Basic Audio Recording Notes

A brief history of audio recording

• http://www.aes.org/aeshc/docs/recording.technology.history/notes.html• https://en.wikipedia.org/wiki/History_of_sound_recording

Wikipedia definition of audio engineer (https://en.wikipedia.org/wiki/Audio_engineer)

Chapter 1: Introduction

• Use the web to compliment your learning (http://www.wikipedia.org, http://www.youtube.comand http://www.modrec.com)

• The recording studio has the following elements:◦ A professional staff◦ Professional equipment◦ Professional, yet comfortable working environment◦ Optimized acoustic and recording environment◦ Optimized control room environment

• Shape, size and acoustic design of recording studios:◦ Studios that record a wide variety of music (ranging from classical to rock) might have a

large live room with smaller isolation rooms off to the side, for unusually loud or softinstruments, vocals, etc.

◦ Studios designed for film scoring might be larger and will have high ceilings toaccommodate the large sound buildups generated by a big group of studio musicians

◦ Studios used to produe audio for video, film, vocals and mixdown, might be small recordingspaces

• The control room:◦ Is ideally separated from the live recording room (though that is not always necessary)◦ Acts as a critical listening environment, with carefully placed and balanced monitor

speakers◦ Houses the majority of the studio's recording, control and effects equipment, including the

recording console (desk, board), which is:▪ The artist's palette for a recording engineer▪ Combines and controls the audio input, output and distribution signals▪ Is used for mixing audio, spatial positioning, routing and switching▪ Do you need a big mixing board? You don't need one but there are some advantages to

having one• Recording machines

◦ Reel to reel (tape, analog machines)◦ Digital Audio Workstations (DAWs), usually a computer running audio software that is

hooked up to a mixing board• The changing face of the music business:

◦ Personal production and home project studio have reduced the need for professional studios◦ The need for post-production audio for film, broadcast and web projects has created new

work for recording studios◦ Studios have branched out into offering web design, music distribution, video, music

business advisory services, networking, label and tour support, and other services• Project studio – audio production facility built for personal use • Portable studio – laptop, USB-microphone, instruments, software• On-location recording – use field equipment

Audio for Video and Film

• MIDI (Musical Instrument Digital Interface)• Hard-disk recording• Timecode• Synchronization• Automated Mixdown• Advanced effects • Demands a high level of specialization

Audio for Games

• More games using orchestral recordings• Has its own set of technical and scheduling requirements• A single game can work with tens of thousands of voice and music cues• Specialized schools, such as Digipen in Seattle

Multimedia and the Web

• Integration of text, graphics, MIDI, digital audio, digital video, etc.• A website could offer music, audio books, interactive books, interactive videos, etc.

Social Media, Streaming and VR

• Social media is a huge part of how music reaches the masses these days• Streaming is the new form of consuming music• If media, like music and video, are free, how do artists make money? Could it be with virtual

reality?

Studio Personnel

• The artist – creates the music – the quality of the piece begins and ends with the artist• Studio musicians and arrangers – add extra spice and depth to performance – could be

“sidemen”, a choir, an orchestra or even just a better drummer (for example, Andy White sittingin for Ringo on “Love Me Do”)

• The producer – help the artist/record label create the best possible salable product (technically and aesthetically) that fits within the artist's vision. Assists in song selection. Focuses the assets and helps with goal setting. Increasingly, people in the studio wear many hats, such as engineer/producer, or artist/engineer/producer (Prince)

• The engineer must:◦ Conceptualize the best approach to record the music (technical and aesthetic)

◦ Document the recording process via real or virtual notes◦ Choose microphones and placement, run all the gear and the Digital Audio Workstation

(DAW)◦ Capture the performance◦ Record overdubs◦ Mix the final product◦ Archive the recording

• Assistance engineer – engineer in training• Maintenance engineer – takes care of equipment• Mastering engineer – makes final tweaks to recording before it hits the public• VJ – video jockey – can mix video and audio in a live environment and make videos for the

web• DJ – disc jockey – three types:

◦ on the air – powerful voices on the radio◦ on the floor – gets people to interact with music◦ on the web – powerful voice over social media ◦ Who is the most powerful of the above three DJs today? Why?

