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mp3 audio format, the most popular

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mp3 audio format, the most popular

With the rapid development of file compression technology, MP3 has become the most popular music format today.

mp3 audio format, the most popular

MP3 File Format Analysis MP3 file data is made up of multiple frames, and the frame is the smallest unit of the MP3 file. Each frame consists of a frame header, additional information, and sound data. The playback time of each frame is 0.026 seconds, and its duration varies with the bit rate. Some MP3 files have extra bytes at the end to store description information for non-audio data. The structure of the MP3 file is shown in Figure 2. 3.1 Frame header format The frame header is 4 bytes long. For fixed bitrate MP3 files, the frame header format of all frames is the same. The data structure is as follows: typedef FrameHeader{ unsigned int sync:11;//Sync information unsigned int version:2 ;//version unsigned int layer:2;//layer unsigned int protection:1;//CRC check unsigned int bitrate:4;//unsigned bitrate int frequency:2;//unsigned frequency int padding:1;//unsigned frame length setting int private:1;//unsigned reserved word int mode:2; //unsigned channel mode int mode extension:2;//unsigned extended mode int copyright:1;//unsigned copyright int original:1 ;//unsigned original logo int emphasis:2;//emphasis mode }HEADER, *LPHEADER; See Table 1 for a description of the 4 byte frame header. Table 1 Explanation of the use of MP3 frame header bytes Name Length (bits) Description Synchronization information 11 All bits in the 1st and 2nd byte are 1, and the 1st byte is always FF. Version 200-MPEG 2. 5 01-undefined 10-MPEG 2 11-MPEG 1 layer 2 00-undefined 01-Layer 3 10-Layer 2 11-Layer 1 CRC check 1 0-check 1-no check Bit rate 4 The third bit Tuple sampling rate, the unit is kbps, such as MPEG-1 Layer 3, 64 kbps, the value is 0101. Frequency 2 Sampling frequency, for MPEG-1: 00-44.1 kHz 01-48 kHz 10 -32 kHz 11-setting frame length undefined 1 is used to set the length of the file header, 0-no setting, 1-setting, the specific setting calculation method see below. Reserved word 1 is not used. Channel Mode 2 The fourth byte indicates the channel, 00-Stereo 01-Joint Stereo 10-Dual Channel 11-Mono Expansion Mode 2 Only used when the channel mode is 01. Copyright 1 Whether the file is legal or not, 0-Illegal 1-Original logo legal 1 If original, 0-Not original 1-Original emphasis method 2 Used for classification of sound compensation after noise reduction and compression, which is rarely used and is it may not work in the future. 00-Undefined 01-50/15ms 10-Reserved 11-CCITT J.17 MP3 frame length depends on bit rate and frequency, the calculation formula is: frame length = 144×bit rate∕ frequency+padding For example: bit rate is 64kbps, frequency is 44.1kHz, when padding is 1, frame length is 210 bytes. After the table header there is additional information of variable length. For standard MP3 files, their length is 32 bytes, followed by compressed audio data, which will be decoded when the decoder reads here. For Constant Bit Rate (CBR) MP3 files, not all frames are the same length, and some frames may be one or more bytes longer. There is also Variable Bitrate (VBR) MP3, to minimize the length of MP3 file and ensure sound quality, compared to CBR file, except for the first frame, the rest is the same. The first frame of VBR does not contain audio data and its length is 156 bytes, which is used to store information such as standard audio frame header (4 bytes), VBR file identifier, frame number, number file byte, etc. See table 2 for the description of the structure. Table 2 Description of the first byte of the frame structure of the VBR 1-4 file The same standard sound frame header as CBR 5-40 Store the logo of the VBR file “Xing” (58 69 6E 67), the specific position of this logo depends on the adopted MPEG standard and the sound depends on the channel mode.

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mp3 audio format, the most popular

With the rapid development of file compression technology, MP3 has become the most popular music format today.

