The five equal, horizontal lines and four spaces are the Staff. Each individual line and space corresponds to a note. Notes above and below the Staff are placed on ledger lines. The combination of the Treble Clef Staff and the Bass Clef Staff connected by brace are known as the Grand Staff.

Notation is read from left to right. Measures include the lines of the staff onto which the notes are entered and each measure has a finite number of notes / beats per the time signature. Each measure is separated from the previous or successive measure by bar lines. The staffs are like a graph, with notes moving vertically up the staff rising in pitch and declining in pitch as they are entered or move vertically down the lower lines. In moving from left to right on the graph, time elapses as notes are played according to the time signature.

  The clef is placed at the beginning of the staff lines, just prior to the time signature, and will indicate the pitch of the note. The Clefs are: G or Treble (pictured to the far left) for higher pitched instruments, F or Bass (pictured next)for lower pitched instruments, and C (Tenor Clef and Alto Clef). The Treble Clef has its own staff lines for notes in its octave group and the Bass Clef has its own staff for notes in its octave groups.

The Time Signature is entered at the beginning of the staff lines. The most common time signature of a composition is 4/4 time. This means that there are 4 beats total per each measure. It secondly indicates that that each quarter note receives one beat. In 3/4 time, the 3 indicates three beats per measure and the 4 indicates that each quarter note receives one beat.

Notes include whole notes (which receive all the maximum beats per measure), half notes (which receive one half the maximum beats per measure), quarter notes (which receive one beat per measure). An eighth note would receive one half beat per measure. Thus, in 4/4 time, a whole note gets 4 beats or counts after striking it on its first beat, a half note gets 2 beats after the initial strike, a quarter note receives one beat simultaneously with the strike and 2 eighth notes would be struck in the time it took to play one quarter note. The duration of any note will always be relative to the amount of beats indicated by its representation (whole, half, quarter, eighth, sixteenth, etc.). After all of the allotted notes per measure are played, the next measure is played (again, each measure is separated by a bar from an adjoining measure). This underlying time provides the rhythm of the composition.

The whole note is represented (when written on sheet music) as an oval. All the other notes are also represented (in written form) as ovals but all the other notes have stems. The stem may point either upwards or downwards. The half note looks like the whole note with a stem. The quarter notes are also colored in as are the eighth notes and sixteenth notes. Eighth notes are also represented by adding a flag to a quarter note. The flag indicates that the note receives one half the beat of the quarter. Two flags on a sixteenth note indicates that it receives one half the beat of the eighth note. When two eighth are written in succession on a measure, the flag on each note is not written. Rather a beam (bold line) is written between the two notes, thus connecting them. Similarly, two beams are written connecting two sixteenth notes.

Adding a dot to a note adds an additional one half of its beat to the duration of the note. Thus, a half note in 4/4 time that receives 2 beats per measure. Adding a dot, which is one half of the two beats, or one beat, is added on to the duration. Thus, a dotted half note receives its two beats plus one additional beat, or three beats.

Similarly, there is a whole rest, a half rest, and quarter, eighth and sixteenth rest notation. A Rest indicates that a beat should be counted but no note should be played (silence).

The vertical graph aspect of the lines of the staff indicates the scale of notes. Each line of the staff will represent a note and each space between the lines represents a note. The natural notes are in western, alphabetical representation: A, B, C, D, E, F, G. However, the actual scale is based on 12 notes. These include the half step (semitone) sharps and flats between A and B, C an D, D and E, and F and G. These seven major notes and five interval minor notes create an octave. The interval semitone between all of the notes, except for B and C and E and F, is always the Sharp of the preceding note and the Flat of the successive note. Thus, for instance, A Sharp and B Flat are the same Enharmonic Note. The symbols written before the notes to signify whether a Sharp (#) or a Flat (b) should be played are known as Accidentals. The step up between notes is a tone, such as between C and D.

There are several levels of octaves in which a scale (sequence) of notes can be played. Every note played by any instrument corresponds with a frequency. For instance, A just below Middle C has the frequency of 440 cycles per second (Hz). This frequency also indicates the pitch of the note. Each rise in octave doubles the frequency of a note. Thus, A before C one octave above the previous example has a frequency of 880 Hz. Again, this frequency is what identifies the note’s pitch.

