Nabble - Music-DSP

Fellowship opportunities for C4DM, Queen Mary University of London

2009-12-18T08:33:47Z

Dear List,

[Apologies for cross-posting. Please pass on to anyone who may be interested.]

To help people who would like to carry out postdoctoral research work here at the Centre for Digital Music (C4DM) at Queen Mary University of London, we have gathered together some information about possible relevant postdoctoral fellowship schemes at:
www.elec.qmul.ac.uk/digitalmusic/fellowships.html

Those currently open are summarized below.

If you are interested in any of these opportunities, please FIRST check the scheme details for eligibility information to make sure you are eligible for that scheme. For example, some schemes are only valid for researchers from certain countries (e.g. non-UK citizens).

These schemes are often quite competitive, so you will need to write a strong proposal. You will need to be prepared to write most of the proposal yourself, although we will be able to help with information about C4DM as the host for your research.

An overview of our research is at: www.elec.qmul.ac.uk/digitalmusic/

To discuss any proposal for research in one of our general areas of interest, please contact one of the people in the areas below:

* Audio Engineering: www.elec.qmul.ac.uk/digitalmusic/audioengineering.html
Contact: Josh Reiss - www.elec.qmul.ac.uk/people/josh

* Interactional Sound: www.elec.qmul.ac.uk/digitalmusic/interactional.html
Contact: Nick Bryan-Kinns - www.dcs.qmul.ac.uk/~nickbk

* Machine Listening: www.elec.qmul.ac.uk/digitalmusic/machinelistening.html
Contact: Mark Plumbley - www.elec.qmul.ac.uk/people/markp

* Music Informatics: www.elec.qmul.ac.uk/digitalmusic/musicinformatics.html
Contact: Simon Dixon - www.elec.qmul.ac.uk/people/simond

Alternatively, if you know a particular person that you would like to work with, see the list of Academic Staff at www.elec.qmul.ac.uk/digitalmusic/people.html and contact them directly. Other people listed there include Mark Sandler (head of C4DM), Anssi Klapuri (music signal analysis, auditory scene analysis) and Panos Kudumakis (Music rights management & standardization).

Best wishes,

Mark Plumbley

---------------

Selected open fellowship opportunities

Newton International Fellowship
Awards enable early stage postdoctoral researchers to work at UK research institutions for a period of two years. Applicants should be working outside the UK and should not hold UK citizenship.
Deadline: 08 February 2010 [Extended deadline]
More details: www.newtonfellowships.org (see also below)

Dorothy Hodgkin Fellowships
The Dorothy Hodgkin Fellowship scheme supports excellent scientists and engineers at an early stage of their career, and is designed to help them to progress to a permanent position. It is aimed specifically at researchers who require a flexible working pattern due to personal circumstances including parental/caring responsibilites and health issues. Female candiates are particularly invited to apply.
Deadline: 20 January 2010
More details: http://royalsociety.org/Dorothy-Hodgkin-Fellowships/

1851 Research Fellowships in Science or Engineering
These enable PhD level scientists or engineers of outstanding promise to conduct research for a further period. A candidate must be a citizen of the United Kingdom or the Commonwealth, or of the Republics of Ireland or Pakistan.
Deadline: 25 February 2010
More details: www.royalcommission1851.org.uk/res_fellow.html

Leverhulme Trust Early Career Fellowships
These aim to provide career development opportunities for those who are at a relatively early stage of their academic careers but with a proven record of research. Applicants should normally hold a degree from a UK higher education institution, or hold an academic position in the UK.
Deadline: 11 March 2010
More details: www.leverhulme.ac.uk/grants_awards/grants/early_career_fellowships/

For more fellowships, see www.elec.qmul.ac.uk/digitalmusic/fellowships.html

--
Prof Mark D Plumbley
Electronic Engineering & Computer Science (Eng Bldg)
Queen Mary University of London
Mile End Road, London E1 4NS, UK
Tel: +44 (0)20 7882 7518
Fax: +44 (0)20 7882 7997
Email: mark.plumbley@...
http://www.elec.qmul.ac.uk/people/markp/
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Final Reminder: Audio Engineering Society 128th Convention, London, May 2010- Call for Papers

2009-12-15T10:11:00Z

Apologies if you receive multiple copies of this message.

AUDIO ENGINEERING SOCIETY 128th Convention, 2010 CALL for PAPERS London, UK
Dates: 2010 May 22 - May 25 **Note revised dates
www.aes.org/events/128

The AES 128th Convention Committee invites submission of technical papers for presentation at the 2010 May 20 to 23 meeting in London. By 2009 December 18, a proposed title, 60- to 120-word abstract, and 500- to 750-word précis of the paper must be submitted electronically to the AES 128th proposal submission site at www.aes.org/128th_authors. Submissions will be accepted starting approximately 2009 November 4. Presenting authors (one per paper) who are members of the AES or student members will be required to pay 60% of the member or student convention registration fees and they will receive a CD-ROM of the papers. Presenting authors who are nonmembers will pay the full nonmember registration fee and receive a CD-ROM. Acceptance of proposed papers will be determined by a peer-review committee based on an assessment of the abstract and précis. Presenting authors who are student members and whose papers are accepted for presentation will be eligible for the Student Paper Award at the 128th. The précis must clearly describe the work performed, methods employed, conclusions and significance of the paper with respect to other published work in the field. During the online submission process you will be asked to specify whether you prefer to present your paper in a lecture or poster session. Highly detailed papers are better suited to poster sessions, which permit greater interaction between author and audience. The convention committee reserves the right to reassign papers to any session. Whether a lecture or a poster, a complete electronic manuscript submitted before 2010 March 12 is required before the paper can be accepted for presentation at the convention. During the submission process, authors will be asked if their convention papers should be considered for possible publication in the AES Journal.

Proposed topics for papers include but are not limited to:
----------------------------------------------------------
Applications in Audio
   Audio for games
   Digital broadcasting
   Forensic audio
   Automotive audio
   Audio for mobile and handheld devices
   Audio in education
   Networked, Internet, and remote audio Audio content management
   Archiving and restoration
   Digital libraries
   Automatic content description
   Audio information retrieval
Audio Processing
   Analysis and synthesis of sound
   Machine listening
   Music and speech signal processing
   High resolution audio
   Audio coding and compression
Recording, Production, and    Reproduction
   Live event and stage audio
   Mixing, remixing, and mastering
   Multichannel and spatial audio
   Room and architectural acoustics
   Sound design and reinforcement
   Studio recording techniques
Audio Equipment
   Microphones, converters, and amplifiers
   Loudspeakers and headphones
   Wireless and wearable audio
   Instrumentation and measurement
   Protocols and data formats
Perception
   Audio perception
   Hearing loss, protection, and enhancement
   Listening tests and evaluation
   Speech intelligibility
   Psychoacoustics
Emerging Audio Technologies
   Innovative applications
   Interactive sound
   New audio interfaces
   Web 2.0 technologies

Submission of papers schedule
-----------------------------
Proposal deadline: 2009 December 18
Acceptance emailed: 2010 January 20
Paper deadline: 2010 March 12
By 2010 January 20 authors will be advised whether or not their proposed papers have been accepted.
Please submit proposed title and abstract at www.aes.org/128th_authors no later than 2009 December 18.
If you have any questions, contact 128th_papers@...

