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Low-bitrate RM to MP3 128—how to minimise size/waste but keep quality?, Was: help making decision/advice about converting mass amount of files
Ahmose
post Jan 7 2012, 23:50
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Hello,
i have about 30000 real media (*.rm) files
with different bit rates.. and i have to convert them to mp3
i decided that the max bitrate will be 128
but, as this files is already rm files with low bitrates
i think i will lose a lot of space and bandwidth by resampling the files to a higher sample-rate
so whats the solution to convert this files with the same quality, and lowest possible file sizes.

this is the commands i use to convert
rm to wav with mplayer

CODE
$config[path_mplayer] rmfile -ao pcm:file=wavfile


then
wav to mp3 with lame

CODE
$config[path_lame] -b 128 --resample 44.1 --id3v2-only --tt \"$title\" --ta \"$artist\" -h -p wavfile mp3file


i use php exec to process this commands.

This post has been edited by Ahmose: Jan 8 2012, 00:04
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db1989
post Jan 8 2012, 02:16
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QUOTE (Ahmose @ Jan 7 2012, 22:50) *
i think i will lose a lot of space and bandwidth by resampling the files to a higher sample-rate
Why are you doing it, then?
QUOTE
$config[path_lame] -b 128 --resample 44.1 --id3v2-only --tt \"$title\" --ta \"$artist\" -h -p wavfile mp3file
I assume mplayer will output the WAV files at whichever sampling rate is appropriate, so I don’t see why you’re forcing LAME to resample them—especially when you say you want to get the maximum quality per unit file-size, something that is not promoted by unnecessarily resampling.

As for the bitrate, I can’t speak for everyone else, but I suspect there will be no easy answer. Except, perhaps, to try and evaluate some different settings on a representative sample of tracks, perhaps with the aid of ABX tests if you want to find a level that achieves transparency. Even then, no one can guarantee that any one setting will be satisfactory for all of them; that’s why lossy-to-lossy transcoding is problematic and is advised against whenever it’s avoidable.
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Ahmose
post Jan 8 2012, 03:08
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QUOTE (db1989 @ Jan 8 2012, 03:16) *
QUOTE (Ahmose @ Jan 7 2012, 22:50) *
i think i will lose a lot of space and bandwidth by resampling the files to a higher sample-rate
Why are you doing it, then?
QUOTE
$config[path_lame] -b 128 --resample 44.1 --id3v2-only --tt \"$title\" --ta \"$artist\" -h -p wavfile mp3file
I assume mplayer will output the WAV files at whichever sampling rate is appropriate, so I don’t see why you’re forcing LAME to resample them—especially when you say you want to get the maximum quality per unit file-size, something that is not promoted by unnecessarily resampling.

As for the bitrate, I can’t speak for everyone else, but I suspect there will be no easy answer. Except, perhaps, to try and evaluate some different settings on a representative sample of tracks, perhaps with the aid of ABX tests if you want to find a level that achieves transparency. Even then, no one can guarantee that any one setting will be satisfactory for all of them; that’s why lossy-to-lossy transcoding is problematic and is advised against whenever it’s avoidable.


i used the --resample 44.1 because i converted file once and the sound was faster than the original file..
but that was converting mp3 to mp3 sorry for that..
now what if i used the default settings
QUOTE
$config[path_lame] --id3v2-only --tt \"$title\" --ta \"$artist\" -h -p wavfile mp3file

what would be the results ?!
i don't know how lame choose the bitrate and sampling rate.
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saratoga
post Jan 8 2012, 04:37
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QUOTE (Ahmose @ Jan 7 2012, 21:08) *
now what if i used the default settings
QUOTE
$config[path_lame] --id3v2-only --tt \"$title\" --ta \"$artist\" -h -p wavfile mp3file

what would be the results ?!


I believe its 128k CBR. You can look in your specific build using 'lame --help'

QUOTE (Ahmose @ Jan 7 2012, 21:08) *
i don't know how lame choose the bitrate and sampling rate.


Sampling rate will be whatever your source audio is.
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