// Sample Configuration File for Shairport Sync
// Commented out settings are generally the defaults, except where noted.
// Some sections are operative only if Shairport Sync has been built with the right configuration flags.
// See the individual sections for details.

// 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. 3.3-OpenSSL-Avahi-ALSA-soxr-metadata-sysconfdir:/etc
//		Overall length can not exceed 50 characters. Example: "Shairport Sync %v on %H".
//	password = "secret"; // (AirPlay 1 only) leave this commented out if you don't want to require a password
//	The interpolation setting below controls how Shairport Sync adds or removes frames of audio to keep in sync.
//			"auto" (default) measures the processor's floating point speed and chooses "soxr" if available and it is fast enough. Otherwise, "vernier" is selected.
//			"soxr" uses the SoX library to recode a packet of frames to a new packet containing more or fewer frames. This needs a processor with fast floating point capability.
//			"vernier" recodes a packet of frames to a new packet containing more or fewer frames. This is recommended for low powered devices.
//			"basic" causes the simple removal or insertion of frames in a packet of frames. Not recommended.
//	interpolation = "auto"; // aka "stuffing". Default is "auto". Alternatives are "vernier", "basic" or "soxr". Choose "soxr" only if you have a reasonably fast processor and Shairport Sync has been built with "soxr" support.
//	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.
//	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.
//	port = <number>; // Listen for service requests on this port. 5000 for AirPlay 1, 7000 for AirPlay 2
//	udp_port_base = 6001; // (AirPlay 1 only) start allocating UDP ports from this port number when needed 
//	udp_port_range = 10; // (AirPlay 1 only) 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.
//	airplay_device_id_offset = 0; // (AirPlay 2 only) add this to the default airplay_device_id calculated from one of the device's MAC address
//	airplay_device_id = 0x<six-digit_hexadecimal_number>L; // (AirPlay 2 only) use this as the airplay_device_id e.g. 0xDCA632D4E8F3L -- remember the "L" at the end as it's a 64-bit quantity!
//	regtype = "<string>"; // Use this advanced setting to set the service type and transport to be advertised by Zeroconf/Bonjour. Default is "_raop._tcp" for AirPlay 1, "_airplay._tcp" for AirPlay 2.
//	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
//	playback_mode = "stereo"; // This can be "stereo", "mono", "reverse stereo", "both left" or "both right". Default is "stereo".
//	For FFmpeg channel and layout names, (e.g. "7.1", "FL", "3.0(back)", etc.), please see channel_names and channel_layout_map at https://ffmpeg.org/doxygen/trunk/channel__layout_8c_source.html
//	eight_channel_mode = "on"; // Enable reception of eight channel audio. Can be "off", "on" or an eight-channel FFmpeg channel layout. If "on", the channel layout used is: "7.1".
//	six_channel_mode = "on"; // Enable reception of six channel audio. Can be "off", "on" or a six-channel FFmpeg channel layout. If "on", the channel layout used is: "5.1".
//	mixdown = "auto"; // Enable mixdown. Can be "auto", "off" or an FFmpeg channel layout, e.g. "quad". If "auto", mixdown will occur, if needed, to the default channel layout for the output channels available.
//	output_channel_mapping = "auto"; // Specify how audio channels are mapped to the output device's channels:
//	  Shairport Sync uses standard FFmpeg channel names for each channel in the audio output. The names are, in order, "FL", "FR", "FC", "LFE", "BL", "BR", "SL", "SR".
//	  If "auto", the audio channels are matched, where possible, to the channels in the device's channel map. Any leftover output channels are mapped, in order, to leftover device channels.
//	  If "off", or if there is no device channel map, audio channels are output to the device channels in order.
//	  If a list of audio channels is given, e.g. ( "FL", "FR", "LFE", "FC", "BL", "BR", "SL", "SR" ), they are  mapped in the order given to the device channels from 1 upwards.
//	  The audio channel list can include the same channel more than once and can include the silent channel "--".

