Hi Anders,

There is now an undersample opcode to go along with the oversample. I am not sure what you mean about 2., it does allow
a-rate opcodes.

best
========================
Prof. Victor Lazzarini
Maynooth University
Ireland

> On 20 Mar 2024, at 08:55, Anders Genell  wrote:
> 
> *Warning*
> This email originated from outside of Maynooth University's Mail System. Do not reply, click links or open attachments unless you recognise the sender and know the content is safe.
> This is mainly for Victor Lazzarini, 
> 
> I saw you added code in a PR to implement local sample rate in UDOs which I think is absolutely fantastic.
> It seems there are two limitations that I really hope could be addressed in the future:
> 1. The code allows for upsampling but not for downsampling
> 2. The code does not allow for a-rate opcodes
> 
> The use case I am hoping will be possible in the future is when having a reasonably high global sr (say 48kHz or even 96kHz) where I would like to apply relatively narrow band filters with center frequencies at say 20Hz, 25Hz 32Hz or the like. Such filters tend to be somewhat unstable at such high SR if they also need to be reasonably steep, and by downsampling before filtering a better result is often achieved. This is commonly done in signal processing for analysis in e.g. Matlab or GNU octave.
> The idea is e.g. to be able to manipulate individual bands, like in the pvs group of opcodes, and combining the individual bands back to one signal afterwords, so just docking global reduction of SR would not be a viable route, unless of course starting several instances of csound and using channel opcodes to combine everything in the end.
> 
> Would this be a possibility in some future iteration of the newly implemented code?
> 
> Regards,
> Anders

Make sure you have libsamplerate for the best quality conversion. There’s a simple fallback converter, but Secret Rabbit Code is better.

The limitations are:

- can’t setksmps alongside it
- can’t use a-rate or k-rate arrays (that may be possibly changed in the future).
- if you use global arrays or bus channels it won’t work either. All access has to be
through xin and xout, because that’s where the conversion happens.

Testing is helpful, thanks for that.
========================
Prof. Victor Lazzarini
Maynooth University
Ireland

> On 20 Mar 2024, at 11:27, Anders Genell  wrote:
> 
> Ah, brilliant!
> I just saw the flurry of emails connected to the code submissions and it said something about not allowing a-rate, but that was likely a misinterpretation of mine.
> 
> I will try to build and test to see if my scenario seems to work as I hope.
> 
> Thank you!
> 
> Regards,
> Anders
> 
> 
> On Wed, Mar 20, 2024 at 10:51 AM Victor Lazzarini  wrote:
> Hi Anders,
> 
> There is now an undersample opcode to go along with the oversample. I am not sure what you mean about 2., it does allow
> a-rate opcodes.
> 
> best
> ========================
> Prof. Victor Lazzarini
> Maynooth University
> Ireland
> 
> > On 20 Mar 2024, at 08:55, Anders Genell  wrote:
> > 
> > *Warning*
> > This email originated from outside of Maynooth University's Mail System. Do not reply, click links or open attachments unless you recognise the sender and know the content is safe.
> > This is mainly for Victor Lazzarini, 
> > 
> > I saw you added code in a PR to implement local sample rate in UDOs which I think is absolutely fantastic.
> > It seems there are two limitations that I really hope could be addressed in the future:
> > 1. The code allows for upsampling but not for downsampling
> > 2. The code does not allow for a-rate opcodes
> > 
> > The use case I am hoping will be possible in the future is when having a reasonably high global sr (say 48kHz or even 96kHz) where I would like to apply relatively narrow band filters with center frequencies at say 20Hz, 25Hz 32Hz or the like. Such filters tend to be somewhat unstable at such high SR if they also need to be reasonably steep, and by downsampling before filtering a better result is often achieved. This is commonly done in signal processing for analysis in e.g. Matlab or GNU octave.
> > The idea is e.g. to be able to manipulate individual bands, like in the pvs group of opcodes, and combining the individual bands back to one signal afterwords, so just docking global reduction of SR would not be a viable route, unless of course starting several instances of csound and using channel opcodes to combine everything in the end.
> > 
> > Would this be a possibility in some future iteration of the newly implemented code?
> > 
> > Regards,
> > Anders



WSL should work. I use the ubuntu image to test linux builds. You need to either install libintl or else disable it's usage via USE_GETTEXT=0.

