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 Options

[Ahmose]

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

[db1989]

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?

$config[path_lame] -b 128 --resample 44.1 --id3v2-only --tt \"$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.

[Porcus]

i have about 30000 real media (*.rm) files
with different bit rates.. and i have to convert them to mp3

I have very limited knowledge of the Real formats (avoided them as much as possible ...), but depending on the content, you might not have to transcode. There isn't mp3 content in Real media, but it might be AAC. Can you use mp4/m4a instead of mp3? Maybe you can use ffmpeg to merely reencapsulate without transcoding.

[Ahmose]

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?

$config[path_lame] -b 128 --resample 44.1 --id3v2-only --tt \"$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

$config[path_lame] --id3v2-only --tt \"$title\" --ta \"$artist\" -h -p wavfile mp3file

what would be the results ?!
i don't know how lame choose the bitrate and sampling rate.

[saratoga]

now what if i used the default settings

$config[path_lame] --id3v2-only --tt \"$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'

i don't know how lame choose the bitrate and sampling rate.

Sampling rate will be whatever your source audio is.

[pdq]

i don't know how lame choose the bitrate and sampling rate.

Sampling rate will be whatever your source audio is.

That's not necessarily true. Under some circumstances lame will downsample the audio, but at cbr 128 and 44.1 kHz, the sample rate will not be changed.

[Ahmose]

now what if i used the default settings

$config[path_lame] --id3v2-only --tt \"$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'

i don't know how lame choose the bitrate and sampling rate.

Sampling rate will be whatever your source audio is.

its 128k as you said, and i think this is the max bitrate, because i tried to convert some of the files to test with the default settings
results are lower than 128k and different from file to another..
i think i will convert with this settings..
i will update if there are any issues came up
thanks every body for your help