RedFreelancer
commented
3 years ago Here is my log for the config file:
// Sample Configuration File for Shairport Sync
// Commented out settings are generally the defaults, except where noted.
// 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
// ignore_volume_control = "no"; // 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 o$// volume_max_db = 0.0 ; // use this advanced setting, which must have a decimal point in it, to set the maximum volume, in dB, you wish to use.
// The setting is for the hardware mixer, if chosen, or the software mixer otherwise. The value must be in the mixer's range (0.0 to -96.2 for the software mixer).
// Leave it commented out to use mixer's maximum volume.
// volume_control_profile = "standard" ; // use this advanced setting to specify how the airplay volume is transferred to the mixer volume.
// "standard" makes the volume change more quickly at lower volumes and slower at higher volumes.
// "flat" makes the volume change at the same rate at all volumes.
// run_this_when_volume_is_set = "/full/path/to/application/and/args"; // Run the specified application whenever the volume control is set or changed.
// The desired AirPlay volume is appended to the end of the command line – leave a space if you want it treated as an extra argument.
// AirPlay volume goes from 0 to -30 and -144 means "mute".
// regtype = "_raop._tcp"; // Use this advanced setting to set the service type and transport to be advertised by Zeroconf/Bonjour. Default is "_raop._tcp".
// playback_mode = "stereo"; // This can be "stereo", "mono", "reverse stereo", "both left" or "both right". Default is "stereo".
// alac_decoder = "hammerton"; // This can be "hammerton" or "apple". This advanced setting allows you to choose
// the original Shairport decoder by David Hammerton or the Apple Lossless Audio Codec (ALAC) decoder written by Apple.
// interface = "name"; // Use this advanced setting to specify the interface on which Shairport Sync should provide its service. Leave it commented out to get the default, which is to$
// 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.$// audio_backend_buffer_desired_length_in_seconds = 0.15; // If set too small, buffer underflow occurs on low-powered machines. Too long and the response time to volume changes become$// 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.$// dbus_service_bus = "system"; // The Shairport Sync dbus interface, if selected at compilation, will appear
// as "org.gnome.ShairportSync" on the whichever bus you specify here: "system" (default) or "session".
// mpris_service_bus = "system"; // The Shairport Sync mpris interface, if selected at compilation, will appear
// as "org.gnome.ShairportSync" on the whichever bus you specify here: "system" (default) or "session".
};
// Advanced parameters for controlling how Shairport Sync runs a play session
sessioncontrol =
{
// run_this_before_play_begins = "/full/path/to/application and args"; // make sure the application has executable permission. If it's a script, include the shebang (#!/bin/...) on th$// 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 f$// wait_for_completion = "no"; // set to "yes" to get Shairport Sync to wait until the "run_this..." applications have terminated before continuing
// allow_session_interruption = "no"; // set to "yes" to allow another device to interrupt Shairport Sync while it's playing from an existing audio source
// session_timeout = 120; // wait for this number of seconds after a source disappears before terminating the session and becoming available again.
};
// Back End Settings
// These are parameters for the "alsa" audio back end.
alsa =
{
// output_device = "hw:USB PnP Sound Device"; // the name of the alsa output device. Use "alsamixer" or "aplay" to find out the names of devices, mixers, etc.
// mixer_control_name = "PCM"; // the name of the mixer to use to adjust output volume. If not specified, volume in adjusted in software.
// mixer_device = "default"; // the mixer_device default is whatever the output_device is. Normally you wouldn't have to use this.
// output_rate = 44100; // can be 44100, 88200, 176400 or 352800, but the device must have the capability.
// output_format = "S16"; // can be "U8", "S8", "S16", "S24", "S24_3LE", "S24_3BE" or "S32", but the device must have the capability. Except where stated using (*LE or *BE), endiannes$// 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 $};
// Parameters for the "sndio" audio back end. All are optional.
sndio =
{
// device = "snd/0"; // optional setting to set the name of the output device. Default is the sndio system default.
// rate = 44100; // optional setting which can be 44100, 88200, 176400 or 352800, but the device must have the capability. Default is 44100.
// format = "S16"; // optional setting which can be "U8", "S8", "S16", "S24", "S24_3LE", "S24_3BE" or "S32", but the device must have the capability. Except where stated using (*LE o$// 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.
mikebrady
Hi all, im new to the whole coding scene and wanted to jump right in but i keep getting the same issue. I want to play the audio out usb aux but it keeps deciding that it wants to play through HDMI my usb is hw:2
please find below my -h log