lossless sound quality

lossless sound quality

lossless sound quality
lossless sound quality

The principle of these audio compression methods is to keep as much as possible the parts that are easy to hear for the human ear.

lossless sound quality
lossless sound quality

 

In the main constant bit rate (CBR) mode, audible data in the audio is removed and it is easier for the human ear to hear. The more information that is retained for the part, the less information that is retained for the less audible part. As sound complexity increases, the degree of sound quality degradation becomes more apparent. For example, when listening to pure human music voice performances and performances combined with background music, pure The degree of degradation of the compressed music of the human voice is less obvious than that of the original file.

As for the lossless sound quality provided by today’s streaming platform, there are many formats, but strictly speaking, these formats, including flac and ALAC, belong to lossless compression (non-destructive data compression) rather than quality. lossless sound in a narrow sense, but due to its performance it is almost the same as WAVE, but the file is small, the decoding speed is fast, and it can be used in streaming services, so it is also a representative format of lossless sound quality that has entered the era without CD support, and it is also the current music streaming service.

Also, MQA like Tidal is a more controversial format, because although MQA is lossless audio compression in a broad sense, it focuses on driving high-quality music files that are above CD quality at a size similar to those of CD quality flac files. , but it can also be restored to the original high-quality audio format, but the controversial point is whether the higher compression ratio can maintain the same real auditory sense as the original high-quality audio has always been controversial. At the same time, MQA requires a specific device to perform hardware decoding. Yes, many devices can only decode MQA through software.

▲Currently called Hi-Res Audio generally refers to sound quality higher than that of a CD, but to be certified, the device must support the processing capability of the 24-bit 96kHz format.

However, since MP3 and AAC are previously planned destructive music compression formats, their purpose is to compress CD-level sound quality into smaller files, so most encoding specifications are presented in one presentation. CD-level 16-bit 44.1 kHz or 16-bit 48 kHz; but MQA is a format with a small file size that locks in high sound quality requirements. Basically, the MQA format will be music equivalent to 24-bit PCM or higher.

lossless sound quality

lossless sound quality

lossless sound quality
lossless sound quality

Although lossless sound quality and high-quality streaming services have appeared on the market for a while

lossless sound quality
lossless sound quality

Like Apple isn’t the first to make true wireless earbuds, but it has ridden the wave of the market, with Apple’s launch of so-called hi-fi. compressed sound quality in Apple Music. In fact, after the lossless compression commonly known in the audio industry, lossless sound quality and high sound quality have instantly become a hot topic for many people to discuss, but whether the benefits of quality Lossless sound systems are really Obviously, the author will devote some space from theoretical concepts to practical levels. The difference is a simple overview, but the following arguments focus on easy understanding, so some of the descriptions are not entirely accurate.

What is lossless sound quality? What is the sound quality of the music we usually listen to?
The photo mentions * Introduction to KKBOX sound quality (including format), release year, sound quality, including numbers, KKBox, streaming media, Hi-Res Audio and Hi-Fi

▲ Both AAC and MP3 are destructive compression, while FLAC and ALAC are lossless compression

Before we get into the details, let’s talk about what is called lossless sound quality; From the author’s concept and cognition, the so-called lossless sound quality is a relative word rather than an absolute word. The main source is the relative word born due to the appearance of the MP3 music compression format; after the music industry went digital, digital music files were generally recorded via PCM encoding, and WAVE files were generally used on CDs. The time and space situation that MP3 was born into was because hard drive capacity was generally small at the time. To store a large amount of music data in storage space, you probably need to compress the files.

The reason why MP3 cannot be called a lossless format is because it essentially compresses the original WAVE in a way that limits the bitrate by combining concepts like the psychology of sound. The bitrate compression mode, compared to the bitrate of the original WAVE file, compresses at least in a range of 1:4 to 12:1. The higher the bitrate, the greater the amount of retained information. For example, AAC format also adopts similar audio psychology for compression, but the efficiency is better than MP3. Therefore, most of the lossless sound quality of most streaming platforms currently uses AAC as the mainstream.

