AF’s Weblog

June 1, 2012

How to Get the Pumping Drums Effect with Sidechain Compression

To read the full article see:  Sidechain Compression

Sidechaining has been around for years; this is the process of using one signal to control another. A couple classic examples are using a kick drum to gate a bass part, or doing de-essing – isolating the high sibilant frequencies from a vocal, and using those to trigger compression so that the sibilants come down in volume.

But in the digital age, we can do a lot more with sidechaining. One of the most popular applications is with dance music, where sidechaining can create the “heavy pumping” electronica drum sound used by artists like Eric Prydz and others.

We’ll describe how to do this with Cakewalk Sonar, although the same principle applies to other programs that allow for sidechaining. Sonar allows sidechaining for several effects, including compression, so that one instrument can control the compression characteristics of another instrument. This offers a variety of effects, including a “pumping” drum sound for multitracked drum parts; we’ll do that by setting up the snare to control compression for all drum tracks.

Fig. 1: You’ll need a drum submix bus to create an overall drum sound.

The first step is to create a drum submix bus, and send the drum tracks to it (Fig. 1). We need this submix so the entire drum track can be processed by the sidechained compressor. To create the submix bus, right-click in an empty space in the bus pane and select “Insert Stereo Bus.” To create a send in track view, right-click in a blank space in the track title bar and select “Insert Send.” From the menu that appears, select the send destination. Make sure you feed the bus pre-fader, and turn the individual drum channel faders down so that only the bus contributes the drum sound to the master.

Fig. 2: Assign the Drum Submix out to your main stereo output.

Let’s take a closer look…

….

Create a second pre-fader send in the snare track, and assign its out to the bus feeding the sidechain input.

Fig. 7: We’re almost there – it’s time to adjust the compressor.

To adjust the compressor, start with the compression attack time set to 0 ms; the drum sound will essentially disappear when the snare hits because the gain is being reduced so much. Gradually increase the attack time to let through more of the initial snare hit, and add a fair amount of release (250-500 ms) to increase the apparent amount of pumping.

And there you have it – the pumping drum sound. May it go over well on the dance floor!

To read the full article see:  Sidechain Compression

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May 10, 2012

Avoid Common Recording Mistakes

Filed under: Recording reviews — Tags: , , — audiofanzine @ 1:07 pm

To read the full detailed article see : Avoid Common Recording Mistakes

Why is it so hard to get tracks that kill? Mixes that scream with emotional impact–music that holds up to the work of the masters of our craft?

Experienced pro or newbie neophyte, we all share a desire to improve the sound, relevance, and “vibe” of our recordings. But sometimes the way to do this isn’t just by doing the right thing, but avoiding doing the wrong thing–and that in turn will indeed make things easier.

Bad Gear

Everyone’s favorite whipping boy, bad gear is often the first place many of us look to and point the finger at when something about our recordings doesn’t knock us out. And let’s face it: First-class gear sounds great, and that can’t help but make things sound better–but only if you know what you’re doing with it. I’ve been amazed by the quality of some recordings I’ve heard that were done on primitive or inexpensive gear, however, that says more about the engineer than the gear. Still, it’s important to scrutinize your system from time to time and probe for weak links. Did you upgrade your mixer, but not your monitor speakers? Do you have a great microphone, but are using it with an old, noisy mic preamp? Nothing works in isolation, so consider where the best improvements can be made to enhance your system’s sound quality as a whole, and don’t obsess on any single area (like having the best mic cabinet in the world if you don’t have preamps that are equal to the task).

The Curse of the Adaptive Ear

Even in a well-designed control room with great monitors, our ears adapt to EQ changes very quickly–that’s how you can enjoy hearing your favorite song on a cheap TV speaker or a high quality system. Our ears perceive the extremes of the audible frequency spectrum differently at different playback levels, with the flattest response being at about 85dB SPL. Our ears also tire after long hours, especially at unsafe monitoring levels. That EQ tweak that sounded great last night after 10 hours of playback at 105dB might not sound so hot the next morning. Having high-quality reference material that you can A/B with your mix can help you get back to reality when EQ changes start to throw off your perspective over time, and so can watching your levels and knowing when to quit when your ears have had enough for the day.

Now let’s take a look at some other mistakes…

No Substitute for Performances

I consider myself to be a pretty good editor with tape or DAW; I’ve been doing it for over three decades, and I’ve gotten a fair amount of kudos from clients over the years. But I still need “something to work with”, and the best edit is a performance that doesn’t need one. If you have to edit, it’s a lot easier to do if you have tracks with generally solid performances with few errors and great feel. Piecing something together from sub-standard performances is not my idea of a good time, and the musicality of your work is going to be much better if the musicality of the people you’re working with is already happening. When I work with brilliant musicians, my work sounds better – and so will yours. If things are not quite “there” with the artists you are working with, take some time to do some pre-production rehearsals before you get into the studio so that you can help get things as tight as possible before you start waxing tracks. Rehearse more, edit less.

Number one, with a Bullet

Probably the number one issue is material. A so-so recording of a great song still leaves you with a great song. A great recording of a so-so song leaves you with a so-so song. Of course, we’re not in the business of making so-so recordings, and everything matters, so take a moment to evaluate the weaker areas of your whole rig – and that includes your personal skills and musicianship – and plan out a strategy for improving each of them. Your recordings and productions will only get better as a result.

