Posts Tagged ‘Recording’

Recording Technologies Used for Audio Bible Podcasts

New scientific high-technology recording devices are used to bring the Word of God to people who might not otherwise have any way of hearing God’s Word. The result of this scientific technology on the Word of God is the audio Bible. There are various formats in which you will find the audio Bible available, including a Bible on audio CD and audio Bible mp3 formats, as well as Internet distribution by podcast. This technology also brings another benefit: it is highly portable, requiring only a small digital sound system connected to a computer. This means that recording of the Bible can occur anywhere in the world. The result of this ease of mobility is that there can be recording centers where the audio Bible is recorded in hundreds of different indigenous languages. People worldwide, who might not have any other means for reading and understanding the Bible, can now be exposed to the power of God’s Word in an audio bible mp3 version, which they can hear.

Digital recording provides clear, crisp audio, and is a preferred method for capturing the sound required for an audio Bible. There are basic techniques used for capturing a an audio Bible recording. A speaker reads the Bible into a microphone, which produces an analog signal. Then this signal is sent from the microphone to an analog to digital converter. This component’s job is to convert the analog signal into a binary code. This code is then sent through a bundle of transmitters, usually wires connected to the computer in the form of cables, and eventually ends in storage on the computer, often on a hard drive or a CD burner.

The beauty of digital recording is that any mistakes can be taken out by simply removing the exact part of the code stored on the computer, which makes for much more precise editing and cleaner sound as a result. During the editing process, music and sound effects can also be added, and are, in order to produce a final audio Bible version that is dramatized, and more compelling to listen to. You can have speakers from anywhere in the world recording their part of the final version, then sending it via the Internet to the mixing center, where all of the parts are combined together and edited to result in the final version of the audio Bible. Speakers who come from oral cultures can relate to drama very well, and this is an important aspect of the recordings that make the audio Bible so influential.

Lasers are used in the recording and playback of CDs. The laser burns tiny holes onto the CD’s surface. This allows a CD or DVD player to distinguish between parts of the disc that allow reflected weak laser light to permeate through holes and parts of the CD surface that do not, resulting in a reading of the digital data and playback through a speaker so that you can hear it. These are some methods used for audio Bible production, to share God’s Word worldwide.

An Introduction to the Properties of Sound for Recording Musicians

An Introduction to the Properties of Sound for Recording Musicians

by Clay Butler

Obviously, if you’re reading this article, you have an interest in recording.   I’m sure you also have some notion as to what sound is.  Sound is what we hear, right?  Well, yes.  But there’s a lot more to sound.  Knowing what sound is and how sound works is the key to getting a quality recording.  Let’s get our feet wet in sound waves (pun intended).

WHAT IS SOUND?

Simply put, sound is Acoustical Energy, or vibration.  More specifically, sound is vibration propagated through a medium, which is then received by our ears and interpreted by our brain as sound.  The reason I say the vibrations are interpreted by our brain as sound is because acoustical energy still exists that we cannot perceive as sound (more on this later).  For example, while a dog might go nuts from hearing a dog whistle, we as humans cannot interpret those vibrations as sound.  Radio, cell phones, Wi-Fi, microwave ovens, and RADAR all use sound waves that we as humans just can’t hear.  So, in essence, when we refer to sound, we really refer to our perception of vibrations.

Sound vibrations that are propagated through the air (or any other medium for that matter) are in the form of transverse waves.  Thus, you could also say that sound is really rapid fluctuations in air pressure.  The vibrations from a vibrating body, such as a guitar string, push and pull on the surrounding air, creating positive and negative pressures.  These waves occur as the molecules of air are slammed closely together as they are pushed by the vibrating body.  The part of the wave consisting of positive pressure, where the air molecules are slammed together, is called a compression.  Negatively pressured parts of the wave, where the air molecules are pulled apart from each other, are called rarefactions.

