Detailed music format


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Detailed music format

Audio File Formats
Audio File Formats

classic wave

Audio File Formats
Audio File Formats

As the most classic Windows media audio format, the WAVE file is widely used, which uses three parameters to represent sound: the number of sampled bits, the sample rate, and the number of channels.
The channels are divided into mono and stereo, and the sample rates are generally 11025 Hz (11 kHz), 22050 Hz (22 kHz), and 44100 Hz (44 kHz). The capacity occupied by the WAVE file = (sampling frequency × sampling bits × channel) × time/8 (1 byte = 8 bits).

traditional mod

MOD is a wavetable-like music format, but its structure is similar to MIDI, it uses real samples, and the volume is small. In the earlier DOS era, MOD was often used as background music for games. Modern mods can contain many audio tracks in many formats, such as S3M, NST, 669, MTM, XM, IT, XT, and RT.

midi music computer

MIDI is short for Musical Instrument Data Interface. Records the sound played by the instrument digitally (each note is recorded as a number), and then synthesizes these records via FM or wavetable during playback: FM synthesis is the sound of the instrument is simulated by mixing the multi-frequency sounds; wavetable synthesis consists of storing the sound samples of the instrument in the wavetable of the sound card and extracting the sound from the wavetable as you play.

Boss Boss MP3

It can be said that MP3 is famous, it uses MPEG Audio Layer 3 technology to compress the sound with a compression ratio of 1:10 or even 1:12, with a sampling rate of 44kHz and a bit rate of 112kbit/s. .
MP3 music is music stored in digital form. If you want to play it, you must have a corresponding digital playback and decoding system. Generally, MP3 digital music is decoded by special software and then restored to a waveform sound signal for playback output. This type of software is called For MP3 players, such as Winamp, etc.

Overlord RA series online

RA, RAM, and RM are Real’s mature network audio formats, using “streaming audio” technology, making them well suited for network streaming. Information such as copyright, singer, producer, mail and song title can be added during production.
RA can be called the supreme lord of multimedia communication on the Internet. It is suitable for streaming on the Internet and is currently the best format for listening to online music online.

VQF with high compression ratio

VQF or TwinVQ is an audio compression technology developed by Nippon Telegraph and Telephone and Yamaha Corporation.
The audio compression rate of VQF is almost twice that of standard MPEG audio and can reach approximately 1:18 or even higher. And popular compression formats like MP3 and RA are usually only around 1:12. But it still won’t affect the sound quality, when VQF compress music at 44kHz-80kbit/s audio sampling rate, its sound quality will be better than 44kHz-128kbit/s MP3, when compress at 44kHz-96kbit/s , the music is close to 44kHz-256kbit/s MP3.

MD minidisc

MD (ie MiniDisc) is a comprehensive portable music format released by SONY in 1992. The compression algorithm it uses is ATRAC technology (the compression ratio is 1:5). MD is divided into Recordable MD (Recordable, with two heads of magnetic head and laser head) and Single Play MD (Prerecorded, only laser head).
The powerful editing function is the strong point of MD. You can quickly select tracks, move tracks, merge, split, delete and edit track titles. It is more personalized than CD and you can have your own MD album at any time. MD products include MD Walkman, MD bedside audio, MD car audio, MD recording deck, MD camera gun and MD driver, etc.


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Compressed audio formats

Compressed audio formats

Compressed Audio File Formats

Understanding compressed audio formats
The digital age dictates its own laws, according to which, in particular, audio and video information is more convenient to store and transmit in compressed form. Let’s briefly discuss the principle of sound compression.

Compressed Audio file formats

As you know, the music we listen to consists of a set of signals, each of which has its own characteristics, including loudness. The human auditory system is designed so that we do not distinguish or misdirect a weak (low) signal from the background of a strong (strong) signal. This principle forms the basis of modern means of compression (compression) of audio data.

If we imagine that a signal of a certain length is divided into many parts, and each part is processed in such a way that a weaker signal, which is difficult to distinguish from a strong one, falls under the knife and a stronger signal remains, then this will be a rough model of audio signal compression. … Consequently, the level of data compression will depend on how many parts (samples) the original file will be divided into and how many weak signals from each individual sample will be removed (what the bit rate will be: the number of bits in a sample of a specified duration).

The first versions of codecs for data compression acted quite crudely: they just cut off a weak signal and did not take into account the type of music, therefore, rather energetic music, without special nuances, in a compressed form does not it sounded worse than the original, whereas more complex classical and acoustic music simply lost all color and depth.

As a result of this, a transition to a more intelligent compression algorithm, with a variable bit rate, was made. Depending on the musical texture, that is, the ratio of weak and strong signals, the codec changes the amount of weak signals cut, so that we hear a more believable sound.

Obviously, with a higher sample rate (sampling) of 44.1-48.0 KHz and a higher bit rate (160-192 Kbps), we will get a sound more consistent with the original than with a sample rate 22 KHz and 64 Kbps bit rate. However, the size of the final compressed file is directly proportional to the selected sample rate and bit rate, and this is what people who distribute music in the form of compressed (compressed).

It should also be remembered that most algorithms cut the upper part of the audible range as well, starting at around 15 kHz.

There are currently several original compression algorithms, most of which are compatible with Linux.

Ogg Vorbis
Ogg Vorbis is a completely open audio format that allows you to store and transmit audio information with high sound quality (44.1-48.0 kHz sample rate, 16+ bits, polyphony (multi-channel audio)) and bit rates ranging from 16 to 512 kbps per channel. The number of channels processed can be as high as 255. This allows Vorbis to be on par with MPEG-4 (AAC and TwinVQ), WMA and PAC audio, and clearly superior to MPEG-1 Layer 3 (MP3) audio. .

Ogg Vorbis is also a streaming format, allowing it to be used, for example, for Internet broadcasts, especially since this format is compatible with Icecast. The characteristics of the codec algorithm allow you to get the final file smaller than MP3 files of similar quality.

For the reproduction the console program ogg123 is used, to encode – oggenc; both have graphic housings. More details on both are in the following sections.

MP3
MP3 or MPEG-1 audio layer 3 is by far the most popular format for storing and transmitting compressed data. This format was developed by the Frauenhofer Institut, Germany. However, despite the ubiquity of the format, it should not be forgotten that the patent for MP3 encoding and decoding algorithms belongs to a single company, so the end user at any time may find themselves in a very disadvantageous environment, such as It has already happened with the developers of free MP3 data compression tools …. You can get details about the license conditions on the developers website.

WMA
The WMA format is a proprietary product of Microsoft. It failed to occupy a market segment comparable to MP3, but it has some popularity despite serious security concerns identified. At the moment, only the universal MPlayer player can play WMA files. There are no free data compression tools for this algorithm and its appearance is unlikely.