[music-dsp] JOB: Mobile Software Developer @ Native Instruments

2016-11-12 Thread Kevin Dixon
Hi all,

Wanted to announce an opening on my team at Native Instruments based in Los
Angeles

https://www.native-instruments.com/en/career-center/los-angeles/product-creation-development/mobile-software-developer/

We need strong C++ developers with passion for music and DSP!

Thanks

Kevin Dixon
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Re: [music-dsp] ARM for DSP

2012-09-25 Thread Kevin Dixon
Here at work, we do all of our DSP for ARM in fixed point, though we
have tested the Cortex A9 with floating point builds of our
algorithms, and the performance is comparable, apparently the A9
series has improved the FPU significantly

(my 2cents)
-Kevin

On Tue, Sep 25, 2012 at 12:19 PM, Eric Brombaugh  wrote:
> I'm using a bare-bones development flow based on a text editor, GCC +
> utilities and a Black Magic Probe JTAG/SWD download pod. I've also used the
> ST-LINK V2 download feature that's built in to the Discovery boards, but the
> FOSS tools to support that hardware are a bit less reliable.
>
> I've tried a number of pre-built GCC toolchains - the CodeSourcery Lite
> version is free and readily available. ARM sponsors the Launchpad project
> which seems to work well and tracks a slightly newer code revision. The
> Black Magic Probe supports the GDB extended target protocol for realtime
> debug and I've found it to be very useful and reliable. Black Magic firmware
> is available to reflash the SWD hardware in the ST Discovery boards for
> improved functionality under Linux but I haven't taken that step.
>
> For those working on MacOS X or Windows, there are alternative development
> systems. ST recommends a number of commercial IDEs that charge "pro" prices
> but also provide crippled (code size limited) free versions. Full-capability
> IDEs based on Eclipse + GCC are available from Yagarto and CooCox and on
> Windows they support the ST-LINK SWD downloader.
>
> Eric
>
>
> On 09/25/2012 11:51 AM, Andy Farnell wrote:
>>
>>
>> How do you develop code for these Eric? What is your toolchain?
>>
>> best
>> Andy
>>
>> On Tue, Sep 25, 2012 at 09:43:09AM -0700, Eric Brombaugh wrote:
>>>
>>> The STM32F4 series parts are cheap, fast, powerful for their class.
>>> Not really on par with the Cortex A8 and Atom machines but great for
>>> embedded.
>>>
>>> I've got a little audio signal processing project based on the
>>> STM32F4 going now:
>>>
>>> http://ebrombaugh.studionebula.com/synth/stm32f4_codec/index.html
>>>
>>> I've been writing code on this for the last few weeks and have a few
>>> basic audio effects running on it. I've tried some frequency domain
>>> processing as well - 128-sample real floating point FFT/IFFT takes
>>> about 120us with the CPU running at max rated clock speed.
>>>
>>> Definitely worth checking out if you've got modest DSP to do on a budget.
>>>
>>> Eric
>>>
>>> On 09/25/2012 09:28 AM, Nigel Redmon wrote:

 I haven't had time to do much with it yet, but the STM32F4DISCOVERY
 board is bargain, with single precision floating point and DSP features
 (single cycle MAC, saturated arithmetic, SIMD), and a bunch of nice goodies
 on the baord, $14.55 at Mouser.
>
>
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Re: [music-dsp] maintaining musicdsp.org

2012-04-05 Thread Kevin Dixon
I would vote for a CAPTCHA... specifically recaptcha
http://www.google.com/recaptcha/whyrecaptcha

-Kevin

On Thu, Apr 5, 2012 at 1:03 PM, Bastian Schnuerle
 wrote:
> just did wordpress for a friend .. looks nice .. +1 ..
>
> Am 05.04.2012 um 21:50 schrieb douglas repetto:
>
>
>>
>> I think even Wordpress would work very well for the content on
>> musicdsp.org. I agree a full drupal site seems like overkill!
>>
>> douglas
>>
>> On 4/5/12 10:05 AM, Bjorn Roche wrote:
>>>
>>>
>>> On Apr 5, 2012, at 4:53 AM, Ross Bencina wrote:
>>>
 Hey Bjorn,