Detailed music format

Detailed music format

Audio File Formats
Audio File Formats

classic wave

Audio File Formats
Audio File Formats

As the most classic Windows media audio format, the WAVE file is widely used, which uses three parameters to represent sound: the number of sampled bits, the sample rate, and the number of channels.
The channels are divided into mono and stereo, and the sample rates are generally 11025 Hz (11 kHz), 22050 Hz (22 kHz), and 44100 Hz (44 kHz). The capacity occupied by the WAVE file = (sampling frequency × sampling bits × channel) × time/8 (1 byte = 8 bits).

traditional mod

MOD is a wavetable-like music format, but its structure is similar to MIDI, it uses real samples, and the volume is small. In the earlier DOS era, MOD was often used as background music for games. Modern mods can contain many audio tracks in many formats, such as S3M, NST, 669, MTM, XM, IT, XT, and RT.

midi music computer

MIDI is short for Musical Instrument Data Interface. Records the sound played by the instrument digitally (each note is recorded as a number), and then synthesizes these records via FM or wavetable during playback: FM synthesis is the sound of the instrument is simulated by mixing the multi-frequency sounds; wavetable synthesis consists of storing the sound samples of the instrument in the wavetable of the sound card and extracting the sound from the wavetable as you play.

Boss Boss MP3

It can be said that MP3 is famous, it uses MPEG Audio Layer 3 technology to compress the sound with a compression ratio of 1:10 or even 1:12, with a sampling rate of 44kHz and a bit rate of 112kbit/s. .
MP3 music is music stored in digital form. If you want to play it, you must have a corresponding digital playback and decoding system. Generally, MP3 digital music is decoded by special software and then restored to a waveform sound signal for playback output. This type of software is called For MP3 players, such as Winamp, etc.

Overlord RA series online

RA, RAM, and RM are Real’s mature network audio formats, using “streaming audio” technology, making them well suited for network streaming. Information such as copyright, singer, producer, mail and song title can be added during production.
RA can be called the supreme lord of multimedia communication on the Internet. It is suitable for streaming on the Internet and is currently the best format for listening to online music online.

VQF with high compression ratio

VQF or TwinVQ is an audio compression technology developed by Nippon Telegraph and Telephone and Yamaha Corporation.
The audio compression rate of VQF is almost twice that of standard MPEG audio and can reach approximately 1:18 or even higher. And popular compression formats like MP3 and RA are usually only around 1:12. But it still won’t affect the sound quality, when VQF compress music at 44kHz-80kbit/s audio sampling rate, its sound quality will be better than 44kHz-128kbit/s MP3, when compress at 44kHz-96kbit/s , the music is close to 44kHz-256kbit/s MP3.

MD minidisc

MD (ie MiniDisc) is a comprehensive portable music format released by SONY in 1992. The compression algorithm it uses is ATRAC technology (the compression ratio is 1:5). MD is divided into Recordable MD (Recordable, with two heads of magnetic head and laser head) and Single Play MD (Prerecorded, only laser head).
The powerful editing function is the strong point of MD. You can quickly select tracks, move tracks, merge, split, delete and edit track titles. It is more personalized than CD and you can have your own MD album at any time. MD products include MD Walkman, MD bedside audio, MD car audio, MD recording deck, MD camera gun and MD driver, etc.

Lossless audio formats

Lossless audio formats

Lossless audio

Audio files can be converted in a more or less space saving way using lossless or lossy codecs. While the data reduction is absolutely true to the original with lossless codecs, a difference can be heard from the original material, especially with heavy compression with lossy codecs.

Lossless compressed audio files lose approximately 25% to 50% of their original file size. Typical representatives are Monkey’s Audio, FLAC, or WavPack. Modern lossy codecs like AAC, Ogg Vorbis, or MP3, on the other hand, reduce source material by 90% without sounding noticeably worse.

Lossless audio compression

Unlike documents or images, for example, audio files are very difficult to compress, since identical repetitions are very rare in music. But it is precisely on such occurrences that the Huffmann or Lempel-Ziv algorithms depend, which are used in ZIP or RAR, for example. Therefore, most non-clustered compressors employ predictive coding. The signal is divided into a music component and a noise component. Depending on how well this prediction has been made, the noise component should ideally be Gaussian white noise that can be easily compressed using conventional methods.
In the ideal case, the result is 50% compression, which is highly dependent on the piece of music.