• Studio manager – runs the day-to-day operations• Music lawyer – deals with contracts, band agreements, copyrights and other legal matters• On page 26 of the book there is an extended list of other careers in the music industry

Networking for Work

• Showing up is huge (Woody Allen - “75% of success is just showing up”)• Make a WordPress webpage• Send out resumes• Move to where the jobs are• Look for internships and mentors• Hang out around the companies you want to work for • Remember to follow up

The Recording Process

1. Preparation (the battle is won or lost before it begins)2. Recording – recording a single instrument to its own track allows for variation of the following

without affecting the other instruments:1. Level2. Panning3. EQ4. Signal processing5. Routing

3. Overdubbing – “punching in” to fix errors, create better tracks. “Comping” is doing a bunch of tracks and picking the best one to use

4. Mixdown – altering the overall session's:1. Relative level2. Spatial positioning (panning)3. EQ4. Dynamics

5. Effects processing5. Mastering:

1. Relative level balancing between songs2. Dynamic level3. EQ4. Overall level5. Song sequence editing

6. Product Manufacturing (physical or virtual):1. Creating a manufacturing master2. Art layout and printing3. Product packaging

7. Marketing and Sales (social media):1. Who is my audience?2. Will I sell the product myself?3. What should the final product look and sound like4. What's my budget?5. How will it be marketed?

Transducer – changes one form of energy into another corresponding form of energy

• Ear – sound waves in ear – nerve impulses in brain• Microphone - Sound waves in air – electrical signals in wires

Chapter 2

The Basics of Sound

• Sound arrives at the ear as sound pressure waves• If you pop a balloon, the molecules around it absorb the compressed molecules inside the

balloon, increasing the atmospheric pressure around it. This is called compression. If you were to let air out of the balloon and it got smaller, the process of creating less atmospheric pressure is called rarefaction. As a speaker cone moves in out and out, it creates sound waves of higher and lower pressure in a process called wave propagation

• A waveform is the graphic representation of a sound-pressure or voltage level as it moves through a medium over time. Waveforms have the following characteristics:

◦ Amplitude – the distance above or below the centerline of a waveform (top or bottom of waveform is peak amplitude value or peak level)

◦ Frequency – the rate at which an acoustic generator, electrical signal or vibrating mass repeats within a cycle of positive and negative amplitude. The more repetitions, the higher the pitch and vice-versa. The number of cycles per second is measured in hertz (Hz)

◦ Velocity – how fast the sound wave travels through the air, 1130 feet/second (faster at higher temperatures)

◦ Wavelength – the physical distance in a medium between the beginning and the end of a cycle

◦ Sound reflection – see diagrams on page 49◦ Sound diffraction – bending a sound around objects◦ Phase – waves can be in or out of phase (see page 54) do online tutorial◦ Harmonic content – frequencies above or below the main frequency (overtones, partials). If

the partials are whole-number multiples of the original frequency, that are perceived as pleasant (octaves). If they are odd multiples, they are called odd harmonics and are considered unpleasant, dissonant, harsher tones

◦ Timbre – the harmonics and their relative intensities◦ Envelope:

▪ Attack▪ Decay▪ Sustain▪ Release

◦ Loudness levels – human ear hears over a range of 1013: 1, so logarithms are used to simplify the numbers

◦ Decibel (1/10th of a Bell – a unit of measurement used with older phones, named after Alexander Graham Bell)

◦ Sound-pressure Level (SPL) – the acoustic pressure that's built up within (usually) a square centimeter. The higher the SPL, the louder the sound (see this SPL chart). As the distance from a sound is doubled, the sound will decrease by 6 dB, if the distance is halved, it will increase by 6 dB

◦ Voltage – the pressure behind electrons within a wire◦ Power – a measure of wattage or current; the flow of electrons through a wire over time.◦ Takeaways

▪ a 1 dB change is noticeable to most ears (but not as much)▪ Turning something up by 3 dB will double its level▪ Turning something down by 3 dB will halve its level▪ Turning something up 10 dB will be a ten-fold increase, turning it up 20 dB will be a

100-fold increase, 30 dB increase will be 1000 times louder (because of logarithms)▪ Show diagram of ea r and talk about how it works▪ Take care of your hearing

• Acoustic trauma – ear exposed to sudden, loud noise, could lead to permanent hearing loss (Pete Townshend – Kieth Moon blowing up drums)

