Re: Audio editing (Ernesto Accolti )


Subject: Re: Audio editing
From:    Ernesto Accolti  <ernestoaccolti@xxxxxxxx>
Date:    Thu, 20 Dec 2012 08:49:32 -0300
List-Archive:<http://lists.mcgill.ca/scripts/wa.exe?LIST=AUDITORY>

--14dae93407350ab74404d147535e Content-Type: text/plain; charset=ISO-8859-1 Hello, I agree with Kevin. I also prefer codes to work with many files but actually with Audition you can automate processing such as normalization (even loudness normalization) or any other. http://helpx.adobe.com/audition/using/automating-common-tasks-cs6.html Best regards, Ernesto. 2012/12/20 Kevin Austin <kevin.austin@xxxxxxxx> > Hi > > I was confused by the question which can be read in a number of ways. It > is not clear if there are [say] 10 files of 2 minutes duration, or 40 files > of 2 seconds. If there are 40 files, are their levels related to each other > such that the amount of signal amplification [read: normalization] needs to > remain constant across all 40 files, or is each file to be normalized > independently of the others. The approach would be different in each case. > > Normalization, applied correctly, will not clip a signal. The file will be > scanned for the peak level and this peak will be amplified by 'n dB' so > that the peak signal is "0 VU". This will not result in clipping if the > software is designed correctly. One of the difficulties experienced with > voice recording is that when very high quality mics have been used, there > is a strong likelihood of DC offset producing an asymmetrical waveform. > > The are manual techniques for achieving a kind of 'RMS normalization', but > they are labor intensive. As normalization does not change the relative > amplitude of signals within the file being processed, there should be no > detectable change in naturalness, IME. > > Kevin > > > > On 2012, Dec 18, at 12:34 PM, Matt Winn wrote: > > > Abin and List, > > Forgive double-postings, as I apparently made an error trying to attach > a file. It may be easier for you to do this in a scriptable (and free) > environment like Praat instead of Audition. I would like to share a simple > tool that I have made for this kind of intensity normalization. > > In the Praat script linked here, you can scale the intensities of all > sounds in a folder to a selected level. It will alert you if any of the > sounds clip, and offer you the option of decreasing your target intensity > level until none of them clip. In the end, you will have a folder full of > normalized sounds and an info text file to let you know what changes were > applied. The original sounds are preserved. > > This is designed to use for a folder full of short sounds (e.g. words), > and might not be ideal for longer sounds. It does not perform compression. > > Find the script here: > > > http://www.mattwinn.com/Scale_intensity_of_all_sounds_check_maxima_v2.txt > > To use it in Praat, either copy the text into a new Praat Script window > or open it directly. > > > > Regarding naturalness - you should be aware that compression and (to a > lesser extent) normalization actually decrease the naturalness of the > signals by altering each of them in different ways. There are some inherent > volume differences between some speech sounds (e.g. /s/ is louder than /f/, > /a/ is louder than /u/), so normalizing levels for these sounds would > decrease naturalness to some extent. > > Good luck, > > Matt > > > > > > > > > > On Mon, Dec 17, 2012 at 5:05 PM, Abin Kuruvilla Mathew < > amat527@xxxxxxxx> wrote: > > Dear All, > > > > I have a set of audio files (consonants and vowels) to be editied in > Adobe audition and was wondering to what extent and how much of > Normalization (RMS) and dynamic compression (if necessary) would be needed > so that the naturalness is preserved and clipping doesn't occur. > > > > kind regards, > > Abin > > > > -- > > Abin K. Mathew > > Doctoral student > > Department of Psychology (Speech Science) > > Tamaki Campus, 261 Morrin Road, Glen Innes > > The University of Auckland > > Private Bag 92019 > > Auckland- 1142 > > New Zealand > > Email: amat527@xxxxxxxx > > > > > > > --14dae93407350ab74404d147535e Content-Type: text/html; charset=ISO-8859-1 Content-Transfer-Encoding: quoted-printable Hello,<div><br></div><div>I agree with Kevin.=A0</div><div><br></div><div>I= also prefer codes to work with many files but actually with Audition you c= an automate processing such as normalization (even loudness normalization) = or any other.=A0</div> <div><br></div><div><a href=3D"http://helpx.adobe.com/audition/using/automa= ting-common-tasks-cs6.html">http://helpx.adobe.com/audition/using/automatin= g-common-tasks-cs6.html</a></div><div><br></div><div>Best regards,</div><di= v> Ernesto.