bpa wrote:
> I didn't look in great detail but anything other than 1 seems very
> complicated.
>
> If HiRes MQA stream can play on Windows - I would look at network
> traffic to find ways to do the same on LMS and cut out all the VAC &
> remote control stuff - my gut feel a quick initial investi
I didn't look in great detail but anything other than 1 seems very
complicated.
If HiRes MQA stream can play on Windows - I would look at network
traffic to find ways to do the same on LMS and cut out all the VAC &
remote control stuff - my gut feel a quick initial investigation will
tell if it
Continued explorations of ways to get hi-res Tidal audio out of the
Squeezebox players, via Waveinput or otherwise, I have now moved to a
new more specific and updated thread and under what I believe is the
correct category:
https://forums.slimdevices.com/showthread.php?110959-Master-high-resolut
bpa wrote:
> Ask technical questions but include also describe the ultimate goal -
> there are many users who have found solutions to problem and are willing
> to share. Vague questions like "xxx doesn't work" can be offputting as
> it usually means someone at the bottom of the learning curve.
...testet with success on LMS server 7.9.+ on Windows 10.
Thanks to @pbjbryan and @bpa for the WaveInput pluging, which is still
working for ever new applications, and for support!
http://wiki.slimdevices.com/index.php/WaveInput_plugin
The problem was that the default wavin2cmd.exe transcoder
Vegz78 wrote:
> I would if I could, but unsuccessfully spending many hours studying the
> plugin scripts and custom-convert.conf rules trying to get $FILE$ or any
> other variables to output only the channel # and not the full "wavein:#"
> string to be used in a favorite entry for choosing the a
bpa wrote:
> Good that you found a solution.
>
Me too, thanks for all your help!!!
bpa wrote:
> You needn't have copied and renamed sox - you can use "[sox]" instead of
> "[wavin2cmd]" and LMS will search all its "Bin" path for sox.exe and
> should find it in the main LMS Bin path. That'll m
Good that you found a solution.
You needn't have copied and renamed sox - you can use "[sox]" instead of
"[wavin2cmd]" and LMS will search all its "Bin" path for sox.exe and
should find it in the main LMS Bin path. That'll make it easier for you
to maintain - just need keep a copy of the custo
bpa wrote:
> I opt for spelling out all possibilities.
>
> As I read your posts (albeit very quickly) you can play 44.1/16 but
> possibly you haven't managed to get wavin2cmd to record 96/24 - then
> * point 3 in terms of modes may not be an issue but perhaps the audio
> format part of propert
Vegz78 wrote:
> Since everything is working fine hw access and all in default 44,1/16,
> but not in 96/24, can it really be the things you're suggesting here?
I opt for spelling out all possibilities.
As I read your posts (albeit very quickly) you can play 44.1/16 but
possibly you haven't man
Thanks again for your quick reply, bpa!
I'll do all these tests and get back to you, as its good to test outside
LMS and directly on files.
But, referring to my updated post, 2 posts ago: vegz78 wrote:
>
Since everything is working fine hw access and all in default 44,1/16,
but not in 96/24,
The "few secs of audio" is to do with Wav format. With WAV the WAV
header has details including length. Usually when a wav file is being
written - the header is updated after the whole file is written. This
cannot be done if output is to a stream (i.e. STDOUT is no "seekable")
To check audio
bpa wrote:
> I haven't been near tWin2cmd for many years so I have to refresh my
> memory especially wrt the "few secs of audio" IIRC the wiki entrrey has
> an error
>
> What version of windows ?
I understand, and I'm sorry to bother you with old memories, but very
happy to get replies and s
Vegz78 wrote:
> Any thoughts or ideas?
I haven't been near tWin2cmd for many years so I have to refresh my
memory especially wrt the "few secs of audio" IIRC the wiki entrrey has
an error
What version of windows ?
bpa
bpa wrote:
> On Windows the wavin2cmd program defaults to outputs 44.1kHz/16 stream
> so you are trying to encode a 44.1kHz/16 PCM stream and telling Flac
> that is 96kHz/24 - so you will get noise on playback.
