I have worked out a patch to convert my array from 22050HZ to 44100Hz by copying the value to the next index :
new_array[2x] =new_array[2x+1] = old_array[x]
The new array displays correct datas.

Unfortunately I have been trying to use this converter patch with libpd and some audio I am recording with an ipad. It failed to work. Somehow the iteration through the audio array jumps some values and I just couldn't figure it out why. I'll try to do the conversion using some Core Audio ressources.

That's my intention. Thanks.

Thanks for your reply. This is already a very useful for me. I'll give it a go.

How do you add an anti-aliasing filter?

Thanks for the answers. I'll try to repeat each sample twice to see how it sounds. Maybe I'll have to do some interpolation. I'll keep the thread informed.

Does anyone know if it is possible to open a 22050Hz audio file with [soundfiler] and convert 44100Hz ? Regardless of the current sample rate.

Thanks Gilberto. Following the same technic than yours, it is working. Great!

So I am reading my array until there is a non-zero value. It is working although silence is not necessarily a pure 0 value. I had to set a a low value threshold (positive and negative) . So far it is working ;^)

Thanks for the suggestion. Reading the table of course! I'll try that right now.
I had a solution with Bonk~but I found that the result was not precise enough.

Hi,

I have an audio file that I read with [tabread~]). The audio starts with some silence and I am trying to find the first non silent sample index.

I have tried to use [bonk~], [fiddle~] or [sigmund~] but the results were approximate and not constants.

Does anyone have a solution for this? Thanks.

Pascal