This is a basic Compression Chart for student reference; this can be used as a basic guide when approaching the task of processing different types of audio material with Compression. This reference will be periodically updated. Remember this is guide and not definitive, nothing is set in stone, but it should prove useful.

John Harding

Website: www.musicproducerpro.net

If your serious about music production i am sure your using Windows XP and opt to skip Windows Vista from being installed or upgraded on your Music Production Computer. (I know i did.)

With Windows 7 it seems like Bill G. has got the company back in a decent playing field with musicians. Its my understanding that the Microsoft team work closely with the music producers & video gamers this time around and did drastic changes that many had been requesting for some time now.

If you have real world experience running Win XP / Win 7 in your music studio let us know what kind of difference / benefits you have discovered since making the transition. There are tons of variables out there to take into consideration when even starting this debate but i am just looking for a “ballpark” remote idea of how things are going for others.

To help better serve others in the FP community list a few music programs your running, ram memory, and operating system version
so we can all get a “real life” idea of what’s going on out here on the frontline of music studios.

Note: Nobbs: (You may have bought a PC with Vista, but that doesn’t mean that was the best OS to be running on your computer. A downgrade was really called for if you ran into the dozens of Audio / Video issues plagued by Vista (Some issues may have been occurring with or without your knowledge). Matter fact, if you didn’t have any easily detected issues running Vista in your studio Windows XP could have saved you major time & money by clocking in a lot faster as a OS. If you think your computer runs fast with Vista, with XP it would be Twice the speed.)

I am not telling you anything to “make” you change the Operating System on your studio computer. If you don’t know what your doing with Software / Hardware technology forget messing with your computer. You must safely backup data, beats, recording sessions, software licenses, software programs, vst plug ins all before “tampering” with OS software. If you don’t know how to “properly” backup Everything you need i advise you to enjoy your current setup or get with a Music Production / Computer Guru who can help you.

If you have 1st hand experience with Win 7 or both Win XP & 7 let us know your findings.

For information visit – www.musicproducerpro.net

Twitter now has a retweet link you can click to retweet tweets. This will make tweets going viral on Twitter much easier. Could be good news for bands when it comes to “#musicmonday”.

New website of loops at technomusic.me

This is a basic EQ Chart for student reference; this can be used as a basic guide when approaching the task of processing different types of audio material with EQ. This reference will be periodically updated throughout the term. Remember this is guide and not definitive, nothing is set in stone, but it should prove useful.

John Harding

POD PRODUCTIONS

Learn Logic, Live, production and mixing from professional musicians.

If you are just beginning or an seasoned producer looking for some new tricks we have a wealth of experience to offer on:

- Genre and non genre approach to making electronic music
- Sequencing
- Midi Mashing
- Synthesis
- Effects
- Tracking / Recording
- Music Theory
- Sampling
- Live Electronic Performance

Electronic music making is a deep and exciting new world of music a lot of people are keen on these days and we are very passionate and love to share the knowledge with like minded people!

We have lots of releases in the fields of:

- Trance
- Dubstep
- Drum n Bass
- IDM
- House
- Live electronic
- Breakcore
- Jungle
- Funk
etc etc!

We have played along side artists including

Aphex Twin, AndyC&GQ, Pendulum, Kosheen, Marky, Monolake, Dizzee Rascal, Rage against the machine, Dkay & Stamina MC, MIA, High Contrast, Shpongle, Coldcut, Cut Chemist, AdamF&Fresh, Krust+Die, PlumpDJ’s, Ltj Bookem, Ed Rush n Opticall, Infusion, Amorpheus Androgynous, Hallucinogen, Koolism, Phil K, Concord Dawn, 28-Days, Hilltop Hoods, Silverchair, The Bird, Ruffneck Souljah, Regurgitator, Bulletproof…….

Our rates are $50 an hour

Location: Melbourne Australia

CONTACT: trimaxis@gmail.com

Today, we will all learn how to easily stream audio online… FREE.  Okay, but before we even get to that point, we have to create some ORIGINAL audio and in order to do that, we will be using FL STUDIO 9.

What I would like y’all (yes, y’all, I’m from TEXAS!) to do is log on to the Windows partition in Boot Camp and open up the FL STUDIO 9 program.  Here, I would like for you to create a simple drum track with a BPM between 90-100.  Feel free to add an instrument or two using the Piano Roll feature (or grab a USB MIDI Keyboard and playout your sounds).  Create 2-3 PATTERNS and then switch from PAT (PATTERN) to SONG.  Go to View and select PLAYLIST and layout your patterns to have a few change ups.  Repeat and add different patterns until you have  atleast a One Minute track.

