24 January 2008
Setting file defaults on Audacity-win-1.3.4
I've just downloaded the latest release of the Audacity free-and-open-source audio editing software. When you open the program, you can look on the upper left of the toolbar and find a pulldown menu called Edit. At the bottom of the list you find Preferences. From there, I found locations to set the preference for .wav file exports at 16-bit and for .mp3 exports at 128 kbps. This is what I recommend in order to generate broadcast-quality files that can nevertheless be edited and played in pretty much all other programs, including older ones. The default setting in Audacity is 32-bit, which is supposedly higher quality but can cause problems in Winamp, older SoundForge releases, etc.
06 January 2008
File formats: Codec conversion and codec chaining
Codecs are computer programs for reducing the size of audio files, for faster transmission over the internet and for saving storage space. There are quite a few different versions of codecs, and as described in the re-posted messages below, converting from one file format to another can cause you to permanently lose different parts of the audio information.
A radio program with various contributors like WINGS receives files saved with various codecs, so in order to avoid multiple audio losses, it's best to convert all files to a lossless format like .WAV for editing, so you don't do "codec chaining" and lose information several times. (My favourite editing software only recognizes 16-bit, 44.1 mHz files, so I convert to that.)
Even codecs that are ostensibly the same - have the same name, but are programmed by different companies or people - can have different aural effects. So, it's good to use your ears to help decide not only the file format but also what computer program to use for conversion.
As an example: When I edit with Soundforge 4.5, for which I bought the mp3 plugin, I prefer to save my shows as 112 kbps mp3s, because I find the 128 kbps mp3s made on that codec sound too hissy. But now that I have Audacity (a free and open-source editing program) with a Lame mp3-type converter installed, I usually drag my edited .wav files into Audacity and export them from there as 128 kbps mp3 files. That does a nice clean job.
Here are the discussion items about codecs that I've copied from the listserve of the GRC (Grassroots Radio Conference):
Here's a reply from Brian Shirutski:
A radio program with various contributors like WINGS receives files saved with various codecs, so in order to avoid multiple audio losses, it's best to convert all files to a lossless format like .WAV for editing, so you don't do "codec chaining" and lose information several times. (My favourite editing software only recognizes 16-bit, 44.1 mHz files, so I convert to that.)
Even codecs that are ostensibly the same - have the same name, but are programmed by different companies or people - can have different aural effects. So, it's good to use your ears to help decide not only the file format but also what computer program to use for conversion.
As an example: When I edit with Soundforge 4.5, for which I bought the mp3 plugin, I prefer to save my shows as 112 kbps mp3s, because I find the 128 kbps mp3s made on that codec sound too hissy. But now that I have Audacity (a free and open-source editing program) with a Lame mp3-type converter installed, I usually drag my edited .wav files into Audacity and export them from there as 128 kbps mp3 files. That does a nice clean job.
Here are the discussion items about codecs that I've copied from the listserve of the GRC (Grassroots Radio Conference):
dateNov 13, 2007 1:36 PM
subjectRe: [grc] Anyone using flac audio codec?
I like to second this view. Every time you import an mp3 into any editor it is converted to an approximation of the original PCM (or wav)file. Every time you save it as an mp3 it is throwing our over 3/4ths of the data, you edit again and you just threw out a different set of 3/4ths of the data. Pretty soon there is not much left of the original signal. After reading about how they make mp3's I was amazed that it
even works.
All editing should be in full PCM format, wav, flac, or whatever apple calls it. This includes segments that are being shared. You can save space by using a lossless compression like Flac (there are others such as Shorten, Monkey's Audio, and WavePack, but I think Flac is the best, at least it's the most used).
Flac is the default PCM format for my Ubuntu box at home, which is handy because it takes up less then half the space. And most of the Linux sound editors use it directly. I just wish there was a plug-in for Sound Forge to give it Flac support so you didn't have to manually convert it.
BTW there are some awesome conversion tools at: http://www.rarewares.org
I like the drop utilities Flacdrop, OggDrop, and LameDrop, it's just a panel on your desktop you drop the file into it, it converts it, and very fast too. Oggdrop even works in Linux under Wine, I haven't tried Flacdrop.
