Punching in & Punching out! A guide for dummies

Hi everyone and welcome to this week's article, in which we celebrate our article n.200 AND our Facebook page fan n. 500!

Today we're focusing on an option featured in almost every Daw (although in the pic above we are showing the Cubase/Nuendo u.i. version): the Punch in (and Punch out) funcion.

This funcion basically lets you replace a section of a take on the fly, by deciding before when to start recording and when to stop.

In the Cubase/Nuendo interface you can see, in the transport panel, a section in which you can type the positioning of the left and right locators (otherwise you can set them manually by dragging them in the top part of the interface). There is also a downhill icon and an uphill one: activate both those icons to tell the Daw that you want it to start recording at the left locator and stop when the marker reaches to the right one. 

Now "arm" the track you want to record in by clicking in the red button of the channel, set your marker anywhere prior the beginning of the area that will be re-recorded and press play (not record, play). 

The song will start playing, until the moment it will reach the left locator, then it will start recording in the armed track, and it will stop when it will reach the right locator, but the song will go on playing.

Why is this function useful? 

Essentially for one reason: let's imagine we have tracked the perfect guitar take, and there is only one small flaw to re-do, but we can't re play it on its own, it would lose the swing of playing the whole take together: we can re-play the whole part and tell the Daw to record just the small bit we need to re-do, and if we're precise enough in setting our locators, we won't even need further editing.

If the new track appears to be BELOW the old one, thus we can't hear it, we can right click to the old track and select it from the drop down menu, choosing it and clicking on To Front.

Additional info:

If we select only the left locator from the transport panel, it will go on recording until we press stop.

One last thing: in the transport panel, in the locators section,  under the numbers that sets where the locators are, we can also find a smaller number: this is the PRE ROLL COUNT, it makes a pre-count before playing, but make sure you don't have it in the metronome too, otherwise the DAW will give priority to that one.
In the Preferences section, finally, we can also decide the Daw to stop playing when the marker reaches the Punch Out locator.


Become fan of this blog on Facebook! Share it and contact us to collaborate!!

Review: TSE X50 v.2.4 (with audio sample)

Hello everyone and welcome to this week's article!

Today we are reviewing a new, very interesting guitar suite, from a talented producer (Onqel of the norwegian company Tse Audio) that I have followed with interest since his first freeware plugins (and still today I use some of them with great satisfaction).

Tse X50 2 is the first commercial product of Tse Audio, and it is the result of years of debug on the free versions, done also with the help of the Ultimatemetal Forum Community.

The software, in this version has been totally re-written since the freeware versions, and now everything is part of a complete suite, with 3 stompboxes (a noisegate, a tube screamer and a Rat distortion), 2 preamps (X50, which resembles closely a Peavey 5150 head, and X30, which looks and sounds like an Engl preamp), a rack with tuner, delay and an equalizer.

Going in depth we also have a power amp section in which we can swap the power tubes among EL34 and 6L6GC, set the Bias and so on, and for the cabinet section the program lets you browse through a good choice of some of the best cabinet impulses that can be found, fruit of a collaboration among Tse and the best impulse producers (Guitarhack, Kalthallen and Rosen Digital, among the others).

When playing and recording, the plugin sounds really good (the original X30 has been for a long time my to-go metal preamp simulator), and it sums up the best that Tse has ever produced (except for their amazing bass preamp Tse B.O.D. which simulates a Sansamp, and that will surely find its place on some future bass expansion).

The plugin is very light on the cpu (especially compared with some of the other today's competitors) and features also an automatic input adjustment, that learns the peaks of your signal as you play and adjusts the input level accordigly, which is brilliant.

All in all this plugin has really everything you need and nothing you don't, focusing on few high quality features rather than becoming a huge, heavy collection of models that clogs the cpu, and it is very player-oriented, granting low latency, lightweight high quality tone.

If you must choose a single guitar amp simulator to buy in 2015, Tse X50 v.2.4 is probably one of the best ones, and we can't wait to see what the future will bring from Tse Audio!

What suggestion can we give? Well in the future, the plugin could also incorporate a simulation specific for clean and overdrive tones, like a vox or a peavey classic 30, and maybe the rack could incorporate a new module with a multi effect like phaser, flanger, vibrato and so on.

Become fan of this blog on Facebook! Share it and contact us to collaborate!!

Difference between Linear Phase Eq and Regular Eq

Hello and welcome to this week's article!
Today we're going to focus on a topic that we have mentioned in many occasions, but that we never really explained yet: what does the term Linear Phase Eq and Regular Eq mean.

Let's start with this assumption: Regular EQ’s have phase shift between the different bands. When we boost or cut a certain area, that frequency range is technically delayed by a tiny amount, due to the way the equalizer itself works.

