MP3 Compression: Bitrate and Audio Quality Tradeoffs
MP3 CompressionMP3 Compression
MP3 Compression
MP3 is a popular format for digital audio. It is a lossy format, which means that some of the original audio data is discarded in order to reduce the file size. The amount of data that is discarded is determined by the bitrate, which is a measure of the amount of data per second. A higher bitrate results in a higher quality audio file, but also a larger file size.
How MP3 Compression Works
MP3 compression works by using a technique called psychoacoustic coding. Psychoacoustic coding takes advantage of the fact that the human ear is not equally sensitive to all frequencies. For example, we can hear lower frequencies better than higher frequencies. Psychoacoustic coding uses this information to discard frequencies that are not as important to human hearing.
Bitrate and Audio Quality
The bitrate is the most important factor that determines the audio quality of an MP3 file. A higher bitrate results in a higher quality audio file, but also a larger file size. For example, a 128 kbps MP3 file will sound better than a 64 kbps MP3 file, but the 128 kbps file will be twice as large.
Choosing the Right Bitrate
The best bitrate to choose depends on how you plan to use the MP3 file. If you are going to listen to the file on a high-quality audio system, then you will want to use a high bitrate. If you are going to listen to the file on a portable device, then you may want to use a lower bitrate to save space.
Other Factors That Affect Audio Quality
In addition to the bitrate, there are other factors that can affect the audio quality of an MP3 file. These factors include the sampling rate, the bit depth, and the encoder used.
The sampling rate is the number of times per second that the audio signal is sampled. A higher sampling rate results in a higher quality audio file.
The bit depth is the number of bits used to represent each sample. A higher bit depth results in a higher quality audio file.
The encoder is the software that is used to compress the audio file. Different encoders use different algorithms, and some encoders produce better quality audio files than others.
Conclusion
MP3 compression is a popular and effective way to reduce the file size of digital audio files. By using a high bitrate, you can ensure that the audio quality of your MP3 files is good enough for your needs.
Frequently Asked Questions
What is the difference between MP3 and lossless audio formats?
MP3 is a lossy format, which means that some of the original audio data is discarded in order to reduce the file size. Lossless audio formats, such as FLAC and WAV, do not discard any data, so they retain the original audio quality. However, lossless audio files are much larger than MP3 files.
What is the best bitrate for MP3 files?
The best bitrate for MP3 files depends on how you plan to use them. If you are going to listen to the files on a high-quality audio system, then you will want to use a high bitrate. If you are going to listen to the files on a portable device, then you may want to use a lower bitrate to save space.
What are some tips for improving the audio quality of MP3 files?
There are a few things you can do to improve the audio quality of MP3 files. First, use a high bitrate. Second, use a high-quality encoder. Third, avoid using compression plugins or software that may degrade the audio quality.
What are some common problems with MP3 files?
Some common problems with MP3 files include:
Crackling or popping noises
Loss of high-frequency sounds
Muffled or distorted sound
These problems can be caused by a number of factors, including:
Low bitrate
Poor quality encoder
Damage to the file
If you are experiencing problems with your MP3 files, try using a different encoder or a higher bitrate. You can also try repairing the file using a file repair utility.
As someone who has been working with audio files for years, I can tell you that MP3 compression is one of the most important topics in the industry. It’s a technique that has revolutionized the way we listen to music, and it’s something that every audio enthusiast should understand.
How MP3 Compression Works
At its core, MP3 compression is all about removing data that the human ear can’t hear. This is done by analyzing the audio file and identifying sounds that are outside of the range of human hearing. These sounds are then removed, resulting in a smaller file size without any noticeable loss in quality.
As the book “The Art of Digital Audio” explains, “MP3 compression is based on the psychoacoustic principle that the human ear cannot discern certain sounds that are masked by other sounds.” This means that by removing these masked sounds, we can significantly reduce the file size of an audio file without sacrificing quality.
The Benefits of MP3 Compression
One of the biggest benefits of MP3 compression is the ability to store more music on your device. Before MP3 compression, most audio files were too large to be stored on a computer or portable music player. With MP3 compression, you can store hundreds or even thousands of songs on a single device.
Another benefit of MP3 compression is the ability to stream music over the internet. Without MP3 compression, streaming music would be nearly impossible due to the large file sizes of most audio files. MP3 compression allows for fast and efficient streaming, making it possible to listen to music on the go.
The Future of MP3 Compression
While MP3 compression has been around for decades, it’s still an evolving technology. As new audio formats and compression techniques are developed, we can expect MP3 compression to continue to improve.
One area where MP3 compression is likely to see significant growth is in the field of virtual and augmented reality. As these technologies become more advanced, the need for high-quality, low-latency audio will become increasingly important. MP3 compression is likely to play a key role in meeting this need.
