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10 Common Audio Formats: Which Format Should I Choose?

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10 Common Audio Formats: Which Format Should I Choose?

Audio files come in various formats and sizes, in addition to our common MP3, there are AAC, FLAC, OGG, WMA, etc.

Why are there so many different audio standards? What format is the best? What should you choose when saving audio files?

In fact, all audio formats can be divided into 3 categories, and once you understand the characteristics of each of these 3 types, you can choose one of the three, and then choose a further subdivided category according to your specific needs.

Comparison of audio formats – operation, equipment – Introduction to 10 common audio formats: which format should I choose?
uncompressed audio format
The uncompressed format preserves the original audio waveform. The so-called original audio waveform refers to the result after the sound is captured and converted to a digital signal for storage. Raw audio waveforms are stored without any post processing. From the results, such an audio file will consume a lot of space. For 24-bit (24-bit) 96 KHz audio, you need about 34 MB of storage space per minute.

Uncompressed audio format: PCM
PCM is short for Pulse-Code Modulation, which is a technical method of simulating sound using digital signals. The sounds that we can hear in nature exist in the form of physical sound waves, and in order to simulate this sound wave with a digital signal, scientists decided to sample it and record it with another wave, which is what we call a pulse.

Therefore, digital audio has two basic concepts: sample rate and bit depth. The so-called sampling rate refers to the speed of how many times the sound is sampled. The so-called bit depth refers to the space occupied by a single sample. In uncompressed formats, the product of the sample rate, bit depth, and audio duration is the space occupied by the digital signal that represents the audio content.

PCM is a format commonly used on CDs and DVDs, but its derivative LPCM is often used in practice. LPCM is short for Linear PCM, which is Linear Pulse Code Modulation. Most PCM files in use today are actually LPCM files, so there is no need to differentiate between them.

Uncompressed audio format: WAV
WAV is short for Waveform Audio File Format, which is a standard audio format developed by Microsoft and IBM in 1991.

Many people think that WAV is an uncompressed audio format, but this is not entirely true. In fact, WAV is a “wrapper” type format, a common format developed by Microsoft to use audio for various platforms. So it is possible for WAV to contain compressed audio, but this is rarely the format of choice when using compression these days.

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All the Audio Format Differences: Which One Should You Use?

Three classifications of audio formats

Characteristics and differences of different audio formats

Which audio format is right for you?

You already know MP3, but what about AAC, FLAC, OGG or WMA? Why are there so many audio file formats and is there any best audio format?

Comparison of audio formats

Audio files come in various types and sizes. While we’re all probably familiar with MP3, what about AAC, FLAC, OGG, or WMA? Why are there so many audio standards? Is there a better audio format? Which ones are important and which ones can be ignored?

All audio formats are divided into three main categories, and once you know what the categories mean, you can choose the format within the categories that best suits your needs.

uncompressed audio format

Uncompressed audio consists of actual sound waves that have been captured and converted to digital format without any additional processing. So uncompressed audio files tend to be the most accurate, but take up a lot of disk space: about 34MB per minute for 24-bit 96KHz stereo.

Audio file format: PCM

PCM stands for Pulse Code Modulation, a digital representation of the original analog audio signal. Analog sounds exist as waveforms. To convert a waveform into digital bits, the sound must be sampled and recorded at specific intervals (or pulses).

This digital audio format has a “sample rate” (how often the samples are made) and a “bit depth” (how many bits are used to represent each sample). There is no compression involved. Digital recordings are almost accurate representations of analog sounds.

Audio compression, how it works

audio compression

It must have a corresponding inverse transform, called decompression or decoding. The audio signal can introduce a lot of noise and some distortion after passing through a codec system