The octaves also are number by group. For instance, the first octave group adds a zero to any note in its group. Thus, the first C is C0. Middle C is C4 at 261.6 Hz, and A4 is the 440 Hz A note mentioned in the above paragraph.

A Scale is a sequence (played either ascending or descending) of notes within an octave. For instance, playing only the major notes from Middle C (C4) on a piano (white keys only) to C one octave higher (C5) is a scale. The scale begins with Middle C and is known as the C Major Scale (C, D, E, F, G, A, B, C). The Middle C is also known as the Tonic (root note) of the scale. Each note in the scale has a name, number position and a relationship to the Tonic:

C Tonic I
D SuperTonic II
E Mediant III
F Subdominant IV
G Dominant V
A Submediant VI
B Leading Note VII

A Major Scale always has the same intervals between each of the notes in the scale. For instance, in the C Major Scale, the two notes following the Tonic C (the Supertonic D and Mediant E) are always a full step intervals. The Subdominant F is a half step from the Mediant E. The Dominant G, the Submediant A and the Leading Note B are all full intervals between each other. The next octave C (C5) is a half interval. Similarly, the G Major Scale is G, A, B, C, D, E, F#, G. The Transposition of any Major Scale is just changing the Tonic (root note) as all the following notes in the scale will follow the same interval pattern. The transposition of a major scale is accomplished by the insertion of a key signature.

Sometimes one or two Sharp or one or two Flat symbols will be written one a specific staff line or space at the beginning of a composition right after the Clef symbol but just before the Time Signature. The presence of these Sharps or Flats indicates the Key Signature of the composition. This means that any note on the staff line or space that has a Key Signature Sharp or Flat at the beginning of the composition will consistently be played a semitone higher or lower (depending on the Key Signature). The absence of any Key Signature means that the composition is in the Key of C, which has no sharps or flats.

The Circle of Fifths demonstrates the relationship between key signatures. There are twelve key signatures in the Circle starting with the key of C at the top. When moving in a clockwise direction, the next note is a perfect fifth from the previous note.

In addition to the major scales in western music, there are also three Minor Scales: Natural (or Relative) Minor, Harmonic Minor and Melodic Minor.

List of Scales:

Chords are a group of three or more notes played simultaneously. A chord will always be structured around a key note (tonic), which is normally the lowest note in the chord. The most basic chord are known as Triads. Triads are three note chord combinations. For instance, the major chords, C Major, G Major, etc., are triads (.wav and .aiff digital audio file samples are presented on the Sound Page for use in your projects).

Once a WAV or AIFF sample is available in either RAM or from the hard drive, the digitized file can be opened and manipulated. Applying the effects to a track or series of tracks in real time, however, may result in a “destructive” edit where the original recording and spectral profile is changed permanently.

The sound can be processed many ways.

• Chorus
• Delay
• Envelope
• Filter
• Flanging
• Harmonizer
• Panning
• Phase (Phase Shift)
• Reverb (Reverberation)
• Reverse
• Time stretch
• Transposition
• Tremolo
• Vibrato

There is usually a graphical interface on the computer screen when one opens a file in an effects processing application that shows the reconstructed waveform based on the sampled values of the original analog waveform.

Chorus is obtained by layering a track or waveform with two or more identical tracks or waveforms that have had a slight time delay and modulation of the tracks (pitch shift), thus giving the effect of several instruments playing at once. The application of the effect is measured by the effect’s speed (rate) and alternation (depth). The effect is created with either a circuit oscillator or software oscillator.

Cross fading is the slow reduction of the gain of one track while a second track gain is faded in.

De-esser, as in the pronunciation of the letter “S”, is a limiter or compressor that will reduce the frequencies related to this letter that are produced when either spoken or sung.

Delay is the attempt to create an echo by adding a reproduction of the original signal at a constant rate of time after the original signal. The actual effect can be set from milliseconds to several seconds.

Distortion adds additional frequencies by clipping the waveform.

