Why does digital music need to be normalized?


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Why does digital music need to be normalized?

For younger consumers, the focus is often on the computer, which plays MP3s through the PC’s speakers. “They’re made to rumble a lot during games,” says “c’t” expert Zota. This can be useful when reproducing the explosions in a shooting game. However, when listening to music, such boxes disappoint.

Digital Music

Other consumers use their iPod with clip-on speakers, and mini systems like Bose’s “Wave Music System” are enjoying best-sellers. Of course, they cannot match the tonal volume of a full floor standing speaker.
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Digital music

Those who decide to buy a high-quality music system generally turn to home theater systems. These are multi-channel systems with up to eight speakers and multiple power amplifiers. Their specialty is DVD playback, where they evoke powerful bass thanks to the subwoofers.

The viewer also physically experiences an earthquake in the movie because the shelves begin to shake. Solo: Compared to pure stereo systems, some home theater systems are disappointing. Some subwoofers are too inaccurate to play music. Above all, the quality is significantly more expensive compared to stereo systems. “The budget has to be divided into many more individual parts than with a stereo system,” says Besic, specialist in “Stereoplay”. For 1000 euros there is a decent stereo, but only a lousy home theater system. According to GfK, Germans spend an average of just over 400 euros on complete home theater systems, and 800 euros if these consist of the individual components of an amplifier, CD player and speaker cabinets.

Music producers flatten recordings

But it’s not just bad speakers that degrade sound quality. Music producers also contribute. They have been making their songs louder and louder since the mid-1990s. In pop, hip hop, rock, and electronic dance music, there are practically no quiet passages. At the same time, musical recordings have lost their dynamism. The mids are emphasized, but very high and fine sounds, as well as very deep bass, are often missing. The idea behind it: the songs should appear and assert themselves against loud advertising on the radio or background noise in the pub.

Additionally, sound engineers increasingly manipulate the sound of rock bands and pop singers with just a few clicks. Engineers use computer programs to smooth the edges and eliminate the smallest errors. For example, the pitch of the song is fine-tuned later; and hand-played drums sound accurate after computer processing, but like a machine and somehow always the same. Not much remains of the musicians’ own sound.

“In addition, the generally short time due to lower budgets also plays a role. In the past, you had much more production time, which of course was reflected in the end result in better quality and creativity, ”says Gerhard Wölfle, director of Dorian Gray Studios in Eichenau, near Munich. Wölfle has recorded CDs with the bands Guano Apes, Reamonn and The Donots. In the past, around six weeks of production time was the guideline for such albums. Today, studio professionals are satisfied when the music industry and artists spend half their time on them. Gerhard Wölfle says: “The excessive volume due to the massive use of compressors and limiters definitely gives many productions to the rest”.

An excellent example of an extremely loud album is the album “What People Say I Am, That’s What I’m Not” by English band Arctic Monkeys from 2006. The fully adjusted mix quickly rose to the top of audience favor. . The single “I bet you look good on the dance floor” (see the band’s MySpace profile) became a number one hit.

All this has generated a problem in matters such as the loudness of the music, which almost necessarily must be normalized to get them to sound at a similar volume.

Mp4Gain is the perfect choice to get a boost to the loudness of a song or to make all instruments sound clearly and audible.

Mp4Gain offers the latest technology and algorithms to make your music sound great today.


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Basics of digital audio

Basics of digital audio:

Before the computer can record, manipulate, and reproduce sound, sound must be transformed from an audible analog form to a computer-acceptable digital form, using a process called analog-to-digital conversion (ADC). Once the sound data has been stored as bytes in the computer, the power of the computer’s CPU can be used to transform this sound in thousands of ways. Finally, when you are ready to listen to the result, the digital-to-analog conversion (DAC) process transforms the sound bytes back into an analog electrical signal from the speakers.

Sampling: Analog to Digital Conversion

Given an analog signal, discrete values ​​of its amplitude are taken at small time intervals, obviously the more reliable the reproduction the more samples per second are taken. These obtained values ​​are assigned a digital value that the computer can understand and process as required. We can use 8 or 16 bit words, thus obtaining 256 or 65536 different combinations and obtaining higher resolution.

 

SAMPLE FREQUENCY: According to the Nyquist theorem, it is possible to accurately repeat a waveform if the sampling frequency is at least twice the frequency of the component with the highest frequency. The highest frequency that the human ear can perceive is close to 20 kHz, so the 44.1 kHz sampling rate of sound cards is more than enough. This value is the one used today by CD audio players.

SAMPLE SIZE: The sample size controls the dynamic range that can be recorded. For example, 8-bit samples limit the dynamic range to 256 steps (50 dB range). In contrast, a 16-bit sample has a dynamic range of 65,536 steps (90 dB range) a substantial improvement. The human ear perceives a whole world of differences between these two sample sizes. Ears are more sensitive to detecting differences in pitch than intensity, but are even more sensitive to the strength of sound.

From the previous processes we can get an audio file, such as (and since it is the best known), a WAV audio file. It is the own format of Windows. They can be 8 or 16 bit with sampling rates of 11,025 kHz, 22.05 kHz, or 44.1 kHz and generally have good sound quality.

Digital audio compression

It could be assumed that all you have to do to get good sound is to record at the 44.1 kHz speed limit with 16-bit (2-byte) samples. The only problem that appears if recording in stereo, sampling simultaneously on the left and right channels at 44.1 kHz, a one minute sound sample needs a 10.58MB storage space. This involves using large disk spaces to store these sound files. Many compressed file formats (codecs) have been developed that enable high-quality recording without the need for so much disk space.

Most common audio formats:

With the simple objective of listing a series of codecs used by different operating systems to perform audio compression. Later, a more complete description of the most used is made: MP3.

Therefore, some of the most used are:

Advanced Audio Coding (AAC): used by Apple computers. More efficient than MP3.

Audio for Unix (AU): Acoustic standard for the JAVA programming language.

Windows Media Audio (WMA)

Ogg Vorbis: It is free, open and not patented.

Atrac: compression and playback technology for minidisc.

 

The codec par excellence: the MP3

Its origin and current

The abbreviations MP3 respond to the abbreviation of MPEG (Moving Picture Expert Group) 1 Layer 3, which is a perceptual coding algorithm. This among others was developed by the Moving Picture Expert Group (MPEG) (http://www.cselt.it/mpeg/) together with the Fraunhofer Institute of Technology (http://www.ipa.fhg.de/english/ ).

Moving Picture Expert Group is an ISO / IEC research committee. MPEG is in charge of the international development of compression, decompression, processing and encoded rendering standards for movies, audio and the combination of both. It is a non-profit institution created in 1988, which brings together 300 experts from 20 countries three times a year.