> someone please correct me if im massively confused here... the first
> thing you do to the input when encoding is convert it to frequency
> spectra, right? so why not just treat 22khz input exactly the same
> as 44khz, except half all computed frequencies and fft windows? why
> ever convert b
>> Does anyone have a GOOD quality Sample Rate conversion algorithm ?
>
>See any elementary discrete time audio processing book and look under 'integer
>ratio rate conversion'. It's mostly an exercise in upsampling, lowpassing and
>downsampling. The computational complexity is proportional to th
> For example, going from 22050 to 44100, just make every other sample a zero,
> then apply a lowpass (however you like) with a cutoff just below 11.025 kHz.
> Now downsampling is needed in this case (well, the downsample factor is 1).
someone please correct me if im massively confused here...
> There is a much better way to do SRC using polyphase filtering. It also allows
> SRC between ANY two sample rates.
This is a generalization of what I described done in the time domain (if one
understands the MPEG audio polyphase subbander, this should be easy to work
through). It introduces j
On Thu, 30 Sep 1999, Mark Gilbert wrote:
> Does anyone have a GOOD quality Sample Rate conversion algorithm ?
The resample code in the *current* version of sox is pretty good. Make
sure that you get the current version, though, because previous versions
were not good at all
Mike
--
Mike Olipha
Monty wrote:
>
> > Does anyone have a GOOD quality Sample Rate conversion algorithm ?
>
> See any elementary discrete time audio processing book and look under 'integer
> ratio rate conversion'. It's mostly an exercise in upsampling, lowpassing and
> downsampling. The computational complexity
> For example, going from 22050 to 44100, just make every other sample a zero,
Sorry, make that 'double the number of samples, making every other sample a
zero'. Eg, 'A,B,C,D' becomes 'A,0,B,0,C,0,D,0'...
Monty
--
MP3 ENCODER mailing list ( http://geek.rcc.se/mp3encoder/ )
> Does anyone have a GOOD quality Sample Rate conversion algorithm ?
See any elementary discrete time audio processing book and look under 'integer
ratio rate conversion'. It's mostly an exercise in upsampling, lowpassing and
downsampling. The computational complexity is proportional to the pro
Does anyone have a GOOD quality Sample Rate conversion algorithm ?
Please mail me if you have,
Thanks
Mark Gilbert
>>
>> > - resampling (sox)
>>
>> if this would just be added as a pipeline stage, ie part of reading the
>> input file, i dont see a lot of point in integrating it. on the other