The encoder transforms the original sound into the frequency domain through a hybrid filter bank. Using a psychoacoustic model, it is estimated that it may be sufficient to be The perceived noise level is then quantized and converted to Huffman coding to form an MP3 bitstream. The decoder is much simpler and its task is to extract the sound signal from the encoded spectral line components through inverse quantization and inverse transformation.
2.4 Modified Discrete Cosine Transform Modified Discrete Cosine Transform (MDCT) refers to converting a set of time-domain data to frequency-domain data for time-domain variation. MDCT is an enhancement of the DCT algorithm. The first fast algorithm is the Fast Fourier Transform (FFT), but FFT has operations on complex numbers and MDCT are all operations on real numbers, which is convenient for programming. When compressing audio data, first divide the original audio data into fixed blocks, and then perform forward MDCT (Forward MDCT) to convert the value of each block into MDCT 512 coefficients. When decompressing, the reverse MDCT (Reverse MDCT) The 512 coefficients are restored to the original sound data, and the original sound data before and after are inconsistent, because redundant and irrelevant data are removed during the compression process. The FMDCT transformation formula is: k=0, 1,…, N/2-1 where N is the length of the transformation window, that is, the number of sample points per block, N=8, 16 ,… ., 1024, 2048. n0=(N/2+1)/2, X(n) is the value in the time domain, X(k) is the value in the frequency domain. If N takes 1024 points, it will become 512 frequency domain values. The IMDCT transformation formula is: 4 Modified Discrete Cosine Transform Modified Discrete Cosine Transform (MDCT) refers to converting a set of time-domain data to frequency-domain data to learn the changes in the domain. weather. MDCT is an enhancement of the DCT algorithm. The first fast algorithm is the Fast Fourier Transform (FFT), but FFT has operations on complex numbers and MDCT are all operations on real numbers, which is convenient for programming. When compressing audio data, first divide the original audio data into fixed blocks, and then perform forward MDCT (Forward MDCT) to convert the value of each block into MDCT 512 coefficients. When decompressing, the reverse MDCT (Reverse MDCT) The 512 coefficients are restored to the original sound data, and the original sound data before and after are inconsistent, because redundant and irrelevant data are removed during the compression process. The FMDCT transformation formula is: k=0, 1,…, N/2-1 where N is the length of the transformation window, that is, the number of sample points per block, N=8, 16 ,… ., 1024, 2048. n0=(N/2+1)/2, X(n) is the value in the time domain, X(k) is the value in the frequency domain. If N takes 1024 points, it will become 512 frequency domain values. The IMDCT transformation formula is: 4 Modified Discrete Cosine Transform Modified Discrete Cosine Transform (MDCT) refers to converting a set of time-domain data to frequency-domain data to learn the changes in the domain. weather. MDCT is an enhancement of the DCT algorithm. The first fast algorithm is the Fast Fourier Transform (FFT), but FFT has operations on complex numbers and MDCT are all operations on real numbers, which is convenient for programming. When compressing audio data, first divide the original audio data into fixed blocks, and then perform forward MDCT (Forward MDCT) to convert the value of each block into MDCT 512 coefficients. When decompressing, the reverse MDCT (Reverse MDCT) The 512 coefficients are restored to the original sound data, and the original sound data before and after are inconsistent, because redundant and irrelevant data are removed during the compression process. The FMDCT transformation formula is: k=0, 1,…, N/2-1 where N is the length of the transformation window, that is, the number of sample points per block, N=8, 16 ,… ., 1024, 2048. n0=(N/2+1)/2, X(n) is the value in the time domain, X(k) is the value in the frequency domain.

mp3 audio format, the most popular

With the rapid development of file compression technology, MP3 has become the most popular music format today.