//	alac_decoder = <setting>; //This advanced setting is for Classic Airplay only. It can be "ffmpeg" (default on systems with FFmpeg support), "hammerton" (default on systems without FFmpeg support, deprecated) or "apple" (deprecated).
//		The original Shairport decoder is by David Hammerton.  This is deprecated for security reasons as it is no longer maintained. However, it is compact and may be useful in storage-constrained devices. It is included by default on systems without FFmpeg support.
//		If you build Shairport Sync with the flag --with-apple-alac, the Apple ALAC decoder will be chosen by default unless FFmpeg support is included. The Apple ALAC decoder is deprecated for security reasons.
//		If you build Shairport Sync with the flag --with-ffmpeg, the FFmpeg ALAC decoder will be chosen by default. This is recommended except where there is insufficient storage on the device. With this setting, the Hammerton decoder will not be built. 
//		In AirPlay 2 operation, the FFmpeg ALAC decoder is always used.
//	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 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.
//		"dasl_tapered" is similar to "standard" - it makes the volume change more quickly at lower volumes and slower at higher volumes.
//			The basic idea behind dasl_tapered is that a given percentage change in volume should result in the same percentage change in
//			perceived loudness. For instance, doubling the volume level should result in doubling the perceived loudness.
//			With the range of AirPlay volume being from -30 to 0, doubling the volume from -22.5 to -15 results in an increase of 10 dB.
//			Similarly, doubling the volume from -15 to 0 results in an increase of 10 dB.
//			For compatibility with mixers having a restricted attenuation range (e.g. 30 dB), "dasl_tapered" will switch to a flat profile at low AirPlay volumes.
//	volume_control_combined_hardware_priority = "no"; // when extending the volume range by combining the built-in software attenuator with the hardware mixer attenuator, set this to "yes" to reduce volume by using the hardware mixer first, then the built-in software attenuator.
//	default_airplay_volume = -24.0; // this is the suggested volume after a reset or after the high_volume_threshold has been exceed and the high_volume_idle_timeout_in_minutes has passed
//	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 0.0 to -30.0 and -144.0 means "mute".
//	audio_backend_latency_offset_in_seconds = 0.0; // This is added to the latency requested by the player to delay or advance the output by a fixed amount.
//		Use it, for example, to compensate for a fixed delay in the audio back end.
//		E.g. if the output device, e.g. a soundbar, takes 100 ms to process audio, set this to -0.1 to deliver the audio
//		to the output device 100 ms early, allowing it time to process the audio and output it perfectly in sync.
//	audio_backend_buffer_desired_length_in_seconds = 0.2; // This is the desired size of the buffer to be maintained in the external output system, e.g. the DAC in ALSA. If set too small, buffer underflow occurs on low-powered machines.
//		Too long and the response time to volume changes becomes annoying.
//	audio_decoded_buffer_desired_length_in_seconds = 1.0; // Advanced feature. This is the desired size of the buffer of fully deciphered and decoded audio maintained within Shairport Sync prior to sending it to the external output system , e.g. the DAC in ALSA.
//		Valid for AirPlay 2 Buffered Audio streams only.
//	audio_backend_buffer_interpolation_threshold_in_seconds = 0.075; // Advanced feature. If the buffer size drops below this, stop using time-consuming interpolation like soxr to avoid dropouts due to underrun.
//	audio_backend_silent_lead_in_time = "auto"; // This optional advanced setting, either "auto" or a positive number, sets the length of the period of silence that precedes the start of the audio.
//		The default is "auto" -- the silent lead-in starts as soon as the player starts sending packets.
//		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".
//	resend_control_first_check_time = 0.10; // Use this optional advanced setting to set the wait time in seconds before deciding a packet is missing.
//	resend_control_check_interval_time = 0.25; //  Use this optional advanced setting to set the time in seconds between requests for a missing packet.
//	resend_control_last_check_time = 0.10; // Use this optional advanced setting to set the latest time, in seconds, by which the last check should be done before the estimated time of a missing packet's transfer to the output buffer.
//	missing_port_dacp_scan_interval_seconds = 2.0; // Use this optional advanced setting to set the time interval between scans for a DACP port number if no port number has been provided by the player for remote control commands
};

