What are MP3 and AAC? Which one should I choose to use?


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What are MP3 and AAC? Which one should I choose to use?

aac VS mp3

Music has passed through discs and CDs, and now more and more people listen to it by putting data on their smartphones or streaming it over the Internet.

AAC vs. MP3

One of the things that comes to mind are the words MP3 and AAC that are attached to music files. What the hell is this? In this article, I will explain these two words!

Lossy compression for MP3 and AAC
Generally speaking, MP3 and AAC are the forms of music converted into digital files. The MP3 and AAC forms have their own characteristics, but what they can say in common is that they are both lossy compressed sound sources.

Lossy compression, that is, a method that reduces the amount of data but is irreversible due to data loss. However, this missing part is mostly a super treble part that is difficult for humans to hear, so it doesn’t mean there is no vocal voice. Are you sure.

MP3 with many compatible devices
MP3 is the official name of MPEG1 Audio / Layer3, and it was born in 1992! It is quite an old format.

At that time, the Internet was completely different. If you say that it was natural to exchange at low speed which is incomparable with the speed at which “communication with smartphone slowed down due to speed limitation”, you can understand the horror.

In such circumstances, how to reduce the file size is important, and thus the compression format called MP3 was born. It can be said that the format sacrifices sound quality in order to reduce file size.

However, even in MP3, if the bit rate (the unit is bps), which is the amount of data per second, is set to 320 kbps, the sound quality will improve a bit. This is the same for AAC, which will be explained later.

MP3 is an old format, but on the contrary, it is a format with a fairly high penetration rate. Almost all compatible devices have the advantage that they can be used in any terminal without any problem.

AAC, the successor to MP3
The official name of AAC is Advanced Audio Coding. Born in 1997 as a successor to MP3. Improvements have been made since then, and the AAC currently in use is the 2006 AAC-HE V2.

Generally, AAC is said to have superior sound quality than MP3 if the compression ratio is the same, and it is a format that is also used for digital terrestrial broadcasting on TV.

Apple, which makes iPhones, is active in using AAC. ITunes Store download sales are AAC / 256kbps, which is the standard when importing a CD with iTunes.
Compared to MP3s, the number of compatible devices is reduced, but it can still be played on most terminals.

Other music file formats
There are also other music file formats. Lossless compressed and Apple Lossless FLACs, and uncompressed AIFF and WAVs are well known. Lossless compression is the opposite of lossy compression and is a format that can be undone. Therefore, the sound does not deteriorate, but the compression ratio is lower than lossy compression. Uncompressed is a format that has the same sound and capacity.

Which is better, MP3 or AAC?
Now that the speed of Internet communication is increasing, it seems unlikely that MP3 and AAC will come into play at first glance. However, there are restrictions on the amount of data communication from mobile phone companies, so you cannot listen to large music files over the Internet.

So I think I should save it to the terminal, but this time the data area of ​​the terminal will be bloated. Due to such circumstances, the MP3 and AAC formats are still active. So which of these two should you choose?

There are various theories, but this is still AAC. Since it is a newcomer, the sound quality is better than lossy compression, and the data capacity can be suppressed. MP3 is an option only if you are using a device that does not support AAC.


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What is the difference between AAC and mp3? Introducing a music file format comparison!

What is the difference between AAC and mp3? Introducing a music file format comparison!

AAC Vs. MP3

One of the music files is “AAC”, but do you know the difference between the two files, the AAC file and the popular mp3 file? This article provides a comparison of the formats and the differences between the two music files, AAC and mp3.

AAC or MP3

What is AAC?
What is the difference between AAC and mp3?

What is AAC?

There are various music file formats, but do you know what kind of file is “AAC”, what is one of the file formats? AAC is an abbreviation for “Advanced Audio Coding” and is one of the irreversible digital audio compression formats. The extensions are “.m4a” and “.m4p”.

This is a standardized file format for higher sound quality and higher compression than MPEG-1 Audio, including mp3. AAC is also used in MPEG-2 and MPEG-4 and is responsible for the audio part of video files in those formats. AAC can have 48 channels of full-bandwidth audio in one stream and can contain 16 channels of low-frequency sound effects, 16 interactive channels, and 16 data streams.

