Hi Brad,
Out of curiosity I checked that the interp setting actually had an
effect by adding printf statements to the relevant functions. The
fluid_rvoice_dsp_interpolate_linear function is called just fine if
you set "interp 1" in the FluidSynth shell. Seems like you've got a
problem with your devel setup...
Cheers,
Marcus
2016-01-02 0:33 GMT+01:00 Brad Stewart <brads...@fastmail.us>:
> So I recompiled the code and added a return at the beginning of the function
> fluid_rvoice_dsp_interpolate_linear (fluid_rvoice_dsp_t *voice).
>
> I also tried adding a for(;;) and a printf.
>
> I tried several interp num commands (with num from 0 to 7) but it seems it
> never gets to this function. All audio sounds as it should.
>
> Could you comment please?
>
> Thanks,
> Brad
>
> P.S. I imported the project into Eclipse Mars using import "existing tools
> as Autotools project". It compiled with some warnings. It seems to run OK.
> But I was unable to build a debug version of it so I could use gdb to trace
> through the code.
>
>
>
>
>
> On 12/31/2015 03:36 PM, Brad Stewart wrote:
>
> Could you confirm that "interp 1" command line invokes the Linear
> Interpolation function?
> Thanks,
> Brad
>
>
>
> On Thu, Dec 31, 2015, at 01:36 PM, Element Green wrote:
>
> Hello Brad,
>
> To my knowledge the audio sample data is used as is in FluidSynth. The
> sample rate of the data and the current pitch of a voice determine how it
> gets interpolated. So I believe your analysis is correct, that
> interpolating between two identical values will yield that value with the
> linear interpolation algorithm, regardless of where in between the two
> points the new interpolated point lies. The higher order interpolation
> algorithms will be affected by additional sample points though, so it would
> not be true in those cases unless additional consecutive points had the same
> value (for the number of points used by the interpolation algorithm).
>
> At this point I'm not sure what you are asking anymore though.
>
> Best regards,
>
> Element
>
>
> On Thu, Dec 31, 2015 at 12:38 PM, Brad Stewart <brads...@fastmail.us> wrote:
>
>
> Hi Mr Green,
>
> Yes, I am carrying the fractional part. In my case, it's 12 bits and I
> shift right by 4 to get the upper 8 bits to index to the coefficient table.
>
> But let me cite an example,
> Suppose I have x[i]=100 and x[i+1]=100 (which occurs often in a lower note
> frequency).
>
> Assume after looking up the coefficient table, you get coeffs[0] = .3 and
> coeffs[1] = .7
> So if you do the math,
> out=100*.3 + 100*.7 = 100. So no change.
> In my case, I can have 4 consecutive values of 100 (at note = 60) as shown
> in the previous images.
>
> So any of the coefficients in the table will return a value and 1-value, if
> the current sample and then next are the same, the output is always the
> original value. So in the general case where sampe[i]=sample[i+1],
>
> out = sample* frac + sample*(1-frac) = sample*(frac +1 -frac) = sample =
> out.
>
> This is why I was asking if you're adding to the data set by up-sampling,
> pre-filtering, or something else. This is hinted at in the paper by Olli
> Niemitalo.
> I suppose you that the interpolated value could be stuffed back into the
> original data (recursion) but I don't see that occurring in
> fluid_voice_dsp.c.
>
> Brad
>
>
>
>
> On Wed, Dec 30, 2015, at 07:21 PM, Element Green wrote:
>
> Hello Brad,
>
>
> On Wed, Dec 30, 2015 at 7:42 PM, Brad Stewart <brads...@fastmail.us> wrote:
>
> Hi,
>
> I'm developing an embedded project using some of the code in FluidSynth.
>
> The target is an ARM Cortex M4 and am using integer math. Most of the
> coefficients are in Q15 format.
>
> So far, I'm able to generate different notes with sound font files and use
> some of the CMSIS DSP libraries to generate high order IIR filters.
>
> But I am having no luck with any interpolation routines. I ran some tests
> using the command line version of FluidSynth on a laptop running Ubuntu 14.
>
> I created a sine wave sound font (using Swami) with a sample rate of 20050
> Hz, root note is 86. It's looping over several cycles. The image
> Midi_60_no_interpolation.jpg picture shows the result with no interpolation
> (I get the exact same results on my embedded platform). This is using
> "interp 0". (Note that the ringing is due to the sinc function in the Codec
> filter/decimator.)
>
> If I then set "interp 1", I get the image as shown in
> Midi_60_linear_interpolation.jpg which is a major improvement.
>
> I have reviewed the code inside fluid_voice_dsp.c and have set up my program
> to emulate the equations. I converted the floating point coefficients to
> Q15 format. It seems to do exactly what it's supposed to do. Yet I see no
> improvement what-so-ever except at higher note values.
>
> However...I don't see how linear interpolation can work. If you have
> straight lines (i.e. the "steps" in the first waveform), a linear
> interpolation won't change anything since the current sample and the next
> sample are the same over several samples.
>
> Are you up-sampling before you to do the linear interpolation? I didn't see
> any evidence in the code. Or do you simply take the loop sample as is and
> step through it using dsp_phase_index? I don't think the bi-quad post
> filter has an effect since it has a very high cut-off. Nor should the
> reverb or chorus blocks have any effect (since they're disabled anyway in my
> test).
>
> What am I missing?
>
> Thanks for your time and consideration. Any advice or insight is
> appreciated. I really like FluidSynth.
>
> Brad Stewart
> P.S. I also implemented the 4th order interpolation but it didn't change
> anything--I still get the "stepped" waveform. (At 50MHz clock, the M4 can
> process 128 samples in <1uS with sound fonts stored in internal Flash
> memory. Pretty impressive.)
>
>
>
>
> Let me make sure I understand your situation correctly.. So you have taken
> the interpolation code in fluid_rvoice_dsp.c and adapted it to integer math
> and there is something wrong with the ported code, since it is not
> interpolating as expected.
>
> It has been a while since I delved into the interpolation code and I did not
> originally write it (though I did muck about in that area a bit). If you
> look at the fluid_rvoice_dsp_config() routine you will see that it
> initializes the interpolation tables, including interp_coeff_linear, which
> essentially stores 2 tables of linearly increasing and decreasing values
> from 0 to 1. These values are then used depending on the current sample
> index fraction (think of the current sample pointer as being at a position
> which may be in between 2 consecutive samples - the fraction component). If
> the fractional position is closer to the first sample, then it will have a
> higher weight in the linear calculation of the sample value which gets
> synthesized. I hope that makes sense and perhaps you already knew this -
> seeing as you have done this port.
>
> I would make sure that your ported code retains this fractional sample
> pointer component and is using it properly and that your linear coefficients
> table is properly initialized. You should be able to see this stuff at work
> by single stepping the code and looking at the calculations to see if they
> are behaving as expected.
>
> Hope that helps. Maybe I did not get your scenario right though, if so,
> please explain in more detail.
>
> Best regards,
>
> Element Green
>
>
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>
> --
> Brad Stewart
> brads...@fastmail.us
>
>
>
>
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