Visual editing of audio-rate parameter files?

[Charles Turner-3]

Hey, sorry to ask a tech question here, but it's pretty general in  
nature:

I've recorded an audio-rate (AIFF) 8-channel datafile with synthesis  
parameter data in it. That is, instead of data that ranges from -1.0  
to 1.0, the data range is more like 0.0 to 50.0.

It'd be great to open this file in a conventional audio editor, but it  
seems like most (or all) of them clip the data above 1.0. Some will  
wrap it, which creates another type of weirdness.

Any thoughts? I poked around a bit with MSP, and it seems the  
[buffer~] object only handles 4-channels. I haven't found an audio  
editor that has this feature. Currently wondering whether there's a  
more general tool for DSP use that might do the trick.

Thanks! Charles

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[justin smith]

Charles Turner wrote:

> Hey, sorry to ask a tech question here, but it's pretty general in nature:
>
> I've recorded an audio-rate (AIFF) 8-channel datafile with synthesis
> parameter data in it. That is, instead of data that ranges from -1.0 to
> 1.0, the data range is more like 0.0 to 50.0.
>
> It'd be great to open this file in a conventional audio editor, but it
> seems like most (or all) of them clip the data above 1.0. Some will wrap
> it, which creates another type of weirdness.
>
> Any thoughts? I poked around a bit with MSP, and it seems the [buffer~]
> object only handles 4-channels. I haven't found an audio editor that has
> this feature. Currently wondering whether there's a more general tool
> for DSP use that might do the trick.
>

I would say that if any tool could handle it snd would be able to edit it
visually. I think snd is linux / unix only.

https://ccrma.stanford.edu/software/snd/

Also, sox should be able to split it out into individual channels or normalize
to a range other tools can use (ie. if you scale amplitude by .02 it should be
editable in a normal editor, and then sox should be able to multiply it by 50 again
when you are done -- but of course this would probably lead to some rounding errors
in quieter parts of the signal).
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[Charles Turner-3]

Hi Justin-

Thanks for the response. I've used snd a bunch in the past, and while  
it might be programmed to do such a thing, it doesn't right out of the  
box.

Your suggestion about scaling is well taken, although I think I would  
generate my parameter data with values between 0.0 - 1.0 and then post-
scale it. Inserting the record stage between the data generation and  
post-scaling would enable me to edit the recorded parameters while  
editing the resulting audio.

The only inelegance would be having to remember the scaling factors,  
although the rounding you discuss wouldn't be an issue then.

Best, Charles

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[Phil Thomson]

You haven't mentioned which (Mac?) audio editors you've tried.
Offhand, I can think of SoundHack, Audacity and Amadeus (or Amadeus
Pro) that can import raw data. Have you tried all those?

On Thu, Nov 12, 2009 at 2:21 PM, Justin Glenn Smith
<noisesmith@...> wrote:

> Charles Turner wrote:
>> Hey, sorry to ask a tech question here, but it's pretty general in nature:
>>
>> I've recorded an audio-rate (AIFF) 8-channel datafile with synthesis
>> parameter data in it. That is, instead of data that ranges from -1.0 to
>> 1.0, the data range is more like 0.0 to 50.0.
>>
>> It'd be great to open this file in a conventional audio editor, but it
>> seems like most (or all) of them clip the data above 1.0. Some will wrap
>> it, which creates another type of weirdness.
>>
>> Any thoughts? I poked around a bit with MSP, and it seems the [buffer~]
>> object only handles 4-channels. I haven't found an audio editor that has
>> this feature. Currently wondering whether there's a more general tool
>> for DSP use that might do the trick.
>>
>
> I would say that if any tool could handle it snd would be able to edit it
> visually. I think snd is linux / unix only.
>
> https://ccrma.stanford.edu/software/snd/
>
> Also, sox should be able to split it out into individual channels or normalize
> to a range other tools can use (ie. if you scale amplitude by .02 it should be
> editable in a normal editor, and then sox should be able to multiply it by 50 again
> when you are done -- but of course this would probably lead to some rounding errors
> in quieter parts of the signal).
> _______________________________________________
> microsound mailing list
> microsound@...
> http://or8.net/mailman/listinfo/microsound
>

--
"In art only one thing matters: that which cannot be explained."

~Georges Braque


~~~~~~~~~~~~~~~~~~~~~~~~~
~ Phil Thomson
~ http://philthomson.ca/
~~~~~~~~~~~~~~~~~~~~~~~~~
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[Charles Turner-3]

On Nov 12, 2009, at 7:02 PM, Phil Thomson wrote:

> You haven't mentioned which (Mac?) audio editors you've tried.
> Offhand, I can think of SoundHack, Audacity and Amadeus (or Amadeus
> Pro) that can import raw data. Have you tried all those?

Yep. I wouldn't say that SoundHack enables visual editing. Amadeus and  
Audacity both clip the data values. Tried also DSP Quattro, Sound  
Sampler, and Sound Studio. (I'm sure I could write a small article on  
how different editors handle clipping.)

I'm skeptical that an audio editor would permit what I want to do.  
After all it's supposed to be feeding the data into the audio  
subsystem of the computer, which is just going to clip anyway. That's  
is why I thought of Matlab or some such...

C

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[Greg Pond]

I cant say I have an answer but I would post to the PD list for this.

greg

On Thu, Nov 12, 2009 at 6:27 PM, Charles Turner <vze26m98@...> wrote:

> On Nov 12, 2009, at 7:02 PM, Phil Thomson wrote:
>
>> You haven't mentioned which (Mac?) audio editors you've tried.
>> Offhand, I can think of SoundHack, Audacity and Amadeus (or Amadeus
>> Pro) that can import raw data. Have you tried all those?
>
> Yep. I wouldn't say that SoundHack enables visual editing. Amadeus and
> Audacity both clip the data values. Tried also DSP Quattro, Sound Sampler,
> and Sound Studio. (I'm sure I could write a small article on how different
> editors handle clipping.)
>
> I'm skeptical that an audio editor would permit what I want to do. After all
> it's supposed to be feeding the data into the audio subsystem of the
> computer, which is just going to clip anyway. That's is why I thought of
> Matlab or some such...
>
> C
>
> _______________________________________________
> microsound mailing list
> microsound@...
> http://or8.net/mailman/listinfo/microsound
>

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[Michal Seta]

On Thu, Nov 12, 2009 at 7:56 PM, Greg Pond <gregpond@...> wrote:
> I cant say I have an answer but I would post to the PD list for this.

Yes, in fact, pd arrays do not have any limits as to the ranges of
stored data.  Just make sure you create the arrays that can
accommodate such large values (you can do it in the properties) or
else the data will be drawn all over the place.  Also, I guess you
will want to avoid the use of [soundfiler] object to load that data
into arrays (this is a "standard" way of loading audio data) because
it will probably chop your data into -1.0 to 1.0 although I have not
tried it and may be wrong.  In fact, soundfiler has the ability to
completely ignore the header of the file and load raw data.
Otherwise, you send the array a "read" message and it should do but
you will need to know how many values there are in your files and
create the arrays large enough to accommodate the data.

Feel free to shoot me an email if you need any assistance.

./MiS
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