Tag: audio compression codecs

Which audio codec for Bluetooth is better?


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Which audio codec for Bluetooth is better?

Bluetooth Audio Codec

The best codec is the one that can deliver the best sound quality. But if they were considered among the popular formats, it was difficult to choose the right option. Each one has positive and negative characteristics that influence the final choice.

Bluetooth audio codec

Which audio codec is better?
When choosing codecs, the following nuances should be taken into account:

Experts believe that the sound quality should be similar to that of CD audio. The sampling frequency must be 44.1 kHz and 16 bits. These are average values, they are observed only in the rarely used LDAC algorithm;

The aptX codec has modest performance, but it provides high-quality audio files. It is popular as the only high-definition Bluetooth codec;

All algorithms perform audio compression. This is necessary to reduce the bitrate to the maximum allowed for a particular codec. This means that when using any algorithm, the music will sound a bit distorted, in some it will be more obvious (for example, in the SBC format), in others it will be almost invisible (LDAC, aptX);

When choosing codecs, do not forget about the model and operating system of the smartphone, tablet and other devices. For Android devices, SBC or aptX is better, but for Apple it is recommended to use Advanced Audio Coding with an improved algorithm.
Below is a table with the main indicators and supported formats of popular algorithms.

Codec Sampling frequency (kHz) Bit rate indicator (kbps) Audio formats
SBC 46-48 328 MP3
CAA 42-44,1 250 MP3, AAC
LDAC 94-96 990 Lossless Formats, Hi-Res Audio
aptX 42-44,1 352 Audio CD
aptX HD 46-48 576 Lossless Formats, Hi-Res Audio

The SBC format codec is considered obsolete and is rarely used for playing music and audio files. It was originally created for the transmission of voice and sound data via Bluetooth. Over time, improved algorithms have appeared. If you want to buy wireless headphones for normal use, then it is better to give them to aptX based devices, these will transmit sound without obvious distortion, noise, squeak.

If you are using Apple devices, only AAC headphones will work. The algorithm is adjusted for this technique, you will be able to transmit the quality of the music. But when using it for Android OS devices, the sound will be distorted with interference.

For music lovers who value sound quality, the aptX HD algorithm is suitable. It has good sample rates, bitrate levels, and supports modern audio file formats. The codec characteristics convey high quality sound, the acoustics are delivered without distortion.

But if the price is not an obstacle, you can afford wireless headphones, devices, smartphones, tablets, LDAC-based players from the famous Sony company. The technique is quite expensive, the cost can amount to several hundred dollars, but the characteristics of the algorithm fully justify it.

Codecs are an important prerequisite for high-quality sound reproduction when using Bluetooth-based wireless devices and headphones. Without them, the music will be poor quality, distorted, and constant interference will make the melody shrill and vague. When using popular brand devices (Huawei, Xiaomi, iPhone), it is worth applying suitable algorithms that suit the device and the device’s operating system.


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What are audio codecs?

What are audio codecs?

Audio Codec

High-quality music without interruptions or interference is every music lover’s dream. Devices with a Bluetooth system are popular. Wireless headphones provide free, wire-free listening for lightness.

Audio Codecs

When using them, it is fashionable to do movements, running, playing sports, this will not affect the operation of the device. For its operation, the Bluetooth LDAC, AAC, APTX, SBC codecs are used. Each of them has distinctive characteristics, specific functions that must be considered in advance, this will help to break the principle of operation of these systems.

Why are codecs needed
Codecs are used to improve the quality of music file transfer via Bluetooth to wireless headphones. The system was originally created for data transmission, but it had problems with audio quality. The sound was distorted by noise and the development of codecs helped eliminate the annoying problems.

The operation of the algorithm is determined by the following criteria:

Sample rate indicator. Expressed in Hz. Indicates the data recording frequency for 1 second of sound. The higher the criteria, the better the sound quality;
The bitness of the recording ((Bit-depth). The bit is used for the measurement. If we consider a CD, then 16 bits is enough to record. The indicator is enough to record music up to 96 dB. But they have progressive recording methods , for which 24 to 32 bits are used.;
Bit rate. The indicator is expressed in kb / s. Reflects the amount of data the device processes to play 1 second of audio. A high value records a large amount of audio data for 1 second.
For reference! Voice transmission between carrier networks is regulated by the session border controller. This is carrier-class software that is part of carrier’s NGN networks. It issues signaling protocols and their dialects, analyzes the quality of the media channels through which voice traffic is routed.