What is the difference between AAC and mp3?

You may be curious to know the difference between AAC files and mp3 files. As mentioned above, AAC files are a standard intended for higher sound quality and higher compression than mp3 files, which is why they are basically positioned as the successor format to mp3. AAC is the new generation and mp3 is the old generation.

Both are lossy compression, that is, data compression is done using a compression method that produces data that is truncated, even though the data compression rate is high. Therefore, if you rip from a music CD and create these files, the sound quality will inevitably deteriorate compared to the original music CD. (Since the audible range that humans cannot hear is cut off, it is difficult to notice if you are not very particular to

respect). The data compression rate for both AAC and mp3 is about 1/10 of the original data, and AAC is slightly better. It feels like the compression rate is high. AAC is the default file format used by Mac products, such as the iPhone, and mp3 is more common on Android smartphones. Also, there are many music players that support mp3 because it has a long history and AAC tends to be less than mp3.

Regarding sound quality, the maximum bit rate for both is “320 kbps”. Although 320 kbps is well below the 1142 kbps for music CDs, the sound quality is high enough considering the human audible range. And if both AAC and mp3 are encoded at 320 kbps, AAC is said to have better sound quality.

Difference between lossless compression and lossy compression.

Difference between lossless compression and lossy compression.

Lossless

Basically, the higher the kbps, the higher the sound quality in the sense that it is closer to the original sound.

lossless vs lossy

Therefore, it is easy to think that it is only necessary to collect sound sources with high kbps.

However, since the amount of data will be large, it must actually be compressed.

That’s where two compression technologies come in.

Lossless compression (lossless compression)
Rossy compression (lossy compression)
About the uncompressed sound source
Let’s explain each compression technology.

We will also look at uncompressed sound sources.

Lossless compression (lossless compression)
Lossless compression has the advantage of retaining a large amount of information by compressing the data without damaging it.

In other words, the tendency is for the kbps value to be higher.

Of course, because it is compressed, the file size can be reduced to about half.

However, the capacity is considerably higher than the Rossie compression.

Lossy compression (lossy compression)
Rossy compression is famous for formats like mp3 and AAC.

These audio files cut and compress sounds that are inaudible to the human ear, which can save a lot of space.

However, even in the sense of cutting off the sound, it deviates from the original sound.

However, it is quite difficult to distinguish it from those without loss in the ears of ordinary people and in an environment where you can prepare.

Therefore, it is still used in many software and services, taking advantage of the reduction in size.

About the uncompressed sound source
It is no exaggeration to say that the audio file is the closest to the original sound.

Windows .wav and Mac OS .aiff are famous as uncompressed sound sources.

Uncompressed sound sources are so flexible that they can store data at any sample rate and bit rate.

So it is the best file format to record first.

But it goes without saying that the file size is large because it is not compressed.

Therefore, lossless compression, which can reduce the size and preserve the original data, is now popular.Difference between lossless compression (lossless compression) and lossy compression (lossy compression)
Basically, the higher the kbps, the higher the sound quality in the sense that it is closer to the original sound.

Therefore, it is easy to think that it is only necessary to collect sound sources with high kbps.

However, since the amount of data will be large, it must actually be compressed.

That’s where two compression technologies come in.

Lossless compression (lossless compression)
Rossy compression (lossy compression)
About the uncompressed sound source
Let’s explain each compression technology.

We will also look at uncompressed sound sources.

Lossless compression (lossless compression)
Lossless compression has the advantage of retaining a large amount of information by compressing the data without damaging it.

In other words, the tendency is for the kbps value to be higher.

Of course, because it is compressed, the file size can be reduced to about half.

However, the capacity is considerably higher than the Rossie compression.

Lossy compression (lossy compression)
Rossy compression is famous for formats like mp3 and AAC.

These audio files cut and compress sounds that are inaudible to the human ear, which can save a lot of space.

However, even in the sense of cutting off the sound, it deviates from the original sound.

However, it is quite difficult to distinguish it from those without loss in the ears of ordinary people and in an environment where you can prepare.

Therefore, it is still used in many software and services, taking advantage of the reduction in size.

About the uncompressed sound source
It is no exaggeration to say that the audio file is the closest to the original sound.

