Minimserver users...transcode FLAC to WAV?

[lmitche]

Yes, it enables Mick Jagger mode as well!

[Cebolla]

Oops, lets try again: convOut=-af aresample=resampler=soxr{semI-colon}precision=28. You will need the ffmpeg-full library for this to work.

 

Damn emoticons!

If you Go Advanced in the editor, you can set it to Disable smilies in text in the Additional Options.

 

convOut=-af aresample=resampler=soxr:precision=28

 

 

Incidentally, in case anyone else is interested in resampling via UPnP/DLNA streaming, using MinimServer's support for the SoX resampler:

Resampling

 

It requires the use of the MinimStreamer optional transcoder package and uses SoX via ffmpeg with libsoxr (needs to be installed separately).

[lmitche]

Thanks Cebolla!

[sydlow]

Nevertheless, the wav file data being received by the Aries should still be 24-bit, assuming that MinimServer's transcoder (MinimStreamer) has been properly set to wav24. It's possible the Lightning DS app might be caching older links to MinimServer's 16-bit transcoder URLs, so will require a rescan of the Lightning DS app to apply MinimServer's latest settings of links to the 24-bit transcoder URLs.

 

Yep, thanks. Refreshing the Cache seems to have helped.

However, Lightning DS now shows 32bit with the WAV24 setting. Does that sound right?

 

However, the tracks in a Playlist isn't refreshed after rebuilding the cache and still links to the 16-bit URLs. Any idea how to force those to refresh as well?

 

Syd

[Distinctive]

The obvious answer would be to delete the LDS app and restore it from App store....(?)

[lmitche]

Yep, thanks. Refreshing the Cache seems to have helped.

However, Lightning DS now shows 32bit with the WAV24 setting. Does that sound right?

 

However, the tracks in a Playlist isn't refreshed after rebuilding the cache and still links to the 16-bit URLs. Any idea how to force those to refresh as well?

 

Syd

 

My SMS-100 shows a 32 bit stream after Minimstreamer wav24 transcoding on playback. It seems to be a Linux thing. It appears there are only 16 and 32 bit output modes.

[sydlow]

The obvious answer would be to delete the LDS app and restore it from App store....(?)

 

Yeah, but would mean I'd have to construct my playlists again wouldn't it?

[scan80269]

My colleague and I use Auralic Aries to drive Vega DAC, and Aries delivers better sound when MinimServer/MinimStreamer does FLAC to WAV transcoding. The sound stage is more realistic. My hypothesis is that Aries performs better when its processor has less computational work to do.

 

I'm still experimenting between upping the bit depth of redbook to 24 bits (FLAC:WAV24) and preserving bit depth (FLAC:WAV). The sound is different, but I'm currently on the fence as to which is better. More listening...

[Cebolla]

Yep, thanks. Refreshing the Cache seems to have helped.

However, Lightning DS now shows 32bit with the WAV24 setting. Does that sound right?

 

However, the tracks in a Playlist isn't refreshed after rebuilding the cache and still links to the 16-bit URLs. Any idea how to force those to refresh as well?

Syd

 

I don't own an Aries, but my Pioneer N-50 streamer displays MinimServer's transcoder wav24 setting correctly as 24-bit and so does the Foobar2000 software player using the foo_UPnP plugin to enable its UPnP/DLNA rendering function. So I suspect its probably an issue with the Aries and/or the Lightning DS app not displaying the correct value for some reason.

Have you tried network streaming an actual 24-bit wav file to test the Aries with (as opposed to the 24-bit converted output from MinimServer's transcoder)? You can download some test samples of 24-bit wav files from the Naim Label record label site:

naimlabel.com

 

Ah, I didn't realise you also had some Lightning DS playlists with the links to MinimServer's transcoder's original 16-bit URLs. That situation would be similar to having (old) playlists containing links to files that have been deleted from the UPnP media server, so you can't expect the Lightning DS cache refresh to account for those sorts of changes. Unfortunately, those playlists will need to be created from scratch, unless you can somehow manually edit their contents to change their tracks' linked URLs. You'll need to ask Auralic if that's at all possible.

 

John

[sydlow]

Thanks John, I figured so.

 

I'm now going to go to your next suggestion and get soxr resampling working on the Synology. Stand by!

 

Are you using a Synology or another platform for your minimserver?

[sydlow]

Have you tried network streaming an actual 24-bit wav file to test the Aries with (as opposed to the 24-bit converted output from MinimServer's transcoder)?

 

Yes, 24bit encoded files play as 24 bit in Lightning DS - others have said the 32bit issue might be a Linux one.

[lmitche]

Looking at the docs, Linux Alsa pads 24 bits out to 32.

