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Move common audio backend settings like buffer size and offsets to the "general" settings stanza. Improve the resync code a little.

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This commit is contained in:
Mike Brady
2017-07-21 12:29:03 +02:00
parent d810d43048
commit b7864b4ecf
11 changed files with 153 additions and 365 deletions
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scripts/shairport-sync.conf
+11 -26
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@@ -23,6 +23,7 @@ general =
// 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
// log_verbosity = 0; // "0" means no debug verbosity, "3" is most verbose.
// 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.
@@ -31,11 +32,16 @@ general =
// 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; // Use this optional advanced setting, between 0.050 and 4.0 seconds, to set 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.
};
@@ -100,9 +106,6 @@ alsa =
// 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.
// 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 times with software mixer become annoying.
// audio_backend_silent_lead_in_time = 2.0; // Use this optional advanced setting, between 0.050 and 4.0 seconds, to set 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.
// 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
@@ -110,7 +113,7 @@ alsa =
// mute_using_playback_switch = "yes"; // Use this optional advanced setting to control whether the snd_mixer_selem_set_playback_switch_all call can be used for muting. Default is yes.
};
// These are parameters for the "sndio" audio back end. All are optional.
// Parameters for the "sndio" audio back end. All are optional.
sndio =
{
// device = "snd/0"; // the name of the output device
@@ -120,39 +123,21 @@ sndio =
// bufsz = <number>; // set the buffer size near to this value
};
// These are parameters for the "pa" PulseAudio backend.
// 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.
// audio_backend_latency_offset_in_seconds = 0.0; // Set this offset in seconds 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 = 0.35; // Having started to send audio at the right time, send all subsequent audio this much ahead of time, creating a buffer this length.
// audio_backend_silent_lead_in_time = 2.0; // Use this optional advanced setting, between 0.050 and 4.0 seconds, to set 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.
};
// These are parameters for the "pipe" audio back end, a back end that directs raw CD-style audio output to a pipe. No interpolation is done.
// 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
// audio_backend_latency_offset_in_seconds = 0.0; // Set this offset in seconds 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 = 1.0; // Having started to send audio at the right time, send all subsequent audio this much ahead of time, creating a buffer this length.
// audio_backend_silent_lead_in_time = 2.0; // Use this optional advanced setting, between 0.050 and 4.0 seconds, to set 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.
};
// These are parameters for the "stdout" audio back end, a back end that directs raw CD-style audio output to stdout. No interpolation is done.
stdout =
{
// audio_backend_latency_offset_in_seconds = 0.0; // Set this offset in seconds 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 = 1.0; // Having started to send audio at the right time, send all subsequent audio this much ahead of time, creating a buffer this length.
// audio_backend_silent_lead_in_time = 2.0; // Use this optional advanced setting, between 0.050 and 4.0 seconds, to set 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.
};
// These are no configuration file parameters for the "stdout" audio back end. No interpolation is done.
// These are parameters for the "ao" audio back end. No interpolation is done.
ao =
{
// audio_backend_latency_offset_in_seconds = 0.0; // Set this offset in seconds 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 = 1.0; // Having started to send audio at the right time, send all subsequent audio this much ahead of time, creating a buffer this length.
// audio_backend_silent_lead_in_time = 2.0; // Use this optional advanced setting, between 0.050 and 4.0 seconds, to set 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.
};
// 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.