MP3 Compression vs. Other Audio Formats
When it comes to audio formats, there are a lot of options out there. From WAV to FLAC to AAC, each format has its own strengths and weaknesses. So how does MP3 compression stack up against the competition?
MP3 Compression vs. WAV
WAV is a lossless audio format that is often used in professional audio production. While WAV files offer the highest possible audio quality, they also come with a large file size. This makes them impractical for most consumer applications.
MP3 compression, on the other hand, offers a good balance between file size and audio quality. While MP3 files are not as high-quality as WAV files, they are much smaller and more practical for everyday use.
MP3 Compression vs. FLAC
FLAC is another lossless audio format that is often used by audiophiles. Like WAV, FLAC files offer high-quality audio, but they also come with a large file size.
While FLAC files are great for archiving and preserving high-quality audio, they are not practical for everyday use. MP3 compression, on the other hand, offers a good compromise between file size and audio quality, making it the ideal format for most consumer applications.
MP3 Compression vs. AAC
AAC is a newer audio format that was developed by Apple. Like MP3 compression, AAC is a lossy format that offers a good balance between file size and audio quality.
While AAC files are generally smaller than MP3 files, they also tend to offer slightly better audio quality. However, because AAC is a proprietary format, it is not as widely supported as MP3 compression.
The Science Behind MP3 Compression
At its core, MP3 compression is all about the science of sound. By understanding how sound works and how the human ear perceives it, we can create audio files that are smaller and more efficient without sacrificing quality.
The Psychoacoustic Model
The key to MP3 compression is the psychoacoustic model. This model is based on the fact that the human ear is not equally sensitive to all frequencies of sound. In fact, our ears are much more sensitive to sounds in the midrange frequencies than they are to sounds in the high or low frequencies.
By taking advantage of this fact, MP3 compression is able to remove sounds that are outside of the range of human hearing. This results in a smaller file size without any noticeable loss in quality.
The Bitrate
Another important factor in MP3 compression is the bitrate. The bitrate is the amount of data that is used to represent each second of audio. A higher bitrate means that more data is being used, which results in a higher-quality audio file.
However, higher bitrates also mean larger file sizes. This is why most MP3 files are encoded at a bitrate of 128 kbps or 192 kbps. These bitrates offer a good balance between file size and audio quality.
The Future of MP3 Compression
As technology continues to evolve, we can expect MP3 compression to continue to improve. New compression techniques and audio formats are likely to emerge, offering even better audio quality and smaller file sizes.
However, even as new technologies emerge, MP3 compression is likely to remain a key part of the audio industry. Its ability to offer high-quality audio in a small file size makes it the ideal format for most consumer applications.
MP3 Compression Techniques
There are a number of different techniques that can be used to compress MP3 files. Each technique has its own strengths and weaknesses, and the best technique to use will depend on the specific needs of the user.
Constant Bitrate Encoding
Constant bitrate encoding is the simplest and most common technique used to compress MP3 files. With constant bitrate encoding, the bitrate is kept constant throughout the entire audio file.
While constant bitrate encoding is easy to implement, it can result in larger file sizes than other techniques. This is because the bitrate is not adjusted to match the complexity of the audio.
Variable Bitrate Encoding
Variable bitrate encoding is a more advanced technique that adjusts the bitrate based on the complexity of the audio. This means that more data is used to represent complex sounds, while less data is used to represent simpler sounds.
Variable bitrate encoding can result in smaller file sizes than constant bitrate encoding, while still maintaining high audio quality. However, it can be more difficult to implement than constant bitrate encoding.
Joint Stereo Encoding
Joint stereo encoding is a technique that takes advantage of the fact that most audio files are recorded in stereo. With joint stereo encoding, the left and right channels of the audio are analyzed separately, and the data is compressed based on the similarities between the two channels.
This technique can result in smaller file sizes than other techniques, while still maintaining high audio quality. However, it can also result in some loss of stereo separation.
The Benefits of MP3 Compression
As someone who has been working with audio files for years, I can tell you that MP3 compression is one of the most important topics in the industry. It’s a technique that has revolutionized the way we listen to music, and it’s something that every audio enthusiast should understand.
Storing More Music
One of the biggest benefits of MP3 compression is the ability to store more music on your device. Before MP3 compression, most audio files were too large to be stored on a computer or portable music player. With MP3 compression, you can store hundreds or even thousands of songs on a single device.
This is something that I’ve personally experienced. As someone who loves music, I used to have to carry around a large collection of CDs or cassette tapes. With MP3 compression, I can now carry my entire music collection in my pocket.
Streaming Music
Another benefit of MP3 compression is the ability to stream music over the internet. Without MP3 compression, streaming music would be nearly impossible due to the large file sizes of most audio files. MP3 compression allows for fast and efficient streaming, making it possible to listen to music on the go.