>
> You need to tell wavin2cmd to use the sample rates & size you want which
> must
Vegz78 wrote:
> Thanks again, bpa!
>
> Ill check it out again later tonight and return with the result!
Since you have cross posted variations of the same issue on multiple
threads - I suggest you post an update on those threads so that other
users do not waste their time on the same issue whi
bpa wrote:
> You need to tell wavin2cmd to use the sample rates & size you want which
> must be exactly matched by the flac encoding.
> >
Code:
> >
> [wavin2cmd] -s 96000 -b 24 -o $FILE$ | [flac] -cs --totally-silent
--compression-level-0 --endian=little --sign=sign
On Windows the wavin2cmd program defaults to outputs 44.1kHz/16 stream
so you are trying to encode a 44.1kHz/16 PCM stream and telling Flac
that is 96kHz/24 - so you will get noise on playback.
You need to tell wavin2cmd to use the sample rates & size you want which
must be exactly matched by th
bpa wrote:
> What have you tried ?
>
> Show the custom-convert.conf file that has failed ?
Hi bpa,
Thanks for your swift reply!
I'm on a newly installed nightly Logitech Media Server Versjon: 7.9.2 -
1565967976 @ Fri Aug 16 17:08:05 WEDT 2019 for Windows with the
"official" 3rd party WaveInp
Vegz78 wrote:
> Did anyone ever find a solution for this?
> Here or
> https://forums.slimdevices.com/showthread.php?102689-Looking-for-plugin-which-parses-stream-for-sample-rate-bits-per-sample&p=950172#post950172
> ?
>
> I'm looking for a similar setup where I stream Hi-res/Master Tidal from
>
Did anyone ever find a solution for this?
Here or
https://forums.slimdevices.com/showthread.php?102689-Looking-for-plugin-which-parses-stream-for-sample-rate-bits-per-sample&p=950172#post950172
?
I'm looking for a similar setup where I stream Hi-res/Master Tidal from
a Windows server via Waveinp
I've been browsing through plugins to try to find a parseMetadata
routine (or similar) that pushes the sampling rate, bits per sample, and
number of channels of the stream to the server (and then the stream). Do
you have an idea of a plugin which would contain this kind I code. I
should be able to
> All you need to do is look in other plugin for the parseMetadata
> routine, copy the code to the WAVIN.pm file, chop out the parsing code
> from the other plugin and hardcode the values you want in the returned
> data.
I attempted to look through and understand the parseMetadata routine in
the
mike_b16 wrote:
> I'd definitely like to share this with the community when it's
> completed.
If you plan to offer the completed plugin you'll need to learn LMS Perl
to rename and create your own plugin package and be able to provide
support.
---
Ok, I'll have a look and let you know how it goes. Thanks for being
patient.
Cheers,
Michael
mike_b16's Profile: http://forums.slimdevices.com/member.php?userid=63606
View this thread: http://forums.slimdevices.com/showth
mike_b16 wrote:
> I do believe you. I understand that your plugin is expecting the audio
> at 16/44,1. I don't believe that I have enough knowledge to create a
> parseMetadata routine which can just automatically detect the source
> information. That's why I'm asking if I can just hardcode the va
I do believe you. I understand that your plugin is expecting the audio
at 16/44,1. I don't believe that I have enough knowledge to create a
parseMetadata routine which can just automatically detect the source
information. That's why I'm asking if I can just hardcode the values in
and avoid having
mike_b16 wrote:
> I had a look at the WAVIN.pm file. I see that there is a subfunction
> "getMetadatafor" where the bitrate for CD quality (16 bit, 44.1 kHz, and
> 2 channels I'm assuming) is hardcoded. All I ever want to pass to LMS is
> audio at 24 bits, 192 kHz, and 2 channels. Is there a way
Ok, I've spent some time wrapping my head around this. Also, I tried to
test the plugin at 24/192 by changing the rate and format on all 3 lines
which contained arecord (in the custom-convert.conf file). Just for
curiosity more than anything. When I tried to use the plugin, I got
static out of my
mike_b16 wrote:
> Now, will I definitely need to modify your plugin to achieve this, or is
> it just a possibility? Does WaveInput have some kind of downsampling
> built in if the input is above that 16/44.1 mark?
