update README.md

Introduce opus codec and adopt new default values for chunk duration
and sample rate.
This commit is contained in:
Hannes Ellinger 2015-06-14 12:08:52 +02:00
parent 4fda33a20e
commit aa7cb58146

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@ -8,10 +8,11 @@ The server's audio input is a named pipe `/tmp/snapfifo`. All data that is fed i
How does is work How does is work
---------------- ----------------
The snapserver reads PCM chunks of 50ms duration from the pipe `/tmp/snapfifo`. The chunk is encoded and tagged with the local time The snapserver reads PCM chunks of 10ms duration from the pipe `/tmp/snapfifo`. The chunk is encoded and tagged with the local time
* PCM: lossless uncompressed * PCM: lossless uncompressed
* FLAC: lossless compressed [default] * FLAC: lossless compressed [default]
* Vorbis: lossy compression * Vorbis: lossy compression
* Opus: lossy low-latency compression
The encoded chunk is sent via a TCP connection to the snapclients. The encoded chunk is sent via a TCP connection to the snapclients.
Each client does continuos time synchronization with the server, so that the client is always aware of the local server time. Each client does continuos time synchronization with the server, so that the client is always aware of the local server time.
@ -65,20 +66,20 @@ Disable alsa audio output by commenting out this section:
# type "alsa" # type "alsa"
# name "My ALSA Device" # name "My ALSA Device"
# device "hw:0,0" # optional # device "hw:0,0" # optional
# format "44100:16:2" # optional # format "48000:16:2" # optional
# mixer_device "default" # optional # mixer_device "default" # optional
# mixer_control "PCM" # optional # mixer_control "PCM" # optional
# mixer_index "0" # optional # mixer_index "0" # optional
#} #}
Add a new audio output of the type "fifo", which will let mpd play audio into the named pipe `/tmp/snapfifo`. Add a new audio output of the type "fifo", which will let mpd play audio into the named pipe `/tmp/snapfifo`.
Make sure that the "format" setting is the same as the format setting of the snapserver (default is "44100:16:2", which should make resampling unnecessary in most cases) Make sure that the "format" setting is the same as the format setting of the snapserver (default is "48000:16:2", which should make resampling unnecessary in most cases)
audio_output { audio_output {
type "fifo" type "fifo"
name "my pipe" name "my pipe"
path "/tmp/snapfifo" path "/tmp/snapfifo"
format "44100:16:2" format "48000:16:2"
mixer_type "software" mixer_type "software"
} }