This is something that I’ve personally experienced as well. As someone who travels frequently, I rely on streaming music services to keep me entertained on long flights or train rides. Without MP3 compression, this would not be possible.
The Future of MP3 Compression
While MP3 compression has been around for decades, it’s still an evolving technology. As new audio formats and compression techniques are developed, we can expect MP3 compression to continue to improve.
One area where MP3 compression is likely to see significant growth is in the field of virtual and augmented reality. As these technologies become more advanced, the need for high-quality, low-latency audio will become increasingly important. MP3 compression is likely to play a key role in meeting this need.
MP3 Compression for Beginners
If you’re new to the world of audio files, MP3 compression can seem like a daunting topic. However, with a little bit of knowledge, you can quickly become an expert.
Choosing the Right Bitrate
One of the most important things to consider when compressing MP3 files is the bitrate. The bitrate is the amount of data that is used to represent each second of audio. A higher bitrate means that more data is being used, which results in a higher-quality audio file.
However, higher bitrates also mean larger file sizes. This is why most MP3 files are encoded at a bitrate of 128 kbps or 192 kbps. These bitrates offer a good balance between file size and audio quality.
Using the Right Software
Another important factor to consider when compressing MP3 files is the software that you use. While there are many different programs available for compressing audio files, not all of them are created equal.
If you’re looking for a reliable and easy-to-use program for compressing MP3 files, I would recommend checking out MP4Gain. This program offers a wide range of compression options, making it easy to find the right settings for your needs.
Conclusion
In conclusion, MP3 compression is an important topic for anyone who works with audio files. Whether you’re a professional audio engineer or just someone who loves music, understanding MP3 compression is essential.
By taking advantage of the techniques and technologies available for MP3 compression, you can store more music on your device, stream music over the internet, and enjoy high-quality audio without sacrificing file size. So if you haven’t already, I would encourage you to start exploring the world of MP3 compression today.
1. The common name for portable MP3 player. a portable player
used to play music in MP3 format (now compatible with wma, wav and other formats).
Portable MP3 Player was originally developed by Korean Wenguang Su and Huang Dingxia (Moon & Hwang) Invented in 1997 and applied for related patents Detailed explanation of the
technology development of the MP3 format Format, which is designed to drastically reduce the amount of audio data, while for most users the playback quality is not appreciably degraded from the original uncompressed audio. It was invented and standardized in 1991 by a group of engineers from the Fraunhofer-Gesellschaft research organization in Erlangen, Germany. MPEG-1 Audio Layer 3, often referred to as MP3, is one of the most popular lossy compression and digital audio encoding formats today. There is no noticeable drop in sound quality compared to the original uncompressed audio. It was invented and standardized in 1991 by a group of engineers from the Fraunhofer-Gesellschaft research organization in Erlangen, Germany. General information MP3 is a data compression format. It discards pulse code modulation (PCM) audio data that is not important to the human ear (similar to how JPEG is lossy image compression), resulting in a much smaller file size. Various techniques are used in MP3, including psychoacoustics, to determine which parts of the audio can be discarded. MP3 audio can be compressed at different bit rates, providing a variety of trade-offs between data size and sound quality. The MP3 format uses a hybrid conversion mechanism to convert the time domain signal to the frequency domain signal: * 32-Band Polyphase Integrating Filter (PQF)
* Modified 36 or 12 tap discrete cosine filter (
MDCT); each subband size is independently selectable between 0…1 and 2…31 However, due to the unprecedented popularity of MP3, the success of any other format is currently unlikely. MP3 not only has extensive client software support, but also has a lot of hardware support, such as portable media players (referring to MP3 players), DVD and CD players.
The development of
MPEG-1 Audio Layer 2 encoding started with the German Deutsche Forschungs- und Versuchsanstalt für Luft- und Raumfahrt (later called Deutsches Zentrum für Luft- und Raumfahrt, German Space Center) Digital Audio Broadcasting (DAB) managed by Egon Meier-Project Engelen . This project is funded by the European Union as a EUREKA research project, and its name is commonly known as EU-147. The study period for EU-147 was from 1987 to 1994.
By 1991 there were already two proposals: Musicam (called Layer 2) and ASPEC (Adaptive Spectrum Sensing Entropy Coding). The Musicam method proposed by Philips of the Netherlands, CCETT of France, and the Institut für Rundfunktechnik of Germany was chosen due to its simplicity, robustness against errors, and lower computational effort in high-quality compression. The Musicam format based on subband coding is a key factor in determining the MPEG audio compression format (sampling rate, frame structure, data header, sample points per frame). This technology and its design philosophy are fully integrated into the definition of ISO MPEG Audio Layer I, II and later Layer III (MP3) formats. The standard was developed by Leon van de Kerkhof (Layer I) and Gerhard Stoll (Layer II) under the auspices of Prof. Mussmann (University of Hannover).
