I don't think a linear interpolator would work here. There's quite a bit of attenuation in the 0-Fs/4 band (~8dB by Fs/8) and only around -28dB on the aliases. -28dB might be enough for the aliases, but I don't think it's flat enough in the Fs/4 band to be useful.
https://ccrma.stanford.edu/~jos/pasp/Linear_Interpolation_Frequency_Response.html On Sat, Sep 19, 2015 at 4:03 PM, Stuart Longland <[email protected] > wrote: > On 20/09/15 03:43, Tomas Härdin wrote: > > Couldn't you do further low-pass filtering in software, then decimate to > > 8 kHz? (caveat: I haven't checked if the code actually does this) > > That's exactly what it does. It samples at 16kHz from the ADC, filters > it to produce an 8kHz stream. > > Actually a thought just occurred, could we just do nearest-neighbour > upsampling to get back to 16kHz? > > I'll admit I havent looked at what algorithm is used to do the > resampling, however this is not a device that will be used by > audiophiles (it doesn't do 192kHz 32-bit FLAC for starters) so linear > interpolation or nearest-neighbour would probably work for getting it > back to 16kHz. > -- > Stuart Longland (aka Redhatter, VK4MSL) > > I haven't lost my mind... > ...it's backed up on a tape somewhere. > > > ------------------------------------------------------------------------------ > _______________________________________________ > Freetel-codec2 mailing list > [email protected] > https://lists.sourceforge.net/lists/listinfo/freetel-codec2 >
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