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Digital audio formats, compression and connectivity

This article discusses in the detail the questionable topics for the users such as, Digital audio format, compression and connectivity.
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The popularization of the network of networks changed everything, for the worse at first (at least if we are interested in quality) and for the better later. But at the beginning the damage was catastrophic, a true regression. Why? By data compression, materialized in formats such as MP3. Many fans will undoubtedly wonder about the digital audio format, the word compression, its meaning and what it entailed.

Leaving aside the value judgments on the ethical-moral component of massive free downloads, we will concentrate on the technical part of the question. And the reality is that before the advent of broadband, ADSL in its different variables, the only way we had to download Internet content were modems with a binary transfer speed of 56 kb / s. Yes, none of those megabits per second that businesses and consumers compete for right now: 56,000 bits per second was the most. You don’t have to be a computer engineer to realize that with such a figure downloading a simple song from a CD (and let alone a movie in VHS resolution… at the most) was a real drama.

Data compression or the technique that changed everything about Digital audio format

Without intending to extend ourselves in the description of this critical concept, it is important to know that compression is understood as a scheme that allows, by means of a “decision” algorithm in turn based on a series of “rules” (which in the case of audio are masking and threshold of audibility) reduce the amount of data to transmit a given message. In other words: if song “x” occupies 1 million bits in the Digital audio formats used to encode the sound of a CD , data compression allows that song to be reproduced with maximum intelligibility using only 50,000 of those bits.
                                                                                                      This graph shows Format vs Resolution
Thus, downloading a CD Completion from a given website could be completed in a reasonable amount of time. But, of course, the price to pay was high in terms of quality because such a “castration” of the original message (which in turn was not “continuous”, analog, but also digital, although “linear”, without compression) implied suppressing many nuances of music, a disaster that in reality did not care for many consumers but it did worry, and much, those who bet on that High Fidelity in the reproduction of the sound that we are so passionate about and that received a wound that was almost fatal . In this sense, it is worth knowing that the “philosophical” keys to data compression are summarized in two terms: redundancy and irrelevance. In the first case, It is about rearranging the available data to eliminate those that are repeated (for whatever reason: security, etc.), a bit like a computer “zip” file. It is a formal remodeling that does not affect the sound message at all (but it does save space to transmit / save data, so it is very practical), so in this case, we are talking about lossless compression. ”. It is the second term that has the greatest scope in terms of sound quality because the idea of ​​irrelevance implies suppressing irrelevant data from a certain message. And, of course, who decides what is relevant or not? Well, an algorithm, a program that, obviously, can be more or less sophisticated but still makes decisions with which everyone will agree. It is easy to understand: what may be irrelevant to such a person and / or the team may not be irrelevant to another. The fact is that here musical information is suppressed, which, as a fundamental issue, can then no longer be recovered. Well then: algorithms in which there are losses of musical information are known by the name of “lossy” or lossless encoding algorithms. From what has been said it is easy to deduce that the difference between the concepts “lossless” and “lossy” is the one that marks the boundary between high and low quality digital audio, between high resolution (with the formats with recording studio quality or “Studio Master” at the top) and that “practical” sound (in principle for portable players and cars) and very often unnatural from Digital audio formats such as the once omnipresent MP3, which, we insist,


ADSL, the key to access to High End audio via the Internet

At bottom it was a purely technical progress that, logically, had to come. A progress that would allow breaking the limitations that prevented downloading a song recorded in PCM at 16 bits / 44’1 kHz and, over time, files with a much higher resolution than for a good decade and a half are the usual ones in studios of recording. So, thanks to ADSL, the High End in audio via Internet, and, therefore, “without physical support” is available to everyone. At this point, it will be well to briefly review the small “soup” of acronyms that we can find, otherwise the result of the availability of open and “closed” environments (Windows, Mac), in terms of CODEC’s (algorithms that compress and decompress data, in this case music) refers to how compression is the norm.