High-quality music quickly spreads to all parts of the world with the arrangement of 0 and 1, shaking people’s hearts. What is MP3? The full name of MP3 is MPEG Audio Layer 3. It is an efficient computer audio coding scheme. It converts audio files into smaller files with .MP3 extension with a higher compression ratio and basically maintains the sound quality of the file. original. MP3 is part of the ISO/MPEG standard. The ISO/MPEG standard describes audio compression using a high-performance perceptual coding scheme. This standard has been continuously updated to meet the pursuit of “high quality and small quantity”, and now has formed MPEG Layer 1, Layer 2. Layer 3 three audio encoding and decoding schemes. The compression rate of MPEG Layer 3 can reach from 1:10 to 1:12. A 1M MP3 file can be played for 1 minute, while a 1 minute CD-quality WAV file (44100Hz, 16bit, 2ch, 60sec) occupies 10M of space, so Calculated, the time The playback time of a 650M MP3 disc should be more than 10 hours, while the playback time of a CD with the same capacity is about 70 minutes. The advantages of MP3 are unmatched by CD. 2 Analysis of the principle of MP3 2.1 MPEG audio standard MPEG (Moving Picture Experts Group) is a moving picture expert group under ISO, and the MPEG standard formulated by it is widely used in various multimedia. MPEG standards include video and audio standards, among which MPEG-1, MPEG-2, MPEG-2 AAC, and MPEG-4 audio standards have been developed. The MPEG-1 and MPEG-2 standards use the same family of audio codecs: Layer 1, 2 and 3. A new feature of MPEG-2 is the use of low sample rate expansion kits to reduce data traffic , and another feature is the multi-channel expansion kit, which increases the number of main channels to five. Fraunhofer IIS and AT&T released the MPEG-2 AAC (MPEG-2 Advanced Audio Coding) standard in 1997 to significantly reduce data traffic. The MDCT (Modified Discrete Cosine Transform) algorithm adopted by MPEG-2 AAC, The sampling frequency can be between 8 KHz and 96 KHz, and the number of channels can be between 1 and 48. MPEG Audio Layer 1, 2 and 3 use the same filter bank, bitstream structure, and header information, and the sample rate is either 32 KHz, 44.1 KHz, or 48 KHz. Layer 1 is designed for DCC (digital compact cassette) digital compression tape, the data rate is 384 kbps, and layer 2 has made a compromise between complexity and performance, and the data rate has been reduced to 256 kbps- 192kbps. Layer 3 was designed for low data rate from the beginning, and the data rate is 128Kbps-112Kbps. Layer 3 adds MDCT transform, which makes its frequency resolution 18 times higher than that of Layer 2. Layer 3 also uses information averaging similar to MPEG video entropy coding to reduce redundant information. The vast majority of MP3 uses the MPEG-1 standard. 2.2 The purpose of audio compression The MP3 format began in the mid-1980s, and the Fraunhofer Institute in Erlangen, Germany, was committed to high-quality, low-data-rate audio coding. Let’s look at an example: You want to sample a song you like that is about 4 minutes long, store it on a disc, and sample it in CD-quality WAV format at a sample rate of 44.1 kHz, which means receiving 44100 per second. , stereo, each sample data is 16 bits (2 bytes), so the space occupied by this song is: 44100×2 channels x2 bytes x60 seconds x4 minutes=40.4MB If you download this song from the Internet, assume the transmission speed is of 56kbps, the download time is: 40.4x106x8/56x103x60=96 minutes. Even a 1M broadband network takes more than 5 minutes. It can be seen that audio compression is especially important to reduce the storage space of audio data. 2.3 MP3 encoding and decoding MP3 audio compression involves encoding and decoding in two parts. Encoding is turning the data in a WAV file into a highly compressed bitstream, and decoding is taking the bitstream and reconstructing it into a WAV file. MP3 uses a distortion algorithm called Perceptual Audio Coding. The frequency range of sound perceived by the human ear is from 20 Hz to 20 kHz. MP3 cuts out a lot of redundant and irrelevant signals.

MP3 encoder

1. MP3 Encoder FAQ

: what is an MP3 encoder?
An MP3 encoder is a piece of software that uses the MP3 codec algorithm (compression/decompression) to create mp3 files. Most encoders only convert
a WAV file to an MP3 file, although many can convert other formats such as WMA, Real Audio, Ogg, etc.

There are only a few standalone encoders, and a lot of software also only uses 4 main encoding engines, largely due to
to Fraunhofer Gesellschaft patents and various companies helping with ISO sources. Although no company owns the license, the
Developers must pay expensive license fees no matter what proprietary MP3 encoder they use. Major MP3 encoding engines include: LAME (
non-ISO source), BladeEnc, Fraunhofer, and Real Networks’ Xing encoder.

