Normal
Ok. So a cd sample is 44100hz and 16 bit. The sample polarities are between -1 to 1. I think it just polls the mic at these intervals to get the air pressure in analogue before changing it to digital though im completely ignorant of that process. The way your describing it in analogue form -1v to 1v seems to be whats happening and if so that would make the conversion relatively easy as its a unit to unit match...(or is it -5 to 5v)? Either way thats a linear exchange which if true is very helpful. In digital if i have multiple samples if add them together the resulting waveform can be above 1 and below minus1 but you can normalise this sample(ie go through every sample and find the most extreme peak in either direction) and then just multiply every sample by this value to squeeze it back to 1>-1. If you were to play a sample that goes out of these boundaries youd hear a horrible distortion where it goes above them as it just turns every sample that goes above to 1 and every sample that goes below to -1.16bit sample is 65536 possible combinations. 0 is neutral32767 is 1-32768 is minus 1And all the inbetween states away from zero in either direction are the steps(converted into floating point numbers)65536 combinations can be represented as either 0->65535 or -32768->32767 depending on whats required from the user for a 16bit value.So as far as the computer is concerned it always fires samples to the analogue hardware in this fashion and its up to the amplifier to magnify the voltage for volume. (I think you said +-40v). Im trying to do my low pass filter(and any other filter) before sending it to the player(which i suppose you could connect a mixer to this output to get the correct results) but wheres the fun in that?
Ok. So a cd sample is 44100hz and 16 bit. The sample polarities are between -1 to 1. I think it just polls the mic at these intervals to get the air pressure in analogue before changing it to digital though im completely ignorant of that process. The way your describing it in analogue form -1v to 1v seems to be whats happening and if so that would make the conversion relatively easy as its a unit to unit match...(or is it -5 to 5v)? Either way thats a linear exchange which if true is very helpful. In digital if i have multiple samples if add them together the resulting waveform can be above 1 and below minus1 but you can normalise this sample(ie go through every sample and find the most extreme peak in either direction) and then just multiply every sample by this value to squeeze it back to 1>-1. If you were to play a sample that goes out of these boundaries youd hear a horrible distortion where it goes above them as it just turns every sample that goes above to 1 and every sample that goes below to -1.
16bit sample is 65536 possible combinations.
0 is neutral
32767 is 1
-32768 is minus 1
And all the inbetween states away from zero in either direction are the steps(converted into floating point numbers)
65536 combinations can be represented as either 0->65535 or -32768->32767 depending on whats required from the user for a 16bit value.
So as far as the computer is concerned it always fires samples to the analogue hardware in this fashion and its up to the amplifier to magnify the voltage for volume. (I think you said +-40v). Im trying to do my low pass filter(and any other filter) before sending it to the player(which i suppose you could connect a mixer to this output to get the correct results) but wheres the fun in that?
