transcoding MP3 to 96k FLAC

[bpa]

flc flc * <MAC goes here>
	# FT:{START=--skip=%t}U:{END=--until=%v}
    	[flac] -dcs $START$ $END$ -- $FILE$  |  [sox] -q -t wav - -t flac -e signed -C 0 -b 24 - rate -v -b 98 -L -a 96000

mp3 flc * <MAC goes here>
	[lame] --mp3input --decode -t --silent $FILE$ - |  [sox] -q -t wav - -t flac -e signed -C 0 -b 24 - rate -v -b 98 -L -a 96000

aac flc * <MAC goes here>
	# IF
	[faad] -q -w -f 2 $FILE$  |  [sox] -q -t wav - -t flac -e signed -C 0 -b 24 - rate -v -b 98 -L -a 96000

Main problem is the conversion rules have problems and so are breaking the conversion and so it fails.

If you are going modify convert.conf rules - you really need to study the options.

In the MP3 flc. Lame by default decodes to WAV. The "-t" option tells lame to output PCM and not WAV. In the sox part receiving - the first "-t wav" tell sox to expect WAV format. So immediately there is a mismatch.

After fixing that then sox complained that "-e signed" is not a supported encode option.

So a MP3 FLC command that should work would be

	[lame] --mp3input --decode --silent $FILE$ - |  [sox] -q -t wav - -t flac -C 0 -b 24 - rate -v -b 98 -L -a 96000

sox can process MP3 directly so there is no need for lame so in theory the following could work but I think you may lose the jump to offset capability

[sox] -q -t mp3 - -t flac -C 0 -b 24 - rate -v -b 98 -L -a 96000

However - depending on what you play (e.g. files, podcasts, stream, spotty, etc.) I think there maybe potential problems with your choice of "# FT" and "# IF" and missing "#" options.

I think you have the same issues with the faad lines . The "-f 2" (raw PCM) should be "-f 1" (WAV) and there should be no "-e signed"

[socialxray]

BPA you totally RAWK!!!

MP3 is squared away and working beautifully!!!

AAC setting in custom-convert.conf is still being ignored. The file plays but the sample rate is still 44.1k. So I though about what you said about MP4 and disabled the MPEG-4 to AAC added this to my custom-convert.conf.

mp4 flc * <MAC goes here>
# IF
[faad] -q -w -f 1 $START$ $END$ $FILE$ | [sox] -q -t wav - -t flac -C 0 -b 24 - rate -v -b 98 -L -a 96000

I know I am on the right track because the test file no longer plays. Same error as before on the Transporter display.
PROBLEM: CAN"T OPEN FILE:
All That We Are

I uploaded the server log.

Also what '#' options would you recommend? I not not very sure what these options mean. I am really just going through the convert.conf and copying and pasting into custom-convert.conf wiht the addition of the sox command.

THANKS AGAIN!@

[socialxray]

Oh snap! I got it working!!!

[socialxray]

SO I messed with the custom-convert.conf and came up with this.

mp4 flc * *
# FT:{START=-j %s}U:{END=-e %u}
[faad] -q -w -f 1 $START$ $END$ $FILE$ | [sox] -q -t wav - -t flac -C 0 -b 24 - rate -v -b 98 -L -a 96000

Adding those '#' options from the "mp4x flc" entry from convert.conf worked. Honestly it was just guessing; I don't really know what I am doing.

But thank you BPA!!!

I would have never gotten this far without your help. Do you have a donation page?
Also I would like to get your advice on the '#' options. I am not sure if I have them correct and it seems they can make or break the up-sampling.

My current custom-convert.conf looks like this.

flc flc * <MAC goes here>
# FT:{START=--skip=%t}U:{END=--until=%v}
[flac] -dcs $START$ $END$ -- $FILE$ | [sox] -q -t wav - -t flac -C 0 -b 24 - rate -v -b 98 -L -a 96000

mp3 flc * <MAC goes here>
[lame] --mp3input --decode --silent $FILE$ - | [sox] -q -t wav - -t flac -C 0 -b 24 - rate -v -b 98 -L -a 96000

mp4 flc * <MAC goes here>
# FT:{START=-j %s}U:{END=-e %u}
[faad] -q -w -f 1 $START$ $END$ $FILE$ | [sox] -q -t wav - -t flac -C 0 -b 24 - rate -v -b 98 -L -a 96000

aac flc * <MAC goes here>
# IF
[faad] -q -w -f 1 $FILE$ | [sox] -q -t wav - -t flac -C 0 -b 24 - rate -v -b 98 -L -a 96000

[bpa]

Also I would like to get your advice on the '#' options. I am not sure if I have them correct and it seems they can make or break the up-sampling.

Not using "native" playing will result in compromises.

LMS expects every player to be able to play MP3 natively - so disabling native MP3 may mean going into unknown territory and probably reduced functionality.

All LMS rules should have a "#" line. MP3 rules seem to not have them because MP3 was expected to be always available.

You need to spend time to understand what is happening and not just cut & paste lines in the vague hope they might work.

From the convert.conf file header

# Capabilities
# I - can transcode from stdin
# F - can transcode from a named file
# R - can transcode from a remote URL (URL types unspecified)
#
# E - extensions syntax E:{<key>=<value>,<key>=<value>}
#		NOSTART=I/F/R : no $START$ field when transcoding from I/F/R
#		NOHEADER=I/F/R : strip out header when transcoding from I/F/R (waf/aif only)
#
# O - can seek to a byte offset in the source stream (not yet implemented)
# T - can seek to a start time offset
# U - can seek to start time offset and finish at end time offset
#
# D - can downsample
# B - can limit bitrate

For your requirement F,I,R and T are probably main one of interest.

"R" may not be of interest as you should get bitrate change within a remote stream/services but I don't know internet streams/services you are using.

For MP3 - all input usually from "I" (i.e. stdin) so I think current setting may be OK although not sure without "T" (or "U") whether you'll be able to skip to a time offset in a file (i.e. maybe no ffwd or rew)

For AAC - current "#" lines in convert.conf for AAC, MP4, MP4x to FLC should be duplicated in your custom-convert.conf as you are trasncoding to FLC just like in main

for Flac files - like MP3 not sure without "T" (or "U") whether you'll be able to skip to a time offset in a file (i.e. maybe no ffwd or rew)