Types of data storage and transmission formats
There are three formats for storing and transferring data: uncompressed, lossy (lossy compression), and lossless (lossless compression).

What audio codecs are common?

Audio Codecs

Depending on whether you want to burn your audio file to CD, make it available on the Internet, or edit it with an audio editor, the different audio formats are in question. Codecs are responsible for converting to and from the various formats:

Audio Formats

PCM (pulse code modulation)

Pulse code modulation is a coding process in which an analog signal can be digitized with almost no loss. Audio material encoded in this way is ideal for further processing because it is not compressed. Data generated with this method is generally saved as wave files with the extension “.wav”.

MP3 (MPEG-1 Audio Layer 3)

The encoding process is actually called MPEG-1 Audio Layer 3 or MPEG-2 Audio Layer 3 and was developed by the Fraunhofer Institute for Integrated Circuits. The name is derived from the associated MP3 file extension of the format. It is one of the first lossy compression processes to rely on psychoacoustic effects on perception to reduce the amount of data. In addition to the original codec from the Fraunhofer Institute, there is also the open source encoder LAME. Files containing data streams encoded in this way usually end in “.mp3”. There are also other container formats that can hold MP3 data streams, such as AVI or MP4.

AAC (advanced audio coding)

AAC is a lossy encoding method that can compress audio data (on a CD) to one-sixteenth of its original size. Compared to MP3, the process can demonstrate higher compression and improved sound quality. Therefore, various online music stores and online radio stations rely on this format. MP4 is designed as a container format to store compressed audio signals. Files containing such an audio track usually end in “.mp4” or “.m4a”.

Vorbis

This open source format is patent-free and therefore can be used by software developers without license fees. The format is also suitable for streaming. Compression is lossy and better than MP3. Although many hardware playback devices now support this format, it is not as widespread as MP3. The data stream is usually embedded in an OGG container. Associated files end in “.ogg” or “.oga”.

WMA (Windows Media Audio)

WMA is an encoding process developed by Microsoft and also offers lossy compression. Many hardware playback devices now support this format, because it is very popular in the music industry due to its built-in copy protection (Digital Rights Management (DRM)). If the file contains only audio data, it ends with “.wma”. ASF is used as the container format.

Why do you need “file formats”?

Digital data used to represent analog video or audio signals can be organized in different formats. The best way to explain this is with a single image – there are multiple options for storing individual pixels in a file. For example, if the image points are stored one after the other from left to right or first from top to bottom in the file it is of course a convention that must be specified. The way a color value is stored must also be clearly defined. These and many other definitions are determined by a specification, which is then implemented in the respective file format. To store the data, a predefined encoding rule is always followed, which is ultimately decisive for the data to be interpreted correctly. You can think of individual formats as different data carriers: CDs, large and small video cassettes, audio tapes, etc. can contain audio data; however, you cannot load a cassette in the CD player. WAV, MP4, WMA or MP3 file formats are equally different.

Many file formats are actually container formats. The term is intended to make it clear that different formats can be used within a convention. For example, an MP4 file can contain different video and audio formats that can also appear in the same file at the same time.

How to choose the perfect compressor configuration

Compressors and how to use them, explained.

Compression is one of your most powerful mixing tools. It is the essential element behind any good mix.

But for your compressors to work, you must first understand what compression is.

It can seem intimidating to start learning such a broad subject, especially when the controls and how they affect the signal are difficult to understand in relation to the sound.

This article will help you understand what compression does, how to choose the perfect compressor setting, and some common mistakes to avoid.

But before…

What is compression in music?

Compression in music is the process of reducing the dynamic range of a signal. Dynamic range is the difference between the loudest and quietest parts of an audio signal.

audio compression

You must reduce the dynamic range of most audio signals to sound natural to a recording.

For example: imagine a whisper and a scream on the same audio track. If they had the same volume difference as they do in real life, it would be very annoying!

Compressors fix all of this by attenuating the loudest parts of the signal and boosting what is output so that the quieter parts are more noticeable.

Imagine a whisper and a scream on the same audio track. If they had the same volume difference as they do in real life, it would be very annoying!
Using compression
Experienced engineers often talk about how one compressor is more “musical” than another.

It is an important concept. Its dynamics is one of the fundamental aspects for its sound to be unique.