Audio compression technology refers to the application of suitable digital signal processing technology to the original digital audio signal stream (PCM encoding), without losing the amount of useful information, or under the condition that the loss introduced insignificant, reducing (compressing) its code rate, and also called compression encoding. It must have a corresponding inverse transform, called decompression or decoding. Audio signals can introduce a great deal of noise and some distortion after passing through a codec system. The advantages of digital signal are obvious, but it also has its own corresponding disadvantages, ie increased storage capacity requirements and increased channel capacity requirements during transmission. Taking a CD as an example, the sampling frequency is 44.1KHz and the quantization precision is 16 bits, so a stereo audio signal for 1 minute needs to occupy about 10M bytes of storage capacity, that is, the capacity of a CD turntable is only about 1 hour. Of course, the problem is even more pronounced in the world of much higher bandwidth digital video. Are all these bits necessary? The study found that there is a large redundancy in the direct use of the PCM code stream for storage and transmission. In fact, sound can be compressed at least 4:1 under lossless conditions, that is, only 25% of the digital amount is used to retain all the information, and the compression ratio in the video field can even reach to several hundred times. Therefore, in order to use limited resources, compression technology has received much attention since its inception. The research and application of audio compression technology has a long history, like A-law coding, u-law is a simple almost instant compression technology, and has been applied in ISDN voice transmission. Research on speech signals has been developed before and has matured, and has been widely used, such as adaptive differential PCM (ADPCM), linear predictive coding (LPC), and other technologies.

Audio compression for music lovers

Audio compression for music lovers

the truth about high bitrate lossy compression

In the opinion of most people, the word music lover is most often associated with a person who not only loves and collects music, but also appreciates high-quality music, and not only in artistic and aesthetic terms, but also the quality of the recording of the phonogram itself. Just think, a few years ago, an audio CD was considered the standard for music quality, whereas a computer, even in dreams, could not compete with the quality of a CD. However, time is a great joker, and he often likes to turn things upside down. It would seem that quite a while, a year or two passed and … that’s it, the CD on the PC went into the background. Don’t ask “why?”, You know the answer to this question yourself. Everything is to blame for the revolution in the world of computer sound: audio compression (hereinafter referred to as audiolo compression which means lossy compression to reduce the size of the audio file), which made it possible to store music on disk hard, lots of music! In addition, it was possible to exchange it over the Internet. New sound cards have been released, capable of almost “squeezing” studio quality out of a piece of hardware that seems useless in terms of music. Today, even having a computer that is not very smart in performance, having bought a Creative SoundBlaster Live! and remembering that since Soviet times there is a good amplifier and good acoustics, you will get nothing but a high-quality music center, the sound of which is inferior only to very expensive audio equipment (average or even the highest Hi-Fi category ). Add to this the general availability of music files and you understand that you have the power in your hands. And then there is a revolution, and you understand that a compact disc is no longer so convenient, you are fascinated by something completely different: the magic “MP3” signs. You cannot eat or sleep; you are faced with the seemingly insoluble “chicken and egg” question: how to “squeeze” and, most importantly, how to “squeeze” …

This is where I will help you. This article is the beginning of my new series of informational materials on music on the computer. For over a year developing OrlSoft MPeg eXtension and maintaining an extensive database of MP3 files, I have accumulated a great deal of research on audio compression. It is these studies that I will try to share with you. Many articles have been written on audio compression by different respected authors, so I will try not to write what I can easily find in other sources of information. I would like to put my position on the subject we are considering simply and clearly. We will not consider audio compression to be as compact a tool as possible put audio information on your hard drive (so that you can record so many hours of music there). Yes, compression allows you to record music more compactly, but my goal is to minimize quality loss by converting “pure” audio to compressed audio. This is why only high bit rates and qualitatively compressing encoders are considered in these modes. So it is much more convenient to work with compressed audio – instant access to any track from any album, convenient software for playback. And, of course, the financial issue has not been forgotten either.

Of the audio compression formats that exist today, in my opinion, three deserve attention: MP3 (or MPEG-1 Audio Layer III), LQT (as representative of the MPEG-2 AAC / MPEG-4 family) and a Completely new OGG format (Ogg Vorbis) developed by a group of enthusiasts:

MP3 is by far the most used of these (mainly because it is free). Let me remind you that it was thanks to the MP3 format that the victorious procession of compressed audio took place. However, as often happens with pioneers, little by little it is losing ground and giving way to new and better formats.
The second format, LQT, is a representative of a new direction of audio coding algorithms, a representative of the AAC family. This is a fairly high quality, but commercial and highly classified format.
OGG became widely known to the public this summer and is currently developing rapidly, soon (with the launch of the Encoder and Decoder) it should beat MP3 with better sound quality with smaller file size.