AAC (Advanced Audio Coding) :   It was designed to be the successor to MP3 and, although it is a lossy CODEC , the results in terms of sound quality are superior to those of MP3 for the same bit rate. The AAC has been adopted for use by a wide range of portable audio devices such as the iPod and its derivatives.

AIFF (Audio Interchange File Format): It is the version of WAV created by Apple. It works with files without compression (that is, “lossless”) that maintain full resolution and size.

ALE (Apple Lossless Encoder), also known as ALAC (Apple Lossless Audio Codec): Uses lossless compression to save storage space. Once unzipped for listening, the file will be identical bitwise to a full-size one encoded in WAV or AIFF. As in AIFF or FLAC, in ALE / ALAC files the metadata is attached to them.

DSD  (Direct Stream Digital): Like Linear PCM, it is a “raw” digital encoding format, and therefore without any compression. Originally developed for the SACD “Super Audio Compact Disc”), it is used by recording studios that consider it musically superior to the aforementioned Linear PCM, the result being a growing number of content encoded in it that can be accessed via specialized sites from Internet. In this case, two options are available: DSD64 and DSD128, which can be found, respectively, in the DFF 2.82 MHz and DFF 5.64 MHz file formats.

FLAC (Free Losssless Audio Codec): It is a free and lossless format, so it will generate full-size files, like WAV. It was originally developed to offer compression of audio files without loss of musical information, being able to compress files up to 50% of their size. Like the ALE format, once unzipped, the file is identical to a full-size one. As a consequence of this feature and the ease of adding metadata to files (i.e. information regarding song, composer, album, cover art, etc.), FLAC has become the de facto standard for recording audiophile quality music.

LPCM (Linear Pulse-Code Modulation) or Linear PCM: It is not a CODEC, but a method for digitally encoding audio signals. It is the standard form of digital audio data used on media such as CD, DVD, and Blu-ray Disc. It is the “raw” and uncompressed format of digital audio signals that are input to digital audio processors or DACs through the S / PDIF (both unbalanced and balanced), TosLink and PC-USB inputs.

MP3 (MPEG-1 / MPEG-2 Audio Layer 3): By far the most popular of the digital audio formats that was available for use in computers and players. Lossy type, that is, lossy, it offers small file sizes at the expense of reducing file content and therefore sound quality.

Ogg Vorbis : This is a lossy format developed as a free alternative to MP3 and AAC that offers similar options in terms of reducing file sizes, as well as bitrates of up to 500 kHz.

WAV (Waveform Audio File Format) or WAVE: Create files in full size and with full resolution. In this case, the files are not compressed, so they consume a lot of storage space but are easily transferred to other platforms. The only weak point is that the metadata is not attached to the music files automatically, so if the user moves the files, they must also do the same with the metadata.

WMA (Windows Media Audio): It was developed by Microsoft as an alternative to MP3, which means that we are facing a CODEC with losses (“lossy”). It can sample signals at a maximum frequency of 48 kHz and is widely used by Internet radio stations.

WMA Pro : It is a version of WMA with the improved compression algorithm, although it is still of the “lossy” type

WMA Lossless : A lossless version of WMA with multi-channel capability and support for signals up to 24 bit / 96 kHz. It allows to make bit-by-bit copies, that is, with audiophile quality, to understand us.
Also note that each of these files is identified with a specific extension (.aac, .wma, .mp3, .flac, etc.).

