Wednesday, November 5, 2014
Q. What are the best freeware plug-ins?
There are loads of freeware plug‑ins floating around out there now, so I find I'm getting swamped by choices. One site I checked out listed 670 of them! I'd rather not slow down my sessions looking for the perfect delay when just sticking with a good one and working with it would be much more productive. I've checked out a few of the ones mentioned in Mix Rescue and have been quite impressed, so I was wondering whether you could give me some further suggestions for a couple for each basic category of plug‑in. In particular, I'd be interested in any 'go to' freeware choices. I'm on a PC, so VST would be best.
Eoghan Brady via emailSome good freeware and donationware VST equalisers: Cockos ReaEQ, Bootsy Nasty CS, Antress Modern Black Dragon, and DDMF LP10.Some good freeware and donationware VST equalisers: Cockos ReaEQ, Bootsy Nasty CS, Antress Modern Black Dragon, and DDMF LP10.Q. What are the best freeware plug-ins?Q. What are the best freeware plug-ins?Q. What are the best freeware plug-ins?
SOS contributor Mike Senior replies: First of all, you could do worse than just download the ReaPlugs VST suite, which is a big chunk of the Reaper plug‑in complement and includes everything you're after, in one form or another. I've done whole mixes with just Reaper's plug‑ins, so I can vouch for their effectiveness. Other particularly worthwhile sets I've found are those from Antress Modern (http://antress.er‑webs.com), Bootsy (http://varietyofsound.wordpress.com), GVST (www.gvst.co.uk), MDA (http://mda.smartelectronix.com) and Voxengo (www.voxengo.com), which cover a lot of bases between them.
But on to some specific things I like, all of which have proved their worth in the heat of Mix Rescue! For general‑purpose EQ'ing, I do like Reaper's ReaEQ a lot, but for extra colour, try Bootsy's Nasty series and the Antress Modern emulations. DDMF (www.ddmf.eu) have a great donationware linear‑phase EQ called LP10, too. For synth‑style filtering, I usually just tend to automate ReaEQ, but Camel Audio's Camel Crusher (www.camelaudio.com) and Ohm Force's Frohmage (www.ohmforce.com) have more obvious attitude, if required. As far as dynamics are concerned, ReaComp and ReaXcomp in the ReaPlugs set are, again, good all‑round workhorses, but things like Georg Yohng's W1 (www.yohng.com), Buzzroom's BuzMaxi 3 (www.x-buz.com), Bootsy's Density, Jeroen Breebaart's PC2 (www.jeroenbreebaart.com) and the Antress Modern vintage emulations all get regular use on my projects. ReaGate and ReaFIR are a solid bet for most expansion and noise‑reduction tasks, so I've never really bothered looking elsewhere.
My freeware fallback for chorus, phaser, and flanger effects is Kjaerhus Audio's Classic series, and although I could no longer find a web presence for them at the time of writing, it's still possible to find the plug‑ins hosted on other sites via Google. MDA's Leslie and The Interruptor's Wow & Flutter (www.interruptor.ch) are cool for general modulation grunginess and I use those a lot. For tremolo/chopper effects, try Tweakbench's Cairo (www.tweakbench.com) or Oli Larkin's Autopan and LFO Chopper (www.olilarkin.co.uk). When it comes to distortion/saturation, there's lots of good stuff and I admit to being a bit of a collector in this respect. Some of my favourites are Bootsy's Ferric, GVST's GClip and GRecti, Jeroen Breebaart's Ferox, MDA's Combo and Bandisto, Mokafix Noamp (www.mokafix.com), Silverspike's Rubytube (www.silverspike.com), and Voxengo's Tubeamp: so much dirt, so little time! For more outrageous grainy and grungy effects, DBlue's Glitch (http://illformed.org) is a good bet, as are Jack Dark's outrageous Darkware series (www.gersic.com/plugins/hosted/darkware/darkware.html) and Tweakbench's Pudding and Sideslip.