Windows .wav and Mac OS .aiff are famous as uncompressed sound sources.

Uncompressed sound sources are so flexible that they can store data at any sample rate and bit rate.

So it is the best file format to record first.

But it goes without saying that the file size is large because it is not compressed.

Therefore, lossless compression, which can reduce the size and preserve the original data, is now popular.

High resolution compatible terminal

High resolution compatible terminal

High-Resolution Audio

In the case of the iPhone, it does not support the output of high-resolution sound sources.

Hi-Res Audio

Therefore, it is necessary to prepare a headphone amplifier equipped with a DAC that supports high resolution.

In the case of Android, there are terminals that support high resolution such as Xperia.

To enjoy high-resolution sound quality, you need to check whether the terminal that plays music is also compatible.

High resolution recommended elements
We will present specific recommended high resolution elements.

ACTIVE “CT10”
Let’s take a look at the appeal of the product.

ACTIVE “CT10”
ACTIVE CT10 [Cool White] High-resolution portable audio player with high sound quality
ACTIVE CT10 [Cool White] High-resolution portable audio player with high sound quality

19,800 yen (as of 08/27 01:10)
Amazon
Rakuten

ACTIVE “CT10” is a digital audio player that allows you to easily enjoy high resolution.

ACTIVE “CT10” has the following characteristics.

Achieves smooth operation despite its compact size
Astell & Kern exclusive sound quality
Up to 400GB expansion with MicroSD card
groovers will end on June 30, 2019
Let’s take a look at each feature.

Achieves smooth operation despite its compact size
Many people support the compactness of the ACTIVE “CT10”.

It’s smaller than most smartphones and fits in the palm of your hand, making it easy to carry.

And despite its small size, it uses a “quad-core CPU” for exceptional stability.

The “Quad Core CPU” has high processing power and offers smooth operation.

The processing power is important because it is high resolution, which usually has a large capacity.

Astell & Kern exclusive sound quality
ACTIVE “CT10” is a collection of Astell & Kern technical capabilities.

Astell & Kern is the original brand of IRIVER Korea and has developed many audio players compatible with Hi-Fi / Hi-Res sound sources.

Due to its high sound quality, it is now one of the standard brands of high-resolution players.

ACTIVE “CT10” boasts of sound quality because it requested development support from the IRIVER group company.

Up to 400GB expansion with MicroSD card
Since the capacity is 16GB, it is a little small to put a high resolution sound source.

However, since it supports MicroSD cards up to 400GB, it can be expanded up to 416GB in total.

It is sold separately, but if you want to put a lot of music on it, buy a MicroSD card.

High resolution has a large capacity.

groovers will end on June 30, 2019
ACTIVE “CT10” was able to buy a high resolution sound source directly from the player body.

However, it cannot be used now because the distribution site “groovers” has ended.

However, ACTIVE “CT10” can easily transfer music from your computer.

Just because “groovers” have finished doesn’t mean you can’t listen in high resolution.

Difference between analog data and digital data
Analog is a line.

Digital is a point.

Digital data can be brought closer to analog (line) by arranging points in succession.

The record is analog because it records continuous sound waves in the groove as is.

A CD divides continuous sound waves at regular intervals (sampling) and applies numerical values ​​to each division (quantization).

Then the flow consists of converting the numerical value into digital data suitable for a computer (encoding).

In this way, digital data is only one point, so in theory it is further from the original sound than analog.

However, since it is indistinguishable to the human ear, it is possible to enjoy the sound digitally without any discomfort.

Over the limits of human hearing! High resolution doesn’t make sense?

Over the limits of human hearing! High resolution doesn’t make sense?

Hi-Res

What is the limit of the ability to discriminate human sounds?

Hi-Res Audio

Although there are individual differences, the human audible range is said to be 20 Hz to 20 kHz.

The sampling frequency of the CD is 44.1 kHz.

In theory, this value can be reproduced up to 22.05 kHz of the original sound, so it can sufficiently cover the human audible range.

In that case, the question arises that high-resolution sound quality, which has higher specs than CDs, may not be audible to the human ear.

Actually, there are many negative opinions from a scientific point of view, and it remains controversial.

<Reference: High Resolution Audio-Wikipedia>

However, the sound is highly dependent on experience, so if you are interested, you should give it a try.

