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The truth about audio formats and their quality

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As you well know, there are three digital ways to play a song. From the original recording, through a copy with lossless compression (what we usually find when we buy a CD) or through a copy with lossy compression (which we usually download legally from the G.G internet).

The three files differ basically in the same property, which is none other than bitrate, that is, the information it contains per second. As more bitrate, more weight, so an audio file of a song without compression can quietly occupy 200mbs. In the second case (lossless understanding), the weight is reduced a lot (over 40, 50mbs), and is obtained by reducing the bitrate in those parts with silence, or with a wave oscillation in a single spectrum. To understand each other, the healthy young ear recognizes between 20Hz and 20KHz. A file with lossless compression (usually .flac files or those found on a CD) maintains this spectrum, reducing it when it is not necessary (silences). And finally there are the files with compression and loss (.mp3, .mp4, .flv, …) files that reduce this spectrum to the one that most ears recognize, leaving it around between 15Hz and 15KHz, obtaining a weight per file of song around 4mb, 5mb.
The latter has always been questioned, especially in musical circles, which assured that compression with loss greatly diminished the quality of what was reproduced, not allowing to admire the most serious or the most acute, thus losing all the completeness of the work.
As if that were not enough, mp3 files have different encodings (64, 128, 192 or 320 Kbps), with a greater or worse loss, and even constant (CBR) or variable bitrate (VBR) that is usually optimal when compressing with various bitrates Different moments of the songs.

Well, it has been more than 50 years before a good music lover programmer named Jeff Atwood decided to see if there really is a substantial change for the human ear between the different formats. In his blog, after several entries and several weeks of study in what would be called The Great Experiment of bitrate in MP3, we have finally obtained an empirical version of this eternal question.

But let’s make a brief summary of what we have in hand.
To test his hypothesis, Atwood decided to hang five audio files from his website, one of them being the original (without any digital treatment that modifies the bitrate), and another four tablets at various bitrates between 128 and 320 Kbps. The objective was that the user entered, listened to the five, and chose which one seemed to him to have higher or lower quality. Best of all, he obtained a not insignificant opinion of 3,500 visitors, hanging the results weeks later.

And from his observations you can get some gold reefs:
No doubt people knew how to differentiate the worst of all, such as the mp3 encoded with the worst bitrate 128 Kbps CBR.

The variable bit rate coding proved to be higher than the constant.
The most positive audio obtained was that of 160Kbps VBR, even higher than 320 Kbps CBR, and paradoxically also superior to the original audio of the CD.

From all this follows a corollary:
People are unable to ensure that it has more quality above 160Kbps, so it sends lossless formats to the horn that occupy one more scumbag, and that in practice, our ear cannot discern.
So you know. That’s over 15, 20 songs for a CD. There is no excuse.

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MP3 vs FLAC vs AAC vs OGG: what differences does each audio format have?

Although streaming platforms such as Spotify are more fashionable than ever thanks to their great musical variety, reduced price and convenience of not having to manually download the files, many users still prefer to have music stored locally, for which there are formats like MP3, FLAC, AAC and OGG.

These formats are currently the most widespread for music on our devices, being able to pass files between PC and mobile without relying on the Internet and without being afraid of depleting our data rate. Most of the formats that we are going to deal with are formats that compress information, and therefore have quality losses. About the compression of images and files we talked a while ago.

Why does a compressed file occupy less?

Audio formats with losses: MP3, AAC and OGG

The first of the formats that we are going to try is MP3. This format, whose acronym stands for MPEG Audio Layer III, is the most commonly used format with loss of quality. It is not the one that offers the best quality or best compression, but its great compatibility has made the standard format for music for decades.

Another widely used format for sound in recent years is AAC. It is very similar in MP3 performance, but has the advantage that it is able to offer the same quality in a smaller size. This is the reason why platforms like Apple’s iTunes use it, and the fact that Apple uses it has made its compatibility as great as MP3’s today. AAC is also used to compress stereo sound in movies of 1 or 2 GB in size that we find in various torrent portals, direct download or streaming.

The next most used format is OGG, or OGG Vorbis, it is a free alternative to AAC and MP3 (although the MP3 patent ended last May). Its size is similar to that of MP3, but its compression is smaller, keeping a higher audio quality than MP3, especially at high frequencies, which destroys the MP3 the lower the bitrate. In addition, while MP3 reaches 320 Kbps, OGG reaches up to 500 Kbps.

