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ralphpnj
2015-08-08, 14:36
Oh the wonders of digital streaming!

My backyard deck sound system:

Source: LMS streamed to iPhone using iPeng with the iPhone set as the player. The iPhone is connected via bluetooth to the Pyle Street Blaster. The selection I was listening to when the picture was taken was a 24bit/96kHz download of the new Dawes release, so LMS was also downsampling the files on the fly. And it sounds very good.

netchord
2015-08-12, 11:53
Oh the wonders of digital streaming!

My backyard deck sound system:

Source: LMS streamed to iPhone using iPeng with the iPhone set as the player. The iPhone is connected via bluetooth to the Pyle Street Blaster. The selection I was listening to when the picture was taken was a 24bit/96kHz download of the new Dawes release, so LMS was also downsampling the files on the fly. And it sounds very good.

very creative!

pippin
2015-08-12, 13:56
<3
Cool!

ralphpnj
2015-08-12, 14:01
<3
Cool!

Hey it would not have been possible without the wonderful and amazing iPeng app! Thanks Pippin!

marflao
2015-08-12, 21:40
Indeed... Very nice!
Is this done via iPeng Playback?

ralphpnj
2015-08-13, 04:25
Indeed... Very nice!
Is this done via iPeng Playback?

Yes! Here's how it works:

First make sure that one is able to access LMS on the iPhone, in other words that the iPhone is on the same WiFi network as LMS

then pair the iPhone and the Street Blaster together with bluetooth

then using iPeng select the iPhone as the "squeezebox" player

Finally listen to any music in the LMS library.

Apesbrain
2015-08-13, 04:50
Cool! Have been rocking this thing in the backyard for a couple summers:
http://forums.slimdevices.com/showthread.php?100725-Battery-Powered-quot-Boom-quot-Alternative&highlight=boom

marflao
2015-08-13, 05:03
Yes! Here's how it works:

First make sure that one is able to access LMS on the iPhone, in other words that the iPhone is on the same WiFi network as LMS

then pair the iPhone and the Street Blaster together with bluetooth

then using iPeng select the iPhone as the "squeezebox" player

Finally listen to any music in the LMS library.
Thanks for the clarification, Ralph.

Just read that this is also possible on Android. Since I use OrangeSqueeze I only need to get the SqueezePlayer app.
I might check this out.

ralphpnj
2015-08-13, 05:05
Cool! Have been rocking this thing in the backyard for a couple summers:
http://forums.slimdevices.com/showthread.php?100725-Battery-Powered-quot-Boom-quot-Alternative&highlight=boom

I think we have stumbled upon an whole underground Squeezebox movement: the LMS/smartphone/tablet powered Bluetooth speaker system. Now let's see who else comes forward to bring their system into the light.

By the way, I noticed that you had the Band playing when you took the photo. What is about Americana and backyard systems?

ralphpnj
2015-08-13, 05:21
Thanks for the clarification, Ralph.

Just read that this is also possible on Android. Since I use OrangeSqueeze I only need to get the SqueezePlayer app.
I might check this out.

I also use the QrangeSqueeze app but I use it on a Nook and the Nook leaves a lot to desired especially when compared to either an iPhone or an iPad. However OrangeSqueeze is a really nice app.

marflao
2015-08-13, 06:22
So far I connected my Nexus 7 via Bluetooth to a Nude Audio "Super M" while playing songs from Google Music.
But it doesn't work without hiccups... Lot's of buffering.

So this LMS => SqueezePlayer combo might be a better option.

I'm just not 100% sure if my NAS has enough power for the downsampling?!

ralphpnj
2015-08-13, 11:22
So far I connected my Nexus 7 via Bluetooth to a Nude Audio "Super M" while playing songs from Google Music.
But it doesn't work without hiccups... Lot's of buffering.

So this LMS => SqueezePlayer combo might be a better option.

I'm just not 100% sure if my NAS has enough power for the downsampling?!

