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Normalize audio: Normalization and normalizer

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Normalization – Normalize audio

Normalization is an atypical dynamic process, very different from compression, limitation, expansion or noise reduction:

It does not reduce the relative dynamic range of the audio signal.
It is not applied in “real time”, or at the moment, but it is a process that is carried out “a posteriori”, on the previously recorded material.
The normalizer is a device that falls into the category of dynamics processors.

Analyze the target signal, detect its highest volume peak and increase its gain to the maximum level possible without distorting.
With the same proportion the level of the rest of the signal increases.
The signal, in general, will sound with a greater volume.

It is very unpleasant to have a playlist with mp3 files playing at different volumes. However, it is possible to normalize (equalize) the volume of our mp3s.

Loudness normalizer

You can use the Loudness Normalizer to obtain a specific loudness.

Increasing the loudness to a specific value may cause clipping. To solve it, a peak limiter can be part of the process. The Loudness Normalizer increases the loudness and limits the peaks of the signal at the same time if necessary, to obtain the desired loudness.

This process takes place in several stages, first an analysis and then the final rendering.

What is an audio compressor?

When we talk about compression we also talk about dynamic range. Recall that the dynamic range is the difference in amplitude between the lowest and highest part of a signal. In the compression process, basically and technically it consists in decreasing the dynamic range of a signal.

These are the most common controls or points to control in an audio compressor:

Threshold: This is the point from which the compressor will start working. Any signal that exceeds this point will be compressed.

Ratio: This is the reason for the compression that will be applied to the signal that exceeds the threshold. We see it and find it as 2: 1, 4: 1, etc. The first number means the number of input decibels that exceed the threshold, and the second number the output decibels. In other words, for “X” number of decibels that exceed the threshold, “Y” number of decibels will be output.

Attack: It is the time it takes for the signal to compress after having exceeded the threshold. We see it and find it commonly expressed in milliseconds (ms).

Release: How soon the compressor stops working after the signal falls below the threshold.

Knee: This is a parameter that subtly modifies the compressor threshold. A low setting (Hard knee or 0) means that the compressor will act only from the set threshold. A high setting (Soft knee) will allow the compressor to act gradually from before the signal reaches the threshold. In this way, the threshold can be treated as a range and not as a specific point. As the signal approaches the threshold compression increases. When it exceeds it, it continues to increase until the entire compression ratio is applied.

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WebM: everything you need to know about the Google format

What is the WebM?

WebM is a container format (with extension * .webm) for multimedia files, that is, for videos and audio files. In the same container the video codecs VP8 and VP9 are used, as well as the Vorbis and Opus audio codecs. At the Google I / 0 2010 conference, the company announced its plan for WebM to be an alternative to the existing MP4 format with its H.264 codec from the beginning. The consumer can use the latter at no cost when watching a video, but developers who want to work with the codec must pay the license fees. On the contrary, WebM is an open source project with which anyone can work without paying rights for it.

WebM is designed for use with HTML5. The VP8 and VP9 codecs are designed so that in those cases where considerable compression must be carried out, the extraction can still occur with little computing power. The objective of this design is to allow the reproduction of Internet videos on virtually any device (regardless of whether it is a desktop computer, a tablet, a smartphone or a multimedia device such as a Smart TV). It is not surprising that YouTube, being a subsidiary of Google, converts all its videos to the WebM format, regardless of the format of the original file. Despite everything, YouTube still supports H.264 for those who cannot play WebM.

WebM has become a political issue within the Internet community. While Google tries hard to consolidate this audio and video format, other important market players such as Apple or Microsoft cling to formats like MP4. The main reason is, above all, the patent system: both software companies use a group of MPEG-LA patents, since it is responsible for maintaining the patents of the used codecs and charging royalties for them. Google is trying to circumvent these patents with WebM.

This situation has already led to legal problems in the past, the VP8 codec being the point of contention. Several companies have criticized that their codec patent has been ignored. Google would have reached an agreement with MPEG LA, however, Nokia is not part of this patent pool and believes its rights have been ignored. A first lawsuit, in which the company faced its competitor HTC before the courts, whose devices support V8, was dismissed by the Mannheim regional court.

WebM vs. MP4: advantages and disadvantages

While WebM is relatively young, MP4 (MPEG-4 Part 14) and H.264 have been used for many years. Due to its age, this format and the codec have become a standard: you will find few applications that do not support MP4. In addition to Internet services and PC and MAC software, many other devices (such as camcorders) can also use MP4. The high degree of acceptance makes the format interesting for both manufacturers and users.