I will be selecting a few of your audio tracks to add to the Class Recordings section of this blog, so take your time and come up with a banger.

Once you are done with that, go to FILE and then EXPORT.  Once you roll your mouse over EXPORT, a  drop down menu will give you format options.  Click on MP3 and save your file to a folder on the desktop with YOUR NAME.  (you may need to create this folder)

Once you have the ORIGINAL MP3 created, go to www.divshare.com and create a free account.  Use this website to upload your MP3 and use the DASHBOARD link to find your different sharing options (embed, direct link, etc.)

I will be here if you have any questions or need help.  Now get to producing!  ♥

Bedroom Beats is a brand new tutorial website offering the tips and techniques pioneered in modern dance music productions. The site is sorted in to different sections each filled with professional knowledge and free advice. The blog is predominantly aimed at the more upfront styles of dance music – Breakbeat, Drum n Bass, Dubstep and House. The weekly info varies from quick tips and videos to full length professionally written reports. For those involved in audio recording or computer music, Bedroom Beats offers a “sound” foundation to brush up your skills.

Make Beats

Live recording of your tracks is finally possible. Choose your set-up, add, remove and arrange the devices you need and record your track to the ‚myFiles’ section of your Hobnox account. From there, publish and share. The first step towards timeline & sequencer, saving arrangments & remixing tracks is done!

Check it out HERE.

P.s. Here’s a tutorial to get you started!

Bedroom Beats is a new tutorial website offering the tips and techniques pioneered in modern dance music productions. The blog is arranged in to different sections each full with expert knowledge and free tips. The site is mainly aimed at the more upfront genres of dance music – Breakbeat, Drum n Bass, Dubstep and House. The weekly info varies from quick tips and videos to full length professionally written reports. For those interested in home recording or computer producing, Bedroom Beats offers a “sound” platform to brush up your skills.

Audio Production

Starting a new audio samples remix music production website. (maybe). I’m coding it to see what it would look like. It’s not that i want to do it but i feel it’s something i have to do, lol. If it fails i’ll just scrap it and fall back to the old website.

The thing to remember is running an audio samples website costs more because of all the downloading. It’s not just blog text. This is why most people don’t get into it.

ADDITIVE

Does what it says on the tin. By adding one or more basic and simple waveforms together and their harmonics you create a complex waveform. However, you need to add an enormous amount of harmonics to create the simplest of sounds and this type of synthesis can be complicated to create in the form of a synthesizer but the Kawai K5000 does exactly that. You can create extremely rich textures or wild and crazy sounds on this beast. Personally, I love additive synthesis but then again I am receiving very intense therapy. The process of additive synthesis is also referred to as summing the waveforms and harmonics. This method adopts Fourier analysis. Described as the representation of a sound’s frequency components as a sum of pure sinusoidal waves . An analysis of a sound’s frequency components is taken at a steady state to give an approximation of that sounds spectrum. As most natural sounds are spectrally dynamic, one single Fourier analysis could not possibly represent a sound in sine waves. By ‘windowing’, a Fast Fourier Transform (FFT) takes several of these approximations and strings them together to better predict a sound’s spectrum over time. Although this is daunting to take in it is crucial to know and I don�t expect you to understand Fourier analysis but just thought I would bring it in now as we will come back to this at the advanced stages of these tutorials.

SUBTRACTIVE

This process involves the generating of complex waveforms and then filtering the frequencies so that you are then left with the sound you want. You take away the frequencies. Obviously the filters are crucial in subtractive synthesis and the better the filters and the wider the choice of filters available, the better the end result will be.

FREQUENCY MODULATION (FM)

The output of one oscillator (modulator) is used to modulate the frequency of another oscillator (carrier). These oscillators are called operators. FM synthesizers usually have 4 or 6 operators. Algorithms are predetermined combinations of routings of modulators and carriers. To really explain this I would have to go into harmonics, sidebands, non-coincident and coincident series and the relationships between modulators and carriers. So I won�t go there. What I will say is that FM synthesis can create lovely digital type of sounds, from brittle to lush. A little bit of info for DX owners, is that the oscillators on these synthesizers were all sine waves.

PHYSICAL MODELLING (PM or PHM)

This form of synthesis simulates the physical properties of natural instruments, or any sound, by using complex mathematical equations in real-time. This requires huge processing power. You are not actually creating the sound but, you are creating and controlling the process that produces that sound. Waveguides and algorithms come into this process heavily but, again, I won�t go into that. What I do find interesting is that the Nord Lead uses PM synthesis to emulate an analogue synthesizer. Weird huh?

