Wiki > Main > PluginSplitEdit (compare)
Difference: PluginSplitEdit (r4 vs. r3)
When you record a mp3 file with your recorder you usually start the recording a little bit early and stop it a little bit late to ensure all the desired sound is recorded. This results in recordings that contain undesired parts. Unfortunately these parts can not be deleted easily because they are stored in the same file with the wanted parts. The purpose of the split editor is to split a mp3 file (the input file) at a point in time (split point). Two new files can be generated from the input file. The first file contains the first part which is before the split point and the second file contains the part after the split point. Once this process has been successful the original file can be deleted or kept as a backup.
The whole process of splitting a mp3 file consists of two steps:
When the device plays the song just hit the pause button, when the playback has roughly reached the split point. This needn't be very precise as the split point can be fine tuned later.
Open the plugin "splitedit". A screen similar to this will appear. Lets have a closer look at the items of the screen.
The waveform displays the volume of the song over time. It will build up as the song plays and helps you to visually identify the point in time where you want to split the song.
The split point indicator is a vertical line with a small triangle at the top end. It is the most important control element of the split editor. You can move it with the left / right buttons. Later, when you have finetuned the split point, the song will be split at this position.
At the top of the window the point a time value is displayed. This is the point in time within the song at which the split point indicator is positioned.
Another vertical bar represents the position locator. It moves along as the song plays. In contrast to the split point indicator it doesn't have any triangles.
The time bar displays which part of the song is displayed as waveform. The entire length of the time bar represents the song length. The length of the solid part of the timebar represents the position and length of the displayed part of the song.
Directly above the button F3 the scale mode is displayed. The waveform can be scaled either logarithmic or linear. In logarithmic scale mode the letters "db" are displayed, in linear mode "%". You can use F3 to switch between these modes. Linear mode usually gives better optical hints with commercially recorded music. For soft recordings, especially with human speech, the logarighmic scale often is preferable.
Directly above the F2 button the loop mode icon is displayed. There are 4 different loop modes. Pressing F2 changes to the next loop mode.
The icon directly above the F1 button indicates its function to execute the split. When you have finished to finetune the split position, you can open the save dialog with F1.
In the save dialog you can specify which of the files you want to save and with which names they are saved. When you have done these selections you can use "Save" and the files will be written to disk. Note that files can't be overwritten. Thus you must choose filenames that don't exist yet. If you are not sure wether the file already exists you may simply try to save it. If another file with this name exists the dialog will return and you can choose another filename.
The waveform is not calculated directly from the mp3 files. That would be a task which is far beyond the capabilities of the processor used in the jukebox. But the Hardware can provide "quasi peak" information that can be read and used in realtime. There is no other way to extract volume information from mp3 files at all. This information is available only in realtime while a song is playing. Thus the only possibility to display volumes values that haven't been played yet is to rely information that was stored before. That is the reason why neither the volume curve of the whole song nor of a part of it can be displayed before it has been played. Unfortunately the peak values provided by the hardware are not very precise. Very often a peak remains undetected. To overcome this weakness the waveform is updated each time it is played over again. If new peaks are detected that were undetected at first try, they now can can be visualized. Thus the waveform becomes more and more precise, the more often it is played.
Zooming and scrolling invalidates the the waveform. All values must be redetected. Especially when you zoomed deeply into the waveform little gaps can occur within the waveform. These gaps don't resemble volume information. At very high zoom levels the locator has to move so fast that there is no time to read out a volume information for each pixel column position. Usually these gaps vanish when the playback loops over a few times.
The values in the waveform are scaled according to the settings of the peak meter. These can be altered in the menu "General Settings -> Display -> Peak Meter". If you set extreme minimum / maximum values the waveform might be cut off. I recommend to use a minimum setting of -60 db and a maximum setting of 0 db. With these settings you should be able to get useful waveforms for very soft sounds in logarithmic mode (db). When you use the editor on loud sounds (e.g. commercial rock / pop productions) you might want to switch over to linear scale as the logarithmic scale compresses loud signals and makes it more difficult to identify characteristic shapes. Note that you can always toggle the scale with F3.
Choose "save" to save the files.
CategoryPlugin: An MP3 file split editor [Recorder , Ondio, iRiver]
r6 - 01 Oct 2005 - 18:07:47 - DaveChapmanRevision r4 - 12 Feb 2005 - 19:38 - JonasHaeggqvist
Revision r3 - 17 Aug 2004 - 21:00 - LinusNielsenFeltzing
Copyright © by the contributing authors.