It shouldn't really be failing as we have code to test if it's found but it seems to be a bug that was introduced in the last few months. I'll try and take a further look.

Stephen

---

From: Csound-developers <CSOUND-DEV@LISTSERV.HEANET.IE> on behalf of Victor Lazzarini <Victor.Lazzarini@MU.IE>
Sent: Wednesday 20 March 2024 17:08
To: CSOUND-DEV@LISTSERV.HEANET.IE <CSOUND-DEV@LISTSERV.HEANET.IE>
Subject: Re: [Csnd-dev] [EXTERNAL] [Csnd-dev] UDO local sr

 

I really don't know, this branch was started off develop last Saturday. There has been a lot of change in the build system and it might have broken people's builds. OTOH the CI builds are passing.

Stephen, do you have a suggestion?

Prof. Victor Lazzarini

Maynooth University

Ireland

On 20 Mar 2024, at 16:04, Anders Genell <anders.genell@gmail.com> wrote:



*localsr* branch, that is

On Wed, Mar 20, 2024 at 5:03 PM Anders Genell <anders.genell@gmail.com> wrote:

Just checked out locals branch and got a cmake error:

"CMake Error at CMakeLists.txt:1093 (add_library):

  Target "csound64" links to target "Intl::Intl" but the target was not

  found.  Perhaps a find_package() call is missing for an IMPORTED target, or

  an ALIAS target is missing?"

I suppose I'm doing something wrong, the first of which may very well be the fact that I am building on the Windows Subsystem for Linux - that's just because I'm at work doing ordinary work stuff on my work computer - I can build on an ordinary linux system later...

Regards,

Anders

On Wed, Mar 20, 2024 at 2:12 PM Anders Genell <anders.genell@gmail.com> wrote:

Ah, it was just arrays that were limited. Great!

I'll be sure to install libsamplerate and will see what I can do with regards to testing

Readers,

ANders

On Wed, Mar 20, 2024 at 2:10 PM Victor Lazzarini <Victor.Lazzarini@mu.ie> wrote:

Make sure you have libsamplerate for the best quality conversion. There’s a simple fallback converter, but Secret Rabbit Code is better.

The limitations are:

- can’t setksmps alongside it
- can’t use a-rate or k-rate arrays (that may be possibly changed in the future).
- if you use global arrays or bus channels it won’t work either. All access has to be
through xin and xout, because that’s where the conversion happens.

Testing is helpful, thanks for that.
========================
Prof. Victor Lazzarini
Maynooth University
Ireland

