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FLAC [Free Lossless Audio Codec] The difference between FLAC and mp3

FLAC (Free Lossless Audio Codec) is a lossless method for compressing the size of digital music information. Read about comparing mp3 and FLAC and then open it.
The file with the * .flac extension can store music compressed using the free lossless audio codec, as well as high-resolution DSD DoP or MQA formats.

“Lossless” means that “the original and restored digital audio material is completely identical.”

Example:

If the sequence “1234” is compressed in size in a certain sequence (for example, “97”), after unpacking the latter we have again “1234”.

Read on to learn more about audio quality, conversion, playback, and more.

Features> Converters> FLAC sound quality problems> How to play FLAC (software players)> Download FLAC files (examples)> Links> FLAC and iTunes> FLAC or mp3>

In addition to Free Lossless Audio Codec software decoders, there are hardware decoders built into portable audio players (DAPs). Although in reality a DAP or a mobile phone is a small computer that runs a program to encode or decode a music format.

A single large * .flac file can be a music album container along with a CUE index file containing the start time points of each track.

The free lossless audio codec is maintained by the Xiph.Org Foundation.

The open source software libraries at the time of writing are available for Windows, Unix (Linux, * BSD, Solaris, Mac OS X, IRIX), BeOS, OS / 2, Amiga operating systems.

FLAC features

Sampling frequency: up to 384 kHz
Bit resolution: up to 32 bits
Compression type: lossless
Audio compression: compressed / uncompressed (uncompressed, non-standard)
FLAC file size compression ratio / compression ratio: 9 levels
(As a first approximation, the compressed FLAC size is approximately 60% of the original WAV size or more)
Using the FLAC file as a container for other formats – yes
Preservation of text metadata: yes
Save artwork as album art: yes
Save multiple images – yes
Program code: open source

FLAC sound quality problems

Compressed vs uncompressed
One of the most popular and never ending discussions is “FLAC vs WAV Sound Quality”. There is an opinion that WAV sounds better than FLAC.

However, WAV has compatibility issues with displaying metadata with some software. Therefore, uncompressed FLAC was implemented, which supports metadata as standard.

FLAC and WAV contain the exact same digital material. Therefore, FLAC and WAV should sound exactly the same.

FLAC in your hands!

No matter how malicious the copyright holders are, we won’t stop copying AudioCD, if only because we don’t want to wear down the licensed disc again. It’s more expensive to store music on analog media – vinyl and magnetic tapes deteriorate fairly quickly, and CDs tend to get scratched or even broken on faulty drives. What has already been copied (digitized), in most cases, are popular MP3 files (less often OGG or even less often WMA). At the same time, most citizens do not think about the very real mockery of sound that occurs when converting music into compressed formats. The best option is to keep your favorite works unchanged.

If we use a CD ripper and tell it to rip audio tracks to WAV files, we end up with what was originally burned on the audio CD (of course, we don’t take disc read errors into account). But the size! Even now, when the cost of 1TB hard drives has dropped below $ 100, it cannot be forgotten that the volume of “captured” WAV files from an audio CD reaches 700MB. Now let’s go back to the days when 10GB hard drives were considered a luxury and understand the motivation behind compressed formats. But at what cost did they achieve such impressive results? And where is the line that is best not to cross to maintain maximum sound quality? We will talk about this. But first, a little theory …

It is usual to subdivide the coding algorithms into two groups: with and without loss of information. The advantage of the latter is that the signal will be extracted in the same way as before compression.

“Digital” vs. “analog” recordings are not subject to the changes that analog audio media undergoes. Tracks are stored in binary form and transmitted in the form of electrical pulses that have only two values, “1” or “0”: the signal is present or absent. If interference occurs, even though it may affect the signal, digital circuitry can still detect whether it is present (“1”) or not (“0”). This is the main advantage of digital technology over analog technology: “Digital technology allows you to store and replicate records for an unlimited time without loss of quality.

Digitizing the sound supplied to the line input of a sound card is done in three stages. In the first stage (sampling), the signal is divided into many millions of elements, the number of which depends on the sampling frequency, which is measured in kilohertz (kHz). For example, a sample rate of 48 kHz means that every second of the analog signal has been divided into 48,000 samples. The higher the sampling frequency, the more elements the original analog audio signal will be divided, the more accurate the conversion, and the higher the quality of the digitized audio.

In the second stage (quantization), each element of the sampled signal is assigned a certain numerical value corresponding to its amplitude. This number can vary within certain limits, for example, from 0 to 16,535. With such quantization, 16,535 signal levels are possible (this type of quantization is called 16 bits or 16 bits (16,535 = 216). The resulting quantization is not binary, but decimal. Each decimal number is stored in computer memory in binary form: for example, the number “1” as “00000001” and “2” as “00000010”. The digitization of sound is called “Pulse Code Modulation” (PCM), since the audio signal is represented as a series of pulses of constant frequency, the amplitude of which is encoded in decimal numbers, that is, in the form of digital.

Finally, the third stage is encoding (compression). During it, the resulting sequence of numbers is archived according to a specific algorithm. The most popular digital audio compression format today is the one developed by the Institute. Comrade Fraunhofer IIS, www.iis.fraunhofer.de/amm) MPEG Layer3, or more commonly MP3. With high-quality encoding of individual blocks (bit rate up to 320 Kbps), only mathematical compression algorithms are used. At the same time, the quality does not suffer, but the file size is reduced by only four times, that is, we have a compression ratio like what a normal filing cabinet would give.