Digital sound: DSD vs PCM
Digital sound. How many myths revolve around this phrase. How many disputes arose between lovers of convenience and quality numbers and supporters of "live air" vinyl sound multiplied by "warm tube" sound. In addition, there is a lot of controversy between the lovers of the "figures": is 16x44.1 enough or is 24x192 necessary? Which is better: multibit or delta sigma? CDDA or SACD? PCM or DSD? In this article I will try to explain in simple language the basics of digital sound, as well as I’ll dwell more on the comparison of two types of encoding of an analog signal to digital: DSD and PCM .
To begin, let's answer the question, what is digital sound? How does it differ from analog? In short, in a mathematical language, the analog sound is a continuous function , the digital sound is a discrete function . What does it mean?
If you draw in your imagination a graph of a sine wave (this is how a sound wave is most often depicted): then, no matter how much we increase it, trying to see all the details, we will always see a smooth, smooth line: this is an analogue sound signal (Fig. 1).
Fig. 1. Analog signal
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Analog sound (recording) has many parameters with which you can assess its quality. Consider the three most important: frequency range, dynamic range, distortion.
Frequency range - a set of frequencies contained in the sound. It is considered that the frequency range of human hearing is 20 ... 20.000 Hz (sometimes 16 - 22.000 Hz is indicated). The frequency range of the music itself does not represent any interest in terms of quality assessment (for example, the frequency range of the same airplane will be very wide, and the vocal tenor part is much narrower). The quality parameter of, say, headphones is the potential frequency range, and it is estimated using the amplitude-frequency response (AFC). An ideal frequency response — a straight line throughout the frequency range of the ear — means that the sound source does not amplify or attenuate any particular frequencies, which means that the extracted sound coincides with the original.
Fig. 2. Frequency response MP3 file 256 kbps
Dynamic range (DD) - the difference between the quietest and loudest sound. Measured volume in decibels (dB). It is considered that the maximum volume that does not cause injuries to a person is 130 dB - the sound of an airplane taking off, and the minimum audible volume - 5 ... 10 dB - at the level of leaves rustling in windy weather. Naturally, the rustling of leaves against the background of an airplane taking off will be impossible to disassemble, and listening to music with a level of 130 dB is extremely unpleasant. Therefore, it is assumed that a comfortable DD for listening to music is 80 ... 100 dB.
Distortion is nothing but the deviation of the signal from the original.
What happens when digitizing analog audio? We will not delve into the technical aspects, we will analyze everything, as they say, on paper: for this we draw our imaginary “ideal” sinusoid and measure the signal size at regular intervals (this process is called discretization or quantization ): we will get some consistent set of values ​​- this will be our digital signal obtained by the method of pulse code modulation (PCM) (Fig. 3).
Fig. 3. Convert analog signal to PCM
The two main parameters for the quality of a PCM signal are frequency and bit depth. Frequency is the number of measurements in one second; the more there are, the more accurately the signal is transmitted. Frequency is measured in hertz: 44100 Hz, 192000 Hz, etc. The width is the number of possible values ​​of the signal (the accuracy of the transfer value). The more options - the more accurate the signal. Bit width is measured in bits: 16 bit (65.536 possible values, DD 96 dB), 24 bit (16.777.216 values, DD 144 dB), etc.
But this is not the only way to represent a sound wave in digital form. There is a way to get rid of such a parameter as digit capacity, leaving only two amplitude levels: -100% and + 100% (0 or 1). To achieve this, without losing quality, you need to repeatedly increase the frequency of reading the signal magnitude (Fig. 4).
Fig. 4. Convert analog signal to DSD
This type of representation of digital sound is called pulse-density modulation, most often it uses the abbreviation DSD. In fact, the only quality parameter of such a signal is frequency. But since the frequencies used are very high (from 2.822.400 Hz), these numbers are difficult to remember, it is common to divide the DSD signal frequency by 44.100 Hz. The resulting number is an indicator of quality: DSD64 (DD 120 dB), DSD128, DSD256, etc.