> On 20 Mar 2024, at 11:27, Anders Genell <anders.genell@GMAIL.COM> wrote:
>
> Ah, brilliant!
> I just saw the flurry of emails connected to the code submissions and it said something about not allowing a-rate, but that was likely a misinterpretation of mine.
>
> I will try to build and test to see if my scenario seems to work as I hope.
>
> Thank you!
>
> Regards,
> Anders
>
>
> On Wed, Mar 20, 2024 at 10:51 AM Victor Lazzarini <Victor.Lazzarini@mu.ie> wrote:
> Hi Anders,
>
> There is now an undersample opcode to go along with the oversample. I am not sure what you mean about 2., it does allow
> a-rate opcodes.
>
> best
> ========================
> Prof. Victor Lazzarini
> Maynooth University
> Ireland
>
> > On 20 Mar 2024, at 08:55, Anders Genell <anders.genell@GMAIL.COM> wrote:
> >
> > *Warning*
> > This email originated from outside of Maynooth University's Mail System. Do not reply, click links or open attachments unless you recognise the sender and know the content is safe.
> > This is mainly for Victor Lazzarini,
> >
> > I saw you added code in a PR to implement local sample rate in UDOs which I think is absolutely fantastic.
> > It seems there are two limitations that I really hope could be addressed in the future:
> > 1. The code allows for upsampling but not for downsampling
> > 2. The code does not allow for a-rate opcodes
> >
> > The use case I am hoping will be possible in the future is when having a reasonably high global sr (say 48kHz or even 96kHz) where I would like to apply relatively narrow band filters with center frequencies at say 20Hz, 25Hz 32Hz or the like. Such filters tend to be somewhat unstable at such high SR if they also need to be reasonably steep, and by downsampling before filtering a better result is often achieved. This is commonly done in signal processing for analysis in e.g. Matlab or GNU octave.
> > The idea is e.g. to be able to manipulate individual bands, like in the pvs group of opcodes, and combining the individual bands back to one signal afterwords, so just docking global reduction of SR would not be a viable route, unless of course starting several instances of csound and using channel opcodes to combine everything in the end.
> >
> > Would this be a possibility in some future iteration of the newly implemented code?
> >
> > Regards,
> > Anders

ah, thanks for this. I hadn’t tried with more than one instance and you caught a bug in
the deletion of the converters.

Fixed now.
========================
Prof. Victor Lazzarini
Maynooth University
Ireland