To read the full detailed article see : Avoid Common Recording Mistakes

February 14, 2012

Olympus LS-20M Review

Somewhere between the pocket cam and pocket recorder market segments, Olympus has introduced a hybrid called LS-20M. The concept is simple: offer a full-HD pocket cam capable of recording good-quality audio, making the LS-20M the first real competitor of the Zoom Q3 HD, which is currently the only product in this market segment…

The battle between the two, promises a lot: while Zoom is the leading manufacturer of pocket recorders with the H2, Olympus is the leader of dictation systems. Moreover, Olympus is also one of the leading manufacturers in the cameras/lenses market, so it might become a serious challenger for Zoom, and even for the top dogs in the pocket cam market like Kodak, Cisco, Sanyo and Sony.

In The Box

Olympus LS-20M

Olympus included almost everything you can expect inside the box. Besides the device, you’ll find a battery, a 2GB SD card and a dual-function USB cable. The cable will be useful to transfer all data recorded on the LS-20M to your computer, and also to load the battery either via the USB port of your computer or an external PSU. The package also includes the user’s manual in six different languages. And that’s it! No transport bag for the device, no wrist-strap, no HDMI cable — and, since we are complaining, the 8″ USB cable is really short…

The design is quite nice: the device is a bit thicker and longer than a smartphone but less bulky than a Zoom Q3HD (it has a finer design). It has many controls and connections on its small housing made out of different mat and glossy plastic materials in metal finish. The main colors are black and anthracite. On the top of the device, the two mics are placed on both sides of the camera under chrome-like baskets. Everything looks very serious, even if it would be more reassuring to get a silicone or padded leather case to prevent any damages in case of a fall.

Front and Side Views

Olympus LS-20M

Now it’s time to have a closer look at the device. Starting with the left side that provides a power on/off+hold switch, a connector for an optional remote control, a mic in and a headphones out on stereo minijacks. The mic input can be switched to line input and fed with phantom power, which is a decisive advantage over the Q3HD that only has a line input (making the connection of external mics impossible). With the LS-20M, you can use a shotgun mic, a lavalier mic or a good old SM58. This feature will attract users who want to use external mics — just notice that using the mic input mutes the internal mics, so don’t expect to be able to mix both signals…

Olympus LS-20M

On the right side, a switch allows you to toggle between audio/video modes while a small slot allows you to access the SD card. The bottom side of the device includes a miniUSB and a HDMI connector hidden behind a blind plate. Everything looks pretty good, and this also applies to the rear side, which provides an access to the battery, a tiny 2/3″ speaker (it’s not a ghetto blaster but it’s convenient for raw monitoring in quiet environments), and a thread insert allowing you to mount the LS-20M on a camera stand… instead of a microphone stand, which would be more convenient in most cases.

Olympus LS-20M

Between the two mics on the top side of the device, you’ll find a LED indicating signal overloads and (surprise!) the lens of the camera. The surprising position of the camera changes the handling of the device quite radically. To shoot what is happening in front of you, you have to hold the LS-20M horizontally —not in parallel to your body like with most pocket cams— and aim at the scene you want to capture like you would do with a remote control. At first glance this seems more intuitive.

Now let’s take a closer look…

Conclusion

The LS-20M provides good quality video and high quality audio recording. With numerous useful options, especially in the video department, the LS-20M is a dangerous competitor for the Zoom Q3HD. The awkward position of the camera is certainly its main disadvantage in many situations: except in some rare occasions (shooting above a crowd or recording people who are seated while you’re standing), the camera position is not very practical and makes things more difficult for the user. Now you have all the information you need to choose between these two rivals or you might even consider a third solution: an Apple iStuff plus a microphone kit. It’s up to you…

 Advantages:
  • Nice overall look
  • Seems rather rugged
  • Compact size (it even fits inside your hip pocket)
  • Picture quality on the same level as the best pocket cams on the market, but with a much higher sound quality
  • No need to switch to macro mode for close-ups
  • Video pickup angle wider than most other pocket cams
  • High-quality sound with detailed high frequencies
  • Many audio and video settings and functions
  • All-in-one concept: cam, field recorder, webcam, multimedia jukebox
Drawbacks:
  • Position of the camera — more disturbing than advantageous
  • Few accessories: no protection bag, no wrist-strap, etc.
  • The small buttons are not backlit and their silk screen is hardly readable
  • Many buttons, many menus for a somewhat old-fashioned design
  • Renaming files and folders is impossible
  • High-frequencies a bit too sharp, slight lack of low-end

To read the full detailed article with video demos please see:  Olympus LS-20M Review

August 2, 2011

Interview with Ned Douglas

Filed under: Recording reviews — Tags: , , , , , — audiofanzine @ 4:40 pm

Last may, we heard that Dave Stewart (Eurythmics) and Mick Jagger had teamed up to form SuperHeavy, an international “superband” also formed with Joss Stone, Damian Marley and AR Rahman (film composer, who previously scored Slumdog Millionnaire among others ND). We’ve had the chance to interview producer Ned Douglas, chief engineer at Weapons of Mass Entertainment recording studio but also responsible for some recordings and production on this new “super album”.

Ned Douglas & Dave Stewart

Hi Ned! I know that you’re Dave’s staff programmer and engineer @ Weapons of Mass Entertainment in Hollywood. You’ve been using the Dangerous D-Box along this project. How did you end up setting up this gear for Dave Stewart’s album?