// Advanced parameters for controlling how Shairport Sync stays active and how it runs a session
sessioncontrol =
{
//	"active" state starts when play begins and ends when the active_state_timeout has elapsed after play ends, unless another play session starts before the timeout has fully elapsed.
//	run_this_before_entering_active_state = "/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_exiting_active_state = "/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
//	active_state_timeout = 10.0; // wait for this number of seconds after play ends before leaving the active state, unless another play session begins.

//	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

//	run_this_if_an_unfixable_error_is_detected = "/full/path/to/application and args"; // if a problem occurs that can't be cleared by Shairport Sync itself, hook a program on here to deal with it.
//	  An error code-string is passed as the last argument.
//	  Many of these "unfixable" problems are caused by malfunctioning output devices, and sometimes it is necessary to restart the whole device to clear the problem.
//	  You could hook on a program to do this automatically, but beware -- the device may then power off and restart without warning!
//	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 = 60; // wait for this number of seconds after a source disappears before terminating the session and becoming available again.
};

// Back End Settings

// Rates, Formats and Channels
// Shairport Sync can handle a wide range of wide range of output rates, formats and channels.

// If it has been built for AirPlay 2 or with FFmpeg support, Shairport Sync can output in a wide range settings, including 48,000 and 44,100 frames per second, 32- and 24-bit sample sizes, 1 to 8 channels.
// Otherwise, if Shairport Sync has been built (1) for classic AirPlay only and (2) without FFmpeg support, it is restricted to output rates of 44,100 frames per second or exact multiples and two channels only.

// You can check Shairport Sync's support for FFmpeg by entering the command: shairport-sync -V
// If the resulting string contains "FFmpeg" or "AirPlay2", then it has FFmpeg support built in.

// With FFmpeg support:
//     Possible output rates are: 5512, 8000, 11025, 16000, 22050, 32000, 44100, 48000, 64000, 88200, 96000, 176400, 192000, 352800 and 384000 frames per seconds.
//     Possible output formats are:   "S8", "U8", "S16_LE", "S16_BE", "S24_LE", "S24_BE", "S24_3LE", "S24_3BE", "S32_LE" and "S32_BE".
//     Possible output channel counts are: 1 to 8.

// Without FFmpeg support:
//     Possible rates are: 44100, 88200, 176400 and 352800 frames per seconds.
//     Possible formats are:   "S8", "U8", "S16_LE", "S16_BE", "S24_LE", "S24_BE", "S24_3LE", "S24_3BE", "S32_LE" and "S32_BE".
//     Possible channel counts are: 2.

// Automatic settings
//     Shairport Sync will dynamically select output formats, attempting to match input and output rates, formats and channel counts, picking the best alternatives otherwise.
//     (Settings are static by default on the STDOUT and pipe backends, as there is no obvious way to signal a downstream consumer of the data when the format has changed.)

// Rate Selection:
//     Shairport Sync checks the rates the output system can accept and those that have been specified in the configuration file.
//     From that set of possibilities, Shairport Sync will attempt to match the rate at which the audio is being received and will switch the output to that rate if necessary.
//     If the exact rate is not available, an exact multiple will be selected if available. Finally, and only with FFmpeg support, a higher rate or a lower rate will be chosen.
//     To avoid output rate switching, specify just one rate in the configuration file.

// Format Selection:
//     Shairport Sync checks the formats the output system can accept and those that have been specified in the configuration file.
//     From that set of possibilities, Shairport Sync will use the deepest format unless ignore_volume_control is true and maximum_volume is not used, if which case it will try to switch the output to the exact format of the incoming audio.
//     To avoid output format switching, specify just one format in the configuration file.