> On 23 Mar 2024, at 16:24, Anders Genell  wrote:
> 
> I ended up compiling with gettext disabled as I couldn't figure out what package to install to fulfil the dependency...
> 
> So I modified an example csd with an UDO containing an undersample command. It ran for the first instance in the score but then it aborted... 
> 
> 
> 
> ; Select audio/midi flags here according to platform
> -odac  ;;;realtime audio out
> ; -iadc    ;;;uncomment -iadc if real audio input is needed too
> ; For Non-realtime ouput leave only the line below:
> ; -o sr.wav -W ;;; for file output any platform
> 
> 
> 
> sr = 48000
> ksmps = 32
> nchnls = 2
> 0dbfs  = 1
> 
> 
>         opcode Resample, a, a
> 
>         undersample 4
> 
> ain     xin                     ; read input parmeter
> aout    init 0                  ; initialize output
> aout    =  ain                  ; simple tone-like filter
>         xout aout               ; write output
> 
>         endop
> 
> 
> instr 1
> 
> ihar    = int(sr/2/p4)          ; maximum possible number of harmonics w/o aliasing
> prints  "maximum number of harmonics = %d \\n", ihar
> kenv    linen .5, 1, p3, .2     ; envelope
> asig    buzz  kenv, p4, ihar, 1
> 
> 
> aout Resample asig
> 
>         outs  aout, aout
> 
> 
> endin
> 
> 
> f1 0 4096 10 1  ;sine wave
> 
> i 1 0 3 100     ;different frequencies
> i 1 + 3 1000
> i 1 + 3 10000
> e
> 
> 
> 
> UnifiedCSD:  testlocalsr.csd
> Elapsed time at end of orchestra compile: real: 0.000s, CPU: 0.000s
> sorting score ...
>         ... done
> Elapsed time at end of score sort: real: 0.000s, CPU: 0.000s
> --Csound version 7.0 (double samples) Mar 23 2024
> [commit: 4ac0d3eb4a1c771deaba9f681dc48e5732935806]
> libsndfile-1.0.28
> displays suppressed
> sr = 48000.0, kr = 1500.000, ksmps = 32
> 0dBFS level = 1.0, A4 tuning = 440.0
> orch now loaded
> audio buffered in 256 sample-frame blocks
> ALSA output: total buffer size: 1024, period size: 256
> writing 512 sample blks of 64-bit floats to dac
> SECTION 1:
> ftable 1:
> new alloc for instr 1:
> maximum number of harmonics = 240
> B  0.000 ..  3.000 T  3.000 TT  3.000 M:  0.44296  0.44296
> maximum number of harmonics = 24
> free(): double free detected in tcache 2
> 
> csound command: Aborted
> 
> 
> On Wed, Mar 20, 2024 at 10:06 PM Anders Genell  wrote:
> Thanks, I’ll install libintl and see if things clear up. 
> 
> Regards,
> Anders
> 
>> 20 mars 2024 kl. 19:41 skrev Stephen Kyne :
>> 
>>  WSL should work. I use the ubuntu image to test linux builds. You need to either install libintl or else disable it's usage via USE_GETTEXT=0.
>> 
>> It shouldn't really be failing as we have code to test if it's found but it seems to be a bug that was introduced in the last few months. I'll try and take a further look.
>> 
>> Stephen
>> From: Csound-developers  on behalf of Victor Lazzarini 
>> Sent: Wednesday 20 March 2024 17:08
>> To: CSOUND-DEV@LISTSERV.HEANET.IE 
>> Subject: Re: [Csnd-dev] [EXTERNAL] [Csnd-dev] UDO local sr   I really don't know, this branch was started off develop last Saturday. There has been a lot of change in the build system and it might have broken people's builds. OTOH the CI builds are passing. 
>> 
>> Stephen, do you have a suggestion?
>> 
>> Prof. Victor Lazzarini 
>> Maynooth University
>> Ireland
>> 
>>> On 20 Mar 2024, at 16:04, Anders Genell  wrote:
>>> 
>>>  *localsr* branch, that is
>>> 
>>> On Wed, Mar 20, 2024 at 5:03 PM Anders Genell  wrote:
>>> Just checked out locals branch and got a cmake error: 
>>> 
>>> "CMake Error at CMakeLists.txt:1093 (add_library):
>>>   Target "csound64" links to target "Intl::Intl" but the target was not
>>>   found.  Perhaps a find_package() call is missing for an IMPORTED target, or