I agree that this is overkill for the final product. Last step should be to convert it to mp3 for distribution.
Any original music recording should always be archived in a LOSSLESS compressed format, such as Flac. And in case you can't tell, I'm one of those DeadHead taper dudes that Scooter mentioned.
As far as lossy formats go it should be noted that ogg files sound better and make a smaller file then mp3's or wma's. But only some players support them.
(I have a song running around in the back of my head, it's called "MX'd by baby Jesus")
Dave Willard
Pseudo-Engineer
Radio Free Moscow
Here's a reply from Brian Shirutski:
dateJan 6, 2008 1:11 PM
subjectRe: [grc] Anyone using flac audio codec?
BBC paper on cascaded codecs (september 2005):
>Broadcasters have experienced significant problems with cascaded
> audio coding in the broadcast chain following the introduction of
> digital transmission. It has been found that cascading different
> codecs can result in an overall degradation in sound that many
> listeners find objectionable. A comprehensive investigation of this
> problem has been conducted by members of the EBU project group B/
> AIM. This paper describes typical cascades of codecs found in radio
> broadcast chains, and aims to identify the most critical
> combinations. The intent is to guide broadcasters in deciding which
> codec combinations should be avoided to maximise sound quality...
>
> ...An extensive, thorough, and time-consuming investigation has been
> conducted by members of the EBU B/AIM project group into cascaded
> audio coding. A model of a broadcast chain consisting of 5 cascaded
> codecs was assumed. From the thousands of possible combinations of
> codecs, a subset of the more likely ones was tested for audio
> performance using objective and subjective methods.
>
> Objective testing using PEAQ was successfully employed to reduce the
> number of combinations to be subjectively tested. The subjective
> testing was performed using the MUSHRA test method, with the subset
> of codec combinations being divided amongst a small number of test
> laboratories. Some codecs were tested by all sites for comparison
> purposes. The results clearly show that the cumulative effect of
> cascaded audio coding can be highly detrimental to audio quality,
> even when each stage in the chain makes only a small reduction in
> quality.
>
> The comparison of objective and subjective results showed a good
> correlation between scores. Caution should be exercised here because
> the scales and descriptive terms associated with the two test
> methods used are quite different. The objective and subjective test
> results were both analysed to try to identify codec performance that
> was significantly better or significantly worse than expected. It
> was found that none of the combinations showed any unusual
> behaviour. This should simplify the selection process for users of
> low bit rate coding - it implies that choosing the best codecs will
> give the best results.
<http://www.bbc.co.uk/rd/pubs/whp/whp-pdf-files/WHP118.pdf>
05 January 2008
What are Podcasting and Internet Radio?
A friend emailed me this question. Here's my reply:
Podcasting and internet radio are both kind of vaguely defined terms right now - there can be more than one meaning.
Internet radio can mean continuously streaming audio either coming from a broadcast station onto the internet, or generated independently just for web streaming. But internet radio can also sometimes mean making pre-recorded audio files available for listening by streaming, and/or by downloading.
Podcasting generally refers to single items of audio - for example, a radio program. It can sometimes refer to streaming audio, sometimes to downloadable files that you access online, and sometimes also to a kind of automatic downloading system that delivers the audio directly to your computer or mobile device. The best known method for this automatic syndication is RSS - which supposedly stands for Really Simple Syndication. There are a few other computer applications now that do basically the same thing. The term Podcasting was created in conjunction with Apple's iTunes/iPod application, but it has been adopted to refer to pre-existing technology for audio file downloading and streaming, and also for newer technologies like RSS.
Someone needs to begin a good taxonomy of web applications, to help people figure out what they are describing and to aid in useful naming conventions. Perhaps such exists and I just haven't found it yet.
Podcasting and internet radio are both kind of vaguely defined terms right now - there can be more than one meaning.
Internet radio can mean continuously streaming audio either coming from a broadcast station onto the internet, or generated independently just for web streaming. But internet radio can also sometimes mean making pre-recorded audio files available for listening by streaming, and/or by downloading.