I’ve been working with Dave for a long while now, nearly 15 years as engineer and programmer. He’s marvelously creative and diverse and it’s a constantly exciting and interesting job. When we moved studios last year I took the decision of losing the desk we had as the faders used to stay a zero most of the time, it felt like it was a waste of space. Having the D-Box has allowed me to keep an analog stage in chain which is great, it has also means I have quick hands on access to talkback, headphone levels and inputs.

How did this idea of creating a “supergroup” come to reality?

Dave has always done a lot of work with Mick Jagger and the idea of forming the supergroup came out of a session they did together about 3 years ago. We’d worked with Mick and Joss Stone previously on the Alfie soundtrack so we knew they sounded great together. Damian and AR were then brought in as they wanted to create a truly international and diverse musical project. Everyone worked together for a two week session at Hensons’s studio (former A&M in Hollywood, NDA) where the bulk of the tracks were laid down, all the songs were created in the studio and born out of jam sessions with the band. The rest of the album was finished in a variety of countries and studios (including on a boat and a Caribbean island) and I’ve been involved throughout the entire project. Keeping track of it all has been white a mission !

With all these sessions in different places around the world, how did you manage to keep a coherent vibe and sound ?

With the bulk of the band tracks being laid down in the initial monster jam sessions the backdown of the songs has remained pretty consistent (thanks to Damian’s drummer and bass player). For most of the the sessions we’ve done elsewhere we’ve had it so that we can pull up a great sounding mix with almost noplugins from a stereo Protools session (thanks to engineer Cliff Norrel!) which meant that I could turn up with a laptop anywhere and have things as they were last heard with full access to the parts. A lot of what we did outside of Henson’s were vocals and lyrics but some  songs like “Beautiful People” and “Warring People” were started fully programmed on my laptop and then recorded with the band later.

For this project, did you have to deal with unusual instruments or recording moments ?

AR Rahman has an interesting setup, he uses a midi controller called a Continuum which allows him to do micro tonal performances. He had it hooked up to a Indian sound module (who’s name I forget) which had the most bewildering midi setup I’ve ever seen. It meant he could do some pretty cool stuff though.

Now let’s take a closer look…

Is there something you try to achieve in every work you do ? From an artistical, technical and human point of view, which aspects in the music production process do you take care about the most ?

Something that I’ve learnt from working with Dave Stewart is that the vibe in the room in pretty much the most crucial element (and he’s a master at it) and that nothing kills a vibe faster than having to wait on technical reasons. Being able to quickly interpret people’s ideas and get things sounding exciting with a minimum of fuss is essential. In essence: people do their best work when relaxed and having fun and if that means recording in someone’s living room on a SM58 to get the best vibe then that’s the way it should be done (especially during the writing process).

The most important lesson I have learned however is: Always be in record!

As long as you can remember, what was your best studio moment ? And your worst technical nightmare ?

Be able to witness the writing process with great songwriters, as I’ve been lucky enough to, is always a magical experience. To start the day with nothing but an empty session and finish it with a song, created from thin air is something that never gets boring. Technically working in Jamaica has posed a few moments, unreliable power, blown speakers and the like… We ended up hiring a sound system from the local DJ when we worked with Shakira out there !

To read the full interview see:  Interview with Ned Douglas

July 28, 2011

Mysteries of Dynamics Processing Revealed

Filed under: Compressors, Processors, Recording reviews — Tags: , , , , , — audiofanzine @ 7:34 pm

A dynamic processor is something that outputs a signal, where the level of the outgoing signal is based on the level of the incoming signal. In other words, a loud signal coming in will come out differently than a quiet signal coming in.

Basic types of Dynamic Processors

Compressors: The most common – the louder the signal is coming in, the less level it provides going out. In a compressor, a target level is set – called the “threshold” – and any signal coming in that exceeds that level will be reduced. The higher the level is above that threshold, the more reduction will occur. More on this later.

 

Limiters: Limiters are like super compressors. The idea is to ensure that the level does not exceed the threshold. Because this amount of compression is extreme, a limiter relies on certain functions and design that regular compressors do not have.

Expanders: The quieter the signal is coming in, the less level it provides going out. In other words – it makes quiet signals even quieter. Much like a compressor, the threshold is set at a certain level. Any signal that does NOT exceed that threshold is reduced, and the quieter the signal, the more reduction is done.

 

Gates: Gates are like super expanders. Anything that does not exceed the threshold is reduced to inaudible. Again, because gates are extreme, they often require a slightly different design than a regular expander.

 

Now – I’ll focus primarily on Compression, because that’s going to be the most commonly used dynamic processor.

Compression

Every signal you hear is compressed??? Yes, every signal you hear is compressed.

Bare with me. Imagine you have a rapper in front of a microphone. The rapper raps, you record. You play it back. You haven’t used any processing – you’re just playing back the raw vocals.  You are listening to a signal that has gone through at bare minimum 3 stages of compression – and more likely than not – closer to 6.