The Interruptor's delay plug‑ins are good, as are GSi's WatKat (www.genuinesoundware.com), Tweakbench's Maelcum and GVST's GDuckDelay. That said, I tend to use ReaDelay for basic delay requirements most of the time. Smart Ambience is a great functional reverb demo, but Christian Knufinke's SIR (www.knufinke.de/sir/sir1.html) with impulses from Echo Chamber (www.memi.com/echochamber/responses/index.html) takes the cake for me in the freeware reverb department. For stereo image adjustment and M/S processing, my clear favourites are Voxengo's MSED and Flux's Stereo Tool (www.fluxhome.com). The latter has one of the best stereo vectorscope displays I've encountered anywhere. Speaking of displays, Roger Nichols' Inspector (www.rndigital.com) was my metering and spectrum-analysis plug‑in of choice for a long time, although Voxengo's SPAN is also good. I tend to use Schwa's payware Schope instead for most things these days, however. And speaking of Schwa (www.stillwellaudio.com), they have a great freeware bitscope plug‑in called Bitter that can be handy for digital troubleshooting. The TT Dynamic Range Meter is great if you're interested in the mastering 'loudness wars'; you can get it free on request via the Brainworx site (www.brainworx‑music.de).
Finally, here's a couple of odds and ends. Although I've yet to come across a decent, simple, freeware pitch‑shifter, if you're after freeware pitch correction, look no further than GVST's GSnap, which is pretty effective and has seen use in a number of Mix Rescues before now. If you're a fan of Aphex‑style psychoacoustic enhancement, also be sure to fire up Stillwell Audio's exciter, one of the plug‑ins available within the ReaPlugs ReaJS host, which does the same kind of thing.
Eoghan Brady via emailSome good freeware and donationware VST equalisers: Cockos ReaEQ, Bootsy Nasty CS, Antress Modern Black Dragon, and DDMF LP10.Some good freeware and donationware VST equalisers: Cockos ReaEQ, Bootsy Nasty CS, Antress Modern Black Dragon, and DDMF LP10.Q. What are the best freeware plug-ins?Q. What are the best freeware plug-ins?Q. What are the best freeware plug-ins?
SOS contributor Mike Senior replies: First of all, you could do worse than just download the ReaPlugs VST suite, which is a big chunk of the Reaper plug‑in complement and includes everything you're after, in one form or another. I've done whole mixes with just Reaper's plug‑ins, so I can vouch for their effectiveness. Other particularly worthwhile sets I've found are those from Antress Modern (http://antress.er‑webs.com), Bootsy (http://varietyofsound.wordpress.com), GVST (www.gvst.co.uk), MDA (http://mda.smartelectronix.com) and Voxengo (www.voxengo.com), which cover a lot of bases between them.
But on to some specific things I like, all of which have proved their worth in the heat of Mix Rescue! For general‑purpose EQ'ing, I do like Reaper's ReaEQ a lot, but for extra colour, try Bootsy's Nasty series and the Antress Modern emulations. DDMF (www.ddmf.eu) have a great donationware linear‑phase EQ called LP10, too. For synth‑style filtering, I usually just tend to automate ReaEQ, but Camel Audio's Camel Crusher (www.camelaudio.com) and Ohm Force's Frohmage (www.ohmforce.com) have more obvious attitude, if required. As far as dynamics are concerned, ReaComp and ReaXcomp in the ReaPlugs set are, again, good all‑round workhorses, but things like Georg Yohng's W1 (www.yohng.com), Buzzroom's BuzMaxi 3 (www.x-buz.com), Bootsy's Density, Jeroen Breebaart's PC2 (www.jeroenbreebaart.com) and the Antress Modern vintage emulations all get regular use on my projects. ReaGate and ReaFIR are a solid bet for most expansion and noise‑reduction tasks, so I've never really bothered looking elsewhere.
My freeware fallback for chorus, phaser, and flanger effects is Kjaerhus Audio's Classic series, and although I could no longer find a web presence for them at the time of writing, it's still possible to find the plug‑ins hosted on other sites via Google. MDA's Leslie and The Interruptor's Wow & Flutter (www.interruptor.ch) are cool for general modulation grunginess and I use those a lot. For tremolo/chopper effects, try Tweakbench's Cairo (www.tweakbench.com) or Oli Larkin's Autopan and LFO Chopper (www.olilarkin.co.uk). When it comes to distortion/saturation, there's lots of good stuff and I admit to being a bit of a collector in this respect. Some of my favourites are Bootsy's Ferric, GVST's GClip and GRecti, Jeroen Breebaart's Ferox, MDA's Combo and Bandisto, Mokafix Noamp (www.mokafix.com), Silverspike's Rubytube (www.silverspike.com), and Voxengo's Tubeamp: so much dirt, so little time! For more outrageous grainy and grungy effects, DBlue's Glitch (http://illformed.org) is a good bet, as are Jack Dark's outrageous Darkware series (www.gersic.com/plugins/hosted/darkware/darkware.html) and Tweakbench's Pudding and Sideslip.
