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Old 5th July 2011, 06:47 AM   #1971
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Hi folks.

One more remark to the float/integer thing.

I just learned that e.g. even ecasound (my prefered engine) was not transparent ( Kai Vehmanen - the guy in charge - called it asymmetric) prior to 2.8.0. .

2.8.0/2.8.1. is not even available in the Ubuntu repos btw.

Those folks running e.g. MPD should get that issue highlighted with the MPD folks. Just to make sure that they run symmetric.

Cheers
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Old 5th July 2011, 02:40 PM   #1972
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@Roberto

I just downloaded the 32bitSSE package which comes with a binary.

Then installed the dependecies

Code:
sudo apt-get install ffmpeg libmpcdec6 libmpg123-0 libsndfile1 libtagc0
And then I ran it:

Code:
cd /usr/src/libebur128-0.4.0-Linux-sse2 ; ./r128-scanner -t album <dir>
Works great incl. tagging and hirez material.
I've just written a script that runs libebur128
over my collection.


Thx again for the hint.

Cheers
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Old 6th July 2011, 05:57 AM   #1973
RR is offline RR  Europe
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Quote:
Originally Posted by soundcheck View Post

Works great incl. tagging and hirez material.
I've just written a script that runs libebur128
over my collection.


Thx again for the hint.

Cheers
I'm glad you liked it.

And thanks for the debian-ubuntu tips, I failed to install it in my ubuntu machine. The music server is a gentoo box, though. Those in that case can find the package in the pro-audio overlay, it is not in the main distro yet.

It is great stuff indeed. I particularly like the well tested algorithm in use, that make me feel confident about the results. You can find it detailed here:
http://www.itu.int/dms_pubrec/itu-r/...3-I!!PDF-E.pdf
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Old 6th July 2011, 06:31 AM   #1974
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Quote:
Originally Posted by RR View Post
It is great stuff indeed. I particularly like the well tested algorithm in use, that make me feel confident about the results. You can find it detailed here:
http://www.itu.int/dms_pubrec/itu-r/...3-I!!PDF-E.pdf
Yeah, but is everything really well-designed under accuracy aspects?
The K-filter is designed as a 2-stage filter, a first stage shelving filter and a 2nd stage highpass filter (RLB-weighting curve). The coefficients are given for a samplerate of 48 kHz and it is the task of the programmers to carry out the proper filters for other samplerates.
It would be perfect to define the filters as biquad filters with the given specs about frequency, Q and gain. Then it is easy to compute the filter coefficients for all samplerates. But now the coefficients are given only for 48 kHz. So if you assume now the filters as biquad filters you can try to solve the equations to get f0, Q and gain. This is possible for one of the filters. For the other filter you get just an approximate solution but no accurate one! So the K-filters for other samplerates (calculated from the approximate values for f0, Q and gain) will be more or less identical between different programs and give different results.

Uli
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Old 6th July 2011, 12:00 PM   #1975
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Quote:
Originally Posted by uli.brueggemann View Post
It would be perfect to define the filters as biquad filters with the given specs about frequency, Q and gain. Then it is easy to compute the filter coefficients for all samplerates. But now the coefficients are given only for 48 kHz.
Great point, it is extrange the way the weighting filter is defined. Given the nature of the estimation, I think the very small variations in results that have to be expected are not important, but assuring complete repeatability between different programs would have been better.

Test material exist to check implementations (EBU TECHNICAL - Loudness) but it should be resampled (it is all 48 khz) for each case.
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Old 6th July 2011, 01:07 PM   #1976
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There is an unofficial paper for the parameter estimation: http://www-public.tu-bs.de:8080/~y00...-1-filters.pdf

libebur128 is using exactly the estimated parameters. So far ok. It just should be noted that calculating the coefficients from the estimated parameters differ from the coefficients in the specs.

So I'm wondering why the specs define the coefficients with such precision. You can use them for 48 kHz but not for other samplerates. Or you can use the estimated parameters (like libebur128) but then you do not precisely follow the specs at least @ 48 kHz

Uli
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Old 24th July 2011, 11:39 AM   #1977
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Default Linux and Acourate, a perfect pair!

Hi there,
I am diying PC-audio since several years and began with Windows XP and Tact 2.0, followed by Bidule from ploque and DRC on OSX.
The latest version was Winows7 64bit on an overclocked Intel dual core system. The crossovers were built with Sonoris Mastering EQ and I used still DRC on SIR2 as convolver. I was really satisfied with the sound quality. Especially when I changed the soundcard from an ESI ESP1010e to the wonderful X-Meridian 2G from Auzentech with a change of the OPAmps.

Then I read about the qualities of Acourate and tried it out with a stereo correction file in this constellation. I was flashed from the effect on the sound and could not listen to music without this room correction anymore.

But I wanted to use the phase linear crossovers of acourate as a try and wanted to work with a Linux system.
After some days of trying and with "a little help from my friends", I got to the GNU/Parsix 64bit system, which ist more comfortable than the text-based distributions without having a big overload of unnessecary stuff like Ubuntu and others.
Now I am running the Acourate crossovers with integrated room and measuring-microfone correction in brutefir, which is giving the signal via jack to the soundcard. The player is aqualung, which is the only one I found to be able to play uncompressed wav-files.
I am overwhelmed by the sound, Never thought, that this could be possible.
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Old 24th November 2011, 11:04 AM   #1978
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Lightbulb beets: the music geek’s media organizer

A very interesting new entry in Debian "Sid":

beets: the music geek's media organizer

beets: the music geek?s media organizer &mdash; beets v1.0b10 documentation

Quote:
Beets is the media library management system for obsessive-compulsive music geeks.

The purpose of beets is to get your music collection right once and for all. It catalogs your collection, automatically improving its metadata as it goes using the MusicBrainz database. (It also downloads cover art for albums it imports.) Then it provides a bouquet of tools for manipulating and accessing your music.
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"We should no more let numbers define audio quality than we would let chemical analysis be the arbiter of fine wines." N.P.
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Old 24th November 2011, 02:34 PM   #1979
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did you tried it?


Pierre



Quote:
Originally Posted by UnixMan View Post
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Old 24th November 2011, 03:15 PM   #1980
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I'm using dbpoweramp which comes with the so called Perfect Meta feature.

Musicbrainz is just one out of several databases used to assemble the "perfect"

tags.

My opinion: I havn't seen any database (accessible to normal users) that's generating consistent tags.

The DPB Perfect Meta thing I consider useless.


I doubt that beets can do it any better.
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