But digitizing an analog signal is half the battle. To listen to digital music, you need to perform the inverse transformation. To begin, consider how to turn a digital DSD stream into sound. As we already know, this stream is a high-frequency (2.8 MHz or more) two-level signal, the average value of this signal varies with the sound frequency. That is, if you approach the solution of the problem as simply as possible, you need to filter out all the high-frequency components of the DSD stream, leaving only the useful sound signal (frequencies up to 20 ... 22 kHz). This is done using an analog low- pass filter (LPF). The simplest low-pass filter is an RC string . The signal received, after passing through this chain, is shown in fig. five.
Fig. 5. Recovery of analog signal from DSD
As you can see, the resulting graph only remotely resembles the original sinusoid. But do not forget that we have “applied” the simplest filter; by improving the filter circuit, we can achieve the almost complete absence of high-frequency noise and get analog sound with good quality indicators.
To recover an analog signal from digital PCM, it is not enough just an analog LPF, you need to decrypt the digital data, for this purpose digital-to-analog converters (DACs) are used. They can be of different types, but it is not included in the tasks of this article to describe them all. Let us dwell on the 2 most common types of audio equipment. Firstly, this is the so-called ladder type DAC (it is also called multibit). As you probably guessed, such a DAC converts a PCM digital data stream into a stream of sound signal values ​​that look like a ladder on the graph (Fig. 6). As in the case of DSD, it is imperative to use an analog filter to smooth the “steps”.
Fig. 6. Recover Analog Signal from PCM
Often, such converters use intermediate resampling of a digital PCM signal to higher frequencies (for example, 192 kHz): this reduces the “steps”, which allows us to simplify the analog filter circuit.
The second type of D / A converter, delta-sigma, uses oversampling to even higher values ​​of the frequency while simultaneously decreasing the bit depth to one bit. Nothing like? This is the same DSD signal! How to further process such a signal and turn it into analog, we have already considered above.
Where can we find each of the encoding methods? PCM format is very common: CDDA discs, DVD Audio, MP3, FLAC, ALAC, AAC files, sound in movies, and further, and further, it is easier to say when non-PCM. Super Audio CDs, DSDs, DSF, DFF files are DSD format. What is better? When playing any format we get better sound?
The articles on the DSD format describe many advantages over PCM, but are all the described advantages true or are these myths invented for ordinary people who do not understand the technical component to win over the market that is densely occupied by the PCM format? Let's go through the list briefly.
So, what to choose DSD or PCM? There is no unambiguous answer and cannot be: PCM 24 bit 92 kHz and DSD128, for example, are very similar in quality characteristics, and these characteristics are better than the equipment on which these formats will be played, which means a further increase in the quality of digital formats for playback on This stage is not appropriate. When evaluating the quality of sound in various high-definition formats, subjective sensations come to the fore, because the human brain is not the only person with quality: the equipment design, its cost, and, most importantly, the listener's mood and mood give a much greater effect on the sensations of listening to music. So, choose what you personally like and do not impose your opinion on others. Enjoy everyone!
To begin, let's answer the question, what is digital sound? How does it differ from analog? In short, in a mathematical language, the analog sound is a continuous function , the digital sound is a discrete function . What does it mean?
Analog signal
If you draw in your imagination a graph of a sine wave (this is how a sound wave is most often depicted): then, no matter how much we increase it, trying to see all the details, we will always see a smooth, smooth line: this is an analogue sound signal (Fig. 1).
Fig. 1. Analog signal
')
Analog sound (recording) has many parameters with which you can assess its quality. Consider the three most important: frequency range, dynamic range, distortion.
Frequency range - a set of frequencies contained in the sound. It is considered that the frequency range of human hearing is 20 ... 20.000 Hz (sometimes 16 - 22.000 Hz is indicated). The frequency range of the music itself does not represent any interest in terms of quality assessment (for example, the frequency range of the same airplane will be very wide, and the vocal tenor part is much narrower). The quality parameter of, say, headphones is the potential frequency range, and it is estimated using the amplitude-frequency response (AFC). An ideal frequency response — a straight line throughout the frequency range of the ear — means that the sound source does not amplify or attenuate any particular frequencies, which means that the extracted sound coincides with the original.
