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Formats: what is digital sound Part 3

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Formats: what is digital sound Part 3

digital sound

JointStereo, which is one of the features of MP3, means that instead of encoding stereo as two independent channels, it encodes calls. center channel and the difference from the original stereo channels.

DIGITAL SOUND

Many audio components on stereo channels are the same, and encoding them on the common channel allows you to free up additional bandwidth for more detailed encoding of the difference, leading to improved quality.

Be sure to mention the variable bit rate or VBR. This means that the encoder changes the compression ratio on the fly, depending on the nature of the sound. This approach results in a smaller total file size or, if quality requirements increase, the same file size may produce better sound.

MP3 Pro

Introduced in 2001, the MP3 Pro codec was developed by Coding Technologies in association with Thomson Multimedia. It is MP3-based and as a result turned out to be fully MP3 backward compatible and only partially forward compatible. It uses SBR (Spectral Band Replication) technology, so the codec provides good quality at low bit rates. However, the encoding quality at medium to high bit rates is inferior to almost all other codecs. As a result, MP3 Pro is used more for streaming on the Internet and demonstrating snippets of new musical compositions.

WMA

The WMA codec, or Microsoft Windows Media Audio, is a serious alternative to MP3. Files in this format have the extensions .WMA and .ASF, have a clear advantage over MP3 at low data rates (bitrates) and lose it when the data feed rate to the codec is increased.

Based on WMA, the WMA DRM standard has been developed to provide copy protection so appreciated by record companies. Files based on this format can be recorded on playback devices such as MP3 flash players, but cannot be copied from there.

ATTRAC

ATRAC (Adaptive TRansform Acoustic Coding) is based on a stereo audio format with 16-bit quantization and a sample rate of 44.1 kHz. When compressed, each frame is divided into 52 frequency bands, resulting in a transmission rate of 292 kbps (1: 5 compression). Used in MiniDisk system.

Ogg Vorbis

The Ogg Vorbis format is a relatively new universal lossy audio recording format. It belongs to the same type of audio compression formats as MP3 and WMA, and the psychoacoustic model that describes the characteristics of the human ear, according to which compression is performed, is similar in principle to MP3. The radical difference of this format was the mathematical processing and the practical implementation of this model. In this format, the maximum threshold sample rate is not 44 kHz as in MP3, but 48, which theoretically improves the sound quality. It should also be noted that the theoretical number of channels is not limited to two, as usual, but reaches 255. Files encoded in this format are smaller than the same MP3 files. The spread of the format was slowed by insufficient support from hardware manufacturers.

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Formats: what is digital sound Part 2

Formats: what is digital sound Part 2

Digital Sound

Historically, digital sound was undoubtedly the initiative of company engineers who adopted Philips-developed Audio-CDs, also called CDDA – Compact Disk Digital Audio.

DIGITAL SOUND

As a result, digital audio media led to a massive transition in recording studios to digital DAT tape recorders and digital editing equipment with S / PDIF and other interfaces. And then digital sound began to penetrate deeper into our lives from CD players, and as it was transmitted via S / PDIF, it became digital switches, equalizers, and noise reduction systems. Today this series ends with Dolby Digital surround sound processors.

Who needs it

CDDA’s sound quality is satisfactory for most end users, ie listeners, but the amount of data required to present sound in this way is critical. As a result, several compressed digital audio formats appear, one of which is the old MS ADPCM, and among which are quite acceptable Sony ATRAC, PASC or Fraunhoffer MP3. Each of the encoding methods has an important characteristic – the bit rate, with which the compressed information enters the decoder when the audio signal is restored.

For example, when you talk on a cell phone, the sound of your voices is digitally converted and compressed, degrading its performance. Various algorithms compress speech hundreds of times, preserving the basic characteristics.

Let’s move on to specific audio file formats and audio compression formats. The most common format today is, of course, MP3. However, historically, to understand the evolution of sound formats, it is necessary to start with a different type of file, with the extension .WAV.

