Tag: how does digital audio work

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How digital compression works. Part 2

The next after CDDA in 1987 appeared the DAT format – Digital Audio Tape.

The sample rate was 48 kHz, the quantization bit did not change. And although the format failed, the 48 kHz sample rate took hold in recording studios, as they say, due to the convenience of digital processing.

In 1999, the DVD-Audio format was released, which made it possible to record on a disc six stereo tracks with a sampling frequency of 96 kHz and a 24-bit bit depth, or two stereo tracks with a frequency of 192 kHz, 24 bits.

In the same year, the SACD – Super Audio CD format was introduced, but the discs began to be produced only three years later. I will tell you more about this format in the DSD section.

These are the main formats that are considered the standard for digital audio recordings on media. Now let’s see how data is transmitted on a digital audio path.

The structure of the digital audio path.
When playing music, something like the following happens: the player, using a codec created in the form of a device or program, decompresses the file into a specific format (FLAC, MP3 and others) or reads data from a CD, DVD-Audio or disc SACD, receiving a standard PCM data stream … This stream is then transferred via USB, LAN, S / PDIF, PCI, etc., to the I2S converter. In turn, the converter converts the received data into so-called I2S data interface frames (not to be confused with I2C!)

I2S
I2S is a digital audio transmission serial bus. Now I2S is a standard for connecting a signal source (computer, turntable) to a digital-to-analog converter. It is through it that the vast majority of the DAC connects directly or indirectly. There are other digital audio transmission standards, but they are much less common.

I2S output (input) on PCB
I2S output (input) on PCB
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The I2S bus can consist of three, four, or even five pins:

continuous serial clock (SCK) – bit sync clock (can be called BCK or BCLK);
word selection (WS) – frame sync clock (may be called LRCK or FSYNC);
Serial data (SD): transmitted data signal (can be called DATA, SDOUT, or SDATA). As a general rule, data is transmitted from a transmitter to a receiver, but there are devices that can act as a receiver and transmitter at the same time. In this case, another contact may be present;
Serial data in (SDIN): On this pin, data moves in the receive direction, not the transmit direction.
SD or SDOUT is used to connect a D / A converter, and SDIN is used to connect an A / D converter to the I2S bus.

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How digital compression works.

Have you ever wondered how sound is reproduced on digital devices?

How is a sound signal formed from a combination of ones and zeros? I’m sure I was thinking, since I started reading! But often, even professionals only have a general idea of ​​the modern sound route. In this article, you will learn how the different formats appeared, what a digital-to-analog converter is, what types of DACs exist, and what determines the quality of sound reproduction.

PCM
As you know, in digital audio, almost any format, with rare exceptions, is recorded using a pulse code stream or a PCM stream – pulse code modulation. FLAC, MP3, WAV, Audio CD, DVD-Audio and other formats are just ways to package, “preserve” a PCM stream.

How it all began
The theoretical foundations of digital sound transmission were developed at the dawn of the 20th century, when scientists tried to transmit an audio signal over a long distance, but not by telephone, but in a rather strange way for that time.

By dividing the sound wave into small parts, it could be sent to the receiver in some kind of mathematical representation. The recipient, in turn, could restore the original waveform and listen to the recording. In addition, scientists were faced with the task of increasing the bandwidth of the “ether”.

In 1933, the theorem of V.A. Kotelnikov. In Western sources, it is called the Nyquist-Shannon theorem. Yes, Harry Nyquist was the first to raise this issue: in 1927 he calculated the minimum sampling frequency to transmit a waveform, which later got his name “Nyquist frequency”, but Kotelnikov’s theorem was published 16 years ago before.

The essence of the theorem is simple: a continuous signal can be represented as an interpolation series consisting of discrete reports, from which the signal can be reconstructed. In order to roughly restore the original state of the signal, the sampling frequency must be at least twice the upper cutoff frequency of this signal.

For many years, the theorem was not in demand, until the advent of the digital age. It was then that it found a use. In particular, the theorem was useful when developing the CDDA (Compact Disc Digital Audio) format, in common people it is called Audio CD or Red Book. The format was released by engineers at Philips and Sony in 1980 and became the standard for audio CDs.

Format characteristics:

sampling frequency – 44.1 kHz;
quantization capacity – 16 bits.

INFO
The sampling rate is the number of signal samples taken during your sampling. Measured in Hertz.
Quantization bit: the number of binary bits that express the amplitude of the signal. Measured in bits.
The 44.1 kHz sampling frequency was calculated from Kotelnikov’s theorem. It is believed that the hearing of the average person cannot pick up sound beyond 19-22 kHz. The frequency was probably 22 kHz and was chosen as the upper limit.

22,000 × 2 = 44,000 + 100 = 44,100 Hertz

Where does 100 Hertz come from? There is a version that this is a small margin in case of errors or oversampling. In fact, Sony chose this frequency for its compatibility with the PAL transmission standard.

The bit depth of the CDDA format is 16 bits, or 65,536 samples, which equates to a dynamic range of approximately 96 dB. Such a large number of samples were not chosen by chance. Firstly, due to the strong influence of quantization noise, and secondly, to provide a formal dynamic range superior to that of the main competitors at the time – cassette records and vinyl records. I’ll cover this in more detail in the section on digital to analog converters.

