digital audio out pass through Archives

Digital audio formats

Free Download Mp4Gain

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.

Free Download Mp4Gain

Mp4Gain Main Window

Mp4Gain Features

Free Download Mp4Gain

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. .

Benefits of “digital audio”

Benefits of “digital audio”

Digital Audio

The digitized audio signal has the following advantages:

DIGITAL AUDIO

-the possibility of infinitely long storage without loss of original quality,

-the ability to reproduce for a long time without losing the original quality,

-the possibility of infinite reproduction without loss of original quality,

-simplicity and wide possibilities of processing by modern means,

-Resistance to interference in signal transmission lines.

From CD to Super Audio CD and DVD Audio

CD (Compact Disk) is a type of removable plastic disk with optical reading of information.

In 1979, Sony and Philips proposed the Red Book standard for digital audio recording.

Analog sound is digitized and recorded as a spiral track of alternating zeros and ones (micron holes and a smooth surface) on a 12 cm polycarbonate disc, slightly thicker than a millimeter, covered with the thinner layer gold (later aluminum).

The player’s laser illuminates the disc and detects binary “zeros” and “ones”, which, after processing, are converted back to sound. It is almost impossible to mistake zero for one. Possible problems associated with read errors and scratches on the disc surface were compensated for using digital error correction.

As a result, not only did the physical dimensions of the record holder decrease compared to vinyl record, but also the musical capacity increased significantly: up to 74 minutes (the then owner of Sony wanted his favorite Beethoven Ninth Symphony to fit into a disk).

In 1982 in Langenhagen (Germany) the mass production of compact discs (CD) began with the “Alpine Symphony” by I. Strauss.

Real

High-quality audio is now recorded in Super Audio CD and DVD Audio formats, which:

use a DVD media,

use multichannel recording (up to 5.1),

sampling rate up to 192 kHz,

quantization level: up to 24 bits (each bit doubles the precision of sound transmission and, at such a depth of quantization, the dynamic range of the reproduced sounds can exceed 130 dB).

The new recording formats offer the highest quality, are expensive ($ 15 per disc), and are not popular because most listeners, sadly, don’t care too much about sound quality.

Digital audio options

The important parameters of the digital representation of sound are the sample rate of the audio signals and the quantization of bits.

Quantization rates indicate how many times per second a signal is sampled (measured in amplitude) for conversion to digital code.
For CD standard it is 44KHz (44 thousand times per second), for SACD 192KHz

The quantization bit characterizes the number of signal steps and is measured by the power of 2.

For the CD standard, 16-bit audio adapters are used, which have 65,536 quantization steps (2 to the 16 power), as in an audio CD. For standard and 24-bit SACD.

Digital audio storage

About digitizing sound has a set of signal amplitude values taken at regular intervals and can be written to file sequence numbers (amplitude values).

Two methods are widely used to encode audio information:

PCM (pulse code modulation)

ADPCM (Adaptive Relative Pulse Code Modulation)

PCM (Pulse Code Modulation) is a method of digitally encoding a signal by recording the absolute values of the amplitudes. This is how data is recorded on all audio CDs.

ADPCM (Adaptive Delta PCM) – Records signal values in relative amplitude changes (increments), allowing you to simplify data to take up less memory.

Lossless encoding (for lossless data odirovanie) allows data recovery from fully compressed (20-50%) stream.

Popular L ossless encoding algorithms:

Windows Wave (WAV) is the primary audio file format for Windows.
The Audio Interchange File Format (AIFF) is the primary audio format for the Macintosh.

L ossy encoding (lossy data encoding) enables you to achieve sound similarity of the reconstructed signal to the original with the highest possible data compression (10-1 5 times).

The basis of lossy-encoders is the use of psychoacoustic models: certain portions of the signal, in certain frequency ranges that are inaudible to the human ear, nuances (masked or inaudible frequencies) and occurs to remove them from the original signal.

Digital audio

Digital audio

Digital Audio

what happens to sound within computer programs

Digital Audio

Digital audio is a representation of analog sound used by computers and various digital devices to record and reproduce audio information. Like the frames of a movie, a digital audio signal is created from a series of sound fragments that are played when we press the play button. There are many different digital audio formats, they differ from each other in the transmission quality of the audio information.

About Pulse Code Modulation – PCM

If we talk about an acoustic sound or an analog signal, we are always talking about the propagation of sound waves in space. Whereas digital audio is only a rough description of what happens to sound or should happen within computer programs or digital devices.

This article will discuss pulse code modulation (PCM), the most common digital audio decoding system. Besides PCM, there are also DTS and Dolby Digital systems, but these are mainly applicable in the field of film and video production. Today we will not talk about them.

In pulse code modulation, a signal is read many times per second. At each reading moment the amplitude of the sound wave is recorded and reproduced. As mentioned above, a digital signal is just a rough copy of an analog signal, since an analog wave cannot be recreated with perfect precision. The values of each fragment are rounded to the nearest most accurate, then all the fragments are played and we hear a copy of the original analog sound.

“What meanings are we talking about?” – you ask. Just as analog audio is defined by frequency and amplitude, digital audio is determined by two important values: the sample rate and the bit depth. The sample rate means how many times per second the fragments of the audio signal are read, and the bit depth is the value of the dynamic range of each fragment of the audio signal.

Sampling rate

The standard 44.1 kHz sample rate used for recording audio to CDs (remember those?) Might seem like a random number. But this is not the case at all. This value was chosen based on Kotelnikov’s theorem, which essentially states that the sampling frequency must be more than 2 times higher than the maximum value of the reading frequency. As you know, the upper limit of audibility of the human ear’s frequency range is 20 kHz. It turns out that the sampling frequency must be higher than 40 kHz. An additional 4.1 kHz is added to avoid distortion, the so-called aliasing effect. In theory, 44.1 kHz should be sufficient to accurately reproduce an audio signal, however there are higher values.

For example, 48 kHz is the dominant standard in film and video production. As in the case of cinema, sound is synchronized at a frame rate of 24 frames per second. We won’t go into the details of why exactly 24 frames per second was chosen, in other words, this is the minimum frequency at which we can see a smooth, eye-pleasing image. The sample rate must match this frame rate. Using a frequency of 44.1 kHz can cause a noticeable out of sync of the picture and sound. Again, based on Kotelnikov’s theorem.

Even higher sample rates are repelled by these two base frequencies of 44.1 or 48 kHz, multiplying them by multiples of 2. That is, 88.2, 96, 192 kHz are the standard sample rates for all audio equipment. modern audio.

Bit depth

The bitness or bitness of an audio file tells us about its dynamic resolution or, more simply, clarity. You can draw an analogy with digital photography: the higher the resolution of the photo, the clearer and better the image will be.

It is important to note here that we are not talking about the loudness of the signal, but about a more realistic, clean and clear sound. More accurate transmission of the audio signal.

Bit depth can be compared to text in the book. The lower the bit depth, the less meaningful the text will make. That is, lowering the bitness leads to the fact that some letters begin to disappear from words, punctuation marks from sentences. At the moment, we will still be able to grasp the meaning of the text, but if the bit depth continues to decrease, the information will become so distorted that we simply stop understanding what we are talking about. The same goes for sound: the lower the bit depth, the more distorted we hear the sound.