gorman wrote: 
> Ok, found solution for MP3s as well. Since I have music only in flac and
> mp3 my solution is complete.
> 
> > 
Code:
--------------------
  >   > 
  > mp3 flc * bc:5f:f4:bf:3e:70
  >     # FT:{START=-ss %t}
  >     [ffmpeg] -loglevel quiet $START$ -i $FILE$ -f wav - | [sox] -D -q -t 
wav - -t flac -C 0 -b 24 - gain -6.9 equalizer 30 0.43q +6.2 equalizer 699 
2.00q +1.0 equalizer 2924 4.11q -1.7 equalizer 6031 5.05q -6.2 equalizer 9825 
4.08q +2.2 equalizer 83 4.24q +1.3 equalizer 209 1.54q -1.2 equalizer 2842 0.1q 
+0.2 equalizer 12443 1.29q +2.7 equalizer 19698 0.46q -11.8
  > 
--------------------
> > 
> 
> I had to use ffmpeg because, as far as I can see, LAME does not accept
> time offsets when decoding
> (https://svn.code.sf.net/p/lame/svn/trunk/lame/USAGE). Don't know how
> the internal functionality of mp3 transcoding works (when lowering
> bitrates) because that is capable of seeking but the convert.conf line
> does not show any instruction for that. I could probably be using
> ffmpeg for flac files too but I guess there's no point or added
> value.
> 
> If there are mistakes in my solution above, please do point them out.
> 
> Edit: can anybody explain the use of the --until=v% command when using
> flac decoding? With the above line in ffmpeg I've got complete seek
> functionality and I don't get if I'm missing something and what would
> that be.
> 
> Edit 2: since bpa asked for it, this is now the finalized portion in
> custom-convert.conf
> 
> > 
Code:
--------------------
  >   > 
  > #Headphones
  > flc flc * bc:5f:f4:bf:3e:70
  >     # FT:{START=--skip=%t}U:{END=--until=%v}
  >     [flac] -dcs $START$ $END$ -- $FILE$ | [sox] -D -q -t wav - -t flac -C 0 
-b 24 - gain -6.9 equalizer 30 0.43q +6.2 equalizer 699 2.00q +1.0 equalizer 
2924 4.11q -1.7 equalizer 6031 5.05q -6.2 equalizer 9825 4.08q +2.2 equalizer 
83 4.24q +1.3 equalizer 209 1.54q -1.2 equalizer 2842 0.1q +0.2 equalizer 12443 
1.29q +2.7 equalizer 19698 0.46q -11.8
  >     
  > mp3 flc * bc:5f:f4:bf:3e:70
  >     # FT:{START=-ss %t}
  >     [ffmpeg] -loglevel quiet $START$ -i $FILE$ -f wav - | [sox] -D -q -t 
wav - -t flac -C 0 -b 24 - gain -6.9 equalizer 30 0.43q +6.2 equalizer 699 
2.00q +1.0 equalizer 2924 4.11q -1.7 equalizer 6031 5.05q -6.2 equalizer 9825 
4.08q +2.2 equalizer 83 4.24q +1.3 equalizer 209 1.54q -1.2 equalizer 2842 0.1q 
+0.2 equalizer 12443 1.29q +2.7 equalizer 19698 0.46q -11.8
  > 
--------------------
> > 
> 
> If somebody is privy to the inner workings of sox, I wonder about two
> things:
> 
> 1) The need to use the -D argument for no dithering.
> 2) The need to use the -b 24 argument for 24-bit output. This makes
> sense to me as it provides headroom for all equalization calculations.
> But I might be mistaken.

As to your two questions, here's what I concluded when I looked into
this a couple of years ago:

1. I don't think the -D option is needed when you've specified 24-bit
output. Here's what the SoX manual says:

"[B]y default, SoX automatically adds TPDF dither when the output
bit-depth is less than 24 and any of the following are true:
• bit-depth reduction has been specified explicitly using a command-line
option
• the output file format supports only bit-depths lower than that of the
input file format
• an effect has increased effective bit-depth within the internal
processing chain"

2. SoX uses 32-bit processing internally for its effects, incl.
equalization, so that already offers plenty of headroom. But if you
didn't specify "-b 24", then SoX would apply dither to the output file,
esp. if the input file, like a lot of CD-quality music, is 16 bit. That
said, I gather that savage wars have been fought over dithering,
noise-shaping, etc., so I wouldn't want to comment any further.

With that out the way, I have to admit that I've never seen the point of
the commented line (# FT:{START=...) and the first part of the following
line, up to the pipe that sends output to SoX. Instead, I've been using
this stripped-back custom-convert for equalization for almost two years
now:


Code:
--------------------
    flc flc * 00:00:00:00:00:00
  [sox] -q -t flac $FILE$ -b 24 -t flac - equalizer 40.8 5q -11.4 equalizer 
62.8 2.487q -11.3 equalizer 87.7 5q -3.2 equalizer 263 5q -7.4
  
--------------------


This works just fine, without the additional code. So what have I been
missing?


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