The Interruptor's delay plug‑ins are good, as are GSi's WatKat (www.genuinesoundware.com), Tweakbench's Maelcum and GVST's GDuckDelay. That said, I tend to use ReaDelay for basic delay requirements most of the time. Smart Ambience is a great functional reverb demo, but Christian Knufinke's SIR (www.knufinke.de/sir/sir1.html) with impulses from Echo Chamber (www.memi.com/echochamber/responses/index.html) takes the cake for me in the freeware reverb department. For stereo image adjustment and M/S processing, my clear favourites are Voxengo's MSED and Flux's Stereo Tool (www.fluxhome.com). The latter has one of the best stereo vectorscope displays I've encountered anywhere. Speaking of displays, Roger Nichols' Inspector (www.rndigital.com) was my metering and spectrum-analysis plug‑in of choice for a long time, although Voxengo's SPAN is also good. I tend to use Schwa's payware Schope instead for most things these days, however. And speaking of Schwa (www.stillwellaudio.com), they have a great freeware bitscope plug‑in called Bitter that can be handy for digital troubleshooting. The TT Dynamic Range Meter is great if you're interested in the mastering 'loudness wars'; you can get it free on request via the Brainworx site (www.brainworx‑music.de).
Finally, here's a couple of odds and ends. Although I've yet to come across a decent, simple, freeware pitch‑shifter, if you're after freeware pitch correction, look no further than GVST's GSnap, which is pretty effective and has seen use in a number of Mix Rescues before now. If you're a fan of Aphex‑style psychoacoustic enhancement, also be sure to fire up Stillwell Audio's exciter, one of the plug‑ins available within the ReaPlugs ReaJS host, which does the same kind of thing.
Tuesday, November 4, 2014
Q What happens when audio is converted from AIFF format to AAC and back again?
Sound Advice : Recording
Hugh Robjohns
I understand that data-compressing audio files is really just chopping out frequencies we think we don't need, but I don't understand what encoders do to go the other way — converting an AAC file to an AIFF, for example. I say this because I've found that an AIFF ripped straight off a CD on my iTunes sounds pretty close to an AIFF that used to be an AAC. Both AIFFs sound better on my classic hi-fi system than the AAC — but the AIFFs don't quite sound the same as each other either... But it's hard to know if I'm hearing a difference or imagining it! So, can you tell me what the difference is between a first-generation AIFF (the original 'rip' from CD) and a second-generation AIFF (first-generation converted to AAC, and that AAC converted back to AIFF)?.
Aaron Johnson via email
Loss-less audio data compression options, such FLAC or ALAC (as being selected here in iTunes), may not reduce the file size as much as lossy codecs like MP3 or AAC — but neither do they sacrifice any actual audio data, and so the quality remains every bit as good as the CD, WAV or AIFF original from which the compressed file is created.SOS Technical Editor Hugh Robjohns replies: The first AIFF is, in theory, a bit-accurate copy of the CD audio and should sound identical to the CD. In practice, the accuracy of the rip depends on the cleanliness and quality of the CD itself, and the capability of the disc player and ripping software. The second AIFF is a 'capture' of the output from the AAC file decoder. There is no 'up-conversion' as such: lossy data codecs such as AAC throw information away and it can't be retrieved.
Here's a crude analogy. If the original CD were an arty photograph hanging in a gallery, the equivalent of your first AIFF would be a direct reprint of that picture taken from the same negative. The image is identical in every way. The equivalent of the AAC file would be a photo, taken in the gallery of the original on the wall, using a high-quality large-format camera. The resulting negative would then be blown up and printed to the same size as the original, but with the printer deliberately punching hundreds of small holes in the paper, so it wasn't quite so heavy. From a distance, the image would appear the same as the original, but closer inspection would reveal that many small parts of the image were physically missing. Your second AIFF is the equivalent of another high-quality photo, but this time of the hole-infested image. The resulting negative is enlarged and printed somewhere else — but with a better printer that doesn't punch holes in it. Again, the image will look, from a distance, like the original, but when you get closer you can see that the image has captured all the holes on the AAC version — because even though the holes aren't physically in the new print, the actual image data is still missing.