 On 5/04/2012 1:52 AM, Bjorn Roche wrote:
>
> Any thoughts about modernizing the whole thing with a fresh CMS?
> I think it would be easier to maintain, have built-in spam
> filters, and it would be easier to have multiple people do the
> work. Plus it would look more attractive. I don't think it would
> take much effort to redo the whole thing in, say, drupal.


 Have you ever set up a Drupal site? I have. It is not for
 small-time, non-commercial, low-maintenance overhead projects
 imho.
>>>
>>>
>>> Yes. Quite a few.
>>>
 Imho it would be a huge job to port the current site to Drupal and
 there is a lot of ongoing maintenance required to keep security
 patches up to date etc etc.
>>>
>>>
>>> Yes. The biggest problem is security updates. You are right: major
>>> PITA factor. This can be mitigated by a hosted solution, or a
>>> multi-site install where someone is already monitoring the site for
>>> security updates. But, at the end of the day, that might not be
>>> realistic.
>>>
 Doing the theme port alone would be a lot of work.
>>>
>>>
>>> I would not dream of porting the existing theme, but rather use a
>>> new, or built-in theme.
>>>
 Unless I'm completely out of touch it is really non-trivial to set
 up something like musicdsp.org in Drupal with adequate spam
 filtering. The standard Drupal capcha solution (Mollom) is not
 great -- in my experience it flags a lot of false positives (spam
 that isn't spam).
>>>
>>>
>>> Mollom sucks. Captchas alone catch the vast majority of spam. The
>>> rest can be handled with moderation.
>>>
 Anyway, this is really just a vote against Drupal for musicdsp.org,
 not against using a CMS.

 I actually think the current ad-hoc php solution is not so bad --
 but Bram knows more about these things than me.
>>>
>>>
>>> Recaptcha could be added to the existing site with fairly little
>>> effort, but there are other advantages to a CMS: they are easier to
>>> team-manage, organize, and they have a number of potentially useful
>>> features like taxonomies (giving the ability to tag and categorize
>>> algos by language and purpose for example.)
>>>
>>> bjorn
>>>
>>> - Bjorn Roche http://www.xonami.com Audio
>>> Collaboration http://blog.bjornroche.com
>>>
>>>
>>>
>>>
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>>>
>>
>> --
>> ... http://artbots.org
>> .douglas.irving http://dorkbot.org
>> .. http://music.columbia.edu/cmc/music-dsp
>> ...repetto. http://music.columbia.edu/organism
>> ... http://music.columbia.edu/~douglas
>>
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>
>
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Re: [music-dsp] good introductory microcontroller platform for audio tasks?

2011-04-21 Thread Kevin Dixon
As usual, thanks for the wealth of information.
I'm ultimately targeting a stompbox and/or rackmountable type of
device for gigging, so it needs to be roadworthy.
An ideal solution would be a breadboarding+dev kit where I could
develop the functionality I need, but also have a more permanent
"final device", e.g. in an enclosure, permanently soldered, etc.
In this case, I would appreciate keeping the unit cost relatively low,
since I plan on building several of the devices.

On a side note, are there any "repositories" ala SourceForge for open
source hardware/software solutions?

-Kevin
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[music-dsp] good introductory microcontroller platform for audio tasks?

2011-04-19 Thread Kevin Dixon
Hello list,
I'm experienced in developing DSP routines in C/C++ for desktop
computers, and have gotten my feet wet on TI MSP430, doing boring
things.
My project chiefly involves implementing a MIDI interface and also
implement an LFO.
I also need to read analog waveform, but sound quality is not really
important, I'm guessing an 8bit AD would suffice for the task. That
being said, in the future I will inevitably want to do some real DSP,
so families that can support 16-24bit AD at higher sample rates would
be preferable.