Digital audio format connectivity options

The final point of our little digital adventure will be to send our valuable musical data to a digital audio processor or DAC (Marantz, Pro-Ject, Primare, Electrocompaniet, Esoteric, to name a few “our” names), which brings us to the last point of our blog this week: connectivity. The most important thing here is to know that, with the exception of the TosLink optical format, which by physical configuration accepts a maximum of data encoded at 24 bits / 96 kHz, all the digital inputs with which we will find accept what is currently the “standard ”In high resolution audio: encoding with 24-bit digital words and 192 kHz sampling rate (or its equivalent in monobit DSD encoding). Translation: quality identical to that handled by a recording studio and, therefore, the Holy Grail of musical reproduction. An old dream come true, to understand each other, although not everything is perfect: the quality of the sound recording continues to make a difference and when push comes to shove, the analog and digital elements of each processor / DAC will also provide their mark. But this is already a reason to dive into another blog, so, for now, we will leave it here. Let’s see now those digital connectivity options that must be taken into account: we will leave it here. Let’s see now those digital connectivity options that must be taken into account: we will leave it here. Let’s see now those digital connectivity options that must be taken into account:


  • S / PDIF coaxial format : This is a coaxial socket that can carry both a stereophonic signal (CD format, that is, PCM), and multi-channel (Dolby, DTS). A cable specifically designed for the transport of digital audio signals should be used if the best possible signal transfer is truly to be achieved. In other words, you would have to use a 75 ohm impedance cable. It uses a standard RCA connector, which in some products is accompanied by a BNC (bayonet) type as a consequence of its superior precision.
  • Optical TosLink : Optical TosLink ports accept the exact same digital audio data format as RCA coaxial jacks, ie S / PDIF, but use pulsed light rather than fluctuating electrical current to carry the data in question. TosLink connections use special fiber optic cables for support. The TosLink format is also known as EIAJ-TosLink, where EIAJ stands for the “Electronics Industries Association of Japan”; on the other hand, the prefix “Tos” comes from Toshiba, a Japanese firm that developed this connection system.
  • AES / EBU (XLR) : It is the format for the connection of digital audio signals adopted by the Audio Engineering Society (AES) and the European Broadcasting Union (EBU) and uses the same XLR jacks and plugs “for microphone” that are used on balanced analog audio connections. The AES / EBU connection (the characteristic impedance of the cable used must be 110 ohms if you want to fully exploit the possibilities of the format) is very common in professional audio and in some very high-level components for Hi-Fi and Cinema applications. Home.
  • USB (Universal Serial Bus): It is the latest connection format that has joined the digital “party” and the one that is having the most boom to be massively used for connection from computers and a growing number of mobile devices. Initially created for the transmission of “non-musical” data, the high-speed USB 2.0 Audio format has managed to place itself at the same level as the rest in terms of sound quality thanks to the development of high-performance asynchronous schemes (born for compensate for the original USB “computing” condition). Some products require the prior installation of a “driver” or controller on the computer that we are going to use, which is carried out without major problems. There are two types of USB connectors: A (“rectangular”) and B (“square”). In mobile devices we also find both options in their “mini” and “micro” versions. It is important, actually vital, to highlight the “asynchronous” nature of the USB sockets used for high-quality audio applications since it is an engineering strategy designed to compensate, as we indicated previously, the deficiencies of the USB format for this application. concrete.


  • IEEE1394 / FireWire / i.Link : It is an interconnection system that is based on the use of 4, 6.9 or 12 conductors of a serial bus that responds to a protocol invented by Apple and which is also called FireWire . In IEEE 1394, data (which can be both audio and video) is transmitted in packets at a speed of up to 400 Mbits per second. At present it is used, within the scope of High End audio, in the digital audio processors of the Japanese company Esoteric.


  • ES-LINK : It is an interconnection format for very wide passband signals (that is, very high resolution) used by the reference digital audio processors of the Japanese firm Esoteric. In its ES-LINK3 version, it supports PCM signals up to 48 bits / 176.4 kHz, as well as being compatible with the “Dual AES 8Fs” professional standard, which works with digital signals up to 24 bits / 352.8 kHz .
  • HDMI (High-Definition Multimedia Interface): More common in audiovisual components than in “pure” digital audio, HDMI is, however, used in some high-performance products for the transport of audio signals (both stereophonic and multi-channel) high resolution.

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