The downsampling that I was referring to was the result of the fact that the iPhone does not play 24bit/96kHz files so LMS was downsampling the files to 16bit/44.1kHz. If you are using standard CD resolution (16bit/44.1kHz) files or mp3 files then there would not be any downsampling. I would give the LMS/Squeezeplayer combo a try.

pippin
2015-08-13, 12:34
Hm? iPeng does play 24/96 natively.
Did you maybe enable bitrate limiting?

ralphpnj
2015-08-13, 13:19
Hm? iPeng does play 24/96 natively.
Did you maybe enable bitrate limiting?

Maybe it was playing the 24/96 without downsampling and I just didn't realize it. Is there any way to tell the bit and sampling rate of the file being played on the iPhone?

pippin
2015-08-13, 13:20
Sure, it's in the context menu for the current track under "more info"

philippe_44
2015-08-13, 16:26
So far I connected my Nexus 7 via Bluetooth to a Nude Audio "Super M" while playing songs from Google Music.
But it doesn't work without hiccups... Lot's of buffering.

So this LMS => SqueezePlayer combo might be a better option.

I'm just not 100% sure if my NAS has enough power for the downsampling?!

Just as a curiosity, wouldn't LMS downsampling from 96 to 48 be a simple puncturing of every other sample, so very limited CPU requirement ? (no spectrum aliasing expected)

pippin
2015-08-13, 16:28
LMS uses sox which does rather complex interpolation and actually eats quite a bit of CPU in the process. Even more than MP3 encoding and dramatically more than FLAC encoding.
Keeps surprising me, too

Mnyb
2015-08-13, 19:07
LMS uses sox which does rather complex interpolation and actually eats quite a bit of CPU in the process. Even more than MP3 encoding and dramatically more than FLAC encoding.
Keeps surprising me, too


Just as a curiosity, wouldn't LMS downsampling from 96 to 48 be a simple puncturing of every other sample, so very limited CPU requirement ? (no spectrum aliasing expected)

Yes pippin is right LMS does state of the art downsampling when a player needs it it's inaudible to the listener a very good choice.

But if your server can't take it that's a problem .

Very very old server versions >10 years ago ,that where around at the time of slimp3 and SB1 might just dumped 1/2 of the samples , but at those time squeezeboxes where fun gadgets before anyone at slimdevices realised that it had hifi potential ,that came with the SB2.

philippe_44
2015-08-13, 19:50
Yes pippin is right LMS does state of the art downsampling when a player needs it it's inaudible to the listener a very good choice.

But if your server can't take it that's a problem .

Very very old server versions >10 years ago ,that where around at the time of slimp3 and SB1 might just dumped 1/2 of the samples , but at those time squeezeboxes where fun gadgets before anyone at slimdevices realised that it had hifi potential ,that came with the SB2.

Agreed but in the special case of 192/96/48 there is no need for filtering, so I was wondering if that optimization was there and could help by setting downsampling to 48 instead of 44.1

Mnyb
2015-08-13, 22:07
Agreed but in the special case of 192/96/48 there is no need for filtering, so I was wondering if that optimization was there and could help by setting downsampling to 48 instead of 44.1

There is always need for filtering for various reasons, but I'm afraid I can't explain it very good .
Simplest case there is signal content even if it's noise above the nyqvist limit for 44.1 or 48 k (22,05khz , 24khz) or in practice a little bit lover ,this will aliase back into the signal if you just drop samples . Therefore they are filtered before downsampling . There must be no signal content at all above the nyqvist limit of the target sample rate .

Mnyb
2015-08-13, 22:51
There is a low end alternative to lame , shine . Results is worse , but it actually works on Sheeva plug and other machines without floating piont .

I have not yet seen a cruder alternative to SoX for low end servers ?

Depending on CPU architecture and FPU I've had a server where lame used much more CPU than SoX.

But enough off topic from me , let's see some backyard systems :)

philippe_44
2015-08-13, 22:56
There is always need for filtering for various reasons, but I'm afraid I can't explain it very good .
Simplest case there is signal content even if it's noise above the nyqvist limit for 44.1 or 48 k (22,05khz , 24khz) or in practice a little bit lover ,this will aliase back into the signal if you just drop samples . Therefore they are filtered before downsampling . There must be no signal content at all above the nyqvist limit of the target sample rate .

You're right but here is what I meant more precisely.