Echo is obtained by mixing a time-delayed signal output with the original version of the signal. The parameter adjustment also allows for the control of the signal repetition. It can be further adjusted by the volume decrease (or increase) of the repetition. The effect differs from Delay or Reverb by representing more distinct and wider reflections of the sound wave.

Envelope is the span of time through which the audio signal goes through various levels of Attack, Sustain and Decay.

Flanging is obtained by combining very slight, varying signals that have been time-delayed (by milliseconds) with the original audio signal to give a resonating effect. The application of the effect is measured by the effect’s speed (rate) and alternation (depth). The effect is created with either a circuit oscillator or software oscillator.

Gain is the increase in the (volume) level (measured in dB / decibels) of the audio signal.

Harmonization is the effect of detuning the delay and with the result being a two-pitch harmony.

Pan or Panning (Panorama) is establishing a balance of left and right output of the sample or establishing a higher degree of the signal or sound in one side of the stereo mix at the expense of the other side.

Phase (Phasing) is the combining an audio signal with a copy of that same signal but has had the relative phase reversed.

Pitch Shifting / Stretching is the changing (transposing) the root note of the waveform by semitone step to a new root note without changing the playback duration rate.

Reverb combines several Echo delay affects of the original signal in order to create the ambience and depth of how the actual sound would be diffused/reflected from surfaces within a certain type and size of confined structure or environment. The increase of the volume and intensity of the reverb simulates a particular environment.

Reverse is the reversal of the audio signal.

Ring Modulation a form of amplitude modulation, is one signal literally sent through another signal, however only the sum of the frequencies, and the difference between the two frequencies, both known as the Sideband frequencies, combined is now the new signal frequency output.

Time stretching is adjusting the length of time of which a waveform plays to fit a specified time frame. The idea is to change the length without changing the tone, vibrato, tremolo and other characteristics of the sound.

Tremolo is a rapid modulation of the volume level of the signal which is obtained by a modulating wave signal being applied to the amplitude of the carrier audio signal. The rate of the effect can also be adjusted.

Vibrato is a rapid modulation of the pitch of a note.

5.1 and 7.1 Editing / Panning

Mixing and editing is moving in the direction of mixing a finished product for Dolby Digital Surround Sound. This involves having hardware and software capable of separating out the 6 discrete channels and panning the signal to the correct speaker. First, 6 separate digital audio tracks must be created.

Dynamic Range is the range between the the highest amplitude and the lowest amplitude of a digital audio data signal.

Filtering

Most editing and effects are conducted by the application of a Filter. Filters remove, increase or shape frequencies from the digital audio signal.

The High Pass Filter (Treble Shelf Filter) cuts out frequencies below a preset level (cut-off frequency) or boosts frequencies above a preset level.

The Low Pass Filter (Bass Shelf Filter) cuts out frequencies above a preset level (cut-off frequency) or boosts frequencies below a preset level.

A Band Pass Filter (Mid Range Filter, Notch Filter) allows only a signal of a specific range (band) of frequencies to pass, and these frequencies can also either be cut or boosted. A sweepable band pass filter will allow one to reset the center frequency, which in turn will adjust the rest of the frequencies in the specified band that range both equally higher and lower from the center frequency. A Parametric Filter allows one to increase or decrease the specified range of frequencies that are encompassed in the band.

Anti-aliasing filtering is the application of a combination of pass band and stop band filters that do not allow incoming signal frequencies (say during analog to digital conversion) in access of 20.5 to 21.5 KHz so that Quantization Error will not occur.