• Temporary threshold shift – temporary loss of hearing – like after a loud rock concert

• Permanent threshold shift – extended exposure to loud noises can lead to permanent loss of hearing in that range

• Wear earplugs when around loud sounds• Take 15-minute quiet breaks when around a lot of sound-press• Have hearing checked periodically by an audiologist

Basic Audio Recording NotesChapter Three

Studio Types

– If you spend hundreds or thousands of dollars on your equipment, you probably have a ProjectStudio (under this definition, I personally have a Project Studio, though if you spend enough time spending money on it ((like 20-30 years) you can easily end up with a Professional Studio)

– If you spend tens of thousands, or even hundreds of thousands, on your equipment, you probably have a Professional Studio

– A third type of studio is the Audio-for-Visual Studio, which is for scoring films and other visual projects.

– Professional studios are often defined by how much the budget is for creating them and by building constraints. Pro equipment and facilities are very pricey. Fewer of these studios are around these days as home-recording equipment has become so affordable.

– Anything from a lady with a laptop and microphone up to someone who spends tens of thousands of dollars on equipment would probably be considered a Project Studio. Project Studios are constrained by construction budgets and often have to make do with what's available.

– Professional Studios are like other professional businesses in that they rent or buy pricey buildings and equipment and have professional personnel on staff. A local professional studio owner I know has microphones alone that are worth $100,000.00.

Form Factors Governing Studio Acoustics

– Acoustic isolation – keep external noises (airplanes, cars, trains, neighbors) to a minimum– Symmetry in control room/monitoring design – allows you to hear the sounds in the optimal

environment (see Auralex analysis of my control room)– Frequency balance – maintain the original frequency balance via symmetry in room design,

otherwise frequencies will not be heard accurately (will be imbalanced)– Reflection – too much creates Standing waves (aka “room modes” - reflections off non-

absorbent surfaces) can continue to bounce around the room and artificially highlight certain frequencies while de-emphasizing others. Sound can reflect around the room the way visuals reflect in a room filled with funhouse mirrors. Standing waves cause phase problems (phase cancellation). They are generated by rooms with parallel walls (which is most rooms) and also the ceiling/floor

– Diffusion – helps break up standing waves (see Auralex T-fusor) and can help smooth out the reverberation of the room (bookshelves are natural diffusors)

– Isolation rooms (aka acoustic partitions, gobos, baffles – can be made from palettes)– Absorption – opposite of reflection – sucks some of the acoustic energy out of the room. Some

engineers agree that a ration of 25% absorption with 25% diffusion, 50% reflections is a good balance

– Reverberation – the persistence of a signal (the time required for a sound to die)– Acoustic echo chamber – an isolated room that has highly-reflective surfaces where

microphones are placed. You can create your own echo chambers in garages, showers and by feeding speaker outputs into a large room. You could play a track into a church and record the reverberation to add to the main tracks later.

Basic Audio ProductionChapter 4: Microphones

Definition• A microphone is a transducer that changes energy from one form (sound waves) to another

(electricity)• The quality of the recorded sound will depend on external factors (placement, distance,

instrument and acoustic environment) and on design variables (microphone operating type, design characteristics and quality)

• Three rules to apply when miking:1. There are no rules, only guidelines2. The overall sound of an audio signal is no tter than the weakest link in the signal path3. Use the “good” rule. Good musician + good instrument + good performance + good

mike + good placement = good sound

Mike Design

• Three types of mikes are used:1. Dynamic2. Ribbon3. Condenser

Dynamic MicThis mic uses electromagnetic induction – when an electrically conducive metal cuts across the flux lines of a magnetic field, a current of a specific magnitude and direction will be generated within that metal.

These mikes contain a stiff Mylar diaphragm. Attached to the diaphragm is a finely-wrapped of metal called the voice coil that is precisely suspended within a high-level magnetic field. When an acoustic pressure wave hits the diaphragm, the attached voice coil is displaced in proportion to the amplitude and frequency of the wave, causing the coil to cut across the lines of magnetic flux and creating an analogous electrical signal through the output leads.