Sampling a signal s at a rate of T has the effect of duplicating its spectrum an infinite amount of time with an n/T spacing. Obviously if the spectrum is larger than 1/T (max complex frequency 1/2T), then overlapp (aliasing) occurs and original info is lost.
So, assuming that spectrum is below that 1/T limit, sampling at T/2 has the effect of duplicating spectrum every 2n/T and sampling at 4T spaces it at 4n/T. So, mathematically speaking, a signal with a spectrum below 1/T and sampled at T/2 or T/4 can be simply punctured at 1/2 or 1/4 without any loss of information, assuming that you do a perfect cardinal sine filtering when you switch back to analogue domain.

What I meant by "no filtering needed" is that if you want do downsample at non integer multiple of initial rate, then you have to re-interpolate (filtering ...) in the digital domain at the new rate . When it is an integer multiple, there is no need of that. And if you downsample in respect with spectrum size, there is no need to lowpass filter

In "real life", the analogue conversion filtering is not a perfect cardinal sine and the benefit of oversampling is that by increasing space between spectrum "replica", you ease the analogue filtering requirements. But, without entering into an audiophile debate, assuming a decent DAC, I was suggesting that in case the host CPU is a problem, moving from 96K to 48K without filtering should be done by 1/2 puncturing which requires no CPU, re-interpolating and lowpass is not needed

(PS: I'm not trying to be pedantic, sorry if it looks like that :()

Mnyb
2015-08-14, 00:50
You're right but here is what I meant more precisely.

Sampling a signal s at a rate of T has the effect of duplicating its spectrum an infinite amount of time with an n/T spacing. Obviously if the spectrum is larger than 1/T (max complex frequency 1/2T), then overlapp (aliasing) occurs and original info is lost.
So, assuming that spectrum is below that 1/T limit, sampling at T/2 has the effect of duplicating spectrum every 2n/T and sampling at 4T spaces it at 4n/T. So, mathematically speaking, a signal with a spectrum below 1/T and sampled at T/2 or T/4 can be simply punctured at 1/2 or 1/4 without any loss of information, assuming that you do a perfect cardinal sine filtering when you switch back to analogue domain.

What I meant by "no filtering needed" is that if you want do downsample at non integer multiple of initial rate, then you have to re-interpolate (filtering ...) in the digital domain at the new rate . When it is an integer multiple, there is no need of that. And if you downsample in respect with spectrum size, there is no need to lowpass filter

In "real life", the analogue conversion filtering is not a perfect cardinal sine and the benefit of oversampling is that by increasing space between spectrum "replica", you ease the analogue filtering requirements. But, without entering into an audiophile debate, assuming a decent DAC, I was suggesting that in case the host CPU is a problem, moving from 96K to 48K without filtering should be done by 1/2 puncturing which requires no CPU, re-interpolating and lowpass is not needed

(PS: I'm not trying to be pedantic, sorry if it looks like that :()

He he so you mean by the typical 24/96 download which is fake it's really a 16/44 master but HD tracks don't tell you that :) there is not much that could aliase down ? Or for other reasons there are nothing much above the limit .
There might be real world issues anyway ,but that's beyond my detailed understanding . I think the current use of SoX is best practice .

But the LMS architecture may need a compromise solution for low CPU servers that just do as you suggest with multiples of the sample rate giving end results that's playable but may be compromised . Or is there a less CPU demanding resampler out there .
Or is it so simple as give SoX the right commands and it runs a less demanding procedure .

But how many low CPU servers is there today ? Would not mores law fix this faster than the comunity finds a solution ?

pippin
2015-08-14, 02:10
No, what he's saying is that if you just drop every second sample that would be the same process as sampling at a lower sampling rate all along.
If there is really any improvement in a particular recording from going from a 48 kHz sample rate to a 96 kHz sample rate that improvement will be lost in the process but you still get something slightly superior to CD quality which should be fine for applications where you don't have a 96kHz DAC anyway. You might be able to get a very slight improvement over that by interpolation so you would "save" some of the benefits of the higher sample rate recording but if you don't have the CPU required, the result is actually a broken playback which is the worst SQ you'll ever have. Working always beats "theoretically better but not working".