When you use a compressor to change the dynamics, the sound engineer becomes part of the musical performance.

If your compressors work properly, they will positively contribute to performance and improve recordings.

Transients: understanding high energy moments.

To understand compression, you need to know what transients are.

Transients are the first high-energy moments of a certain sound in its waveform. These explosions give our brain a lot of information about the quality of a sound.

Since transients are usually louder than the rest of the waveform, they are greatly influenced by compressors.

For example: think of a nice roaring trap. As soon as the trap enters, there is an initial peak in the waveform that narrows slowly. That initial energy spike is your transient.

transient compresor

Compression helps you find the perfect balance for a track that has good dynamic range with a beautiful, full body.
A waveform with good dynamics will have a lot of transients when some sounds hit and then decay in the composition. Transients and their final decay are what make a waveform similar to a fish bone.

There is even an overly dynamic trail. If your song is transient without a body, its sound will not be of interest to your ear.

The reverse is also true, no dynamics can lead to lifeless, exhausting sound for the human ear and a waveform that looks like a big brick.

Compression helps you find the perfect balance for a track that has good dynamic range with a beautiful, full body.

Limiter

The threshold determines the signal level at which the compressor will start operating. The threshold is measured in dB, therefore any signal above the set threshold will be compressed.

When setting the threshold, decide what part of the signal you want to reduce.
With the threshold low, the compressor gain reduction is applied to a larger portion of the signal. Setting it higher affects only the most aggressive peaks and leaves the rest intact.

To determine what the perfect threshold is, think about what you’re trying to accomplish by compressing the audio and which parts of the signal are the most troublesome.

Are strong signal transients distracting you from the rest of your mix? Or maybe your final decadence is imperceptible in the mix?

A good rule of thumb for compression is “do no harm.”
Set the threshold to hear compressor operation on the part of the signal that needs to be addressed and not lowered.

Setting the perfect threshold will depend on your needs. Play the track and tweak it on the go to find the perfect amount.

Relationship

The ratio determines the amount of gain reduction applied by the compressor when the signal exceeds the threshold. It is called a relationship because it is expressed in comparison with the unaffected signal.

The higher the first number in the report, the greater the gain reduction factor.

For example, we can say that an uncompressed signal would have a 1: 1 ratio

What is the compressor and how does it work?

The compressor, together with the equalizer, is one of the most important and most used processors in professional audio, but its operation is not always so intuitive and knowing how to master the compression technique sometimes requires years of experience. In this new article we begin to explore this fundamental processor.

What is the compressor for?

First of all, let’s start to see what the compressor’s function is: to reduce the dynamic range of an audio track, that is, to decrease the distance in volume between the weakest signal and the strongest signal. Initially created to optimize recording on magnetic tape and to avoid saturation of the input stages, the compressor is still used today during recording and mixing. Reducing dynamic range also allows us to keep multiple tracks in the mix, such as a voice, for example, always at the same volume throughout the song so that they are not dominated by the other instruments in the most crowded sections, as well as to avoid Output saturation.

Compressor

Back to basics: what is the compressor and how does it work

The controls

Now let’s see in detail what the various compressor controls are and what they are for:
— Threshold: or threshold, expressed in dB, indicates the point beyond which the compressor begins to operate.
— Ratio: is the compression ratio and indicates how much the signal will compress when it exceeds the Threshold. For example, with a 2: 1 ratio, each signal that exceeds the threshold will be halved at the output, that is, every 2 dB at input 1 will be returned at the output.
— Make Up Gain: This is the output of the compressor and is used to recover the volume lost due to compression.
— Attack: expressed in milliseconds is the time it takes for the compressor to start once the signal has passed the threshold.
— Release: always expressed in milliseconds, it indicates the time it takes for the compressor to stop compression once the signal has returned below the threshold.
— Gain reduction meter: it is not a control but a visual indicator, led or pointer, which informs how much the signal is compressed, through a scale in dB.
— Bypass: shuts down the processor, making the signal pass through the machine without alteration.

