What is digital audio?


Free Download Mp4Gain
picture

What is digital audio?

Digital Audio

Digital sound is nothing more than a combination of numbers. With a certain algorithm, sound, such as air pressure, is converted into data streams and encoded for further processing and playback. Depending on the algorithm used, the music file has one format or another, one or another extension.

Analog Vs. Digital Sound

Remember that along with digital sound, there is analog sound, which is represented by a continuous electrical signal that reflects the change in the sound wave. The analog to digital sound conversion is a setting of the numerical value of the amplitude at a given time with a given density of values. Consequently, the more values ​​that are recorded, the more reliable and accurate the image of the digitized sound fragment is recreated. With such digitization, very voluminous data matrices emerge that, depending on the format used, differ in the sound quality / volume ratio of the final file.

Perhaps the main advantage of digital audio over analog is the ability to store and copy data indefinitely without losing the original quality (whereas when copying from one analog medium to another, a decrease in recording quality is quite noticeable).

The most widespread and popular digital audio format today is MP3 (MPEG Layer 3). It was developed, after a series of intermediate formats and investigations, started in 1987, by the Fraunhofer Institute in Germany.

The developers of the format were faced with the task of simplifying and reducing the cost of shipping long musical fragments. As you know, one minute of a stereo signal from a CD (16 bit, 44.1 kHz sample rate) takes up about ten megabytes of memory. At the same time, unlike text or graphic files, the audio signal cannot be compressed without loss of quality. Thus, modem transmission of an uncompressed composition from an audio CD lasting 3 minutes at a data transfer rate of, say, 24 kbps will take several hours. Scientists at the Fraunhofer Institute managed to achieve multiple file size compression: on average, one minute of a compressed audio signal in MP3 format takes about 1 megabyte. The principle of compression is based on the elimination of “unnecessary” sounds from the music file, to which the human ear is immune, or that duplicate each other.

The main factor that determines the relationship between file size and sound quality within a given format is the bit rate. Bit rate is an indicator of how much information a second of sound encodes. The higher it is, the less distortion and the closer the encoded composition is to the original. The most common on the Internet are compositions with 128 and 192 Kbps bitrates. The maximum bitrate supported by programs and devices that work with MP3 is 320 Kbps. In practice, only an expert or a professional who works with sound can notice the differences between an MP3 file with a 320 bit rate.

To optimize the size of MP3 music files while maintaining decent quality, a variable bit rate (abbreviation VBR – variable bit rate) is used. In this case, the encoding program divides the file into fragments of different spectral saturation and encodes them with a suitable bit rate. Most modern MP3 players support variable bit rate playback. A significant advantage of MP3 files is that they can contain the name of the artist, the name of the track and the album, the year of its release, etc. The set of this data is called ID3 tags. Most modern gamers can read and display them on the screen.

In 2001, Swedish Coding Technologies and Thomson Multimedia developed the MP3 Pro codec. It is MP3-based and as a result is fully MP3 backward compatible and only partially forward compatible. It uses SBR (Spectral Band Replication) technology, so the codec provides good quality at low bit rates. However, the encoding quality at medium to high bit rates is inferior to that of most other codecs. For this reason, this format is mainly used for broadcasts on the Internet and demonstrations of fragments of new musical compositions.

Another type of MP3 was the development of MP3 Surround, recently introduced by the creators of MP3: the Fraunhofer Institute. This format repeats all the characteristics of multichannel sound, while still being compatible with standard stereo MP3: information describing the spatial characteristics of the sound is recorded on an additional track. By playing files of this format on special equipment capable of reading this track, you can obtain surround sound that conforms to the Surround 5.1 standard.


Free Download Mp4Gain
picture


Mp4Gain Main Window
picture


Mp4Gain Features
picture


Free Download Mp4Gain
picture

Digital audio encoding: data reduction

Mp3 encoding

Since the introduction of the compact disc audio (CD) and the advent of digital audio tape (DAT), digital technology has become increasingly popular in the audio industry. Both CD and DAT use pulse code modulation (PCM) as a basic digitizing process. This technology translates the original analog audio signal into the digital world through sampling, quantization, and encoding. Since PCM does not use data reduction, excellent sound quality is achieved, but is purchased at the cost of high memory requirements. In PCM, a CD can contain a maximum of 80 minutes of audio data.

Mp3 Encoding

Why reduce the audio data?

The high memory requirements of PCM, in particular, made direct use of this technology in multimedia or digital radio systems ineffective, time-consuming or impossible. These systems require a radical thinning of the audio signals. The reasons for this are insufficient broadcast transmission capabilities, the limited transfer rate of current bus systems (PCI, IDE, SCSI) and, above all, the still lack of storage space. Not only is there a shortage of hard drive space, but the main memory in today’s PC systems also offers insufficient reserves to allow sensible work with PCM audio data. If you consider that a 6-minute piece of music in PCM requires up to 60 Mbytes of memory (WAV file), it is easy to imagine that streaming this piece, for example over the Internet, is not profitable. not to mention classic works that last several hours. The result would be extremely long download times.

On the other hand, digital technology has unbeatable advantages over analog technology. Very good sound quality, immunity to interference and relatively easy technical manageability were reasons enough for several research institutions to develop more and more methods in recent years that allow to reduce the storage requirements of digital audio signals and, therefore, its use in new areas such as digital broadcasting. The main objective was to maintain sound quality, using the CD as a reference. The result is a whole series of codecs, some of which save a considerable amount of data. At the moment, the MP3 codec, developed by Motion Pictures Expert Group (MPEG), which is widely used on the Internet, is probably the best known, but also MPEG 2, AC-3,

The amount of memory required by a digital audio signal is primarily determined by the bit rate and the sample rate. Both parameters can be adjusted while encoding the signal. The next section examines the effects of changing the sample rate and bit rate when processing signals.

According to Shannon’s sampling theorem, sampling must take place with at least twice the maximum frequency of the function to be discretized. In the audio range, where 20 kHz is the upper limit, at least 40 kHz is required. The CD uses 44.1 kHz to avoid aliasing effects. Sampling can be used to reduce the data. Lowering the sampling rate results in fewer samples that need to be stored. Needless to say, this dramatically reduces the storage requirement. Unfortunately, this tactic has one major drawback. If you reduce the sampling rate, you can easily conflict with the sampling theorem. If you wanted to sample an audio signal with the full frequency range (20Hz – 20kHz) with, for example, only 20kHz, extreme alias distortion would occur. Playing music would be completely impossible. However, sampling is sometimes a way to reduce the data rate. If, for example, only speech intelligibility is desired without high-quality music reproduction, the 20 kHz frequency range is unnecessary. 3 kHz is sufficient as the upper limit frequency. Here the audio signal can be band limited to 3 kHz with the help of a low pass filter and the sample rate can be reduced to a minimum of 6 kHz. One possible use of such low sample rates would be telephone applications, for example. Here, the audio signal can be band limited to 3 kHz