Podcasting generally refers to single items of audio - for example, a radio program. It can sometimes refer to streaming audio, sometimes to downloadable files that you access online, and sometimes also to a kind of automatic downloading system that delivers the audio directly to your computer or mobile device. The best known method for this automatic syndication is RSS - which supposedly stands for Really Simple Syndication. There are a few other computer applications now that do basically the same thing. The term Podcasting was created in conjunction with Apple's iTunes/iPod application, but it has been adopted to refer to pre-existing technology for audio file downloading and streaming, and also for newer technologies like RSS.
Someone needs to begin a good taxonomy of web applications, to help people figure out what they are describing and to aid in useful naming conventions. Perhaps such exists and I just haven't found it yet.
08 December 2007
Comb Filter to effect a frequency and its harmonics!
After I did all that work described in my last post, my friend Jerry emailed to tell me that you can effect a frequency and all its harmonics at once with a Comb Filter. Sure enough, once I knew the name I looked for Comb Filter and found that David R. Sky, the same guy that designed the Nyquist-language Notch Filter that plugs in to Audacity has also designed a Comb Filter for it. I have now downloaded the comb filter and installed it the same way I did the Notch Filter. It can be found at: http://audacity.sourceforge.net/download/nyquistplugins
I don't have time to play with it right now, because I have to do my content editing tonight, but next time I'll try that first.
Here's what David wrote about it on the Sourceforge download page:
I don't have time to play with it right now, because I have to do my content editing tonight, but next time I'll try that first.
Here's what David wrote about it on the Sourceforge download page:
Comb filter | View comb.ny | Download comb.zip
Example audio clips: [MP3 Clip 1]
The name 'comb' filter comes from how it acts on the audio spectrum of what it's applied to: it looks like a comb with the teeth pointing up. For example, if you set the comb frequency at 1000 Hz, the comb filter emphasizes 1000 Hz as well as 2000, 3000, 4000 and so on Hz. This particular plug-in produces an 'airy' effect, which is more pronounced the higher the comb decay value is set, and resonance is increasingly produced as well.
A comb filter can be produced using flanger-like settings on a delay effect, but this filter does not use a delay to get the result, so it does sound somewhat different.
Written by David R. Sky
60 cycle hum removal using Audacity and Notch Filter
Audacity is the free and open-source audio editing program available from Sourceforge. I prefer the Beta version, which has important new features compared to the so-called "stable" version. You can download Audacity itself at: http://audacity.sourceforge.net/download/
For this editing project, I was working with audio recorded by someone else using a telephone line, and somewhere in the phone line or the setup there was a very typical 60-cycle hum, of the type induced by electricity, which got recorded on the original track. (Note, we are both in North America. In some countries the electricity induces a 50-cycle hum instead.) Some audio editing programs have a plug-in that comes packaged with them that will let you choose to eliminate this kind of hum and its harmonics in one pass. Audacity, as far as I can tell, currently does not.
To improve this hum, what I did was:
First, I installed the Nyquist plug-in called Notch Filter. (There's a link to this download page at the end of this article.) The Notch Filter lets you narrow in on a very specific frequency and notch it out of the sound. (The analog tool I used to use for this kind of problem is a parametric equalizer.)
With this digital notch filter, you choose a "notch frequency" and a "notch q" - the Q determines how wide a spectrum of frequency you are choosing, centred around the frequency you chose.
I found and highlighted a section with only the ambience and the hum, and tested on that until I got an acceptably low noise level. Here's what I had to do:
For 60 cycle hum, I chose 60 for the notch frequency and set the notch q to the highest number (the narrowest bandwidth) - which was 5. It was hard to tell any difference so I went through and did the same for each of the harmonics: 120, 180... all the way through 720. This was fast when applied to the test segment, but took about 30 seconds each pass for the entire almost-30-minute piece.
When I had finished, I thought too much of the low end had been lost.
So, I re-imported the original file (post normalization), and using the Amplify function, I took it down by trial and error to -14db below its original level. Playing this together with the multiply-Notched file restored the hum slightly, though at a lower level, but it also restored some low end to the voice. As the low end had been a bit overloud on the original, this was a nice combination.
I exported the two files in a mixdown to a combined .wav file, and with that proceeded to editing the content itself.
