I'm having a strange problem playing back FLAC files gapless(ly) and I
need some help...

I don't know if this is related to upgrading squeezelite to 1.8-dev,
upgrading lubuntu or replacing my AV receiver (XMC-1).

My setup is:
    
- ODROID/max2play connected directly to an Emotiva/XMC-1 DAC through
  USB or through HDMI
- squeezlite 1.8-dev (Build options: LINUX ALSA EVENTFD RESAMPLE
  FFMPEG VISEXPORT)
- squeezelite parameters (USB-DAC): -o plughw:CARD=XMC1,DEV=0 -a
  320::: -p 90
- squeezelite parameters (HDMI): -o plughw:CARD=Audio1,DEV=0 -a
  120::16:
  

If I play some flac files back-to-back (same sampling rates), squeezlite
shows this debug log:


Code:
--------------------
    
  [12:48:14.053728] output_thread:630 open output device: plughw:CARD=XMC1,DEV=0
  [12:48:14.053779] alsa_open:347 opening device at: 44100
  [12:48:14.054538] alsa_open:398 opened device plughw:CARD=XMC1,DEV=0 using 
format: S32_LE sample rate: 44100 mmap: 1
  [12:48:14.054863] alsa_open:477 buffer: 320 period: 4 -> buffer size: 14112 
period size: 3528
  
  [...]
  [12:55:40.889564] set_volume:238 setting internal gain left: 65536 right: 
65536
  [12:55:40.891499] set_volume:238 setting internal gain left: 65536 right: 
65536
  [12:55:40.984041] _output_frames:59 start buffer frames: 198144
  [12:55:40.984127] _output_frames:144 track start sample rate: 44100 
replay_gain: 0
  [12:55:43.736240] set_volume:238 setting internal gain left: 65536 right: 
65536
  [12:55:43.784020] _output_frames:59 start buffer frames: 422336
  [12:55:53.224294] _output_frames:144 track start sample rate: 44100 
replay_gain: 0
  [12:55:58.446883] set_volume:238 setting internal gain left: 65536 right: 
65536
  [...]
  
--------------------


So the songs should actually play without gaps. But actually they do not
in my setup. I get some short garbled sound at the beginning of the new
song, as if the DAC had to resync or it would receive some bad data. The
first few ms of every song do not get played. It's exactly the same
using audio over HDMI.

It is not the same garbled sound every time if I skipped back to the
beginning of the song (resulting in "output_flush:418 flush output
buffer"). If there is short silence at the beginning of the new song, no
garbled sound can be heard anyway. This helps while listing...

My personal analysis: If playback would be gapless, then the DAC/AVR
would not recognize the start of a new song. It would just receive an
'endless' stream of data. Right? So this seems to be a buffering/driver
problem or an error at the player side?

I tried adjusting buffer sizes, but I think 320ms should be more than
enough.  

Any ideas of possible ajustments or changes in my setup?



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