Variety of formats

Wav

It is the primary format for many, many digital audio playback systems and is used as a standard audio file format on personal computers. In addition, it has a strong set of specifications, which has grown considerably lately. Its full name is Microsoft RIFF / WAVE – Resource Interchange File Format / Wave – Resource Interchange File Format / Waveform, and it was created by Microsoft and Intel engineers. In turn, WAV is short for Waveform Audio File Format.

Apple AIFF

This type of file is standard for Apple Macintosh systems and sound processing systems based on it. Apple AIFF stands for Audio Interchange File Format: an audio interchange file format, it is somewhat similar to WAV. Its peculiarity is that it allows additional information to be placed next to the sound wave, in particular WaveTable samples (examples of the instrument sound together with synthesizer parameters), which improves the quality of the final result. Although today Apple computers are capable of playing files in almost any format, including MP3.

RAW

Yes, this is not just the image format in which some digital cameras take pictures. In fact, RAW is the call. “Pure Digitization”, which does not contain a title and only contains a sequence of samples of a sound wave. Typically, the scan is stored in 16-bit format.

MP3

The most popular compression format today is MP3. The MP3 (MPEG Layer 3) format was developed, after several intermediate formats, by the Fraunhofer Institute in Germany. Actually, the .MP3 format is based on fooling the human ear. After some research, it turned out that human hearing tends to adapt to the appearance of new sounds, which is expressed in an increase in the hearing threshold. Therefore, some sounds are capable of masking (that is, making them subjectively inaudible) others. So in this format, some of the sounds that, according to the corresponding theory, are made inaudible, are simply removed from the general sound. The resulting “semi-finished product” is then encoded using the Hoffman method. Be sure to note that in MP3 format, programs that compress the sound of the original are not standardized, that is, each competent programmer can implement their own compression scheme. And only decoders meet the standards, which leads to the fact that the quality of MP3 playback does not always depend on the player that plays this file. Due to the different skills and predilections of implementers of various encoders, some of them are better at handling symphonic music, some with rock and metal, some with rap and rave, etc.

Formats: what is digital sound

Formats: what is digital sound

Digital Sound

Sound plays an increasingly important role in the modern world, having long since separated itself from the close link with the image that emerged during the heyday of television and cinema.

digital sound

Modern multimedia equipment has the widest possibilities not only for playback, but even for changing the sound. It is no longer a dead record, a static reproduction of events from the past, firmly imprinted on its medium. The most important role in transforming our ideas about sound was played by the development of a digital method of recording sound, turning it into a data stream that can be easily and naturally operated with modern devices.

The basics of “numbers”

Each of the multimedia devices on sale today, be it a CD player, a voice recorder or a flash memory player, uses many different types of presentation of data streams, which are then converted into sound. And even more sound formats used for professional purposes have been invented. An inexperienced buyer is forced to gather information about designations on boxes and devices from a variety of sources, often receiving incorrect information or even more confusion.

Almost all devices in the “Portable Audio” section of the ZOOM.CNews.ru catalog support multiple sound formats at the same time, and many devices that do not belong in this category are also tagged with support for playing sound files. To help our reader, we decided to create a short glossary of abbreviations and talk about the most common formats. We plan to leave it open for updates and modifications, adding new formats and describing in more detail the advantages and disadvantages of the already common or forgotten ones.

A little theory

To begin with, remember that digital sound is nothing more than a collection of numbers. The determining factor is the system by which sound as air pressure is converted into data streams and encoded for further processing and reproduction. Consequently, digital sound is usually included in computer files with various extensions, which more often (but not always) can determine their format. And the same concept of format can have, paradoxically, two meanings. First, the format may exist as a general characteristic that includes both the type and the physical characteristics of the medium (disk or cassette), the method of recording, the principles of encoding, and protection against errors. Second, the format can only be understood as the method of audio encoding and compression, as standard means are used for transfer, for example,

Analog sound, unlike digital, is reproduced on analog devices and has a number of significant differences. While not a data stream, analog audio is represented as a continuous electrical signal that represents a change in the sound wave. To translate it into digital format, the sound is “digitized”, that is, it is divided into certain segments, in which the numerical value of the amplitude is fixed at that moment. We will not delve into the principles of digital sound creation, but it is absolutely necessary to note that the more often a sound segment is divided and its characteristics described, the clearer and more complete the sound image itself is created.