THE PROPERTIES OF SOUND

How loudly a sound is perceived is determined by how hard the air gets pushed by the vibrating body.  The harder the air gets pushed, the louder the sound.  Although this is typically referred to as volume, in reference to acoustical energy, it is called Sound Pressure Level (SPL).  The scale used to measure Sound Pressure Level is the Decibel scale, or dB SPL (more on the other Decibel scales in a later article).

The pitch of a sound is a function of frequency.   How frequently a vibrating body pushes the air determines how high or low the tone of the sound is perceived.  The more frequently the air gets pushed, the higher the tone of the sound.  As the air gets pushed less frequently, the tone sounds lower.  Therefore frequency is expressed as the number of sound waves occurring over time.  The scale used is known as Hertz (Hz), which signifies the number sound waves per second.  For example, the note “A” below “Middle C” on a piano is 440Hz.

The frequency spectrum is broken up into three parts.  The Audible Range for humans is roughly 20Hz to 20,000Hz (or 20 Kilohertz, abbreviated as KHz).  The frequencies below 20Hz are categorized as Infrasonic.  All frequencies above 20KHz are referred to as Ultrasonic.

ACOUSTICS

Let’s look again at sound being all about our perception.  Generally, we don’t hear the sound emanating directly from its source.  The majority of the time, we hear sound after it bounces off the surrounding walls, along with any other nearby surfaces, and interacts with the room.  We call this acoustics.  Understanding how acoustics influence sound, especially those sounds you’re trying to record, better enables you to get the sound you want without any surprises.  Each time a sound wave is reflected back into an acoustic space, our perception of that wave changes, especially when you hear sound coming directly from a sound source in addition the reflected waves.  This is why it’s not uncommon for a pro recording studio to spend hundreds of thousands of dollars on architectural design and acoustic treatment, so that those extra reflections and wave interference are eliminated.

http://www.claybutlermusic.com

© 2009 Butler Productions

Advanced Audio Recording Techniques

Hard Disk / Computer-Based Recording

One of the biggest trends in recent audio production has been to merge digital audio with computer technology to create a samplebased approach to sound recording. The encoding of audio data into digital memory or onto a storage medium provides us with a means for storing or manipulating defined blocks of digital data. This data can be stored as a soundfile such as .wav, .aiff or SDII.

Perhaps the most important difference that can be distinguished between a tape-based system (digital or analogue) and samplebased recording system is random access. Random access production refers to the fact that digital audio can be stored within a random access memory (RAM), or a disk based memory medium in such a way that the data can – virtually instantaneously – be accessed, processed, or reproduced in any order at any point in time.

Once developers began to design updated sample editor software, it was discovered that through additional processing hardware, digital audio editors were capable of recording digitized audio directly to a computer’s hard disk. These devices, sometimes known as digital audio workstations (DAW), serve as computer based hardware and software packages that are intended specifically for the recording, manipulation, and reproduction of digital audio that resides on hard disk.

Commonly, such devices are designed around and controlled by a standard personal computer with the addition of a sound card which provides the input and output interaction with the computer.

There are multiple advantages to using digital audio workstations in an audio production environment.

- The capability to handle longer sound files. Hard disk recording is limited only by the size of the hard disk itself (commonly one minute of stereo recording at 44.1 kHz occupies 10.5 MB of hard disk memory or 5MB / track minute).

- Random Access editing. As audio is recorded on the hard disk, any point within the program can be accessed at any time, regardless of the order in which it was recorded.

- Nondestructive editing allows audio segments (often called regions) to be placed in any order, manipulated in any fashion without changing the originally recorded sound file in any way.

- DSP. Digital signal processing can be performed on a segment or entire sound file in either real time or non-real time in a nondestructive fashion.

- In addition to these advantages, computer-based digital audio devices serve to integrate many of the tasks related to both digital audio and MIDI production. Many DAW’s are capable of importing, processing, and exporting sound files into formats such as mp3 or Real Players G2.