I'm worried that you don''t understand how LMS works. LMS does all the
audio pro
So it turns out I was a little bit dense before. When I was reading up
on the plugin before I attempted to use it, I saw something that said it
"copied" the audio on the sound card in order to use it for the plugin.
I understood this as some kind of current playback being required to use
the plugi
>From the Wolfson audio driver source code - a shell script to recrod
from spdif input. It looks like it is OK to record in 32 bit format
even though hardware inputs up to 24 bit..
Code:
#!/bin/bash
#Record from SPDIF in
#SPDIF Record:
amixer -Dhw:0 cset
mike_b16 wrote:
> That is true, but I believe that the Wolfson card doesn't support
> multi-track recordings. AKA if I try to run multiple instances of
> arecord on the same card, I'll get errors. That's why I'm trying to fine
> a way to pass it off somewhere else.
What are you talking about ? I
That is true, but I believe that the Wolfson card doesn't support
multi-track recordings. AKA if I try to run multiple instances of
arecord on the same card, I'll get errors. That's why I'm trying to fine
a way to pass it off somewhere else.
Cheers,
Michael
If you can arecord from the Wolf audio line in - there is no need for
any Pulseaudio or snd-aloop complications.
If you want WaveInput to stream a format other than 44.1kHz/16bit - you
need to modify it to tell LMS that you are using a different format.
-
Thanks for looking into that for me. That did take the air out of my
sails a bit, but I'm not giving up yet. I'm going to attempt the
PulseAudio solution. Even if that did pan out though, you're thinking
that I would still need to modify your plugin, yes?
Cheers,
Michael
--
AFAICT looking at snd_loop.c source code lines 570-571 ( see
http://lxr.free-electrons.com/source/sound/drivers/aloop.c) the loopback
driver does not support 24bit format.
Code:
570 .formats = (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE |
571
I have a bit of programming experience. Hopefully this is something I
could pick up fairly easily.
Before I get to this however, I need to try to understand why the
snd-aloop module won't let me play to it/record from it in 24 bits. Even
if I managed to modify the plugin correctly, it would be a
mike_b16 wrote:
> Would you have an idea of where to start looking or a direction you
> could point me in? Getting 24/192 out of this stream was quite important
> here, as it was my goal from the beginning.
What are your software skills ?
I think you'll need to add a parseMetadata routine to th
bpa wrote:
> So I think the plugin needs modification for non 44.1khz / 16 bit
> streams and offhand I do not know what needs to be changed.
Would you have an idea of where to start looking or a direction you
could point me in? Getting 24/192 out of this stream was quite important
here, as it wa
mike_b16 wrote:
> Any ideas? Also, thanks for all of the help so far bpa. It's gotten me
> this far?
You're doing a lot of work generating a PCM stream that is not
44.1kHz/16 but I have a feeling that you need tell LMS the stream format
before you start stream audio data so that LMS know whethe
I have been somewhat successful!
After going with the solution in post 47 of this thread (the snd-aloop
method) and overclocking my Pi's, I was able to play the stream with no
problems to my clients. I essentially used arecord to grab my line in
(with -D hw:0) and then pipe it to the virtual soun
mike_b16 wrote:
> I also tried change the filetype in the pcm portion of
> custom-convert.conf to a wav (-t wav) with the same results. If you're
> saying LMS is picky with the formatting I can change it back.
Basically LMS will create a process using the command line created using
the "template
Thanks for the answers. I'll parse through the LMS convert.conf file
right after I answer your question.