With the advent of digital and accessories, we can find controls that not all hardware compressors have:
— Knee: indicates the type of curve at the point where the compressor begins to operate, which can be abrupt (Hard Knee), soft (Soft Knee) or various intermediate values.
— Automatic: sets the time control to which it refers (attack, release or both) automatically, depending on the input signal (program dependent).
— Sidechain eq or External Sidechain: Sidechain is the signal that drives the compression circuit, where in most cases it is the signal itself to compress, but sometimes it can be a version of the input signal with different equalization, for example without low frequencies, so that they don’t start the compressor too soon. Or it can be an external signal, such as the one used on the radio where the speaker’s voice signal drives a compressor on the background music signal, so it automatically turns off when it starts to speak (Ducking), or Classic Speaker Use to activate the compressor on various instruments in the mix or the Master Buss.
— Mix: used to mix the compressed signal with the original signal. This way, you can use Parallel Compression directly on the compressor, without having to use two mixer tracks (one for the dry signal and one for the compressed signal).
Back to basics: what is the compressor and how does it work

Compressor

Compressor or limiter?

What is the difference between a compressor and a limiter?

Essentially, the compression ratio: over 10 dB ratio, the processor is considered a limiter. A separate case is the Brickwall Limiter, a compressor with immediate attack and a compression ratio of infinity to 1, so that no signal can exceed the Threshold. It is mainly used on the master buses so as not to exceed 0dBFS on the output and then send the converters to clips.

Usage examples

As we already said, the compressor is used to keep the volume excursion under control. One track in the mix: in this case, using a fairly fast attack, slow release and not too aggressive ratio, allows us to compress the signal constantly and transparently, that is, without making your intervention feel excessively.
The compressor can also serve to emphasize the attack of a percussion instrument: in one case, for example, by setting a medium slow attack.

Audio normalization or compression

The function of a compressor is to reduce the dynamic range of the signal, that is, the level difference between the strongest and weakest signal parts.

Why compression or normalize?

At the time of analog, the limited dynamics of the main musical supports (vinyl, audio and video cassettes) did not allow to reproduce the dynamics of a classical, jazz or even rock orchestra in the case of the audio cassette. Therefore, the signal was compressed to avoid distortion in the transmission medium.

audio compression or normalization

Now that the music is converted to 16-bit or more, recorded in digital format, and then streamed to CD / DVD or downloaded, the dynamics of the media is enough to faithfully reproduce the dynamics of almost any orchestra. The old technical limitations have disappeared, therefore compression is no longer essential.

However, whatever the musical genre, some sources (voices) are compressed almost systematically. The goal of modern compression is therefore to optimize sound recording, either to get closer to reality or, conversely, to create a less faithful but denser, more controlled, more powerful sound, etc., or even a sound. totaly new.

And to do all this, the compressor is satisfied with a simple principle: it reduces dynamics by attenuating the signal level when the latter exceeds a given threshold level.

Level settings

– Threshold (threshold level, in dB)

This parameter determines the threshold level from which the compressor is triggered. As long as the input signal level remains below the threshold, the compressor does not start and no treatment is applied. As soon as the source signal exceeds the threshold level, compression is applied.

– Ratio (compression ratio)

The ratio determines the amount of level reduction applied to the part of the signal that exceeds the threshold level, the rest of the signal is not processed. Depending on the compressor, the ratio can vary from 1: 1 to Inf: 1. Quésaco?

Set up a compressor

With a 1: 1 ratio, no compression is applied, the level of the input signal is equal to that of the output signal. With a ratio of 2: 1, the level of the signal portion that exceeds the threshold is divided by 2 in the output signal. With a 3: 1 ratio, it is divided by 3, etc. When the compression ratio is infinite (Inf: 1 ratio), the compressor behaves like a limiter: the output signal never exceeds the threshold level, regardless of the input level.

Therefore, the compression intensity applied to the signal is a compromise between the threshold and the compression rate setting:

The lower the threshold, the larger the compressed signal portion.
The higher the ratio, the greater the level reduction applied to the signal portion above the threshold.
Depending on the compressors, you may find other parameters, for example, an input level setting instead of the threshold, or a gain setting (also called the offset or output level) that amplifies the signal to compensate for the drop in level resulting from compression.

Time settings

– Attack (attack, in ms)

Attack corresponds to the time the compressor needs to reach the given ratio when the signal level exceeds the threshold level. A quick attack of a few milliseconds triggers strong compression as soon as the signal level exceeds the threshold; With a slower attack, the compressor passes the first transients of the signal peaks, keeping one side alive and well cut.

Set up a compressor

– Launch (launch, in ms and s)

Release corresponds to the time the compressor needs to return to the 1: 1 unit ratio when the source signal falls below the threshold level. A quick launch of a few tens of ms allows the original character to stay alive. Slower relaxation improves instrument resonance and reverberation, but can cause compression of the first peak transients when the latter are close together.