A working group consisting of Leon Van de Kerkhof from the Netherlands, Gerhard Stoll from Germany, Yves-François Dehery from France, and Karlheinz Brandenburg from Germany absorbed design ideas from Musicam and ASPEC and added their own design ideas to develop MP3 , which can play MP2. Sound quality from 192kbit/s to 128kbit/s.
All of these algorithms eventually became part of the first group of MPEG standards, MPEG-1, in 1992, resulting in the ISO/IEC 11172-3 international standard published in 1993. Further work on MPEG audio eventually became part of the MPEG-2 standard, a second group of MPEG standards developed in 1994, officially known as ISO/IEC 13818-3, first published in 1995.
The compression efficiency of an encoder is usually defined by the bit rate, since the compression rate depends on the number of bits.
The MP3 Encoder is that program that analyzes the uncompressed digital file (for example, a Wav file) and transforms it into an MP3 file.
The audio signal is filtered and divided into 576 areas (called subbands) through a process that uses DCT (Discrete Cosine Transformation) and manages to eliminate all unnecessary frequencies. The human ear, as already stated, perceives sounds only beyond a certain threshold so that all the audio below is not encoded.
At this point, the resulting signal passes through the psychoacoustic model in which the masking thresholds of which we have spoken previously are identified. This is done using the discrete Fourier transform (DFT, Discrete Fourier Transform).
During the masking of the 576 subbands, the frequencies to be masked are determined and therefore can be removed.
After masking, the defined Stereo Ensemble process is applied. Below a certain frequency, the ear cannot perceive the spatial position of sounds, so they can be recorded on a single channel (therefore in mono format) with significant space savings.
Once the file is ready, the data is further analyzed and compressed using Hufmann encoding which allows for a data reduction (without loss of information) of approximately 20%.
At this point, after all the data has been collected, the encoder proceeds to create the bit stream that will form the final MP3 file.
Compression criteria
To perform such compression, the MP3 format is based on a simple concept: filter a digital musical piece and eliminate all unnecessary information, thus reducing space.
The human ear is an almost perfect instrument but it also has its limits. The human ear pass band extends from 20 Hz to 20,000 Hz, but is much more sensitive to those in the mid-range, 700 to 6,000 Hz, where most of the information is concentrated.
The study of auditory perception is a matter of psychoacoustics that mainly analyzes 2 factors that are later used in MP3 encoding:
Auditory perception
In the area of sounds, only a few can be heard by the human ear. The following figure shows these areas that represent the different sound frequencies. Only those in the white area are audible from our ear.
Masking
Masking is nothing more than the superposition of weak sounds with loud sounds. It almost always happens that the sounds of different instruments overlap each other. In cases where the loudest sound completely covers the lowest, there is a so-called masking. In MP3 files, masking allows you to remove the information from the weakest sounds, which, however, because they are not perceived by the ear, are virtually irrelevant.
Many people do not have a clear idea that in general most audio formats compress music.
In fact, thanks to that compression, the mp3 became so popular. It is not because it sounded better, as an uncle of mine creates … but because it allows you to store much more music on a USB stick, on a CD, etc. even when it sacrifices a bit of quality.
That is to say, technically the mp3 sounds worse than the original raw format like a wav.
But handling wavs is usually unmanageable, unless you are an audio professional.
But, going back to talk about my uncle, who wants to listen to Frank Sinatra in his car, using the mp3 is much more friendly. Even because it has a metadata (artist name, track, lyrics, etc.) and also, if a good bitrate is used (160 m or more) it is almost imperceptible to most of the people the difference between an mp3 and a wav .
Experiments have been carried out in famous universities that managed to show that not even the people who claimed to have an auditory training (for being musicians, djs, etc.) managed to distinguish in most cases a 192-bit mp3 from the original wav.
This explains why mp3 is still king, even before the appearance of FLAC for example, that it is free (without patents) and that it has a much better quality.
But, again to mention my uncle, he believes that FLAC is a colorful cereal … and he still says that he really likes that cereal for breakfast !!
Compression
But then, the fame of the mp3 is due exclusively to its ability to save space?
Yes.
And how does the music compress the mp3?
Follow several methods. Here I will tell you superficially and only by way of introduction how it manages to save space.
The first tactic is almost logical. As the human ear only listens to a part of the sound spectrum, the mp3 erases everything that is outside that spectrum, thus saving a lot of space.
Then it uses another well-known mechanism of the human ear (if you look at the mp3 it is based on the ability to perceive the human ear … THAT’S why people DO NOT manage to perceive a good mp3 from the original wav !!).
That mechanism is called masking, and it’s about the following. If there are two or more sounds at nearby frequencies and one of them suddenly sounds loud enough, the ear will NOT hear the other sounds that are lower in volume at nearby frequencies. So the mp3 uses that acoustic principle of the human ear and gets rid of those other sounds with which it again removes information.