In summary these are my requirements:
-UART
-A to D
-D to A

My EE friend is recommending I go PIC, but the Arduino looks
promising, especially for fast return on effort :) I guess startup
cost is an issue too, I'd like to be up and running for about 50 USD.

Any thoughts/recommendations? Thanks,

-Kevin
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Re: [music-dsp] Regarding low pass filter for biquad programming

2011-01-27 Thread Kevin Dixon
Excuse me, in LPF mode, the bandwidth parameter is actually
Q/resonance -- try a value of 0.7
-Kevin

On Thu, Jan 27, 2011 at 1:08 PM, Kevin Dixon  wrote:
> it looks like you would call BiQuad_new(LPF, 0, 400, 44100, 0);
> where 400 is your corner frequency, and 44100 is your sample rate
> bandwidth is not applicable to this filter, so it can be 0.
>
> then using the returned biquad* struct, you would then iterate your
> samples, and call BiQuad(sample, *b) for each sample
>
> pseudo code:
>
> biquad * b = BiQuad_new(LPF, 0, 400, 44100, 0);
>
> for(int i = 0; i < len; i++)
> {
>     bufferOut[i] = BiQuad(bufferIn[i], b);
> }
>
> -Kevin
>
> On Tue, Jan 25, 2011 at 6:52 AM, Samuel Kitono  wrote:
>> Hi everyone,
>>
>> I am wondering if you guys have any example of creating a low pass
>> filter based on this www.musicdsp.org/files/biquad.c page.
>> --
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>>
>
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Re: [music-dsp] Regarding low pass filter for biquad programming

2011-01-27 Thread Kevin Dixon
it looks like you would call BiQuad_new(LPF, 0, 400, 44100, 0);
where 400 is your corner frequency, and 44100 is your sample rate
bandwidth is not applicable to this filter, so it can be 0.

then using the returned biquad* struct, you would then iterate your
samples, and call BiQuad(sample, *b) for each sample

pseudo code:

biquad * b = BiQuad_new(LPF, 0, 400, 44100, 0);

for(int i = 0; i < len; i++)
{
 bufferOut[i] = BiQuad(bufferIn[i], b);
}

-Kevin

On Tue, Jan 25, 2011 at 6:52 AM, Samuel Kitono  wrote:
> Hi everyone,
>
> I am wondering if you guys have any example of creating a low pass
> filter based on this www.musicdsp.org/files/biquad.c page.
> --
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Re: [music-dsp] What's the best way to render waveforms with accuarcy

2010-12-21 Thread Kevin Dixon
My 2 cents -- Logic, ProTools and many other professional DAWs perform
the task "Constructing Waveform Overview" when you import an audio
file, I can only assume they are performing some sort of
pre-calculated wave form, much as Ross describes here:

-Kevin

On Tue, Dec 21, 2010 at 9:00 AM, Ross Bencina
 wrote:
> Thomas Young wrote:
>>
>> When you have more samples (than pixels) each horizontal pixel column will
>> represent multiple samples, you will simply need a resampling algorithm to
>> determine the max and min for that column.
>
> You can also compute min and max for a set of fixed zoom levels and choose
> the appropriate one. Basically a 1d version of mipmapping:
> http://en.wikipedia.org/wiki/Mipmap
> This ensures that no matter what your zoom level is, your computational load
> is more or less constant.
>
>> As a begginer in DSP I can't figure out how to preserve the accuarcy
>> of the waveforms without having all the samples preserved.
>
> The way I've implemented it in the past is that when you're zoomed right in,
> I just read the real samples off disk for that small visible portion. You
> can thread it so the rendering is only performed once the read from disk is
> completed if you want to mask the latency, but disks are fast enough that
> you might be able to get away with performing the read inside the paint
> routine. Otherwise you can use a prefetch routine to make sure you usually
> have data near the visible region in memory.
>
> Ross.
>
>
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