– How does the MP3 encoder work?
The core technology under MPEG-Layer 3 is included in the MP3 encoder. The decoding process uses a series of algorithms and rules to compress audio.
The encoder also detect sounds that occur at the same time
and they try to rule out any that might be “masked” or “inaudible” by other sounds.

– What is a good MP3 encoder?
Xing is the fastest encoder in terms of speed, but the worst in quality. For smaller file sizes, Fraunhofer FastEnc
offers the best quality. LAME is a very good encoder, and one version is faster than the previous one, BladeEnc
it is the best quality for large files, but very slow.

2. Dissection of MP3 files
In addition to proficiency in using the basic features of the MP3 encoder, ordinary users do not need to know how the internal structure of the MP3 file is encoded, just like the situation when
face JPEG or DOC files. Out of morbid curiosity, here’s an X-ray view of an MP3 file:

– Box header
As mentioned above, MP3 files are made up of thousands of “frame frames”, each frame containing a part (second part) of valuable audio data.
for the decoder to reconstruct the audio data. The first part above is the box header. (Frame Header), which consists of 32-bit metadata related to the
later data, see the figure below. The MP3 header begins with an 11-bit “sync timing” block, which allows the player to seek and lock the first
legal framework available, which is useful in MP3 streaming, which can quickly move or jump ID3 from the playback source block to a normal one.
position . However, simply detecting synchronized blocks is theoretically not enough, so it is necessary to check the header.

– transmission lock
MP3 was originally designed for broadcast, and as a result it became important that the MP3 receiver could be synchronized with the signal at any part of the broadcast,
so the frame header is placed at the beginning of any frame transmission, so when an MP3 receiver “tunes” to a data stream, it picks up the
signal instantly and you can play it immediately. Interestingly, this fact makes it possible to cut MPEG files into small segments, each of which can be played independently. But unfortunately
not possible in 3-layer (MP3) files, where frames often depend on other frames, so you can’t just
Edit .

– Frames per second
Just as the movie industry has a standard for the number of frames per second in film to ensure proper viewing on any projector,
A similar standard is used in the MP3 standard, regardless of the file’s bitrate, MPEG-1 A frame in the file is 26 ms, approximately 38 fps frames per second. If the bit rate
is , the frame size is correspondingly larger, and vice versa. Also, the number of samples contained in an MP3 frame is constant, 1152 samples per frame.

The total size of any given frame can be calculated with the following formula:

FrameSize = 144 * BitRate / (SampleRate + Padding).

Find out in detail what is the MP3 and ACC music format

Find out in detail what is the MP3 and ACC music format

MP3 o AAC

Songs have become part of our daily life and we rarely listen to a single song during our day, during our breaks or in our free time. New music never stops appearing and it is likely that on many occasions we would like to download these songs.

MP3 VS AAC

Many of us listen to hundreds of songs by our favorite bands every day, and we may never really analyze the format of each song in detail. We have heard of the existing formats, but we really do not know the benefits of each of them and their characteristics.

For this reason, Solvetic on this day will analyze in detail the two most common formats at a musical level, such as MP3 and ACC.

What is AAC?

AAC (Advanced Audio Coding) is a new audio format developed by the Fraunhofer Institute in Germany in collaboration with companies such as AT&T, Nokia, Sony and Dolby.

AAC, whose extension is m4a, is responsible for compressing a part of the audio files of an element called lossy compression, that is, some data that affects its optimal quality since inaudible frequencies are removed from the audio element, etc.

This AAC format is based on the international standard ISO / IEC 13818-7 and is basically an extension of MPEG-2. It is important to note that Apple chose AAC as the default format for the iPod and for iTunes, demonstrating its high level of quality.

Among its main characteristics we find:

It uses a bit rate encoding variable called VBR, which adapts the number of bits used in one second to encrypt the audio data.
Supports up to 48 channels for polyphonic sounds
It offers frequencies ranging from 8Hz to 96.0kHz.
They are smaller in MP3 size
AAC focuses on broadband usage
Provide high quality sound
As we can see little by little, AAC is establishing itself as one of the best music formats of the time.