Monkey’s Audio, FLAC and WavPack are popular formats for lossless compression of WAV (PCM) files, for example digitized original recordings from the recording studio (for archiving or later processing) or archiving copies of music CDs. The sound quality is always the same as the original and the checksums reveal corrupted files. Increasing hard drive capacity at affordable prices makes “Lossless Codecs” interesting for everyday use.

Uncompressed: WAV (PCM)

WAV is the largest common denominator of Windows audio formats. The Macintosh equivalent is called AIFF. WAV is actually the collective term for various subformats, of which PCM is the most common and is generally equated with WAV.

WAV (PCM) is an uncompressed recording of sound samples: the time signal of a noise is sampled, quantized, digitized, and saved at discrete points in time. The more often and finer you record these values, the better the sound. With CD quality music, this instantaneous value is recorded 44,100 times per second and recorded with 16-bit “precision”, that is, 2 ^ 16 = 65536 possible values.

With the help of special programs (eg CDex, EAC, Audiograbber), Audio CDs can be transferred to the hard disk as WAV (PCM) files. Viewed in this way, WAV (PCM) files are copies of the original, provided the CD-ROM drive is not read incorrectly or the CD is damaged. One minute at CD quality requires roughly 10MB of storage space, which is not as happy to give away even in the age of ever-larger hard drives.

On the PC, WAV (PCM) and CD-quality (44.1 kHz, 16-bit, stereo) audio files are often the starting material for creating space-saving audio files in formats such as MP3. However, for sound processing on the home PC, WAV (PCM) is the first choice.

Monkey’s Audio

Monkey’s Audio is a lossless audio codec for PCM wave files. Monkey’s Audio comes with a convenient program interface (in English), over which files can be compressed, decompressed, verified or tagged. PCM wave files (any sample rate, 8/16/24 bit, mono or stereo) or corresponding Shorten or WavPack files are accepted as source files. Monkey’s Audio does not support multiple channels.

The APE tags that are used to store the title information can be supplemented with their own fields and are therefore very flexible. They are now used in conjunction with other audio formats as well.

The included command line encoder allows integration into other programs. Various audio players support the format through plugins. A plug-in for Winamp can be installed at the same time as installation. Monkey’s Audio is a Windows program by default. However, there is a platform independent version of Java.

FLAC

FLAC stands for “Free Lossless Audio Codec”. There are several lossless audio codecs available. FLAC is suitable here for several reasons: FLAC follows the open source philosophy (free open source code, available for many operating systems), works very fast, has a good and secure framework structure.

FLAC, WAV, MP3, DSD, ALAC … What audio format should I use?

You probably know the famous MP3 audio format. There’s even a good chance that you only use it on a daily basis. But did you know that it is possible to take your music to the next level thanks to other audio formats? If the terms FLAC, DSD, sample rate, or even lossless don’t mean anything to you, then you’ve come to the right place. Designed specifically for newbies, this guide tells you everything you need to know about the basics of digital audio.

soundwave

FLAC, DSD, ALAC … Listening to a debate between audiophiles can seem difficult when you do not know this universe and the many acronyms that refer to it. But if you try the adventure, you will not regret it. Say goodbye to your boring and lifeless MP3s and hello to quality music. Trust us, your ears will thank you!

Sample Rate and Bit Depth: The Basics of Digital Music

Before knocking you out (we promise we won’t hit too hard) with barbaric acronyms in every way, let’s first focus on two essential notions of modern audio, namely sampling rate and bit depth. These two elements give an idea of ​​the recording precision of a song.

but depth

As you know, computers run on bits, which are sets of 0 and 1. During a passage in the studio, music produced by an artist must be digitized, therefore transformed into 0 and 1 in order to be recorded on CD or transmitted to through transmission services. This is where the sampling rate and bit depth come into play.