Linear-phase EQ, instead, compensates this delay keeping all the frequencies 100% in phase (it basically anticipates the wave of a certain time to correct the phase shift). In other words, all the frequencies pass through the EQ at the same speed, resulting in zero phase issues between bands.

The basic question is: "if they have created a way to prevent the phase to shift, why don't everyone uses just linear phase eq?"

the answer is because it consumes more cpu resources and, especially, because it creates a "pre-ring", which means that by correcting the phase variation it creates a part of the wave that actually starts BEFORE the peak of the transient, and this, for certain types of sounds as for example percussive drums, it can ruin them.

The conclusion is that linear phase eq is best suited for certain instruments in which this problem is less apparent (like acoustic guitars) and less for others, and we must try it and compare it with the regular one in order to understand on which instrument it brings a benefit and on which one not.

About the mastering phase, finally, there are different schools of thought, for example many producers likes the result obtained with the Waves Lin-Mb, a linear phase multiband compressor, because they think linear phase sounds more transparent, while other engineers think that the "pre-ring" downside can ruin the mastering, especially if we are speaking of a heavily percussive track.

As always, it depends and we must trust our ears.



Become fan of this blog on Facebook! Share it and contact us to collaborate!!

Review: M-Audio Studio Pro 3

Hello everyone and welcome to this week's article!
Today we're going to review a recently discontinued product from M-Audio, that can be still found very often in various music equipment stores, it's one of the cheapest set of monitors you can find, and the street price is around 90€, for the set of 2.

One can tell the quality of an audio monitor according to its "realism", which is the capacity to let us hear what we are working on in our Daw, for example, in a way that, once we have exported the final track and we listen to it with other devices, it "translates" well, without too many differences.

Unfortunately these speakers are not realistic at all, and we can't trust them, when mixing.

Surely they are a very good and cheap tracking device, if we use them just when recording they do a good job, and from a consumer side also, they are pretty decent for movies, games, or just music listening.

The build quality is pretty solid, and each unit features a 3,25 inches woofer and an 1 inch tweeter, and the couple consists into a main unit, with an ac adaptor for power, and a satellite, which is powered by connecting it to the main unit.

One feature that I have appreciated very much is the front aux in: this way you can connect the speakers both to the audio interface and to the computer's internal soundcard, so you can record music and then watch a video, without having to switch inputs, all with the same speakers, and this is pretty rare and very comfortable.

The unit features a bass boost feature to compensate the small dimensions of the speaker and to emphasize certain music genres or films, and the volume is decent for everyday home use, but to be honest there can be moments in which 10watts per speaker are not enough. 

My suggestion when buying multi purpose speakers is to get some with a power ranging from 20 to 40w rms per speaker, in order to never feel the need for "more power".

In the end, yes, these speakers can be used as a small upgrade for the consumer user, but in all honesty I wouldn't suggest them for music production purposes other than sheer tracking.


Specs Taken from the Website:

3.25" low-frequency drivers; 1" high-frequency drivers

Magnetic shielding prevents computer/video monitor interference (RF interference, output current limiting, over temperature, turn on/off transient, subsonic filter)

Internal standing wave acoustic absorption for greater efficiency

Bass Boost switch

RCA rear-panel inputs; convenient front-panel 1/8" inputs

Frequency response: 100Hz – 20kHz ±3dB, Crossover frequency: 2.3kHz

Dynamic range (Maximum Signal-to-Noise Ratio): >80dB (typical, A-weighted)

Input connectors: left and right line input connectors (RCA)

Dynamic power: 10 watts into 4Ω (per channel) at 0.5% distortion

Indicator: blue power LED ring around on/off/volume knob on front panel

Dimensions: 5.5"W x 7.9"H x 5.9"D

Weight: 8.8lb./pair (approx., without packing)


Become fan of this blog on Facebook! Share it and contact us to collaborate!!

How to Use Pre-Recorded Backing Tracks Live 3/3

Hello and welcome to this week's article!
Today we're going to see a third and last way of using backing track, with the support of Francesco Ferrini of Fleshgod Apocalypse.

This method it's the most complex and the one which requires more hardware, but it is the most professional we can use in order to manage stage monitoring and use backing tracks.

First off all instruments (microphones, Pods, and so on) goes into a rack with several inputs, and this rack goes into a splitter. One "copy" of the outputs goes to the stage box, and from there to the external mixer, another "copy" goes to the audio interface.

Their interface (a Motu with 8 independent outputs) has a software called Cue Mix, which is a virtual mixer that manages the various inputs and outputs (as almost all audio interface have), and from there they can create a mix for each band member, and each mix will go, through one of the independent ouputs, to the in ear monitor of each band member.
Why do they use CueMix instead of doing everything directly with Cubase?
For 2 reasons:
1) CueMix has no latency, unlike Cubase
2) Because it's a better way to manage the mix that each band member wants in their in-ear monitor.