I can record from the command line using a few commands:
arecord -D hw:0 -c 2 -r 44100 -f S16_LE example_file.wav (rate and
format can change without problems)
arecord -D default -c 2 -r 4410
mike_b16 wrote:
> Also, is there a way to tell if my client unit is receiving the audio
> stream even if it's outputting no audio?
Enable logging.
bpa's Profile: http://forums.slimdevices.com/member.php?userid=1806
View t
mike_b16 wrote:
> Also, I just want to understand the custom-convent.conf file. Excuse the
> lack of Linux knowledge coming up here:
>
> 1. Why is there one section each for PCM, mp3, and flac? Do they all
> need to be there or can I comment some of it out?
> 2. The "# R" before the PCM arecord
Also, is there a way to tell if my client unit is receiving the audio
stream even if it's outputting no audio?
Cheers,
Michael
mike_b16's Profile: http://forums.slimdevices.com/member.php?userid=63606
View this thread: ht
Also, I just want to understand the custom-convent.conf file. Excuse the
lack of Linux knowledge coming up here:
1. Why is there one section each for PCM, mp3, and flac? Do they all
need to be there or can I comment some of it out?
2. The "# R" before the PCM arecord function is just a comment do
I tried to look through the active processes for anything that would
involve pulseaudio. I tried "ps aux | grep pulse" and "ps aux | grep pa"
and found nothing. I also shut down ecasound, the squeezeboxserver, and
squeezelite. I can record a .wav file fine with arecord. However, the
issue still oc
In The waveinput plugn thread some other have posted solution using
Pulseaudio on other lnux box.
1. Convert audio to a httpmp3 stream and then play audio direct just
like a normal stream
see
http://forums.slimdevices.com/showthread.php?49584-Announce-WaveInput-for-Linux&p=766290&viewfull=1#post7
Awesome. I'll go give these a try now. Thanks.
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View this thread: http://forums.slimdevices.com/showthread.php?t=102551
_
I think you have now run into Pulseaudio. I'm not a fan of PulseAudio -
others have no problems with it.
I have found Pulseaudio can solve problems I never have and causes
problems with what I want to do. So on my Linux boxes I have given upon
Pulseaudio and install ALSA.
To get WaveInput worki
Ok, so.
Due to the fact that I accidentally broke Perl trying to find ways to
fix the problem, I decided to flash my SD card again and install LMS and
squeezelite again. While installing it, I realized that 107 was the user
ID for the username "squeezeboxserver" which LMS created as a user upon
i
Looks like installation didn't go properly as no user name has been
associated with the user id 107.
To use a user id instead of a name with sudo prefix number with # (i.e.
sudo -u #107 arecord ...)
bpa's Profile: http://
Hey bpa,
As far as I can tell from the command line, user "107" is running LMS. I
tried using arecord, and it says it doesnt recognize user 107. I have
attached a screenshot of the results.
16720
Cheers,
Michael
+---+
|Filename:
I think you have Pulseaudio which can cause problem for WaveInput but
probably not if using the Wolfson line-in.
You need to check arecord recoding the Wolfson line-in running as the
user which LMS uses. When LMS is running do a "ps" command and find the
user name which is running LMS. Then run
Hey bpa,
Thanks for getting back to me. My Pi's are running on Raspbian with
Wolfson kernels. I never installed pulseaudio myself. There is a folder
called "pulse" in /etc, but I'm not sure if that means that this version
of Raspbian came with pulseaudio. I just installed LMS 7.8.0 from the
repos
What Linux distro ?
Do you have Pulseaudio ?
How is LMS run as a service or a process under a user login ?
Does the LMS user id have permissions to access the audio device ?
bpa's Profile: http://forums.slimdevices.com/memb
Hey there,
I am having issues using the WaveInput plugin on my Raspberry Pi unit.
For background information, I currently have two RPis (both with Wolfson
sound cards) set up for audio playback. One of the units has LMS
installed on it and both have squeezelite player installed. I can have
both p
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