LINEAR ARITHMETIC SYNTHESIS

This type of synthesis takes short attack sampled waveforms called PCM, or Pulse Code Modulation and combines them with synthesized sounds that form the body and tail of the new sound. By layering these and combining them with the synthesized portion of the sound you arrive at the new sound. The final sound is processed by using filters, envelope generators etc. This is one of the most common forms of synthesis used in the 90s and even today. Roland were the most famous for adopting this type of synthesis and the D50 was one of the most common of the synthesizers that used LA synthesis. By the way, a great synthesizer and still used today.

WAVETABLE SYNTHESIS

This form of synthesis incorporates the use of pre recorded digitized audio waveforms of real or synthetic instruments. The waveforms are then stored in memory and played back at varying speeds for the corresponding notes played. These waveforms usually have a looped segment which allows for a sustained note to be played. Using envelopes and modulators, these waveforms can be processed and layered to form complex sounds that can often be lush and interesting. The processes are algorithmic and memory is crucial to house the waveforms. I could get into linear crossfading sequentially, quasi-periodic and sine functions etc. But I won�t.

GRANULAR SYNTHESIS

This is the method by which tiny events of sounds (grains or clouds) are manipulated to form new complex sounds. By using varying frequencies and amplitudes of the sonic components, and by processing varying sequences and durations of these grains, a new complex sound is formed. Simply put, this form of synthesis creates some crazy and interesting sounds.

ADVANCED VECTOR SYNTHESIS ( AVS )

This method of synthesis incorporates the combining and processing of digital waveforms. Using PCM samples, effects and filtering this method of synthesis can create stunning sounds, from lush and evolving pads to strange stepped sequences. Korg made the famous Wavestation range of synthesizers and these were based around the Sequential Circuits Prohet VS. Working off a 2 dimensional envelope using an X and Y axis (joystick) and 4 voices, this synthesizer also had wave sequencing, playing a loopable sequence of PCM samples in a rhythmic and/or crossfaded fashion. The idea was to be able to crossfade 2 or more waveforms using the joystick. Freaky but incredible.

Synthesizers used to be categorized into 2 categories, analogue and digital, but in the modern world we have the hybrid digital analogue synthesizer. Confused? Don�t be. It actually makes good sense. Whether the processing is digital or analogue, the sounds are produced in much the same way, albeit with different terminology and some different routings. A good example of the terminology being adapted is: VCO, voltage controlled oscillator becomes a DCO, digitally controlled oscillator.

Of course, with today’s influx of synthesizers, be they analogue, digital or a hybrid system, we are offered many varying forms of synthesis, from granular to transwave and so on.

Ultimately, synthesis is about sound design and manipulation and we now have fully fledged systems whereby effects are incorporated into the signal matrix, dynamics at the output stage and modulation matrices that make modular synthesizers look primitive.

Extract taken from Synthesis Tutorials – Part 3

Personal 1-2-1 Tuition Courses

Eddie Bazil (Zukan)

www.samplecraze.com

While we are on this subject, wouldn’t it be great if you could extract quantise templates from other sources and use them in your own music? What if I told you that you could extract a groove quantise template from an audio loop? Would that excite you a touch?

Most decent audio/sequencing packages like Cubase will offer you an ‘Extract Groove or Create Groove Quantise from Hitpoints’. Hitpoints are the same as markers. You apply these hitpoints to the peaks in the audio file and then use ‘Create Groove Quantise from Hitpoints’ in the audio menu. Of course, the software you are using will have different terminology or tools, but they all do the same thing.

You can then use this quantise groove template exactly as you would any other groove template within the quantise menu option.

This tool is extremely effective when dealing with drum loops as drum loops have distinct peak points (where the kick, snare, hi hats etc peak), so it makes it easier to extract grooves and create templates.

Additionally, you can import groove templates from other sources and use them as exactly the same as using a pre defined template. There are many companies that now offer groove templates of your favourite quantise templates. MPC drum quantise templates spring to mind.

Here is an example of how to extract and create a groove template from a piece of audio. The principles are the same for most software so try to transcribe the tools and terminology across to your software. I am using a very famous piece of audio; a drum loop from Missy E…………………..

Excerpt taken from The Beat Production Bible.