// Channel Count Selection:
//     Shairport Sync checks the channels counts the output system can accept and those that have been specified in the configuration file.
//     From that set of possibilities, Shairport Sync will attempt to match the output channels to the number of channels in the audio and will switch the output to that number of channels if necessary.
//     If the exact number of output channels is not available, a greater output channel count will be selected if available. Failing that, a lower channel count will be chosen.
//     To avoid channel count switching, specify just one channel count in the configuration file.


// These are parameters for the "alsa" audio back end.
// For this section to be operative, Shairport Sync must be built with the following configuration flag:
// --with-alsa
alsa =
{
//	output_device = "default"; // the name of the alsa output device. Use "shairport-sync -h" to discover the names of ALSA hardware devices. 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. No default. If not specified, no mixer is used and volume in adjusted in software.
//	mixer_control_index = 0; // the index of the mixer to use to adjust output volume. Default is 0. The mixer is fully identified by the combination of the mixer_control_name and the mixer_control_index, e.g. "PCM",0 would be such a specification.
//	mixer_device = "default"; // the mixer_device default is whatever the output_device is. Normally you wouldn't have to use this.

//	Note: if you specify settings here, the output device must be capable of them. Otherwise, Shairport Sync will quit and leave a message in the system log.
//	output_rate = "auto"; // Specify "auto", or a single rate, e.g. 48000, or a bracketed comma-separated list of rates, e.g. (44100, 48000, 64000). Default is "auto" -- try to match the input. See the "Rates, Formats and Channels" discussion above.
//	output_format = "auto"; // Specify "auto", or a single format, e.g. "S32_LE", or a bracketed comma-separated list of formats, e.g. ("S32_LE", "S16_LE"). Default is "auto". See the "Rates, Formats and Channels" discussion above.
//	output_channels = "auto"; // Specify "auto", or a specific number of channels, e.g. 2, or a bracketed comma-separated list of numbers of channels, e.g. (2, 6). Default is "auto" -- try to match the input. See the "Rates, Formats and Channels" discussion above.

//	disable_synchronization = "no"; // Set to "yes" to disable synchronization.

//	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 = "no"; // Use this optional advanced setting to control whether MMAP-based output is used to communicate  with the DAC. Default is "no".
//	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.
//	maximum_stall_time = 0.200; // Use this optional advanced setting to control how long to wait for data to be consumed by the output device before considering it an error. It should never approach 200 ms.
//	use_precision_timing = "auto"; // Use this optional advanced setting to control how Shairport Sync gathers timing information. When set to "auto", if the output device is a real hardware device, precision timing will be used. Choose "no" for more compatible standard timing, choose "yes" to force the use of precision timing, which may cause problems.

//	disable_standby_mode = "never"; // This setting prevents the DAC from entering the standby mode. Some DACs make small "popping" noises when they go in and out of standby mode. Settings can be: "always", "auto" or "never". Default is "never", but only for backwards compatibility. The "auto" setting prevents entry to standby mode while Shairport Sync is in the "active" mode. You can use "yes" instead of "always" and "no" instead of "never".
//	disable_standby_mode_silence_threshold = 0.040; // Use this optional advanced setting to control how little audio should remain in the output buffer before the disable_standby code should start sending silence to the output device.
//	disable_standby_mode_silence_scan_interval = 0.030; // Use this optional advanced setting to control how often the amount of audio remaining in the output buffer should be checked.
//	disable_standby_mode_default_channels = 2; // Use this optional advanced setting to set the initial channel setting when disable_standby_mode is "always" or "yes". After a track has been played, the track's output channel setting will be used.
//	disable_standby_mode_default_rate = <rate>; // Use this optional advanced setting to set the initial rate, in frames per second, when disable_standby_mode is "always" or "yes". Default is 44100 for classic AirPlay, 48000 for AirPlay 2. After a track has been played, the track's output rate setting will be used.
};

// Parameters for the PipeWire backend.
// For this section to be operative, Shairport Sync must be built with the following configuration flag:
// --with-pipewire
pipewire =
{
//	application_name = "Shairport Sync"; // Set this to the name that should appear in the Sounds "Applications" or "Volume Levels".
//	node_name = "Shairport Sync"; // This appears in some PipeWire CLI tool outputs.
//	sink_target = "<sink target name>"; // Leave this commented out to get the sink target already chosen by the PipeWire system.