Compression / Expansion

Compression is used to set the upper and lower frequencies of the dynamic range of the signal. Thus, compression will restrict the overall dynamic range of the signal so that it does not exceed a predetermined threshold level. High amplitudes are decreased to a maximum level while low amplitudes are increased to a minimum level. It is obtained by setting a maximum Threshold Level (measured in dB/decibels) and setting the compression ratio. Any signal that exceeds the Threshold is compressed (gain is reduced) in terms of the ratio. The ratio of compression to signal input reduces the dB level of the output signal by the ratio factor. Thus, in a 2 to 1 ratio, a signal that exceed the Threshold by 2 dB is compressed to only 1 dB gain increase. The higher the ratio, the more the gain reduction is applied. Conversely, on the low amplitude level, a 1 dB signal can be increased to 2 dB, thus increasing the volume of the low amplitude. The Attack (how quickly the Compressor will act on the signal) and Release (how quickly the Compression brings the gain back to the Threshold level) of the application of Compression must also be specified. These applications are measured in ms (milliseconds). The compression needs to be applied accurately and equally to both signals if it is an incoming stereo signal. In multi-band compression, different levels of compression can be applied to different frequencies of the signal.

Digital compression hardware have an ADC (Audio to Digital Converter) circuit that samples the incoming analog signal (or even an already existing digital signal). The variable voltage analog signal is sampled, the sample is a mathematical model of the amplitude of the analog signal represented by a bits / bytes sequence, and the compressor DSP application conducts a furhter manipulation of the amplitude model and the processed digital signal has the new amplitude model based on the parameters (Threshold Level, ratio, attack and decay) set by the user. A software compression application installed on a computer conducts compression in the same manner by emulating the function of an analog compressor and using the host processing capabilities of the CPU of the computer.

Expansion (Expander) is the opposite of compression, such that it increases the dynamic range (increases amplitude of the signal, amplifies, makes louder) of a signal if it drops below a threshold level. The attack and release of this affect can also be adjusted.

Limiter

A Limiter is used to establish a predetermined Threshold Level that the dynamic range of an incoming signal will not exceed without affecting the low end of the range.

Gating

Gating reduces or eliminates a signal if it decreases below a predetermined threshold level.

Noise Gate only allows certain frequencies through, which will reduce background noise or hiss and can actually be used to increase the dynamic range of a signal. The utilization of the noise gate requires specifying a threshold level (signal frequency) that will allow the signal to pass and then stop the signal once it falls below a certain frequency, thus keeping out any and all other signals.

Equalization

EQ or Equalization isolates frequencies, provide gain affects for how much the frequency is increased or reduced, and which range of frequencies are affected in order to balance the relationship between the highs and lows for a sound (frequency response). The “Q” in EQ is the measurement of the width of frequencies above and below an individual frequency. This width above and below the central frequency is measured in octaves. The actual application to the signal is done by shelf, peak and notch filters. By adjusting certain frequencies the actual perceptible analog sound quality is changed.

Equalization requires the application of a filter to modify the low, mid-range and high frequencies of an audio signal:

• High Pass Filter filters out specified low frequencies and only allows frequencies “higher” than the specified level (cut-off frequency) to pass into the mix.
• Low Pass Filter filters out specified high frequencies and only allows frequencies “lower” than the specified level (cut-off frequency) to pass into the mix.
• Band Pass Filter filters out frequencies above and below and specific range. A Band Reject Filter will perform the opposite, it will eliminate all frequencies within a specified range.

Graphic Equalization allows the amplitude (loudness) adjustment individually of several fixed frequency bands in a given audio signal. Each bandwidth can be adjusted independently of the other. The graph of the EQ Curve, which represents the intersection of amplitude of a signal (plotted on the y/vertical axis) and the frequency of the signal (the corresponding x/horizontal axis).

Parametric Equalization – allows macro control of bandwidth, frequency and amplitude (gain) for specific, overlapping bands of frequencies.

Normalization

Normalization insures that the amplitude of the cycles of the entire signal (or digital audio file) remain within the dynamic range.

Side Chain

A Side Chain is the ability of the dynamic range or effects processing (see below) hardware or software, or a mixing board, to apply a second (or more) effect(s) to a signal just prior to it is being routed / processed or immediately after.

DSP (Digital Signal Processing) is conducted by specific microprocessors wired to circuit boards within the computers. The DSP chip itself is either fixed function or can be multi-function, programmable. This requires algorithmic software written specifically for the DSP architecture. The design of the chip has migrated to 32-bit, fixed or floating point processing. The application of DSP processor/software is used in sample rate conversion, effects processing, filtering, audio mixing, and dynamic range processing.