Personal 1-2-1 Tuition Courses

Eddie Bazil (Zukan)

www.samplecraze.com

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What if I told you that you could extract a groove quantise template from an audio loop? Would that excite you a touch? <o:p></o:p></p> <p class=”MsoNormal”> <o:p></o:p>Most decent audio/sequencing packages like Cubase will offer you an ‘Extract Groove or Create Groove Quantise from Hitpoints’. Hitpoints are the same as markers. You apply these hitpoints to the peaks in the audio file and then use ‘Create Groove Quantise from Hitpoints’ in the audio menu. Of course, the software you are using will have different terminology or tools, but they all do the same thing. <o:p></o:p></p> <p class=”MsoNormal”> <o:p></o:p>You can then use this quantise groove template exactly as you would any other groove template within the quantise menu option. <o:p></o:p></p> <p class=”MsoNormal”> <o:p></o:p>This tool is extremely effective when dealing with drum loops as drum loops have distinct peak points (where the kick, snare, hi hats etc peak), so it makes it easier to extract grooves and create templates. <o:p></o:p></p> <p class=”MsoNormal”> <o:p></o:p>Additionally, you can import groove templates from other sources and use them as exactly the same as using a pre defined template. There are many companies that now offer groove templates of your favourite quantise templates. MPC drum quantise templates spring to mind. <o:p></o:p></p> <p class=”MsoNormal”> <o:p></o:p>Here is an example of how to extract and create a groove template from a piece of audio. The principles are the same for most software so try to transcribe the tools and terminology across to your software. I am using a very famous piece of audio; a drum loop from Missy E…………………..</p> <p class=”MsoNormal”>Excerpt taken from <a href=”../product.php?xProd=160&amp;xSec=5″>The Beat Production Bible</a>.</p> <p class=”MsoNormal”><a href=”../section.php?xSec=79″>Personal 1-2-1 Tuition Courses</a> <br> </p> <p> Eddie Bazil (Zukan) </p> www.samplecraze.com <br> Design html Code Preview Use Shift+Enter for a <BR> tag guidelines visible Paste From Word… Insert/Edit a Hyperlink… Link To A Document… Paste From Word… Image Properties… Insert/Edit a Hyperlink… Link To A Document… Paste From Word… Bookmark Properties… Paste From Word… Table, Row and Cell Properties… Add Row Above Add Row Below Add Column to the Left Add Column to the Right Delete Row Delete Column Merge with Cell to the Right Merge with Cell Below Unmerge with Cell to the Right Unmerge with Cell Below Insert/Edit a Hyperlink… Link To A Document… Default Arial Times New Roman Courier Georgia Verdana Geneva Default 1 2 3 4 5 6 7 Normal Heading 1 Heading 2 Heading 3 Heading 4 Heading 5 Heading 6 Pre Formatted Address Clear Styles highBox (highBox) blue (blue) // <p><b>WARNING:</b> JavaScript must be enabled to use this form.</p>
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Time stretching is the process of changing the speed or duration of an audio signal without affecting its pitch. Pitch scaling or pitch shifting is the reverse: the process of changing the pitch without affecting the speed.

If you take a sample and assign it to a key in a sampler and then play the sample further up the key span you will invariably hear the sample get higher in pitch and shorter in length. It is the same as playing a record at a faster speed. The pitch changes and the time it takes for the record to play from beginning to end is far shorter than at the original speed.

Time stretching is simply a mathematical process and there are many software that provide time stretching as a standard function. For time stretching to work sensibly you need to know the original tempo of the sample so as to then advocate a new tempo for the sample to be stretched to or compressed to (time compression). Be aware that stretching a sample too much will cause all sorts of anomalies and sometimes these anomalies are used as an effect (robotic effect used in Electro genres etc).

Let’s take at a simple drum beat at 80 bpm and time stretch it to 125 bpm. Simply changing the pitch of the sample to speed it up will not work as the sounds within the beat will also be pitched up. We therefore must time stretch so as to maintain the pitch of the sounds within the drum beat…………..

Excerpt taken from The Beat Production Bible.

Personal 1-2-1 Tuition Courses

Eddie Bazil (Zukan)

www.samplecraze.com

The following tutorial is based on not having your studio acoustically treated, and is only a guide to try to maximise the best assets in mixing in a home studio.

Sound Waves

Let us have a very brief look at how sound travels, and how we measure it’s effectiveness.

Sound travels at approximately1130 feet per second.

Now let us take a frequency travel scenario and try to explain it’s movement in a room.

For arguments sake, let’s look at a bass frequency of 60 Hz.

To understand what I am talking about here in regards to Hz, please read my Synthesis tutorial on Sound.

When emitting sound, the speakers will vibrate at a rate of 60 times per second.   Each cycle (Hz) means that the speaker cones will extend forward when transmitting the sound, and refract back (rarefaction) when recoiling for the next cycle.

These vibrations create peaks on the forward drive and troughs on the refraction. Each peak and trough equates to one cycle.

