diff options
Diffstat (limited to 'scripts/shairport-sync.conf')
| -rw-r--r-- | scripts/shairport-sync.conf | 26 |
1 files changed, 14 insertions, 12 deletions
diff --git a/scripts/shairport-sync.conf b/scripts/shairport-sync.conf index 0502720..50d0f79 100644 --- a/scripts/shairport-sync.conf +++ b/scripts/shairport-sync.conf @@ -8,8 +8,8 @@ general = // The default is "<Hostname>" -- the first letter is capitalised (ASCII only.) // You can use %h for the hostname, // %H for the Hostname (i.e. with first letter capitalised (ASCII only)), -// %v for the version number, e.g. 2.8.4 and -// %V for the full version string, e.g. 2.8.4-OpenSSL-Avahi-ALSA-soxr-metadata +// %v for the version number, e.g. 3.2.1 and +// %V for the full version string, e.g. 3.2.1-OpenSSL-Avahi-ALSA-soxr-metadata // 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. @@ -19,8 +19,8 @@ general = // 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 (only three are needed). // statistics = "no"; // set to "yes" to print statistics in the log -// drift = 88; // allow this number of frames of drift away from exact synchronisation before attempting to correct it -// resync_threshold = 2205; // a synchronisation error greater than this will cause resynchronisation; 0 disables it +// drift = 0.002; // allow a timing error of this number of seconds of drift away from exact synchronisation before attempting to correct it +// resync_threshold = 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 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. @@ -60,8 +60,10 @@ alsa = // 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. -// audio_backend_latency_offset = 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 -4410. -// audio_backend_buffer_desired_length = 6615; // If set too small, buffer underflow occurs on low-powered machines. Too long and the response times with software mixer become annoying. +// output_rate = 44100; // can be 44100, 88200, 176400, 352800, but the device must be capable of it +// output_format = "S16_LE"; // can be "S16_LE", "S24_LE" or "S32_LE", but the device must be capable of it +// audio_backend_latency_offset = 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 = 0.15; // Given in seconds. If set too small, buffer underflow occurs on low-powered machines. Too long and the response times with software mixer become annoying. // 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 @@ -72,22 +74,22 @@ alsa = pipe = { // name = "/path/to/pipe"; // there is no default pipe name for the output -// audio_backend_latency_offset = 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 -4410. -// audio_backend_buffer_desired_length = 44100; // Having started to send audio at the right time, send all subsequent audio this many frames ahead of time, creating a buffer this size. +// audio_backend_latency_offset = 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 = 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. }; // 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 = 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 -4410. -// audio_backend_buffer_desired_length = 44100; // Having started to send audio at the right time, send all subsequent audio this many frames ahead of time, creating a buffer this size. +// audio_backend_latency_offset = 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 = 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. }; // These are parameters for the "ao" audio back end. No interpolation is done. ao = { -// audio_backend_latency_offset = 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 -4410. -// audio_backend_buffer_desired_length = 44100; // Having started to send audio at the right time, send all subsequent audio this many frames ahead of time, creating a buffer this size. +// audio_backend_latency_offset = 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 = 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. }; // Static latency settings are deprecated and the settings have been removed. |