Tag: audio formats hi res

Principle of mp3 and file format analysis. Part4

The three bytes starting at 1397H are 54 41 47, which store the “TAG” character, indicating that this file has ID3 V1.0 information.

The 30 bytes starting at 139AH store the name of the song, the first 4 bytes that are not 00 are 54 45 53 54, which means “TEST”;
the 4 bytes starting at 13F4H are 04 19 14 03 and the year of storage is “04/25/2003” ”;
the last byte is 4E, which represents the music category, and the code name is 78, that is, “Rock&Roll”; the
other bytes are all 00, and no information is stored.

4 Conclusions
As an important multimedia data type, people are always looking for more efficient compression methods and new sound file formats. In the MP3 file, the MDCT transform is used, which is a quasi-optimal transform with a simple structure and easy programming, which avoids the problem that the optimal transform (KL) is difficult to solve for the eigenvalues ​​and eigenvectors of the covariance. matrix.

Through the analysis of the MP3 file format, it is not difficult to find its shortcomings. Each frame of an MP3 file has the same 4-byte frame header, which requires some space overhead for an MP3 file with a large number of frames. ID3 stores the music description information. The proprietary, copyright, and other information in the frame header is also description information. The music description information is a bit messy.

In any case, the development of MP3 is unstoppable. MP3 has become a recognized sound data format. MP3 is becoming a hot spot in the field of multimedia information processing along with JPEG images and PDF documents.

Principle of mp3 and file format analysis. Part 2

MP3 uses perceptual audio coding (Perceptual Audio Coding) this distortion algorithm.

The frequency range of sound perceived by the human ear is 20 Hz to 20 kHz. MP3 cuts out a lot of redundant signals and irrelevant signals. The encoder transforms the original sound into the frequency domain through a mixed filter bank and uses a psychoacoustic model. to estimate that it may be only The perceived noise level is quantized and converted to Huffman coding to form an MP3 bit stream. The decoder is much simpler, its task is to extract the sound signal from the encoded spectral line components through inverse quantization and inverse transformation. The MP3 encoding and decoding process is shown in Figure 1.
2.4 Modified Discrete Cosine Transform The cosine transform
Modified Discrete CT (MDCT) refers to converting a time-domain data set to frequency-domain data in order to know the changes in the time domain. MDCT is an enhancement of the DCT algorithm. The first fast algorithm is fast Fourier transform (FFT), but FFT has complex operations, MDCT are real operations, easy to program.
When compressing audio data, first divide the original sound data into fixed blocks, and then perform direct MDCT (direct MDCT) to convert the value of each block into MDCT 512 coefficients. The 512 coefficients are restored to the original sound data, and The original before and after sound data is inconsistent because redundant and irrelevant data is removed during the compression process. The FMDCT transformation formula is:
k=0, 1,
.
n0=(N/2+1)/2, X(n) is the time domain value, X(k) is the frequency domain value. If N takes 1024 points, it becomes 512 frequency domain values.
The IMDCT transformation formula is:

n=0, 1, …, N-1
MDCT itself does not compress data, it simply maps the signal to another domain, and quantization compresses the data. When bit allocation is done on the quantized transformed samples, the entire quantized block must be considered the smallest, which is called lossy compression.
3 File Format Analysis
MP3 MP3 file data is made up of multiple frames, and the frame is the smallest unit of the MP3 file. Each frame, in turn, 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 that contain description information for non-audio data.

Principle of mp3 and file format analysis.

