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artmusicsouth

Because I am attached to an institution I am able to download the score for Ricercare Una Melodia for trumpet but it doesn't come with the electronics. I was wondering if someone has built a Max patch for this work.

jonathan-harveys-ricercare-una-melodia

Hello there, I did.
I've made a MaxMSp StandAlone, 3 years ago. I'll check it if it still works, and then I'll send it to you


Hello there, I'm sorry for the delay...
I had to rebuild, now it is a MAX collective.

please do tell me what do you think of it ( and if it works and is relliable)
and if you have any kind of comments please do so :)

https://dl.dropbox.com/u/1111139/Ricercare%20una%20melodia%20JHarvey%20-%20app%20JLF.zip


For some reason I missed you posting the patch. I just downloaded it but it keeps crashing on my machine.


UPDATE: Can open in Max just not standalone.

the patch is actually very easy to do, i suggest to do it, it's a good exercise, you can use just delay (tapin-tapout) objects or buffer record and play (or groove)
A.


I was wondering if I could see your Max Patch for Ricercare una Melodia.  I tried clicking the link, but it is expired.  

Hello Alexandria, probably i'll have to upgrade it to MAX 7 standalone - can you wait a few days?


I was also hoping to see you Max patch for Ricercare Una Melodia. Would you be able to post an updated link?

im totally new on this. what hardware and software do I need in order to perform this piece?

The original performance involved two four-track reel-to-reel tape recorders with one reel of tape running between them (to create tape delay), a mic and 4 loudspeakers (& amps!). The first part of the piece is simple delay to each of the 4 speakers, 3 seconds apart. Trumpet is recorded on machine 1 track one and played back on machine 2, and also fed into machine 1 track 2, which is played back on machine 2, and so on. In the second part, the playback machine plays at half speed so tape starts accumulating on the floor and each recorded layer plays back an octave lower... With max you just need a mic and an interface with 1 input & 4 outputs feeding 4 speakers.

Thanks a lot for your response!! :) I am totally new on this and is a big help. I have a focus rite 2i2 and a AT2020 mic. Does it Work? Could work with 2 speakers only?  Where could I find the preset for the max?

Hi. It's a 4 channel piece with speakers surrounding the audience so no, 2 speakers won't do it. There is no official patch, but lots of people have made their own versions with delays & record~/play~ objects. If you like I could dig out my old patch which probably still works.

Thanks for the info! It would be nice to have the patch. My e-mail is javitromba@hotmail.com

is this patch still available somewhere ? :) 


I can send you my patch - let me have your email address

Jonathan Harvey's Ricercare Una Melodia

pablodhv

Hello, I am trying to analyze/read rhythm or rhythm patterns out of the audio from a movie file.  At the moment I can visualize the audio from the video. But I am not sure what tools to use to complete the rhythmic pattern analysis part. The image shows what I have so far. 

Any help will be greatly appreciated. Thank you in advance. 

Hello, I am trying to analyze/read rhythm or rhythm patterns out of the audio from a movie file.  At the moment I can visualize the audio...

analyze-speech-rhythm-over-a-68-signature

Analyze speech rhythm over a 6/8 signature. 

Hector MacInnes

I'd like to try experimenting with using different audio sources that approximate pink noise as a dither. In order to see how these function in extreme circumstances I need a way of 

 the audio bits: i.e. reducing from MSP's internal float32 (I believe?) down to like extremes like 8 or 4 bit 

, so I can then introduce other pseudo-dithers.

I would have thought this could be a pretty simple maths operation I could apply in gen~ while passing audio from one buffer to another, but I'm struggling to get my head around it.

The original crude idea was that, e.g., a four bit system would round each sample value to the nearest 1/16th, i.e. increments of 0.0625, but I'm fairly sure that's not actually how truncation actually works... 

Any thoughts on this much appreciated. It used to be the case that you could export low bit depth files straight from Pro Tools and it was your own fault if they weren't dithered properly, but I don't think it allows that any more... now that it's the one thing I want to do... :/

Hi everyone,

I'd like to try experimenting with using different audio sources that approximate pink noise as a dither. In order to see how...

creating-truncated-rather-than-dithered-audio

creating truncated rather than dithered audio

Dan Nigrin

I am writing an app that requires writing square waves of two different frequencies to an audio file. The audio file alternates these frequencies frequently, and at very precise times - often just a few cycles (2 cycles of the lower frequency (1300 Hz), 4 cycles of the higher frequency (2100 Hz)) at a time.