This process generates a large amount of data that describes the sound, and it is clear that each digital audio format is nothing more than a compromise between the need to present the sound as loudly as possible and the memory limitations of a computer or device. Of reproduction.

A little more theory. In most cases, the human ear perceives sound with a frequency no higher than 22,000 Hz and, to describe it fully in digital form, a sampling frequency of at least 44.1 kHz is required. Since it is absolutely impossible to determine the value of the signal at any given moment, during digitization quantization occurs, that is, the replacement of the actual values of the signal by approximate values. The more levels of audio quantization, the more accurately the signal level is described. As a result, each standard CD carries an audio signal with a sampling frequency of the same 44.1 kHz and a 16-bit quantization level, and in some devices, sampling is done at 48 kHz.

The benefits of digital audio

And now, briefly on why this algorithm was developed. Digital sound has enormous advantages over analog, although we must not forget its certain disadvantages. The main value of digital sound is the possibility of infinitely long storage and playback

Digital sound

Digital sound

Digital Sound

Unlike the analog signal, the digital signal does not simulate acoustic sound.

Digital Sound

Digital sound assigns digital values to individual points in time that reflect the height of the amplitude at a given point. The second difference between digital and analog audio is that digital audio is discrete.

As you know, digital information is stored in bytes, each of which consists of 8 bits. A bit is the smallest unit of digital information that can take only two values: zero or one.

So how do you convert a continuous analog signal into a sequence of zeros and ones, and even link this information correctly to the timeline? Converting audio to digital format is divided into two operations: sampling and quantizing. Sampling – sampling and quantization time – amplitude. It is these operations that your audio interface performs.

Any audio interface has an ADC (analog-to-digital converter) and a DAC (digital-to-analog converter). Let’s consider how audio recording works when used to record a microphone and a computer with an audio interface attached.

When you speak, your voice creates fluctuations in air pressure, which the microphone picks up and translates into an alternating voltage electrical signal. The received electrical signal is very weak, so it is amplified and then sent to the audio interface for digital conversion. Based on its internal clock, the ADC divides time into many different points. Time sampling occurs according to the set frequency, which indicates how many dots will be divided by 1 second of sound. At each received time point, the ADC measures the voltage of the input signal and assigns the corresponding digit to the amplitude value. The data obtained as a result of this conversion can be saved on a computer.

Digital sound

When you start playing the audio file, the reverse process will start. The digital information will be sent from the computer to your audio interface. Your DAC will provide a reverse conversion of the received information into a continuous electrical signal with alternating voltage. The signal will then be amplified and reproduced through your speaker system.

So what is the sample rate to get digital sound that can then be converted back to analog? According to Kotelnikov’s theorem, each band-limited signal can be sampled and then recovered in digital form, as long as the sample rate is at least twice the highest frequency of the original signal.

This means that our signal must have a maximum frequency that will never be exceeded. When we set the highest frequency, all that remains is to multiply it by two and get the desired sample rate. Also, according to the theorem, all frequencies above half the sample rate must be removed from the input signal.

Since a person hears sounds from 20 Hz to 20 kHz, a sample rate of 40 kHz should be adequate to encode any sound audible to a person. With a small margin for the filter, which is calculated before converting to digital format, in the CD audio standard, sounds above 22,050 Hz are cut off and the sample rate is 44,100 Hz.

Now let’s see exactly what numbers the ADC assigns to the amplitude values when converting an analog signal.

The computer can assign a finite number of values to the amplitude. As mentioned above, any information in a computer is a sequence of bits, each of which takes on values of zero or one.

A numeric expression of n bits assumes 2 n different variants of values, that is, 2 n different variants of sequences of zeros and ones. The table shows the sequence options for n = 2,3,4.

The quality of digital sound reproduction.

The quality of digital sound reproduction.

Digital Sound Quality

Audio coding. Before converting music to another format, you must “unzip” it to WAV.

SOUND QUALITY

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

A person perceives sound waves (air vibrations) with the help of hearing in the form of sound of varying 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 (Fig. 1.1).