//	Note: if you specify settings here, the the PipeWire system must be capable of giving effect to them. Otherwise, Shairport Sync will quit and leave a message in the system log.
//	output_rate = "auto"; // Specify "auto", or a single rate, e.g. 48000, or a bracketed comma-separated list of rates, e.g. (44100, 48000, 64000). Default is "auto" -- try to match the input. See the "Rates, Formats and Channels" discussion above.
//	output_format = "auto"; // Specify "auto", or a single format, e.g. "S32_LE", or a bracketed comma-separated list of formats, e.g. ("S32_LE", "S16_LE"). Default is "auto". See the "Rates, Formats and Channels" discussion above.
//	output_channels = "auto"; // Specify "auto", or a specific number of channels, e.g. 2, or a bracketed comma-separated list of numbers of channels, e.g. (2, 6). Default is "auto" -- try to match the input. See the "Rates, Formats and Channels" discussion above.
};

// Parameters for the "sndio" audio back end. All are optional.
// For this section to be operative, Shairport Sync must be built with the following configuration flag:
// --with-sndio
sndio =
{
//	device = "default"; // optional setting to set the name of the output device, e.g. "rsnd/0", "rsnd/1", etc.
//	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

//	Note: if you specify settings here, the sndio system must be capable of giving effect to  them. Otherwise, Shairport Sync will quit and leave a message in the system log.
//	output_rate = "auto"; // Specify "auto", or a single rate, e.g. 48000, or a bracketed comma-separated list of rates, e.g. (44100, 48000, 64000). Default is "auto" -- try to match the input. See the "Rates, Formats and Channels" discussion above.
//	output_format = "auto"; // Specify "auto", or a single format, e.g. "S32_LE", or a bracketed comma-separated list of formats, e.g. ("S32_LE", "S16_LE") from the list: "S32_LE", "S32_BE", "S16_LE" or "S16_BE". Default is "auto". See the "Rates, Formats and Channels" discussion above.
//	output_channels = "auto"; // Specify "auto", or a specific number of channels, e.g. 2, or a bracketed comma-separated list of numbers of channels, e.g. (2, 6). Default is "auto" -- try to match the input. See the "Rates, Formats and Channels" discussion above.
};

// Parameters for the PulseAudio  backend.
// For this section to be operative, Shairport Sync must be built with the following configuration flag:
// --with-pulseaudio
pulseaudio =
{
//	server = "host"; // Set this to override the default pulseaudio server that should be used.
//	sink = "Sink Name"; // Set this to override the default pulseaudio sink that should be used. (Untested)
//	application_name = "Shairport Sync"; //Set this to the name that should appear in the Sounds "Applications" tab when Shairport Sync is active.

//	Note: if you specify settings here, the PulseAudio system must be capable of giving effect to them. Otherwise, Shairport Sync will quit and leave a message in the system log.
//	output_rate = "auto"; // Specify "auto", or a single rate, e.g. 48000, or a bracketed comma-separated list of rates, e.g. (44100, 48000, 64000). Default is "auto" -- try to match the input. See the "Rates, Formats and Channels" discussion above.
//	output_format = "auto"; // Specify "auto", or a single format, e.g. "S32_LE", or a bracketed comma-separated list of formats, e.g. ("S32_LE", "S16_LE"). Default is "auto". See the "Rates, Formats and Channels" discussion above.
//	output_channels = "auto"; // Specify "auto", or a specific number of channels, e.g. 2, or a bracketed comma-separated list of numbers of channels, e.g. (2, 6). Default is "auto" -- try to match the input. See the "Rates, Formats and Channels" discussion above.
//	default_channel_layouts = "alsa"; // Set to "alsa" (default) for the alsa-compatible channel layouts (see "PA_CHANNEL_MAP_ALSA"), or set to "pulseaudio" if you want PulseAudio's own channel layouts ("PA_CHANNEL_MAP_DEFAULT") to be used instead. 
};

