philippe_44 wrote:
> Almost. LMS has the source format and sample rate of a file, the target
> player's maximum sample rate and its supported list of codecs, and an
> optional time offset to start from. With these information, it scans the
> conversion rules to try to find a perfect match or at least a match with
> lowest "cost". The match can say: nothing to transcode, just send file
> or it can say "invoke this external chain of programs to transform from
> source format to that accepted format. Ultimately, the player has to
> decode to PCM, but in LMS, the transcoding might never make PCM
> "visible".
>
> So to make sure I get that correctly: These 48kHz MP3 do not play at
> all, right?
> It seems that in your case the maximum sample rate reported by the
> player is less than 48kHz and LMS does not find any transcoding rule,
> which would be one of these
>
> >
Code:
--------------------
> >
> # special case for smart bitrates on mp3 input
> mp3 mp3 transcode *
> # IFB:{BITRATE=--abr %B}D:{RESAMPLE=--resample %D}
> [lame] --silent -q $QUALITY$ $BITRATE$ $RESAMPLE$ --mp3input $FILE$ -
>
> flc flc transcode *
> # IFT:{START=--skip=%t}U:{END=--until=%v}D:{RESAMPLE=-r %d}E:{NOSTART=I}
> [flac] -dcs $START$ $END$ --force-raw-format --sign=signed
--endian=little -- $FILE$ | [sox] -q -t raw --encoding signed-integer -b
$SAMPLESIZE$ -r $SAMPLERATE$ -c $CHANNELS$ -L - -t flac $RESAMPLE$ -C 0 -
>
--------------------
> >
>
> So it fails
Not really.
The original problem was that 48 kHz MP3 were played too slowly,
probably as if they were 44.1 kHz. The problem does not occur with flac
48 or aac 48.
Then it was proposed to force LMS to transcode instead of Squeezelite.
Therefore, I used the Squeezelite settings to exclude MP3 from the
codecs it would use.
But this does not work either because LMS somehow can't find a
conversion rule. So with that Squeezelite setting, the MP3 files are not
played at all anymore.
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