Type sudo nano /etc/pulse/daemon.conf
then using the arrow keys locate the line in question. Make the changes as indicated. When you are done, hit Ctrl + O, confirm that you want to over write the file (thus saving it). Then hit Ctrl + X.
You are now free to restart pulse audio, but you can also just restart the machine.
So the last tip has helped me edit the file but its still sending it to the hifi at 44.1k instead of 96k.
here is what my file reads like:
; avoid-resampling = true
; enable-remixing = yes
; remixing-use-all-sink-channels = yes
; enable-lfe-remixing = no
; lfe-crossover-freq = 0
; flat-volumes = yes
; rlimit-fsize = -1
; rlimit-data = -1
; rlimit-stack = -1
; rlimit-core = -1
; rlimit-as = -1
; rlimit-rss = -1
; rlimit-nproc = -1
; rlimit-nofile = 256
; rlimit-memlock = -1
; rlimit-locks = -1
; rlimit-sigpending = -1
; rlimit-msgqueue = -1
; rlimit-nice = 31
; rlimit-rtprio = 9
; rlimit-rttime = 200000
; default-sample-format = s16le
; default-sample-rate = 44100
; alternate-sample-rate = 96000
I also tried removing the semi colon at the start of the line but that didnt work either.
Is it because of the default sample format being limited to 16 bits and my flac file is 32 bits?
You must remove the semi-colon.
Please do that and restart the phone and then try again.
sudo nano /etc/pulse/daemon.conf is the file-name you need to edit and write-out to disk
the ; or ;; or # are commented out lines that need to be un-commented out to take effect.
simply erase the comments so the line looks like so :
avoid-resampling = true
make sure there are no spaces or anything else before or after the edited line.
i’ve learned something new in this thread too ! all input appreciated ! 
Thanks for that, however its still sending it at 44.1k. here is how it reads:
; resample-method = speex-float-1
avoid-resampling = true
; enable-remixing = yes
; remixing-use-all-sink-channels = yes
; enable-lfe-remixing = no
; lfe-crossover-freq = 0
; flat-volumes = yes
; rlimit-fsize = -1
; rlimit-data = -1
; rlimit-stack = -1
; rlimit-core = -1
; rlimit-as = -1
; rlimit-rss = -1
; rlimit-nproc = -1
; rlimit-nofile = 256
; rlimit-memlock = -1
; rlimit-locks = -1
; rlimit-sigpending = -1
; rlimit-msgqueue = -1
; rlimit-nice = 31
; rlimit-rtprio = 9
; rlimit-rttime = 200000
; default-sample-format = s16le
; default-sample-rate = 44100
; alternate-sample-rate = 96000
; default-sample-channels = 2
; default-channel-map = front-left,front-right
could it be the resample method?
It shouldn’t be as resampling is what is downgrading your audio files. With avoid-resampling set to true, the resample method shouldn’t matter. Although avoid doesn’t read as implicit, so there may be another condition still permitting the resampling. Not my forte unfortunately.
Its a desktop PC. Intel i7 ivybridge
OK, for some reason I thought it was the Librem 5 (phone), I guess because I have been playing around with sound files on my phone.
Do you have actual audio files saved locally on the computer?
Did you restart after that change?
how can you tell ? what program are you using for playback ?
in my case the L-Mini-v1 is sending the original-digital-audio-output to an external AMP/DAC which does some internal processing on it’s own hardware and spits it further out ANALOG into my ears …
Yeah its on one of the internal HDDs’
I restarted twice. The details within the file have changed.
My amp has a screen on it that gives me details of whats playing etc. In windows I can play the same file and the screen reads 96k (after some faffing about) but when i switch it to PureOS its back to 44.1k. Im sending the music to the amp via an optical toslink cable, if that matters.
i mean, 44.1 kHz is original CD quality (it’s actually double the frequency most people can distinguish sound - we aren’t all bats you know 
)
if you’re into BASS then you need to be concerned about the lower frequency spectrum. i still think this warrants further investigation.
see the older thread > Audiophile question
youre probably right, but Id like the dac in the amp to take charge of the decoding of hi res music over my general purpose pc. Im into the overall soundstage not BASS as my neighbours keep complaining.
have you tried playing hi-fi audio files with MPV (Celluloid) - for AV files if you press i while in playback-mode you will see some media info.
VLC offers you some info as well while playing music ctrl + i if i remember correctly (haven’t used VLC in a while)
We have lift off!
I tried MPV but dragging a file into did nothing. So I switched to VLC and its sending it to the amp at 96k therefore it must be something to do with Audacious or rhythm box as they are still downgrading it. So now we are getting somewhere how can i be sure its sending the file as 32 bits and the default decoder is s16xx which i assume is 16 bit.
Also FYI your can enter pulseaudio -k in the terminal to restart pulse without rebooting if you make any more changes.
You could try to force an error as shown below. Warning: This is off my Librem 5 i.e. completely different hardware, both audio and everything else.
aplay -D hw:0,0 foo.wav
Playing WAVE 'foo.wav' : Unsigned 8 bit, Rate 8000 Hz, Mono
Available formats:
- S16_LE
- S24_LE
- S32_LE
I’ve been lucky here in that the hardware doesn’t support a crappy WAV file recorded off my microphone on a different computer, thus provoking the software into reporting the formats that it does support.
Anyway the goal is to see whether S32_LE is listed.
I would suppose that the LE stands for “Little Endian” i.e. the 2, 3 or 4 byte samples are stored low-byte-first, but that may not matter for the purposes of the question asked.
This may be of help.
the program foobar came highly praised for it’s ‘don’t touch’ playback … i see it’s not on the PureOS software store by default.