And removing information means SAVING SPACE.
And if you finally use some mechanism to compress (type .zip or type .rar), a great saving of space is achieved.
For example, let’s imagine (it is a false example, but it illustrates what I mean), if we had this string in the audio “xxxxxxxxxxxxxxxxxxxx”, one way to compress it would be to say that there are 20 x, instead of writing 10 x, note :
xxxxxxxxxxxxxxxxxxxx
20x
Which takes up more space and which takes less?
Both strings of signs or characters say the same thing, there are 20 x, but it is shorter to write it as 20x, than to write “xxxxxxxxxxxxxxxxxxxx”
Onbiamente in all loss of information, there is a loss of quality. But the same thing happens with colors.
They say there are computers capable of handling not how many millions of different colors … it would be smart to ask how many different colors the human eye can perceive.
So, there will always be a purist who says that the mp3 loses quality … but it would be good to see if her ear can distinguish it. Music is made to be heard by human ears, with its limitations.
Well, in short, this is how you make an mp3 to save space. I will send a copy of this article to my uncle.
First of all, remember that “MP3” is short for the term “Audio MPEG-1/2 Layer 3 Compression”, which is an audio data encoding format that allows you to divide the weight of a computer file by more than ten.
The word MP3 also refers by extension to portable audio players that play the audio in MP3 format.
The main role of the MP3 format is to compress music so that it is lighter (to store more in our player) without the listener noticing the differences.
Therefore, we will remove everything considered “superfluous” from the audio signal, but this is the whole controversy: what is really superfluous or unimportant or superfluous in the sound to be encoded?
Some people who oppose this method of storage speak of signal mutilation. Others describe this operation with a nice comparison: “The more potatoes you put in a pan? It’s simple, we make it puree!
In fact it is not that simple, the compression method is much more complex than you think.
Music compression
To make MP3 music lighter, it is compressed, but without the user hearing or perceiving the difference. The principle is to eliminate sounds that are inaudible to the human ear, such as ultrasound (treble) or infrasound (bass). But be careful, this “light” music (12 times less heavy than the standard format music) should remain “of good quality” to satisfy listeners.
To achieve this, MP3 does not encode all the data necessary for full sound reproduction, but only what is perceived by the human ear. This is how we achieve what we call the “skinny”.
1st phase: the first skimming takes place in all sounds that are not perceived by the ear. They are simply removed.
Compression allows the spectrometric components of an audio signal to be analyzed and a psychoacoustic model applied to them, so that only “audible” sounds are preserved.
The human ear can distinguish sounds on average between 0.02 kHz and 20 kHz, knowing that the sensitivity is maximum for frequencies between 2 and 5 kHz, according to a curve given by Fletcher and Munson’s law. Therefore, this first compression phase consists of determining the sounds we do not hear and eliminating them, therefore it is a destructive compression, that is, with loss of information.
2nd step: Next we will more accurately encode the sounds to which the ear is most sensitive (those between 2 and 5 kilohertz). The rest of the sounds contain the frequencies that are less perceived by the ear and will be encoded with less precision. Then they will be of lesser quality, and, that is the goal, they will take up less space because they are almost undetected. The listener will not notice this “degradation” of the original sound because these are frequencies to which the ear is not sensitive.
In this same phase, a second treatment is added: dynamic compression. Dynamic compression consists of raising the weak levels and the low levels to keep them lower, to erase the contrasts the music has.
These two stages will lighten music without altering the perception of sound.
Sound masking
After heavily compressing the sound, the MP3 continues using the masking phenomenon. When a sound reaches a certain intensity, it masks the sounds with the lowest intensities closest to it. The ear does not detect the weakest sound and MP3 will therefore easily remove these so-called “masked” sounds.
If you look at the sun and a bird goes along its axis, you will not see it because the light from the sun is too important. It is the same in acoustics. If there are loud sounds, you cannot hear the weakest. For example, if a sound of 80 dB with a frequency of 1000 Hz is followed by a sound of 20 dB and has the same frequency, formatting in MP3 will preserve the sound of 80 dB and hide the others
Therefore, the blue sound is masked by the black sound.
The danger of this size
The MP3 format poses two kinds of danger to our hearing: – The first is that it encourages the listener to increase the volume of the sound from his player.
Second, our ears are getting used to this type of sound, which we could describe as “dematerialized,” and it is getting slow.
Special hearing disorders related to MP3 formatting. The human ear is used to perceiving strong dynamic contrasts and is not made for compressed MP3 format signals. In fact, the compression of the music will act as an optical illusion. If we listen to this compressed music, we will unconsciously
The Moving Picture Expert Group 1/2 Audio Layer 3, the audio compression format that has changed the music world forever, has officially disappeared, at least for the Fraunhofer Institute for Integrated Circuits.