Papers Chair: Peter Mapp, Peter Mapp Associates
General Chair: Josh Reiss, Centre for Digital Music, Queen Mary University of London

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[admin] music-dsp FAQ

2009-12-15T04:00:00Z

Hi,

Just a reminder that if you are new to the list you should read the
music-dsp FAQ. It contains answers to both technical _and_
adminstrative questions that often come up on the list. If your question
appears in the FAQ it is safe to assume that it has been discussed on the
list many times in the past, and you should probably have a look through
the list archives before posting your question to the list.

http://music.columbia.edu/cmc/music-dsp/musicdspFAQ.html

Also of interest to new and not-so-new list members:

The music-dsp list archives
http://music.columbia.edu/cmc/music-dsp/musicdsparchives.html

The music-dsp source code archive
http://www.musicdsp.org

music-dsp books and reviews
http://music.columbia.edu/cmc/music-dsp/dspbooks.html

All this and more at:
http://music.columbia.edu/cmc/music-dsp

Hasta la pasta,
douglas

(this is an automated message sent out on the 1st and 15th of each month)
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Re: hardware question

2009-12-11T08:39:12Z

I have even more than my typical level of ignorance on RF, but
depending on the task perhaps a desktop computer-interfaced wideband
receiver would be useful?

Here is a link to a few such:

http://www.universal-radio.com/catalog/commrxvr.html

And a not-terribly-expensive example--

http://www.universal-radio.com/catalog/commrxvr/0106.html

More generally, one old way to get part of a high-freq spectrum down
low enough to capture with a garden-variety ADC would be Heterodyning.

http://en.wikipedia.org/wiki/Intermediate_frequency

Awhile ago happened upon a NASA school project for radio astronomy. Am
referencing it here because the explanations are clear and well-
written, and there is an example simple heterodyne circuit for
frequency-shifting 20 MHz radio-astronomy signals to the audio range.
Parts of the circuit could be adapted for more general purposes perhaps.

http://radiojove.gsfc.nasa.gov/

And the circuit with explanations:

http://radiojove.gsfc.nasa.gov/telescope/rcvr_manual.pdf

See page 11 for the circuit diagram. I am extremely ignorant of RF, as
said before, but something like this with a computer-controlled local
mixing oscillator perhaps wouldn't be a beast to construct, and allow
computer-controlled examination of a movable 'window' into the wider
spectrum. The parts count looks pretty low compared to what would have
been necessary before the advent of IC's.

James Chandler Jr.

On Dec 11, 2009, at 6:01 AM, Brian Willoughby wrote:

> I realize that I'm really behind in reading this list, but maybe
> someone is interested still.
>
> Texas Instruments has a line of high speed ADC chips. I'm using a
> pair of the ADS7951 chips which can handle 12-bit conversions at 1
> MHz. I believe they have faster chips. Check out their site for
> more details.
>
> These use SPI or compatible serial interfacing, but you probably need
> a DSP on board to handle that kind of throughput, and then a USB or
> FireWire interface which can handle transmission. I doubt you'll be
> able to get 3 GHz into your computer for real-time processing.
> You'll need to build a little DSP board and possibly send only the
> summary of the processing to the host computer.
>
> You might be able to find a USB oscilloscope, but I don't know if any
> of those offer programmable real-time processing. Alternatively,
> there are a couple of DIY platforms which might work with a A/D
> daughter board.
>
> None of this is going to be cheap, I'm afraid.
>
> Brian Willoughby
> Sound Consulting
>
>
> On Jan 8, 2007, at 13:41, Tyler Adams wrote:
>> I have a hardware question and I hope that it's not too much of a
>> newbie
>> question for the list.
>>
>> I need some DSP hardware for getting signals containing high
>> frequencies
>> into my computer (for example somewhere in the range of 1Mhz -
>> 3Ghz). I'm
>> looking for something on the cheap side. And I'm hoping to process
>> the
>> signals in real-time. Does anyone have any recommendations?
>>
>> Has anyone worked with Labview? I think they have a small USB unit
>> but
>> I'm not certain.
>
>
> --
> dupswapdrop -- the music-dsp mailing list and website:
> subscription info, FAQ, source code archive, list archive, book
> reviews, dsp links
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> http://music.columbia.edu/mailman/listinfo/music-dsp

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Re: hardware question

2009-12-11T08:07:57Z

FWIW, if you really want wide-bandwidth sampling, along with capability
to handle RF in various bands up into the low microwave you might
consider looking into the GnuRadio project. They've got hardware and
software that will get you all the way from the antenna to bits on your
hard drive. A fairly comprehensive solution, including an API. More here:

http://gnuradio.org/trac

Interestingly enough, there's a fair amount of audio processing code in
the main line. Might be worth a closer look as a general-purpose DSP
environment for music too.

Eric

On 12/11/2009 04:01 AM, Brian Willoughby wrote:

> I realize that I'm really behind in reading this list, but maybe
> someone is interested still.
>
> Texas Instruments has a line of high speed ADC chips. I'm using a
> pair of the ADS7951 chips which can handle 12-bit conversions at 1
> MHz. I believe they have faster chips. Check out their site for
> more details.
>
> These use SPI or compatible serial interfacing, but you probably need
> a DSP on board to handle that kind of throughput, and then a USB or
> FireWire interface which can handle transmission. I doubt you'll be
> able to get 3 GHz into your computer for real-time processing.
> You'll need to build a little DSP board and possibly send only the
> summary of the processing to the host computer.
>
> You might be able to find a USB oscilloscope, but I don't know if any
> of those offer programmable real-time processing. Alternatively,
> there are a couple of DIY platforms which might work with a A/D
> daughter board.
>
> None of this is going to be cheap, I'm afraid.
>
> Brian Willoughby
> Sound Consulting
>
>
> On Jan 8, 2007, at 13:41, Tyler Adams wrote:
>> I have a hardware question and I hope that it's not too much of a
>> newbie
>> question for the list.
>>
>> I need some DSP hardware for getting signals containing high
>> frequencies
>> into my computer (for example somewhere in the range of 1Mhz -
>> 3Ghz). I'm
>> looking for something on the cheap side. And I'm hoping to process
>> the
>> signals in real-time. Does anyone have any recommendations?
>>
>> Has anyone worked with Labview? I think they have a small USB unit
>> but
>> I'm not certain.
>
>
> --
> dupswapdrop -- the music-dsp mailing list and website:
> subscription info, FAQ, source code archive, list archive, book reviews, dsp links
> http://music.columbia.edu/cmc/music-dsp
> http://music.columbia.edu/mailman/listinfo/music-dsp
>

Re: hardware question

2009-12-11T07:28:06Z

I believe USB 2.0 runs at 480 mbps. but you will most likely need to
process your data on a DSP chip if you want to sample at a rate that
high. However, TI has evaluation modules (EVM) and product development
kits (? PDK) that can help you get started.

e.g. http://focus.ti.com/docs/toolsw/folders/print/ads5463evm.html

There are two things to realize,

What is the bandwidth of the information you want to capture? This is
because your sampling rate needs to be at least two times the bandwidth
of what you want to capture. So if your information is 200 kHz centered
about 1 Ghz, you will only need to sample at 400 kHz (SHOCKING!). In
order to remove data thats at 0 - 200khz, 200khz - 400khz and so on, you
will need to pass your signal through an analog filter before you
sample. This will allow you to sample at much lower sampling rate and
have your information at managable size.

The second thing is that you can interleave ADCs to achieve higher
sampling rate. However, it doesn't seem to me that you want to do this
yourself, but again, TI has some evaluation modules that can come to
your rescue.