Then the band has a session of Cubase with the whole setlist, the tempo track with the metronome of each song and the backing tracks; they send the click and backing tracks through the Cubase Sends to each output of the Audio Interface without passing through Cue Mix. 

Why are those relatives mixes from Cubase passing through the Sends and not through the Master Out? Simple: because they want the metronome only in the relative mixes that will go to the in ear monitors of the band member, but only the backing track must pass through the master, and from there to the mixer, the metronome must obviously not be heard by the audience!

Finally, always with the Midi Automations of Cubase, the band controls also the patches of the Pod Hd that they use for guitars, for example to switch automatically at the right time from Rhythm patch to Solo patch, they also control the Co2 Geysers, the Strobo Lights and so on...

...but this is another topic!


Hope this was Helpful!



CLICK HERE FOR PART 1 OF 3 

CLICK HERE FOR PART 2 OF 3


Become fan of this blog on Facebook! Share it and contact us to collaborate!!

How to Use Pre-Recorded Backing Tracks Live 2/3

Hello and welcome to this week's article!

Last week we have seen how to use a backing track on a live context, but this time we must take a look on HOW to make this backing track, to load on the drummer's mp3 player.

Let's load up on our favourite DAW a project and create a midi track and and some audio one.

The audio tracks should be in mono, since only one channel of the mixer will go to the PA.

We should set the tempo track properly, with the right metronome and checking out for tempo changes inside the same tracks, leaving 8 metronome hits before the actual starting of the song, so that the drummer can hear 4 hits, then give the other band members the other 4, and then start the song.

First we make the tempo track, then we create in the midi track an event, of the lenght of the whole song, and we insert the metronome writing the midi part, with the beats and the accents, using a sound that will be easily recognizable by the drummer, even when playing live.

About the audio tracks, we will have to make a mix with all the things we want to have in the base, for example backing vocals, vocal harmonies, orchestral sections, and so on, and to make sure everything is perfectly on time with the click.

When it's time to export we must keep the audio tracks central so that the sound goes evenly both on the left and right channels, while the click part must be all panned LEFT, so in the mixer we will send the left channel on the drummer's headphones and the right channel on the P.A., and the audience will not hear the click, only the backing track.

My suggestion is to use a limiter to make the final volume even among all songs, and to make sure that all songs not only have the same levels, but also the metronome on the same side!! :)


CLICK HERE FOR PART 1 OF 3 

CLICK HERE FOR PART 3 OF 3


Become fan of this blog on Facebook! Share it and contact us to collaborate!!

How to Use Pre-Recorded Backing Tracks Live 1/3

Hello and welcome to this week's article!
This time we will talk about LIVE music, and how to integrate pre-recorded music (like orchestral parts, or backing vocals) with a real band, making sure everything is on time.

There are actually 2 methods, today we will see the easiest one, which does not involve the use of a computer, but lets us only have backing tracks in mono.

what we need is:

- an Mp3 Player (or a smartphone put in airplane mode) with the backing tracks inside

- a Y stereo minijack cable (a cable that takes the stereo sound from the mp3 player and splits it into 2 mono sounds, one with the left track and one with the right track).

- 2 adaptors from minijack to normal jack, to connect the aforementioned cable to the mixer

- a mixer, with at least 2 channels, as the one depicted on the first photo

- one jack with a jack connector on one side and an xrl to the other, to connect the mixer to the PA, usually through the channel box.

- a pair of headphones for the drummer

Once we have all these tools ready, we need obviously a backing track on our device, with on the left side the metronome track and the backing track, and on the right side just the backing track (of course you can switch them if you need, the important is that all tracks have the metronome on the same side.
Now we can connect the stereo cable to our mp3 player and send the Left Track (the one with the metronome too) in the first channel of the mixer, and the Right Track (the one without the metronome) in the second channel, as you can see from our Routing Scheme below:

In the 2 channels of our mixer now we must pan the left track full left and the right track full right in order to avoid the metronome to spill from the other track into the channel box.

Now we can connect on the left mono output the drummer's headphones, so that he can choose the song with the mp3 player and adjust the volume of the base with the mixer, and the right mono output of the mixer to the channel box, using the jack with one regular jack connector on one side (the mixer side) and the xrl connector on the other (for the channel box side). If we are connecting into another mixer without passing through a channel box, much probably we can use a regular jack.


CLICK HERE FOR PART 2 OF 3 

CLICK HERE FOR PART 3 OF 3


Become fan of this blog on Facebook! Share it and contact us to collaborate!!