Lossless audio formats: FLAC, ALAC and WAV

On the other hand, we have FLAC. This lossless format is free, as indicated by its name (Free Lossless Audio Codec). The size of your files is between 5 and 10 times larger than MP3, but it has no losses, although the audio is “compressed.” Thus, it occupies much less than uncompressed formats such as WAV or AIFF, and maintaining the same sound quality.

The equivalent of FLAC in Apple is ALAC. Although it is not as efficient as FLAC (its files occupy more), ALAC owns Apple, and is the only alternative that can be used in iTunes, since the platform does not read FLAC.

In short, the best format to use is always FLAC if you can afford its large size, followed by AAC and OGG. If you have no choice, MP3, although it is the least desirable option, is the most widespread today, and what you will be forced to use for a lot of music on the network.

What audio formats exist? All you need to know

FLAC, WAV, AIFF, DSD … these are just some of the acronyms you can find when looking for a digital format. They are also accompanied by technical data such as sample rates and bit depth. So many terms can leave you more misplaced than a chicken in a dance. And unless you are an expert in digital sound, the process to choose the audio format that best suits your needs can be a mess. But if they explain it to you, the subject is relatively simple. That is why in Culturasonora we have prepared a complete guide on the different audio formats used. This will prevent any acronym from taking you on the dark side, dear Padawan.

Sample Rate and Bit Depth.
MP3s vs WAVs vs AIFF.
OGG vs FLAC vs ALAC.
What is the DSD format?
How to listen to the DSD?
MQA audio Hi-Res.
What is Bit Depth and Sample Rate?

These two concepts are basic. To understand how audio formats work, you need to know what Bit Depth and Sample Rate are. They are two measures that indicate the quality of a digital audio file. We will try to summarize it so that you stay with the general idea

When you read the specifications of the audio formats you find a couple of figures. For example: 32-bit / 192kHz or 24-bit / 96kHz. These numbers indicate the bit depth and the sample rate. These references tell us how much information the different formats transmit and the sound quality. For example, the audio we hear on a normal CD, or on a Spotify stream, is 16bit / 44.1kHz. Samples are always measured in Hertz (or hertz) and bit depth in Bits.
Softwares or hardwares do not usually work with a continuous flow of information but often use pieces, samples or samples to effectively manage the data that is transmitted. The sample rate is the number of samples per second that are obtained from a recording. The higher the number of times a device plays the samples, the higher the sound quality. Each of these extracts or samples has a certain amount of information, which is the bit depth, or bit depth.
To understand it better, we are going to make a slightly beast analogy, which is not entirely true, but which will help you to make sense of all this. What interests us. If you control a bit of photography and image you will get it right away: the sample rate would be something similar to the frames or frames per second of a video, and the bit rate would be similar to the pixels of a photograph. The higher the bit depth number, the more information each sample will have. The more pixels an image has, the more resolution each frame of a video will have. The more frames per second a movie has, the greater the definition. In short: the higher the number of the Bit Depth and the Sample Rate, the higher the quality of the audio file.

Audio formats: MP3 vs WAV vs AIFF

What is the MP3 format?
If you are interested in getting some audio fidelity and decent sound from your files, you will want to avoid this format. Why? Because basically an MP3 is a file that sacrifices audio quality to minimize size. They weigh very little for any device to read. The negative? The compression of these files provides a poor, almost lifeless sound. Nowadays almost nobody uses that format seriously. Even its creators recently finished the license declaring her dead. But surely every now and then you find a zombie file with this format.
What is the WAV format?
WAV (Waveform Audio File Format) are equally common but better for anyone who wants a decent audio format. They are higher resolution files than MP3s. A WAV is an audio piece that is encoded with something known as Pulse Code Modulation (PCM), a medium that encodes analog audio parts and converts them into digital so that they can have the Sample rates and the Bit Depth of the that we have talked about before.
What is the AIFF format?
The audio format AIFF (Audio Interchange File Format) is very similar to WAV, since it also uses the PCM to encode analog audio pieces and present them in digital format. This format was born as an answer from Apple to the Microsoft WAV, and at the beginning it could only work on MAC computers. Currently, the AIFF and WAV are more or less interchangeable.
In summary…
To close this topic we will tell you that if you have a file in WAV or AIFF audio formats you will hear a piece of good quality sound. Normally these formats are used in files that we play through our services, such as the iTunes music library. We will not see them in online streaming services, which tend to use special types of files. Now we will review that point

Do you differentiate between an mp3 encoded at 128 and one at 320 kbp?