// Parameters for the JACK Audio Connection Kit backend.
// For this section to be operative, Shairport Sync must be built with the following configuration flag:
// --with-jack
jack =
{
//	client_name = "shairport-sync"; // Set this to the name of the client that should appear in "Connections" when Shairport Sync is active.
//	autoconnect_pattern = ""; // Set this to a POSIX regular expression pattern that describes the ports you would like to connect to
//                                   automatically. Examples:
//                                   "system:playback_[12]"
//                                   "some_app_[0-9]*:in-[LR]"
//                                   "jack_mixer:in_2[78]"
//                                   Beware: if you make a syntax error, libjack might crash. In that case, fix it and start over.
//                                   For a good overview, look here: https://www.ibm.com/support/knowledgecenter/SS8NLW_11.0.1/com.ibm.swg.im.infosphere.dataexpl.engine.doc/c_posix-regex-examples.html
//  soxr_resample_quality = "none"; // Enable resampling by setting this to "very high", "high", "medium", "low" or "quick"
//	bufsz = <number>; // advanced optional setting to set the buffer size to this value
};

// Parameters for the "pipe" audio back end, a back end that directs raw PCM audio output to a unix pipe. No interpolation is done.
// For this section to be operative, Shairport Sync must have been built with the following configuration flag:
// --with-pipe
pipe =
{
//	name = "/tmp/shairport-sync-audio"; // this is the default

//	Note: if you specify "auto" or multiple settings here. Shairport Sync may switch between them to match the input, but there will be no notification in the pipe as changes occur. To avoid this, consider setting one just rate/format/channel count. If built with AirPlay 2 or FFmpeg support, Shairport Sync will automatically transcode and mixdown as necessary.
//	output_rate = <rate>; // Specify a single rate, e.g. 44100, or a bracketed comma-separated list of rates, e.g. (44100, 48000, 64000) or "auto" -- try to match the input. Default is 44100 for classic AirPlay, 48000 for AirPlay 2. See the "Rates, Formats and Channels" discussion above.
//	output_format = <format>; // Specify a format, e.g. "S16_LE", or a bracketed comma-separated list of formats, e.g. ("S32_LE", "S16_LE") or "auto". Default is "S16_LE" for classic AirPlay, "S32_LE" for AirPlay 2. See the "Rates, Formats and Channels" discussion above.
//	output_channels = 2; // Specify a specific number of channels, e.g. 2, or a bracketed comma-separated list of numbers of channels, e.g. (2, 6) or "auto" -- try to match the input. Default is 2. See the "Rates, Formats and Channels" discussion above.
};

// Parameters for the "stdout" audio back end, a back end that directs raw PCM audio output to STDOUT. No interpolation is done.
// To include support for the "stdout" backend, Shairport Sync must be built with the following configuration flag:
// --with-stdout
stdout =
{
//	Note: if you specify "auto" or multiple settings here. Shairport Sync may switch between them to match the input, but there will be no notification in STDOUT as changes occur. To avoid this, consider setting one just rate/format/channel count. If built with AirPlay 2 or FFmpeg support, Shairport Sync will automatically transcode and mixdown as necessary.
//	output_rate = <rate>; // Specify a single rate, e.g. 44100, or a bracketed comma-separated list of rates, e.g. (44100, 48000, 64000) or "auto" -- try to match the input. Default is 44100 for classic AirPlay, 48000 for AirPlay 2. See the "Rates, Formats and Channels" discussion above.
//	output_format = <format>; // Specify a format, e.g. "S16_LE", or a bracketed comma-separated list of formats, e.g. ("S32_LE", "S16_LE") or "auto". Default is "S16_LE" for classic AirPlay, "S32_LE" for AirPlay 2. See the "Rates, Formats and Channels" discussion above.
//	output_channels = 2; // Specify a specific number of channels, e.g. 2, or a bracketed comma-separated list of numbers of channels, e.g. (2, 6) or "auto" -- try to match the input. Default is 2. See the "Rates, Formats and Channels" discussion above.
};