What is MP3

MP3 (Motion Picture Experts Group) is an audio format that delivers quality while drastically reducing file size.

MP3 uses a lossy algorithm with which we can reduce the size of an element without losing its quality. This format, like AAC, was developed at the Fraunhofer Institute in Germany. MP3 has the ability to compress using a lower or higher bit rate, which will affect the sound quality.

Its main characteristics are:

Supports frequencies from 16 to 48 kHz
Allows compression of the audio object with a ratio of 11: 1
With the MP3 format, music is divided 44,100 times per second and each of these parts is 16 bits.
MP3 can contain tags with information about the included file
With these concepts in mind, we will see that AAC and MP3 behave in certain situations.

Audio file size

Both formats perform the function of reducing the size of the original file while maintaining sound quality. At this point AAC reduces the file size more than MP3, for example a 20MB MP3 file will weigh 16MB in AAC format.

compatibility

As we already mentioned, the ACC is being implemented by Apple for its devices, and therefore there is no doubt that the most compatible format is MP3, since since the 90s it has accompanied us on various devices such as cell phones, audio systems, televisions. , team. calculations, etc.

Sound quality

In this regard, AAC surpasses Mp3 for technical reasons such as a higher audio frequency, a higher level of audio compression to eliminate elements that affect its quality, better encoding, among other things.

Next, we will see the relationship between these two audio files:

The death of the MP3 has been mentioned in some places, but this is not really the case where the licenses of this format have stopped being active, so the MP3 will continue to be active in many of the songs we listen to, and there is no doubt that that ACC will gradually gain strength until it surpasses it. MP3 medium term, but for now, AAC users can enjoy and appreciate AAC.

Let’s continue enjoying our favorite songs and remember that the purpose of these files is to offer quality sound in a small storage space.

AAC vs mp3 quality

AAC vs mp3 quality

MP3 vs AAC

Answer 1 :
Q: What is the difference between AAC and MP3?

AAC vs MP3

The other answers here helped to talk about the technical differences between the two lossy compression formats.

I’ll take a different tactic with this answer and explain how they sound different to the ear.

To explain the difference in abbreviated form, at any given bitrate, AAC will sound better in the higher ranges, while MP3 will sound better in the lower ranges.

MP3 compression adds a specific sound to the sound. This is very noticeable at bit rates of 128 kbps and below; everything sounds confusing. At higher bit rates like 256 kbps (where it’s hard to hear) or 320 kbps (where you need high-end hardware to listen to artifacts), MP3 compression is much less of a problem.

AAC compression is much better at high frequencies. “AAC” in AAC is that music sounds weak, especially at low bit rates. If you like music with significant low frequency content (drums, electronic drums, bass, bass, etc.), you will miss some of that bass in AAC files; they just sound like they lack solidity. However, as with MP3, the higher the bit rate, the less problem you will be able to hear.

At any bit rate below 256 kbps, I personally prefer AAC. The lack of solidity in AAC compressed music is less undesirable than in Futz with MP3 compression.

At 320 kbps, these artifacts are very difficult to hear in any compression format, so the fact that MP3 is more compatible in most cases gives this compression algorithm an advantage.

But we also live in today’s world where conventional hard drives have more than 12 terabytes. A completely uncompressed album (that is, AIFF or WAV format) is less than 650 megabytes in size. (** grip calculator **) You can put 18,461 uncompressed WAV or AIFF albums on a 12TB hard drive. So why do we continue to use MP3 and AAC today?