The German institution that was working on the format and that funded its development in the late 1980s recently announced his death at the end of the licensing program for some registered patents related to the MP3 format. According to the official statement, the reason is: “More efficient audio codecs are available today.”
Despite the enormous popularity that was gained in about 30 years, the MP3 format was surpassed by the formats of the Aac family used by modern multimedia services such as streaming or TV and radio broadcasts, and soon also by the extraordinary Mpeg-H .
The new formats guarantee better audio quality and a lower bit rate, hence a heavier audio file with the same quality compared to MP3 and offer greater functionality. According to Bernhard Grill, director of the institute, AAC is today the de facto standard for downloading music and videos on smartphones. If MP3 was the symbol of a revolution, today nobody cares about the name of the institute format in which an audio file is encoded, only “sounds” good.
Let’s return to the history of MP3 thanks to these 10 “Maybe not everyone knows”:
1) An idea from the late 19th century. Studies of an algorithm that reduced the weight of audio files in order to transmit them more easily through very slow networks in the late 1980s relate to the concept of “auditory masking” or the phenomenon by which the perception of a Presence of another sound masked.
The first observations on this phenomenon were made in 1894 by the American physicist Alfred M. Mayer.
2) Hello, I’m MP3 The father of MP3 can be seen as a codec for the psychoacoustic masking introduced in 1979. The aim was to create an audio format for telephone messages that does not “weigh” the lines. The basic idea that was later taken up when creating the MP3 format is that the human ear cannot perceive some audio frequencies.
For this reason, it is sufficient to eliminate these frequencies in order to reduce the weight of an audio file while maintaining an apparent quality. In fact, the basic assumption has proven to be wrong in recent years. Read also: The virtual reality changes the music and fights the secondary ticket sales. And Keith Richards teaches you how to play
3) An Italian is listening Leonardo Chiariglione Mp3 seen at “The Visible City” at the Turin International Book Fair 2012. Valerio Pennicino / Getty Images Leonar do Chiariglione, an engineer from Almese, Turin, is considered one of the fathers of the MP3 format as the founder of the working group MPEG (Moving Pictures Expert Group) in 1988, which developed several audio / video compression formats in world standards.
In December 1988, the MPEG group launched a public request to develop an audio compression algorithm. Because of their similarity, the 14 algorithms obtained were divided into four main categories.
4. Brandenburg uses it. Suzanne Vega. Carlos Alvarez / Getty Images It is the thesis of the doctoral student Karlheinz Brandenburg that was discussed in 1989 at the German University of Erlangen-Nuremberg to illustrate the specifications of the MP3 format in detail.
The first song encoded in the new format was Tom’s Diner by singer Suzanne Vega. Brandenburg coded it countless times to understand whether the omitted frequencies had affected the sound of Vegas’ voice. Also Read: 10 Songs To Keep Fit: Here’s The Spotify Playlist
5. Light weights With the introduction of the MP3 format, the weight of a song was reduced to approximately 4 MB compared to ten MB of an audio file on a CD. It was a revolution because it was finally possible to transmit the songs over the Internet, although the transmission speed was still tied to the limits of the 56 kbit / s modems or even to a lower download speed.
6. The hacker in a coat In the summer of 1996, the NetFrack user published a message in the Affinity online fanzine that he had found a way to reduce the size of audio files thanks to a new compression format and thus hard drives. from that time on they could have contained many more songs. Subsequently, NetFrack founded the online group Compress Da Audio, which only distributed music files, and made Metallica’s song Doesi It Sleeps available in MP3 format.
August 10, 1996 is the official date of birth of music piracy.
7. The beginning of the revolution. In 1997 NullSoft created Winamp, the first software to encode audio files in MP3 format. The following year, Diamond Multimedia introduced the first portable MP3 player, the Rio PMPm300, which could hardly hold the contents of an album, used a pencil battery, and cost around $ 200. In 1999 it was Shawn Fanning and Sean Parker. Years later, when Mark Zuckerberg advised to remove “The” from the Facebook name, Napster founded it.
8. A useful service. Despite about $ 35 million in claims and considered utterly evil, Radioheads Kid A wouldn’t have had the success it had had without Napster. The group was not yet known worldwide and the record company had not planned to advertise the new album, release or video clips. In October 2000, the album was Radiohead’s first to top the billboard charts, also thanks to the fact that it was released three months before Napster’s official release.
And Thom Yorke said unlike Madonna, Metallica and Dr. Dre, who had filed million dollar lawsuits: “The best thing about Napster is that it instills enthusiasm for music in a way that the music industry has stopped. Hour”.
9. Apple, thank you In 2001, Apple introduced the iPod, the MP3 file player that played a key role in tracking china down to the Cupertino home. Almost 400 million units were sold in around 13 years of life. In 2003, Apple always invented the first paid and legal music download service. Today, 70% of online music is purchased on iTunes, which is an average of approximately 20,000 songs per minute.