<br><br><div class=3D"gmail_quote">2012/12/20 Kevin Austin <span di= r=3D"ltr">&lt;<a href=3D"mailto:kevin.austin@xxxxxxxx" target=3D"_blank= ">kevin.austin@xxxxxxxx</a>&gt;</span><br><blockquote class=3D"gmail_qu= ote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex= "> Hi<br> <br> I was confused by the question which can be read in a number of ways. It is= not clear if there are [say] 10 files of 2 minutes duration, or 40 files o= f 2 seconds. If there are 40 files, are their levels related to each other = such that the amount of signal amplification [read: normalization] needs to= remain constant across all 40 files, or is each file to be normalized inde= pendently of the others. The approach would be different in each case.<br> <br> Normalization, applied correctly, will not clip a signal. The file will be = scanned for the peak level and this peak will be amplified by &#39;n dB&#39= ; so that the peak signal is &quot;0 VU&quot;. This will not result in clip= ping if the software is designed correctly. One of the difficulties experie= nced with voice recording is that when very high quality mics have been use= d, there is a strong likelihood of DC offset producing an asymmetrical wave= form.<br> <br> The are manual techniques for achieving a kind of &#39;RMS normalization&#3= 9;, but they are labor intensive. As normalization does not change the rela= tive amplitude of signals within the file being processed, there should be = no detectable change in naturalness, IME.<br> <span class=3D"HOEnZb"><font color=3D"#888888"><br> Kevin<br> </font></span><div class=3D"im HOEnZb"><br> <br> <br> On 2012, Dec 18, at 12:34 PM, Matt Winn wrote:<br> <br> &gt; Abin and List,<br> &gt; Forgive double-postings, as I apparently made an error trying to attac= h a file. It may be easier for you to do this in a scriptable (and free) en= vironment like Praat instead of Audition. I would like to share a simple to= ol that I have made for this kind of intensity normalization.<br> &gt; In the Praat script linked here, you can scale the intensities of all = sounds in a folder to a selected level. It will alert you if any of the sou= nds clip, and offer you the option of decreasing your target intensity leve= l until none of them clip. In the end, you will have a folder full of norma= lized sounds and an info text file to let you know what changes were applie= d. The original sounds are preserved.<br> &gt; This is designed to use for a folder full of short sounds (e.g. words)= , and might not be ideal for longer sounds. It does not perform compression= .<br> &gt; =A0Find the script here:<br> &gt; <a href=3D"http://www.mattwinn.com/Scale_intensity_of_all_sounds_check= _maxima_v2.txt" target=3D"_blank">http://www.mattwinn.com/Scale_intensity_o= f_all_sounds_check_maxima_v2.txt</a><br> &gt; To use it in Praat, either copy the text into a new Praat Script windo= w or open it directly.<br> &gt;<br> &gt; Regarding naturalness - you should be aware that compression and (to a= lesser extent) normalization actually decrease the naturalness of the sign= als by altering each of them in different ways. There are some inherent vol= ume differences between some speech sounds (e.g. /s/ is louder than /f/, /a= / is louder than /u/), so normalizing levels for these sounds would decreas= e naturalness to some extent.<br> &gt; =A0Good luck,<br> &gt; Matt<br> &gt;<br> &gt;<br> &gt;<br> &gt;<br> </div><div class=3D"HOEnZb"><div class=3D"h5">&gt; On Mon, Dec 17, 2012 at = 5:05 PM, Abin Kuruvilla Mathew &lt;<a href=3D"mailto:amat527@xxxxxxxx= .nz">amat527@xxxxxxxx</a>&gt; wrote:<br> &gt; Dear All,<br> &gt;<br> &gt; I have a set of audio files (consonants and vowels) to be editied in A= dobe audition and was wondering to what extent and how much of Normalizatio= n (RMS) and dynamic compression (if necessary) would be needed so that the = naturalness is preserved and clipping doesn&#39;t occur.<br> &gt;<br> &gt; kind regards,<br> &gt; Abin<br> &gt;<br> &gt; --<br> &gt; Abin K. Mathew<br> &gt; Doctoral student<br> &gt; Department of Psychology (Speech Science)<br> &gt; Tamaki Campus, 261 Morrin Road, Glen Innes<br> &gt; The University of Auckland<br> &gt; Private Bag 92019<br> &gt; Auckland- 1142<br> &gt; New Zealand<br> &gt; Email: <a href=3D"mailto:amat527@xxxxxxxx">amat527@xxxxxxxx= i.ac.nz</a><br> &gt;<br> &gt;<br> &gt;<br> </div></div></blockquote></div><br></div> --14dae93407350ab74404d147535e--


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