Omer Osman
George Washington University

--

> Message: 3
> Date: Fri, 11 Dec 2009 03:01:06 -0800
> From: Brian Willoughby <brianw@...>
> Subject: Re: [music-dsp] hardware question
> To: A discussion list for music-related DSP
> <music-dsp@...>
> Message-ID: <59FA1DD9-E87C-445C-BA5A-A21FD45331D8@...>
> Content-Type: text/plain; charset=US-ASCII; delsp=yes; format=flowed
>
> I realize that I'm really behind in reading this list, but maybe
> someone is interested still.
>
> Texas Instruments has a line of high speed ADC chips. I'm using a
> pair of the ADS7951 chips which can handle 12-bit conversions at 1
> MHz. I believe they have faster chips. Check out their site for
> more details.
>
> These use SPI or compatible serial interfacing, but you probably need
> a DSP on board to handle that kind of throughput, and then a USB or
> FireWire interface which can handle transmission. I doubt you'll be
> able to get 3 GHz into your computer for real-time processing.
> You'll need to build a little DSP board and possibly send only the
> summary of the processing to the host computer.
>
> You might be able to find a USB oscilloscope, but I don't know if any
> of those offer programmable real-time processing. Alternatively,
> there are a couple of DIY platforms which might work with a A/D
> daughter board.
>
> None of this is going to be cheap, I'm afraid.
>
> Brian Willoughby
> Sound Consulting
>
>
> On Jan 8, 2007, at 13:41, Tyler Adams wrote:
>
>> I have a hardware question and I hope that it's not too much of a
>> newbie
>> question for the list.
>>
>> I need some DSP hardware for getting signals containing high
>> frequencies
>> into my computer (for example somewhere in the range of 1Mhz -
>> 3Ghz). I'm
>> looking for something on the cheap side. And I'm hoping to process
>> the
>> signals in real-time. Does anyone have any recommendations?
>>
>> Has anyone worked with Labview? I think they have a small USB unit
>> but
>> I'm not certain.
>>
>
>
>

Re: Bandlimiting C0, C1, and C2?

2009-12-11T03:52:54Z

> I've been scanning old articles in this list, and I notice folks like
> Andrew Simper and Antti mention techniques for bandlimiting C0, C1,
> and C2 in passing comments. Could anyone provide references to
> papers or anything discussing these techniques? I don't see mention
> of these terms in the band-limited articles that I searched.

This one has some theoretical coverage for an arbitrary-order discontinuity:
http://www.native-instruments.com/fileadmin/ni_media/downloads/pdf/SineSync.pdf

Regards,
Vadim

--
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Senior Software Developer | R&D

Tel +49-30-611035-0
Fax +49-30-611035-2600

NATIVE INSTRUMENTS GmbH
Schlesische Str. 28
10997 Berlin, Germany
http://www.native-instruments.com

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Bandlimiting C0, C1, and C2?

2009-12-11T03:30:00Z

I've been scanning old articles in this list, and I notice folks like
Andrew Simper and Antti mention techniques for bandlimiting C0, C1,
and C2 in passing comments. Could anyone provide references to
papers or anything discussing these techniques? I don't see mention
of these terms in the band-limited articles that I searched.

Anyway, I working with waveshaping unknown analog frequencies using
hard sync and analog-style waveforms, so the above terms sound
promising. I'd like to do more research if the stuff is documented
somewhere.

Thanks.

Brian

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Re: hardware question

2009-12-11T03:01:06Z

I realize that I'm really behind in reading this list, but maybe
someone is interested still.

Texas Instruments has a line of high speed ADC chips. I'm using a
pair of the ADS7951 chips which can handle 12-bit conversions at 1
MHz. I believe they have faster chips. Check out their site for
more details.

These use SPI or compatible serial interfacing, but you probably need
a DSP on board to handle that kind of throughput, and then a USB or
FireWire interface which can handle transmission. I doubt you'll be
able to get 3 GHz into your computer for real-time processing.
You'll need to build a little DSP board and possibly send only the
summary of the processing to the host computer.

You might be able to find a USB oscilloscope, but I don't know if any
of those offer programmable real-time processing. Alternatively,
there are a couple of DIY platforms which might work with a A/D
daughter board.

None of this is going to be cheap, I'm afraid.

Brian Willoughby
Sound Consulting

On Jan 8, 2007, at 13:41, Tyler Adams wrote:

> I have a hardware question and I hope that it's not too much of a
> newbie
> question for the list.
>
> I need some DSP hardware for getting signals containing high
> frequencies
> into my computer (for example somewhere in the range of 1Mhz -
> 3Ghz). I'm
> looking for something on the cheap side. And I'm hoping to process
> the
> signals in real-time. Does anyone have any recommendations?
>
> Has anyone worked with Labview? I think they have a small USB unit
> but
> I'm not certain.

Audio Engineering Society 128th Convention, London, May 2010- Call for Papers reminder

2009-12-02T09:17:42Z

[admin] music-dsp FAQ

2009-12-01T04:00:01Z

Re: bandlimited interpolation ofnon-uniformly-spacedsamples?

2009-11-29T23:15:50Z

robert bristow-johnson wrote:
> all these BLITs and BLEPs and BLAMs... what comes next, kaBLUIe?

Yeah I'm not sure where BLAMs came from.. I seem to remember Eli's original
paper explicitly refrained from giving band limited ramps an abbreviation
for the reason just stated.

> i just wanna understand this:
>
> 1. first connect your original un-uniformly spaced points with
> lines,
>
> 2. then replace the lines with BLAMs.
>
> and that's what you do for a waveform period? somehow we get the
> different ramps to add up to the slope we want between each pair of
> adjacent points.

music-dsp, Thu Mar 16 06:40:42 EST 2006, Andrew Simper wrote:
>Blam is a band limited ramp ie it is an pre integrated Blep. Blep is a band
>limited step ie a pre integrated Blit. Blit is a band limited impulse ie
>some finite sinc like
>function.http://music.columbia.edu/pipermail/music-dsp/2006-March/065138.html
There's a coupe of related threads around that time.> must we stop with
BLAMs and linear ramps? could we connect our > original points with
piece-wise something-higher-order-than-linear > and then replace those with
BL-something-else?Yes, we could do that. Xenakis' DSS did use straight lines
however, so it depends on whether you want bandlimited DSS or something
more.. (I imagine Xenakis would be just as happy with splines though).

Ross.

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Re: bandlimited interpolation of non-uniformly-spacedsamples?

2009-11-29T22:48:53Z

On Nov 30, 2009, at 12:50 AM, Ross Bencina wrote:

> Hi Spencer
>
>> I've written an implementation of Iannis Xenakis's Dynamic Stochastic
>> Synthesis as an object for Pure Data and Max/MSP.
> ...
>> Implementations that I'm familiar with have all just interpolated
>> linearly, but I'm interested in exploring other interpolation
>> methods,
>> especially looking to generate bandlimited waveforms to eliminate
>> aliasing problems.
>
>
> I have a little familiarity with the DSS method but havn't
> implemented it.
> If I recall correctly it's a bit like generating triangle waveforms
> where
> length of each line segment is randomised and the amplitude of each
> endpoint
> (direction change) is also randomised (at least that's a sufficient
> definition for discussing bandlimited synthesis).
>
> You might not actually want to use bandlimited interpolation of the
> non-uniformly spaced line-segment endpoints, but to continue to use
> line
> segments and to just synthesize the discontinuities between line
> segments in
> a band limited way -- that would give you alias free DSS.
>
> This can be achieved using a BLAM (band limited ramp), a variation
> on a
> miblep.
> http://www.experimentalscene.com/articles/minbleps.php

all these BLITs and BLEPs and BLAMs... what comes next, kaBLUIe?

i just wanna understand this:

1. first connect your original un-uniformly spaced points with
lines,

2. then replace the lines with BLAMs.

and that's what you do for a waveform period? somehow we get the
different ramps to add up to the slope we want between each pair of
adjacent points.

must we stop with BLAMs and linear ramps? could we connect our
original points with piece-wise something-higher-order-than-linear
and then replace those with BL-something-else?

