Category: Audio and video

Digital audio information (Part 3)

Codec sample rate and bit depth

Sampling is the acquisition of instantaneous values ​​(samples) of an analog signal with a certain time step in the digitization process. The frequency of this step is called the sample rate (it is also the sample or sample rate). The larger it is, the better the sound recorded and reproduced. In studio equipment, the frequency is 48 kHz, in home systems – 44.1 kHz.

Bit depth determines the quality of the recorded audio. Higher is better. The bit value, for example 32, denotes the number of bits that are allocated to record the amplitude of the signal at the time of its measurement.

Consequently, the more often (sample rate) and more accurately (bit depth) the audio signal is measured, the higher quality audio file is obtained.

Bitrate

The bit rate (literally, the information bit rate) determines the maximum amount of information that can be transmitted through the audio channel per unit of time. A high bit rate is needed to transmit a rich sound image and is not required when encoding speech. Audio recordings with a 128 Kbps bit rate are suitable for inexpensive speakers, but when accessing expensive equipment, it makes sense to get music at a 192-256 Kbps bit rate.

Convenient solution: variable bit rate encoding, change the bandwidth of the audio channel according to the quality and saturation of the musical fragment.

Audio formats

MP3 is the most popular digital audio format right now. It is widely used in file-sharing networks due to the small size of the final files (approximately 1/10 of the original audio CD file) and due to its special data compression algorithm, it provides playback quality very close to that of original. The MP3 format is compatible with absolutely all RoverMedia players, as well as all modern stereos and DVD players.

WMA is a file format developed by Microsoft to store and transmit audio information. The main advantage of WMA over MP3 is its greater compression capacity, which results in a smaller file size. The latest versions of the format, starting with Windows Media Audio 9.1, provide lossless encoding, multi-channel surround sound encoding, and speech encoding.

WAV is an audio container file format for storing a recording of a digitized audio stream. This format is mainly used to record sound from the voice recorder built into RoverMedia players and most modern devices.

FLAC (Free Lossless Audio Codec) is one of the most popular formats for lossless audio compression. Unlike MP3 and WMA formats, it does not remove any information from the audio stream when encoding the audio. Thanks to this, FLAC files are suitable not only for listening to high-quality music on RoverMedia portable media players, but even on high-quality audio equipment.

Number of audio channels

Infectious mononucleosis
Mono (from the Greek (Monos) – one) is a prefix that means the relationship with the singular.
Mono eng. Mono (monophony) is most often used as a term related to the recording and reproduction of sound.
Mono means monophonic, single channel.

Stereo
Stereo (from Greek solid, spatial)
Stereophony or stereo sound (from the ancient Greek words “stereoros” – solid, spatial and “background” – sound): recording, transmission or reproduction of sound, in which the auditory information about the location of its source is stored through sound design over two (or more) independent audio channels. …
In stereo recording, the recording is made from 2 microphones spaced a certain distance, each with a separate channel (right or left).
The result is what is called “panoramic sound”.

Digital audio information (Part 2)

Sampling frequency. A microphone connected to the sound card is used to record analog sound and convert it to digital format. The quality of the digital sound obtained depends on the number of measurements of the sound volume level per unit of time, that is, the sampling frequency. The more measurements that are made in 1 second (the higher the sampling frequency), the more accurately the “ladder” of the digital audio signal repeats the curve of the dialogue signal.

The audio sample rate is the number of sound volume measurements in one second.

The audio sample rate can vary between 8000 and 48000 sound volume measurements per second.

Audio encoding depth. Each “step” is assigned a specific value for the sound volume level. Loudness levels of sound can be viewed as a set of possible states N, for which a certain amount of information I is required, which is called audio coding depth.

Audio encoding depth is the amount of information required to encode the discrete volume levels of digital audio.

If the encoding depth is known, then the number of digital audio loudness levels can be calculated using the formula N = 2I. Let the sound encoding depth be 16 bit, then the number of sound volume levels is:

N = 2I = 216 = 65536.

During the encoding process, each sound volume level is assigned its own 16-bit binary code, the lowest sound level will correspond to the code 0000000000000000 and the highest – 1111111111111111.

The quality of digitized sound. The higher the sampling frequency and depth of the sound, the better the sound of the digitized sound. The lowest quality of digitized sound, corresponding to the quality of telephone communication, is obtained at a sampling rate of 8000 times per second, a sampling rate of 8 bits, and by recording an audio track (“mono” mode). The highest quality of digitized sound, corresponding to the quality of an audio CD, is achieved with a sampling rate of 48,000 times per second, a sampling rate of 16 bits and the recording of two audio tracks (stereo mode) .

