Ask A Sound Guy

Archive for the ‘Q & A’ Category

If you read yesterday’s post, you know that AASG has a new writer, Andrew Levine. He’ll be popping in from time to time with articles regarding recording and otherwise awesome audio stuff.

One of the questions we posed on the AASG Google Wave group was this: Would you, at any time in your production workflow, normalize your audio? The response to the question was good and definitely varied. Andrew decided to weigh in on the subject via this post, so read it and feel free to comment. If you comment with your email, I may just send you a Wave invite!

Regarding the question of when to normalize audio I’d like to put that in perspective with regards to my work-flow.

When I do the final down-mix of a project (mostly in the domain of classical music, jazz or avantgarde), with material at the final sample rate and a dither plug-in in the last slot of the master bus (to reduce to 16 bit for CD-A), I check the peak levels of the loudest passages to make sure they stay below -0.2 dBFS.

You need some headroom when you plan to possibly trans-code the audio, e.g. to an mp3, as peaks can crop up and you definitely don’t want to go above 0 dBFS.

If you plan on sending the file to someone for mastering, or if you are bouncing a file that is to be placed in a compilation of material from other projects, it is useful to preserve headroom in the range of 3 to 6 dB–without clipping the material! Bounce to 24 bit fixed or 32 bit float and don’t maximize levels before you know where this piece of music will fit in.

If I see that there are occasional (and few!) peaks that would either create overs or force me to lower the overall level there are two choices: manually and musically lower the level of the clipping passages (I’ll pick up the topic of “automated gain riding” in a followup) or insert a brick-wall limiter in the processing chain (before the dithering plug-in).

So, when do I personally normalize?

If I have a file that was rendered with some headroom and want to hand it on I normalize it (to -0.2 dBFS) before reducing the bit depth and transcoding it. If I want to send off a small group of similar files I’ll do the same, normalizing all audio to the same level, so that the recipient has an easier job. And that’s about it :-)

Question: How do I get the other outputs on Ultrabeat to work when using the Multichannel version of the plugin?

So Ultrabeat is a monster of a plugin, but there are a few simple tricks to using that save a lot of time and make the plugin really flexible for something that comes bundled with a DAW.

Setting up a multichannel version of Ultrabeat allows you to group your samples and play them out of ultrabeat into different aux tracks.  I chose to have kicks come out of the main track, snares out of another aux, toms a second aux and cymbals out of a third. Basically I can put different compressors and EQ’s on the kicks than I do on the snares.  Then I can tweak the hi-end of the cymbals and gate the toms – all on separate channels.  Pretty cool, right?  Check out the video!

The Question: I have heard that dithering is essential, when changing the sample rate of an audio file. Why?

The Short Answer: Dither is low-level (as in quiet) noise that gets added to a signal to aid in the conversion of “bit-depths.”  Basically you only NEED dither when you’re going from one bit-depth to another. For example you’ll want to include dither when going from 24-bit to 16-bit.

One important note: You do NOT need to add dither when only converting sample rates – like 48 kHz to 44.1kHz sample rate.

One important tip: You’ll get better results if you do sample rate conversions of mixes second-to-last and then bit-depth conversions last.

The Longer Answer: Dithering is a procress by which you can increases the theoretical potential dynamic range of an audio system.  A 16-bit file has potentially about 96dB of potential dynamic range (bit depth * 6). Through more math and physics than I want to get into (because it still confuses me at times) dithering can increase that potential dynamic range significantly – by making resolution clearer in the quieter range of sounds. It basically helps the reduce the distortion inherent in bit-depth reduction – think of the sounds you get when you put a bit-crusher on a track.

(I’ll try to add more to this later, but it’s going to take some delving into my old school notes and a bit of book reading.)

Blue Cat Audio released news today that they’ve added full 3D visualization to two of their plug-ins: FreqAnaylist and StereoScope Pro. This feature is offered as a free upgrade to those who already own the plug-ins and are registered.

Blue Cat is also offering a special promotion for the rest of August. Demos of the plug-ins are available in VST, Audio Unit, and DirectX from Blue Cat’s website.

Here’s a list of new features within the updated plug-ins:

- New 3D waterfall view to monitor the evolution of the spectrum or stereo field over time.
- Drag the rulers to move the curve when zoomed (can keep measurement/selection mode and still drag the curves).
- Output automation is now disabled by default.
- Updated documentation.
- Bug Fix (PC): Cubase freezes when loading a preset using a different skin while the plugin window is open.

So even if you’re not into the whole 3D thing, it looks like Blue Cat still gave you more than one reason to upgrade your plug-ins.

Now and for the rest of August, you can buy FreqAnaylist and StereoScope Pro for $75.65 from their online store. Registered Blue Cat Audio members will recieve an additional discount, as well.

I got this question in my inbox today and it seems like something a lot of people might have this very same question.

Thanks
Will

This is a problem that many DJs face, and why all hardware mixers have gain knobs!

Let’s review how a track usually goes:

• A track is crafted. Parts are written, drums are programmed, etc etc.
• During the mix, individual tracks will usually be compressed and, depending on the mix, you might see some group compression (like drums in a parallel compression scheme), and maybe even some buss compression on the master channel.
• After the track is mixed, it’ll need to be mastered. Hopefully the track is being sent to a qualified mastering engineer (like Sanjay or I). From there, the mastering process takes place. The engineer will take the final, mixed, stereo track and use frequency-specific compression as well as limiting among other things (EQ, other processors like the Oxford Inflator).
• Keep in mind that mastering processes are different depending on the medium. Most notably, mastering engineers will have somewhat different processes for mastering to vinyl and to CD/digital.

When you record a mix inside of a DAW, the reason the tracks all sound different in terms of volume is because every mastering engineer does things slightly differently. Also, when downloading tracks offline (especially from mp3 blogs), you’ll likely get a mix of vinyl rips and CD/digital copies. Lets take a look at two different waveforms.

This waveform is from a 12″ that I recently recorded into Audacity thru my Mbox:

Notice how, even thought it’s limited, it’s more of a sausage waveform. Now how do you think it’d sound if mixed into the next waveform which is a digital release I bought from Beatport:

Yeah. The first waveform is going to sound imposibly quiet when compared next to the second. So how do you fix this when recording a mix? I usually end up placing a limiter on the master channel to even everything out if I’m mixing inside Live 7. If I’m mixing on my CD decks with my mixer, then I use the gain knob.

Remember: you DO have gain control inside Ableton Live. Just double-click your audio clip and bring the gain up/down as you see fit:

Subsequently, if you’re sequencing a mix in the Arrangement view, there’s always automation!