Here's the page where I found the advice on harmonics:
http://recordrestoration.com/01-Get%20Rid%20of%20Noise.htm
Here's the information and link about the Notch Filter - you have to follow the link and then search or scroll down to the plug-in you want (Notch Filter), then click through where it says Download Notch and that will offer to save to your computer:
http://audacity.sourceforge.net/download/nyquistplugins
After you download and run the installer program for the Nyquist Notch Filter, you have to save the file for the plug-in to C:\Program Files\Audacity\Plug-ins [or, instead of Plug-ins you might install it in a folder named Nyquist]. When the program is installed it appears as Notch Filter, in the bottom section of Audacity's Effects list (see the pulldown menu under Effects on toolbar).
For this editing project, I was working with audio recorded by someone else using a telephone line, and somewhere in the phone line or the setup there was a very typical 60-cycle hum, of the type induced by electricity, which got recorded on the original track. (Note, we are both in North America. In some countries the electricity induces a 50-cycle hum instead.) Some audio editing programs have a plug-in that comes packaged with them that will let you choose to eliminate this kind of hum and its harmonics in one pass. Audacity, as far as I can tell, currently does not.
To improve this hum, what I did was:
First, I installed the Nyquist plug-in called Notch Filter. (There's a link to this download page at the end of this article.) The Notch Filter lets you narrow in on a very specific frequency and notch it out of the sound. (The analog tool I used to use for this kind of problem is a parametric equalizer.)
With this digital notch filter, you choose a "notch frequency" and a "notch q" - the Q determines how wide a spectrum of frequency you are choosing, centred around the frequency you chose.
I found and highlighted a section with only the ambience and the hum, and tested on that until I got an acceptably low noise level. Here's what I had to do:
For 60 cycle hum, I chose 60 for the notch frequency and set the notch q to the highest number (the narrowest bandwidth) - which was 5. It was hard to tell any difference so I went through and did the same for each of the harmonics: 120, 180... all the way through 720. This was fast when applied to the test segment, but took about 30 seconds each pass for the entire almost-30-minute piece.
When I had finished, I thought too much of the low end had been lost.
So, I re-imported the original file (post normalization), and using the Amplify function, I took it down by trial and error to -14db below its original level. Playing this together with the multiply-Notched file restored the hum slightly, though at a lower level, but it also restored some low end to the voice. As the low end had been a bit overloud on the original, this was a nice combination.
I exported the two files in a mixdown to a combined .wav file, and with that proceeded to editing the content itself.
Here's the page where I found the advice on harmonics:
http://recordrestoration.com/01-Get%20Rid%20of%20Noise.htm
3-The first thing you will do is eliminate any 60 cycle hum present in the waveform. Go to Effects/Filters/Notch Filter. In the "Presets" box, click on "60Hz + Harmonics." The notch width box should be set to "super narrow." Click "OK" and the notch filter will remove any 6o cycle hum present. (The filter actually very selectively removes 60 cycle hum + its harmonics, 120, 180, 240, etc.). The frequency ranges filtered out in this step are so narrow that the loss will not be audible. If you are in a country that uses 50 cycle AC current, instead of the "60Hz + Harmonics" preset selection, choose "50Hz + Harmonics." If you are sure there is no AC hum present in your waveform, you can forego this step.
Here's the information and link about the Notch Filter - you have to follow the link and then search or scroll down to the plug-in you want (Notch Filter), then click through where it says Download Notch and that will offer to save to your computer:
http://audacity.sourceforge.net/download/nyquistplugins
Notch Filter | View notch.ny | Download notch.zip
Notch filter
Like its name suggests, a notch filter cuts out a "notch" in the spectrum of your audio. The default frequency is 60Hz, great for removing 60Hz electrical hum, in case your recording equipment has picked this up.
The second control, q, determines the width of the notch cut from your audio. Default q is 1, below 1 creates a wider notch, above 1 creates a narrower notch.
Works on mono and stereo audio.
By David R. Sky
After you download and run the installer program for the Nyquist Notch Filter, you have to save the file for the plug-in to C:\Program Files\Audacity\Plug-ins [or, instead of Plug-ins you might install it in a folder named Nyquist]. When the program is installed it appears as Notch Filter, in the bottom section of Audacity's Effects list (see the pulldown menu under Effects on toolbar).
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