Fig. 2. Frequency response MP3 file 256 kbps
Dynamic range (DD) - the difference between the quietest and loudest sound. Measured volume in decibels (dB). It is considered that the maximum volume that does not cause injuries to a person is 130 dB - the sound of an airplane taking off, and the minimum audible volume - 5 ... 10 dB - at the level of leaves rustling in windy weather. Naturally, the rustling of leaves against the background of an airplane taking off will be impossible to disassemble, and listening to music with a level of 130 dB is extremely unpleasant. Therefore, it is assumed that a comfortable DD for listening to music is 80 ... 100 dB.
Distortion is nothing but the deviation of the signal from the original.
Principles of digital audio
What happens when digitizing analog audio? We will not delve into the technical aspects, we will analyze everything, as they say, on paper: for this we draw our imaginary “ideal” sinusoid and measure the signal size at regular intervals (this process is called discretization or quantization ): we will get some consistent set of values ​​- this will be our digital signal obtained by the method of pulse code modulation (PCM) (Fig. 3).
Fig. 3. Convert analog signal to PCM
The two main parameters for the quality of a PCM signal are frequency and bit depth. Frequency is the number of measurements in one second; the more there are, the more accurately the signal is transmitted. Frequency is measured in hertz: 44100 Hz, 192000 Hz, etc. The width is the number of possible values ​​of the signal (the accuracy of the transfer value). The more options - the more accurate the signal. Bit width is measured in bits: 16 bit (65.536 possible values, DD 96 dB), 24 bit (16.777.216 values, DD 144 dB), etc.
But this is not the only way to represent a sound wave in digital form. There is a way to get rid of such a parameter as digit capacity, leaving only two amplitude levels: -100% and + 100% (0 or 1). To achieve this, without losing quality, you need to repeatedly increase the frequency of reading the signal magnitude (Fig. 4).
Fig. 4. Convert analog signal to DSD
This type of representation of digital sound is called pulse-density modulation, most often it uses the abbreviation DSD. In fact, the only quality parameter of such a signal is frequency. But since the frequencies used are very high (from 2.822.400 Hz), these numbers are difficult to remember, it is common to divide the DSD signal frequency by 44.100 Hz. The resulting number is an indicator of quality: DSD64 (DD 120 dB), DSD128, DSD256, etc.
Recovery of analog signal from the "numbers"
But digitizing an analog signal is half the battle. To listen to digital music, you need to perform the inverse transformation. To begin, consider how to turn a digital DSD stream into sound. As we already know, this stream is a high-frequency (2.8 MHz or more) two-level signal, the average value of this signal varies with the sound frequency. That is, if you approach the solution of the problem as simply as possible, you need to filter out all the high-frequency components of the DSD stream, leaving only the useful sound signal (frequencies up to 20 ... 22 kHz). This is done using an analog low- pass filter (LPF). The simplest low-pass filter is an RC string . The signal received, after passing through this chain, is shown in fig. five.
Fig. 5. Recovery of analog signal from DSD
As you can see, the resulting graph only remotely resembles the original sinusoid. But do not forget that we have “applied” the simplest filter; by improving the filter circuit, we can achieve the almost complete absence of high-frequency noise and get analog sound with good quality indicators.
To recover an analog signal from digital PCM, it is not enough just an analog LPF, you need to decrypt the digital data, for this purpose digital-to-analog converters (DACs) are used. They can be of different types, but it is not included in the tasks of this article to describe them all. Let us dwell on the 2 most common types of audio equipment. Firstly, this is the so-called ladder type DAC (it is also called multibit). As you probably guessed, such a DAC converts a PCM digital data stream into a stream of sound signal values ​​that look like a ladder on the graph (Fig. 6). As in the case of DSD, it is imperative to use an analog filter to smooth the “steps”.
Fig. 6. Recover Analog Signal from PCM
Often, such converters use intermediate resampling of a digital PCM signal to higher frequencies (for example, 192 kHz): this reduces the “steps”, which allows us to simplify the analog filter circuit.
