ffmpeg vs. SoX for resampling Options

[jkauff]

I just found out that TAudioConverter, although it has SoX support, uses ffmpeg for bitdepth changes, resampling, and dither. I don't use those functions very much, but I'm curious if anyone has done a quality comparison between the two.

[[JAZ]]

As always, there is tradeoff between speed and accuracy, and in this field, there's even different techniques in play.


There are two good ways to resample:

- Using SINC ( sin(x)/x ) interpolation.
- Using a decimator/interpolator combination.

In both cases, a filter is needed to reconstruct the signal, and the quality of that filter reflects how fast and how clean it is in removing frequency imaging, without adding other unwanted distortions.


Both are relatively slow (They migh be "fast" for downsampling a single stereo sample from 96Khz to 44.1Khz, but now think about a realtime sampler like BASSMIDI, FluidSynth, or any other Soundfont player, where you not only play several streams, but also change their speed while playing).


So there have always existed other "resamplers" (not in the academic sense, but approximations) like:

- ZO (Zero order, or sample Hold). The fastest and worst of them all (See the graph of Secret Rabbit Code, ZOH).
- Linear interpolation. Quite an improvement over ZO, and still fast (This was being used in 16 and 32 channel trackers back in 1992-1996 with 386 and 486 PC's! See Secret Rabbit Code, Linear. One can apply a filter with this type of interpolation, like in Wavosaur Linear, but it sort of defeats the purpose)
- Cubic interpolation. Cubic interpolation, and other polynomial interpolators are an advancement over linear interpolation, approximating the signal using polinomials. This generally lowers aliasing notably, but they still miss a filter. (See Renoise cubic, or OpenMPT polyphase, which are, again, realtime multichannel trackers with realtime virtual effects).


Any SINC interpolator should give a clean signal, but it is at least 4 times slower than cubic interpolation. It can filter by itself, but to be of good quality, it gets slower (because it needs more taps).


So, while a product that is specific to change the sample rate of a signal should have a good resampler, one should not think that it is the only reasonable way to resample.

This post has been edited by [JAZ]: Feb 6 2013, 23:41

[halb27]

... Any SINC interpolator should give a clean signal ...

Can you tell me please which software uses SINC?

[bandpass]

Can you tell me please which software uses SINC?

First, go to http://src.infinitewave.ca/
For the top graph, select Test Result = Impulse, then press the > button to look at each resampler in turn:

1. If the impulse looks broadly like Audacity 2.0.3 (perhaps with shorter tails), it's a close approximation to a sinc. Could be obtained by windowing sinc, but could also be other techniques such as Parks-McClellan optimal FIR. Longer tails equate to steeper/deeper filters.
2. It it looks broadly like AbletonLive 8.2 then it's a sinc approximation that's been phase-adjusted to be causal (i.e. non-linear phase).
3. It it looks roughly like AbletonLive 7 or Waveburner 1.2, then it's low-order polynomial. Even this is a simple approximation to a sinc, but gives correspondingly poor results.
4. Others, like FL Studio 10 (6 Point Hermite), use other ways to approximate a sinc, but again with not so good results.

• Most are type #1.
• Soundhack looks like it should be type #1 but has implementation errors.
• SIR2 and a few others are type #1 but inverted (bug).
• Wavosaur 1.0.3.0 looks like a design error: polynomial followed by a sinc. (A multi-stage approach is perfectly valid; however, a good multi-stage design will still give a sinc impulse response.)

So the answer is they all use use sinc approximation, but the closer the approximation, the closer the result to the 'ideal'.