Surely more than once you starred in or attended a dispute between people who say that you notice a lot of difference between an MP3 encoded with one or another level of compression, or between a CD and an MP3. However, there are very few people able to distinguish these nuances. That’s why at mp3ornot.com we propose this challenge:

Are you able to differentiate between an mp3 encoded at 128 kbps from another at 320 kbps? If you think you have your ear developed enough to capture that difference, I challenge you to take the test … and then tell me.

Data:

The Mp3 (MPEG-1/2 Audio Layer 3) was one of the first types of audio compression with almost imperceptible losses to the human ear. Its compression rate is measured in kbps (kilobits per second), with 128 kbps being the standard quality, in which the file size reduction is about 90%, that is, a ratio of 10: 1. That compression rate can currently reach up to 320 kbps, the maximum quality, in which the file size reduction is about 25%, that is, a ratio of 4: 1, going before 192 kbps, 256 kbps, that is, the maximum quality that can be removed in Mp3.

The lossy compression method used in the compression of the Mp3 consists in removing from the audio everything that the human ear would normally not be able to perceive, due to phenomena of masking sounds and limitations of human hearing (although people with absolute hearing can perceive such losses).

How to compress an MP3 file

Knowing that the MP3 audio format has become the most standardized and used worldwide in recent years, we have thought it pertinent to talk about the different parameters that make an MP3 file respond to one quality or another.

The first thing we have to know is the meaning of MP3, and it is nothing more than a compressed digital audio format that although by nature suffers a loss of information in the conversion process, it is not audible by the human ear, which It implies an assumable loss since we will not be able to perceive it in broad strokes.

Generally, an MP3 file is capable of reducing the size of an original audio file without altering quality. What this means is that in the conversion process for example of an audio file with CD quality, the result of the MP3 file would be practically identical to the original, leaving as standard ratio 1 minute = 1 MB.

That said, we can begin to clarify some parameters that will determine the quality of an MP3 file, which in its vast majority, depends on the bitrate or Bitrate.

Impact of Bitrate in MP3 quality
The MP3 file format allows you to select the compression ratio of the source file. The margins at the domestic level are between 8 Kbps and 340 Kbps, with 128 Kbps being the transfer rate equivalent to CD quality.

Bitrate is the unit of measure for the rate of data transfer read from an MP3 file. The higher bitrate an MP3 file has, the greater the amount of data that a player can obtain in the unit of time (Second).

The more instrumental content or quality an MP3 audio file contains (sound effects, recorded audio tracks, high frequencies, low frequencies, etc.), the higher the transfer rate it will require to fully reproduce the information, and at this point, it is where it is defined The quality of the MP3 file, since if we compress that file, we reduce that bandwidth, we will be sacrificing some of that data, resulting in loss of information that will influence the final result of the MP3 conversion.

In summary:

If the file lasts 5 minutes and weighs 3 MB, we would be talking about a low quality MP3 file.

If the file lasts 5 minutes and weighs 9 MB, we would be talking about a high quality MP3 file.

The great experiment on MP3 quality: no, there really isn’t that much difference with CDs

This article was originally published in Cooking Ideas, a Vodafone blog where we collaborate weekly with the goal of creating stories that “feed the mind of ideas.”

A programmer named Jeff Atwood said some time and several entries from his blog, the always recommended Coding Horror, to a healthy entertainment he called The Great Experiment of bitrate in MP3. Its objective: to verify empirically if for ordinary people there are really qualitative differences when listening to music in various MP3 formats compared to traditional ones.

The contestants were the traditional formats called “no loss of quality”, basically CD (Compact Disc) and FLAC versus compression formats with loss of quality: MP3 with different bitrates. The bit rate, better known by its name in English, is a key feature because it basically determines how much information is transmitted per unit of time: in this case it is the waves that define the music and become human voices and instrument notes . In the world of MP3 encodings of 64, 128, 192 or 320 Kbps (kilobits per second) are usually used.