Those upsampled tracks have gone through so many potentially distortion- and aliasing-adding conversions it probably doesn't matter what you are doing to them anyway, they will be worse than the original 44.1/16 recording anyway.

Mnyb
2015-08-14, 02:30
No, what he's saying is that if you just drop every second sample that would be the same process as sampling at a lower sampling rate all along.
If there is really any improvement in a particular recording from going from a 48 kHz sample rate to a 96 kHz sample rate that improvement will be lost in the process but you still get something slightly superior to CD quality which should be fine for applications where you don't have a 96kHz DAC anyway. You might be able to get a very slight improvement over that by interpolation so you would "save" some of the benefits of the higher sample rate recording but if you don't have the CPU required, the result is actually a broken playback which is the worst SQ you'll ever have. Working always beats "theoretically better but not working".

Those upsampled tracks have gone through so many potentially distortion- and aliasing-adding conversions it probably doesn't matter what you are doing to them anyway, they will be worse than the original 44.1/16 recording anyway.

Very likely , but I do have some real content that ain't to broken that do contain above 24khz signal there are actual hires recordings just far fewer than audiophiles care to admit ( even fewer that contains music anyone cares for ) . But obviously I won't care in the garden or on mobile :)

One obvious choice we all have is to just down convert everything offline , I convinced that I can't hear the difference even on state of the art recordings down converted to 16/44.1 if you do 24/44.1 and 24/48 of everything you are probably good to go and enjoy life and music

marflao
2015-08-14, 04:44
Well... I just installed SqueezePlayer on my tablet and it's playing well up to 24/44,1.
Above it stops playing.

Is there something I need to setup in LMS that songs with bitrates up to 24/192 can be played or will this not be possible?

It's not that I will hear a difference between 16/44,1 and higher rates. But it would be more convenient if they would be played. Honestly I don't like the idea to downsampling them (lazy me).

Apesbrain
2015-08-14, 04:58
Is there something I need to setup in LMS that songs with bitrates up to 24/192 can be played...
You might be able to do it with a "custom-convert.conf" file that tells LMS to use SoX to resample all FLAC going to that device to 16/44. The contents of this text file would look something like this:



flc flc * 00:00:00:00:00:00
# FT:{START=--skip=%t}U:{END=--until=%v}
[flac] -dcs $START$ $END$ -- $FILE$ | [sox] -q -t wav - -t flac -C 0 -b 16 -r 44.1k -

Where 00:00:00:00:00:00 is the MAC address of your tablet.

This file goes on your server in same folder as "convert.conf". Restart.

marflao
2015-08-14, 05:09
Thanks for that hint, apesbrain.

Just one question: in case I would choose my Touch as the player this custom conversion would not be applicable (because the Mac address of the tablet is used), right?
Or am I wrong and the songs will also be downsampled once I'll choose the Touch?

philippe_44
2015-08-14, 09:08
He he so you mean by the typical 24/96 download which is fake it's really a 16/44 master but HD tracks don't tell you that :) there is not much that could aliase down ? Or for other reasons there are nothing much above the limit .
There might be real world issues anyway ,but that's beyond my detailed understanding . I think the current use of SoX is best practice .

But the LMS architecture may need a compromise solution for low CPU servers that just do as you suggest with multiples of the sample rate giving end results that's playable but may be compromised . Or is there a less CPU demanding resampler out there .
Or is it so simple as give SoX the right commands and it runs a less demanding procedure .

But how many low CPU servers is there today ? Would not mores law fix this faster than the comunity finds a solution ?

You're right on CPU, it should not be the issue unless you want to have many players in parallel.

I also meant that tracks made to be played should have nothing in spectrum above 20KHz, it is useless. The benefit of A/D oversampling is to push away spectrum images so that you can use easier analogue filters and then in the digital domain, you should eliminate anything above between 20KHz (up to Fs/2 or course) by digital filtering where you can use all the complicated, non real-time, post-processing in the world. After that, your file can be downsampled for size improvment with no information loss. Then when you do the D/A process, the benefit of up-sampling is that again, with the images being rejected further, the analogue filters can be less complicated. But up-sampling and interpolation could be done realtime, do not need to store the over-sampled file.

