zzzap wrote:
> Try these
> SL_SOUNDCARD=default
> card "E30"
EUREKA
I finally got it working with
Code:
ffmpeg -f alsa -acodec pcm_s32le -channels 2 -sample_rate 44100 -i hw:3,1,0
fromFLAC.wav
Here is the result from Deltawave :cool:
+---
I am making slow progress. If I just try to capture audio from
Squeezelite without listening to it I can select my loopback device as
the Squeezelite output device (hw:3,0,0) and I can run
arecord -vv -D hw:3,1,0 -f S32_LE -r 44100 -c 2 /dev/null
where hw:3,1,0 is the other half of the loopback d
I can run
squeezelite -o hw:CARD=E30,DEV=0
from a terminal but default still doesn't work.
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I installed Squeezelite using
Sudo apt install Squeezelite
but I have no idea how to run it...
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zzzap wrote:
> Try these
> SL_SOUNDCARD=default
> card "E30"I still don't hear any output so presumably no recording either. It
might be easier to start again installing Squeezelite your way. There
isn't even an easy way to remove my Squeezelite.
Sent from my Pixel 3a using Tapatalk
---
slartibartfast wrote:
> In my Squeezelite settings file the line would be
> SL_SOUNDCARD="default"
> with quotation marks. Is yours the same? Where is the default soundcard
> defined?
> If my soundcard is
> hw:CARD=E30,DEV=0
> is that what I would enter for pcm.output?
Try these
SL_SOUNDCARD=d
zzzap wrote:
> @slartibartfast
>
> As before mentiod I installed Squeezelite from the package manager with
> output to 'default'.
> ~$ sudo apt install squeezelte
>
> Output in the asound.conf use name of you sound card. Or simply try
> 'card 1'.
> >
Code:
> > pcm.out
@slartibartfast
As before mentiod I installed Squeezelite from the package manager with
output to 'default'.
~$ sudo apt install squeezelte
Output in the asound.conf use name of you sound card. Or simply try
'card 1'.
Code:
pcm.output {
type hw
card "sndrpihifiberr
Paul Webster wrote:
> This might still work
> https://raspberrypi.stackexchange.com/questions/42773/modules-not-loaded-at-boot-timeThanks
> I'll try that if I can get the loopback recording working. Any
ideas about that [emoji1696]
Sent from my Pixel 3a using Tapatalk
-
slartibartfast wrote:
> It is annoying having to run
> modprobe snd-aloop pcm_substreams=1
> after every reboot.
>
This might still work
https://raspberrypi.stackexchange.com/questions/42773/modules-not-loaded-at-boot-time
Paul Webster
author of \"now playing\" plugins covering radio france
It is annoying having to run
modprobe snd-aloop pcm_substreams=1
after every reboot.
My recorded file is silent, presumably some issue with Squeezelite
output selection or the card name in .asoundrc is wrong. I installed
Squeezelite using instructions here
http://www.gerrelt.nl/RaspberryPi/wordpr
zzzap wrote:
> I'm on latest RPi-OS Lite. Pulseaudio comes with the grapical versjon of
> RPi-OS to help mix sound from multiple applications. I've uninstalled it
> on some ocations, but it breaks audio from other apps.
> Maybe start with a clean Lite install on another SD card is the less
> tr
zzzap wrote:
> I'm on latest RPi-OS Lite. Pulseaudio comes with the grapical versjon of
> RPi-OS to help mix sound from multiple applications. I've uninstalled it
> on some ocations, but it breaks audio from other apps.
> Maybe start with a clean Lite install on another SD card is the less
> tr
zzzap wrote:
> I'm on latest RPi-OS Lite. Pulseaudio comes with the grapical versjon of
> RPi-OS to help mix sound from multiple applications. I've uninstalled it
> on some ocations, but it breaks audio from other apps.
> Maybe start with a clean Lite install on another SD card is the less
> tr
slartibartfast wrote:
> I don't have .asoundrc either so I think the likelihood of me getting
> this working is slim. Which version of the Raspberry OS are you running
> on your Pi. Mine is Buster and pulseaudio is default on that. I have no
> idea what difference that makes.
I'm on latest RPi-
zzzap wrote:
> Yes, we can also have it in user home directory under another name. See
> the description on the that page.I don't have .asoundrc either so I think the
> likelihood of me getting
this working is slim. Which version of the Raspberry OS are you running
on your Pi. Mine is Buster an
slartibartfast wrote:
> I am just starting to have a look at this and I don't seem to have
> /etc/asound.conf
> Did you create it?
>
Yes, we can also have it in user home directory under another name. See
the description on the that page.
-
zzzap wrote:
> Quite possible, if a problem at all. I where lead in a direction due to
> how the WAV file repeatedly got 100% match while the FLAC always failed
> when recaptured multiple times. With now a more a extended dataset using
> different input files I'm starting to a lesser degree trus
zzzap wrote:
> Thanks @bpa. More to digest. The tee command can prove useful to compare
> Ogg/FLAC stream to ffmpeg capture at alsa output. Ahh of course,
> directing the convertion to an application of own making opens up many
> possibilities. Thanks, inspires to think of new ways of using LMS.
bpa wrote:
> This feels like an ALSA issue not an LMS one.
Quite possible, if a problem at all. I where lead in a direction due to
how the WAV file repeatedly got 100% match while the FLAC always failed
when recaptured multiple times. With now a more a extended dataset using
different input fil
zzzap wrote:
> With data corruption I'm thinking at byte level not interupting the
> stream. As in why does DeltaWave find 1 dB difference in RMS level and
> still return 40% bit accuratsy? I would expect it to be null when
> amplitude differs. And why would higher frequencies be less accurate?
slartibartfast wrote:
> Are the waveforms you feed into Deltawave exactly the same number of
> samples?