– Knee (literally knee!)

The Knee parameter determines the increase in compression, that is, the transition between the compression ratio of the unit (1: 1, no compression) and the compression ratio set to ratio.

Applications

At the output, the compressor can be used as a limiter to control signal peaks and prevent distortion from occurring in the analog / digital conversion stage.
When taking and mixing, light compression can bring out weak parts of the signal and thus reveal certain details.
In the mix, the compressor allows you to increase the average level of the audio volume output.

Destructive compression vs non-destructive

Destructive compression is compression obtained by losing information. This means that if you extract the compressed signal with this technique, you will not find the start signal.

Destructive Vs Non-Destructive Audio compression

In destructive compression techniques, there are basically methods that take advantage of the properties of the human ear. The latter listens to frequencies between 20 Hz and 20 kHz. If a song contains frequencies outside this range, we can easily delete them without losing the audio quality because the ear does not hear them. In fact, frequencies between 2 kHz and 5 kHz are generally heard correctly. In fact, less than 5 dB is required to hear frequencies in this band, while more than 20 dB is required to hear frequencies below 100 Hz or above 10 kHz. These results can be used to reduce the size of the files. For example, we can conclude that all frequencies above 15 kHz are suppressed.

audio compression

MP3 also uses the principle of masked frequencies. If in a frequency group some have a much higher noise level than others, it is not necessary to keep the frequencies low – we will not hear them. Imagine yourself in your garden listening to the birds sing. The chord goes over your head (even very high). We no longer listen to birds because the sound they make is much quieter than that of the plane. It is as if the birds no longer exist or have stopped singing. Obviously, it is not necessary to code all the frequencies present in a song so that the human ear can always perceive it well. Finally like the two ways

What do we find among non-destructive techniques?

Mainly coding techniques.

Let’s explain. A sound is a frequency. A second of music is therefore a sequence of frequencies. Imagine that in the series of samples that make up a second of music (remember that there are 44,100), we have the same frequency several times in succession, for example 10 times. If instead of storing these 10 points, we only store 1 and the number of times it is repeated, we must encode 2 digits and not 10. If we also apply this method to frequencies which are no longer identical, but very dense together (so close that the average human ear cannot distinguish them), we can still save space. This time, the compression is destructive because we are replacing one frequency with another frequency (almost identical).

MP3 also uses the algorithm of Huffman (1952) as a method of encoding information. This method is used in all compression algorithms (compression of text files, compression of images, compression of sound). It is based on the use of a variable length code and the probability that an event (in this case a frequency) will occur. The more a frequency appears, the shorter the code (low number of bits to display it). The file is read for the first time and a table appears with the frequencies that appear and the number of times they appear. We derive the right code. This encryption was last used. It is the final phase of compression. This is non-destructive coding.

MP3 works on the properties of the ear, first to reduce the size of a part, then processes the stereo sound and possibly applies encodings which end with Huffman encoding.

The use of all the reduction options mentioned depends on the location you want to give within 1 minute of your tablet and therefore on the compression speed to apply.
To encode MP3 audio files, we are talking more in terms of bit rate than compression rate.
Bit rate is the number of bits allowed in 1 second.
Therefore, we have the following relationship: the more we want to compress a song (so that it takes up the least space possible), the lower the bit rate.

Choice of compression ratio (bit rate)

Obviously, the more you compress, the worse the sound quality.
You have to compromise the file size and audio quality.
This commitment can be dictated by your needs, but also by the use you want to make of your MP3 files. It may not even be demanding if your MP3s are intended for your portable music player and are too demanding to be listened to on a stereo system.

Mp3: Audio Compression.

Audio Digitization.

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 in a
point. Sound waves have the proper and measurable characteristics of waves in general,
such as reflection, refraction and diffraction. As it is a continuous wave, a
digitization process to represent it as a series of numbers. Currently, most of
the operations carried out on sound signals are digital, since both storage and
processing and transmission of 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 ability to include error protection codes,
as well as encryption. With the appropriate decoding mechanisms, moreover, they can be treated
simultaneously signals of different types transmitted on the same channel. The disadvantage
main aspect of the digital signal is that it requires a much greater bandwidth than that of the signal
analog, hence an exhaustive study is carried out regarding data compression,
some of whose techniques will be the center of our study.
The digitization process consists of two phases: sampling and quantization. In the sampling,
Divide the time axis into discrete segments: the sampling frequency will be the inverse of time
that mediates between one measurement and the next. At this time the quantization is performed, which, in its
In the simplest way, it is simply to measure the signal value in amplitude and save it.