The second type of D / A converter, delta-sigma, uses oversampling to even higher values ​​of the frequency while simultaneously decreasing the bit depth to one bit. Nothing like? This is the same DSD signal! How to further process such a signal and turn it into analog, we have already considered above.
PCM and DSD application advantages / disadvantages
Where can we find each of the encoding methods? PCM format is very common: CDDA discs, DVD Audio, MP3, FLAC, ALAC, AAC files, sound in movies, and further, and further, it is easier to say when non-PCM. Super Audio CDs, DSDs, DSF, DFF files are DSD format. What is better? When playing any format we get better sound?
The articles on the DSD format describe many advantages over PCM, but are all the described advantages true or are these myths invented for ordinary people who do not understand the technical component to win over the market that is densely occupied by the PCM format? Let's go through the list briefly.
- The first advantage that DSD supporters like to bring is rather vague - noise immunity and reduction of the effect of errors. It is strange to hear about different noise immunity in the digital world: both formats are subject to interference just as much as a digital book is susceptible to interference. The storage duration of any digital format or the quality of its transfer between devices depends only on the carrier / transmission method, but not on the format itself. So, the noise immunity is the same. What about reducing the impact of errors? Suppose we store 2 albums on optical disks (one PCM, another DSD), what happens if you scratch the disc? Reading the damaged media will cause errors, but how critical are they? In PCM coding, multi-digit numbers are used, the error in the high-order bit is very critical (as an example, the difference between decimal numbers 11 and 91): by hearing this will feel like a click. In DSD coding, one bit of information has little weight in the general stream, infrequent errors will only cause an increase in the background noise, which is less noticeable by hearing.
- The second advantage is described a little more specifically: greater dynamic range compared to PCM. Well, there is some slyness here, DD is more only in comparison with the classical CDDA format: 120 ... 140 dB versus 96 dB. If you compare, for example, with DVD Audio - DD is about the same.
- Third advantage: DSD is simpler technically. There is nothing to argue with here: simpler decoding of a signal, no need for synchronization and buffering of a stream when transmitting a signal from one device to another - a complete DSD victory. Incidentally, against the background of this advantage, it is strange to see exorbitant prices for equipment that supports DSD playback.
- Well, another advantage that DSD fans like to bring: music in this format is closest to the original analog sound. This is argued by the fact that modern analog-to-digital converters (ADC) operate on the principle of delta-sigma modulation, that is, these ADCs produce digital DSD stream. And here again, slyness: the recording will be completely original only in the case of a direct recording of a live performance or when digitizing a finished analog recording from a quality carrier. The operations of mixing, applying effects, mastering, even simply adjusting the volume — everything that a studio album cannot do without — cannot be done with a DSD digit because of the lack of normal processing algorithms. This means that all these operations are performed with the PCM format, and only after that the finished PCM record is converted to DSD. However, it should be noted that the PCM> DSD conversion and vice versa is quite accurate: only a little noise increases outside the real dynamic range (Fig. 7). This means that it does not matter in what format to listen to the recording: PCM Hi-Res or DSD - both formats are very similar in quality characteristics. Also, in fact, it makes no sense to buy a separate sound card for playing DSD, listening to the advice of a friend, a fan of this format.
Fig. 7. Dynamic range / noise conversion between DSD and PCM
findings
So, what to choose DSD or PCM? There is no unambiguous answer and cannot be: PCM 24 bit 92 kHz and DSD128, for example, are very similar in quality characteristics, and these characteristics are better than the equipment on which these formats will be played, which means a further increase in the quality of digital formats for playback on This stage is not appropriate. When evaluating the quality of sound in various high-definition formats, subjective sensations come to the fore, because the human brain is not the only person with quality: the equipment design, its cost, and, most importantly, the listener's mood and mood give a much greater effect on the sensations of listening to music. So, choose what you personally like and do not impose your opinion on others. Enjoy everyone!
Source: https://habr.com/ru/post/275613/
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