Lossy data-reduction identifies and throws out specific narrow frequency components, and the frequencies it chooses to discard change moment by moment, depending on what the codec's model of human hearing says we can't hear. This, in concert with other tricks, results in a data stream which is much smaller than that of conventional PCM audio. The AAC decoder receives the data stream and untangles it, rebuilding the audio output as far as it can, but there are still missing frequencies and other artifacts. So, returning to your listening experiences, the second AIFF and AAC file should sound identical to each other, and both should be degraded slightly in comparison with the first AIFF and the CD. Audio data was lost when the AAC file was created and it stays lost: there's no 'up-conversion' process from AAC to AIFF, only a format change.
If you're hearing something different then you are either suffering from 'confirmation bias' (where your brain is overriding your ears) or there's something odd going on with the file handling and format conversion in the computer: playing an AIFF is straightforward, but playing an AAC file requires active decoding, and those different processes can contaminate the audio output in subtle but audible ways, most usually by inducing data jitter.
There's no problem in storing AAC files for hi-fi listening if you accept the slightly reduced quality (and, to be fair, high-rate AAC is pretty good). If you want exactly the same quality as the original CD, though, you must rip the CD directly to an uncompressed PCM format (AIFF or WAV), or to a compressed file that's created using a 'lossless' codec. . Lossless codecs such as FLAC or Apple Lossless (ALAC) remove 'redundancy' rather than 'irrelevancy' and work a bit like the ZIP and RAR formats for computer files. They're not as efficient at space saving as AAC — typically only halving the file size, rather than quartering it (or more) — but on replay the audio is rebuilt completely as a bit-accurate output that's identical to the original CD. I use ALAC in iTunes to store my ripped CDs for this very reason.
Hugh Robjohns
I understand that data-compressing audio files is really just chopping out frequencies we think we don't need, but I don't understand what encoders do to go the other way — converting an AAC file to an AIFF, for example. I say this because I've found that an AIFF ripped straight off a CD on my iTunes sounds pretty close to an AIFF that used to be an AAC. Both AIFFs sound better on my classic hi-fi system than the AAC — but the AIFFs don't quite sound the same as each other either... But it's hard to know if I'm hearing a difference or imagining it! So, can you tell me what the difference is between a first-generation AIFF (the original 'rip' from CD) and a second-generation AIFF (first-generation converted to AAC, and that AAC converted back to AIFF)?.
Aaron Johnson via email
Loss-less audio data compression options, such FLAC or ALAC (as being selected here in iTunes), may not reduce the file size as much as lossy codecs like MP3 or AAC — but neither do they sacrifice any actual audio data, and so the quality remains every bit as good as the CD, WAV or AIFF original from which the compressed file is created.SOS Technical Editor Hugh Robjohns replies: The first AIFF is, in theory, a bit-accurate copy of the CD audio and should sound identical to the CD. In practice, the accuracy of the rip depends on the cleanliness and quality of the CD itself, and the capability of the disc player and ripping software. The second AIFF is a 'capture' of the output from the AAC file decoder. There is no 'up-conversion' as such: lossy data codecs such as AAC throw information away and it can't be retrieved.
Here's a crude analogy. If the original CD were an arty photograph hanging in a gallery, the equivalent of your first AIFF would be a direct reprint of that picture taken from the same negative. The image is identical in every way. The equivalent of the AAC file would be a photo, taken in the gallery of the original on the wall, using a high-quality large-format camera. The resulting negative would then be blown up and printed to the same size as the original, but with the printer deliberately punching hundreds of small holes in the paper, so it wasn't quite so heavy. From a distance, the image would appear the same as the original, but closer inspection would reveal that many small parts of the image were physically missing. Your second AIFF is the equivalent of another high-quality photo, but this time of the hole-infested image. The resulting negative is enlarged and printed somewhere else — but with a better printer that doesn't punch holes in it. Again, the image will look, from a distance, like the original, but when you get closer you can see that the image has captured all the holes on the AAC version — because even though the holes aren't physically in the new print, the actual image data is still missing.