Answer 2:
Both are compressed audio files, and although the audio quality is fairly similar, the AAC format was designed to improve over MP3 in the following ways:

Higher sampling frequency (8 kHz to 96 kHz) than MP3 (16 kHz to 48 kHz)
Up to 48 channels (MP3 supports up to two channels in MPEG-1 mode and up to 5.1 channels in MPEG-2 mode)
Arbitrary bit rates and variable frame length. A constant bit rate standardized with a bit pool.
Higher efficiency and simpler filter bank (uses pure MDCT instead of hybrid MP3 encoding)
Higher encoding efficiency for stationary signals (AAC uses a block size of 1024 or 960 samples, which can be encoded more efficiently than 576 MP3 blocks)
Higher encoding precision for transition signals (AAC uses 128 or 120 sample block size, which provides more precise encoding than 192 MP3 sample blocks)
You can use a Kaiser-Bessel derived window function to eliminate spectral leakage by enlarging the main lobe
Much better handling of audio frequencies above 16 kHz
More flexible articulation stereo (different methods can be used in different frequency ranges)
Add additional modules (tools) to improve compression efficiency: TNS, inverse prediction, PNS, etc. These modules can be combined to create different encoding profiles.

Answer 3:
Both are lossy codecs, aimed at significantly reducing file size without affecting sound quality as much as you might think.

AAC is 2 generations younger than MP3, so by then the algorithms had improved significantly, and most tests confirmed that 256 kbps AAC sounds just as good, if not better than 320 kbps MP3, which is why Apple chose this file format for iTunes.

AAC supports higher sample rates than MP3, although I’ve recently seen some weird MP3 implementations (incompatible with just about everything) that do this too.

After all, storage and internet speed are not an issue, lossy compression should be gone by now in favor of FLAC or ALAC. It seems that some bad habits are very difficult to break. 🙂

Answer 4:
AAC stands for Advanced Audio Coding. It was developed by the same people who invented MP3 and is destined to be its successor. Audio in AAC is better than MP3 in almost all cases.

It is more efficient than MP3 in terms of file size precision (bit rate). In other words, an AAC encoded song will sound as good or better than an MP3 encoded with the same bit rate. Therefore, encoding a file at 256 kbps AAC will give you better sound and smaller file size than MP3 at 320 kbps.

AAC vs MP3: which one sounds better?

AAC vs MP3: which one sounds better?

AAC Vs. MP3

AAC and MP3 are now widespread and established in the hardware and software markets. AAC is often touted as the successor to MP3. But is the successor really better? We tell you who sounds better and why.

MP3 to AAC

What are AAC and MP3?

You are probably familiar with AAC and MP3 from your music downloads, audiobooks and audio software for ripping audio CDs or compressing WAV or AIFF files.
Both formats are lossy audio codecs. In a special practical tip, we will explain what exactly a codec is.
Sound in AAC format is often hidden behind M4A and MP4 file extensions.
In a practical advice we explain in detail the differences between MP3 and MP4.
MP3 and AAC are both based on psychoacoustic models of loudness and masking that were developed in the 1960s by Eberhard Zwicker, for example.
Although there are newer and more precise models, the innovations since MP3 mainly reside in more sophisticated signal processing.

AAC vs. MP3: which one sounds better?

AAC is newer than MP3. Does newer mean better? At least the AAC innovations compared to MP3 have the potential for significantly stronger compression with the same sound quality or, conversely, significantly better sound quality with the same compression:
As described above, both codecs are based on practically the same psychoacoustic models.
However, AAC allows more flexible window sizes to better react to transient or stationary signals, depending on the signal.

Unlike MP3, AAC also offers more flexible windows. Used sensibly, this can improve frequency accuracy in applied spectrum analysis.
AAC also allows for frequency-dependent stereo ensemble. This can save quite a bit of storage space with little effort, as the low frequencies in audiobooks, music, and movie sound are often kept mono.
Since AAC offers significantly more flexibility on the encoder side, even a good MP3 encoder cannot keep up with a good AAC encoder.

On the other hand, a poorly conceived AAC encoder can also sound significantly worse than an MP3 of the same size. If you encode an MP3 optimally, the result can compete with many AAC encoders.

However, in our 2003 audio encoder quality comparison test, AAC wins, followed by Warning, OGG over MP3.
Also in our 2005 AAC encoder audio codec test from Nero also wins.
AAC is also more flexible than MP3 for the user. For example, AAC supports sample rates from 8 to 96 kHz, MP3 only from 16 to 48 kHz. If you go for 96 kHz music DVDs, even the highest quality MP3 won’t give you a good sample rate.
AAC also supports up to 48 channels, MP3 only 5.1. In AAC, in theory, it could also encode audio material for 7.1 sound, high-order ambisonics, Dolby Atmos, and Auro-3D.