Nyquist’s theorem guarantees that the frequency necessary to sample a signal that has its
Higher components at a given frequency f is at least 2f. Therefore, the range being
higher than human hearing around 20 Khz., the frequency that guarantees a sampling
suitable for any audible sound will be about 40 Khz. Specifically, to get sound
High-quality frequencies of 44.1 Khz are used, in the case of CD, for example, and up to 48 Khz.
in the case of the DAT. Other typical values ​​are submultiples of the first, 22 and 11 Khz. According to
nature of the application of course the appropriate frequencies can be much lower
such that the voice process is usually carried out at a frequency of between 6 and 20 Khz. or
even less. Regarding quantization, it is evident that the more bits used for the
axis division of amplitude, the “finer” the partition will be and therefore the less error in attributing
a concrete amplitude to the sound at every 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 performed at equal intervals or according to a certain density function,
looking for more resolution in certain sections if the signal in question has more components in a certain
intensity zone, as we will see in the coding techniques.
The complete process is usually called PCM (Pulse Code Modulation) and so we
We will refer to it hereinafter. It has been described in a very simplistic way, mainly
because it is widely discussed and is well known, being 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.
1.2 Coding and Compression.
Before describing compression and encoding systems, we must pause briefly.
analysis of human auditory perception, to understand why a quantity
Significant information that the PCM provides can be discarded. The heart of the matter,
as far as we are concerned, it is based on a phenomenon known as masking.
The human ear perceives a frequency range between 20 Hz. And 20 Khz. First of all, the
sensitivity is higher in the area around 2-4 Khz., so that the sound is more
hardly audible the closer to the ends of the scale. Second is the
masking, whose properties exhaustively use 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. TO
a certain distance from the masking frequency, the effect is reduced so much that
negligible; the range of frequencies in which the phenomenon occurs is called the critical band
(critical band). Components belonging to the same critical band influence each other and
they do not affect nor are affected by those that appear outside it

Audio Data compression

Data compression or the technique that changed everything

Without pretending to extend ourselves in the description of this critical concept, it is important to know that compression is understood as a scheme that allows, by means of a “decision” algorithm based on a series of “rules” (which in the case of audio are masking and audibility threshold) reduce the amount of data to transmit a certain message. In other words: if the song “x” occupies, in the format used to encode the sound of a CD, 1 million bits, the data compression allows that song to be reproduced with maximum intelligibility using only 50,000 of those bits.

In this way, the download of a complete CD from a certain website could be carried out in a reasonable period of time. But, of course, the price to pay was high in terms of quality because such “castration” of the original message (which in turn was not “continuous”, analog, but also digital, although “linear”, without compression) meant removing many nuances of music, a disaster that in reality did not care for many consumers but it did worry, and a lot, those who bet on that High Fidelity in the reproduction of the sound that we are so passionate about and who received a wound that was almost fatal . In this sense, it is worth knowing that the “philosophical” keys to data compression are summarized in two terms: redundancy and irrelevance. In the first case, it is about reordering the available data to eliminate the ones that are repeated (for whatever reason: security, etc.), a bit like a “zip” computer file. It is a formal remodeling that does not affect the sound message at all (but it does save space to transmit / save data, making it very practical), so in this case, we are talking about lossless compression or “lossless” ” It is the second term that has the greatest scope in terms of sound quality because the idea of ​​irrelevance implies deleting irrelevant data from a certain message. And, of course, who decides what is relevant or not? Well, an algorithm, a program that, obviously, can be more or less sophisticated but still makes decisions with which everyone will agree. It is easy to understand: what may be irrelevant to such a person and / or the team may not be so to someone else. The fact is that here musical information is deleted, which, fundamentally, can no longer be recovered. Well, the algorithms in which there are losses of musical information are known as “lossy” or lossless coding algorithms. From what has been said, it is easily deduced that the difference between the concepts “lossless” and “lossy” is the one that marks the border between high and low quality digital audio, between high resolution (with recording studio quality formats or “Studio Master” on the cusp) and that “practical” sound (in principle for portable players and cars) and very often unnatural formats like the once ubiquitous MP3, which, we insist, almost ruined with the improvements provided by the CD.
ADSL, the key to accessing High End audio via the Internet
Basically it was a purely technical progress that, logically, had to come. A progress that allowed breaking the limitations that prevented downloading a song recorded in PCM at 16 bits / 44’1 kHz and, over time, the files with much higher resolution than for a good decade and a half are the usual ones in studios of recording. So, thanks to ADSL, the High End in audio via the Internet, and therefore “without physical support” is available to everyone. At this point, it will be good to briefly review the small “soup” of acronyms with which we can find ourselves, otherwise the result of the availability of open and “closed” environments (Windows, Mac), in what CODEC’s (algorithms that compress and decompress data (in this case of music) refers to the fact that compression is the norm.