  • The microphone capsule gains tension as the rappers voice pushes it – in other words – it pushes back. The more the rapper’s voice pushes in – the harder the capsule diaphragm pushes back. In other words, the louder the signal is hitting the capsule, the more reduction the capsule does to the signal. That’s compression! (It’s mild compression, but it’s still compression).
  • Along the way through the microphone, you may hit a tube. Tubes have a non-linear response to voltage – the response is quite curved, and also changes the frequency balance of the signal. This is called saturation – which will tend to “round out” a signal, by reducing the loudest peaks. Compression! And before leaving the microphone, the signal may hit a transformer as well, which will saturate in a similar way… more compression.
  • The preamp is going to have multiple stages of saturation – and often times, the more gain you give something – the deeper that saturation curve goes. In other words, the more you drive the signal at the preamp, the more compression the signal experiences.
  • Then the sound has to actually come out of the speaker cones. Well, those speaker cones are going to build up tension when pushed further. See where this is going? This is called “cone compression”.

Ok – so this is a bit of a simplification – but there’s a point here. The point is that “compression” is always part of the signal. Some mics have less of it, some have more – same with speakers, tubes, transformers, etc. And they all do it in different ways. With tubes, people will talk about their saturation curves and %THD (total harmonic distortion – or frequency alterations). With mics, people will refer to how it “grabs” a sound – or more specifically – the sound’s shape.

Now let’s take a closer look…

Maximum Punch

There is a thin line between a transient sound, and a sustained sound. A sound that holds for any noticeable amount of time is sustaining. A sound that moves by too quickly to register as it’s own moment is transient. But transients can vary in length. A transient can be half a millisecond or it could also be ten milliseconds; they won’t sound the same. A big factor in punch is how long that transient exists. A quick transient sounds “spikey” – but a long transient sounds “punchy.” You want to find the point that makes the transient exist as long as possible before “flattening out” or becoming a sustained sound. Only your ear can tell you where that point is.

 

Good samples are already shaped to have that kind of impact – and any additional compression may actually soften that. Of course, punch has a lot to do with frequency as well – but that’s for another article.

 

Now what about the release? The release is super elusive. It determines how long it takes for the compressor to let go. If the release is too short for the signal you are going to get a disjointed sounding shape which usually results in distortion. If it’s too long, your signal never really returns to its natural shape, and you generally lose tone (or you just get permanent drive on the compressor’s output, giving the whole signal a new bit of tone). So the idea is to find a point that emphasizes the sustain (which is where most of the signals tone lives) properly.

 

Lastly, when the attack and release are set in a way that seem to argue – the compression can become very audible. You either hear the decent or the ascent of the signal level. This is called pumping. It’s generally annoying, but can sometimes be used an effect. If audibly desired, consider the rhythm of the release time, and ask yourself if it’s groove is complimenting the song.

———————————-

So, rather than thinking of a compressor as something that effects the “level” of a signal. Think of a compressor as something that effects shape. Why? Because level can be controlled with the volume fader more accurately and transparently. A fader doesn’t really control shape, unless you are being extremely meticulous. Conversely, compression will always effect the shape of the sound it is working on.

Once you start hearing shape, you will understand compression.

To read the full detailed article see:  Mysteries of Dynamics Processing Revealed

July 15, 2011

Vocals Processing Tips: Part 2

Hard disk recording techniques have affected every aspect of recording, including vocals. Although overdubbing vocals has been a common technique for years, today’s programs let you do multiple tracks of vocals, and make a “composite” with all the best bits. We’ll cover how to do that, then talk a bit about compression and reverb.

Composite Vocal Tracks

Cutting and pasting has benefited vocals, as you can do multiple takes, and splice the best parts together to make the perfect “composite” vocal. Some producers feel that stitching together vocals doesn’t produce as natural a “feel” as a take that goes all the way through from beginning to end, while others believe that being able to choose from multiple takes allows creating a vocal with more range than might occur with a single take. If you want to try composite vocals, here are the basic steps.

Record the Takes

Record enough takes so there’s plenty of material to piece together a good performance (loop recording is particularly handy for doing vocals). While you’re in a recording mood, record a little bit of the track without any input signal. This can be handy to have around, for reasons described later.

Audition the Takes

Audition each take, and isolate the good parts (by cutting out unwanted sections). I recommend setting loop points around very short phrases.

Solo each take, one after the other. If you’re not going to use a take, cut the phrase. If a take is a candidate for the final mix, keep it.

Pick the top 3 or 4 candidates, and remove the equivalent sections from the rest of the tracks. Now repeat this procedure, phrase by phrase, until you’ve gone over the entire performance and found the best bits

Ligne de chant compilée

In Sonar, several takes of vocals have been recorded. A mute tool has muted portions of each track (the waveforms are shown as shaded), with the remaining parts making up the final vocal.

Next, listen to combinations of the various different phrases. Balance technical and artistic considerations; choose parts that flow well together as well as sound technically correct. Sometimes you might deliberately choose a less expressive rendition of a line if it comes just before an emotional high point, thus heightening the contrast.

Once you have the segments needed for a cohesive performance, erase the unused parts. If you want to archive everything “just in case,” go for it. But if after putting the part together you think it could be better, you might be better off re-cutting it than putting more hours into editing.

Ligne de chant compiléeSeveral takes of vocals were recorded into Cubase SX, and edited to create one final vocal. The program shows the elements that make up the final vocal by highlighting them in green.

Bounce the Takes

This isn’t absolutely necessary, but converting all the bits into a single track simplifies subsequent editing and processing.

Before bouncing, play the tune through from start to finish and match the segment levels as closely as possible. Also check the meters for any send bus or master bus the tracks are feeding, and adjust levels (if needed) so there’s no distortion. Generally, the bounced track will be derived from a bus or master; if there’s distortion, the bounced track will have distortion too.

