Hypfer wrote:
>
> And indeed it was working surprisingly well with only latency being an
> issue.
> Using the MP3 codec there causes some buffer in the audio path to add
> >10s of audio delay into the setup which is of course unacceptable.
>
> Interestingly, choosing pcm as the output format instead will result in
> the desired latency of 2-3s with no additional changes.
> That suggests that there's some fixed-size buffer somewhere which will
> just buffer as much as it can which will be a lot when using compressed
> audio.
Yes in application called flac which compresses the pcm from input port
into flac but not fixed size buffer.
Most of the delay will be in the application (i.e. lame for MP3 or Flac
for flac) which compress the data. Very little delay in LMS. Rest of
delay will be in player, transmission time and network buffers and don't
forget the roc stuff (its own buffering and network delay).
> There must be something I can tune to get this down to an acceptible
> level.
I think "must" is too optimistic if acceptable is less than maybe
200-500ms. If using windows - don't even bother trying.
The only setting left buffer threshold - min value is about 50 but then
if data is a little slow (i.e. don't use wifi) you'll get rebuffering
If filling buffer is the delay - then try a higher sampling rate (i.e.
96khz andnot 44.1khz) then buffers will be filled faster but take longer
to transmit - swings and roundabouts.
Player should be wired and not use 2.4G Wifi.
> Secondly, I hope that someone has an idea how to adjust the latency when
> using mp3 as the codec.
Buffer is the problems - buffer to compress and then player has to
buffer data before decompress.
> There is a webradio buffer setting in the audio settings of each
> squeezebox, however it doesn't seem related to this issue
Add the following in WAVIN.pm - it can cause more problems than it
solves but for a fixed setup. Value can be 0-255 but less than 50
generally doesn't work.
Code:
--------------------
# use a small buffer threshold to make stream start playing quickly
sub bufferThreshold { 70 }
--------------------
Radio station bufer seconds is the setting which changes how much data
must be in a player before ti start playing. The above means it is a
plugin specific value and not general settings.
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