By the way, there is an important rule to keep in mind: converting an MP3 to AAC or vice versa is quite detrimental to the audio quality. You should only convert for compatibility reasons, if, for example, your portable MP3 player does not support the AAC format.

AAC improvements over MP3

Advanced Audio Coding is designed to be the successor to MPEG-1 Audio Layer 3, known as MP3 format, which was specified by ISO / IEC at 11172-3 (MPEG-1 Audio) and 13818-3 (MPEG-2 Audio).

Blind tests in the late 1990s showed that AAC demonstrated higher sound quality and transparency than MP3 for files encoded with the same bitrate.

The improvements include:

higher sampling frequencies (8-96 kHz) than MP3 format (16 to 48 kHz);
up to 48 channels (MP3 supports up to two channels in MPEG-1 mode and up to 5.1 channels in MPEG-2 mode);
Arbitrary bit rates and variable frame length. Standardized constant bit rate with bit deposit);
higher efficiency and simpler filter bank (instead of hybrid MP3 encoding, AAC uses pure MDCT);
higher coding efficiency for stationary signals (AAC uses a block size of 1024 or 960 samples, allowing more efficient coding of sample blocks than MP3 576);

higher coding precision for transient signals (AAC uses a block equal to 128 or 120 samples, allowing more precise coding of blocks of MP3 192 samples);
possibility of using derivatives of the Kaiser-Bessel window function to eliminate spectral dispersion at the expense of enlarging the main lobe;
much better management of audio frequencies above 16 kHz;
more flexible joint stereo (different methods can be used in different frequency ranges);
additional modules (tools) added to increase compression efficiency: TNS, Back Prediction, PNS, etc. These modules can be combined to form different encoding profiles.
In general, the AAC format allows developers more flexibility in codec design than MP3 and corrects many of the design choices made in the original MPEG-1 audio specification. This increased flexibility often leads to multiple simultaneous encoding strategies and consequently more efficient compression. However, in terms of whether AAC is better than MP3, the advantages of AAC are not entirely conclusive, and the MP3 specification, while dated, has proven surprisingly robust despite notable flaws. AAC and HE-AAC are better than MP3 at low bit rates (typically less than 128 kilobits per second). This is especially true at very low bit rates where superior stereo, pure MDCT encoding, and better transform window sizes let MP3 compete.

While the MP3 format has almost universal hardware and software support, mainly because MP3 was the format of choice during the crucial early years of music sharing / distribution over the Internet, AAC is a strong competitor due to some unwavering support from the industry.

How AAC works

AAC is a wideband audio coding algorithm that takes advantage of two main coding strategies to dramatically reduce the amount of data required to represent high-quality digital audio:

Components of the signals that are perceptually irrelevant are discarded.
Excess in the encoded audio signal is removed.
The actual encoding process consists of the following steps:

The signal is converted from the time domain to the frequency domain using the Forward Modified Discrete Cosine Transform (MDCT). This is done using filter banks that take an adequate number of time samples and convert them to frequency samples.
The signal in the frequency domain is quantized based on a psychoacoustic model and encoded.
Internal error correction codes are added.
The signal is stored or transmitted.
To avoid corrupted samples, a modern implementation of the luhn mod N formula is applied to each frame.
The MPEG-4 audio standard does not define a single or small set of highly efficient compression schemes, but rather a complex set of tools to perform a wide range of bitrate encoding operations, from low speech to audio encoding. high quality and musical synthesis.

The ‘MPEG-4 family audio coding algorithm covers the range from low speech coding bit rate (up to 2 kbit / s) to high quality audio coding (at 64 kbit / s per channel and higher).
AAC offers sample rates between 8 kHz and 96 kHz and any number of channels between 1 and 48.
In contrast to MP3’s hybrid filter bank, AAC uses Modified Discrete Cosine Transform (MDCT) in conjunction with increasing window lengths of 1024 or 960 points.