Traditional Ribbon MicThese mikes use an extremely thin piece of aluminum (2 microns) that is corrugated. Sound pressure waves cause it to move across the magnetic flux lines, creating a current that's equal in amplitude and frequency to the acoustic waveform. These mics generate low electricity, so they must be attached to a step-up transformer to get the audio to be at a higher volume.

Newer ribbon mikes use updated technology (please see the book for more information)

Condenser Mic These mikes work using electrostatic principles, as opposed to the electromagnetic principles used by the previous two mics. There are two metal plates: one is thin and movable and the other is static and attached to the back of the microphone. DC power is put through the plates which create a “capacitance” in-between the plates (this is called a “capacitor” it is also known as a “condenser.” As the movable diaphragm is hit by acoustic sound waves, it moves back and forth, changing the amount of space between the diaphragm and the back plate, thereby changing the electrical signal that is output in a way that matches the amplitude and frequency of the incoming signal.

Condenser and ribbon mikes are often powered by the mixing board in a process known as “phantom power.” This supplies +48 volts of electricity to the incoming signal to bring it up to a volume level where it can be used.

Microphone CharacteristicsThe directional response of a mic refers to its sensitivity (output level) at various angles of incidence with respect to the front (on-axis) of the microphone. The angular response can be graphically charted in a way that shows a microphone's sensitivity with respect to direction and frequency over 360 degrees. This is called the mic's polar pattern. Mic directionality can be classified into two categories:

• omnidirectional• directional

Omnidirectional mics pick up information from all sides (omni is Latin for “all). A directional mike that is equally sensitive on two side of a diaphragm has a figure 8 or bi-directional polar pattern.

Frequency ResponseThe on-axis frequency-response curve of a microphone is the measurement of its output over the audible frequency range when driven by a constant, on-axis input signal.

RumbleAt low frequencies, rumble may be picked up due to people walking in a hallway, passing cars or trains, etc. This may be handled in several ways:

– Use a shock mount to isolate the mic from the vibrating surface and floor stand– Choose a mic that has a restricted low-frequency response– Restrict the response of a wide-range mic by using a low-frequency roll-off filter

Proximity EffectThis effect is an increase in bass response when a mic is brought within 1 foot of a sound source.

SibilantsThe popping sounds of “p” and “b” as well as the hiss of “s” can be controlled better with pop screens.

Transient ResponseTransient response is the measure of how quickly a mic's diaphragm will react when it is hit by an acoustic waveform.

Sensitivity RatingThe sensitivity rating is the output in volts that a mic produces. A mic with a high sensitivity rating willproduce a higher output than a low sensitivity rated mic.

Equivalent Noise RatingThe equivalent noise rating of a microphone is the devices electrical self-noise.

Overload CharacteristicsAt high Sound Pressure Levels, a mic can distort. This distortion can be mitigated by use of an attenuation pad, which reduces the signal level at the pre-amps input, thereby eliminating distoration.

Microphone ImpedanceThe output impedance is used to match devices together (a devices's output impedance should match the input impedance of a connected device).

Balanced/Unbalanced LinesBalanced lines balance the electrical signal by canceling out excess noise, and pick up less interference, resulting in a cleaner signal and can be longer cables (XLR cables for microphones 3 lines, two carry the signal and one is a ground.). Unbalanced lines have to be shorter and tend to pick up electrical interference (TRS phone cables for instruments).

Microphone PreampsMic preamps are used to boost the signal of a microphone to line level for recording and can also color the sound in a positive way. Older-style tube amp preamps are in demand and sound great, plus they are pricey to buy.

Microphone AcquisitionThere are two methods:

– Buy a lot of general purpose mics– Buy specific mics for every purpose

Microphone Techniques

– Distant miking– Close miking– Accent miking– Ambient miking

Distant MikingMiking from three feet or more away from an instrument. This will pick up a large portion of the soundcoming from a musical instrument or ensemble, preserving the overall tonal balance of the source. Often, a natural tone balance can be achieved by placing the mic at a distance that's roughly equal to the size of the instrument or sound source.

Distant miking also allows for the room environment to be picked up by the mic.

This is used for jazz ensembles and classical symphonies but can also bring a small band to life (by, forexample, distant miking a set of drums on a separate track).

An engineer might need to add foam or or absorption materials to break up the reflections of a room if the sound is muddy.