// Parameters for the "ao" (also known as "libao")  backend.
// To include support for the "ao" backend, Shairport Sync must be built with the following configuration flag:
// --with-ao
ao =
{
//	Note: if you specify settings here, the ao (aka "libao") system must be capable of giving effect to them. Otherwise, Shairport Sync will quit and leave a message in the system log.
//	output_rate = "auto"; // Specify "auto", or a single rate, e.g. 48000, or a bracketed comma-separated list of rates, e.g. (44100, 48000, 64000). Default is "auto" -- try to match the input. See the "Rates, Formats and Channels" discussion above.
//	output_format = "auto"; // Specify "auto", or a single format, e.g. "S32_LE", or a bracketed comma-separated list of formats, e.g. ("S32_LE", "S16_LE"). Default is "auto". See the "Rates, Formats and Channels" discussion above.
//	output_channels = "auto"; // Specify "auto", or a specific number of channels, e.g. 2, or a bracketed comma-separated list of numbers of channels, e.g. (2, 6). Default is "auto" -- try to match the input. See the "Rates, Formats and Channels" discussion above.
};


// For this section to be operative, Shairport Sync must be built with the following configuration flag:
// --with-convolution
dsp =
{
//////////////////////////////////////////
// This convolution filter can be used with inpulse responses files
// 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.
// Long impulse response files require lots of floating-point processing!
//////////////////////////////////////////
//
//      convolution_enabled = "no";                // Set this to "yes" to activate the convolution filter.
//      convolution_thread_pool_size = 1;          // Number of CPU threads that can work on convolution at the same time.
//      convolution_ir_files = "<comma-separated list of full path names to impulse response files, for sample rates of 44100 and 48000 -- stereo or mono only>";
//      convolution_gain = -4.0;                   // Static gain applied after convolution, between -60 and +18 dB. Useful for preventing clipping or amplifying low convolution output levels
//      convolution_max_length_in_seconds = 1.0;   // 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 lose more sensitivity to low frequencies than to mid range ones.
// This filter aims at compensating for this loss, applying a small extra gain to low frequencies, depending on the volume.
// More info can be found here: https://en.wikipedia.org/wiki/Equal-loudness_contour
//
// NOTE: To use this filter, Shairport Sync must _not_ be controlling a hardware mixer.
//
// The setting "loudness_reference_volume_db" is the highest level at which the loudness filter will take effect. Above this level the filter is off.
//////////////////////////////////////////
//
//	loudness_enabled = "no";                  // Set this to "yes" to activate the loudness filter (only works if you are _not_ using hardware volume control)
//	loudness_reference_volume_db = -16.0;     // Above this level the filter will have no effect. Below this level it will gradually boost the low frequencies.
};

// How to deal with metadata, including artwork
// For this section to be operative, Shairport Sync must be built with at one (or more) of the following configuration flags:
// --with-metadata, --with-dbus-interface, --with-mpris-interface or --with-mqtt-client.
// In those cases, "enabled" and "include_cover_art" will both be "yes" by default
metadata =
{
//	enabled = "yes"; // 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 = "yes"; // 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".
//	cover_art_cache_directory = "/tmp/shairport-sync/.cache/coverart"; // artwork will be  stored in this directory if the dbus or MPRIS interfaces are enabled or if the MQTT client is in use. Set it to "" to prevent caching, which may be useful on some systems
//	pipe_name = "/tmp/shairport-sync-metadata";
//	pipe_timeout = 5000; // wait for this number of milliseconds for a blocked pipe to unblock before giving up
//	progress_interval = 0.0; // if non-zero, progress 'phbt' messages will be sent at the interval specified in seconds. A 'phb0' message will also be sent when the first audio frame of a play session is about to be played.
//		Each message consists of the RTPtime of a a frame of audio and the exact system time when it is to be played. The system time, in nanoseconds, is based the CLOCK_MONOTONIC_RAW of the machine -- if available -- or CLOCK_MONOTONIC otherwise.
//		Messages are sent when the frame is placed in the output device's buffer, thus, they will be _approximately_ 'audio_backend_buffer_desired_length_in_seconds' ahead of time.
//	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.
};