Necessary environment to enjoy high resolution
Whether it makes sense or not, there are a few things you need to prepare for to enjoy high resolution.

The required environment is as follows.

High resolution sound source
High-resolution compatible headphones / earphones / speakers
High resolution compatible terminal
Let’s explain each environment.

High resolution sound source
There are various media for high resolution audio sources.

The easiest of these is probably the Hi-Res Audio source that is delivered online.

If you are a high resolution distribution site, you can get many sources of sound.

Note that listening with high-resolution headphones will not result in high-resolution audio.

High-resolution compatible headphones / earphones / speakers
In addition to high-resolution audio sources, prepare items such as compatible headphones and earphones.

It is quick to check if there is a high resolution mark on the product package to see if it is compatible.

Please note that most wireless headphones do not support high resolution playback because they compress data during transmission.

In the case of wireless technology, only aptX / aptX HD compatible items or SONY LDAC compatible devices support high resolution.

In any case, the high-resolution branding is a milestone.

High resolution doesn’t make sense?

High resolution doesn’t make sense?

Hi-Res

Sound quality comparison!

What is the difference between bit, khz and kbps? High resolution doesn’t make sense?
It is quick to judge whether the sound quality is good or bad from the data.

However, there are several factors that make up sound quality.

Therefore, in this article, I explained the sound quality in a rather broken way.

If you are a little interested in sound quality, check it out.

High-resolution sound quality and CD comparison
Over the limits of human hearing! High resolution doesn’t make sense?

Necessary environment to enjoy high resolution
High resolution sound source
High-resolution compatible headphones / earphones / speakers
High resolution compatible terminal
High resolution recommended elements
ACTIVE “CT10”
Difference between analog data and digital data
Difference between lossless compression (lossless compression) and lossy compression (lossy compression)
Lossless compression (lossless compression)
Lossy compression (lossy compression)
About the uncompressed sound source
Difference between bit, kHz (kilohertz) and kbps (Kirobuki PS)
The next three units are important for measuring sound quality.

little
kHz (kilohertz)
kbps (Kilobit per second)
Let’s briefly explain each unit.

What is a bit?
bit is the number of quantization bits.

Here, it is easy to understand if you think of it as “the resolution (fineness) of the sound itself”.

The higher the value of this bit, the smaller and finer the sound can be reproduced.

In other words, it is more faithful to the original sound.

However, there is a disadvantage that the amount of data increases when the bit is large.

What is kHz (kilohertz)?
kHz is the sampling frequency.

It’s easy to understand if you think of it as “the number of data used in the expression of sound per second”.

The higher this number, the greater the range that can be recorded.

Also, kHz is one of the numbers often used to express headphone specifications.

If you look at the specification table, you will often see numbers like “20Hz to 20kHz”.

This indicates the range of supported ranges.

The lower the number on the left, the better the bass can be heard, and the higher the number on the right, the greater the playback range.

What is kbps?
kbps is the so-called bit rate.

Also known as “bits per second” or “kilobits per second”.

Unit that expresses how many kilobits of data can be transmitted per second.

Kbps can be calculated using the following formula.

kbps = bit x kHz x number of channels

The number of channels is the number of speakers, for example, in stereo, it is “2”.

In theory, the higher the number of processed bits per second, the higher the sound quality.

High-resolution sound quality and CD comparison
Let’s compare CD and high resolution sound quality from a data point of view.

little kHz kbps
16 bit CD 44.1-48 kHz 1,411.2 kbps
High resolution Exceeds 44.1 kHz / 16 bit
Those that exceed CD specifications are defined as high resolution.

Specific examples are as follows.

44.1 kHz / 16-bit: CD specifications
48 kHz / 16-bit: CD specifications
44.1 kHz / 24-bit: high resolution (the number of quantization bits is greater than the CD specification)
48 kHz / 24 bit: high resolution (the number of quantization bits is greater than the CD specification)
96 kHz / 16 bit: high resolution (sampling frequency is higher than CD specification)
96 kHz / 24-bit: high resolution (both higher than CD specs)
96 kHz / 12 bit: not high resolution (low number of quantization bits)
32 kHz / 24 bit: not high resolution (sample rate is low)

Digital audio formats

Digital audio formats

Digital Audio

The digital audio format is a format for presenting audio data used in digital audio recording, as well as for additional storage of recorded material on a computer and other electronic media, so-called audio media.

digital audio

The audio file (a file containing a sound recording) is a computer file consisting of information about the amplitude and frequency of sound, saved for later playback on a computer or player.