This is also where the recorded noise might come in handy. Sometimes I’ve had to do a quick fade on the end of one segment, and a fade in on the beginning of another, leaving a dead silent gap between phrases. Layering in a bit of the noise signal gives better continuity, and keeps the part from sounding too “assembled.”

After everything’s set, implement the program’s bounce or mix to hard disk function. You can typically bounce to an empty track, or “render” the audio to disk and bring it back into the project.

Edit the Composite Track

At this point, I bring the composite track into a digital audio editor for clean-up. Here are some typical processes:

  • Phrase-by-phrase gain adjustments. If a phrase has mismatched levels, use the program’s level change DSP or mix automation to fix the problem.
  • Fix breath noises and inhales. There might be “flammed” inhales from combining two different takes, so cut one. However, don’t eliminate all inhales and breath noises — they keep things “human.”
  • Add overall dynamics control, reverb, EQ, echo, etc. if needed. Do not add these while cutting individual takes; it will be much harder to match the effect, and in the case of reverb, tails might get cut off. Adding processing after optimizing the entire track will give the best results.

Tidy Up Your Hard Disk


After the vocals are done, check how your program deals with deleting unused segments, as this can reclaim significant space from your hard drive.

Now let’s take a look at compression…

Reverb Tips for Vocals

Nothing “gift wraps” a vocal better than some tasty reverb. My favorite reverb for voice is a natural acoustic space, but as reverb rooms are an endangered species, you’ll likely use a digital reverb. Reverb settings are a matter of taste, but two parameters are particularly important.

Waves RVerb (Renaissance Reverb)

A reverb’s Predelay and Diffusion parameters are crucial to getting good vocal sounds. This reverb, the RVerb plug-in from WAVES, offers an exceptional amount of control.

Diffusion: With vocals, I prefer low diffusion, where each reflection is more “separated.” Low diffusion settings often sound terrible with percussion, as the individual echoes can have an effect like marbles bouncing on a steel plate. But with vocals, the sparser amount of reflections prevent the voice from being overwhelmed by too “lush” a reverb sound.

Predelay: This works well in the 50-100 ms range. The delay allows the first part of the vocal to punch through without reverb, while the more sustained parts get the full benefit of the reverberated sound.

To read the full article see: Vocals Processing Tips Part 2

March 17, 2011

Panning Laws Revealed

The idea of panning seems pretty obvious, right? You turn a panpot (real or virtual) to place a sound somewhere in the stereo field…

But ignorance of the law is no excuse – in this case, panning laws. These laws govern exactly what happens when a monaural sound moves from left to right in the stereo field, which can be different for different pieces of software. As a matter of fact, not knowing about panning laws can create some significant issues if you need to move a project from one host to another. Panning laws may even account for some of the online foolishness where people argue about one host sounding “punchier” or “wimpier” than another when they loaded the same project into different hosts. It’s the same project, right? So it should sound the same, right?

Well, not necessarily…keep reading.

Origins of Panning Laws

Panning laws originated in the days of analog mixers. If there was a linear gain increase in one channel and a linear gain decrease in the other channel to change the stereo position, at the center position the sum of the two channels sounded louder than if the signal was panned full left or full right.

To compensate for this, it became common to use a logarithmic gain change response to drop the signal by -3dB RMS at the center. You could do this by using dual pots for panning with log/antilog tapers, but as those could be hard to find, you could do pretty much the same thing by adding tapering resistors to standard linear potentiometers. Thus, even though signals were being added together from the left and right channels, the apparent level was the same when centered because they had equal power.

But this “law” was not a standard. Some engineers preferred to drop the center level a bit more, either because they liked the signal to seem louder as it moved out of the main center zone, or because signals that “clumped up” around the center tended to “monoize” the signal. So, dropping their levels a little further created more of an illusion of stereo. And some of the people using analog consoles had their own little secret tweaks to change the panning characteristics.

Panning Meets the Digital Audio Workstation

With virtual mixers we don’t have to worry about dual ganged panpots, and can create any panning characteristic we want. That’s a good thing, because it allows a high degree of flexibility. But it also adds a degree of chaos that we really didn’t need.

For example, Cubase SX3 has four panning laws in the Project Setup dialog; you get there by going Project > Project Setup.

 

Loi de panoramique dans Cubase

The default pan law for Cubase is to drop the center by –3dB, which is the classic equal power setting.

 

Setting the value to 0dB eliminates constant-power panning, and gives the old school, center-channel-louder effect. Since we tried so hard to get away from that, it’s not surprising that Cubase defaults to using the “drop the center by -3dB” classic equal power setting. But you can also choose to drop the center by -4.5dB or -6dB if you want to hype up the extremes somewhat, and make the center a bit more demure. Fair enough; it’s nice to have options.

Adobe Audition has two panning options in multitrack mode, accessed by going View > Advanced Session Properties.

Loi de panoramique dans Audition

Adobe Audition lets you choose from two common panning laws.

L/R Cut Logarithmic is the default, and pans to the left by reducing the right channel volume, and conversely, pans to the right by reducing the left channel volume. As the panning gets closer to hard left or right, the channel being panned to doesn’t increase past what its volume would be when centered. The Equal Power Sinusoidal option maintains constant power by amplifying hard pans to left or right by +3dB, which is conceptually similar to dropping the two channels by -3dB when the signal is centered.