But Google has been marked somewhat with the open source character of WebM: using the format is no cost to manufacturers, developers or end users. In addition, the software is distributed under an open BSD license.

The fabric behind the MP4 or H.264 license is opaque: most users, even those who create videos in a professional way, do not know if they have a valid license with the purchase of hardware or software or if any video violates The license right. WebM eliminates this confusion. The MPEG LA already announced in 2010 that the use of the H.264 codec would also be free in the future, provided that the videos created were already free for users.

For many users, the performance of both formats is more important than the controversies surrounding their patents: it is for some reason that H.264 has positioned itself as the leader of the codecs in recent years. The quality of MP4 videos of this encoding is generally considered very good. H.265 exceeds it in some aspects. WebM also convinces with the image and audio quality, but VP8 does not reach the level of H.264. To what extent the image quality of VP9 approaches H.265 (also known as HEVC) is a controversial issue; some believe that both are equal, while others say that the quality of VP9 does not reach that of H.264.

Two other determining characteristics when comparing codecs are the file size and the speed of encoding and decoding. Both directly influence the utility: for fast data transmission over the Internet, the size should be kept as small as possible. This is especially relevant in the mobile Internet field. H.264 has a bad reputation for creating, in comparison, large files. At the same time, decoding on the user’s site

The video formats for internet

The videos that we play on the Internet either locally on the computer or on any other device, can be encoded in different ways. Each method of coding implies some advantages and disadvantages, and there are better formats than others depending on the use we want to give the video. Thus, we have formats such as AVI, MP4, MKV, 3GP, Google WebM, etc …

When playing video on the Web, using a browser, the most widespread and best supported format by browsers, both mobile and desktop, is the MP4 format, and to be more exact, the MP4 / H format .264, which corresponds to files that normally have the extension .mp4. But:

Are all the .mp4 files the same?

The MP4 format – Parts, containers and extensions
When we talk about an .mp4 file or the MP4 format in general, what we are talking about is what is technically known as MPEG-4 Part 14. It is a standard format (ISO / IEC 14496-14) and is a container format of multimedia tracks. That is, this format defines how audio and video tracks (called data streams) in various formats can be contained in the file, and can even contain subtitles as well.

Within this container format, within the .mp4 file, the audio and video tracks may be encoded in various formats, as appropriate for the application to be given. Although in theory it supports many different formats (almost any) for these audio and video tracks, in practice the players of this format support only some specific types, the most frequent being:

Audio: AAC (Advanced Audio Codec, which when they are loose are files with extensions .m4a or .3gp), or the MP3 format.
Video: the different variants of the MPEG format.
MPEG or Moving Picture Experts Group is a group of “authorities” and manufacturers in the field of audio and video that came together at the request of ISO in the late 1980s to create file encoding standards for this type of multimedia information, and thus guarantee compatibility between media to be reproduced and reproductive devices. The first version of the standard, MPEG-1 came out in 1993, and since then there have been many new versions, and within these what they call “Parts”, which are specific aspects of the standard and also extensions to the specification for specific things, or improvements to the base format that they modify.

The most widespread version of this MPEG format is 4, or MPEG-4, which appeared at the end of 1998, and is what we know as MP4, due to the extension of its files. This version is divided into several sub-standards or “Parts” that describe certain issues of the format (for example, the 14 container, as I said at the beginning) and certain extensions.

Within the parts of the standard, part 10 describes an advanced coding format that is what we also know as H.264, but which is actually also called MPEG-4 Part 10 and what Blue-Ray discs used , for example. That is why in many Internet sites they talk about H.264 and MPEG-4 being the same. And it is true, but not quite, since in MPEG-4 parts 2, 12 or 14 also describe other compression formats that are MPEG-4, and the container format may also contain MPEG in earlier versions of lower quality such as MPEG -1 or MPEG-2.

How MP3 files work

The MP3 movement is one of the most incredible phenomena that the music industry has ever seen. Unlike other similar phenomena, such as the introduction of cassette tape or CD, MP3 technology did not start with the industry, but with a huge audience of music lovers on the Internet. The digital MP3 music format has had, and will continue to have a great impact on how people collect, listen and distribute the music.

If you have wondered how MP3 files work, or simply want to know what uses can be given, read on. This article will give some features of this popular sound format.