Varieties of digital audio formats.

There are several concepts of audio format.

The digital representation of the audio data depends on how the digital-to-analog converter (DAC) quantizes. In sound engineering, two types of quantization are currently the most common:

pulse code modulation

sigma delta modulation

Quantization bit depth and sample rate are often specified for various audio recording and playback devices as a digital audio rendering format (24-bit / 192 kHz; 16-bit / 48 kHz).

The file format determines the structure and presentation characteristics of the audio data when stored on a PC storage device. To eliminate the redundancy of the audio data, audio codecs are used, with the help of which the audio data is compressed. There are three groups of audio file formats:

uncompressed audio formats like WAV, AIFF

lossless compressed audio formats (APE, FLAC)

lossy compressed audio formats (mp3, ogg)

Modular music file formats are highlighted. Created synthetically or from prerecorded live instrument samples, they are primarily used to create modern electronic music (MOD). Also, this can be attributed to the MIDI format, which is not a sound recording, but at the same time, using a sequencer, it allows you to record and play music using a certain set of commands in the form of text.

Digital audio media formats are used for both mass distribution of sound recordings (CD, SACD) and professional sound recording (DAT, minidisc).

For surround sound systems, sound formats can also be distinguished, which are mainly multichannel sound accompaniments for movies. These systems have complete format families from two major competitors, Digital Theater Systems Inc. – DTS and Dolby Laboratories Inc. – Dolby Digital.

The format is also called the number of channels in multichannel sound systems (5.1; 7.1). This system was originally developed for movie theaters, but has since been expanded for home theater systems.

What is digital audio and how does it work

What is digital audio and how does it work

Digital Audio

Regardless of the path chosen, after connecting the source, the sound from the source will be sent to a microprocessor called a digital audio converter (DAC for short), where there will be 2 stages:

Digital Audio

1) Conversion from analog to digital (a / d);

2) Conversion from digital to analog (d / a).

This processor is sometimes called an ad / da converter. Here, the analog audio signal is processed into digital, then redirected to the central processor and memory, and then to the storage medium. Stored digital recordings (often in .WAV format) are sent back to memory and the CPU, and then converted back to analog by the DAC.

The digital audio / MIDI sequencer allows you to record the sound of synthesizers, guitars, and microphones to files with the .wav extension. No matter how sound is transferred to the computer, it will still go to the DAC, computer memory, and hard drive. The resulting data type is called digital audio data. If you record in “CD quality” (among other things one of the lowest possible), every second of the sound is divided into 44,100 pieces. What is this data? Only numbers. But unlike the MIDI format that encodes the notes played, digital audio data is a digital representation of the actual sound wave. This is the same sound described in numbers. Can you guess that this format takes up thousands of times more space than midi data? This is true.

It is a graphical representation of digital audio data. For a computer, this is a sequence of numbers. With this data, you can perform various operations to change and improve. Outwardly, the signals appear to undergo a series of effects, but in reality what happens is a mathematical process.

How MIDI is converted to sound
You may be wondering how to convert MIDI to audio, is there a “convert” utility for that? Connect the output jacks of your synthesizer to your sound card (or audio interface, or mixer with firewire, etc.) and start recording. Analog waves go through a digital converter (DAC), are converted into numbers, and voila! you will receive digital audio data. The nice thing about a sequencer is that you first record a MIDI track and then refine it. in editors and translate it to digital audio for a perfect recording (well maybe not perfect, there is nothing perfect in the world). Yes; you are using synthesizer software, the process will be called slightly differently, but the gist is the same. The computer creates an audio track based on MIDI data and records it in audio format.

Time to process the resulting files perfectly in sync with plugins or effects. You can also save the finished tracks in MIDI format (then you can edit them at any time) and add the sound of vocals, guitars, or whatever else you want. The sequencer can work simultaneously with MIDI files and digital audio.