Does MP3 affect the sound quality?

The compression of songs affects the quality, but the losses are not necessarily audible.

Is compression of MP3 songs harmful to the sound quality? Whether it is HD music or “normal” definition, the question of compression remains. The advantage is that the weight of the songs is reduced, so they take up less space in the memory of a phone or a portable music player. With standard MP3 compression, a music album ranges from 500 MB to 45 MB.

But by the way, the music is damaged. The sound seems a little less natural, less precise, less dynamic. Some of the audio information is literally destroyed. It doesn’t always sound good, but for some songs the difference is clear until everyone will notice.

Fortunately, you can improve the quality of an MP3 song by compressing it with less force. The loss of sound quality becomes less clear, but in return the song weighs more. MP3 isn’t the only compressed music format that corrupts music. The most famous competitors are AAC, Ogg Vorbis and WMA. MP3 is not the most efficient compression format, this title applies to the Ogg Vorbis, but it is still a good option. All music players can play MP3 and online record stores prefer this format.

Lossless compression

However, some music lovers are reluctant to MP3. They swear by “nondestructive” compression, which does not remove sound information. The music has been completely preserved: we hear absolutely no difference. The best known non-destructive formats are Flac, APE and Alac. Unfortunately, not all electronic devices can play music recorded in these formats. Few artists offer their music in “non-destructive” compression. And the weight of the parts thus compressed is still very heavy. An album quickly reaches several hundred megabytes. However, the Flac stands out as the reference format for the most demanding music lovers.

Is it reasonable to keep using MP3? This remains a smart choice for most music lovers, as long as they choose an appropriate compression ratio. Which one to choose: 192 kbit / s, 256 kbit / s or 320 kbit / s? The stronger the compression, the lighter the number, but the lower the quality. With 128 kbit / s, the sound has clearly deteriorated, most of us can hear it. At 192 kbit / s, degradation becomes difficult for most of us to observe except for some rare numbers.

With 256 kbit / s, you have to have a musical ear and good sound equipment to make the difference. With 320 kbit / s, you need a well-trained ear and highly accurate audio equipment to make a difference. We only see a difference in quality in certain titles and only in certain passages. Therefore, most of us can settle for 192 kbit / s recording. Music lovers should expect a minimum of 256 kbit / s. And professionals will choose formats of 320 kbit / s or ‘lossless’.

Do you know what is the best audio format? (for musicians and listeners)

Maybe you are a musician who has had the hard work of learning sound theory, writing and arranging your music, practicing, recording, and now you want to show your work to the world. What is the best audio format to distribute your recording? Or maybe you are a music lover looking for the best way to organize your collection. It may be that your collection comes from CDs or vinyl or even digitally downloaded. What is the best audio format to store your collection?

Choosing the right file format is important. On the one hand, the selected format affects the fidelity of the sound. However, the selected format also limits which devices can play your files. In this guide, we explain the different audio formats available, what they offer and how they are used in the best possible way. First, let’s understand what digital audio files are.

In short, what is the best audio format for you?
To choose the best audio file format you need, think about the purpose of each file. Here are some common uses with suggestions for the best audio formats for each:

If you are an artist …

Do you record a CD with your work? WAV! Conventional CDs use WAV, so export your music from DAW to Wav (or other lossless and then convert properly) and use these files to burn to CD. If you compress CDs from a factory, export in wav but in the original sampling rate and bit depth settings. Leave these settings to them.
Do you share your music with the public? Lossy! You want a smaller file size and maximum compatibility with them from your listeners. Any device plays an MP3 file and is easy to share online.
Should you distribute your music with Magroove or another aggregator for streaming services and online music stores? Streaming services have their own conversion systems, and each one works in a different format (the vast majority, lossy), so you probably won’t get a lossless song. Get a good mp3 that survives encoding (and is easier to load) and save Lossless versions somewhere, whether burning to CD or distributing to acts overseas!
If you are a listener or a collector …
Are you archiving your music? FLAC with maximum compression! A lossless compressed format such as FLAC or ALAC lets you store your files efficiently without losing sound quality. Records in the original multimedia format settings. CD, for example, 16-bit, 44.1 kHz.
Do you digitize music from a vinyl? Record at least in a lossless format. Read about the recording / conversion settings, especially sample rate and bit depth, and understand what lossless quality you need.