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). Decreasing or increasing the sound volume by 10 dB corresponds to a decrease or increase in sound intensity by 10 times.

The sound volume the volume of the
sound in decibels
lower limit of human ear sensitivity 0
leaf whisper 10
Conversation 60
Gudok Vehicle 90
Jet engine 120
Pain threshold 140

Provisional discretization sound. 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” (Fig. 1.2).

Sync Audio Sampling

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 audio 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.

How is sound represented digitally?

How is sound represented digitally?

Digital Representation of Sound

The original shape of an audio signal (a continuous change in amplitude over time) is represented digitally by “cross-sampling”, in time and level.

Representing Sound Digitally

According to Kotelnikov’s theorem, any continuous process with a limited spectrum can be completely described by a discrete sequence of its instantaneous values, following with a frequency at least twice the frequency of the highest harmonic of the process; the sampling frequency Fd of instantaneous values (samples) is called the sampling frequency.

It follows from the theorem that a signal with a frequency Fa can be successfully sampled in time at a frequency 2Fa only if it is a pure sinusoid, because any deviation from the sinusoidal shape leads the spectrum to go beyond the frequency Fa. Therefore, for temporal sampling of an arbitrary audio signal (which normally has, as is known, a spectrum that falls smoothly), it is necessary to select a sampling frequency with a margin or to forcefully limit the spectrum of the input signal below half the sample rate.

Simultaneously with time sampling, amplitude sampling is performed: measurement of instantaneous amplitude values and their representation in the form of numerical values with some precision. The precision of the measurement (binary width N of the obtained discrete value) determines the signal-to-noise ratio and the dynamic range of the signal (theoretically these are reciprocal values, but any real path also has its own level of noise and interference).

The resulting stream of numbers (a series of binary digits) that describe an audio signal is called Pulse Code Modulation (PCM), since each pulse of a time-sampled signal is represented by its own digital code.

Linear quantization is most often used when the numerical value of the sample is proportional to the amplitude of the signal. Due to the logarithmic nature of hearing, logarithmic quantization, when the numerical value is proportional to the magnitude of the signal in decibels, would be more appropriate, but this is fraught with difficulties of a purely technical nature.

Time sampling and amplitude quantization of the signal inevitably introduce noise distortions in the signal, the level of which is generally estimated using the formula 6N + 10lg (Fdiscr / 2Fmax) + C (dB), where the constant C varies for different types of signals: for a pure sinusoid it is 1.7 dB, for sound signals – from -15 to 2 dB. Thus, it can be seen that a decrease in noise in the operating frequency band 0..Fmax leads not only to an increase in the bit depth of the sample, but also to an increase in the sample rate relative to 2Fmax, as the quantization noise is “smeared” across the band up to the sample rate, and the audio information occupies only the smallest part of this strip.

Most modern digital audio systems use the standard 44.1 and 48 kHz sample rates, but the frequency range of the signal is usually limited to about 20 kHz to keep it clear of the theoretical limit. Also the most common is 16-bit level quantization, which provides a limit signal-to-noise ratio of approximately 98 dB. The studio equipment uses higher resolutions: 18, 20, and 24-bit quantization at 56, 96, and 192 kHz sample rates. This is done to preserve the higher harmonics of the sound signal, which are not directly perceived by the ear, but affect the formation of the overall sound image.

To digitize lower-quality, narrow-band signals, you can lower the sample rate and bit depth; for example, telephone lines use 7 or 8 bit digitization with frequencies 8..12 kHz.

The representation of an analog signal in digital form is also called Pulse Code Modulation (PCM), since the signal is represented as a series of pulses of constant frequency (time sampling), the amplitude of which is transmitted digitally (amplitude sampling ). A PCM stream can be parallel, when all the bits in each sample are transmitted simultaneously over several lines with one sampling frequency, or sequential, when the bits are transmitted one after another at a higher frequency on a line.

Digital sound itself and related elements are often denoted by the general term Digital Audio; The analog and digital portions of a sound system are called the Analog Domain and Digital Domain.