Audio formats

In the vast world of digital music, the quality and audio formats play a crucial role in delivering an immersive auditory experience. Whether you’re a casual listener or an audiophile, understanding different audio formats can enhance your appreciation for the nuances of sound. In this blog post, we’ll unravel the complexities of audio formats, shedding light on the most common ones and helping you make informed choices for your musical journey.


Firstly we have to say that in audio formats we have two main cathegories: uncompressed and compressed. The most popular uncompressed audio formats are WAV and AIFF. Both of them based on PCM (Pulse Code Modulation) mechanism which is using for storage audio files. WAV format was created by Microsoft and AIFF by Apple. These formats are high quality like from quality in studio and are very big. Below you can see how many possibilities you have when you render audio in one of the most popular DAW Ableton Live.

Firstly we have to choose right format of our audio file (file type). We can choose from WAV, AIFF and FLAC (we will write about this format later). Then we have to choose a bit depth: 16, 24 or 32. This value impacts on how many audio information do we have in one simple sample rate. And generally from this parameter depend noise and loudness of our music file. For one bit we have 6dB of dynamic range e.g. on CD we have usually 16 bit/44.1 kHz and it’s mean that maximum dynamic range of our music will be 96 dB. Remember that the threshold of pain for hearing is about 120 dB. Obviously you can listen 16 bit file with much more loudness but the level of noise can be not accteptable for you. That’s why in music production 24 bit is usually using, because it gives lower range of noise. In same cases producers also use 32bit which means really high quality audio file without any annoying noises. And now we have the last important parameter: sample rate. This parameter refers to density of downloading information per second. We can choose from 22.05 kHz to 192 kHz. When we use higher sample rate it’s mean that we more often rates our file like streaching audio wave on the graph.

In music world these uncompressed audio files are high quality, sounds great but are also very big. When CD was created it could be filled usually by about 10 tracks with 3-5 minutes lenght. There were a lot of problems with space on discs and drives. That’s why MP3 was created. The most popular compressed format in the world. Special algorithms compress the high quality files to low quality in assumption that the drop of quality can’t be audible by normal listener. Next to MP3 format Apple created AAC. Generally same algorithms and same low quality. Next to these two formats is also less popular OGG format using mainly in games or animations. These type of low quality audio files called lossy audio formats. Next to these compressed formats Microsoft and Apple created also lossless audio formats. They created special algorithms which should be no loss in sound quality. Main formats in that category are FLAC (Free Lossless Audio Codec) and ALAC (Apple Lossless Audio Codec). Generally quality is quite nice, much more better then MP3 and AAC, but not so good than WAV or AIFF.


After years of compressed and lossless formats artists and producers started to fight with low quality of their music. Sony decided to created new audio category (not format): Hi-Res. This definition refers to every compressed or uncompressed audio file which has better quality then CD (16 bit/44.1 kHz). It can be WAV, AIFF, FLAC, ALAC etc. The main difference of Hi-Res audio to no Hi-Res audio is method of coding analog signal to digital. Generally in most popular producers programms (DAWs) like ProTolls, Ableton Live, Fl Studio, Reaper etc. common method of coding is PCM. Pulse coding modulation was created to save music on CDs and another devices. This system shares files for bits and samples this fragments in time. Hi-Res format uses DSD (Direct Stream Digital). In this system there is only 1 bit and sample rate is changed. It depends on analog signal and helps in sampling difficult fragments of a track. It is much more effective and generally reflects better analog curve. DSD was first time used in SACD discs and had 2,8224 MHz sample rate. That is 64 times higher then CD so this lossless compression is called DSD64. There are also higher standards like DSD128, DSD256 and DSD512.

DSD VS PCM – which is better?

All music world today based on PCM coding. Actually coding tracks in DSD is difficult and there’s no so many DAWs which can do this (but it is possible). DSD coding is using mainly to lossless compresion and give us a posibility to listen music on the highest level like in studios. Please check this company which provide an audio converter: I created high quality audio in my DAW – WAV 32 bit and 192 kHz. It was 8 sec. long and sized 11.7 MB. After compression using this converter I got DSD 1 bit and 2.8 MHz. Size of this file was 5 MB. Generally the quality of these two files was similar. I encorage you make the same test and you will better understand how lossless compresson is working.

At the end of this post I would like you to watch a great video about audio formats which will be a summary about this article:

Thank’s a lot for your attention and best regards!

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