Like everything in life, music coding is a compromise between quality and quantity: a song stored in the best possible format – for almost all experts, that is the CD – can occupy about 50 MB (megabytes), maybe 40 or 35 only using some of the lossless compressors that save some space without loss of quality (FLAC, Apple Lossless, etc.). That same song in MP3 can vary between 4, 8 and 12 MB depending on the bitrate (64, 128 and 192 Kbps). To further complicate the matter, you can also choose between a constant (CBR) or variable (VBR) bitrate that is usually optimal when compressing different moments of the songs with various bitrates.

For many users, being able to store between 5 and 10 times more music in the same space is an important saving, easy to translate if one takes into account the price of hard drives, flash memories or storage on iPods, tablets and the like. But there have always been two schools confronted: that of audiophiles who believe that nothing can equal the maximum quality of the CD and that of those who, with a more practical sense, consider the differences between an MP3 and CD ridiculous, if at all there are.

Atwood’s experimental study sought precisely to shed some light on these theories based on the basics: listening to music, quantifying its “quality” and deciding which is the best format based on the various variables. For this, he prepared five different audio files: one of them uncompressed and another four tablets at different bitrates between 128 and 320 Kbps. He put them on his server so that people could listen to them and vote (with a quality “note” of 1 to 5) without knowing which was which. And in total he got more than 3,500 people to contribute to the results – hundreds more than for many of the “quality studies” mentioned in the TV commercials.

The results were analyzed with a spreadsheet and various statistical tools, which showed trends and conclusions quite clearly:

The only sample that could really be considered very different from the rest was the MP3 at 128 Kbps CBR, the worst quality. That quality is not enough to compare with the rest. The best simply ignore it.

The MP3 at 160 Kbps VBR is the highest quality sample, even better than the MP3 at 320 Kbps CBR. This indicates that the coding with a variable bit rate is higher than the fixed one even at those values, and that 160 Kbps VBR up is impossible to improve qualitatively.
Ironically, this would indicate that there are MP3s that are heard “better” than audio CDs. Several things can happen here: that the “artifacts” created by compression seem to improve the audio or that when testing people “imagine things,” which could also happen. The truth is that the data serves to feed the theory that from 160 Kbps people no longer distinguish one quality from another, as it is deduced from the data.

The conclusion of the study confirms the hypothesis that an MP3 at 192 Kbps VBR has such quality that not even the ultrasensitive and powerful ear of a dog would notice the difference with an audio CD. Wow!
In conclusion, we already know at what rate to code and compress if we want a good saving in storage without losing quality: a MP3 of 192 Kbps VBR, the winning format of the test.

What are the differences between mp3 and mp4?

When we want to listen to our favorite songs, we don’t skimp on tastes or ways of listening to those pieces of the moment. Although there is no doubt that when we have two good devices through which we can listen to them we look for the advantages of some compared to the others to select the best one.

The differences between an mp3 and an mp4 are varied, everything will be judged by the qualities that make them more attractive to us according to a value judgment based on their characteristics.

When we make a comparison of differences between an mp3 and an mp4 we observe that the first only supports audio formats while the second allows all types of audio and even video and images with the possibility of having greater storage capacity since it allows compressing much more the formats in order to have many more songs, videos and images in it.

Both are equally functional as they offer a possibility to listen to our favorite songs from the device. Another notable difference between an mp3 and an mp4 is the possibility of storage so it is very regular to check how many files we can save in one or the other.

Likewise, the mp3 due to its MPEG-1 Audio Layer 3 condition decreases the size of the original file, which generates losses when listening to the songs that we store in it, since they are compressed and undermine their quality. The mp4 player for its part, was designed years later so it is seen that it supports the video and image format but in turn does not cause the sound quality to be lost both in the songs and in the videos although it also Compress without altering your audio levels.

In their design there is no notable difference because they have similar models only changes is the name of the device. The two big differences are that the mp4 is multimedia while the mp3 only allows you to play sounds. And its compression causes the audio quality to be lost in the mp3 while the mp4 is an improved device that compresses the format without losing its quality and efficiency.

How to measure sound quality

How to measure sound quality?

There are many ways to measure sound quality.
Many measurements that have been created specifically to measure and rate the sound quality. These are some of the most common types of measures used for sound quality.