MP3 format

If you know something about how CD’s work, then you know how they store music. A CD stores a song in the form of digital information. The data on a CD uses a decompressed high resolution format. This is what happens when a CD is created:

The music is sampled (fractionated) 44,100 times per second. Each of these parts has a size of 16 bits.
Pieces of these fractions or “samples” are taken from the left and right channels in a stereo system.
With a simple formula we realize how great a single song can be.

Fractions * bits * channels = X bits per second

In our case it would be 44,100 for 16 bits per 2 channels, which would give us 1,411,200 bits per second. 1.4 million bits per second equals 176,000 bytes per second. If the average of a song is 3 minutes, then the average of a song on a CD is 32 million bytes of space. That is a lot of space for a song, and it is especially great if we consider that we are downloading music with a 56K Modem, which will take us a few hours.

The MP3 format is a compression system for music. This format allows you to reduce the number of bytes in a song without damaging the sound quality. The goal of the MP3 format is to compress a CD quality song without letting you see the difference. With MP3, a 32 MB song from a CD, compresses up to 3 MB. This allows you to download a song in minutes instead of hours, and store hundreds of songs on your computer’s hard drive.

Compression and quality

Is it possible to compress a song without damaging the quality? To perform this compression, the use of algorithms is needed, in the same way that we use them to compress other formats, such as graphics, text files, applications, etc. A very popular algorithm for compressing sound is the “perceptual noise shaping” technique. This algorithm uses characteristics of the human ear such as:

There are certain sounds that the human ear cannot hear.
There are certain sounds that the human ear hears better than others.
Its there are two sounds playing at the same time, we can hear the one that is louder, and not the lowest.
Using factors like these, certain parts of the song can be eliminated without significantly damaging the quality of the song for the listener. When you have created the MP3 file, what you have is music with a quality close to that of a conventional CD. It doesn’t sound exactly the same because some things have been removed, but it’s very close.

Using the MP3 format

The MP3 movement – consisting of the MP3 format itself and the ability of websites to distribute it – have done several things in the music world:

It has made it easy for anyone to distribute music at a low cost, or even for free.
It has made accessing music simple and instant.
He has taught people to manipulate music on a computer.
One of the strengths of this format is the ability to edit, create and modify music files thanks to powerful computer software tools. Thanks to these tools, it is extremely easy for anyone:

Download an MP3 file from a website and play it instantly.
Transform or “rip” a song from a CD, to the MP3 format, and listen to it later.
Record a song yourself, convert it to MP3, and make it available to everyone on the Internet.
Convert MP3 files into CD files and make your own audio CD’s with MP3 files downloaded from the Internet.
Have thousands of hours of music stored on one or more hard drives.
Upload MP3 files to portable players and listen to them wherever you want.
To do all this, all you need is a computer with a sound card, speakers, an Internet connection, a CD / DVD player / recorder, and an MP3 player.

Sound formats and audio normalization

WAV: It is the “pure” sound format, without any compression. Its weight is huge, as is its quality. Only recommended for professional works or to edit the audio before transferring it to a format with compression.
MP3: We’ve talked about him in the previous pages. Without a doubt, it is the most popular and widespread format. His appearance changed the way we listen to music.
OGG: It is the audio format of GNU / Linux, the free software MP3 version. It has all the virtues of MP3 (and more), but not all portable players can use it, but it is getting more and more.
WMA: Microsoft format, your own version of the MP3. It compresses quite well, but it is not as widespread as the MP3. Nor can all portable players use it.
MID: It is the audio format also known as MIDI (Musical Instrument Digital Interface). It is the only format that can not play more than music simply because what it contains inside are not sounds. Simplifying, it contains a series of instructions for special software included in all systems, a kind of digital synthesizer that can generate sounds like those of many musical instruments. The MID has inside what notes they have to sound and with what instruments: a score.

It is important to clarify the distinction between audio format and audio codec. The codec encodes and decodes the audio data while this data is archived in a file that has a specific audio format.

Most of the formats listed below are container formats, formats that group different types of data. Most of these container formats have only one codec associated, next to which metadata is stored. However, there are formats that group audio and video data produced by different codecs. Some of these container formats that group different types of data are: MP4, Ogg, WAV, QuickTime Format, AVI.

In this article we talk about audio formats, but we are really discussing the properties of the codec associated with the format.

When classifying audio formats we can distinguish three large groups.