 

AAC (Advanced Audio Coding): It was designed to be the successor to MP3 and, although it is a lossy CODEC, the results in terms of sound quality are superior to those of MP3 for the same bit rate. The AAC has adopted a wide range of portable audio devices such as the iPod and its derivatives for use.
AIFF (Audio Interchange File Format): It is the version of WAV created by Apple. Works with uncompressed (ie “lossless”) files that maintain full resolution and size.
 

ALE (Apple Lossless Encoder), also known as ALAC (Apple Lossless Audio Codec): Uses lossless compression to save storage space. Once unzipped for listening, the file will be bit by bit identical to a full size WAV or AIFF encoded file. As in AIFF or FLAC, in ALE / A files

Types of audio codecs

Types of audio codecs:

-DST (Direct Stream Transfer)
-FLAC (Free Lossless Audio Codec)
-LA (Lossless Audio)
-LPAC (Lossless Predictive Audio Codec)
-LTAC (Lossless Transform Audio Codec)
-MLP (Meridial Lossless Packing)
-Monkey’s Audio (APE)

There is a huge amount of audio formats. The most common are formats such as MP3 (MPEG-2 Audio Layer III) and WAV. Usually, the type of format corresponds to the file extension (the letters of the file name after the period, for example .mp3, .wav, .ogg, .wma).

A codec is an algorithm for encoding and compressing data in an audio format. Some file types are assigned a specific codec. For example, the MP3 format always uses the MPEG Layer-3 codec, while the MP4 format can use a range of different codecs.

Many times, the notions of codec and format are used as interchangeable. Especially when a format always uses a single codec. However, it is necessary to understand the difference between a format and a codec. In simple terms, a format can be compared to a container in which a sound or a video signal that uses a particular codec can be stored.

Some formats, such as MP4 or FLV, can store both audio and video sequences.

In the general scope of codecs (for any type of data), we can classify them as follows, depending on whether the original signal can be recovered or not after coding:

With losses (lossy). In this type of codecs, after coding, it is impossible to recover the original signal. Most codecs manage to reduce the size of the bit stream to be transmitted or stored, due to the loss of information in said bit stream. Normally this loss does not produce a large decrease in the quality of the audio perceived by the end user, and if the decrease in quality is appreciable, it is that a lot of information has had to be lost to achieve a small bit stream size, that in many occasions it is necessary, especially in the transmission of audio at a distance (telephony, digital video, television …), although this is a compromise solution between the different codecs, an issue that we will discuss in the comparative section between codecs .
Lossless (loseless). In this type of codecs, after coding, the original signal can be recovered. These types of codecs are the least common. They are usually common especially in high quality audio applications, where the size of the bit stream or stream is not decisive. If the files are to be treated later, it is not advisable to perform loss coding, since one encoding with losses after another would significantly damage the audio quality.

There is another classification of codecs, depending on the type of algorithm used in the coding:

Waveform codecs:

used for all types of digital signals. The waveform of the encoded signal must be as similar to that of the original signal
Vocoders or source codecs: used only for coding voice signals. The original signal is analyzed and synthesized to give rise to the encoded signal
Hybrids: combine characteristics of the two previous types

The waveform codecs seek to produce a reconstructed signal of the signal to be encoded, whose waveform is as similar to that of the signal to be encoded. These codecs work without knowing how the signal to be encoded was generated, which implies that in theory its operation does not depend on the signal and can work well with all types of signals, even if they are not audio.

Hybrid Codecs

These types of codecs are a mix between waveform and source. Within the hybrid codecs, the most used are the codecs in the time domain of Analysis-by-Síntesi