It should be remembered that the higher the quality of the digital sound, the greater the volume of information in the audio file. It is possible to estimate the volume of information of a digital stereo sound file with a duration of 1 second with an average sound quality (16 bits, 24,000 measurements per second). To do this, the encoding depth must be multiplied by the number of measurements in 1 second and multiplied by 2 (stereo sound):

16 bit? 24,000? 2 = 768,000 bits = 96,000 bytes = 93.75 KB.

Sound editors. Sound editors allow you not only to record and play sound, but also to edit it. Digitized sound is presented in sound editors visually, so copying, moving, and deleting parts of the audio track can be easily performed with the mouse. Furthermore, you can layer audio tracks on top of each other (mix sounds) and apply various acoustic effects (echo, reverse playback, etc.).

Sound editors allow you to change the digital sound quality and volume of an audio file by changing the sample rate and encoding depth. Digitized audio can be saved uncompressed as universal WAV or compressed MP3 audio files.

By storing audio in compressed formats, low-intensity audio frequencies “excessive” for human perception are discarded, coinciding in time with high-intensity audio frequencies. Using this format allows you to compress audio files dozens of times, but it leads to irreversible loss of information (files cannot be restored in their original form).
test questions

1. How do sample rate and encoding depth affect digital audio quality?
Self-help assignments

1.22. Selective Response Mapping. The sound card performs binary encoding of the analog audio signal. How much information is needed to encode each of the 65,536 possible levels of signal intensity?
16 bits;
256 bits;
1 bit;
8 bits.

1.23. A task with a detailed answer. Estimate the volume of information in digital audio files with a duration of 10 seconds at an encoding depth and a sample rate of an audio signal that provides the minimum and maximum sound quality:

a) mono, 8 bits, 8000 measurements per second;

b) stereo, 16 bits, 48,000 measurements per second.

Digital audio information (Part 1)

The history of recording technology

The creation of sound by computer is a modern stage in the history of the development of sound technology. Let’s take a brief look at this story.

Since the late 19th century, the technical means of storing and transmitting information have developed rapidly. So in the late 1800s, the famous American inventor Thomas Edison made a phonograph.

The principle of operation of the phonograph is as follows. Speech, music, or song create sound vibrations that are transmitted to the recording pen of the phonograph. The needle, acting on the surface of the rotating wax roller, leaves in it a groove with variable depth: a sound track. When a sound is reproduced, the opposite process occurs: the movement of the reading needle along the soundtrack is accompanied by its oscillations with the same frequency. These vibrations are converted by the phonograph into an audible sound. The Edison phonograph is the first sound recording device.

The same idea served as the basis for the production of celluloid gramophone records and mechanisms that reproduce the sound recorded on them: gramophone and gramophone.

In the middle of the 20th century, an electrophone appeared, an electrical analog of a gramophone.
Analog sound representation

The soundtrack of a phonograph record is an example of a continuous form of sound recording.

The electrical signal is transmitted to the speaker of the microphone and converted into sound.

In the 20th century, the tape recorder was invented, a device for recording sound on magnetic tape. It also uses an analog form of audio storage. Only now the soundtrack is not a mechanical “pit groove”, as shown in fig. 1.1, and a line with continuously changing magnetization. With the help of a magnetic reading head, an alternating electrical signal is generated, which is emitted by an acoustic system.

Until recently, all sound transmission technology was analog. This is both telephone communication and radio communication. During a telephone conversation, the sound vibrations from the microphone membrane are converted into an alternating electrical signal that is transmitted through electrical cables. On the receiving phone, they become sound.
Audio encoding and processing

Sound information. Sound is a wave that travels through air, water, or other medium with a continuously varying intensity and frequency.

A person perceives sound waves (air vibrations) with the help of hearing in the form of sound of different volume and pitch. The greater the intensity of the sound wave, the louder the sound, the higher the frequency of the wave, the higher the pitch of the sound.
Dependence of the volume and pitch of the sound on the intensity and frequency of the sound wave.

The human ear perceives sound at a frequency of 20 vibrations per second (low sound) to 20,000 vibrations per second (high sound).