Effects types
One of the main and most used effects is VIBRATO.
Distinguish amplitude vibrato, when the amplitude of the signal changes periodically. The frequency of change should be small, from a few fractions of a hertz to 10-12 Hz. Tremolo is a type of amplitude vibrato. The frequency of vibration in the case of a tremolo is not less than 10-12 Hz, and the resulting signal is output in portions.

Frequency vibrato. In a non-electronic way, it was done with electric guitars. By changing the tension of the strings with a special lever, the musician changes the pitch (understand – frequency) and achieves the effect of frequency vibrato. The same can be done with synthesizers and midi keyboards using a special wheel or lever. In music editors, you can also adjust the frequency of the sound, change it within the specified or desired limits.

Ring vibrato. The signal passes through a filter, the settings of which are periodically changed. An interesting and beautiful sound is obtained due to periodic changes in the coloration of the timbre.

Effects: Reverb, Chorus, Flanger, Phaser, Delay: effects based on the delay of the signal.

Reverberation: the effect is created by mixing the main signal with copies lagged for different periods of time, obtained as a result of the reflection of various obstacles (walls, objects, etc.) The number of copies can be infinite, the reflected signal can return to reflected from another obstacle (the delay increases naturally) and again summarized with the main one. With a short delay, the effect results in an immersive and booming sound experience. .

Benefits of “digital audio”

Benefits of “digital audio”

Digital Audio

The digitized audio signal has the following advantages:

DIGITAL AUDIO

-the possibility of infinitely long storage without loss of original quality,

-the ability to reproduce for a long time without losing the original quality,

-the possibility of infinite reproduction without loss of original quality,

-simplicity and wide possibilities of processing by modern means,

-Resistance to interference in signal transmission lines.

From CD to Super Audio CD and DVD Audio

CD (Compact Disk) is a type of removable plastic disk with optical reading of information.

In 1979, Sony and Philips proposed the Red Book standard for digital audio recording.

Analog sound is digitized and recorded as a spiral track of alternating zeros and ones (micron holes and a smooth surface) on a 12 cm polycarbonate disc, slightly thicker than a millimeter, covered with the thinner layer gold (later aluminum).

The player’s laser illuminates the disc and detects binary “zeros” and “ones”, which, after processing, are converted back to sound. It is almost impossible to mistake zero for one. Possible problems associated with read errors and scratches on the disc surface were compensated for using digital error correction.

As a result, not only did the physical dimensions of the record holder decrease compared to vinyl record, but also the musical capacity increased significantly: up to 74 minutes (the then owner of Sony wanted his favorite Beethoven Ninth Symphony to fit into a disk).

In 1982 in Langenhagen (Germany) the mass production of compact discs (CD) began with the “Alpine Symphony” by I. Strauss.

Real

High-quality audio is now recorded in Super Audio CD and DVD Audio formats, which:

use a DVD media,

use multichannel recording (up to 5.1),

sampling rate up to 192 kHz,

quantization level: up to 24 bits (each bit doubles the precision of sound transmission and, at such a depth of quantization, the dynamic range of the reproduced sounds can exceed 130 dB).

The new recording formats offer the highest quality, are expensive ($ 15 per disc), and are not popular because most listeners, sadly, don’t care too much about sound quality.

Digital audio options

The important parameters of the digital representation of sound are the sample rate of the audio signals and the quantization of bits.

Quantization rates indicate how many times per second a signal is sampled (measured in amplitude) for conversion to digital code.
For CD standard it is 44KHz (44 thousand times per second), for SACD 192KHz

The quantization bit characterizes the number of signal steps and is measured by the power of 2.

For the CD standard, 16-bit audio adapters are used, which have 65,536 quantization steps (2 to the 16 power), as in an audio CD. For standard and 24-bit SACD.

Digital audio storage

About digitizing sound has a set of signal amplitude values ​​taken at regular intervals and can be written to file sequence numbers (amplitude values).

Two methods are widely used to encode audio information:

PCM (pulse code modulation)

ADPCM (Adaptive Relative Pulse Code Modulation)

PCM (Pulse Code Modulation) is a method of digitally encoding a signal by recording the absolute values ​​of the amplitudes. This is how data is recorded on all audio CDs.

