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Conversion of analog sound to digital sound

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Digital sounds and analog sound

With the advance of science and technology, both the transmission and recording of analog sounds and images have undergone major changes in recent years. The introduction of digital techniques allows you to do many more things, with greater advantages and more versatility than with analog technology.

Many of the devices that we know today as digital, first receive or capture the signals in analogue form and then convert them into digital signals. This is the case, for example, of CD and DVD players, the modem used by computers for the reception / transmission of data, digital cameras and video cameras, mobile or cell phones, etc.

To perform the conversion, these devices use, as an intermediate element, a device called analog-digital converter or ADC (Analogic to Digital Converter), which first receives the electrical signals in the form of an analog sine wave (such as the one provided by the microphone) and It then converts them into digital signals, encoded in binary numerical values, that is, in “zeros” and “ones” (0 – 1).

1. Sound or acoustic wave (voice, music, effects, etc.). 2. Microphone 3. Analog sine wave that is <obtained after the microphone converts the sounds into audio-frequency electrical signals. 4. ADC (Analogic to Digital Converter – Digital Analog Converter). 5. Digital signal formed by zeros and <ones (0 – 1), obtained after the analog signal is processed by the ADC. 6. Output of the <digitized audio signal, ready to be recorded.

In digital cameras and video cameras, as well as in scanners, there is a sensor called CCD (Charge Coupled Device) or, failing that, a CMOS sensor (Complementary Metal Oxide Semiconductor – Semiconductor complementary metal oxide ), which are responsible for converting the images they receive into analog electrical signals.

In that case, as with the microphone, an ADC is responsible for converting those analog signals into digital image signals, so that they can be stored as such in a videotape, on the device’s memory card, or in any other Digital storage device, for later viewing.

The reverse conversion, from digital to analog, is strictly necessary, because the analog sound is the only audible, that is, the only one that recognizes our sense of hearing. Similarly, the analog electrical impulses are the only ones capable of moving the cone of a loudspeaker or loudspeaker to reproduce the original sounds again, which cannot be done by the electrical impulses of “1” and “0” of the binary or digital code. Therefore, to make the coding of the digital sounds audible by the loudspeaker (s), it is necessary to convert them back into analog electrical signals, with their corresponding variations in voltages or voltages.

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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

Digital video formats: how to differentiate them

As with text documents, photographs or audios, digital video is available in different formats or extensions.

In this sense, today we find DVD and Blu-Ray, although some of us still keep in an old VHS closet and maybe some Betacam.

But a second meaning or meaning of video formats refers to their encoding, since in digital video, as with a computer program, any file is written in a certain code.

In videos, the code influences image quality, sound quality, whether or not it includes subtitles and, especially, the relationship between quality and file size.

Thus, today we consume digital audiovisual content through physical discs (DVD, Blu-Ray), through streaming and through IPTV (Internet television), but we also handle digital video files, especially for content that we generate ourselves.

Next we will review the most common digital video formats that we can find, what is their origin and what benefits they offer. I apologize in advance for the gibberish of acronyms.

AVI

We start with the most popular format that we will find. Video files with an .AVI extension have their origin in a format that was launched in 1992 and is so popular that most smart TVs, DVD / Blu-Ray players, video game consoles and operating systems play it.

AVI is an acronym for Audio Video Interleave and not many know that it was created by Microsoft as a digital alternative without dependence on a physical format such as the then popular DVD.

Among its advantages, it allows you to include several audio channels and host content generated with different codecs (AC3 or MP3 for audio, DivX or Xvid for video), which can be an advantage but also an inconvenience with which players.

MP4

MP4 or MPEG-4 is one of the most modern formats, launched in 1998 as a standard for playing video and audio in a single digital file.

MPEG stands for Moving Picture Experts Group, the expert group that has established digital audio and video standards and was formed by two international organizations, the ISO (International Organization for Standardization) and the IEC (International Electrotechnical Commission).

In summary, the MPEG and MPEG-2 format were launched in 1993 and 1995 respectively as standards for encoding digital audio and video. To understand each other, any DVD offers its audiovisual content in MPEG-2.

MP4 also supports several audio channels, but has the advantage of allowing more image and sound quality in a less heavy file, as it compresses data better. Apple, for example, opts for this format and derivatives for its iTunes content.

Related to MP4 we can find M4V (video) or M4A (audio).

MKV

The MKV video format is an open format, free to pay rights, and whose full name is Matroska, like traditional Russian dolls.

MKV saw the light at the end of 2002 and has become popular thanks to the fact that within a single MKV file we can store, together with the audio channel, several channels or audio tracks and several subtitle tracks.

Like MP4, it offers very good audio and video quality in a small space. And as a curiosity, the WebM format that allows you to integrate online video via HTML, is inspired by Matroska.

FLV

The FLV or Flash Video format was created by Macromedia, and subsequently acquired by Adobe. This format is usually found as an FLV or SWF extension.

Like the other Flash content, FLV videos are designed for online playback from the browser through Adobe Flash Player.

As we saw in a previous article, Flash will stop developing in 2020, although we still find pages that use it.

MOV

I said before that Apple is currently betting on MP4 (and AAC) to facilitate multimedia content. But its star format for many years was MOV.

MOV, from QuickTime Movie, is also called QuickTime File Format, and today it is still the default format of QuickTime, the macOS video player.

This format can also be found in many digital video cameras, since it offers very good quality

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.

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.