The development of PCM continued on the principle of multiplying by two. Other sample rates appeared: first, the 48 kHz sample rate was added, and then the frequencies based on it were 96, 192, and 384 kHz. The 44.1 kHz frequency was also doubled to 88.2, 176.4, and 352.8 kHz. Bit depth increased from 16 to 24 and then to 32 bits.

Audio encoding: secrets revealed

Audio settings for video capture and transmission.

As people directly related to the AV sphere, we constantly talk about audio coding and audio codecs, but what is it? An audio codec is essentially a device or algorithm that can encode and decode a digital audio signal.

In practice, the audio waves that travel through the air are continuous analog signals. The signals are converted to digital form by a device called an analog-to-digital converter (ADC), and the reverse converter is called a digital-to-analog converter (DAC). The codec lies between these two functions and it is he who allows you to adjust some important parameters for the successful capture, recording and transmission of an audio signal: the codec algorithm, the sampling frequency, the bit width and the speed of the audio signal. data.

The three most popular audio codecs are Pulse-Code Modulation (PCM), MP3, and Advanced Audio Coding (AAC). The choice of codec determines the compression rate and the recording quality. PCM is a codec used by computers, CDs, digital phones, and sometimes SACD. The PCM signal source is sampled at regular intervals, and each sample is the digital amplitude of the analog signal. PCM is the simplest option for digitizing an analog signal.

With the correct parameters, this digitized signal can be fully converted to analog without any loss. But this codec, which provides almost complete identity with the original audio, is unfortunately not very cheap, which results in large files, and these files are not suitable for streaming. We recommend using PCM to record digital images for your sources or when doing audio post-processing.

Fortunately, we always have the option of choosing a different codec that can compress digital data (versus PCM) based on some helpful observations on the behavior of sound waves. But in this case, you have to make a compromise: all alternative algorithms are associated with “losses”, since it is impossible to completely restore the original signal, but nevertheless the result is still so good that most users will not be able to to catch the difference.

MP3 is an audio encoding format that uses a digital data compression algorithm that allows you to save the audio signal in smaller files. The MP3 codec is the most used by users to record and store music files. We recommend using MP3 to stream audio content as it requires less network bandwidth.

AAC is a newer audio encoding algorithm that is the successor to MP3. AAC has become the standard for MPEG-2 and MPEG-4 formats. In fact, this is also a digital data compression codec, but with less quality loss than MP3 when encoded with the same bit rate. We recommend using this codec for online streaming.

Sampling frequency (kHz, kHz)
Sample rate (or sample rate): the frequency with which the signal is digitized, stored, processed or converted from analog to digital. Time sampling means that the signal is represented by several of its samples (samples) taken at regular intervals.

Measured in hertz (Hz, Hz) or kilohertz (kHz, kHz,) 1 kHz equals 1000 Hz. For example, 44,100 samples per second can be labeled 44,100 Hz or 44.1 kHz. The selected sample rate will determine the maximum playback frequency and, as follows from Kotelnikov’s theorem, to fully restore the original signal, the sample rate must be twice the highest frequency in the signal spectrum.

As you know, the human ear is capable of picking up frequencies between 20 Hz and 20 kHz. Given these parameters and the values ​​shown in the following table, you can understand why 44.1 kHz was chosen as the sampling frequency for CD and is still considered a very good frequency for recording.

What are the problems with digital audio?

As with many areas of technology, there is no single standard for digital audio.

It can be presented in various standards: AES / EBU 110 Ohm, AES-ID3 75 Ohm, S / PDIF 75 Ohm, Optical Toslink, among others. The sampling frequency can be from 32 kHz to 192 kHz with different bit depths. To work with all the variety of standards in a serious studio, you need to have an interface unit, better a digital audio converter or a sample rate converter.

What are the problems with digital video?
Digital video (SDI) is similar in some respects to analog video. In it, the quality of the cables and connectors is also important for normal operation, the loss of high frequencies of the signal in them also affects the quality of the signal. Due to many factors that affect the analog signal, fluctuations can appear in digital systems, at a certain level of which there is a complete blockage of the image (clipping effect *). A little lost in digital video can have far more serious consequences than a pixel lost in analog. When working with digital video, restoration of signal quality (equalization of the frequency spectrum and restoration of clock frequency) is often required. The format (“language”) of a digital signal is very important for its correct transmission, since the transmission protocols are very specific.
Level incompatibility is a rare problem in analog technology. Digital signals, however, can have different and incompatible levels: TTL, ECL or others. Another problem with digital signals is the adaptation of the load capacity of the digital inputs and outputs, which must also be addressed.

What is the easiest way to input a digital video signal into a computer?
The easiest and cheapest way is to use a DV video source and a Firewire® card on your computer (or the built-in interface on many modern computers). The entry procedure is simple and fast. For analog video, you can use an analog video capture card or an external analog video to DV converter connected to the Firewire® card.

Why do I sometimes have difficulties with the DV format?
The digital video format that uses a DV or mini-DV cassette and Firewire® technology has a very high bit rate, which limits the length of the connecting cable. Attempting to use long cables will cause many bit stream problems, such as clipping effect * when the image is completely lost. Another problem is a consequence of two-way communication between devices connected via Firewire® and manifests itself when trying to randomly connect multiple DV devices.

What is a device for embedding (extracting) digital audio into an SDI signal?
The total digital stream of digital serial video can include multiple channels of digital audio. An SDI embedder is used to insert digital audio into an SDI signal, and an SDI embedder is used to extract digital audio from a mixed stream.