Now let’s take a closer look…

Conclusion

We can’t sign off without mentioning one more thing: The pan law you choose isn’t just a matter of convenience or compatibility, although I’ve stressed the importance of being compatible if you want to move a project from one host to another. The law you choose can make a difference in the overall sound of a mix.

This is less of an issue if you use mostly stereo tracks, as panning in that case is really more of a balance control. But for many of us, “multitrack” still means recording at least some mono tracks. I tend to record a mono source (voice, guitar, bass) in mono, unless it’s important to capture the room ambience – and even then, I’m more likely to capture the main sound in mono, and use a stereo pair of room mics (or stereo processing) that go to their own tracks. And if you pan that mono track, you’re going to have to deal with the panning laws.

In any event, you now know enough about those laws to make sure you don’t get cited for contempt of court. Happy panning!

To read the full detailed article please visit: Panning Laws Revealed

December 17, 2010

Gifts for Geeks

Clock is ticking, and there is still time to please and be pleased. Here are some ideas for Christmas gifts for musicians and gear heads to fit all tastes and wallet sizes.

Computer Music

Line 6 MIDI Mobilizer

Line 6 MIDI Mobilizer : and your iThing speaks MIDI

Together with an Apple iPhone, iPad, or iPod touch, and the free MIDI Memo Recorder app, MIDI Mobilizer can play, record, and backup MIDI information any time, any place. Whether you want to capture a quick musical idea or back up the settings of all your MIDI gear, MIDI Mobilizer is a simple and compact solution for everything MIDI.  Price: $70

Peavey AmpKit Link

Peavey AmpKit Link :

Turn your iPhone into a virtual amp for $30. The sound quality is fair considering the price. The marketing strategy of offering a free amp and then have us pay for additional amps is not so bad, considering that guitar players usually have their favorite amps and do not play with 15 different models.

Plugin Lexicon

Plugin Lexicon :

The new software package makes all the effects processing of Lexicon’s PCM96 available as a plug-in designed to add “inspirational new sounds to a user’s DAW that are not available anywhere else.”  The PC- and Macintosh-compatible PCM Native Effects Plug-In Bundle is designed to work with DAWs like Pro Tools and Logic, as well as with any other VST, Audio Unit or RTAS-compatible host.  Price: $1200.

Apogee One

Apogee One : All in one in your pocket

ONE is described as a single input, stereo output USB music interface designed to work seamlessly with Apples iTunes, GarageBand, Logic, Final Cut or any Core Audio compliant application on a Mac. Unlike any product in its category, ONE features an internal reference condenser microphone, ideal for capturing inspired musical moments, according to Apogee. ONE also includes a microphone preamp, an instrument input for guitar, bass, and keyboards, and a studio-quality stereo output for headphones or powered monitors.  Price:  $249

 

Native Komplete 7

Native Komplete 7 : The Bundle of the Decade?

The latest version of the Komplete bundle combines a range of NI products, while the Komplete 7 Elements collection is designed to set a new price point for music production enthusiasts on a budget.  The seventh generation of Komplete now comprises 24 individual products, including the latest Reaktor 5.5 version as well as the new Reaktor Prism, Rammfire, Reflektor, Traktor’s 12 and Vintage Organs. Other products now contained in Komplete include the Abbey Road 60s Drums vintage drum library, the performance effect The Finger, the electric pianos and an electric bass by sampler Thomas Scarbee, the four acoustic pianos from the Classic Piano Collection, the cinematic Acoustic Refractions instrument and the Reaktor Spark synthesizer, amounting to about 10,000 sounds and 90 GB of studio-grade sample material overall.  Price: $559.

Guitar Pro 6

Guitar Pro 6 :

Version 6 is definitely a major update for Guitar Pro. What used to be a small software tool has become the ultimate reference in its category thanks to its intuitive user interface, well thought-out features and an absurdly low price. Should you upgrade your previous Guitar Pro version for $29.95? Yes, a thousand times yes! You’ll benefit from a better design and a much better sounding and efficient audio engine than in previous versions. Should you buy the full version for $59.95 if you don’t own a guitar tab editor? Yes, a thousand times yes!

Pro Tools 9

Pro Tools 9 : Compatible Soundblaster (among others) !

Pro Tools 9 is an open platform that doesn’t require an Avid/M-Audio interface anymore, but can work with or without any Core Audio or ASIO compatible interface – on Mac AND PC.  The new version enables bigger mixes with more tracks, and pro features including Automatic Delay Compensation, multitrack Beat Detective, full Import Session Data dialog, DigiBase Pro, and other separately priced add-ons—now standard.  Price: $599 for the full version.

Pianoteq Play

Pianoteq Play :

Pianoteq Play is a virtual piano based on the physically modeled Pianoteq software instrument, appraised by many musicians for its close intimacy and responsiveness.

Modarrt says there is no need to tweak settings and parameters, as Pianoteq Play is delivered with “perfectly designed instruments.”  Pianoteq Play supports all Pianoteq instruments, and the grand pianos K1, C3, and M3 are embedded.  Price:  $99

RME Babyface

RME Babyface :

RME succeeded in launching a compact and rugged interface with remarkable sound quality. At about $750, this baby provides two quality mic preamps and converters, ADAT in/out, a jog wheel, a transport bag, and a pair of nice-looking VU-meters. Add TotalMix FX —the virtual mixer that allows you to manage all 22 channels and process the signals (EQ, filter, reverb, and echo)— to the package and you get the best mobile audio interface on the market.