• They each have an operating program to manage all of the functions of the system.
• They each incorporate sophisticated, audio-related software applications that perform specific DSP processing functions such as sound synthesis or effects editing.
• They complete all of these functions with the processing power of a microprocessor, anciliary hardware and usually some type of graphical representation or a screen.
• A sophisticated soundcard (audio interface) allows various analog and digital In / Out signal connections, ADC and DAC and DSP capabilities.
• There may be an intermediary modular device, such as a mixing board that incorporates the necessary I/O connections and provides additional processing, effects editing and then the signal is eventually routed to the microprocessor.
• The workstation relys upon the storage capabilities of the computer’s hard drive to hold the data that may be generated from recording live instruments or editing an existing file of a previous recording.
• The processing capacity and configuration all combine to function as a console or Desktop Multi-track/channel Mixer, Digital Audio and MIDI Sequencing, effects editing, Hard Disk Recording (sampling analog signals or re-sampling existing digital audio data) and sometimes direct to CD burning.

In one context, Mastering (post production) is the creation of a a final copy of a composition (or group of compositions) prior to the transfer of the composition(s) to a media for release to the public. Mastering is also sometimes used in the context of creating a Glass Master that will be used to make copies onto the familiar, commercially sold plastic CD (CD Mastering).

Pre-Mastering (or post production) is your best-effort, final, mixed, edited copy of an individual or several compositions, in either DAT or CD format, that will be used as a source copy for replication purposes (which normally means producing either a CD or cassette copy). One may also submit the pre-mastered copy version to a Mastering Studio that may also be capable of replication (manufacturing thousands of copies of the composition in your desired format). The Mastering Studio may also re-master (re-edit) your pre-mastered copy. The Pre-/re-mastered copy is then etched into a glass Master copy by a laser cutter. The glass Master Copy is used to create a metal copy that is used as a stamper to replicate copies in the familiar plastic version. The stamped replication process is done onto the aluminum CD media which produces good results in most consumer audio devices. As music composition continues to migrate to an all digital distribution, Mastering will come to be defined more by its traditional idea of final editing of the composition.

In Mastering (post production final editing) it is expected that the final mix is reviewed and adjusted one more time by a professional engineer on superior editing equipment. It does not matter if it is one composition or several, the idea is to give each composition a consistent sound and presentation when it will be played back on consumer audio equipment. The level of equipment that one already owns, your level of experience, who the intended audience is and what medium (Audio CD, streaming media, etc.) will determine whether you should seek professional pre-mastering or not. With the quality of today’s home recording systems you can produce a fairly good demo for at least review by recording industry representatives. A Mastering company will, hopefully, have superior equipment such that if you send them an analog mix to tape then they can do the analog to digital conversion at 20-bit to 24-bit resolution and 48KHz to 96KHz sampling, compared to what one can do in a home studio. Again, it depends on your intended audience.

The Mastering is the final editing and processing (normalization, EQ, limiting, compression, stereo balance, fading) which are completed in order to make the composition consistent and eliminate any problems. If there is more than one composition, then they are placed in order, spacing between them is timed, and the level of volume across all compositions is established. It is still possible to introduce further editing (reverb, EQ, etc.). The source copy final version should be as perfect as possible as it is the version that you will be submitting to a professional manufacturer so that it could be replicated to create distribution copies for either further review or for sale.

Digital Data Transmission

Digital data signal or bit stream can be transmitted through / over:

• Unshielded Twisted Pair copper wire
• Shielded Twisted Pair copper wire
• Coaxial Cable
• Radio (Microwave and Infrared)
• Satellite
• Fiber Optic Cable (glass and polymer)

Mixing (either with software applications or hardware device) is combining many tracks or inputs into one or two (maximum) outputs (no matter how many tracks you have recorded, when they are played back they will automatically be combined into 2 digital channels stereo). One can cue or solo the sound, which can then be combined with other tracks. Various effects can be applied with the mixer depending on its capabilities. The more recent devices also allow for the processing of audio at levels close to 96 KHz sampling rate and 24-bit resolution, which exceed Redbook CD standards.