No data compression: These are real sound waves that have been captured and converted to digital format without further processing. As a result, uncompressed audio files tend to be the most accurate.
With compression, without loss of data: Compression algorithms are used to reduce file sizes; It basically works by eliminating redundancy.
With compression and data loss: It is a form of compression that loses data during the compression process. In the context of audio, that means sacrificing quality and fidelity to decrease file size. The good news is that, in most cases, we will not notice the difference when listening.

Compression

Compression is a process that involves reducing the dynamic range of an audio signal.

An apparatus, called a compressor, analyzes the gain of the input signal and, according to certain parameters set, those parts that exceed a level or threshold determined according to the desired configuration are attenuated.

In principle, compression is perceived a decrease in overall volume; In fact, this is because the compressor reduces the gain of the “peaks”, that is, of the parts that accumulate greater sound energy.

However, several very interesting objectives are achieved:

The resulting sound sounds more balanced and compensated, there is not much difference between the soft and strong parts of the signal
We gain headroom space (the difference between the nominal level and the saturation point) and we can increase the overall volume of the signal a little more without “touching the ceiling” (the peaks were attenuated). As a consequence, the parts that previously sounded with little force will now be heard better.
It will allow to integrate the signal with greater ease and clarity in the general mix.

Standardization

Normalization is an atypical dynamic process, very different from compression, limitation, expansion or noise reduction:

It does not reduce the relative dynamic range of the audio signal.
It is not applied in “real time”, or at the moment, but it is a process that is carried out “a posteriori”, on the previously recorded material.
The process to normalize audio is summarized as follows:

Normalization analyzes the material and detects its highest volume peak. It then increases its gain to the maximum possible without exceeding the reference level (from which distortion would occur).
Taking as reference the same proportion of increase applied in the previous step increases the level of the rest.
The signal, in general, will sound with a greater volume. The maximum volume level that we can reach depends on the limit marked by the highest peak.

The quality of YouTube videos leaves much to be desired: they need an update

When we watch a video on the platform, we can usually appreciate that, despite finding videos in 1080p resolution, the compression applied by the platform is too aggressive. This causes the final quality of the video we are watching to differ greatly from that of the original file. The codec that YouTube uses is H.264 / MPEG-4 AVC, using various profiles or “levels” that specify the maximum resolution, frames per second and maximum bitrate of each quality.

We have analyzed a few videos, and we have taken a fairly representative one that is available on both Vimeo and YouTube to see how both platforms compress the videos. In addition, we have seen the maximum and minimum bitrate that each video can have according to the YouTube Help page for each resolution. The audio, as we discussed in summer, reaches 128 Kbps, leaving 320 Kbps only for YouTube Red users.

What sound quality (bitrate) do YouTube videos have?

The bitrate for 1080p videos is too low: 4K is the way to go
The bitrates that YouTube says it assigns to each video are the following, with the profile level in parentheses:

4K / 2160p
60 fps: Between 20,000 and 51,000 Kbps (L5.2)
30 fps: Between 13,000 and 34,000 Kbps (L5.1)
1440p
60 fps: Between 9,000 and 18,000 Kbps (L5.1)
30 fps: Between 6,000 and 13,000 Kbps (L5.0)
1080p
60 fps: Between 4,500 and 9,000 Kbps (L4.2)
30 fps: Between 3,000 and 6,000 Kbps (L4.1)
720p
60 fps: Between 2,250 and 6,000 Kbps.
30 fps: Between 1,500 and 4,000 Kbps.
480p: Between 500 and 2,000 Kbps.
360p: Between 400 and 1,000 Kbps.
240p: Between 300 and 700 Kbps.

In our tests, the bitrates we obtained for the previous video were the following:

4K at 30 fps
Vimeo: 19.4 Mbps (file size: 943 MB) (capture)
YouTube: 17 Mbps (file size: 821 MB) (capture)
1080p at 30 fps
Vimeo: 4.31 Mbps (file size: 219 MB) (capture)
YouTube: 3.2 Mbps (file size: 160 MB) (capture)
vimeo vs youtube compression

As we see, Vimeo files occupy more not only because of the lower compression of the videos, whose quality is superior to the naked eye, but that Vimeo’s sound quality doubles that of YouTube, since it reaches 256 Kbps by 128 Kbps from YouTube. So that you can see the difference in image quality, you can open the same New Zealand Ascending video on YouTube and Vimeo, and we have also left four captures at the same moment of each video so you can save them and see comfortably the video difference.

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.

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
IN XATAKA
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)