Re: [linux-audio-dev] peakfiles and EDL's

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Subject: Re: [linux-audio-dev] peakfiles and EDL's
From: Richard C. Burnett (burnett_AT_tality.com)
Date: Sat Feb 24 2001 - 22:44:51 EET


I cannot see anyway to do it without resampling the peak values from the
insert point on. The peak files are just subsets of max/min values so for
the visual representation to be as accurate as possible, any modifications
that are changing the peak point boundaries would require rescanning. In
sampltude when I mix down (or as they put it bouncing tracks) to make a
single file, they recalculate the whole peakfile from the new wavedata and
not the previous peak files. At least I assume this because it does take
a bit of time. Of course I have never "Inserted" sound data. I replace
existing stuff but musically I have never had a need to insert sound. I
mean, don't get me wrong, I can see where it could be used in many
instances. Do you think that resampling the peak would hurt? It is hard
to try and compare samplitude at this because here is where they are so
different. They have HDR files that are just raw audio, and these you can
punch in and edit just like a continuous sound file. Then, you take these
HDR files and select ranges to become objects, and these are what you
align in the editor. Alot of this is automatic. Now that I think about
if, if I rerecord in an HDR, then when its done it recalculate the peak
info when I am done, and I think it does it all, since in this case,
things are different at the boundaries.

Hehe, maybe I am just confusing the situation more and myself at the same
time :)

Rick

On Sat, 24 Feb 2001, Paul Davis wrote:

> Design Question Time.
>
> consider the following situation. you are using pre-generated
> peakfiles, containing the max/min amplitudes for every N samples, to
> dislay waveforms. and you're also using an EDL for editing. lets make
> N=2048 for easy thinking.
>
> a user has edited the audio data stream at a point that is 3074
> samples into one file, by inserting another file at that point. you
> now have to recompute the waveform display. You get the first value
> for the first 2048 samples. The next peak value needs to be based on
> the 1024 samples from the first file and the succeeding 1024 samples
> from the next file.
>
> But wait: the peak values are precomputed, and may represent max/min
> values that are not in the sample data that now forms the audio data
> stream (e.g. the min value for the first file's 2nd block of 2048
> might occur at sample 3786). So how can we possibly decide what
> max/min values to use for the 2nd chunk of 2048 samples in the audio
> stream ? Its presumably based on both files, but can we determine it
> without reading the actual audio data for that part of the audio
> stream ?
>
> And it gets worse: what happens if the inserted material is not
> aligned with the first sample of the second file, but is offset. Now,
> every precomputed max/min pair for this file is essentially useless
> because they are "out of phase" with the way the audio is actually
> being used.
>
> Solutions or even suggestions welcome.
>
> --p
>

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| burnett_AT_tality.com | | |
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