// How to enable the MQTT-metadata/remote-service
// For this section to be operative, Shairport Sync must be built with the following configuration flag:
// --with-mqtt-client

// Note that, for compatability with many MQTT brokers and applications,
// every message that has no extra data is given a
// payload consisting of the string "--".
// You can change this or you can enable empty payloads -- see below.

mqtt =
{
//	enabled = "no"; // set this to yes to enable the mqtt-metadata-service
//	hostname = "iot.eclipse.org"; // Hostname of the MQTT Broker
//	port = 1883; // Port on the MQTT Broker to connect to
//	username = NULL; //set this to a string to your username in order to enable username authentication
//	password = NULL; //set this to a string you your password in order to enable username & password authentication
//	capath = NULL; //set this to the folder with the CA-Certificates to be accepted for the server certificate. If not set, TLS is not used
//	cafile = NULL; //this may be used as an (exclusive) alternative to capath with a single file for all ca-certificates
//	certfile = NULL; //set this to a string to a user certificate to enable MQTT Client certificates. keyfile must also be set!
//	keyfile = NULL; //private key for MQTT Client authentication
//	topic = NULL; //MQTT topic where this instance of shairport-sync should publish. If not set, the general.name value is used.
//	publish_raw = "no"; //whether to publish all available metadata under the codes given in the 'metadata' docs.
//	publish_parsed = "no"; //whether to publish a small (but useful) subset of metadata under human-understandable topics
//	empty_payload_substitute = "--"; // MQTT messages with empty payloads often are invisible or have special significance to MQTT brokers and readers.
//    To avoid empty payload problems, this string is used instead of any empty payload. Set it to the empty string -- "" -- to leave the payload empty.
//	Currently published topics:artist,album,title,genre,format,songalbum,volume,client_ip,
//	Additionally, messages at the topics play_start,play_end,play_flush,play_resume are published
//	publish_cover = "no"; //whether to publish the cover over mqtt in binary form. This may lead to a bit of load on the broker
//	publish_retain = "no"; //whether to set the retain flag on published MQTT messages. When enabled, the broker stores the last message for each topic.
//	enable_autodiscovery = "no"; //whether to publish an autodiscovery message to automatically appear in Home Assistant
//	autodiscovery_prefix = "homeassistant"; //string to prepend to autodiscovery topic
//	enable_remote = "no"; //whether to remote control via MQTT. RC is available under `topic`/remote.
//	Available commands are "command", "beginff", "beginrew", "mutetoggle", "nextitem", "previtem", "pause", "playpause", "play", "stop", "playresume", "shuffle_songs", "volumedown", "volumeup"
};

// Diagnostic settings. These are for diagnostic and debugging only. Normally you should 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".
//	log_output_to = "syslog"; // set this to "syslog" (default), "stderr" or "stdout" or a file or pipe path to specify were all logs, statistics and diagnostic messages are written to. If there's anything wrong with the file spec, output will be to "stderr".
//	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_file_and_line = "yes"; // set this to yes if you want the file and line number of the message source in the log file
//	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 = "yes"; // 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.
//	retain_cover_art = "no"; // artwork is deleted when its corresponding track has been played. Set this to "yes" to retain all artwork permanently. Warning -- your directory might fill up.
};