But you're right, I'm hijacking the original thread on top of risking to start another flame war

Mnyb
2015-08-14, 09:33
You're right on CPU, it should not be the issue unless you want to have many players in parallel.

I also meant that tracks made to be played should have nothing in spectrum above 20KHz, it is useless. The benefit of A/D oversampling is to push away spectrum images so that you can use easier analogue filters and then in the digital domain, you should eliminate anything above between 20KHz (up to Fs/2 or course) by digital filtering where you can use all the complicated, non real-time, post-processing in the world. After that, your file can be downsampled for size improvment with no information loss. Then when you do the D/A process, the benefit of up-sampling is that again, with the images being rejected further, the analogue filters can be less complicated. But up-sampling and interpolation could be done realtime, do not need to store the over-sampled file.

But you're right, I'm hijacking the original thread on top of risking to start another flame war

I get you :) but believers in hirez do want content above 20kHz even in the delivery format to customers . But I agree that most likely this creates problem in the playback chain tweeter resonances and IM and provoke IM in amplifiers etc and actually transformer resonances in tube amps etc .
So hirez dowloads to consumers do contain over 20kHz . Worst case they contain unfiltered DSD noise as the original might have been DSD ,that you *really* want to filter out .

I do understand that recording in very high resolution is necessary for a myriad of reasons , this is not the same topic as playback I constanly say this as this is always confused . (now thats done )

But i agree that not hearing them is the worst option , you really want the music .

On topic should not even the software players report back properly to LMS about their capability so LMS can downsample automatically . If you have to write you own convert conf's something seem broken to me ?? So bug report to the author of such player seems the next step

pippin
2015-08-14, 12:38
The last point is right. I suspect SqueezePlayer reports what it can handle but it doesn't do any own processing and whether it successfully plays depends on the capabilities of the Android device which vary much more than under iOS....

philippe_44
2015-08-14, 17:20
I get you :) but believers in hirez do want content above 20kHz even in the delivery format to customers . But I agree that most likely this creates problem in the playback chain tweeter resonances and IM and provoke IM in amplifiers etc and actually transformer resonances in tube amps etc .
So hirez dowloads to consumers do contain over 20kHz . Worst case they contain unfiltered DSD noise as the original might have been DSD ,that you *really* want to filter out

(sigh) understood - that's why I'm not an audiophile, I guess


I do understand that recording in very high resolution is necessary for a myriad of reasons , this is not the same topic as playback I constanly say this as this is always confused . (now thats done )

But i agree that not hearing them is the worst option , you really want the music .

On topic should not even the software players report back properly to LMS about their capability so LMS can downsample automatically . If you have to write you own convert conf's something seem broken to me ?? So bug report to the author of such player seems the next step

The LMS protocol lets you (at registration) specify a capability list that includes the maximum supported sample rate. Based on that, LMS does downsampling, I think. What is still missing is the possibility to declare the supported sample depth. Hence in my UPnPBridge plugin I still have to mess with sample size in case the player does not support 24 bits

Mnyb
2015-08-14, 22:41
The last point is right. I suspect SqueezePlayer reports what it can handle but it doesn't do any own processing and whether it successfully plays depends on the capabilities of the Android device which vary much more than under iOS....

Aha got that ,there are so many of android devices of varying quality and provenance . Maybe there is no reasonable way to query the hardware about it either , for squeezeplayer .

And to have a manual setting in the app , you get a support issue as it's going to be misunderstood .....

marflao
2015-08-14, 23:42
Hmm.... doesn't seem that easy.

Now I'm thinking of creating a playlist with songs up to 24/44,1 which I will use with SqueezePlayer.

Julf
2015-08-15, 00:58
Hmm.... doesn't seem that easy.

Now I'm thinking of creating a playlist with songs up to 24/44,1 which I will use with SqueezePlayer.

So you didn't get the custom-convert.conf to work?

marflao
2015-08-15, 01:13
Haven't tried that yet, Julf.
Reasons:
(i) i'm not so familiar in Linux (and i guess i need to ssh in that file, or?). But i will check this if
(ii) downsampling won't take place once i choose the Touch.
But from pippin's post above i understood that it will be downsampled as well with the Touch as player. Or was his post not a respond to my question? Maybe i misunderstood something ?!