THD

THD means Total Harmonic Distortion. This is a measure taken to see the total amount of distortion of the original audio signal in playback. Most quality components have less than 1% distortion rate, speakers can have 1 to 5% and the distortion rate of serious subwoofers can produce most distortion sometimes approaching 10%. However, human ears cannot differentiate very low distortion at low levels.

Output power

Output power is the maximum energy per channel, usually shown as Watts. Most speakers or receivers show how many watts they have. There are two peak measurements and RMS. Peak watts is usually what is shown in advertising, usually it is the maximum amount of energy that a speaker or receiver can give out of a very short period of time. RMS stands for Root Media Square and is a more appropriate way to determine power for a longer period of time.

Frequency Response

Frequency response is an important way to determine sound quality. Most human ears can hear frequency (sound), which is about 20 to about 20 K HZ HZ. Bass forms the lower parts of the spectrum frequency response. Most people consider bass frequencies, frequencies below 310 Hz. Bass frequencies include percussion explosions and DVD movie. Midrange frequencies are frequencies ranging from about 310 to about 12K HZ HZ. These frequencies include dialogue and most of the human voice, piano, guitar and other instruments. High frequencies are at the top of the spectrum frequency response. They are usually 12K to about 20K or higher. Cymbals, high notes of the human voice, and some string instruments are included.

Signal to Noise Ratio

Signal to noise ratio is the ratio of sound quality to noise. This measurement is used to measure many devices including receivers, CD players, DVD players, etc. Generally, the highest is decibels (Db), the best sound quality. For example, a signal-to-noise ratio of 90 decibels or 100 is considered high fidelity. Most electronics are generally 80 decibels or more, which is great for many human ears that discern.

It is important to note that terms are not the only way to measure sound. There are several ways to measure sound, components and media that are either digital or analog. However, knowing about the terms that give an idea of the process of measuring sound quality.

High resolution audio: myths and realities – 2

If we stick to the characteristics of the CD we can see that our music is obtained by taking 44,100 samples per second (correspond to 44.1 kHz) from the original analog signal, and each of them is encoded in a data package that It uses 16 bits. And at this point, finally, it is where high-resolution audio comes into play.

Coding

The starting point of this technology is easy to understand: it presupposes that if we increase the resolution, the sampling frequency, or even both parameters at the same time when passing an analog signal to the digital domain, we can “reconstruct” the original analog signal With more precision. And it really is. For this reason, the specifications commonly used in high resolution audio formats are 24 bits and 96 kHz, or 24 bits and 192 kHz. Both options, on paper, should allow us to recreate the original continuous signal more accurately than the 16 bits and 44.1 kHz of the CD, or, what is the same, will discard less information from the original sound.

But this is not all. In addition, increasing the resolution to 24 bits increases the dynamic range and improves the signal-to-noise ratio (our Xataka Smart Home partners explain what these parameters mean in this post). A resolution of 16 bits allows us to encode a total of 65,536 possible levels for each of our samples, while a 24-bit one reaches 16,777,216 levels.

The resolution commonly used in high definition audio formats is 24 bits, and the sampling frequency 96 kHz or 192 kHz

The difference between the two extremes, which is where the lowest and highest levels are located, indicates the dynamic range difference between one resolution and another. With all this data on the table we can think that high resolution sound should offer us more quality than the audio of a standard CD. And it is so, but, as we will see later, there are factors that limit the experience and that users must take into account, beyond what the industry “sells” us.

Internet: key to the success of HD audio

At this point we can understand without difficulty that the size of a sound file depends on the resolution and sampling frequency used to encode the music it contains. The same issue occupies much more if we digitize it at 24 bits and 96 kHz than if we do it at 16 bits and 44.1 kHz. However, we have a very interesting resource that helps us save space: compression. Currently, high resolution audio is usually distributed in six different formats (some of them offer compression without loss of quality): FLAC (compress without loss), ALAC (the lossless compression technology proposed by Apple), AIFF (it is the format of Mac sound file), WAV (this is the sound file format created by Microsoft and IBM for PCs), DSD DFF (SACD format encoding technology) and DSD DSF (DSD variant for Sony VAIO computers).

