DIGITAL AUDIO explained


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Audio is the electronic information that represents sound, or rather, having sound of a temporary nature is the flow of information that represents it.

Sound is made up of pressure waves traveling in space, therefore it is represented by a sinusoidal.

Digital Audio

The characteristics of a sound are:

Amplitude: Measured in Hertz (Hz) and determined by the frequency of a sound, the higher the frequency, the louder the sound, the lower it is, the lower the sound.

Intensity: it is measured in decibels (db) and is determined by the power of a sound, the more intense a sound is, the greater its volume.

Duration: It is measured in seconds (s) and dermal how long a sound lasts over time.

Timbre: It is not directly measurable, but it is that sound parameter that allows us to distinguish a trumpet from a drum. It constitutes the trace of a sound and is characterized by harmonics.

digital audio

ANALOGUE AND DIGITAL

There are two different ways of representing sound as electronic, analog and digital information.

Analog audio was the first, in chronological order, to be developed.

The information varies similarly to the information it represents and can (in theory) assume any value.

If we greatly expand the sine wave that describes an analog sound, we would see that it is a continuous line without interruptions.

Instead, digital audio is encoded with a number system, which allows discretization (transition from analog to digital), during this step information is lost, but once the sound is written as a series of numbers (digital information) it is possible to reproduce it. , transmit and modify it without losing anything in terms of quality, which is impossible with analog information.

If we greatly expand the sine wave that represents a digital sound, we would realize that it is not a continuous line as in the previous case, but a series of points very close to each other.

The amount of these points in one second of information will define the “sampling frequency”.

The amount of information that each point can contain is called “bit depth”.

THE CHARACTERISTICS OF DIGITAL SOUND

Sampling rate

Determine the number of samples contained in one second of information.

It is expressed in hertz (Hz) and generally assumes the following values ​​in the musical field: 22050Hz, 44100Hz, 96000Hz.

According to Nyquist’s theorem, each sampling frequency can record and reproduce sounds that have a maximum frequency equal to half of the chosen sampling frequency, this means that a piece sampled at 44Mhz can assume values ​​of up to 22Mhz only

Bit depth

Determine the amount of information contained in each sample.

It is expressed in Bit (bit) and generally assumes the following values ​​in the musical field 8Bit, 16Bit and 24Bit.

Above all, this is the parameter that depends on the quality of a sound.

Transmission rate (bit rate)

It is a characteristic of codecs, that is, of the “machine language” used to describe a sound.

Sets the total amount of information needed to play a second of a sound.

It is expressed in Bit / s.

AUDIO PROCESSING

Whether you’re talking about studio recording or live performances, the audio signal is never sent directly from the microphone to the speakers / recording medium, but is always processed first, through tools that allow you to perform different interventions. in the sound

These instruments can be analog, therefore they have the instrument physically in the studio (which is usually inserted inside a shelf), which must be connected between the microphone and the mixer or between the mixer and the speakers / recording medium.

Or you can simulate them through some plugins for your computer.

It is necessary to have a Daw (Digital Audio Workstation), which is the workspace in which all editing operations are performed. (Ableton, Cubase, Fruitloops, Logic, Reaper).

Within this software it is possible to install smaller ones, called VST (Virtual Studio Technology) that simulate the circuits of the studio equipment, emulating the effect.

(There are also other proprietary plugins with extensions other than the classic VST like .component or .au).

Some tools are essential and are used in all audio recordings, others are used only in particular situations or to obtain / avoid certain effects.

The main ones are:

Equalizer, is used to emphasize or attenuate some frequencies, this way you get a cleaner sound and a less “mixed” mix where all the instruments occupy only the correct frequencies, without overlapping.

The compressor, as the name suggests, serves to compress the dynamic range, so that the sound is more consistent and less dispersive.

Amp, wavering of different kinds, is used to increase the intensity of a sound.

Limiter works in a similar way to the compressor, but instead of compressing all frequencies, it attenuates those that exceed a predetermined threshold (threshold), avoids entering faults.

Reverb adds a slight reverb that makes a sound recorded in a soundproof studio much more natural than it would be too “dry”.

Filters (high / low cut) allow you to cut some useless and sumptuous frequencies too low or too high. (They are just 1 band parametric equalizers).


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Digital audio formats on the network

Digital audio formats on the network:

WAV: Waveform files (or simply wave) are the most common sound formats on Windows platforms. WAV files can also be played on Mac and other systems with player software.

MPEG (MP3): The Motion Pictures Experts Group (MPEG) format is a standard format with significant compression capability. MPEG level 3 or MP3 files are frequently used for web music distribution. However, due to their size, MPEG files must be downloaded completely before playing them.

RealAudio (.rm): Real Audio is the technology that currently predominates on the Web. You need a proprietary player, but the basic versions of the player are available for free.
MIDI: The Musical Instrument Digital Interface format is not a digital audio format. It represents notes and other information so that music can be synthesized. MIDI has good support and its files are very small, but it is only useful for certain applications because of the quality of its sound when played on PC hardware.

AU: The u-law format is one of the oldest sound formats on the Internet. Players are available for almost all platforms.

RMF: The Rich Music Format supported by Beatnik (www.beatnik.com) is a high quality audio format, primarily for “download-and-play”, which is becoming increasingly popular.

AIFF: The Audio Interchange File Format is very common on Macs. It is widely used in multimedia applications, but it is not very common on the Web.

Flac: Free Lossless Audio Codec (FLAC) (Lossless audio compression codec) Ogg project format without loss. The initial file can be completely recomposed with the disadvantage that the file occupies much more space than would be obtained when applying lossy compression or Lossy.

Digital audio on the network:

The digital sound is measured by the sampling frequency, or how many times the sound is digitized over a certain period of time. The sampling frequencies are indicated in kilohertz (kHz), which indicate the number of times the sound is sampled per second. The CD sound quality is obtained with 44.1 kHz, or 44,100 samples per second. For stereo sound, two channels are required, each 8 bits; At 16 bits per sample, this results in 705,600 bits of data on a CD, producing high quality sound, at the request of the end user. In reality, the transmission of this amount of data would occupy almost half the bandwidth of the T1 network. As the average user of the Web does not have this bandwidth, another solution is necessary. One possible solution is to decrease the sampling rate when digital sound is created for sending through the Web. A sampling frequency of 8 kHz, in mono, would produce acceptable results for simple applications, such as language, especially if we consider that the playback hardware generally consists of a combination of a simple sound card and a small speaker. Low quality audio does not require more than 64,000 bits of data per second, but the end user still has to wait to download the sound. Modern users need several seconds to receive, even in the best conditions, a single second of low quality sound, making continuous sound impossible.