There is No Wrong Answer in Music Composition
Writing music is one of those things you can do and never make a mistake. Some melodies are catchier than others are, and everyone will write some bad stanzas. It's all right; that's why we have revisions. Remember this while writing music: it will make you feel better and help you avoid writer's block.
Do Something Different
It's easy to get stuck in a rut and all of your songs begin to sound the same. Even if you've found a great combination of notes or a catchy beat, changing it can be good and help you grow as a composer.
An easy way to try something new is pick up an instrument you haven't played before. Sometimes you find yourself playing the same old keys or strumming the same chords. A different instrument can lead you to melodies you may not have thought of otherwise.
Practice, Practice and More Practice
There is no substitute for hard work and practice—it is the only formula that will guarantee you will become a better songwriter.
A digital audio workstation is an electronic system designed solely or primarily for recording, editing and playing back digital audio.
A sound card or audio card is an internal computer expansion card that facilitates the input and output of audio signals to and from a computer under control of computer programs.
Audio Stream Input/Output is a computer sound card driver protocol for digital audio, providing a low-latency and high fidelity interface between a software application and a computer's sound card.
Steinberg Cubase & Nuendo & Wavelab
Avid Pro Tools
Sony Acid & Sound Forge
PreSonus Studio One
:Audio Signal Processing:
is the intentional alteration of auditory signals, or sound, often through an audio effect or effects unit. As audio signals may be electronically represented in either digital or analog format, signal processing may occur in either domain. Analog processors operate directly on the electrical signal, while digital processors operate mathematically on the digital representation of that signal.
DSP: Digital Signal Processing
VST: Virtual Studio Technology
AU: Audio Unit (Apple)
RTAS: Real Time AudioSuite (Digidesign)
AVID: Advancement Via Individual Determination
LADSPA: Linux Audio Developers Simple Plugin API
An EQ or Equaliser is a filter than allows you to selectively cut or boost a certain frequency. Some EQ’s have fixed frequencies while others allow you to select these for yourself (these are known as Parametric Equalisers).
All Parametric EQ plugins will have three basic controls, a frequency selector, a Q or bandwidth selector and a Gain (this selects the amount the filter will cut or boost the desired frequency). Many also feature frequency graphs to help you visualise the sound.
At this point the best way to get a feel for how EQ can help you is to run a sound through an EQ plugin in your sequencer. One of the best techniques for identifying weak or problem frequencies is to set your gain control to a high boost and then sweep the frequency control through its spectrum. You’ll hear certain frequencies may sound pleasant when boosted or conversely some may jump out at you as being too harsh. Remember that cutting frequencies can be just as valuable as boosting!
One great technique you can use to help the low frequencies stand out in your mix is to use EQ to cut low frequencies from your lead sounds, try cutting frequencies from 60-200Hz to help create more space in your mix.
Let’s take a look at the controls you’ll find on most compressor plugins.
Threshold - this determines the volume level (in dB) that a sound must attain before the compressor takes effect. When this level is exceeded the signal is reduced (gain reduction).
Ratio - this control determines the amount of gain reduction applied to the signal once the compressor has kicked in. A Ratio setting of 1:1 would mean that no gain reduction would be applied, where as a setting of infinity:1 would mean that the signal would never rise about the Threshold level (this is known as Limiting).
Attack – this control dictates the time that the compressor takes to reduce the gain. Fast attack times will reduce the gain immediately where as slow attack times will leave the first portion of the sound untouched fading the gain reduction in. This is a classic technique used to enhance the qualities of percussive sounds.
Release – this setting will determine the length of time that it takes the compressor‘s gain to return to normal once the signal has fallen below the Threshold setting.
Some compressors have an Auto attack/release button that selects suitable settings for the incoming sound.
Hard Knee/ Soft Knee – this control dictates the character of the compression used, a Hard Knee setting will instantly apply the full amount of gain reduction to the signal once it has exceeded the Threshold. Soft Knee will allow the effect of the compressor to be more gradual.
Side Chain – this function allows you to use an external signal to control the compressor. This effect is frequently used in electronic music. For example producers will often feed the bass part from a song into a compressor whilst routing the kick drum sound through the compressors side chain input. This will have the effect of "ducking" the bass sound when the kick drum plays, producing a rhythmic pumping effect.
is the acoustic environment that surrounds a sound. Natural reverb exists everywhere. Whether the space being described is a bathroom or a gymnasium, the essential characteristics remain the same.Reverb is composed of a series of tightly-spaced echoes. The number of echoes and the way that they decay play a major role in shaping the sound that you hear. Many other factors influence the sound of a reverberant space. These include the dimensions of the actual space (length, width, and height), the construction of the space (such as whether the walls are hard or soft and whether the floor is carpeted), and diffusion (what the sound bounces off of). In addition to natural reverb, software synthesis of reverberation is also possible. Many audio card s, synthesizer s, dedicated effects processors, and digital audio applications can create reverb, simulating both natural and supernatural environments. For example, one could create the reverb for a room fifty feet long, five feet wide, with a four-foot ceiling, lined with carpet.The synthesis of reverb by a digital signal processing (DSP) algorithm usually attempts to mimic the way a real acoustic space works. The algorithm designers simulate the early reflections, the compounding of echoes, and the decay of high versus low frequencies when designing their product. Of course, the more processing power and speed available, the more complex and potentially realistic a reverb signal can be created.