(just for my own education.)

--

r b-j rbj@...

"Imagination is more important than knowledge."

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Re: bandlimited interpolation of non-uniformly-spacedsamples?

2009-11-29T21:50:42Z

Hi Spencer

> I've written an implementation of Iannis Xenakis's Dynamic Stochastic
> Synthesis as an object for Pure Data and Max/MSP.
...
> Implementations that I'm familiar with have all just interpolated
> linearly, but I'm interested in exploring other interpolation methods,
> especially looking to generate bandlimited waveforms to eliminate
> aliasing problems.

I have a little familiarity with the DSS method but havn't implemented it.
If I recall correctly it's a bit like generating triangle waveforms where
length of each line segment is randomised and the amplitude of each endpoint
(direction change) is also randomised (at least that's a sufficient
definition for discussing bandlimited synthesis).

You might not actually want to use bandlimited interpolation of the
non-uniformly spaced line-segment endpoints, but to continue to use line
segments and to just synthesize the discontinuities between line segments in
a band limited way -- that would give you alias free DSS.

This can be achieved using a BLAM (band limited ramp), a variation on a
miblep.
http://www.experimentalscene.com/articles/minbleps.php

Try these Google queries:
BLEP
minblep blam

In essence it means mixing in a little wave-form at each discontinuity which
cancels the ailiasing.

Other things you could try:

- synthesising the line-segments at a higher sample rate (oversampling) and
then using band-limited sample rate conversion to get rid of (some) of the
aliasing.

- use a polynomial spline to interpolate a smooth curve at the direction
change (not bandlimited, but better than having a sharp discontinuity). you
could combine this with oversampling.

HTH

Ross.

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Re: bandlimited interpolation of non-uniformly-spaced samples?

2009-11-28T15:25:03Z

On Fri, Nov 27, 2009 at 2:47 PM, Sampo Syreeni <decoy@...> wrote:

> On 2009-11-27, Spencer Russell wrote:
>
>> Implementations that I'm familiar with have all just interpolated
>> linearly, but I'm interested in exploring other interpolation methods,
>> especially looking to generate bandlimited waveforms to eliminate
>> aliasing problems.
>
> There is a reason why those implementations use linear interpolation and
> risk aliasing in the process. That is because there is no general theory
> of higher order polynomial interpolation for completely arbitrary nodes.

Right, so we've got to make it up as we go.

Here's what I propose:
1. Every polynomial interpolator can be written as a convolution
operator with a piecewise continuous polynomial kernel. This gives us
the method to calculate output.
2. Definition and spectrum of uniform and non-uniform Dirac Delta combs
3. Design of polynomial interpolators for anti-aliasing
--Write the spectral response of the output as product of the
non-uniform Dirac Delta comb and spectrum of the polynomial
interpolator kernel, which you can modify to eliminate aliasing.

I'm just getting back from vacation, so I've got time to shoot the
shit and solve a math problem, at least until Monday. How about you?

I've got the theory of uniform sampling down and I've created a way of
creating an antialiasing table read (tabread4a~ as I called it) for
Pd. So, we're using the same platform. It has not been released
except as code on pd-list, as I don't care enough about it. I would
like to make a better optimized version and have been working on the
theory sporadically to reduce it to O(log N) calculations.
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Re: bandlimited interpolation of non-uniformly-spaced samples?

2009-11-28T03:41:18Z

On 2009-11-27, robert bristow-johnson wrote:

> an Nth-order polynomial increases your bandlimit frequency (say it's
> just below Nyquist) by a factor of N, but when it's non-uniformly
> sampled, i am not sure what the bandlimit frequency is.

Picture the time-variable Lagrange polynomial for {(0,0), (1/2+f(x),1),
(1,1)}. If you take the result to be one cycle of a periodic waveform,
it's easy to see that what we have is a lot like phase modulation. That
we know to be bandunlimited from the start. Another way to see the
problem is to consider the definition of the Lagrange polynomial. If you
let the basis points depend on time as well, you can see that the
Lagrange interpolant is no longer polynomial in time, but actually a
rather higher order fraction.

> Sampo, i realize that what i'm saying is very basic and i'm sure you
> know about it, but i am not sure at all what you mean. it seems to me
> that Lagrange interpolation would work (not be terribly efficient)
> from what i can tell is described by Spencer.

Yeah, I was being imprecise. My point is that Spencer's points are
moving, and that is a big problem if you want get bandlimited waveforms.
--
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Re: bandlimited interpolation of non-uniformly-spaced samples?

2009-11-27T20:06:48Z

On Nov 27, 2009, at 3:47 PM, Sampo Syreeni wrote:

> On 2009-11-27, Spencer Russell wrote:
>
>> Implementations that I'm familiar with have all just interpolated
>> linearly,

then it must be only considering the adjacent two points each side of
your interpolation location.

>> but I'm interested in exploring other interpolation methods,
>> especially looking to generate bandlimited waveforms to eliminate
>> aliasing problems.
>
> There is a reason why those implementations use linear
> interpolation and
> risk aliasing in the process. That is because there is no general
> theory
> of higher order polynomial interpolation for completely arbitrary
> nodes.

then what is Lagrange interpolation? i think it requires that your
nodes be ordered (maybe it doesn't), but they're still arbitrary.

>
> Least of all *any* theory that would guarantee bandlimited output
> given
> an arbitrary set of nodes (polynomials are of course bandlimited,
> but as
> soon as you start doing anything but summin them together, i.e.
> modulating their endpoints as in a spline, you can get just about any
> spectral feature).

an Nth-order polynomial increases your bandlimit frequency (say it's
just below Nyquist) by a factor of N, but when it's non-uniformly
sampled, i am not sure what the bandlimit frequency is. maybe its 1/
(2T) where T is the minimum of the spacing between all samples under
consideration.

> Nor is there a proper analytic solution to how a
> finite number of Markov random variables on the real line which are
> bounded not to cross each other asymptotically behave, AFAIK.
>
>> Can anyone point me towards some efficient methods for interpolating
>> non-uniformly spaced samples to generate bandlimited (or
>> quasi-bandlimited) waveforms?
>
> No, they cannot, because such a theory does not exist.

hmmmmmmm.

> In fact I seem to
> remember that sort of thing can be shown to be impossible at any
> finite
> order of the interpolating polynomial, and even in the case of a
> finite,
> rather well-behaved basis of infinite order polynomial
> approximations to
> real functions, the search problem can be proven NP-complete. (As
> usual,
> no references. But you can check e.g. the literature on finite
> approximations to arbitrary well-behaved functions, in the L^1 norm.)

i just may not be aware of exactly what Sampo and Spencer are talking
about, but if you have non-uniformly-spaced but *known* locations of
the samples (as well as known sample values), there *are* straight-
forward methods of stringing an Nth-order polynomial through N+1
points in general. it's not efficient, requires solving N+1 linear
equations for N+1 coefficients. using gaussian elimination would
make it an O(N^2) problem. letting N be odd, you would likely
include (N+1)/2 points each side of your interpolation location, even
if they're not uniformly space (the (N+1)/2 points to the left might
take up a much larger time interval than on the right). there are
different ways of doing this, but if you want to go straight through
the N+1 points, it's called Lagrange interpolation.