Principle of mp3 and file format analysis

1. Introduction
With the rapid development of file compression technology, MP3 has become the most popular music format today. High-quality music spreads rapidly around the world with the arrangement of 0 and 1, which shakes people’s hearts. What is MP3? The full name of MP3 is MPEG Audio Layer 3, which is an efficient computer audio coding scheme. It converts audio files into smaller files with an .MP3 extension with a higher compression ratio, basically maintaining the sound quality of the original file. MP3 is part of the ISO/MPEG standard, which describes audio compression using a high-performance perceptual coding scheme. This standard has been continuously updated to meet the pursuit of “high quality and low quality”, and has now formed MPEG Layer 1, Layer 2, Layer 3 three audio encoding and decoding schemes. MPEG Layer 3 compression ratio can reach 1:10 to 1:12, 1M of MP3 file can be played for 1 minute and 1 minute of CD-quality WAV file (44100Hz, 16bit, dual channel, 60 seconds) occupies 10M space, so Calculated, the playing time of a 650M MP3 disc should be more than 10 hours, and the playing time of a CD of the same capacity is about 70 minutes. The advantage of MP3 is that the CD is incomparable.
2 Analysis of the principle of MP3
2.1 audio standard
MPEG MPEG (Moving Picture Experts Group) is a group of dynamic picture experts under ISO, the MPEG standard which makes it widely used in various multimedia. The MPEG standards include audio and video standards, of which the audio standards have been established as MPEG-1, MPEG-2, MPEG-2 AAC, and MPEG-4.
The MPEG-1 and MPEG-2 standards use the same family of audio codecs: Layer 1, 2, 3. A new feature of MPEG-2 is the use of low sample rate expansion to reduce the data stream, and another feature is multichannel expansion, which increases the number of main channels to 5. The MPEG-2 AAC (MPEG-2 Advanced Audio Coding) standard was released by Fraunhofer IIS and AT&T in 1997 to significantly reduce data traffic. The MDCT (Modified Discrete Cosine Transform) algorithm adopted by MPEG-2 AAC has a sampling frequency between 8KHz and 96KHz, the number of channels can be between 1-48.
The three layers of MPEG Audio Layer 1, 2, and 3 use the same filter bank, bitstream structure, and header information, and the sampling frequency is 32KHz, 44.1KHz, or 48KHz. Layer 1 is designed for DCC (Digital Compact Cassette) compressed digital tape, the data rate is 384kbps, Layer 2 has made a compromise between complexity and performance, and the data rate is reduced to 256kbps-192 kbps. Layer 3 is designed for low data traffic from the start, and the data traffic is 128Kbps-112Kbps. Layer 3 adds MDCT transformation to make its frequency resolution 18 times that of layer 2. Layer 3 also uses average information similar to MPEG video. Entropy Encoding reduces redundant information. The vast majority of MP3s use the MPEG-1 standard.
2.2 Purpose of audio compression
The MP3 format began in the mid-1980s, when the Fraunhofer Institute in Erlangen, Germany, dedicated itself to encoding high-quality, low-data-rate sound. Let’s look at an example: you want to sample a song you like that is about 4 minutes long, store it on a disk, sample it in CD-quality WAV format, at a sample rate of 44.1 kHz, that is, receive a value of 44100 per second, stereo, each sampled data is 16 bits (2 bytes), so the space this song occupies is:
44100 x 2 channels x 2 bytes x 60 seconds x 4 minutes = 40.4 MB
If you download this song from the Internet, assuming the transmission speed is 56 kbps, the download time is:
40.4x106x8/56x103x60=96 minutes
Even a 1M broadband network requires more than 5 minutes, it can be seen that audio compression is particularly important to reduce audio data storage space.
2.3 Encoding and decoding
MP3 MP3 audio compression consists of two parts: encoding and decoding. Encoding converts the data in a WAV file into a highly compressed bitstream, and decoding takes the bitstream and reconstructs it into a WAV file.

THE MOST COMMON FORMATS FOR MUSIC AND OTHER AUDIO FILES AND HOW THEY ARE RELATED TO EACH OTHER PART 2

AUDIO CONVERTER

With an audio converter the situation is even simpler. Programs of this type are specially designed to convert between audio formats quickly, without explicit user intervention. Unlike audio editors, converters, we can say, use batch mode, that is, they allow you to convert MP3 files in a single operation, for example, not a single copy, and make several pieces at once. Depending on the app’s function, there may be dozens or hundreds.

Audiobooks in MP3 format

Once again, the operation of such a package is simple. Just select the source material (usually it can be a completely different file type) and install the final format. Then press a special button to start the process, the output user gets all files of a certain type. Your save usually occurs in the folder set in the app’s default settings, but the save location can of course be changed by yourself. By the way, the same applies to basis functions, which will be used during the transformation. However, any program initially provides the user with a specific set of criteria to use with a specific type of audio file. They can also change.

The beauty of these apps is that they have a complete process that will automate as much as possible and do all the required processes without much time. However, if we use a music or audio editor, comparing them in terms of improving the same sound quality especially cannot be dispersed here.

MUSICAL ARRANGEMENT
This is another type of software, most of which have built-in editors for MP3, WAV, etc. In this sense, they work on a similar principle to audiorekatorami, but their abilities are slightly broader.

Convert to MP3 format

First of all, it deals with the fact that the entire composition can consist of fragments of different types (MP3, MIDI, WAV, OGG, VST-library or DX-tool, etc. D.). After recording all sound tracks, for example mixing and mastering with virtual synthesizers or prescription parties, the resulting files can be saved in the desired format. Mostly it is an MP3 or WAV, or the program’s project file. In some applications, there is also a recording function to disk. Do you want an audio CD? No problem! In addition to the audio editor, it may take a few minutes to perform the necessary operations and get the tracks on the output disc in CDA format.

If we talk about the benefits of this type of application, it is obvious that only a few formats of the same union, and then saving or exporting to some of the most common are its greatest advantages. Also, you need to pay attention to the fact that the very overlay effect or change of any track parameters happens in real time, that is, the result will not necessarily wait; can be heard immediately by turning some knobs, for example. , or another option. Of course, this is only a small part of what packages are capable of.

HOW SHOULD I USE IT?
Finally, we come to the question of choosing the software to use with the MP3 format, or any other sound to record to. As is clear, normal listening to music or audiobooks is enough and a humble player (software or “iron”), or more commonly a DVD player.

Converting files to other formats, so to speak, in a hurry, is the perfect audio converter. However, if the output needs to achieve crystal clear sound quality, or even convert one file type to another, it is indispensable without powerful dedicated software. Of course, this requires ordering more, and without any experience, time to get the same high-quality MP3 files as the first time and you can’t get. However, with at least some in-depth study from audio editors, let alone professional music studios, the results will exceed everyone’s expectations.