time_since_startup = "no"; 
//  set this to yes if you want the time since startup in the debug message --
//  seconds down to nanoseconds
//log_show_time_since_last_message = "no"; 
//  set this to yes if you want the time since the last debug message in the
//  debug message -- seconds down to nanoseconds
//drop_this_fraction_of_audio_packets = 0.0; 
//  use this to simulate a noisy network where this fraction of UDP packets are
//  lost in transmission. E.g. a value of 0.001 would mean an average of 0.1%
//  of packets are lost, which is actually quite a high figure.
};