Data compression techniques

It is evident that coding techniques for multimedia information contain large amounts of data that require memory space for recording and high transmission speed for transfer to other digital systems.

These needs can be met by reducing the space occupied by the data with special compression techniques. Compressed data cannot be used directly for processing, viewing, or playback. Compression techniques are used by special programs immediately before data storage or transmission. During the read or receive phase, similar programs perform decompression. Compression can be done on the basis that information encoding techniques dedicate an always equal amount of memory to each information element (be it a character, a pixel or a sound sample), regardless of their statistical frequency and its significance.

The compression techniques developed so far are more than a hundred but grouped into two categories:

Compression without loss of information.

Lossless compression techniques are based on compact coding of the same data streams or coding with a small number of bits of the most statistically frequent data.

This compression is completely reversible and the decompression program returns the exact bit sequence as it originally was. For this reason, loss-free technique is applicable to any type of data, including executable texts and programs, although the achievable compression factor is not very high: values usually range from 2: 1 to 4: 1. Of course, these results vary depending on the type of input data.

RLE encoding

The RLE (Run Length Encoding) compression technique is oriented to equal byte sequences. In the original version, it provides the introduction of a special character that indicates the beginning of a sequence, and instead of encoding the same characters in the sequence one by one, it encodes only the first one, followed by a number indicating where many times drawn and repeated. Specifies with the Sc character at the beginning of the sequence, the statement

these ******** are eight stars… these Sc * 8 are eight stars

where 8 is not encoded as an ASCII character but as a binary number.

The decompression program interprets the next byte as a counter and rebuilds the original sequence.

For image compression, RLE encoding only works well with images that contain large areas of uniform color, but are not very effective with complex images.

Compression with loss of information.

Loss-free compression techniques are not sufficient to solve the problem of the huge amount of data generated by encoding multimedia information, e.g. Video images while allowing better use of memory space on disks or data transmission lines. High resolution. , audio or video.

However, to try to solve this problem, it is necessary to remember that multimedia information, although subject to transformation, can remain understandable; This allows for compression factors that are higher in some orders of magnitude than those observed.

These interventions can be studied based on the behavior (vision and hearing) of our sensory systems to reduce the required memory without obvious changes in information content. Compression techniques that do this are called “lossy” since the least significant piece of information is irreversibly suppressed. Therefore, it appears that the bitstream after decompression is different from the original, and therefore these techniques cannot be used for other types of information, e.g. Text. Furthermore, the information thus compressed is not suitable for further processing as the loss introduced with each subsequent step becomes more and more apparent.

What is video encoding and how does it work?

The technique of compressing videos

What do we mean when we talk about video coding or, as industry experts generally call it, video coding?

Simply put, video encoding is the process of compressing and converting video content. The ultimate goal is to use less storage space, use less bandwidth, and make the user experience smoother. It goes without saying that the compression process causes a significant loss of information. The more data that is applied, the more data is deleted in the video. The result is a different version of the original due to missing data.

Why is video coding so important?

Video encoding is essential for transmission because it simplifies the transmission of video on the Internet through a compression process. Compression reduces the bandwidth required while providing a high quality experience. Without this, raw video content would not allow many users to view content on the Internet due to insufficient connection speeds. The protagonist of this process is the bit rate or the speed of digital data transmission that can be transmitted in a certain time interval in a communication channel. When streaming, the bit rate determines whether users can easily view the content or are exposed to video buffering.

Another fundamental aspect of video coding is compatibility. Indeed, sometimes the content is already compressed to an appropriate size, but it still needs to be encoded to be compatible with different devices and applications, although this is often referred to as transcoding.

The video encoding process is governed by video codecs, which are compression standards that are created through software or hardware applications. Each codec consists of an encoder for compressing the video and a decoder for restoring an approximation of the video for playback. The name codec is actually derived from the merging of the words “encoder” and “decoder”.

But what is the best codec?

It depends on the type of video. On this occasion we will describe the most commonly used.