Of all the formats that I have just mentioned, the most used to distribute high resolution music on the Internet are FLAC and ALAC because both offer a very interesting compression rate, and without loss of quality. And we all know that size matters on the Internet. And a lot. In fact, the network is playing an essential role in popularizing high resolution sound.

High resolution audio: myths and realities

High resolution sound is in vogue. A good part of the manufacturers of music equipment, and more and more record labels, especially those that sell over the Internet, seem determined to convince us that high-resolution audio is what all of us who love music should aspire to if We want to enjoy it with the highest possible quality.

On paper there are technical foundations that justify the existence of high resolution audio and indicate that its quality should be higher than that offered by the CD. But there are also solid reasons that invite us not to take their superiority for granted, at least not in such a clear way, and to question some of the virtues that the industry sells us. Let’s see what high resolution sound is, what we need to enjoy it, and, above all, if it really offers us a better experience than music with standard quality (that of the CD).

What is high resolution sound?

To understand in a simple way what high resolution sound offers us, it is good to review how music is stored on CDs that we all know. These discs, unlike vinyl, allow us to store information in the digital domain, while vinyl discs are analog. This means that the music on a CD is encoded in the form of ones and zeros, in exactly the same way as the information we have on the hard drive of our computer, which is also digital.

The sound of your smartphone may resemble that of cinema thanks to neuroscience: this is how Dolby has integrated Atmos audio into it
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The sound of your smartphone may resemble that of cinema thanks to neuroscience: this is how Dolby has integrated Atmos audio into it
But a CD does not have an infinite capacity; in fact, the size of the tiny notches that we see on its surface if we look at it with a microscope and the distance that separates each one from the adjacent ones reveals how much information it is capable of holding. Precisely, in a simplified way, this is what differentiates the CD, the DVD and the Blu-ray Disc: the size of the small holes that encode the information and the distance that separates them. If we compare two discs with the same diameter, the one with these smaller notches and more together will have more capacity. This parameter is precisely what determines the wavelength of the laser that we must use to extract the information.

Formats

The technology of the CD format was developed at the end of the 70s by Philips and Sony, and it was the engineers of the latter company who proposed to encode the information using a resolution of 16 bits and a sampling frequency of 44.1 kHz. But these figures were not chosen at random; These specifications allow this format to reproduce the sounds that are in the frequency range that goes from 20 Hz to 20 kHz, which coincides quite accurately with the frequency limit that the human auditory system is able to perceive, even bearing in mind that Not all people have the same hearing ability.

The CD uses a sampling of 44.1 kHz to, according to the Nyquist-Shannon theorem, be able to reproduce frequencies up to 20 kHz
To understand what is the resolution and the sampling frequency without going into too cumbersome details we can think that to be able to store an analog signal, and, therefore, continuous, in a digital medium, which has a limited capacity, it is essential « Chop up that continuous signal into small fragments, or samples, and introduce as many as fit into the digital medium. The resolution indicates the number of bits that we can use to describe each of these samples, which, in turn, reveals the number of variations or possibilities that each of them can adopt. And the sampling frequency tells us how many we are going to be able to take.

(Part 1)

Audio Normalizer in 2019

We all enjoy music, and we have all modified our behavior by adapting to the new devices.
Today nobody sits to look at the cover of an LP while it turns on the turntable.
Those times are behind.
Today, however, it is quite easy to get an audio file (mp3, flac, ogg, m4a, etc.) of some song that we listen to and like.

But that “get” implies downloading a file from an unknown source. There is no risk in it, at least not a risk of getting infected. But if we run the risk of downloading music files that have been encoded by someone who has no idea what they are doing.

I wonder, why do we have some that have 128 bitrate and another 92 and another 320k? And the same goes for all the settings with which it was encoded. It is as if they were chosen randomly, without really knowing what each thing is.

And then comes the consequence that they all sound “different”, with different volume levels, which we perceive with different volumes and sounds. Our collection, which sometimes becomes very large, is really uneven.

Then appears the desperation to find a software like Mp4Gain that allows us to be a volume leveler, volume enhancer, volume booster. In short, so that everyone can understand: an audio normalizer. And this audio normalizer should be able to normalize the most popular audio and video formats, the mp3 is no longer enough, as was the mp3gain.

Unfortunately some of our audio files will be co-opted by encoders that are not really standard and maybe the file cannot be repaired, but that is the minority, the other 99% will be able to normalize perfectly.