No big deal so far - I can just peek~ the samples for single cycles of the two waveforms into a buffer~. The challenge though is that it turns out that I don't need *exactly* 2 cycles or 4 cycles of the waveforms at a time, but fractional amounts - like 2.75 cycles and 3.75 cycles for example. What this does is to create discontinuities in the resulting waveform of the audio file when these partial cycle waveforms are appended to each other.

So how can I write say, 2.75 cycles of 1300 Hz, then 3.75 cycles of 2100 Hz, then 2.75 cycles of 1300 Hz to a buffer~, but have the result be a continuous, "smooth", audio file?

I am writing an app that requires writing square waves of two different frequencies to an audio file.  The audio file alternates these frequencies frequently, and at very precise times - often just a few cycles (2 cycles of the lower frequency (1300 Hz), 4 cycles of the higher frequency (2100 Hz)) at a time.  

No big deal so far - I can just peek~ the samples for single cycles of the two waveforms into a buffer~.  The challenge though is that it turns out that I don't need *exactly* 2 cycles or 4 cycles of the waveforms at a time, but fractional amounts - like 2.75 cycles and 3.75 cycles for example.  What this does is to create discontinuities in the resulting waveform of the audio file when these partial cycle waveforms are appended to each other.

So how can I write say, 2.75 cycles of 1300 Hz, then 3.75 cycles of 2100 Hz, then 2.75 cycles of 1300 Hz to a buffer~, but have the result be a continuous, "smooth", audio file?


writing-alternating-single-cycle-ish-waveforms-to-disk

Writing alternating single cycle-ish waveforms to disk

Conall   O Maolain

I'm working on a kind of spectrum visualiser. I'm using a multichannel FFT to get a range of signals at each frequency bin and display them in a kind of wave table form. In the jit.gen patch I had to separate each plane of the input matrix using the swiz operator but I'd like to be able to do this dynamically so I can use FFT's with higher resolutions without needing to add a bunch more swiz operators. 

I tried using javascript but it slows down the render rate too much. Can it be done at all?

Hello,

I'm working on a kind of spectrum visualiser. I'm using a multichannel FFT to get a range of signals at each frequency bin and...

dynamic-plane-separation-in-jitgen

Dynamic plane separation in Jit.gen

Baek Santarek

Wildly interpolating 4 presets of [function] object with nodes object often crashes Max.

Faulting application name: Max.exe, version: 9.0.6.0, time stamp: 0x681d12e6

Faulting module name: function.mxe64, version: 9.0.6.0, time stamp: 0x681d138c

Faulting application start time: 0x01dbc7d72d78ceed

Faulting application path: C:\Program Files\Cycling '74\Max 9\Max.exe

Faulting module path: C:\Program Files\Cycling '74\Max 9\resources\externals\max\function.mxe64

Report Id: dcc55faf-c50d-4a51-9ce1-22bad2f8a250

Faulting package-relative application ID: 

Wildly interpolating 4 presets of [function] object with nodes object often crashes Max.

See the event log below.

Faulting application...

function-preset-interpolation-is-crashing-max-9060

[function] preset interpolation is crashing Max 9.0.6.0

Henrique Lorentz

Hello everyone! I'm using RNBO to develop a delay effect with C++ export and JUCE. It works perfectly as standalone and in Ableton, but different testers on FL Studio have shown me that there's a lot of noise when turning the feedback up. I found an old post on this forum mentioning the same issue, with no fix. I attached a video from a tester.

Hello everyone! I'm using RNBO to develop a delay effect with C++ export and JUCE. It works perfectly as standalone and in Ableton, but...

rnbo-delay-noise-on-fl-studio

RNBO Delay Noise on FL Studio

Recall the Coll

Hello folks. I was wondering if I could simulate the function of shift-draging a live.dial. I specifically want to use the live.dial object and I want to press a button/toggle to enable/disable finer values (like when I press shift).  Is that possible? Thanks in advance.

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