A boundary microphone (which is placed on the floor) can be used to eliminate a lot of phase cancellation.

Close MikingThis is when a mike is placed 1 to 3 inches from a sound source. It creates a tight, present sound quality. It effectively excludes the acoustic environment from being picked up. When a close mic picks up the sound of another instrument, inadvertently, it's called leakage.

One way to control leakage is to plug loud instruments (bass, guitar, keyboards) into D.I. or Direct Injection boxes. These send the sound straight to the board, bypassing loud amplifiers.

Leakage can be controlled by using gobos (flats, dividers, baffles) between instruments or an instrument can be surrounded by gobos to isolate it. Instruments can also be spread further apart. Instruments can be placed in isolation booths.

Sometimes you want leakage, to produce a “live” sound. In that case, the mics must be carefully placedin order to avoid phase cancellation.

The three-to-one rule can be used to avoid leakage. It states that for every unit of distance a mic is froma sound source, the next closest mic should be three times that distance away. Some err on the side of caution and use a 5:1 rule.

Leakage isn't always bad. Sometimes it's the “glue” that ties a mix together, by adding in the sound of the environment.

When a mic is close to a sound source, it usually doesn't pick up all of the tonal balance (timbre) of an instrument. Moving the mic a few inches along the length of the instrument of a few inches back can make a big difference. Also, switching out mikes and adding EQ (equalization) the mic can help.

As much as possible, musicians should be able to see each other when performing. Interacting visually provides an important set of cues and energy to musicians.

Accent MikingA microphone placed close enough to an instrument to pick up its sound and the sound of surrounding instruments is called an accent mic. For example, a mic used to pick up a solo violin in an orchestra.

Ambient MikingThis type of miking is used when trying to get the sound of the environment as well as the sound of the instruments.

Usually a cardioid stereo pair of mikes is used or two crossed figure-8 mics. Ambient miking works great for live concerts. Mics can pick up the sound of the audience. They also give a recording a sense of “space.”

Stereo Miking Techniques

Spaced Pair – placed in front of an instrument or ensemble (not pictured below but pictured earlier)

X/Y – placing two coincident-pair mikes (matched pair) with their grills facing each other, as close as possible without touching, usually at an angle of between 90 and 135 degrees, with each mic panned

hard right or left, yields excellent results. The Blumlein Technique is similar but the mikes are on top of each other at 90 degrees and the mikes are bi-directional.

M/S – mid-side technique using a coincident pair. This usually two closely-spaced matched pickups and requires a software plugin or external transformer to work. The M mic is generally a cardioid mic and faces the front of the sound source. The S mic is generally a bi-directional mic and it points to the sides, to pick up the ambient sounds of the room and instrument. This can be combined in the mixdownprocess to add more or less room ambiance to the sound.

Decca Tree This is a classical miking technique that is time-tested that uses time and amplitude cues to create a coherent stereo image. It consists of three omnidirectional mics. A left and right mic are placed three feet apart and a third mic is placed 1.5 feet out in front and centered in the stereo field.

Note: we skipped Chapter 5 as it was not particularly relevant (it was about recording with tape)

Chapter 6 Basic Audio

Digital Basics

In the english language, we use a 26-letter alphabet to form words and communicate and we use a 10-digit number system (base 10). Computers use a 2-digit (base 2 = binary) number system to encode information, using 0 and 1 to indicate different states of information (for example, if a button is in an “off” or “on” state, if there is no voltage present or voltage present, etc.).

For example, a computer might turn the letters C – A – T into the following binary letters (0100 0011) (0100 0001) (0101 0111).

Sampling – a process that affects the overall bandwidth (frequency range) that can be encoded within asound file

Quantization – the resolution of the file (overall quality and distortion characteristics of an encoded signal compared to the original analog signal at its input)

Sampling Details

In the analog audio world (tape) signals are recorded stored and reproduced as changes in voltage levels that continuously vary over time.

The digital process doesn't work quite the same way. Instead it operates by taking periodic samples of an audio waveform over time, and then calculating each of these snapshots samples into grouped binary“words “ that digitally represent these voltage levels as they change over time, as accurately as possible.