Sampo, i realize that what i'm saying is very basic and i'm sure you
know about it, but i am not sure at all what you mean. it seems to
me that Lagrange interpolation would work (not be terribly efficient)
from what i can tell is described by Spencer.

--

r b-j rbj@...

"Imagination is more important than knowledge."

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Re: bandlimited interpolation of non-uniformly-spacedsamples?

2009-11-27T16:29:42Z

I'm not sure how many points per cycle you're thinking of but you could do
it by representing each point with a sinc impulse with amplitude equal to
the difference in height from the previous point, integrating them to give a
series of bandlimited steps, then integrating again to get bandlimited
linear segments (or using a tracking lowpass filter on the original
bandlimited step wave). You'd probably have some DC offset issues,
especially if all the points decided to go the same way!

I suspect that such a waveform wouldn't sound very interesting, but neither
does a sawtooth...

Just my 2c.

Regards,

Steven Cook.
http://www.spcplugins.com/

----- Original Message -----
From: "Spencer Russell" <spencer.f.russell@...>
To: "A discussion list for music-related DSP" <music-dsp@...>
Sent: Friday, November 27, 2009 7:15 PM
Subject: [music-dsp] bandlimited interpolation of
non-uniformly-spacedsamples?

> I've written an implementation of Iannis Xenakis's Dynamic Stochastic
> Synthesis as an object for Pure Data and Max/MSP.
>
> For those unfamiliar, the basic idea is to describe a single cycle of
> a quasi-periodic waveform as a set of points (x_n, y_n), each of which
> is on a random walk. So each time a cycle of the waveform is copied to
> the output buffer, each breakpoint moves a little, and then the next
> cycle is calculated.
>
> Implementations that I'm familiar with have all just interpolated
> linearly, but I'm interested in exploring other interpolation methods,
> especially looking to generate bandlimited waveforms to eliminate
> aliasing problems.
>
> Can anyone point me towards some efficient methods for interpolating
> non-uniformly spaced samples to generate bandlimited(or
> quasi-bandlimited) waveforms?
>
> C/C++ source code would be great, but not necessary.
>
> Thanks!
>
> -spencer
> --
> dupswapdrop -- the music-dsp mailing list and website:
> subscription info, FAQ, source code archive, list archive, book reviews,
> dsp links
> http://music.columbia.edu/cmc/music-dsp
> http://music.columbia.edu/mailman/listinfo/music-dsp

Re: bandlimited interpolation of non-uniformly-spaced samples?

2009-11-27T12:47:05Z

On 2009-11-27, Spencer Russell wrote:

> Implementations that I'm familiar with have all just interpolated
> linearly, but I'm interested in exploring other interpolation methods,
> especially looking to generate bandlimited waveforms to eliminate
> aliasing problems.

There is a reason why those implementations use linear interpolation and
risk aliasing in the process. That is because there is no general theory
of higher order polynomial interpolation for completely arbitrary nodes.
Least of all *any* theory that would guarantee bandlimited output given
an arbitrary set of nodes (polynomials are of course bandlimited, but as
soon as you start doing anything but summin them together, i.e.
modulating their endpoints as in a spline, you can get just about any
spectral feature). Nor is there a proper analytic solution to how a
finite number of Markov random variables on the real line which are
bounded not to cross each other asymptotically behave, AFAIK.

> Can anyone point me towards some efficient methods for interpolating
> non-uniformly spaced samples to generate bandlimited(or
> quasi-bandlimited) waveforms?

No, they cannot, because such a theory does not exist. In fact I seem to
remember that sort of thing can be shown to be impossible at any finite
order of the interpolating polynomial, and even in the case of a finite,
rather well-behaved basis of infinite order polynomial approximations to
real functions, the search problem can be proven NP-complete. (As usual,
no references. But you can check e.g. the literature on finite
approximations to arbitrary well-behaved functions, in the L^1 norm.)
--
Sampo Syreeni, aka decoy - decoy@..., http://decoy.iki.fi/front
+358-50-5756111, 025E D175 ABE5 027C 9494 EEB0 E090 8BA9 0509 85C2
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bandlimited interpolation of non-uniformly-spaced samples?

2009-11-27T11:15:34Z

I've written an implementation of Iannis Xenakis's Dynamic Stochastic
Synthesis as an object for Pure Data and Max/MSP.

For those unfamiliar, the basic idea is to describe a single cycle of
a quasi-periodic waveform as a set of points (x_n, y_n), each of which
is on a random walk. So each time a cycle of the waveform is copied to
the output buffer, each breakpoint moves a little, and then the next
cycle is calculated.

Implementations that I'm familiar with have all just interpolated
linearly, but I'm interested in exploring other interpolation methods,
especially looking to generate bandlimited waveforms to eliminate
aliasing problems.

Can anyone point me towards some efficient methods for interpolating
non-uniformly spaced samples to generate bandlimited(or
quasi-bandlimited) waveforms?

C/C++ source code would be great, but not necessary.

Thanks!

-spencer
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Positions at Computer Music Journal

2009-11-25T15:57:31Z

[Apologies for cross-postings.]

Positions at Computer Music Journal

Computer Music Journal, MIT Press's definitive quarterly on computer
music, invites applications for two positions: (1) an assistant editor
for manuscript editing, and (2) an intern or editorial consultant for
news and announcements. The positions have no geographical limitations,
but native English fluency is required. Members of underrepresented
groups are encouraged to apply, as are all other qualified applicants.
For questions, please contact the Editor at cmj@....

Applications and inquiries are accepted by email; send to
cmj@... with "[CMJ application]" in the subject line.
Hard-copy applications are also accepted; send to Editor, Computer Music
Journal, 2550 9th Street #207 B, Berkeley, CA 94710, USA. (Materials
will not be returned.)

Your application should include:

1. A brief letter explaining your interest and qualifications and
specifying which of the two positions is of interest.
2. A curriculum vitae, including education, any publications, any
editorial experience, a list of research areas of computer music in
which you have competence or expertise, and a summary of mathematical
and technical skills as well as musical training.
3. Names, phone numbers, addresses, and email addresses of references.
4. One or more samples of your writing (not necessarily published) on
any topic related to computer music.
5. The date when you can start, preferably by January 2, 2010.

We recommend that applications be submitted by December 4, 2009. Review
of applications will continue until the position is filled.

Detailed job descriptions follow.

1. Assistant Editor for Manuscript Editing

Responsibilities focus on editing and preparing peer-reviewed
manuscripts for submission to MIT Press for publication, including some
communication with authors on technical topics regarding their articles.
(This position does not include managing the peer-review process or
seeking substantial revisions from authors.) The position entails
approximately 60-120 hours per quarterly issue. Detailed requirements
are listed below.

Applicants should demonstrate both eagerness for success as an editor
and a commitment to computer music. Because of the technical nature of
many CMJ manuscripts, preference will be given to applicants who have a
background in engineering, computer science, or related research areas.
The ideal applicant will be able to comprehend and evaluate manuscripts
on a wide variety of topics, so that he or she can edit each manuscript
that has undergone the peer-review process to conform with the CMJ style
and MIT Press publication standards. We are seeking a native speaker of
English who has top-notch language skills and an eye for detail.

The current editor for manuscript editing, Colby Leider, is stepping
down for lack of time, owing to increased responsibilities in his
academic department. We are sorry to see him leave after over a
decade with the Journal. However, it is gratifying to witness his
successful career development. He will be available to assist the new
editor during the transition.

