Yes, I am building the shell. Interestingly, it stores the
executable in a hidden directory <workspace>/fluidsynth/src/.libs
The console window only displays the shell and I see no error
messages. I'll need to figure out how to display internal printf
messages.
I am having problems building a debug variant. I tried adding -g
-o0 but get many build errors. This might be due to the way Eclipse
sets up autotools. Any quick hints?
--Brad
On 01/02/2016 08:08 AM, Element Green wrote:
Hello Brad,
It sounds like you really need to get debugging working. From what you
said about the "interp num" command, I assume you are building the
fluidsynth command shell. Do you get any relevant error messages on
the console (probably the same terminal you are running fluidsynth
from - but may be hidden if you are accessing the shell through TCP
for example). Make sure you are able to see console messages, since
there may be some useful error messages getting output.
When running the interp command it should
call fluid_synth_set_interp_method(). I would get debugging working
and set a breakpoint on that function and make sure it is doing all
the calls to fluid_channel_set_interp_method() for every channel.
Best regards,
Element
On Sat, Jan 2, 2016 at 4:57 AM, Marcus Weseloh <mar...@weseloh.cc
<mailto:mar...@weseloh.cc>> wrote:
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
<mailto: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 <mailto: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 <mailto: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 <mailto:brads...@fastmail.us>
>
>
>
>
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