Akai APC 20

Akai APC 20 : Enter the Matrix

Yes, the APC40 is much more comprehensive than the APC20! But if you have only $200 for a Live controller, the APC20 has only one competitor in the form of the Novation Launchpad. The latter is less expensive but doesn’t have any faders, which makes it less interesting…

DJing and Live Sound

Traktor Kontrol S4

Traktor Kontrol S4 :

Combining an extended version of the existing Traktor Pro software with a dedicated hardware controller, the Traktor Kontrol S4 is aiming to provide an all-in-one solution for digital DJs. The controller comprises a four-channel mixer, an integrated 24-bit/96kHz audio interface based on NI’s Audio 4 DJ, and interface sections for looping, cueing, track browsing and effects control.  Price: $1000.

Hercules DJ Console 4-MX

Hercules DJ Console 4-MX :

Hercules launched this year the newest version of their DJ Console line for Pro DJs, the DJ Console 4-Mx, a controller featuring large jog wheels (each equipped with touch sensor) a built-in audio interface tailored for DJing, and control over 2 and 4 virtual decks.  The DJ Console 4-Mx has steel and aluminium crafted body with a variety of controls including 89 controls in 2-deck mode and 150 controls in 4-deck mode.  Price: $450.

Pioneer DJM-2000

Pioneer DJM-2000 :

Let’s be clear: this is a great piece of gear! Well thought-out, nicely finished and with a great sound, it offers countless possibilities to allow the most demanding DJ’s to have endless fun. With this product, Pioneer targets night clubs with big budgets who want to offer the best to their DJ’s. The latter will have the possibility to prepare their sets before performing, and to come to the club with only a CD or a USB key — no need for a computer.  Price: $2500.

Denon DN-X1700

Denon DN-X1700 :

The DN-X1700 is a four-channel tabletop mixer with rubberised knobs, 60mm Alps K Series channel faders, 45mm FLEX cross fader, a color LCD display, extended 24-point LED channel and output metering, and LED ring metering around the control knobs.  In operation, the principal features related to the power and flexibility of the DN-X1700 are its Matrix Input Assignment with digital input and MIDI/USB audio, independent and parametric three-band EQ with Kill on each channel, and dual independent EFX processors.  Price: $1800.

Fender Passport 500 Pro

 

Fender Passport 500 Pro :

The eight-channel Passport 500 PRO is the new top-of-the-line Passport system:

  • A port that lets you record your performance with CD quality (.wav) straight to a USB flash drive.
  • CD-quality .wav and mp3 file playback.
  • Sub-out jack for an external powered sub-woofer.
  • Redesigned speaker system with 10″ woofer and improved clarity.
  • Price: $1000.

 

Presonus StudioLive 24.4.2

Presonus StudioLive 24.4.2 :

StudioLive 24.4.2 sports the same user interface, feature set, and I/O configuration as the StudioLive 16.4.2 but with several additions and enhancements. The main difference is that the new mixer provides 24 input channels and 10 aux buses, whereas the StudioLive 16.4.2 has 16 channels and 6 auxes. In addition, the new mixer’s Fat Channel has fully parametric EQ, rather than semi-parametric, and the gate and limiter have been enhanced. Instead of one stereo 31-band graphic EQ on the main bus, you get four dual 31-band graphic EQs that can be assigned to the mains, subgroups, and aux buses.  Price: $3,300.

To see many more gift ideas see:  Gift for Geeks- Xmas Shopping 2010

June 9, 2010

Audio Mastering

Tom Volpicelli of The Mastering House answers the top 10 common questions about mastering.

What is mastering and the role of the mastering engineer?


Mastering is essentially the step of audio production used to prepare mixes for the formats that are used for replication and distribution.  It is the culmination of the combined efforts from the producer, musicians, and engineers to realize the musical vision of the artist.  Each stage of the audio production process, from pre-production to mastering, builds on each other and is dependent on the previous process.  Mastering is the last opportunity to make any changes to positively affect the presentation of your music before it evolves from a studio environment to the outside world.

An awareness of the differences between the roles of mixing and mastering engineers should be noted.  While the tools may be similar, the perspectives between mixing and mastering are very different. When mixing, the focus is on the internal balance of individually recorded tracks and effects used both sonically and creatively for a single piece of music.

An album cannot be heard in its entirety until the job of a mix engineer is completed. The mastering engineer picks up where the mix engineer leaves off. Mastering is geared toward creating the balance required to make the entire album cohesive. The mastering engineer is most concerned with overall sonic and translation issues.  A mastering engineer works with the client to determine proper spacing between songs and how songs will be ordered on the CD. The flow of an album must appeal to the listener; it should engage them and take them on a musical journey as determined by the artist. Any final edits will be addressed during the mastering process as well.

Finally, the role of the mastering engineer is to provide preparation and quality control of the physical media send to the plant for replication.  This includes listening to the premaster CD to verify integrity, along with the more technical aspects such as encoding text, UPC/EAN and ISRC codes, checking for errors within the media and providing any necessary documentation such as a PQ list.

Is mastering always necessary?

A writer’s words are not complete until the editor approves them. A painter’s work is not complete until it has been matted and framed.  A musician’s work requires the same treatment. Audio production should not be rushed, finished haphazardly or completed “just to get it out there”. A finished product should reflect all of the work of the artist, producers and engineers that carry that vision forward.  Even a “perfect” mix needs mastering to a degree. In this case, you want the mastering to be as transparent as possible so that the original sound is maintained while preparing it for the final media.

