Tag: digital audio to hdmi

What is digital audio?

What is digital audio?

Digital Audio
Digital Audio

How does digital audio work?

Digital Audio
Digital Audio

What is the rate? Of course, I can’t directly explain to you that “rate is bitrate”. When you play sound files with some software, you should notice a small message. For example, “128Kbps”, “1411Kbps”… Some friends also know that under normal circumstances, the larger the number in front of “Kbps”, the better the sound effect, for example, CD is “1411Kbps”. So what exactly do these numbers represent? In a nutshell, how much data is converted into sound per second. The reason CDs sound better than MP3s is that CDs have more information per second than MP3s. For example, compared to a 1411 Kbps CD file, a 128 Kbps MP3 file can convert almost 12 times less data per second than a CD. For the same song, the CD is much more delicate to listen to (of course, there is a group of people in the crowd known as “mushrooms” who can feel that the effect is the same) MP3 expresses the same content with less data and, of course, its level of detail is not as good as that of a CD.

 

2. Sampling rate.

 

Sampling rate is also a very common term. The specific form is “XXHz”, where “XX” is a specific number. Such as “44100Hz (44.1KHz)”, “32000Hz (32KHz)” and so on. As mentioned above, digital audio files are made up of many “points”, so the sample rate is actually a standard “quantity” to collect these “points”. Obviously, the sampling rate of “44100 Hz” is higher than that of “32000 Hz”, so more points are collected per time unit (1 second). The more points per unit of time, the more complete the sound information and, of course, the closer to reality. So if the guaranteed rate is the same, the file “44100Hz” is better than “32000Hz” (of course, this is not absolute).

 

————————————————–

 

lossy compression

 

In fact, we are all familiar with lossy compressed audio sources. At present, popular lossy formats mainly include MP3, WMA, OGG, MP3pro, AAC, VQF, ASF, etc.

 

2.WMV format

 

 

 

The full name of WMA is WindowsMedia Audio, which is an audio format promoted by Microsoft. The WMA format achieves a higher compression ratio by reducing the data stream while maintaining sound quality. The compression ratio can usually reach 1:18, and the generated file size is only half of the corresponding MP3 file.

 

3.MP3 format

 

 

 

The full name of MP3 is MovingPicture Experts Group Audio Layer â…¢. In a nutshell, MP3 is an audio compression technology. Since the full name of this compression method is called MPEGAAudio Layer 3, people call it MP3 for short. It was born in 1993, and its “parents” are the German FaunhofeIIS and the French Thomson.

 

MP3 uses MPEGAudio Layer 3 technology to compress music into smaller files with a compression ratio of 1:10 or even 1:12. In other words, you can compress files to a smaller size with little loss of sound quality. And it keeps the original sound quality very well. It is precisely because of MP3’s small size and high sound quality that the MP3 format has become almost synonymous with online music. The MP3 format of music per minute is only 1 MB in size, so the size of each song is only 3-4 megabytes. Use an MP3 player to uncompress (decode) MP3 files in real time so that high-quality MP3 music can be played.

What is digital audio?

What is digital audio?

Digital Audio
Digital Audio

How does digital audio work?

Digital Audio
Digital Audio

In our daily lives, we listen to all kinds of music, and most of this music is transmitted in digital form, whether it is listened to or downloaded to a computer or played on an MP3 or CD player. Of course, you will often see various formats like MP3, WMV, APE, etc., but do you understand the meaning of these formats? Below I have compiled some of this content for you, I hope it helps you.

 

1. Introduction to digital music

 

 

 

Digital audio sources, that is, digital audio formats, first referred to CDs. After the CDs were compressed, a variety of formats suitable for playback on Walkmans were derived. These compressed formats can be divided into two categories: there is lossy and lossless compression. The compression mentioned here refers to converting the audio stream encoded in PCM or WAV format to other formats after special compression processing, so as to achieve the effect of reducing the file size. Lossy/Lossless refers to whether the sound signal retained in the new file is reduced compared to the original PCM/WAV format signal after compression.

 

PCM encoding is short for PulseCode Modulation, also known as Pulse Code Modulation, which is one of the digital communication encoding methods. The sampled value is rounded and quantized according to the hierarchical unit, and the sampled value is represented by a set of binary codes to represent the amplitude of the sampled pulse.

The final form of the digital audio signal is still made up of “0/1”. They can be any permutation and combination, such as “0001110101” or “11100001010”. Of course, different combinations have different effects. Seeing this, some friends should have noticed. If the sound is recorded in the form of “00101010”, then the final form is not a “dot”, that is, a simple “change” process. The sound is continuous, how can it be recorded with “dots”? Shouldn’t the sound we hear be segment by segment? The reason is not difficult to understand. Go home and turn on the fluorescent light, can you find the fluorescent light flickering? can not? In fact, fluorescent lights flicker constantly. Have you seen cartoons? They are all connected by a grid of still images. We can also simply understand the images one by one as “dots” one by one. Man against nature

There are limits to the sense of the world, both visual and auditory. The reason cartoons can produce coherent motion is that these “dots” are an illusion that people create when human vision doesn’t respond in time. With the exception of machines, people cannot distinguish these “dots”. So is the sound. If the frequency of the sound flicker is very fast, people cannot distinguish it. Also, when the sound performs a “digital conversion of analog signals” (D/A conversion), the decoder chip has already connected these “dots” coherently, so we hear a very coherent sound.