Julf
2015-08-15, 02:35
(i) i'm not so familiar in Linux (and i guess i need to ssh in that file, or?).

Yes, you can prepare and edit it on another system, but in the end you have to get into the right place, so a brief ssh session is needed.


But i will check this if
(ii) downsampling won't take place once i choose the Touch.
But from pippin's post above i understood that it will be downsampled as well with the Touch as player. Or was his post not a respond to my question? Maybe i misunderstood something ?!

If the config is based on mac addresses, it only affects those specific players, so you can leave the Touch unaffected.

marflao
2015-08-15, 03:42
Yes, you can prepare and edit it on another system, but in the end you have to get into the right place, so a brief ssh session is needed.



If the config is based on mac addresses, it only affects those specific players, so you can leave the Touch unaffected.

OK..that sounds great, Julf.
Now my questions regarding the code for the conv file Apesbrain has mentioned:






flc flc * 00:00:00:00:00:00
# FT:{START=--skip=%t}U:{END=--until=%v}
[flac] -dcs $START$ $END$ -- $FILE$ | [sox] -q -t wav - -t flac -C 0 -b 16 -r 44.1k -

Where 00:00:00:00:00:00 is the MAC address of your tablet.

This file goes on your server in same folder as "convert.conf". Restart.

Is that enough or do I need to insert some other rows in that conv file (besides my SqueezePlayer´s MAC address)?

Thanks in advance.

Julf
2015-08-15, 04:07
Is that enough or do I need to insert some other rows in that conv file (besides my SqueezePlayer´s MAC address)?

That should be enough.

marflao
2015-08-15, 04:47
Thanks Julf.

Now another hurdle ;-)

I can´t log in via putty to the LMS. I tried it with the IP address and port 9000 but that doesn´t work. Putty shuts down.

Isn´t that how it should work? Once i would be connected login as "root" and PW "1234", or?

Sorry for the noob questions ;-)

Julf
2015-08-15, 05:01
I can´t log in via putty to the LMS. I tried it with the IP address and port 9000 but that doesn´t work. Putty shuts down.

It won't be listening for ssh on port 9000 - the correct port is 22 (but the ssh client should use that by default).


Isn´t that how it should work? Once i would be connected login as "root" and PW "1234", or?

Can you remind me - what system are you running LMS on?

marflao
2015-08-15, 05:04
LMS is running on a Synology NAS.

Tried to connect to port 22 and logged in as "root" but then it failed with PW "1234".

Julf
2015-08-15, 05:11
LMS is running on a Synology NAS.

Tried to connect to port 22 and logged in as "root" but then it failed with PW "1234".

I have no experience with the Synology, but according to google, it seems the default admin password is empty, not "1234".

marflao
2015-08-15, 05:19
I have no experience with the Synology, but according to google, it seems the default admin password is empty, not "1234".

Hmm...doesn´t work.
Need to dig a bit deeper into this later this afternoon. Thanks so far.

Julf
2015-08-15, 05:39
Hmm...doesn´t work.
Need to dig a bit deeper into this later this afternoon. Thanks so far.

You might have to create a ssh user through the synology web interface.

marflao
2015-08-16, 10:59
Well...at least I tried it but wasn´t successful in the end.

What did I do:
(i) used Notepad++, copy&past the code mentioned from Apesbrain and updated the Mac address with the one for my player (the Nexus 7) => saved the file as "custom-convert.conf" on my NAS (mounted to Explorer)
(ii) opened putty, logged in on my NAS with "root" and the pw I use to access DSM and moved the file to the folder where the "convert.conf" file is located (in my case /volume1/@appstore/LmsRepack/)
(iii) logged in the GUI of my NAS and clicked on package center. Stopped "LMS Repack" (I´m using pinkdot´s version) and started it again.
(iv) started Squezzeplayer on my tablet, switched to OrangeSqueeze and started a 24/94 song with the result that the player "Asus Nexus 7" stopped.

Now I´m stucked again ;-)
Any tip?