Location: May live anywhere in the world
Requirements:
- Broad and deep knowledge of computer music
- Master's degree or higher
- At least a bachelor's degree or equivalent in a technical field (e.g.,
computer science, engineering, mathematics, sciences)
- Musical training
- Native English fluency and superb editing skills
- Punctuality and attention to detail
- Familiarity with Microsoft Word, LaTeX, and Adobe Photoshop
Helpful:
- Familiarity with foreign languages
- Editing experience
- Familiarity with the Chicago Manual of Style
Responsibilities:
- Communicate with CMJ Editor, authors, managing editor, and copyeditor
- Assemble materials (documents, figures, captions, etc.) for each
accepted manuscript, starting with the author’s final version.
- Edit each manuscript for conformance to existing CMJ style
- Email authors regarding editing questions and copyright matters
- Collate multiple versions of manuscripts into a final document
- Check bibliographic items for accuracy, completeness, and format;
supply missing details
- Deliver edited manuscripts on time for publication
- Attend the annual CMJ editorial summit meeting (by phone, Skype, or
in person)
- Cultivate future manuscript submissions from potential authors
Compensation:
A small salary is available through a contract with MIT Press; contact
the Editor.

2. Intern or Editorial Consultant for News and Announcements

Responsibilities consist of gathering and carefully editing items for
the Announcements and News sections near the beginning of the Journal.
The position entails an average of approximately 20 hours per quarterly
issue. Detailed requirements are listed below.

The first year of the position will be treated as an internship, unless
the appointee is exceptionally well qualified and has already achieved
widespread recognition for his or her career in computer music. In that
case, the appointee's title will be Editorial Consultant; otherwise, it
will initially be Editorial Intern.

This is a volunteer position: there is no financial compensation, but
the appointee will gain experience in publishing, and will be able to
participate in the "cutting edge" of the computer music field. The
appointee will be listed by name on the inside front cover of each issue
on which he or she works, and on the Journal's Web site.

Applicants should demonstrate both eagerness for success as an editor
and a commitment to computer music. Interests might include composition,
performance, engineering, or research topics in the field. We are
seeking a native speaker of English who has top-notch language skills
and an eye for detail. A solid knowledge of the computer music field
will be invaluable in ascertaining the priority of potential news items,
as well as in editing the chosen items.

The current editor of of news and announcements, Lonce Wyse, will be
taking on other responsibilities with the Journal, and he will also be
available to assist the appointee during the transition.

Location: May live anywhere in the world
Requirements:
- Good knowledge of the computer music field
- Bachelor's degree or higher
- Native English fluency and excellent writing and editing skills
- Punctuality and attention to detail
- Familiarity with Microsoft Word
Helpful:
- Familiarity with other languages (German, French, Spanish, Italian,
Japanese, etc.)
- Editing experience
- Familiarity with the Chicago Manual of Style
- Familiarity with Adobe Photoshop
Responsibilities:
- Receive potential news items and announcements from Editor
- Collect other items from Web, mailing lists, etc.
- Send collected, unedited items to Editor for approval
- Edit each item
- Send edited items to Editor and Managing Editor as Microsoft Word files
- Collect several pieces of artwork to accompany news items (such as
photos, image files from the Web, or hard-copy illustrations in brochures)
- Attend the annual CMJ editorial summit meeting (by phone, Skype, or
in person)

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Re: Polyphase Downsampling/Upsampling By 4, 8, 16...

2009-11-19T13:35:39Z

Hi Laurent,

Many thanks for your answer. I was hoping that it might be possible to use
just two polyphase filters for 4x downsampling but at least three filters is
an improvement on my current arrangement. Thanks also for the link to your
HIIR library, which I've downloaded and will study.

Kind Regards,

Steven Cook.
http://www.spcplugins.com/

> You're right, but filters for both stages must be distinct
> ones :
>
> temp1 = filter1 (in1, in2)
> temp2 = filter1 (in3, in4)
> out = filter2 (temp1, temp2)
>
> You might be interested by the HIIR library -
> <http://ldesoras.free.fr/prod.html#src_hiir> . The readme
> file explains how to design cascaded filters for upsampling
> or downsampling by a power-of-two ratio.
>
>
> --
> Laurent de Soras | Ohm Force
> DSP developer & Software designer | Digital Audio Software
> http://ldesoras.free.fr | http://www.ohmforce.com

Re: Polyphase Downsampling/Upsampling By 4, 8, 16...

2009-11-19T03:36:57Z

Steven Cook wrote:
>
> My question is: I would like to try using filters like this to downsample by
> a factor of four (or more) but I'm unsure how to use two-input-one-output
> filters in this way. Can the filters be used like this? (in1, in2, in3, in4
> are my 4 (oversampled) input samples)
>
> temp1 = filter(in1, in2)
> temp2 = filter(in3, in4)
> out = filter(temp1, temp2)

You're right, but filters for both stages must be distinct
ones :

temp1 = filter1 (in1, in2)
temp2 = filter1 (in3, in4)
out = filter2 (temp1, temp2)

You might be interested by the HIIR library -
<http://ldesoras.free.fr/prod.html#src_hiir> . The readme
file explains how to design cascaded filters for upsampling
or downsampling by a power-of-two ratio.

--
Laurent de Soras | Ohm Force
DSP developer & Software designer | Digital Audio Software
http://ldesoras.free.fr | http://www.ohmforce.com

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Re: Polyphase Downsampling/Upsampling By 4, 8, 16...

2009-11-19T02:39:59Z

Hi Nigel,

Thanks for your continuing input. I have read your webpage and it is a very
clear and well presented explanation of downsampling. I am fairly sure that
I understand the basics of downsampling/upsampling, however, and my question
is one of implementation.

The filter design I am asking about is the IIR half-band filter from "The
Quest for the Perfect Resampler" (Chapter 4) by Laurent De Soras.
http://ldesoras.free.fr/doc/articles/resampler-en.pdf (Which is referenced
[9] as originally coming from a now defunct web link) This combines the
filtering and decimation into one operation (presumably to improve
efficiency), so it has two inputs but only one output. I think filters of
this type are also described in the Music DSP archives.

My question is: I would like to try using filters like this to downsample by
a factor of four (or more) but I'm unsure how to use two-input-one-output
filters in this way. Can the filters be used like this? (in1, in2, in3, in4
are my 4 (oversampled) input samples)

temp1 = filter(in1, in2)
temp2 = filter(in3, in4)
out = filter(temp1, temp2)

Kind regards,

Steven Cook.
http://www.spcplugins.com/

----- Original Message -----
From: "Nigel Redmon" <earlevel@...>
To: "A discussion list for music-related DSP" <music-dsp@...>
Sent: Thursday, November 19, 2009 3:29 AM
Subject: Re: [music-dsp] Polyphase Downsampling/Upsampling By 4, 8, 16...