During this process, an incoming analog signal is sample at discrete* and precisely-timed intervals (as determined by the sample rate). At each interval, this analog signal is momentarily “held” (frozen in time), while the converter goes about the process of determining what the voltage level actually is, witha degree of accuracy that's defined by the converter's circuitry and chosen bit rate. The converter then generates a binary-encoded “word” that's numerically equivalent to the analog voltage level at that point in time. Once this is done, the converter can store the representative word in to a memory medium (tape, disc, etc.), release its hold, and then go about the task of determining the values of the net sampled voltage.

Within a digital system, the sampling rate is defined as the number of measurements (samples) that are periodically taken over the course of a second. For example, the stand CD-quality sampling rate of 44.1kHz (kiloHertz) takes 44,100 samples per second. This is equivalent to resolution in digital images. Themore pixels you have in an image, the more accurate the representation of the image and the better it looks. A higher sampling rate has a greater bandwidth and can capture more frequencies, revealing a more accurate recording of the analog sound.

*A discrete signal or discrete-time signal is a time series consisting of a sequence of quantities. In other words, it is a timeseries that is a function over a domain of integers. Unlike a continuous-time signal, a discrete-time signal is not a function ofa continuous argument; however, it may have been obtained by sampling from a continuous-time signal, and then each value in the sequence is called a sample. When a discrete-time signal obtained by sampling a sequence corresponds to uniformly spaced times, it has an associated sampling rate; the sampling rate is not apparent in the data sequence, and so needs to be associated as a characteristic unit of the system.

Quantization Details

Quantization represents the amplitude component of the digital sampling process. It is used to translate the voltage levels of a continuous analog signal (at discrete sampling points over time) into binary digits (bits) for the purpose of manipulating or storing audio data. By sampling the amplitude of an analog signal at precise intervals over time, the converter determines the exact voltage level of the signal (during a sample interval, when the voltage level is momentarily held) and then outputs the signal level as an equivalent set of binary numbers which represent the original sampled voltage level.

Currently, the most common binary word-length for audio is 16-bit (1111110101000001), however, 20 and 24-bit depths are also common. 32 and 64-bit depths are available for hard-core audiophiles. The greater internal bit-depth helps to reduce errors in level and performance at low-level resolutions when multiple datastreams are mixed or processed within a digital signal processing (DSP) system.

16-bit word = 65, 536 steps20-bit word = 1, 048,576 steps24-bit word = 16,777,216 steps32-bit word = 4, 294,967,296 steps!

Upon playback, the digital words are then converted back into discrete voltages (at precise time intervals) allowing the originally recorded signal voltages to be re-created and played back.

Nyquist Theorem

In order for the desired frequency bandwidth to be faithfully encoded in the digital domain, the selectedsample rate must be a t least twice as high as the highest frequency to be recorded. For example, an audio signal with a bandwidth of 20 kHz would require a sampling rate of at lest 40 kHZ, and a 22 kHzsignal would require at least a 44 kHZ bandwidth. Failure to do this will result in aliasing, or the addition of extra, distorted artifacts in a recording.

Oversampling

Sampling at a higher rate generally results in a higher-quality analog-to-digital (A/D) and digital-to-analog (D/A) converter that sounds better. Since multiple samples are taken of a singe sample-and-holdanalog voltage, the average noise sampling will be lower.

Signal-to-Error Ratio

A digital system's signal-to-error ratio is closely akin (although not identical) to the analog concept of signal-to-noise (S/N) ratio. It indicates the degree of accuracy that's used to capture a sampled level andits step-related effects.

Dither

Dither is commonly used during the recording or conversion process to increase the overall bit resolution (and therefore low-level noise and signal clarity) of a recorded signal, when converting from a higher to a lower bit rate.

Technically, dither is the addition of very small amounts of randomly generated noise to an existing bitstream that allows the S/N and distortion figures to fall to levels that approach their theoretical limit.

Fixed vs. floating-point processing

Floating-point arithmetic is able to use numeric “shorthand” in order to process a wider range of valuesat an point in time. As a result, the processor is able to internally calculate much large bit depth values (32, 64, etc.).

Sound file sample rate

The sample rate of a recorded digital audio bitstream directly relates to the resolution at which a recorded sound will be digitally captured.

• 44.1k: the long-time standard of consumer and pro audio production• 48k: the adopted standard rate for use with professional video and DVD production• 96k: the de facto sample rate for high-resolution recordings

Sound file bit rate

The bit rate (or bit depth) of a digitally recorded sound file directly relates to the number of quantization steps that are encoded into the bitstream.