[master]

A little late to the party, but primarily necessitated by CCA not playing some 24 bit FLAC rips.

 

I've only started with transcoding all FLAC content to 24 bit WAV.

 

The changes are subtle, in fact I had to listen to the same song (George Michael's Father Figure) multiple times to make out the difference, and I'm not even sure I could do it correctly in blind AB testing.

 

That said, more detail with 24 bit WAV, more bass extension, and you get the feeling you can hear George Michael straining more on the high notes as well as hear his breathing/gasping with a little bit more detail.

 

Obviously, needs more listening, a lot more listening I'd say, but even if there was no difference and the transcoding to 24 bit WAV played all of my 24-bit FLAC rips on Chromecast Audio then that is a win I will gladly take.

[udis]

Oops, lets try again: convOut=-af aresample=resampler=soxr{semI-colon}precision=28. You will need the ffmpeg-full library for this to work.

 

Damn emoticons!

 

 

Hi. a few questions:

 

1. How can i check if i have the full ffmpeg library? Here's what my system tab looks like - Image 2016-11-02 at 11.49.01 AM.png

 

2. relating to above, do I need to install the soxr library separately? I am running Synology 214+

 

3. In which field(s) do i enter the above rule? Additionally, do i need to enter the flac to wav24 rule or does the soxr replaces it? If i do, in which field?

 

thank you... transcoding is new to me!

[udis]

Hi. a few questions:

 

1. How can i check if i have the full ffmpeg library? Here's what my system tab looks like - Image 2016-11-02 at 11.49.01 AM.png

 

2. relating to above, do I need to install the soxr library separately? I am running Synology 214+

 

3. In which field(s) do i enter the above rule? Additionally, do i need to enter the flac to wav24 rule or does the soxr replaces it? If i do, in which field?

 

thank you... transcoding is new to me!

It appears that Synology 214+ is ARM which doesn't offer libsoxr support...

[jcn3]

It appears that Synology 214+ is ARM which doesn't offer libsoxr support...

 

you're correct that the arm-based ones (like mine) don't support. from your pic, you can see that ffmpeg is installed and ready to go. to transcode, you can do something similar to what i've done (at bottom):

 

[udis]

you're correct that the arm-based ones (like mine) don't support. from your pic, you can see that ffmpeg is installed and ready to go. to transcode, you can do something similar to what i've done (at bottom):

 

[ATTACH=CONFIG]30394[/ATTACH]

Thanks. there's no need to set the bit rate?

[jcn3]

Thanks. there's no need to set the bit rate?

 

only if you want to specify. otherwise it keeps it at the native bitdepth and sample rate. on the last one, you can see that for the dsd conversion, i had to specify the sample rate as 176k since 192k is the highest my dac will accept.

[lmitche]

 

3. In which field(s) do i enter the above rule? Additionally, do i need to enter the flac to wav24 rule or does the soxr replaces it? If i do, in which field?

 

thank you... transcoding is new to me!

 

The convOut string goes into the stream.options field.

[udis]

only if you want to specify. otherwise it keeps it at the native bitdepth and sample rate. on the last one, you can see that for the dsd conversion, i had to specify the sample rate as 176k since 192k is the highest my dac will accept.

Got it. thank you.

[udis]

The convOut string goes into the stream.options field.

thank you.

[tboooe]

you're correct that the arm-based ones (like mine) don't support. from your pic, you can see that ffmpeg is installed and ready to go. to transcode, you can do something similar to what i've done (at bottom):

 

[ATTACH=CONFIG]30394[/ATTACH]

Guys...sorry for missing this. I've been in talks with a developer who has compiled ffmpeg + libsoxr. I first contacted the developer last year because at the time I had an ARM based QNAP. I hope this helps people.

 

https://www.johnvansickle.com/ffmpeg/

[udis]

you're correct that the arm-based ones (like mine) don't support. from your pic, you can see that ffmpeg is installed and ready to go. to transcode, you can do something similar to what i've done (at bottom):

 

[ATTACH=CONFIG]30394[/ATTACH]

Is there a way to confirm this is working correctly? My settings are as you defined in the screenshot. Would my Aries display all incoming FLAC as WAV?

[jcn3]

Is there a way to confirm this is working correctly? My settings are as you defined in the screenshot. Would my Aries display all incoming FLAC as WAV?

 

if you put the same settings in the stream.transcode line as in the screenshot, then the first parameter specifies that all flac files be transcoded to wav files. your aries would display all incoming files as wav.

[udis]

if you put the same settings in the stream.transcode line as in the screenshot, then the first parameter specifies that all flac files be transcoded to wav files. your aries would display all incoming files as wav.

hmmm.. that doesn't seem to be the case.