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Old 11th December 2010, 01:36 PM   #101
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Join Date: Jul 2003
Location: NJ
Quote:
Originally Posted by omc View Post
. Can I convert this into high res wave files so that it is dealt with differently. If anyone is doing this, could they please let me know works well for them.
Thanks for your post.
I got busy last night w/ dBpoweramp music converter and after some problems found it works very nicely in converting 24/96 flac to 24/96 wave @2 channels. Then after reindexing the music files, music plays back @ full resolution (no truncating to 1411.)
My laptop that is my music server is getting tired. It was very balky this AM after all the fooling around last night. It was also quite chilly last night.
I have another older laptop that I could load Votexbox on but it needs a pcmcia card for USB 2.0 and that's a gamble to make work.
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Old 11th December 2010, 09:46 PM   #102
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I posted too soon. A few times last nite and then once today, rather than music playback, when selecting a folder that contains 24/96 waves processed from an HDTracks download all that could be heard was noise similar to the interference heard between stations on an old radio. Very annoying and going to the next cut clears it immediately. Can't figure this out yet.
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Old 12th December 2010, 02:00 PM   #103
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The issue w/ the noise rather than music playback has cleared up. There remains only a buffer issue on the first cut of 24/96 Wave material. After starting, the playback drops for a few moments then continues. Higher resolution music involves more packets in the buffer and there is probably a way to increase the size of the buffer to correct this.
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Old 12th December 2010, 02:31 PM   #104
omc is offline omc  Canada
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Thanks, this leaves me with 2 possible ways to deal with this.

To answer the earlier question: When you open up squeezebox server has the screen split in two. If you select a song with out playing it on the left screen it will show song info. If you do this for the same song that you are currently playing it will show you how it was delivered (PCM rate etc).
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Old 12th December 2010, 10:13 PM   #105
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mb:
While playing back 24/96 flac, if you click on the song title on the right pane, the playback bit rate shown after the VBR rate will be 1411 or truncated in the left pane.
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Old 12th December 2010, 10:20 PM   #106
mb is offline mb
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Quote:
Originally Posted by omc View Post
To answer the earlier question: When you open up squeezebox server has the screen split in two. If you select a song with out playing it on the left screen it will show song info. If you do this for the same song that you are currently playing it will show you how it was delivered (PCM rate etc).
Ah.. I see it. It looks like a bug on SBserver

1. I have checked the hardware settings on the SBT when playing 24/96 pcm. It is exactly as expected:

# more hw_params
access: MMAP_INTERLEAVED
format: S24_LE
subformat: STD
channels: 2
rate: 96000 (96000/1)
period_size: 960
buffer_size: 1920

2. I've run (again!) the flac decoding command used by SBserver to send the pcm over to the SBT, and it definitely decodes to 24/92 pcm. In fact in setting up my conversion to include room correction, I've found that the SBT will assume the sample rate and word format of the pcm to be that of the flac file, so you _must_ send 24/96 pcm if the flac reports to be 24/96. Otherwise you get the noise which was described elsewhere.

Converting hires flacs to wav is going to waste disk space, and cause audio tags to be lost. Can you please report the bug to Logitech?
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Old 13th December 2010, 12:39 AM   #107
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mb:
Where are you getting your info from?
On SBS when playing a hirez (24/96) flac, I see that the actual VBR of the cut is converted to 1411kbps PCM while playing, when flac is converted to PCM at the server.
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Old 13th December 2010, 04:06 AM   #108
mb is offline mb
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Hi Larry,

The alsa hw parameters are direct from the SBT, via ssh. This is a close I can peek into the hardware settings in alsa as I know.

On an ssh session into the SBT:

# more /proc/asound/card0/pcm0p/sub0/hw_params
access: MMAP_INTERLEAVED
format: S24_LE
subformat: STD
channels: 2
rate: 96000 (96000/1)
period_size: 960
buffer_size: 1920
#

I was puzzled by the server secreen indicating that the flac was converted to 1141kHz pcm too. However the flac program that does the conversion on the server, as I mentioned earlier is:

[flac] -dcs --force-raw-format --endian=little --sign=signed $START$ $END$ -- $FILE$

It means decode to stdout quietly (-dcs) the send it to the SBT. Flac is not able to resample anyway, so 96kHz flac will decode as 96kHz pcm. The message about 1141kHz pcm is from some programmer on the SBserver end, who probably did not consider that higher rate flac might be sent to th SBT. I'm familiar with these details because I needed to modify a brutefir wrapper program to send pcm to the SBT, instead of re-encoding it back to flac, then sending it to the SBT to be decoded (one extra set of encode/decode on server and SBT!)

Last edited by mb; 13th December 2010 at 04:08 AM.
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Old 13th December 2010, 05:03 AM   #109
omc is offline omc  Canada
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I think I will have to try mb's brutefirdrc trick. It says that pcm is being downgraded on squeeze server, and it sure sounds like it. 44.1 sounds great, but 96/24 sounds almost like an mp3. The hardware may be able to handle it, but squeeze server may be calling on another object, class or whatever (stuff I remember from engineering school) that only does 1411kbps.
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Old 13th December 2010, 05:42 AM   #110
mb is offline mb
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Omc: I might not have been clear. I do NOT believe that in your case 24/96 flac is being downsampled. In fact, I am quite sure it is not. What you are hearing is most likely the difference between 24/96 and 44.1.

I have traced the way the server is converting flac to pcm (without brutefir in between), and there is no downsampling. The message you see about 1141 kHz pcm is probably an *bug*, and the data is sent to the SBT as 24/96.

If you're thinking of adding brutefir to your playback chain, let me caution you that for the SBT it is a pretty manual task, as brutefirdrc has not been ported to the SBT. Measuring and getting the right settings for room correction is non-trivial, then making it send pcm to the SBT was another struggle.
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