No, that's not necessary. Don't even have to be the same bit depth,
format or amplitude. DW will convert, seek and find start position on
its own. And then try to even out levels and clock be
zzzap wrote:
> Thanks @bpa. More to digest. The tee command can prove useful to compare
> Ogg/FLAC stream to ffmpeg capture at alsa output. Ahh of course,
> directing the convertion to an application of own making opens up many
> possibilities. Thanks, inspires to think of new ways of using LMS.
Thanks @bpa. More to digest. The tee command can prove useful to compare
Ogg/FLAC stream to ffmpeg capture at alsa output. Ahh of course,
directing the convertion to an application of own making opens up many
possibilities. Thanks, inspires to think of new ways of using LMS.
With data corruption
zzzap wrote:
> Not at all. As mentioned volume fixed at 100% with replay gain off. But
> here is the thing, if amplitude is changes between two takes then all
> audio samples would change for that file unless we were juggling the
> volume knob.
> Here I get more than 40% of the data correct. And
Ron F. wrote:
> My comment here could be construed as naive, but assuming the question
> is bit-perfect output from a Squeezelite instance running on RPi-OS,
> (output samples to stdout using -o -) then what happens if in LMS, the
> player's audio volume control setting option is fixed to 100% ?
My comment here could be construed as naive, but assuming the question
is bit-perfect output from a Squeezelite instance running on RPi-OS,
(output samples to stdout using -o -) then what happens if in LMS, the
player's audio volume control setting option is fixed to 100% ??? I
imagine the OP is
zzzap wrote:
> What OS are your Squeezelite running on?
>
A pretty basic RPi setup: Pi Zero W, PiOS, JustBoom Digi Zero hat, coax
out to RME DAC -> RME USB -> Linux box -> Sox capture.
Here's a capture of a 1kHz 24-bit signal through the above, from my
piCorePlayer Pi-4 LMS server. A couple
zzzap wrote:
>
> Anyone familiar with the LMS code stack that can point me to the files
> where 'stdout' are picket up and sendt to the output?
Code:
→ Slim/Player/Song.pm (open) -→
Slim/Player/SongStreamController.pm →
↓
The top waveform is the output of Squeezeplay on Windows 10 recorded
using WASAPI loopback in Audacity
The bottom waveform is the original FLAC
They look noticeably different. Why would there be such a difference?
+---+
|Filename:
chicks wrote:
> I converted a couple of RME's bit-perfect test files from WAV to FLAC
> using Sox. They still show bit-perfect on the RME ADI-2 DAC FS screen
> when played through Squeezelite from my LMS server.
>
> https://www.manualslib.com/manual/1374592/Rme-Audio-Adi-2-Dac.html?page=63
Wha
The fact that I can send MQA files to my DAC through squeezelite and the
DAC says they are bitperfect strongly casts doubt on this assertion.
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I converted a couple of RME's bit-perfect test files from WAV to FLAC
using Sox. They still show bit-perfect on the RME ADI-2 DAC FS screen
when played through Squeezelite from my LMS server.
https://www.manualslib.com/manual/1374592/Rme-Audio-Adi-2-Dac.html?page=63
piTouch w/JustBoom DigiHa
Final test to exclude the flac application from the equation.
Code:
flac -dcs --force-raw-format --endian=little --sign=signed srv.flac >
srvPIPE.pcm
ffmpeg -f s16le -ar 44.1k -ac 2 -i srvPIPE.pcm srvPIPE.wav
Here we use the same command string
slartibartfast wrote:
> I tried recording a track from Squeezelite-X using WASAPI loopback in
> Audacity. The recorded track is around 2dB higher peak amplitude than
> the original FLAC file but I have no idea why.
I've seen similar from HiFi Berry S/PDIF where there are some -3 dB.
That could
zzzap wrote:
> Thats what DeltaWave are for I think ;)
>
> Fact is I can repeatedly capture a bit perfect stream from WAV files.
> LMS use 'flac -d' output to stdout. This in theory should give us
> exactly the same result as if I use 'flac -d' manually on the FLAC file
> and then stream the W
slartibartfast wrote:
> Have you examined the waveforms in Audacity to see if you can see any
> difference when zoomed in to individual samples?
Thats what DeltaWave are for I think ;)
Fact is I can repeatedly capture a bit perfect stream from WAV files.
LMS use 'flac -d' output to stdout. Thi
zzzap wrote:
> When capturing S/PDIF or analog out whatever medium used will be hard to
> compare. I've tried. Only usefull data I found is the clock and jitter
> difference. And even that might not be of interest using a modern DAC.
>
> I duplicate the Alsa stream comming from Squeezelite. And
slartibartfast wrote:
> How are you capturing the output files?
When capturing S/PDIF or analog out whatever medium used will be hard to
compare. I've tried. Only usefull data I found is the clock and jitter
difference. And even that might not be of interest using a modern DAC.
I duplicate the
How are you capturing the output files? I have messed around with
Deltawave to compare pCP to mpd and results are influenced by the
position of samples on the waveform. The developer of the software has
said that comparing output waveforms to original files doesn't work too
well.
Sent from my Pi
Messing around with Deltawave https://deltaw.org/ I found that capturing
Alsa output from Squeezelite on RPi-OS I have absolutely bit perfect
output from Squeezelite if streaming WAV files from local network.
Sadly results from streaming FLAC where not as joyful and comply with
listening observat
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