10. An announced death. The development of the AAC format, which is now the de facto standard for digital audio, began in 1990, but only understood in 2007 when Apple decided to only make audio files in Aac format with 256 Kbit / s available in iTunes Plus Experts the end. MP3 was close.
To perform such compression, the MP3 format is based on a simple concept: filter a digital piece of music and eliminate all unnecessary information, thus reducing space.
The human ear is an almost perfect instrument but it also has its limits. The human ear pass band extends from 20 Hz to 20,000 Hz, but is much more sensitive to those in the midrange, 700 to 6,000 Hz, where most of the information is concentrated.
The study of auditory perception is a matter of psychoacoustics that mainly analyzes 2 factors that are later used in MP3 encoding:
Mp3 – Auditory perception
In the area of sounds, only a few can be heard by the human ear. The following figure shows these areas that represent the different sound frequencies. Only those in the white area are audible from our ear.
The sounds that the ear perceives are only those of the white areas
Masking
Masking is nothing more than the superposition of weak sounds with loud sounds. It almost always happens that the sounds of different instruments overlap each other. In cases where the loudest sound completely covers the lowest, there is a so-called masking. In MP3 files, masking allows you to remove the information from the weakest sounds, which, however, because they are not perceived by the ear, are virtually irrelevant.
MP3 – The Name
The name MP3 comes from the MPEG standard, which means Moving Picture Experts Group. This group was created specifically for the development of systems and standards used in video compression. DVD movies and satellite broadcasts (DBS) use the MPEG standard to efficiently compress video information.
MPEG compression includes a subsystem for sound compression with three different compression levels (layers) depending on the quality of the information. Layer-3 is the one used for the MP3 standard, which stands for MPEG Layer-3.
MP3 – Step by step compression
The MP3 Encoder is that program that analyzes the uncompressed digital file (for example, a Wav file) and transforms it into an MP3 file.
The audio signal is filtered and divided into 576 areas (called subbands) through a process that uses DCT (Discrete Cosine Transformation) and manages to eliminate all unnecessary frequencies. The human ear, as already said, perceives sounds only beyond a certain threshold so that all the audio below is not encoded.
At this point, the resulting signal is passed through the psychoacoustic model in which the masking thresholds of which we spoke earlier are identified. This is done using Discrete Fourier Transformation (DFT).
During the masking of the 576 subbands, the frequencies to be masked are determined and therefore can be removed.
After masking, the defined Stereo Ensemble process is applied. Below a certain frequency, the ear cannot perceive the spatial position of the sounds, so they can be recorded on a single channel (therefore, in mono format) with significant space savings.
Once the file is ready, the data is re-analyzed and compressed using Hufmann encoding which enables a data reduction (without loss of information) of approximately 20%.
At this point, after all the data has been collected, the encoder proceeds to create the bit stream that will form the final MP3 file.
Sound is a continuous wave that propagates through air or other media, formed by pressure differences, so that it can be detected by measuring the pressure level at a point. Sound waves have the proper and studyable characteristics of waves in general, such as reflection, refraction and diffraction.
To the Being a continuous wave, a digitization process is required to represent it as a series of numbers. Currently, most of the operations performed on sound signals are digital, since both storage and
Processing and transmitting the signal in digital form offers very significant advantages over analog methods. Digital technology is more advanced and offers greater possibilities, less sensitivity to transmission noise and the ability to include error protection codes, as well as encryption. With the appropriate decoding mechanisms, moreover, they can be processed simultaneously signals of different types transmitted by the same channel. The main disadvantage of the digital signal is that it requires a much greater bandwidth than that of the analog signal, hence an exhaustive study is carried out regarding data compression, some of whose techniques will be the center of our study.
Digitalization of the audio
The digitization process consists of two phases: sampling and quantization. At sampling divides the time axis into segments
discrete: the sampling frequency will be the inverse
the time between a measurement and the
following. At this time the
quantization, which, in its simplest form,
it simply consists of measuring the value of the signal
in breadth and save it.
Nyquist’s theorem
Nyquist’s theorem ensures that the frequency required to sample a signal that has its highest components at a given frequency f is at least 2f. Therefore, being the upper range of human hearing around 20 Khz, the frequency that guarantees adequate sampling for any audible sound will be around 40 Khz.
Specifically, to obtain high quality sound, frequencies of 44’1 Khz are used,
in the case of CD, for example, and up to 48 Khz, in the case of DAT. Other typical values are submultiples of the first, 22 and 11 Khz.
Depending on the nature of the application, of course, the appropriate frequencies can be much lower, such that the voice process is usually performed at a frequency between 6 and
20 Khz. or even less. Regarding quantization, it is evident that the more bits used for the division of the amplitude axis, the “finer” the partition will be and therefore the less error when attributing a specific amplitude to the sound at each moment.
