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Bug 556756 - audioresample does not low-pass filter ("antialias") when down-sampling
audioresample does not low-pass filter ("antialias") when down-sampling
Status: RESOLVED DUPLICATE of bug 346218
Product: GStreamer
Classification: Platform
Component: gst-plugins-base
git master
Other Linux
: Normal normal
: git master
Assigned To: Sebastian Dröge (slomo)
GStreamer Maintainers
Depends on:
Blocks:
 
 
Reported: 2008-10-17 17:03 UTC by Kipp
Modified: 2008-10-27 16:20 UTC
See Also:
GNOME target: ---
GNOME version: ---



Description Kipp 2008-10-17 17:03:38 UTC
When down-sampling an audio time series to a lower sample rate, one must first remove frequency components above the Nyquist frequency of the new time series.  This is accomplished by low-pass filtering the data before resampling, a process sometimes referred to as "anti-aliasing".  Failure to anti-alias the time series results in high-frequency components being "aliased" down to lower frequencies, corrupting the audio.

The audioresample element does not anti-alias the time series when down-sampling.  In a pipeline, one can work-around this bug by preceding the audioresample element with a low-pass filter element like audiowsinclimit, but audioresample, like any element, should Just Work (tm).
Comment 1 Sebastian Dröge (slomo) 2008-10-19 12:03:00 UTC
Scary, you're right, audioresample really doesn't lowpass when downsampling. For now you could use speexresample (from gst-plugins-bad) or ffaudioresample (from gst-ffmpeg), those two resamplers lowpass the input.
Comment 2 Sebastian Dröge (slomo) 2008-10-27 16:20:34 UTC
Thanks for the bug report. This particular bug has already been reported into our bug tracking system, but please feel free to report any further bugs you find.


*** This bug has been marked as a duplicate of 346218 ***