Digital audio formats

Digital audio formats

Digital Audio

The digital audio format is a format for presenting audio data used in digital audio recording, as well as for additional storage of recorded material on a computer and other electronic media, so-called audio media.

digital audio

The audio file (a file containing a sound recording) is a computer file consisting of information about the amplitude and frequency of sound, saved for later playback on a computer or player.

Varieties of digital audio formats.

There are several concepts of audio format.

The digital representation of the audio data depends on how the digital-to-analog converter (DAC) quantizes. In sound engineering, two types of quantization are currently the most common:

pulse code modulation

sigma delta modulation

Quantization bit depth and sample rate are often specified for various audio recording and playback devices as a digital audio rendering format (24-bit / 192 kHz; 16-bit / 48 kHz).

The file format determines the structure and presentation characteristics of the audio data when stored on a PC storage device. To eliminate the redundancy of the audio data, audio codecs are used, with the help of which the audio data is compressed. There are three groups of audio file formats:

uncompressed audio formats like WAV, AIFF

lossless compressed audio formats (APE, FLAC)

lossy compressed audio formats (mp3, ogg)

Modular music file formats are highlighted. Created synthetically or from prerecorded live instrument samples, they are primarily used to create modern electronic music (MOD). Also, this can be attributed to the MIDI format, which is not a sound recording, but at the same time, using a sequencer, it allows you to record and play music using a certain set of commands in the form of text.

Digital audio media formats are used for both mass distribution of sound recordings (CD, SACD) and professional sound recording (DAT, minidisc).

For surround sound systems, sound formats can also be distinguished, which are mainly multichannel sound accompaniments for movies. These systems have complete format families from two major competitors, Digital Theater Systems Inc. – DTS and Dolby Laboratories Inc. – Dolby Digital.

The format is also called the number of channels in multichannel sound systems (5.1; 7.1). This system was originally developed for movie theaters, but has since been expanded for home theater systems.

What is digital audio and how does it work

What is digital audio and how does it work

Digital Audio

Regardless of the path chosen, after connecting the source, the sound from the source will be sent to a microprocessor called a digital audio converter (DAC for short), where there will be 2 stages:

Digital Audio

1) Conversion from analog to digital (a / d);

2) Conversion from digital to analog (d / a).

This processor is sometimes called an ad / da converter. Here, the analog audio signal is processed into digital, then redirected to the central processor and memory, and then to the storage medium. Stored digital recordings (often in .WAV format) are sent back to memory and the CPU, and then converted back to analog by the DAC.

The digital audio / MIDI sequencer allows you to record the sound of synthesizers, guitars, and microphones to files with the .wav extension. No matter how sound is transferred to the computer, it will still go to the DAC, computer memory, and hard drive. The resulting data type is called digital audio data. If you record in “CD quality” (among other things one of the lowest possible), every second of the sound is divided into 44,100 pieces. What is this data? Only numbers. But unlike the MIDI format that encodes the notes played, digital audio data is a digital representation of the actual sound wave. This is the same sound described in numbers. Can you guess that this format takes up thousands of times more space than midi data? This is true.

It is a graphical representation of digital audio data. For a computer, this is a sequence of numbers. With this data, you can perform various operations to change and improve. Outwardly, the signals appear to undergo a series of effects, but in reality what happens is a mathematical process.

How MIDI is converted to sound
You may be wondering how to convert MIDI to audio, is there a “convert” utility for that? Connect the output jacks of your synthesizer to your sound card (or audio interface, or mixer with firewire, etc.) and start recording. Analog waves go through a digital converter (DAC), are converted into numbers, and voila! you will receive digital audio data. The nice thing about a sequencer is that you first record a MIDI track and then refine it. in editors and translate it to digital audio for a perfect recording (well maybe not perfect, there is nothing perfect in the world). Yes; you are using synthesizer software, the process will be called slightly differently, but the gist is the same. The computer creates an audio track based on MIDI data and records it in audio format.

Time to process the resulting files perfectly in sync with plugins or effects. You can also save the finished tracks in MIDI format (then you can edit them at any time) and add the sound of vocals, guitars, or whatever else you want. The sequencer can work simultaneously with MIDI files and digital audio.

Effects types
One of the main and most used effects is VIBRATO.
Distinguish amplitude vibrato, when the amplitude of the signal changes periodically. The frequency of change should be small, from a few fractions of a hertz to 10-12 Hz. Tremolo is a type of amplitude vibrato. The frequency of vibration in the case of a tremolo is not less than 10-12 Hz, and the resulting signal is output in portions.

Frequency vibrato. In a non-electronic way, it was done with electric guitars. By changing the tension of the strings with a special lever, the musician changes the pitch (understand – frequency) and achieves the effect of frequency vibrato. The same can be done with synthesizers and midi keyboards using a special wheel or lever. In music editors, you can also adjust the frequency of the sound, change it within the specified or desired limits.

Ring vibrato. The signal passes through a filter, the settings of which are periodically changed. An interesting and beautiful sound is obtained due to periodic changes in the coloration of the timbre.

Effects: Reverb, Chorus, Flanger, Phaser, Delay: effects based on the delay of the signal.

Reverberation: the effect is created by mixing the main signal with copies lagged for different periods of time, obtained as a result of the reflection of various obstacles (walls, objects, etc.) The number of copies can be infinite, the reflected signal can return to reflected from another obstacle (the delay increases naturally) and again summarized with the main one. With a short delay, the effect results in an immersive and booming sound experience. .