Digital audio

General term for technologies that convert audio signals into digital codes, process, transmit, and record them. Also called PCM audio (PCM = pulse code modulation).

Sound is a phenomenon in which the minute vibrations of air particles and the accompanying minute fluctuations in atmospheric pressure propagate through the air, and both particle vibrations and pressure fluctuations often change continuously with time. Even if such sound is received by a microphone and converted into an electrical signal, the waveform is similar to the waveform of the original sound, and the property that the phenomenon changes continuously with time does not change. This method of transmitting or recording an electrical signal is called analog audio in the sense that it handles an electrical signal similar to a sound signal and is used in many fields to this day. It has a good track record. For example, there are AM broadcasting (AM = broadcasting using the amplitude modulation method), FM broadcasting (FM = broadcasting using the frequency modulation method), disk recordings, tape recordings and the like.

On the other hand, it is digital audio in which a continuous electrical signal is converted into a sequence of discontinuous codes, processed, transmitted and recorded. The foundation of digital audio is PCM technology. PCM converts a continuous signal into a discontinuous code sequence through three processes: sampling, quantization, and encoding.

Sampling is an operation in which a signal waveform whose amplitude changes continuously with time is observed at regular intervals as shown in the figure, and the original signal waveform is replaced with a sequence of observed values. . Jumping observations are called sample values, and the number of observations per second is called the sample rate. Since the signal waveform is not faithfully tracked, but only discretely observed, information appears to be missing in such operation, but the sampling frequency is more than twice the upper limit of the frequency component contained in the sign. , it is theoretically proven that the sequence of sample values ​​can maintain the information contained in the signal without omission, and the original signal can be completely restored from now on (called the sampling theorem). In the case of a compact disc (CD), which is a typical digital audio system, the upper limit of the frequency of the signal to be recorded is set to 20 kHz and the sampling frequency is set to 44.1 kHz, which is more than double, with some concession. Even if you are nervous, observe 44,100 times per second in detail.

Quantization is an operation in which a fixed interval step is provided in the range from the minimum value to the maximum value of the signal amplitude, and the signal amplitude that changes continuously from moment to moment is replaced with the value in the closest step. . The picture is like replacing the slope with a ladder as shown in the figure. Since the smooth waveform becomes an irregular step, it is believed that signal deterioration will occur, but if the number of steps is large enough and the width of the step (the height of one step of the stairs) it becomes small enough. are as close to a slope as possible. The number of increments is expressed in bits (the number of digits when the number is expressed in binary) and is called the number of quantization bits. For a CD, the number of quantization bits is 16 bits, which is expressed in decimal as 65,536. Even if the slope is replaced by stairs, the number of steps is more than 60,000 and you can see that it is as close to the slope as possible.

Two operations, sampling and quantizing, result in a continuous signal waveform that is transformed into a discrete, discontinuous (discrete) sequence of signals, both temporal and oscillating. It should be noted that sampling or quantification can be done first, and actually both are often done at the same time.

Encoding is the operation of converting a quantized sample value to a numeric code based on a predetermined promise. From the binary code that corresponds to the voltage (or current) “yes” and “no” to the symbols “1” and “0”, the appropriate code conversion is carried out for each medium such as recording and broadcasting. The CD, which was put into practical use in 1982 (Showa 57), was encoded without any compression operation to reduce the data size, but the MiniDisc (MD)

Digital audio that really sounds better

If you only listen to digital music in 128 kbit / s (bps) AAC or mp3 format on a portable audio player such as an iPod, you will be moving away from real music.

You can understand the sound that is played by connecting an iPod to an audio device by comparing the sound of the original CD with a normal audio device. And there is digital audio with even better sound quality than CDs. It is a big mistake to think that mp3s are the sound quality of digital audio and that digital audio sounds better.

When a record company digitally records a musical performance, it is common to adopt the PCM (Pulse Coded Modulation) method or the DSD (Direct Stream Digital) method. Those recorded with PCM will eventually be sold as CD and DVD-Audio. What DSD records is sold as SACD (Super Audio CD).