>>>   an ALIAS target is missing?"
>>> 
>>> I suppose I'm doing something wrong, the first of which may very well be the fact that I am building on the Windows Subsystem for Linux - that's just because I'm at work doing ordinary work stuff on my work computer - I can build on an ordinary linux system later...
>>> 
>>> Regards,
>>> Anders
>>> 
>>> On Wed, Mar 20, 2024 at 2:12 PM Anders Genell  wrote:
>>> Ah, it was just arrays that were limited. Great! 
>>> 
>>> I'll be sure to install libsamplerate and will see what I can do with regards to testing
>>> 
>>> Readers,
>>> ANders
>>> 
>>> On Wed, Mar 20, 2024 at 2:10 PM Victor Lazzarini  wrote:
>>> Make sure you have libsamplerate for the best quality conversion. There’s a simple fallback converter, but Secret Rabbit Code is better.
>>> 
>>> The limitations are:
>>> 
>>> - can’t setksmps alongside it
>>> - can’t use a-rate or k-rate arrays (that may be possibly changed in the future).
>>> - if you use global arrays or bus channels it won’t work either. All access has to be
>>> through xin and xout, because that’s where the conversion happens.
>>> 
>>> Testing is helpful, thanks for that.
>>> ========================
>>> Prof. Victor Lazzarini
>>> Maynooth University
>>> Ireland
>>> 
>>> > On 20 Mar 2024, at 11:27, Anders Genell  wrote:
>>> > 
>>> > Ah, brilliant!
>>> > I just saw the flurry of emails connected to the code submissions and it said something about not allowing a-rate, but that was likely a misinterpretation of mine.
>>> > 
>>> > I will try to build and test to see if my scenario seems to work as I hope.
>>> > 
>>> > Thank you!
>>> > 
>>> > Regards,
>>> > Anders
>>> > 
>>> > 
>>> > On Wed, Mar 20, 2024 at 10:51 AM Victor Lazzarini  wrote:
>>> > Hi Anders,
>>> > 
>>> > There is now an undersample opcode to go along with the oversample. I am not sure what you mean about 2., it does allow
>>> > a-rate opcodes.
>>> > 
>>> > best
>>> > ========================
>>> > Prof. Victor Lazzarini
>>> > Maynooth University
>>> > Ireland
>>> > 
>>> > > On 20 Mar 2024, at 08:55, Anders Genell  wrote:
>>> > > 
>>> > > *Warning*
>>> > > This email originated from outside of Maynooth University's Mail System. Do not reply, click links or open attachments unless you recognise the sender and know the content is safe.
>>> > > This is mainly for Victor Lazzarini, 
>>> > > 
>>> > > I saw you added code in a PR to implement local sample rate in UDOs which I think is absolutely fantastic.
>>> > > It seems there are two limitations that I really hope could be addressed in the future:
>>> > > 1. The code allows for upsampling but not for downsampling
>>> > > 2. The code does not allow for a-rate opcodes
>>> > > 
>>> > > The use case I am hoping will be possible in the future is when having a reasonably high global sr (say 48kHz or even 96kHz) where I would like to apply relatively narrow band filters with center frequencies at say 20Hz, 25Hz 32Hz or the like. Such filters tend to be somewhat unstable at such high SR if they also need to be reasonably steep, and by downsampling before filtering a better result is often achieved. This is commonly done in signal processing for analysis in e.g. Matlab or GNU octave.
>>> > > The idea is e.g. to be able to manipulate individual bands, like in the pvs group of opcodes, and combining the individual bands back to one signal afterwords, so just docking global reduction of SR would not be a viable route, unless of course starting several instances of csound and using channel opcodes to combine everything in the end.
>>> > > 
>>> > > Would this be a possibility in some future iteration of the newly implemented code?
>>> > > 
>>> > > Regards,
>>> > > Anders
>>> 