Lossy data-reduction identifies and throws out specific narrow frequency components, and the frequencies it chooses to discard change moment by moment, depending on what the codec's model of human hearing says we can't hear. This, in concert with other tricks, results in a data stream which is much smaller than that of conventional PCM audio. The AAC decoder receives the data stream and untangles it, rebuilding the audio output as far as it can, but there are still missing frequencies and other artifacts. So, returning to your listening experiences, the second AIFF and AAC file should sound identical to each other, and both should be degraded slightly in comparison with the first AIFF and the CD. Audio data was lost when the AAC file was created and it stays lost: there's no 'up-conversion' process from AAC to AIFF, only a format change.
If you're hearing something different then you are either suffering from 'confirmation bias' (where your brain is overriding your ears) or there's something odd going on with the file handling and format conversion in the computer: playing an AIFF is straightforward, but playing an AAC file requires active decoding, and those different processes can contaminate the audio output in subtle but audible ways, most usually by inducing data jitter.
There's no problem in storing AAC files for hi-fi listening if you accept the slightly reduced quality (and, to be fair, high-rate AAC is pretty good). If you want exactly the same quality as the original CD, though, you must rip the CD directly to an uncompressed PCM format (AIFF or WAV), or to a compressed file that's created using a 'lossless' codec. . Lossless codecs such as FLAC or Apple Lossless (ALAC) remove 'redundancy' rather than 'irrelevancy' and work a bit like the ZIP and RAR formats for computer files. They're not as efficient at space saving as AAC — typically only halving the file size, rather than quartering it (or more) — but on replay the audio is rebuilt completely as a bit-accurate output that's identical to the original CD. I use ALAC in iTunes to store my ripped CDs for this very reason.
Q. Why does using a high-pass filter make things seem louder?
Hugh Robjohns
Why does applying a high‑pass filter to a sound sometimes result in the output being noticeably higher than it was before? Today I have been working on a sound that peaks at 0dBFS. It has a lot of low‑frequency content. I am applying a high‑pass filter at around 100Hz and the output from the EQ is peaking at around +4dBFS. Why should this happen? Most of the power in this sound is in the low frequencies, and it has little going on above 2kHz, so surely with the high‑pass filter most of the energy from the sound has gone!
Via SOS web site
SOS Technical Editor Hugh Robjohns replies: This is a very common effect and there are several possible reasons for it. Fundamentally, the filtering process changes the shape of the waveform, so although there may be less total energy in the signal, the peak amplitude may well increase.
If you think about a bunch of different‑frequency tones all playing at the same time, their phase relationships vary continuously and add to or cancel each other to create the total waveform. Remove some of those tones and some of those cancellations won't occur. That can result in the waveform becoming bigger. There are various reasons why using a high-pass filter on a signal can make it sound louder. Some equalisers, for example, actually boost the region just above the turnover frequency, which can produce an increase in peak level.There are various reasons why using a high-pass filter on a signal can make it sound louder. Some equalisers, for example, actually boost the region just above the turnover frequency, which can produce an increase in peak level.
Most equalisers also introduce significant phase shifts and that, again, will change the way different frequencies combine and cancel. It can also happen because some equalisers actually boost the region just above the turnover point below which they are attenuating, potentially increasing peak level.
Why does applying a high‑pass filter to a sound sometimes result in the output being noticeably higher than it was before? Today I have been working on a sound that peaks at 0dBFS. It has a lot of low‑frequency content. I am applying a high‑pass filter at around 100Hz and the output from the EQ is peaking at around +4dBFS. Why should this happen? Most of the power in this sound is in the low frequencies, and it has little going on above 2kHz, so surely with the high‑pass filter most of the energy from the sound has gone!
Via SOS web site
SOS Technical Editor Hugh Robjohns replies: This is a very common effect and there are several possible reasons for it. Fundamentally, the filtering process changes the shape of the waveform, so although there may be less total energy in the signal, the peak amplitude may well increase.
If you think about a bunch of different‑frequency tones all playing at the same time, their phase relationships vary continuously and add to or cancel each other to create the total waveform. Remove some of those tones and some of those cancellations won't occur. That can result in the waveform becoming bigger. There are various reasons why using a high-pass filter on a signal can make it sound louder. Some equalisers, for example, actually boost the region just above the turnover frequency, which can produce an increase in peak level.There are various reasons why using a high-pass filter on a signal can make it sound louder. Some equalisers, for example, actually boost the region just above the turnover frequency, which can produce an increase in peak level.
Most equalisers also introduce significant phase shifts and that, again, will change the way different frequencies combine and cancel. It can also happen because some equalisers actually boost the region just above the turnover point below which they are attenuating, potentially increasing peak level.
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