> Hi Steven,
>
> Something's not right here... first, I encourage you to read my web
> article, but basically...
>
> For downsampling, you need to reduce the bandwidth of the signal, to avoid
> aliasing in the result. For instance, if you want to downsample by a
> factor of two, you need to apply a lowpass filter with a cutoff low enough
> to ensure no aliasing at the new rate. That means the lowpass filter
> corner frequency should be well less than one-fourth the sample rate to
> make sure the highs are rolled of sufficiently by half the new sample
> rate. At that point you have a lowpassed version of your signal, but still
> at the original sample rate. It's "oversampled" at this point--the sample
> rate is twice as high as needed to contain the reduced-bandwidth signal.
> So, you just throw away every-other sample.
>
> That's the gist of downsampling--filter it so that it will fit the new
> bandwidth, then throw away what you no longer need, to arrive at the new
> sample rate.
>
> So, the "filter()" function in your code, I presume, rolls the filter and
> the decimator into one--typical of a polyphase filter. But there are only
> two inputs for every output, meaning it's either a horrendously bad FIR
> filter, or it's an IIR (or FIR) with one more input than needed and keeps
> its own history (in which case you couldn't call it the way you are
> without some way to make separate instances of it--obviously not possible
> in your code example). Neither makes sense. (Maybe one input is a pointer
> to a buffer, and the other is length?)
>
> Could you elaborate more on "filter()"? Is that a library function for
> which you have no code, something you have code for, or a flow diagram
> you're trying to implement as code?
>
> Nigel
>
>
>
> On Nov 18, 2009, at 2:51 PM, Steven Cook wrote:
>
>> Thanks Nigel.
>>
>> I've been thinking very hard about your answer but don't understand...
>> I'm
>> thinking of the type of polyphase filter (decimator?) which has two
>> inputs
>> and one output. I assume that to downsample by a factor of 2, it's simply
>> a
>> question of doing: out = filter(in1, in2). When downsampling by a factor
>> of
>> 4, do I process my 4 input samples (in1, in2, in3, in4) like:
>>
>> temp1 = filter(in1, in2);
>> temp2 = filter(in3, in4);
>> out = filter(temp1, temp2);
>>
>> Or some other way?
>>
>> Regards,
>>
>> Steven Cook.
>> http://www.spcplugins.com/
>>
>> ----- Original Message -----
>> From: "Nigel Redmon" <earlevel@...>
>> To: "A discussion list for music-related DSP"
>> <music-dsp@...>
>> Sent: Wednesday, November 18, 2009 8:04 PM
>> Subject: Re: [music-dsp] Polyphase Downsampling/Upsampling By 4, 8, 16...
>>
>>
>>> Re-reading your question, Steven, I think the answer to your question is
>>> "use a circular output buffer between stages", so you'll have enough
>>> samples saved to run an iteration of the next stage.
>>>
>>>
>>> On Nov 17, 2009, at 12:41 PM, Steven Cook wrote:
>>>
>>>> Hi,
>>>>
>>>> I'm experimenting with downsampling by a factor of 4 (8, 16...). I'd
>>>> like
>>>> to
>>>> try polyphase downsampling but can't figure out how to 'wire together'
>>>> 2
>>>> (or
>>>> would it be 3?) polyphase filters between 4 (8. 16...) input samples
>>>> and
>>>> 1
>>>> output sample. Would someone give me a clue? While I'm here, does it
>>>> matter
>>>> in which order the 2 input samples are read into a polyphase filter?
>>>>
>>>> Regards,
>>>>
>>>> Steven Cook.
>>>> http://www.spcplugins.com/
>>>
>
> --
> dupswapdrop -- the music-dsp mailing list and website:
> subscription info, FAQ, source code archive, list archive, book reviews,
> dsp links
> http://music.columbia.edu/cmc/music-dsp
> http://music.columbia.edu/mailman/listinfo/music-dsp

Re: Polyphase Downsampling/Upsampling By 4, 8, 16...

2009-11-18T19:29:00Z

Hi Steven,

Something's not right here... first, I encourage you to read my web article, but basically...

For downsampling, you need to reduce the bandwidth of the signal, to avoid aliasing in the result. For instance, if you want to downsample by a factor of two, you need to apply a lowpass filter with a cutoff low enough to ensure no aliasing at the new rate. That means the lowpass filter corner frequency should be well less than one-fourth the sample rate to make sure the highs are rolled of sufficiently by half the new sample rate. At that point you have a lowpassed version of your signal, but still at the original sample rate. It's "oversampled" at this point--the sample rate is twice as high as needed to contain the reduced-bandwidth signal. So, you just throw away every-other sample.

That's the gist of downsampling--filter it so that it will fit the new bandwidth, then throw away what you no longer need, to arrive at the new sample rate.

So, the "filter()" function in your code, I presume, rolls the filter and the decimator into one--typical of a polyphase filter. But there are only two inputs for every output, meaning it's either a horrendously bad FIR filter, or it's an IIR (or FIR) with one more input than needed and keeps its own history (in which case you couldn't call it the way you are without some way to make separate instances of it--obviously not possible in your code example). Neither makes sense. (Maybe one input is a pointer to a buffer, and the other is length?)

Could you elaborate more on "filter()"? Is that a library function for which you have no code, something you have code for, or a flow diagram you're trying to implement as code?

Nigel

On Nov 18, 2009, at 2:51 PM, Steven Cook wrote:

> Thanks Nigel.
>
> I've been thinking very hard about your answer but don't understand... I'm
> thinking of the type of polyphase filter (decimator?) which has two inputs
> and one output. I assume that to downsample by a factor of 2, it's simply a
> question of doing: out = filter(in1, in2). When downsampling by a factor of
> 4, do I process my 4 input samples (in1, in2, in3, in4) like:
>
> temp1 = filter(in1, in2);
> temp2 = filter(in3, in4);
> out = filter(temp1, temp2);
>
> Or some other way?
>
> Regards,
>
> Steven Cook.
> http://www.spcplugins.com/
>
> ----- Original Message -----
> From: "Nigel Redmon" <earlevel@...>
> To: "A discussion list for music-related DSP" <music-dsp@...>
> Sent: Wednesday, November 18, 2009 8:04 PM
> Subject: Re: [music-dsp] Polyphase Downsampling/Upsampling By 4, 8, 16...
>
>
>> Re-reading your question, Steven, I think the answer to your question is
>> "use a circular output buffer between stages", so you'll have enough
>> samples saved to run an iteration of the next stage.
>>
>>
>> On Nov 17, 2009, at 12:41 PM, Steven Cook wrote:
>>
>>> Hi,
>>>
>>> I'm experimenting with downsampling by a factor of 4 (8, 16...). I'd like
>>> to
>>> try polyphase downsampling but can't figure out how to 'wire together' 2
>>> (or
>>> would it be 3?) polyphase filters between 4 (8. 16...) input samples and
>>> 1
>>> output sample. Would someone give me a clue? While I'm here, does it
>>> matter
>>> in which order the 2 input samples are read into a polyphase filter?
>>>
>>> Regards,
>>>
>>> Steven Cook.
>>> http://www.spcplugins.com/
>>

Re: Polyphase Downsampling/Upsampling By 4, 8, 16...

2009-11-18T14:51:02Z

Thanks Nigel.

I've been thinking very hard about your answer but don't understand... I'm
thinking of the type of polyphase filter (decimator?) which has two inputs
and one output. I assume that to downsample by a factor of 2, it's simply a
question of doing: out = filter(in1, in2). When downsampling by a factor of
4, do I process my 4 input samples (in1, in2, in3, in4) like:

temp1 = filter(in1, in2);
temp2 = filter(in3, in4);
out = filter(temp1, temp2);

Or some other way?

Regards,

Steven Cook.
http://www.spcplugins.com/

----- Original Message -----
From: "Nigel Redmon" <earlevel@...>
To: "A discussion list for music-related DSP" <music-dsp@...>
Sent: Wednesday, November 18, 2009 8:04 PM
Subject: Re: [music-dsp] Polyphase Downsampling/Upsampling By 4, 8, 16...