To stream high quality video over the Internet, H.264 is arguably the most widely used codec for most multimedia traffic. This codec is considered to be of excellent quality, coding speed and compression efficiency, although it is not as efficient as the later HEVC (High Efficiency Video Coding) compression standard, also known as H.265. H.264 also supports 4K video streaming, a real advance for a codec created in 2003.

Now that we have an overview of codecs, let’s look at some compression techniques.

Compression techniques

The most common compression technique is scaling the resolution. The higher the resolution of a video, the more information is contained in each picture. One way to reduce the amount of data is to reduce the size of the image and then scan it again. As a result, fewer pixels are generated, which reduces the level of detail of the image, which has a positive effect on the amount of information required. This process allows you to set multiple quality levels for a video that correspond to different resolutions created. A practical example is if you are watching a movie in streaming before playing it, you can actually choose the resolution at which you want to watch it, provided your device
Support him

One video compression technique that may not be widely used is the interframe. This process reduces “redundant” information from one frame to another.

Another technique is the P-frame, short for predictive frame, which means that it can look back at an i-frame or another P-frame and understand whether the same images are present. In this case, this part is excluded for reasons of space.

B-Frame, on the other hand, is the bidirectional predictive frame that offers good compression without affecting the viewing experience. However, this technique requires a higher coding profile.

Another technique is that which makes it possible to intervene in the color. This process, called “chroma subsampling”, tries to maintain the brightness of the image, which affects the quality of the color. Finally, another method of compressing videos is to reduce the number of frames per second.

Digital audio format guide.

Since digital music has become popular, its fans have gradually developed the desire to hear it in the best possible quality.
For a time, the technical limits and the costs prevented us from achieving this goal, which is no longer an illusion.

The music streaming services market has never been so successful and we talk a lot about it in this article, but there are so many digital music formats … we’ve made this guide for clarity.

The teacher recorded in the studio is recorded at the resolution that defines the artist, but then the album we hear can be converted into a large number of more or less compressed formats, which, as shown in this infographic, into two large families are divided. . ::

Lossy audio formats: If the quality and the data contained are inferior to the original file, which leads to loss of detail, but to the benefit of the lightness of the file. These formats are widely used by streaming services like Spotify and take up less space.

Lossless: If the quality of the generated file matches that of the original master, without loss of quality, however, they require storage systems such as NAS and are not ideal for streaming via a mobile device connection.

Let’s start with the first Lossy family (AAC, MP3, Ogg Vorbis and WMA)

AAC

Advanced Audio Coding (MPEG-4) is the standard standard used by Apple for iTunes. Nostalgic people who had an iPod will remember that when CDs were imported into their MAC, the files were converted to this format. At the same bit rate, it takes up the same space as an MP3, but the conversion can be of higher quality.

MP3

An acronym for Moving Picture Expert Group-1/2 Audio Layer 3 is the world’s most widely used compressed audio format and was first introduced in 1998. The compression algorithm removes some details from the audio track that you can hardly hear The human ear By compressing a WAV file, you get an MP3 file that is up to 90 times lighter than the original. This depends on the bit rates or the amount of digital information (bits) that are transmitted or recorded in a unit of time. They can vary between 32 (poor quality) and 320 KB (excellent quality) per second.

OGG Vorbis

It’s an open source format with a quality comparable to MP3 that was unknown until recently, but is now used by services like Spotify. OGG is the file extension, while Vorbis is the compression algorithm.

WMA

Windows Media Player is the format invented by Microsoft in response to the MP3 standard. The quality is the same, but has the disadvantage that it is only compatible with devices of this brand.

Now let’s discover the features of the lossless family (FLAC, ALAC, AIFF, WAV and DSD).

FLAC

Free lossless audio codec. It is the most widely used open source format for downloading music without loss of quality compared to the original source. A fairly attentive ear can easily tell the difference from a smaller format, lower quality file.
iTunes does not support playback of this format because it uses its (ALAC).

A THE C.

The Apple Lossless Audio Codec or FLAC for the Apple world is only ideal if you use Apple branded devices, even if it does not correspond to the FLAC in terms of weight.

AIFF

This audio swap file format is also owned by Apple, which was developed based on the Electronic Arts swap file format and is particularly suitable for audiophiles and music producers. It has essentially the same properties as the WAV format, but without compression, and is used by artists because it can also be used to embed metadata such as text, notes or other information in the file.