Take the example of a CD. Our beloved empanadas are recorded in 16-bit / 44.1 kHz. The 44.1 kHz sampling rate means that the music produced by our musician is analyzed 44,100 times per second by studio recording devices. As for the bit depth, it gives an indication of the number of information recorded during this same period. The greater the depth, the more information will be encoded at the end.

However, CD quality is not the best in the world, even if it far exceeds MP3. Thus, we find 24-bit / 192 kHz recordings. The DSD goes even further with a frequency that rises to several MHz. But for simplicity, just remember that the higher the values ​​described above, the more accurate the recording will be in your sound reproduction.

Lossy formats: MP3, AAC, OGG

In general, there are two types of formats in the audio world: lossy, lossy in English, and lossy, or lossless. If you want the best audio quality, stay away from compressed formats.

The best known of all is MP3. True dinosaur in the audio world, this type of file was developed at a time when the capacities of our hard drives were determined in MB and not in TB. Therefore, we had to compress the recordings as much as possible, even if that meant putting quality aside.

It is true that MP3 encoded music weighs only a few megabytes. But the applied algorithm is very aggressive, it simply cuts the frequencies considered inaudible by the human ear. In fact, MP3 loses many audible parts. To get an idea, click the link below, you will hear these famous truncated parts. The pieces seem flat, devoid of life. Listening can even become unpleasant after several tens of minutes. Suffice it to say that, apart from its small size, MP3 is no longer really interesting in our time if we are looking for quality music.

To make things better, Apple, meanwhile, released another audio format, AAC, for advanced audio encoding. This is also a lossy format which therefore loses details during data compression. However, the algorithm used is more efficient, cutting fewer important frequencies, at least on paper. In absolute terms, the difference from MP3 is not necessarily stark and the debate has been raging for years in the audiophile environment to find out if the AAC format is really better than MP3.

Finally, there is also the OGG Vorbis, another lossy compressed format. Like AAC, it is supposed to work better than MP3. This is the type of file Spotify uses. Her interest is to enable efficient transmission while reducing quality. However, the songs encoded in this format are not fabulous. The ideal is really to become lossless.

Lossless audio formats 

WAV (.wav) The WAV format is nothing more than a digital recording of real sounds, sounds that come from a source outside the PC. With WAV music, drums, piano, guitar, bass or vocals are heard the same, no matter what computer the file is played on (with the same acoustic quality of hardware components, of course).

wav

DSD (DFF, .Dsf) are used in digital media such as Super Audio CDs. Sampling quality is very high (variable sampling rate is approximately 64 times higher than for audio CDs), although according to several operators in this sector, it must be absolutely determined whether the final quality is higher than the sampling quality. PCM (used in high quality Blu-Ray and DVD Audio). I can reproduce free readers like Foobar 2000 or AIMP on our computer.

AIFF (.aif). It’s the Apple audio format used by Apple for Mac. It’s basically the WAV equivalent used by Windows.

APE (Mono Audio; .ape): Ordinary with a loss that allows us to reduce by about 50% the space occupied by our music (in some cases even more), without losing quality. In this way, an album that takes up about 600 MB in wav format averages 300 MB (much more than about 100 MB of high-speed mpc and 60 mp3 mp3, but the quality is the same as the original); On average, I speak because there are certain types of music where the level of compression is even higher. You can use WinAmp plug-ins to listen to songs in this format, or better yet, a player that uses it natively as Foobar 2000. Right now, it’s probably the best lossless codec, considering the balance between speed and compression (Click here for a lossless comparison table) format.)

Apple Lossless Audio Codec – ALAC (.m4a) This is a lossless audio codec created by Apple a few years ago and available as an open source from Apple since October 2011. Some programs support it in Windows (encoded). , including DbPowerAmp.

FLAC (.flac): a very popular open source format. It used to be very popular (among music purists), but some space has been lost with the MonkeyAudio bee format, which allows for better compression in the same quality. Compared to others, it always came in .ape format (but also after WavPack).