 One thing I noticed was that SRC modes 0,1,2 (sync interp) had varying levels of input/output delay due to windowing (decreasing with mode number). 

Prof. Victor Lazzarini

Maynooth University

Ireland

On 24 Mar 2024, at 11:02, Anders Genell <anders.genell@gmail.com> wrote:



I pulled and built and had no errors in performance, so thank you for the quick mending.

Now on to my original idea for a use case - I'll report back if/when I get to the testing phase.

Regards,

Anders

On Sat, Mar 23, 2024 at 7:39 PM Victor Lazzarini <Victor.Lazzarini@mu.ie> wrote:

ah, thanks for this. I hadn’t tried with more than one instance and you caught a bug in
the deletion of the converters.

Fixed now.
========================
Prof. Victor Lazzarini
Maynooth University
Ireland

> On 23 Mar 2024, at 16:24, Anders Genell <anders.genell@GMAIL.COM> wrote:
>
> I ended up compiling with gettext disabled as I couldn't figure out what package to install to fulfil the dependency...
>
> So I modified an example csd with an UDO containing an undersample command. It ran for the first instance in the score but then it aborted...
>
> <CsoundSynthesizer>
> <CsOptions>
> ; Select audio/midi flags here according to platform
> -odac  ;;;realtime audio out
> ; -iadc    ;;;uncomment -iadc if real audio input is needed too
> ; For Non-realtime ouput leave only the line below:
> ; -o sr.wav -W ;;; for file output any platform
> </CsOptions>
> <CsInstruments>
>
> sr = 48000
> ksmps = 32
> nchnls = 2
> 0dbfs  = 1
>
>
>         opcode Resample, a, a
>
>         undersample 4
>
> ain     xin                     ; read input parmeter
> aout    init 0                  ; initialize output
> aout    =  ain                  ; simple tone-like filter
>         xout aout               ; write output
>
>         endop
>
>
> instr 1
>
> ihar    = int(sr/2/p4)          ; maximum possible number of harmonics w/o aliasing
> prints  "maximum number of harmonics = %d \\n", ihar
> kenv    linen .5, 1, p3, .2     ; envelope
> asig    buzz  kenv, p4, ihar, 1
>
>
> aout Resample asig
>
>         outs  aout, aout
>
>
> endin
> </CsInstruments>
> <CsScore>
> f1 0 4096 10 1  ;sine wave
>
> i 1 0 3 100     ;different frequencies
> i 1 + 3 1000
> i 1 + 3 10000
> e
> </CsScore>
> </CsoundSynthesizer>
>
> UnifiedCSD:  testlocalsr.csd
> Elapsed time at end of orchestra compile: real: 0.000s, CPU: 0.000s
> sorting score ...
>         ... done
> Elapsed time at end of score sort: real: 0.000s, CPU: 0.000s
> --Csound version 7.0 (double samples) Mar 23 2024
> [commit: 4ac0d3eb4a1c771deaba9f681dc48e5732935806]
> libsndfile-1.0.28
> displays suppressed
> sr = 48000.0, kr = 1500.000, ksmps = 32
> 0dBFS level = 1.0, A4 tuning = 440.0
> orch now loaded
> audio buffered in 256 sample-frame blocks
> ALSA output: total buffer size: 1024, period size: 256
> writing 512 sample blks of 64-bit floats to dac
> SECTION 1:
> ftable 1:
> new alloc for instr 1:
> maximum number of harmonics = 240
> B  0.000 ..  3.000 T  3.000 TT  3.000 M:  0.44296  0.44296
> maximum number of harmonics = 24
> free(): double free detected in tcache 2
>
> csound command: Aborted
>
>
> On Wed, Mar 20, 2024 at 10:06 PM Anders Genell <anders.genell@gmail.com> wrote:
> Thanks, I’ll install libintl and see if things clear up.
>
> Regards,
> Anders
>
>> 20 mars 2024 kl. 19:41 skrev Stephen Kyne <stevek@outlook.ie>:
>>
>>  WSL should work. I use the ubuntu image to test linux builds. You need to either install libintl or else disable it's usage via USE_GETTEXT=0.
>>
>> It shouldn't really be failing as we have code to test if it's found but it seems to be a bug that was introduced in the last few months. I'll try and take a further look.
>>
>> Stephen
>> From: Csound-developers <CSOUND-DEV@LISTSERV.HEANET.IE> on behalf of Victor Lazzarini <Victor.Lazzarini@MU.IE>
>> Sent: Wednesday 20 March 2024 17:08
>> To: CSOUND-DEV@LISTSERV.HEANET.IE <CSOUND-DEV@LISTSERV.HEANET.IE>
>> Subject: Re: [Csnd-dev] [EXTERNAL] [Csnd-dev] UDO local sr   I really don't know, this branch was started off develop last Saturday. There has been a lot of change in the build system and it might have broken people's builds. OTOH the CI builds are passing.
>>
>> Stephen, do you have a suggestion?
>>
>> Prof. Victor Lazzarini
>> Maynooth University
