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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.

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