Suren

How can I convert a 96kHz sampling rate audio file to a 48kHz without interpolation? The end result would be twice as long, and this is what I am trying to achieve.

How can I convert a 96kHz sampling rate audio file to a 48kHz without interpolation? The end result would be twice as long, and this is what

converting-sampling-rate-by-stretching-audio

samplerate conversion is simpler than you think: you´d go over the smallest common multiple (rate)

but i am afraid that when you downsample something, you still have to lowpassfilter the input to match the content to the new rate´s frequency spectrum or you get aliasing.


samplerate conversion is simpler than you think: you´d go over the smallest common multiple (rate)

but i am afraid that when you downsample something, you still have to lowpassfilter the input to match the content to the new rate´s frequency spectrum or you get aliasing.


Oh, I mean how to do this on the actual audio file. For instance my project file sample rate is 96kHz.If I stretch the audio file twice as long without pitch shifting, the new audio clip will be an octave lower. However, the sampling rate will still be 96kHz, meaning that my DAW has done some interpolation to fill in those samples. I don't want this to happen, I simply want to stretch out the file with no interpolation. I use Ableton Live 10.


for up:

counter~
+~ 1
/~ 2
index~

for down:

counter~
*~ 2
-~ 1
index~

?


...or for 96 to 44.1, just for reference:


...or for 96 to 44.1, just for reference:

1.)
lcm(44100, 96000) == 14112000

2.)
14112000/96000 == 147
14112000/44100 == 320

3.)
counter~  
+~ 146
/~ 147
index~
record~

4.)
counter~
 /~ 320
-~ 319
index~



Converting sampling rate by stretching audio