Imagine 60 of these every second.

We can now calculate the wave cycles of this 60 Hz wave.   We know that sound travels at approximately 1130 feet per second, so we can calculate how many wave cycles that is for the 60 Hz wave. We divide 1130 by 60, and the result is around 19 feet (18.83 if you want to be anal about it). We can now deduce that each wave cycle is 19 feet apart. To calculate each half cycle, i.e. the distance between the peak and trough, drive and rarefaction, we simply divide by two. We now have a figure of 91/2 feet. What that tells us is that if you sat anywhere up to 91/2 feet from your speakers, the sound would fly past you completely flat.
However, this is assuming you have no boundaries of any sort in the room, i.e. no walls or ceiling. As we know that to be utter rubbish, we then need to factor in the boundaries.

Are you still with me here?

These boundaries will reflect back the sound from the speakers and get mixed with the original source sound. This is not all that happens. The reflected sounds can come from different angles and because of their ‘bouncing’ nature; they could come at a different time to other waves. And because the reflected sound gets mixed with the source sound, the actual volume of the mixed wave is louder.

In certain parts of the room, the reflected sound will amplify because a peak might meet another peak (constructive interference), and in other parts of the room where a peak meets a trough (rarefaction), frequencies are cancelled out (destructive interference).

Calculating what happens where is a nightmare.

This is why your mix will sound poor.

This is why it is crucial for our ears to hear the sound from the speakers arrive before the reflective sounds. For argument’s sake, I will call this sound ‘primary’ or ‘leading’, and the reflective sound ‘secondary’ or ‘following’.

Our brains have the uncanny ability, due to an effect called the Haas effect, of both prioritizing and localizing the primary sound, but only if the secondary sounds are low in amplitude. So, by eliminating as many of the secondary (reflective) sounds as possible, we leave the brain with the primary sound to deal with. This will allow for a more accurate location of the sound, and a better representation of the frequency content.

But is this what we really want?  

I ask this, because the secondary sound is also important in a ‘real’ space and goes to form the tonality of the sound being heard. Words like rich, tight, full etc. all come from secondary sounds (reflected). So, we don’t want to completely remove them, as this would then give us a clinically dead space. We want to keep certain secondary sounds and only diminish the ones that really interfere with the sound.

Our brains also have the ability to filter or ignore unwanted frequencies. In the event that the brain is bombarded with too many reflections, it will have a problem localizing the sounds, so it decides to ignore, or suppress, them.

The best example of this is when there is a lot of noise about you, like in a room or a bar, and you are trying to have a conversation with someone. The brain can ignore the rest of the noise and focus on ‘hearing’ the conversation you are trying to have. I am sure you have experienced this in public places, parties, clubs, football matches etc.

To carry that over to our real world situation of a home studio, we need to understand that reflective surfaces will create major problems, and the most common of these reflective culprits are walls.   However, there is a way of overcoming this, assuming the room is not excessively reflective, and is the standard bedroom/living room type of space with carpet and curtains.  

We overcome this with clever speaker placement and listening position, and before you go thinking that this is just an idea and not based on any scientific foundation, think again. The idea is to have the primary sound arrive to our ears before the secondary sound. Walls are the worst culprits, but because we know that sound travels at a given speed, we can make sure that the primary sound will reach our ears before the secondary sound does. By doing this, and with the Haas effect, our brains will prioritize the primary sound and suppress (if at low amplitude) the secondary sound, which will have the desired result, albeit not perfectly.

A room affects the sound of a speaker by the reflections it causes. We have covered this and now we need to delve a little more into what causes these reflections. Some frequencies will be reinforced, others suppressed, thus altering the character of the sound. We know that solid surfaces will reflect and porous surfaces will absorb, but this is all highly reliant on the materials being used.
Curtains and carpets will absorb certain frequencies, but not all, so it can sometimes be more damaging than productive. For this, we need to understand the surfaces that exist in the room. In our home studio scenario, we are assuming that a carpet and curtains, plus the odd sofa etc, are all that are in the room. We are not dealing with a steel factory floor studio.

In any listening environment, what we hear is a result of a mixture of both the primary and secondary (reflected) sounds. We know this to be true and our sound field will be a combination of both. In general, the primary sound, from the speakers, is responsible for the image, while the secondary sounds contribute to the tonality of the received sound. The trick is to place the speaker in a location that will take of advantage of the desirable reflections, while diminishing the unwanted reflections

Excerpt taken from EQ-Uncovered

Personal 1-2-1 Tuition Courses

Eddie Bazil (Zukan)

www.samplecraze.com