For example, 8 bits offer 256 levels of quantization and 16,65536. The dynamic range of human hearing is about 100 dB. The axis division can be carried out at equal intervals or according to a specific density function, seeking more resolution in certain sections if the signal in question has more components in
certain zone of intensity, as we will see in the coding techniques.
The complete process is usually called PCM (Pulse Code Modulation) and we will refer to it hereinafter. It has been described in a very simplistic way, mainly because it is widely treated and is well known, being
another the field of study of this work. However, we will go into detail at any time that is necessary for the development of the exhibition.
Coding and Compression.
Before describing coding and compression systems, we must pause in a brief analysis of human auditory perception, to understand why a significant amount of the information provided by PCM can be discarded.
The heart of the matter, as far as we are concerned, is based on a phenomenon known as masking.
The human ear perceives a frequency range between 20 Hz. And 20 Khz.
Firstly, the sensitivity is greater in the area around 2-4 Khz., So that the sound is more difficult to hear the closer to the ends of the scale.
Second is masking, the properties of which are used extensively by the most interesting algorithms: when the component at a certain frequency of a signal has high energy, the ear cannot perceive lower energy components at close frequencies, both lower and higher.
At a certain distance from the masking frequency, the effect is reduced so much that it is negligible; the range of frequencies in which the phenomenon occurs is called the critical band.
The components that belong to the same critical band influence each other and do not affect nor are affected by those that appear outside it. The width of the critical band is different according to the
frequency in which we are located and is given by certain data that shows that it is greater with frequency.
It should be noted that these data are obtained by psychoacoustic experiments, which are carried out with experts trained in
sound perception, giving rise to psychoacoustic models with their impressions.
This we have described is the so-called simultaneous or frequency masking.
There is also the so-called asynchronous or time masking, as well as other phenomena of hearing that are not relevant in this point. For now, let’s focus on the idea that certain signal frequency components support higher noise than we would generally consider to be tolerable, and therefore require fewer bits to be encoded if the encoder is endowed.
of the right algorithms to solve masks.
Digitizing the signal using PCM is the simplest form of signal encoding, and is used by both CDs and DAT systems. Like still digitizing, it adds noise to the signal, generally undesirable. As we have seen, the fewer bits used in sampling and quantization, the greater the error in
accept discrete values for the continuous signal, that is, the higher the noise.
To avoid that the noise reaches an excessive level, it is necessary to use a large number of bits, so that at 44.1 Khz. and using 16 bits to quantize the signal, one of the two channels on a CD produces more than 700 kilobits per second (kbps). As we will see,
Much of this information is unnecessary and takes up bandwidth that could be freed, at the cost of increasing the complexity of the decoder system and incurring some loss of quality.
The compromise between bandwidth, complexity and quality
it is the one that produces the different market standards and will form the essential part of our study.
MP3 is a data format that gets its name from an algorithm
encoding called MPEG 1 Layer 3, which, in turn, is an audio compression system that allows you to store sound with a quality similar to that of a CD and with a very high compression ratio, on the order of 1:11
In practice, this means that about 11 audio CDs can be recorded on a CD-Rom, that is, approximately 150 songs.
The encoding system that MP3 uses is a loss algorithm. That is, the original sound and the one that we obtain later are not identical.
This is because MP3 takes advantage of the deficiencies of the human ear and eliminates all the information that we are not able to perceive. A multitude of studies of acoustic perception have been carried out, discovering that there are a series of effects that can aid the coding of sound with the aim of reducing as much as possible the amount of useless or redundant information. The most important are: The limits of hearing. Our ear only works with frequencies that go between 20 Hz and 20 Khz
approximately, so the remaining frequencies are disposable.
Masking effect.
It is one that occurs when two signals of similar frequency are
overlap. So we can only perceive the one that
it has more volume and, therefore, the one with a smaller volume is
liable to be removed
Stereo redundancy.
There are redundancies between the tonal and non-tonal components of the sound on the two stereo channels, and furthermore
below a certain frequency the human ear is not capable of
perceive the directionality of the sound, so below these
frequencies it is even possible to encode a single channel together with
complementary information to restore the spatial feeling for the other channel.
To carry out this “loss of information” action, a system called Subband Coding is used, a process by which the signal is broken down into subbands through a filter bank.
These subbands are then compared to the original using a psychoacoustic model that is responsible for determining which bands can be removed and which cannot.
Depending on the quality we want to obtain, more or less will be eliminated
bands. To end the process, the resulting subbands are quantized and encoded, and the final result is compressed using a standard algorithm, thus obtaining the resulting MP3 file. The encoding process is much more complicated than the decoding process, so it takes much longer to encode an MP3 file than to play it.
This perceptual coding algorithm was developed by the company MPEG (Moving Picture Expert Group) in conjunction with the Franunhofer Institute of Technology, and has been standardized as an ISO standard.