A person can perceive sound in a wide range of intensities, in which the maximum intensity is 1014 times greater than the minimum (one hundred thousand billion times). To measure the volume of sound, a special unit “decibel” (dbl) is used (Table 5.1). A decrease or increase in sound volume by 10 dB corresponds to a decrease or increase in sound intensity by 10 times

Sound volume

Sound volume in decibels:
-Lower limit of human ear sensitivity 0
-Rustling leaves 10
-Talk 60
-90 car horn
-120 jet engine
-Pain threshold 140

Sound time sampling. (Part 1)

In order for a computer to process sound, a continuous audio signal must be converted to a discrete digital form using time sampling. A continuous sound wave is divided into separate small time sections, for each section a certain value of sound intensity is set.

Therefore, the continuous dependence of the loudness of the sound at time A (t) is replaced by a discrete sequence of loudness levels. On the graph, this appears to replace a smooth curve with a sequence of “steps”).

Parameters that affect the quality of digital audio. (Part 3)

The quality of the digital sound obtained depends on the number of measurements of the sound volume level per unit of time, that is, the sampling frequency.

Audio sample rate is the number of audio volume measurements in one second.

The more measurements that are made in one second (the higher the sampling frequency), the more accurately the “ladder” of the digital audio signal repeats the curve of the analog signal.

Each “step” of the graph is assigned a certain value for the sound volume level. Loudness levels can be thought of as a set of possible N states (gradations), which require a certain amount of I information to encode, which is called audio encoding depth.

Audio encoding depth is the amount of information required to encode the discrete volume levels of digital audio.

If the known encoding depth, the number of digital audio volume levels can be calculated by the general formula N = 2 I.

For example, if the audio encoding depth is 16-bit, then the number of audio volume levels is:

N = 2 I = 2 16 = 65 536.

During the encoding process, each sound volume level is assigned its own 16-bit binary code, the lowest sound level will correspond to the code 0000000000000000 and the highest – 1111111111111111.

Digitized audio quality

Therefore, the higher the sample rate and depth of audio encoding, the better the digitized sound will sound and the better you can bring the digitized sound closer to the original sound.

The highest quality of digitized sound, corresponding to the quality of an audio CD, is achieved with a sampling rate of 48,000 times per second, a sampling rate of 16 bits and the recording of two audio tracks (stereo mode) .

It should be remembered that the higher the quality of the digital sound, the greater the volume of information in the audio file.

You can easily estimate the volume of information in a digital stereo sound file with a duration of 1 second with an average sound quality (16 bits, 24,000 measurements per second). To do this, the encoding depth must be multiplied by the number of measurements per second and multiplied by 2 channels (stereo sound):

16 bits × 24,000 × 2 = 768,000 bits = 96,000 bytes = 93.75 KB.

There are three main types of audio digits:

format – no compression;
format (lossy) – lossy compression;
format (lossless): lossless compression.
Lossy compression: technology in which there is a significant reduction of the encoded file compared to the original, due to the removal of information that is not perceived by the human ear.

The downside of this technology is the fact that the compressed file will never be identical to the original.

List of the most common lossy formats:

AAC (.m4a, .mp4, .m4p, .aac): advanced audio encoding (often in MPEG-4 container)
MP2 (MPEG Layer 2)
MP3 (MPEG Layer 3)
MPC (known as Musepack, previously called MPEGplus or MP +)
Ogg Vorbis
WMA (Windows Media Audio)

Lossless – Lossless compressed audio formats, including:

FLAC (Free Lossless Audio Codec)
APE (mono audio)
WV (WavPack)
These formats are capable of converting CD to digital format while maintaining quality. As an example, you can take a CD, convert it to WAV, then WAV to FLAC, then go back from FLAC to WAV, and then burn it to a blank CD and you have an absolutely identical copy of your source.

What format does the music sound with the best quality?
The most popular is the lossless FLAC format, and one of the most widely used CD to FLAC conversion programs is EAC (Exact Audio Copy).

Of all the parameters of digital audio, it is necessary to pay attention primarily to the following indicators:

sampling rate (precision of digitizing an analog signal in time),
bit rate (amount of information contained in a file in terms of one second).

The sample rate is the frequency at which digital audio is processed. The most common sample rate for quality audio formats is 44.1 kHz.

It is generally accepted that a high bit rate guarantees the best quality; this is true, but only if the source file is of good quality. A high quality MP3 should have a bit rate of 320 kbps, but a high quality FLAC format generally has a bit rate of 900 kbps and higher.

What is the best quality music format?
In addition to the audio formats themselves, for high-quality music sound, high-quality reproduction equipment is also needed: speakers, amplifiers, headphones.