Apesbrain
2015-08-16, 12:28
Any tip?
Unfortunately, your NAS may not have enough CPU power to run SoX* in this way. If you want to confirm, temporarily install LMS on your PC and copy some of your hi-res music as well as the new custom-convert.conf to the appropriate places. Restart and try to play these files via your new LMS instance on your Nexus 7.

*Are we sure the Synology distribution of LMS even has SoX support?

marflao
2015-08-16, 13:13
Unfortunately, your NAS may not have enough CPU power to run SoX* in this way. If you want to confirm, temporarily install LMS on your PC and copy some of your hi-res music as well as the new custom-convert.conf to the appropriate places. Restart and try to play these files via your new LMS instance on your Nexus 7.

*Are we sure the Synology distribution of LMS even has SoX support?

Thanks Apesbrain.
Maybe I try it.

Btw... Don't know the answer wrt the Sox support yet.

All the best,
M.

Mnyb
2015-08-16, 14:09
Unfortunately, your NAS may not have enough CPU power to run SoX* in this way. If you want to confirm, temporarily install LMS on your PC and copy some of your hi-res music as well as the new custom-convert.conf to the appropriate places. Restart and try to play these files via your new LMS instance on your Nexus 7.

*Are we sure the Synology distribution of LMS even has SoX support?


Thanks Apesbrain.
Maybe I try it.

Btw... Don't know the answer wrt the Sox support yet.

All the best,
M.

You can check if a SoX binary is shipped with install they are all in the same folder , no I don't know where they reside on your NAS ( sorry , Google )

However this may be the problem all along , if SoX is not supported on your synology model that's may be why it's broken in the first place .
Just try LMS a real computer ,but don't yet add your custom.convert.conf . Maybe there is nothing wrong with the player .

Julf
2015-08-16, 23:31
You can check if a SoX binary is shipped with install they are all in the same folder , no I don't know where they reside on your NAS ( sorry , Google )

Worst case you can always do a "find / -name sox", but it will take a while...

marflao
2015-08-18, 08:47
Well.. the Sox file is located in one of the subdirectories of my LmsRepack folder.

Hmm...

Apesbrain
2015-08-18, 14:51
Well.. the Sox file is located in one of the subdirectories of my LmsRepack folder.
Ok, so you've got sox.exe but it may not be running. I know it's a bother but the best way for us to help is for you to install LMS on a PC, copy over some 16/44 and hi-res, and see if it works with your Nexus: without or with the earlier custom-convert.conf file.

marflao
2015-08-20, 23:35
I finally found some time to install LMS on my Win8 notebook.

I used the offical 7.7.5. version and after I set up everything I checked it with a 24/192 flac file and voilà ...it worked..played without any problems via my Nexus (and a custom-convert.conf file wasn´t even needed).

Am I right now that my NAS is the bottleneck?

Thanks so far for everybody´s help!!! Much, much appreciated.

Cheers,
M.

d6jg
2015-08-21, 15:13
Don't ditch the NAS just run LMS on a PC and point it at the storage on the NAS. Tell LMS the storage path by UNC path eg \\NASIP\MusicFolder use the IP address rather than name - assuming your NAS has a static IP. IPs always work, names can fail. You can map the drive as well. Both the PC and NAS should be wired not wireless.
It should be straight forward.

Mnyb
2015-08-21, 20:37
You can have both LMS running without problem the player can switch between them as needed . So you can start the win8 computer when needed .

Or make down converted copies at a resolution that works for you (as this is what LMS has to do anyway).

marflao
2015-08-21, 23:09
Don't ditch the NAS just run LMS on a PC and point it at the storage on the NAS. Tell LMS the storage path by UNC path eg \\NASIP\MusicFolder use the IP address rather than name - assuming your NAS has a static IP. IPs always work, names can fail. You can map the drive as well. Both the PC and NAS should be wired not wireless.
It should be straight forward.


You can have both LMS running without problem the player can switch between them as needed . So you can start the win8 computer when needed .

Or make down converted copies at a resolution that works for you (as this is what LMS has to do anyway).
Thanks guys,
Will do and go the notebook route for "backyard" activities :-)

Have a great sunny weekend.
Cheers,
M.