As you add recorded tracks, the volume of each individual track may have to be reduced so that the total volume of all tracks combined does not exceed the 16-bit rate (if that is the standard of your software). If it does, you will get a condition known as clipping, which is when the dynamic range of the waveform of the sound exceeds the processing capabilities of the mixer application.

• SMPTE
• MIDI / MTC (MIDI Time Code; MIDI Song Position Pointer)
• Black Burst (this is an frame rate synchronization signal for video production, however it does not include a video signal, only audio is transmitted and can be used a master time code)
• PAL (Phase Alternate Line used in Europe)
• NTSC
• Word Clock

Digital Audio

Digital Audio files can be worked with in:

• Sound Wave Editors
• Multi-track Audio Production
]]> http://www.globalmusicresource.com/basics/synchronization.html/feed 0 http://www.globalmusicresource.com/basics/sound-card.html http://www.globalmusicresource.com/basics/sound-card.html#comments Tue, 27 Oct 2009 17:03:39 +0000 admin http://www.globalmusicresource.com/?p=62 Computers process digital audio and play audible sound through the Soundcard (Audio Interface).

The polyphonic ability of a sound card (a circuit board installed within the computer) refers to the ability of the sound card to play more than one note (also referred to as voice) at a time. Sound cards, and synthesizers, these days are capable of 32, 64 and 128 voice polyphony. A Multi-Timbral soundcard will also refer to is ability to reproduce more than one instrument sound simultaneously. The most recent soundcard professional upgrades will allow for internal processing a high quality 96KHz sampling rate and 24-bit (approximate) resolution which will create very good sounding compositions. Most off-the shlef sound cards have only two channels (stereo) although some professional upgrades will have as many as four to sixteen. An improved soundcard will also allow one to increase the options for data transfer with various connectors such as AES / EBU, S / PDIF, TOSLINK, XLR, RCA, pre-amplifier and Balanced / Unbalanced connections. Each one of these hardware connection standards also have a specific Protocol of data transfer interface.

]]> http://www.globalmusicresource.com/basics/sound-card.html/feed 0 http://www.globalmusicresource.com/basics/sound-module.html http://www.globalmusicresource.com/basics/sound-module.html#comments Tue, 27 Oct 2009 17:02:46 +0000 admin http://www.globalmusicresource.com/?p=60 A Sound Module functions like a storage unit for pre-recorded instrument or special effects sounds. It is usually a stand-alone, modular unit that must be connected to computer or MIDI keyboard to send it instructions. There are also software application Sound Modules available. Again, similar to the Sampler, the Sound Module tends to come without sound synthesis generation capabilties of oscillation for wave creation or any of the synthesis modelling types. Rather, the sample is used as the source of sound and any synthesis-like filtering, gating or modulation is utilized for sophisticated effects processing and file editing.

A Sound Module is a device or instrument that can reproduce either musical or special effects type sound. The stand-alone, harware ROM or sound generating synthesizer comes without the keyboard controller. However, it must be triggered by some type of MIDI controlling device. The module itself merely contains patches or programs of digitized sound in various audio formats depending on the manufacturer and usage. It is sometimes also referred a rack-mount module or stand-alone or external module. The devices can offer General MIDI, unique (non-GM), proprietary, instrumental, percussion or sound effects patches. The designated sound is then triggered by MIDI controller (either keyboard, harware sequencer or from a multi-track sequencer software application installed on a computer and connected to the Sound Module by cable). Many of the recent Sound Module models are capable of up to 128-note polyphony. Thus, similar to a Sampler, one may select “Piano” (for instance) on the Sound Module and the several octaves of piano samples are loaded into the RAM of the Sound Module. One then may trigger all of the notes with a MIDI keyboard (as long as the sampler is MIDI compatable). Sound Modules are also now being designed as software application plug-ins (they ship with CD sample files and are loaded into the RAM of the computer along with the instructions for their management) and interface with a multi-track audio production application. The digital data stream output of the Sound Module can also be routed into a Mixer or directly to a computer with a multi-track audio application and be recorded as a track in a project.

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