As mentioned earlier, it is difficult for a mixing engineer to know how an entire album will sound in its entirety while mixing an individual track. In some cases a given track may be perfect on its own.  However, when that track is placed within the context of an album, slight adjustments in level or frequency balance may be required.  Given the amount of music distributed online, an album needs to stand out from start to finish to be noticed in such a competitive market. If the final goal is to create a product that is ready to be played on the radio, distributed online, or sold as a physical product, it should be mastered.

Mastering helps say something about the professionalism of the artist, from the arrangement of certain styles of songs to the volume of the recording to the pacing of the tracks. If an artist is serious about their music, they should make sure that someone with experience signs off on the finished product.

What kind of improvements can be expected from mastering?

Mastering can help to achieve the correct balance, volume, and depth for a style of music. It can add clarity and punch to music, giving it more vitality.  The idea behind mastering is that a product will sound better after it is treated by the mastering engineer. The degree with which a mastering engineer can achieve this is dependent on the given mixes. In some cases there may be limitations or compromises that need to be made.

One limitation of mastering is the inability to restore severely distorted material. Distortion in a mix is like corrosion; once present it cannot easily be removed and has permanently destroyed a part of the material.  While mastering can mask the effect of some types of distortion, it is essentially covering blemishes that should be addressed before the mastering stage. A common misconception is that mixes should be as “hot” as possible. With the advent of 24 bit digital technology there is no reason why mixes have to “go into the red.”

Most mastering engineers recommend a cushion of anywhere between -6 to -10 dBFS from peak level to help ensure that clipping does not take place and to allow room for processing.  In addition to peak level, the crest factor (peak-to-average ratio) is very important. While dynamic range can always easily be reduced, it is very difficult to undo the effects of over compression or limiting.

If the internal balance of a stereo mix is off, there may be compromises in the sound of the mastered track that will need to be made. For example, if cymbals or a vocal is very sibilant and bright while other parts of the mix are dark, it can be difficult to balance the overall sound in a way that enhances all elements.

In addition to frequency, levels between tracks may also be an issue. If the mastering engineer is given a stereo mix (as is usually the case) specific individual components of the mix cannot be completely isolated and processed separately.  While there are techniques such as de-essing, mid/side processing, equalizing or compressing for a specific imbalance, the results will likely not be as good as with a mix not having these issues and allowing the mastering engineer to address the balance on the whole.

One method of getting around internal balance issues is to provide alternate mixes. Some examples are vocal up/down mixes or mixes where one EQ is favored over another. Another method is supplying the mastering engineer with “stems” or sub mixes of the stereo track.  These might include a separate stereo mix of the vocals or instruments that when summed together are the same as the stereo mix minus any stereo bus processing.  In this case the mastering engineer is placed slightly in the role of a mix engineer and can make adjustments that wouldn’t be possible with a stereo mix alone. Another advantage with using stems is that alternate masters can easily be created such as radio edits, instrumental and vocal-only masters.

Another area where “fixing it in the mix” is better than “fixing it in mastering” is when dealing with the issue of noise. Mute automation on individual tracks should be used where there are noises during sections of a track that are not contributing to the mix.  Some examples are electric guitar hum/buzz on intros, outros, and breaks, bleed from headphones on the vocal track when the vocalist is not singing, drummers laying down their sticks after cymbals have faded but while other instruments are still playing at the end of a track.

Should you choose an engineer based on their “style”?

nevermindTen different mastering engineers working in the same room with the same equipment will create ten totally different masters, each sounding great on their own.  If you ask those same engineers to go back and reproduce any given master, you are likely to get ten almost identical masters back.  While each individual mastering engineer has his own style, it is important that he is able to separate himself from his style when needed.  An engineer should never let his personal taste interfere with the goal of the artist he is working with. Again, this is where communication with the client is a crucial element.

A good mastering engineer should be well versed in a variety of different categories of music. In general, there is no reason why an engineer known for creating great Country albums cannot produce a great Rock album.  While an engineer’s work should be able to transcend musical genres, if a mastering engineer has a certain style that is appealing to you as the artist, you should consider working with him.  It is important that both the engineer and the artist can communicate in a way that is complimentary to both individuals.

Which is more important, a technical background or musical one?

A mastering engineer should be well versed both technically and musically. The craft of the engineer is to be able to know good music and know how to make that music sound better.  Still, while a technical background is extremely important in the mastering world, that background should not interfere with the aesthetics.  Likewise, any personal feelings an engineer has about the stylistic choices of the music he is mastering should ultimately be discussed with the musician. It is because of this that an engineer’s musical background should not hinder his craft.

Given a technical background, some mastering engineers are capable of making modifications to equipment to create a more transparent sound, or provide color according to their taste and needs.  Having a musical background, particularly in the area of pitch, allows an engineer to identify frequency issues relating to musical notes and can speak directly to the musician about these issues in their terms.

An engineer should make sure that he strays away from favoring either background. While most engineers come from one or the other, their craft is in combining the two.  A mastering engineer should remain as objective as possible while still providing necessary feedback and insight from both a musical and technological perspective.

To read the full detailed article see:  Audio Mastering

March 25, 2010

[MUSIK MESSE 2010] – RME – Babyface

For all Musikmesse news, videos and coverage see here:  Musikmesse 2010

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