>> Ireland
>>
>>> On 20 Mar 2024, at 16:04, Anders Genell <anders.genell@gmail.com> wrote:
>>>
>>>  *localsr* branch, that is
>>>
>>> On Wed, Mar 20, 2024 at 5:03 PM Anders Genell <anders.genell@gmail.com> wrote:
>>> Just checked out locals branch and got a cmake error:
>>>
>>> "CMake Error at CMakeLists.txt:1093 (add_library):
>>>   Target "csound64" links to target "Intl::Intl" but the target was not
>>>   found.  Perhaps a find_package() call is missing for an IMPORTED target, or
>>>   an ALIAS target is missing?"
>>>
>>> I suppose I'm doing something wrong, the first of which may very well be the fact that I am building on the Windows Subsystem for Linux - that's just because I'm at work doing ordinary work stuff on my work computer - I can build on an ordinary linux system later...
>>>
>>> Regards,
>>> Anders
>>>
>>> On Wed, Mar 20, 2024 at 2:12 PM Anders Genell <anders.genell@gmail.com> wrote:
>>> Ah, it was just arrays that were limited. Great!
>>>
>>> I'll be sure to install libsamplerate and will see what I can do with regards to testing
>>>
>>> Readers,
>>> ANders
>>>
>>> On Wed, Mar 20, 2024 at 2:10 PM Victor Lazzarini <Victor.Lazzarini@mu.ie> wrote:
>>> Make sure you have libsamplerate for the best quality conversion. There’s a simple fallback converter, but Secret Rabbit Code is better.
>>>
>>> The limitations are:
>>>
>>> - can’t setksmps alongside it
>>> - can’t use a-rate or k-rate arrays (that may be possibly changed in the future).
>>> - if you use global arrays or bus channels it won’t work either. All access has to be
>>> through xin and xout, because that’s where the conversion happens.
>>>
>>> Testing is helpful, thanks for that.
>>> ========================
>>> Prof. Victor Lazzarini
>>> Maynooth University
>>> Ireland
>>>
>>> > On 20 Mar 2024, at 11:27, Anders Genell <anders.genell@GMAIL.COM> wrote:
>>> >
>>> > Ah, brilliant!
>>> > I just saw the flurry of emails connected to the code submissions and it said something about not allowing a-rate, but that was likely a misinterpretation of mine.
>>> >
>>> > I will try to build and test to see if my scenario seems to work as I hope.
>>> >
>>> > Thank you!
>>> >
>>> > Regards,
>>> > Anders
>>> >
>>> >
>>> > On Wed, Mar 20, 2024 at 10:51 AM Victor Lazzarini <Victor.Lazzarini@mu.ie> wrote:
>>> > Hi Anders,
>>> >
>>> > There is now an undersample opcode to go along with the oversample. I am not sure what you mean about 2., it does allow
>>> > a-rate opcodes.
>>> >
>>> > best
>>> > ========================
>>> > Prof. Victor Lazzarini
>>> > Maynooth University
>>> > Ireland
>>> >
>>> > > On 20 Mar 2024, at 08:55, Anders Genell <anders.genell@GMAIL.COM> wrote:
>>> > >
>>> > > *Warning*
>>> > > This email originated from outside of Maynooth University's Mail System. Do not reply, click links or open attachments unless you recognise the sender and know the content is safe.
>>> > > This is mainly for Victor Lazzarini,
>>> > >
>>> > > I saw you added code in a PR to implement local sample rate in UDOs which I think is absolutely fantastic.
>>> > > It seems there are two limitations that I really hope could be addressed in the future:
>>> > > 1. The code allows for upsampling but not for downsampling
>>> > > 2. The code does not allow for a-rate opcodes
>>> > >
>>> > > The use case I am hoping will be possible in the future is when having a reasonably high global sr (say 48kHz or even 96kHz) where I would like to apply relatively narrow band filters with center frequencies at say 20Hz, 25Hz 32Hz or the like. Such filters tend to be somewhat unstable at such high SR if they also need to be reasonably steep, and by downsampling before filtering a better result is often achieved. This is commonly done in signal processing for analysis in e.g. Matlab or GNU octave.
>>> > > The idea is e.g. to be able to manipulate individual bands, like in the pvs group of opcodes, and combining the individual bands back to one signal afterwords, so just docking global reduction of SR would not be a viable route, unless of course starting several instances of csound and using channel opcodes to combine everything in the end.
>>> > >
>>> > > Would this be a possibility in some future iteration of the newly implemented code?
>>> > >
>>> > > Regards,
>>> > > Anders
>>>