Les Cargill wrote:
William Sommerwerck wrote:
You need two all-pass filters. One produces a phase shift of phi. The other
producers phi+90 or phi-90.
It is impossible to produce a fixed 90 degree phase shift over a wide
frequency range and have flat response at the same time. You have to create
the 90-degree shift as the difference between two filters' phase responses.
Trust me.
I have some 'C' source which produces a Hilbert transform of an
audio file ( using libsndfile and FFTW plus some code of my own,
er code I pretty much swiped off the Internet 
.
When I run this twice*, I get perfect** cancellation
What am I missing?
You can't do a Hilbert transform directly in the analogue domain. You can
only use some functions that approximate it, made with Ls and Cs.
--scott
--
"C'est un Nagra. C'est suisse, et tres, tres precis."