this manual lists all the available switches under the -ota( ) section of BeSweet v1.5.
for more info, read the BeSweet FAQ !

pay attention : switches are case-sensitive.

Gain Switches

-g	apply an absolute gain level on decoded stream, but before encoding ("pre-gain")
	example : "-g 6db" will apply 6db of gain
	example : "-g max" will apply the highest gain level that still does not distort the original signal
	example : "-g peak" will apply the gain level written in the wave's peak chunk
	* if inputfile is not wave, or it doesn't have a peak chunk, "-g peak" will act like "-g max"
-norm	apply relative normlization on decoded stream, but before encoding ("pre-gain normalization")
	example : "-norm 0.95" will apply 95% of the highest gain level that still does not distort the original signal
	note that "-g max" & "-norm xx" are implementing two-pass of processing : one for analyzing the input file and determing the maxgain value, and the second for the actual encoding
-G	apply relative normlization on the encoded file ("post-gain normalization")
	example : "-G 0.97" will apply 97% of the highest gain level that still does not distort the original signal
	note that unlike "-norm", the processes of finding maxgain and stream encoding are done in the same single pass
	* if output file is not mp2/mp3/ogg, "-G xx" will act like "-norm xx"
-hybridgain	apply absolute pre-gain and relative post-gain normalization ("hybridgain")
	if inputfile is a stereo ac3, disable dialog normlization and pregain
	if inputfile is six channels ac3, enable dialog normilzation and apply pregain in accordance to dynamic range compression level :
	no compression : 7db. light compression : 8db. normal compression : 10db. heavy compression : 14db. inverse compression : 5db.
	if outputfile is mp3, set lame's scale switch to 1.
	if outputfile is mp2/mp3/ogg, apply postgain of 97% on the processed file
	* if output file is not mp2/mp3/ogg, "-hybridgain" will act like "-g max"

Rate Switches

-fs	force a sample-rate value
	this bypass BeSweet's engine for determing the sample-rate of input files, and sets is manually
	usefull sometime when processing an headerless pcm stream
-r	a frame rate convertor
	example : "-r 23976 25000" will convert a 23.976fps stream (NTSC) to a 25.000fps stream (PAL)

Delay Switches

-d	apply delay correction to file
	example : "-d 10" will add 10miliseconds of silence to the beginning of the encoded file
	example : "-d -20' will skip the first 20milliseconds of the original file
-e	append silence to file
	example : "-e 8" will add 8miliseconds of silence to the end of encoded file
-sc	this sets the minimum resolution of samples for delay assertions
	used in AacMachine to add delay of size of full aac-frames

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