Recording Techniques

FILTERS

Also known as equalization or EQ, filters are used to increase or decrease the level in a specific range of audio frequencies. The most common filters are the simple bass and treble controls found on inexpensive stereo systems, which act on a broad range of frequencies. But other filters are designed to surgically boost or cut very narrow bands of the audio spectrum.

SHELVING FILTERS

As the simplest form of filter, shelving EQ boosts or cuts all frequencies above or below a fixed frequency. A bass shelving filter, also called a low-pass filter, boosts or cuts everything below its fixed center frequency. Likewise a treble shelving filter, also called a high-pass filter, boosts or cuts everything above its fixed center. A single control typically adjusts the amount of boost or cut.

These filters are useful for making broad changes like reducing boomy bass and wind noise. But encoders can easily be overloaded by too much bass or treble, so it’s often wisest to use these filters to cut high and low frequencies to prevent artifacts.

BANDPASS FILTERS

These filters can be used to boost or cut audio on both sides of a center frequency. Bandpass filters are commonly used as midrange filters, because they have little effect on either high or low frequencies. The familiar graphic equalizer is just a set of bandpass filters tuned to different center frequencies.

More sophisticated versions, called sweepable bandpass filters, have an additional control allowing you to change the center frequency. Bandpass filters are useful for increasing the intelligibility of a speaker without increasing hiss or background noise. A variation of the bandpass filter is the notch filter, which boosts or cuts all frequencies except those around the center frequency.

PARAMETRIC FILTERS

A parametric filter is a bandpass filter with an additional control to adjust the width of the frequency band being effected (fig. 3). These are the surgical tools of audio editing. They can be used to eliminate just the noise from an air conditioner, while having a minimal effect on the rest of the audio.

With all filters it’s important to follow the audio engineer’s first rule of EQ — cut rather than boost wherever possible. Cutting undesired sounds is always less obtrusive, and boosting too much can make a track too loud and lead to distortion and artifacts when encoding.

COMPRESSORS

A compressor’s basic function is to reduce the dynamic range of an audio recording, which is the difference between the loudest and softest sounds that pass through the recording chain. Simply put, a compressor is a processor whose output level increases at a slower rate as its input level increases.

By reducing the volume of the loudest sounds, a compressor lets you raise the level of the entire audio track, making it all sound louder than it actually is. Compression can be a big help in achieving intelligible audio tracks with a more uniform volume that will survive the encoding process.

A compressor consists of a level detector that measures the incoming signal, and an an amplifier whose gain is controlled by the level detector.

A Threshold control sets the level at which compression begins. Below the threshold, the compressor acts like a straight piece of wire. But when the input level reaches the Threshold, then the compressor begins reducing its output level by an amount determined by the Ratio control.

The Ratio control establishes the proportion of change between the input and output levels. If you set the compression Ratio to 2:1, then when the input signal gets twice as loud, the output signal will increase by only half.

If you set the Ratio to its maximum (10:1 or more), the the compressor becomes a “limiter” that locks the maximum level at the Threshold.

While a compressor can level out a recording, high levels of compression can also introduce artifacts including “pumping”, in which there is an audible up and down change in volume of a track, or “breathing”, which sounds like someone breathing as the background noise level goes up and

down.

EXPANDERS

An expander is the opposite of a compressor. As the level of the audio signal gets louder, the expander’s amplifier turns up further making loud signals even louder. An expander can be used to reduce noise in a process called downward expansion. In this case you set the Threshold just above the level of background noise. The expander will then raise the volume of everything above the Threshold, but won’t change anything below the Threshold, thereby lowering the perceived background noise.

NORMALIZING

Normalizing increases the gain of the audio file until its loudest point (or sample) is at maximum level. The overall signal level is now higher, which makes for clearer audio, and also gives the encoder more bits of data to work with and reduces encoding artifacts. The only downside of normalizing is that it increases the noise as well as the audio signal so it should be used carefully. It should be your last step before encoding, and you may not need it at all.