is an audio effect which records an input signal to an audio storage medium, and then plays it back after a period of time.The delayed signal may either be played back multiple times, or played back into the recording again, to create the sound of a repeating, decaying echo.
to simulate the effect of reverberation in a large hall or cavern, one or several delayed signals are added to the original signal. To be perceived as echo, the delay has to be of order 35 milliseconds or above. Short of actually playing a sound in the desired environment, the effect of echo can be implemented using either digital or analog methods.
creation for unusual sound , a delayed signal is added to the original signal with a continuously variable delay (usually smaller than 10 ms).Using a delay line creates an unlimited series of equally spaced notches and peaks.
The electronic phasing effect is created by splitting an audio signal into two paths. One path treats the signal with an all-pass filter, which preserves the amplitude of the original signal and alters the phase. The amount of change in phase depends on the frequency. When signals from the two paths are mixed, the frequencies that are out of phase will cancel each other out, creating the phaser's characteristic notches.
a delayed signal is added to the original signal with a constant delay. The delay has to be short in order not to be perceived as echo, but above 5 ms to be audible. If the delay is too short, it will destructively interfere with the un-delayed signal and create a flanging effect. Often, the delayed signals will be slightly pitch shifted to more realistically convey the effect of multiple voices.
An audio filter is a frequency dependent amplifier circuit, working in the audio frequency range, 0 Hz to beyond 20 kHz. Equalization (EQ) is a form of filtering. In the general sense, frequency ranges can be emphasized or attenuated using low-pass, high-pass, band-pass or band-stop filters. Band-pass filtering of voice can simulate the effect of a telephone because telephones use band-pass filters.
effects such as the use of a fuzz box can be used to produce distorted sounds,
such as for imitating robotic voices or to simulate distorted radiotelephone traffic.The most basic overdrive effect involves clipping the signal when its absolute value exceeds a certain threshold.
this effect shifts a signal up or down in pitch. For example, a signal may be shifted an octave up or down. This is usually applied to the entire signal, and not to each note separately.
the opposite of pitch shift, that is, the process of changing the speed of an audio signal without affecting its pitch.
to change the frequency or amplitude of a carrier signal in relation to a predefined signal.
more info coming soon -->
Audio normalization is the application of a constant amount of gain to an audio recording to bring the average or peak amplitude to a target level.
emphasize harmonic frequency content on specified frequencies.
The basis of distorting a sound is to increase the harmonic content of the incoming signal. This basically means it is creating more sound and thus increasing the volume.Normally characterized on bass, yet can be creatively useful on many elements in music.
Bit Crushers lower the bit rate of an incoming signal, achieving a more lo-fi, ‘Gritty’ and in some cases ‘harsh’ texture.
A transient shaper allows you to change the ADSR volume of each hit by detecting the transient of each drum in your track and then boosting or reducing the volume over the length of your drum hit, then starting again as soon as the next hit comes through it.
There are two types of tremolo. a)rapid reiteration-of a single note, particularly used on bowed string instruments and plucked strings such as harp, where it is called bisbigliando or "whispering". between two notes or chords in alternation, an imitation of the preceding that is more common on keyboard instruments. Mallet instruments such as the marimba are capable of either method.a roll on any percussion instrument, whether tuned or untuned. b)variation in amplitude-using electronic effects and effects pedals which rapidly turn the volume of a signal up and down, creating a "shuddering" effect an imitation of the same by strings in which pulsations are taken in the same bow direction.
Vibrato is a musical effect consisting of a regular, pulsating change of pitch. It is used to add expression to vocal and instrumental music. Vibrato is typically characterised in terms of two factors: the amount of pitch variation and the speed with which the pitch is varied.
This effect is in essence the same as distortion. Yet saturation works at a lot softer rate, allowing you to dial in subtle degrees of distortion without introducing harshness. This also allows you to drive the sound harder as you don’t have to worry about introducing unwanted artefacts that distortion can sometimes have.
Exciter can add life and high end harmonic content to an otherwise dull signal, bringing out the harmonics and brightness needed. And all without filling a sound with too much treble.
A limiter acts in a similar way to a compressor, except that nothing can exceed its threshold. They are sometimes referred to as a ‘brick wall’ as nothing can get past. They are normally used in mastering a final track to bring up the overall volume of a track, but can be useful and creative in the mixing stage also.
when you split your audio signal into small pieces of around 1 to 50 ms. These small pieces are called grains. Multiple grains may be layered on top of each other, and may play at different speeds, phases, volume, and frequency, among other parameters.
Is an audio processor that captures the characteristic elements of an an audio signal and then uses this characteristic signal to affect other audio signals. The technology behind the vocoder effect was initially used in attempts to synthesize speech. The effect called vocoding can be recognized on records as a talking synthesizer.