ADPCM (Adaptive Delta PCM) – Records signal values ​​in relative amplitude changes (increments), allowing you to simplify data to take up less memory.

Lossless encoding (for lossless data odirovanie) allows data recovery from fully compressed (20-50%) stream.

Popular L ossless encoding algorithms:

Windows Wave (WAV) is the primary audio file format for Windows.
The Audio Interchange File Format (AIFF) is the primary audio format for the Macintosh.

L ossy encoding (lossy data encoding) enables you to achieve sound similarity of the reconstructed signal to the original with the highest possible data compression (10-1 5 times).

The basis of lossy-encoders is the use of psychoacoustic models: certain portions of the signal, in certain frequency ranges that are inaudible to the human ear, nuances (masked or inaudible frequencies) and occurs to remove them from the original signal.

Digital audio

Digital audio

Digital Audio

what happens to sound within computer programs

Digital Audio

Digital audio is a representation of analog sound used by computers and various digital devices to record and reproduce audio information. Like the frames of a movie, a digital audio signal is created from a series of sound fragments that are played when we press the play button. There are many different digital audio formats, they differ from each other in the transmission quality of the audio information.

About Pulse Code Modulation – PCM

If we talk about an acoustic sound or an analog signal, we are always talking about the propagation of sound waves in space. Whereas digital audio is only a rough description of what happens to sound or should happen within computer programs or digital devices.

This article will discuss pulse code modulation (PCM), the most common digital audio decoding system. Besides PCM, there are also DTS and Dolby Digital systems, but these are mainly applicable in the field of film and video production. Today we will not talk about them.

In pulse code modulation, a signal is read many times per second. At each reading moment the amplitude of the sound wave is recorded and reproduced. As mentioned above, a digital signal is just a rough copy of an analog signal, since an analog wave cannot be recreated with perfect precision. The values ​​of each fragment are rounded to the nearest most accurate, then all the fragments are played and we hear a copy of the original analog sound.

“What meanings are we talking about?” – you ask. Just as analog audio is defined by frequency and amplitude, digital audio is determined by two important values: the sample rate and the bit depth. The sample rate means how many times per second the fragments of the audio signal are read, and the bit depth is the value of the dynamic range of each fragment of the audio signal.

Sampling rate

The standard 44.1 kHz sample rate used for recording audio to CDs (remember those?) Might seem like a random number. But this is not the case at all. This value was chosen based on Kotelnikov’s theorem, which essentially states that the sampling frequency must be more than 2 times higher than the maximum value of the reading frequency. As you know, the upper limit of audibility of the human ear’s frequency range is 20 kHz. It turns out that the sampling frequency must be higher than 40 kHz. An additional 4.1 kHz is added to avoid distortion, the so-called aliasing effect. In theory, 44.1 kHz should be sufficient to accurately reproduce an audio signal, however there are higher values.

For example, 48 kHz is the dominant standard in film and video production. As in the case of cinema, sound is synchronized at a frame rate of 24 frames per second. We won’t go into the details of why exactly 24 frames per second was chosen, in other words, this is the minimum frequency at which we can see a smooth, eye-pleasing image. The sample rate must match this frame rate. Using a frequency of 44.1 kHz can cause a noticeable out of sync of the picture and sound. Again, based on Kotelnikov’s theorem.

Even higher sample rates are repelled by these two base frequencies of 44.1 or 48 kHz, multiplying them by multiples of 2. That is, 88.2, 96, 192 kHz are the standard sample rates for all audio equipment. modern audio.

Bit depth

The bitness or bitness of an audio file tells us about its dynamic resolution or, more simply, clarity. You can draw an analogy with digital photography: the higher the resolution of the photo, the clearer and better the image will be.

It is important to note here that we are not talking about the loudness of the signal, but about a more realistic, clean and clear sound. More accurate transmission of the audio signal.

Bit depth can be compared to text in the book. The lower the bit depth, the less meaningful the text will make. That is, lowering the bitness leads to the fact that some letters begin to disappear from words, punctuation marks from sentences. At the moment, we will still be able to grasp the meaning of the text, but if the bit depth continues to decrease, the information will become so distorted that we simply stop understanding what we are talking about. The same goes for sound: the lower the bit depth, the more distorted we hear the sound.