 

LA (Lossless Audio: .la): The lossless format that compresses the most. Conversion to this format is very slow, but achieves the best compression in history (for example, MonkeyAudio, this album would take up about 290 MB compared to 300 MB in APE). This is not widespread, as some MB obtained on a bee is not worth the long wait (and less support from third-party programs), but it is the winner of my lossless format comparison.flac

OPTIM FROG (.ofr) – Excellent lossless codec, best after LA as compressibility. Compression, when set to slightly high levels, is about 3 times slower than Monkey Audio and the gain in MB is about 2%. Like all other formats on this site, you can listen without the addition of Foobar 2000.

WAVPACK (.wv, wvc) – is a valid open source compression format that allows lossless, lossy and hybrid results. The compression rate is high and the compression is good. Another interesting hybrid format is that the lossy file is merged with another file, which allows you to completely restore the original file without loss in case we want to restore it.

Audio formats

Audio formats

Compression

Compressions are systems for reducing the file size by using different types of algorithms and / or encodings.

compressed audio

There are two types of compression: lossless (compression), which compresses the file without deleting information. Decompression can therefore exactly return the original and lossy (lossy) compression, eliminating redundant parts that are considered irrelevant or irrelevant and the decompression does not return to the original.
It is clear that the first system preserves the integrity of the original, but less compressed, while the second implies a loss of quality, but compresses much more, in proportion to the degree of loss one is willing to accept. Let’s look at a few examples.

Lossless compression

Lossless compression is based on reducing the redundancy typical of human production.

human perception
For example, in a book dedicated to experimental music, the phrase “experimental music” is repeated many times with 19 characters. At this point, simply replace it with a symbol that is normally not included in the text, e.g. ‘# 1 #’ to reduce a term from 19 characters to one of 3 and store 16 characters for each occurrence. Actually we have to say “for every occurrence after the first”, because in order to unpack the text, we also have to create an index of the substitutions in which it is written in this case
# 1 # = “experimental music”.
Obviously, many other words or phrases are repeated several times in the book, and each of them can be replaced by a symbol such as # 2 #, # 3 #, …, # n #, where n is a progressive number, which ultimately makes significant savings.
The Lempel-Ziv (LZ) algorithm uses a similar system, the derivatives of which underlie many modern lossless compression programs, including the well-known ZIP.
In fact, the ancestor of many lossless encoders is the so-called Huffman coding. It is a redundancy elimination system that was developed in 1952 by the researcher of the same name, then an MIT student. His algorithm solves the problem of encoding a series of strings (string = any character set) as compactly as possible, taking into account the frequency with which strings occur: the most common is assigned the shortest symbol in to maximize compression. Here is a good example dealing with Huffman coding issues.

Another type of lossless compression, which is always based on reducing redundancy, is the so-called Run Length Encoding (RLE), which works in a very simple way. Suppose we have the following string of 20 characters
ABBBBBBBBBCDEEEEFGGG

By applying the RLE it will
A 9BCD * * * 4EF 3G

for a total of 13 characters with a saving of 35%.
In practice, a code consisting of the character and the number of repetitions was inserted instead of the repeated characters. The asterisk indicates that the following is the number of repetitions and is not part of the chain (this is of course the basic principle; the details of the coding may vary).
Of course, this system is not productive with text, but it is the case with images where long stripes of the same color are fairly common.

Lossy compression

Lossy compression is based on the elimination of the information components that are considered to be more or less irrelevant depending on the compression level required. At low compression levels, only the really irrelevant details are removed, while at higher levels, the sensitive details are also removed.
An example that is not audio is the encoding of JPEG images, in which nuances are eliminated by assigning neighboring pixel groups the same color if their difference is less than a value that is proportional to the degree of compression. On this page you can see the effect of the size reduction and the corresponding loss of quality when increasing the compression levels.

Further information on compression on Wikipedia (free, community-created encyclopedia) can be found here in English. Wikipedia also exists in Italian, but the content is smaller.
First class compressed audio formats
Lossless (lossless)
These formats work similarly to zip. You compress the content without removing anything. At the time of listening, it is necessary to perform a decompression and to return to the original in one of the linear formats already shown.
Since it is lossless compression, the comparison between these codecs is not made in