WAV

The WAVE audio file format, introduced by Microsoft and IBM in 1991, continues to be used and represents what you get when you import a music CD to your Microsoft computer. When playing back a file of this format that can also be read by a MAC, a special function is used in the Intel CPU. These are very heavy files that reach a maximum of 2 GB and reproduce the sounds in a very faithful manner.

DSD

Direct Stream Digital is a registered trademark of Philips and Sony, developed for the Super Audio CD. An encoding method is used that uses a single bit instead of predicting a higher bit depth, but at a much higher sampling rate to record, store, and reproduce extremely high sound quality. The original sampling rate was 2,822 MHz and its format is known as DSD64

Audio compression, an explanation

Audio compression can be somewhat confusing at first due to the fact that the tools to implement it often have many elements that interact with each other and can be a headache.

Added to all this is the fact that audio / sound compression is often confused with compression in terms of digital formats (MP3 for example), which is a much more complex principle.

That is why we made this guide that aims to attack the most common doubts regarding compressors. The ones I had and the ones you probably have at the moment.

Let’s move on to the important:

What are compressors?

They are essentially an automatic volume or level control.

Let me explain: They are the equivalent of the fader of a console operated by a person in real time, that person has the function of lowering the fader when the volume of an element suddenly rises excessively. All this to control the dynamic range of said element and prevent it from going out of plane.

So what the compressor does in essence is reduce the level of a signal with parameters that are set by the user and that modify how it behaves.

How do they work?

Threshold and knee audio compression
An example of an acting audio compressor showing a 4: 1 reduction contrasting it with the signal without any reduction (1: 1)

Comparing signals, that is to say: a signal enters the compressor, for example the voice we were talking about before and we set a certain level (threshold or treshold) which, if exceeded, causes the compressor to act reducing the level of said voice at the output as if it were the fader on a console.

So the compressor is all the time comparing the input signal against this threshold and reducing the signal at the output if it passes it. On the other hand, the amount of reduction at the output is not always the same, but can be modified by the user with another parameter.

What are all those knobs?

Compressors have various user-modifiable parameters that appear in the form of knobs on both digital and hardware models. Let’s see what they are:

Threshold or Treshold: we tell the compressor that if the signal goes above a certain level, it reduces it in gain. The lower the amount of signal enters the compression and therefore there will be greater reduction in gain. A detail to keep in mind is that in digital models the threshold will appear as a negative number, in essence the more negative that number is, the lower the threshold and the more signal is compressed.
Compression ratio or Ratio: here we tell the compressor to reduce the signal that exceeds the threshold by a certain proportion established by us. For example, if our signal passes the threshold by 10 decibels and we want it to decrease by 5 decibels, we put a ratio of 2: 1 (it works as a division). At higher rates, there will be a greater reduction, but also the compression may start to be noticeable, which that we generally don’t want to happen. What is sought is that it be transparent so that the listener does not realize that the signal was manipulated.

Attack or Attack: it is the time in seconds (generally in the order of milli seconds) that the compressor takes from the moment the signal passes the threshold to the complete reduction in gain that we set with the compression ratio. Keep in mind that the compressor essentially acts immediately, but it is this time that determines how it interacts with the envelope of the signal to be compressed.

Release: is the time in milli seconds that the compressor takes to return to unity gain once the signal stops being above the set threshold. In the same way that with the attack the release can modify the envelope of the sound in question and therefore is very important in the operation of the compressor.

Knee: it is a parameter found in some compressors that modifies the way in which the compressor begins to act, the name is due to the fact that the curve that describes the way in which the compressor begins to act is similar to a knee (knee in English ).
So that we understand better when we talk about soft knee we are talking about that the compressor starts to act gradually before the set threshold and reaches its compression ratio established in this way. Instead, a hard knee compressor will only act when the signal goes beyond the established threshold and therefore more aggressively.

Make up gain or output gain: is the parameter that controls the compressor’s output gain, after having activated and reduced the signal by a number of decibels. What is sought in general is that what was reduced in level is re-gained and therefore make the parts that had less volume now approach those that were compressed.