Thanks Michael!
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What is the logging setting in LMS that can be used to see the actual
command-line used in transcoding? If I set
(player.source) - Player Source Audio Conversion Logging
to Debug, all I see (transcoding FLAC to MP3) in my server.log is:
Code:
[14-11-06
I tried setting all of the available logging settings to Debug and still
didn't get the actual command-line in the log. I could have sworn that
this was logged at one time in the past. Could it be added to 7.9?
JJZolx's
Works for me on 7.8
Code:
[14-11-06 11:26:12.5307] Slim::Player::TranscodingHelper::checkBin (241)
Found command: [lame] --silent -q $QUALITY$ -v $BITRATE$ $RESAMPLE$ --mp3input
$FILE$ -
[14-11-06 11:26:12.5308]
bpa wrote:
Works for me on 7.8
Thanks. I see it now. Confused because it appears in the log file well
before the end of the found conversion statements. I figured it had to
be at the end.
JJZolx's Profile:
[14-11-06 11:26:17.9409] Slim::Player::TranscodingHelper::tokenizeConvertCommand2 (553) Using
command for conversion: /usr/bin/lame --silent -q 9 -v -B 64 --mp3input - -
player.source=info should be good enough
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Michael
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mherger wrote:
[14-11-06 11:26:17.9409]
Slim::Player::TranscodingHelper::tokenizeConvertCommand2 (553) Using
command for conversion: /usr/bin/lame --silent -q 9 -v -B 64
--mp3input - -
player.source=info should be good enough
Yes, I just discovered that, too.
Need to clarify something:
Are non-native formats transcoded to and streamed as FLAC under the
default server configuration? That was my understanding, but is it
universal for all formats?
JJZolx's Profile:
Are non-native formats transcoded to and streamed as FLAC under the
default server configuration? That was my understanding, but is it
universal for all formats?
If no bitrate limiting is enabled and the player does support flac, then
yes, iirc LMS would transcode to flac. SB1 would receive
wait a second - are you telling me even when under advancedfile
formatsmp3 everything is set to native some transcoding takes place?
now i don't understand how any of this works or why we bother to set up
advancedfile formats. i thought the whole point is things go to the box
transparently, and
pablolie wrote:
wait a second - are you telling me even when under advancedfile
formatsmp3 everything is set to native some transcoding takes place?
now i don't understand how any of this works or why we bother to set up
advancedfile formats. i thought the whole point is things go to the box
mherger wrote:
Are non-native formats transcoded to and streamed as FLAC under the
default server configuration? That was my understanding, but is it
universal for all formats?
If no bitrate limiting is enabled and the player does support flac, then
yes, iirc LMS would transcode to
Sorry for asking a silly question but why on earth would anyone want to
transcode to lossy mp3?
Am I missing something?
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JohnB wrote:
Sorry for asking a silly question but why on earth would anyone want to
transcode to lossy mp3?
Am I missing something?
One is Streaming over the Internet to a smartphone via 3G using flac
music on your home LMS server.
Ah, sorry - streaming to smart phones never crossed my mind!
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JohnB wrote:
Sorry for asking a silly question but why on earth would anyone want to
transcode to lossy mp3?
Am I missing something?
And to players with bad wifi far away in ones house or in the garden ,
the radio won't mind sq loss is not big enough to perceived innthat one
:)
And
Nobody voted for this bug... was I wrong?
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Doesn't matter. Server development is dead.
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Fine. But at least people could comment here if I am right or wrong. :)
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I've actually found out why the mistake happened. It didn't happen. Back
then. When it was decided to move to this configuration for transcoding,
the -v switch simply indicated to use VBR encoding. In later versions of
LAME this changed and it became a substitute for -V 4.
Old LAME (3.89)
gorman wrote:
I've actually found out why the mistake happened. It didn't happen. Back
then. When it was decided to move to this configuration for transcoding,
the -v switch simply indicated to use VBR encoding. In later versions of
LAME this changed and it became a substitute for -V 4.
JJZolx wrote:
There was no change to this. The documentation was updated for 3.99.5,
but -v has always meant use VBR, which will be encoded at the default
VBR quality setting of 4.
The mistake was intentional. Boneheaded, but intentional.Doh! Well, that's
a real mistake then. In any
Posted on Bugzilla: http://bugs.slimdevices.com/show_bug.cgi?id=17953
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Really interesting comparison, thanks for sharing.
For me it's just a matter of a couple of my SBs wifi connections not
being up to FLAC bitrates. But they can handle V0 bitrates just fine.
In any case, I'll later report this as a bug. No way the default
settings do what they're supposed to.
gorman wrote:
By the way, would you be so kind as to point me to documentation for the
capabilities portion of convert.conf? The one where you setup stuff such
as $BITRATE.
The only documentation that I know of is in the convert.conf file
itself. The top portion of the file is a very large
Thanks, I knew about the built-in docs, as well as the custom file to be
created. But thanks for the help, it's really appreciated.
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In any case, one thing to give thought to would be to use Ogg Vorbis to
handle transcoding. Vorbis is natively gapless and sounds better than
LAME at low bitrates (at least according to general knowledge at
Hydrogenaudio).
Too bad that it's pretty impossible to handle transcoding to a different
Unless I'm mistaken, there is a problem with the standard settings that
LMS used to transcode when streaming to a bitrate limited player.
This is the string that's used, for instance, to transcode from FLAC:
[lame] --silent -q $QUALITY$ $RESAMPLE$ -v $BITRATE$
Now, the quality setting is the
It doesn't make any sense to me either.
The default Mp3 transcoding rules began using CBR, then it was 'switched
to using ABR' (http://forums.slimdevices.com/showth...MP3-CBR-vs-VBR),
and then 'changed to -V4 VBR with this bitrate cap'
If Cpu power and strict bitrate are not critical, one could also do:
Code:
flc mp3 * *
# FB:{BITRATE=--abr
%B}T:{START=--skip=%t}U:{END=--until=%v}D:{RESAMPLE=--resample %D}
[flac] -dcs $START$ $END$ -- $FILE$ | [lame] --silent -V $QUALITY$ $RESAMPLE$
- -
Yeah, that would work if you want to dictate the VBR level instead of
the bitrate. But I think that most people using bitrate limiting really
are using it to limit the streaming bitrate. Either to deal with a
limited bandwidth internet connection (streaming from their home LMS
server), or to
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