• 16 bits: long-time standard of consumer and professional audio production• 24 bits: this standard bit rate is often used in high-definition audio applications, often in

conjunction with the 96k sample rate (96/24)

Digital audio levels

You never want to go above “0” when outputing your digital audio as it can create distortion. As a rule of thumb, you can have your peaks occur between -12 and -20 dB. Digital distortion is a nasty sounding set of errors that will not do your recording any good.

Digital audio transmission

There are many standards and protocols for distribution digital audio signals throughout a system. Some of the notable ones are:

• ADAT lightpipe – uses standard Toslink connectors to transmit up to 8 channels of uncompressed digital audio at resolutions oup to 24 bit at sample rates up to 48k over a sequential optical bitsrea.

• S/PDIF (Song/Phillips Digital Interface) – uses an RCA cable to transmit information

Signal Distribution

Several devices can be daisy-chained together to record and playback audio channels.

Jitter

When devices that are daisy-chained are not all operating using the same master clock, or there is a mis-match in impedance between devices, jitter can occur. Jitter is a bad audio sound that happens when a soundwave is distorted due to the above reasons.

Wordclock

One way to avoid jitter is to use Wordclock, a standardized way of locking master clocks from various devices together so everything runs smoothly.

Digital audio recording systems

Samplers - can make short samples of audio signals, such as drums, horns, etc.

Hard-disk recording

Recording to a computer hard drive.

Hard-disk multitrack recording

Hardware devices that allow you to record multiple tracks simultaneously.

Portable Studios

These can be hardware, or a combination of hardware and software, as is found with the iPad and numberous accessories, such as high-quality portable mikes with pre-amps.

Flash-memory handhelds

These devices are so small they will fit in your pocket at times. They often come with high-quality electret-condensor microphones attached.

Basic Audio ProductionChapter 7: Digital Audio Workstations (DAWs)

Digital audio workstation software has revolutionized the recording process. Current software offers the following:

• Advanced multitrack, recording, editing and mixdown capabilities• MIDI sequencing, edit and score capabilities• Integrated video and/or video sync capabilities• Integration with peripheral hardware devices, such as controllers, MIDI and audio interface

devices• Plug-in DSP (digital signal processing) supportive• Support for plug-in virtual instruments• Support for integrating timing and control elements with other production software

Some of the things that DAWs offer are:

• Integration – provides centralized control over recording, editing, processing and signalrouting. Also provides for direct communication with production-related hardware and softwaresystems, as well as transport and sync control to/from external media devices

• Communication – a DAW should be able to communicate and distribute pertinent audio, MIDIand automation-related data throughout the connected network system. Digital timing(wordclock) and synchronization (SMPTE time code and /or MTC) should also be supported

• Speed and Flexibility – These are some of the greatest assets of a DAW: they can do thingsfaster and more flexibly than real-life counterparts

• Automation- the ability to instantly save and recall a session and to instantly undo a performedaction is a huge asset here.

• Expandability – can integrate hardware and software with little difficulty

• User-friendly Operation – easy to understand interfaces are easy to work with

Updates

The following aspects of DAWs are updated regularly:

• Need for speed• Increased computing power• Increased disk size and speed• Increased memory size and speed• Operating system (OS) and peripheral integration• General connectivity (networking and the web)

Portable Computers vs. Desktops

– Laptops are not as efficient as desktops at processing music– Mac laptops run faster than PCs when working on music, due to the BIOS– Laptops are great for portable recording– iPads and phones can be used to record mobile sound, and there are many software options

available– Quality portable mics and headphones can be used with iPads, etc.– Sessions recorded on portable devices can be transferred to desktops and laptops for further

work– Portable electronic software instruments can make composing on the go a breeze– iPads can serve as remote controls for desktop software, lighting, live performance triggering,

DJ samples, etc.

Interconnectivity

– USB (Universal Serial Bus – comes in 3 speeds)– FireWire (2 speeds)– Thunderbolt (2 speeds)– Networking (LAN – Local Area Network – can have mobile devices all linking to each other or

a central server)– Shared internet connection– Archiving/Backups

End of class notes