dsp plugin prob

timmay · 20th May 2004, 19:53

Hi - has anyone ever made a dsp plugin before?

if so, how does the short int *samples array hold the information?

because it seems as though it goes from -32768 to 32768. if this is the case how is the sound data stored in this array?

ta

timmay · 21st May 2004, 11:25

anyone know anything about dsp plugins or is this just a forum for media library's?

chris_d_smith · 22nd May 2004, 02:49

this is just guessing from looking at the example dsp plugins but, it looks like:

each item in the array holds the amplitude of the sound wave at a given point in time. The range used to store the amplitude depends on the bps parameter. bps is eather 16 (range -32768 to 32768) or 8 (-128 to 128). The number of cells in the array is the number of channels (nch) multiplied by the number of samples (numsamples).

That all seems pretty certain. Taking a bigger guess, i'd say that the channels are interleaved in the array so that, if you have 2 channels then every even index in the array holds a sample for channel 1 (eg: left) and every odd index in the array holds a sample for channel 2.

The alternative would be that all samples for channel 1 come first and then all samples for channel 2 however that would seem to be much more difficult to construct as the input plugin reads the file.

timmay · 22nd May 2004, 06:53

cheers chris - cleared a few things up

20th May 2004, 19:53	#1
timmay Junior Member Join Date: May 2004 Posts: 5	dsp plugin prob Hi - has anyone ever made a dsp plugin before? if so, how does the short int *samples array hold the information? because it seems as though it goes from -32768 to 32768. if this is the case how is the sound data stored in this array? ta

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21st May 2004, 11:25	#2
timmay Junior Member Join Date: May 2004 Posts: 5	anyone know anything about dsp plugins or is this just a forum for media library's?

22nd May 2004, 02:49	#3
chris_d_smith Senior Member Join Date: Jan 2004 Posts: 161	this is just guessing from looking at the example dsp plugins but, it looks like: each item in the array holds the amplitude of the sound wave at a given point in time. The range used to store the amplitude depends on the bps parameter. bps is eather 16 (range -32768 to 32768) or 8 (-128 to 128). The number of cells in the array is the number of channels (nch) multiplied by the number of samples (numsamples). That all seems pretty certain. Taking a bigger guess, i'd say that the channels are interleaved in the array so that, if you have 2 channels then every even index in the array holds a sample for channel 1 (eg: left) and every odd index in the array holds a sample for channel 2. The alternative would be that all samples for channel 1 come first and then all samples for channel 2 however that would seem to be much more difficult to construct as the input plugin reads the file.

22nd May 2004, 06:53	#4
timmay Junior Member Join Date: May 2004 Posts: 5	cheers chris - cleared a few things up