> Re-reading your question, Steven, I think the answer to your question is
> "use a circular output buffer between stages", so you'll have enough
> samples saved to run an iteration of the next stage.
>
>
> On Nov 17, 2009, at 12:41 PM, Steven Cook wrote:
>
>> Hi,
>>
>> I'm experimenting with downsampling by a factor of 4 (8, 16...). I'd like
>> to
>> try polyphase downsampling but can't figure out how to 'wire together' 2
>> (or
>> would it be 3?) polyphase filters between 4 (8. 16...) input samples and
>> 1
>> output sample. Would someone give me a clue? While I'm here, does it
>> matter
>> in which order the 2 input samples are read into a polyphase filter?
>>
>> Regards,
>>
>> Steven Cook.
>> http://www.spcplugins.com/
>
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Re: Polyphase Downsampling/Upsampling By 4, 8, 16...

2009-11-18T12:04:48Z

Re-reading your question, Steven, I think the answer to your question is "use a circular output buffer between stages", so you'll have enough samples saved to run an iteration of the next stage.

On Nov 17, 2009, at 12:41 PM, Steven Cook wrote:

> Hi,
>
> I'm experimenting with downsampling by a factor of 4 (8, 16...). I'd like to
> try polyphase downsampling but can't figure out how to 'wire together' 2 (or
> would it be 3?) polyphase filters between 4 (8. 16...) input samples and 1
> output sample. Would someone give me a clue? While I'm here, does it matter
> in which order the 2 input samples are read into a polyphase filter?
>
> Regards,
>
> Steven Cook.
> http://www.spcplugins.com/

Re: Polyphase Downsampling/Upsampling By 4, 8, 16...

2009-11-18T11:58:53Z

Not quite sure of what you're asking--when you say "'wire together' 2 (or would it be 3?)", are you talking about multistage downsampling? It sort of seems like that's what you are saying. But then again, if you understand one stage, subsequent stages would be the same thing, so I'm not sure what you're asking.

If you want to understand the nuts and bolts of why it works (I think such an understanding inherently answers a lot of the usual implementation questions), try this:

http://www.earlevel.com/Digital%20Audio/RateConversion.html

On Nov 17, 2009, at 12:41 PM, Steven Cook wrote:

Polyphase Downsampling/Upsampling By 4, 8, 16...

2009-11-17T12:41:06Z

Hi,

I'm experimenting with downsampling by a factor of 4 (8, 16...). I'd like to
try polyphase downsampling but can't figure out how to 'wire together' 2 (or
would it be 3?) polyphase filters between 4 (8. 16...) input samples and 1
output sample. Would someone give me a clue? While I'm here, does it matter
in which order the 2 input samples are read into a polyphase filter?

Regards,

Steven Cook.
http://www.spcplugins.com/

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[admin] music-dsp FAQ

2009-11-15T04:00:01Z

Re: Fixed-point effects lib? Linux friendly floating-point DSP?

2009-11-10T11:58:50Z

On Thursday 08 October 2009, Amusing Muses wrote:

> Hi I'm investigating the Blackfin as a platform for a music DSP
> toolkit, unfortunately all the examples for simple effects like
> choruses & flangers I can find utilize floating-point
> arithmetic--which cripples their performance on the Blackfin because
> it has no FPU.
>
> Does anyone know of any open source DSP effects packages that use
> fixed-point arithmetic? Could save me a bunch of time rewriting these
> effects myself... or should I just use a floating-point DSP? I'm
> looking for something that is supported well by open source tools
> (gcc, u-boot, Linux, etc) and that is low-cost enough for a high
> volume consumer product... The Blackfin meets these requirements
> except the lack of floating-point is turning out to be a hassle. Any
> suggestions?
>
> Thanks.

Hi,

you might also take a look at Intel Atom boards.. They are rather cheap and
can do some number crunching :)

Flo

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Re: intellectual properties in softwaredevelopment

2009-11-03T12:35:32Z

i lost, visit me in jail please ... for all the young developers out
there, never bite the hand that feeds you ...

;)

Am 08.10.2009 um 19:30 schrieb robert bristow-johnson:

>
> On Oct 8, 2009, at 12:36 PM, bastian.schnuerle wrote:
>
>> sorry guys, my last statement was also not a very professional.
>
>
> this is a professional list? i'm just here to have fun.
>
>
> onward to NYC and AES. see ya guys.
>
> --
>
> r b-j rbj@...
>
> "Imagination is more important than knowledge."
>
>
>
>
> --
> dupswapdrop -- the music-dsp mailing list and website:
> subscription info, FAQ, source code archive, list archive, book
> reviews, dsp links
> http://music.columbia.edu/cmc/music-dsp
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Re: platform choice

2009-11-02T18:32:45Z

On Nov 2, 2009, at 5:41 AM, Michael Gogins wrote:
> C++ does not have the degree of linear algebra support as Fortran, but
> it does now have what could be called _decent_ support in the form of
> libraries, foremost Eigen and Gmm++.

(Mike, I'm not directing this at you--you probably know it already--
but just for the benefit of misc. lurkers...)

Another issue for C++ vs FORTRAN for heavy number crunching is the way
C++ deals with numbers (pairwise evaluation with operator overloading,
resulting in intermediate storage in vector calculations, etc.). This
can be addressed in C++ with template meta programming (Blitz, etc.).

Anyway, just mentioning this for people who wonder why anyone in their
right mind would still use FORTRAN (FORmula TRANslation--that's what
it was made for ;-)

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Re: platform choice

2009-11-02T18:13:34Z

On Nov 2, 2009, at 1:18 AM, Richard Dobson wrote:
>>
>> It is really nothing more than a convention.
>
> Hmm, not really. In real life, computers depend on integer types. We
> all
> know that an 8bit number can represent up to 256 distinct values; that
> would be 0-255, not 1-256. If we want to count to ten using one digit,
> it has to be 0-9, not 1-10.

I believe it comes from traditional convention in mathematics (where
the first position in a sequence would be referred to as the first/
1st, not the zeroeth/0th, the last value of a sequence of length 10
would be the 10th, etc.). I suppose this carried over to early
computer languages (I haven't done a survey on that), and I suppose
the benefits of zero-based array become obvious as computing matured
(Pascal was definable, right?). FORTRAN was my first high-level
language in college, and in my youth I programmed automated-test
systems in a FORTRAN-based language. And yes, it's really annoying to
work in 1-based systems, especially when get accustomed to performance
optimizations such as wrapping around arrays with modulo arithmetic
(such as AND-masking the index in power-of-2 length array).

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Re: Equalizer Coefficients

2009-11-02T10:08:32Z

Hi Bruno,

for your code:
> double output = 0
> for each point P of the equalizer do:
> output += applyFilter(intput, P.frequency ) * P.Volume
> done
> return output;
you need normal low/hi/band pass filters, but you'll not be satisfied
with the resulting curves, especially not if attenuate a band's volume.

Better use the peak and shelf filters and do something like:
double output = input;
input = LowShelfe(input);
input = HighShelfe(input);
for(all bands)
input = Peak(input);

You should find examples on musicdsp.org

> Of course, my main problem is how to calculate the Q coefficient ?

The question is what you want to create.
(1) an EQ with adjacent bands or
(2) a conventional parametric EQ with random f0 and Q for N bands.
If you want to set random mid frequencies, the cutoff frequencies won't
match so that fcUpper[n] == fcLower[n+1].
If you'd like to have an EQ with adjacent bands, you'll have to set
'split points' to insert a new band and the mid frequencies will follow.

Best Regards,

Thomas
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