In PCM, the sample rate (rate) and the number of sample bits are determined and recorded. The term sampling frequency is independent of the frequency of the sound and is the number of times a sample is taken per unit of time. The unit is “Hz”, and the higher this value, the more faithful to the original sound, but the larger the data size. For example, 44.1 kHz is selected for CD and 96 kHz or 192 kHz for DVD-Audio. DVD-Audio uses DVD to store large amounts of data. A 44.1 kHz CD can express signals (sound) up to a frequency of approximately 20 kHz. The sound that humans can hear is said to be up to 20 kHz, but the sound of a real musical instrument includes much louder sounds, and when played up to such a high range, the sound quality clearly improves. It has been confirmed that it will be done. With a sampling frequency of 96 kHz, signals up to 48 kHz can be expressed and the tones of musical instruments and voices can be reproduced more faithfully.

On the other hand, the number of sample bits is the number of quantization bits, which determines how many bits the sampled data is represented. The more bits there are, the more faithful the original sound will be, but the larger the data size. If the number of bits is small, the reproducibility of a small signal (sound) deteriorates. The number of bits on an audio CD is 16 bits. If it is 24-bit, it can be expressed with 256 times the precision of 16-bit.

The high-quality PCM digital audio format is 24-bit uncompressed, 96 kHz (2496 for short), or 192 kHz. However, only DVD-Audio or songs sold by Onkyo Marketing on the “e-onkyo music store” website can be purchased in this format (Fig. 1). To play DVD-Audio, you need a multi-format player such as the Pioneer DV-696AV. On your computer, a sound card such as Sound Blaster Audigy 2 is supported. DVD-Audio playback devices are listed on the DVD-Audio website.

Figure 1 ● Website of the e-onkyo music store that sells high-quality 24-bit 96 kHz music files
[Click the pic to enlarge]
DSD, on the other hand, is a completely different method from PCM and uses a delta-sigma modulation method. To put it very roughly, a bit transitions between Low and High states at a very high speed to express the pitch of the sound. It is a method that expresses the magnitude of the audio signal by the density (shading) of a 1-bit digital pulse. When the level of the original sound is high (the sound is high), the high state becomes lower, and when the level of the original sound is 0, it is Low and the number of High will be the same. The sampling frequency of SACD is 2.8224MHz (CD = 64 times 44.1kHz). Sound quality far exceeds that of a CD with a frequency response of 20 kHz and a dynamic range of 94 dB, with a frequency response of more than 100 kHz and a dynamic range of 120 dB or more (the dynamic range it’s wide to loud and quiet sounds). So the bigger the better). A high-speed CPU is required to handle DSD in a personal computer, and the Sony VAIO Type R, Type H, and Type V announced in September 2005 were the first to be equipped with DSD record / playback functions. SACD playback is also not possible with these models.

Requires dedicated hardware
To play the 2496, etc. With good sound quality, the personal computer’s audio hardware needs to be upgraded to 2496-compatible high-quality sound. The most popular audio hardware for personal computers is the IEEE1394 interface. It is convenient because it can be connected with a single cable.

I connected audio hardware for a personal computer and compared 2496 and 1644.1 (normal CD sound quality). The audio hardware tested is as follows (Fig. 2).

・ RME Fireface 800 (about 210,000 yen)
・ RME Hammerfall DSP Multiface (PCMCIA)
・ TerraTec P

Why does the sound change even though it is digital?

If you’re interested in PC audio, once you think about it, “Why does the sound change even though it’s digital?

It is unlikely that the data cannot be corrected and returned to the original data with current digital audio equipment, so the digital data itself is safe to think that they are the same. Then why?
The reason is that even though it is digital, the signal that drives the speaker is an “analog signal”, so it affects the “analog signal” after the digital is converted to analog.
Suppose you have a noisy PC. As shown in the figure below, noise flows through the GND of the USB cable to the USB-DAC. The noise flowing into the USB-DAC affects the signal after analog conversion and the sound quality deteriorates.

On the other hand, a PC with less noise has less effect, so the sound quality is less likely to deteriorate.

In the case of “wireless” at the end, “wireless” is the “ultimate USB cable”. It is not affected at all by PC noise. It is an ideal transmission method. The only embodiment of such a method is “Air High Resolution Link Technology”.

What is PC audio?

For super beginners

Does anyone say “I really don’t understand” or “I can’t hear you anymore” over the audio? This is a super introductory course on audio that started in response to so many unexpected voices, and this time I will explain very briefly about the “PC audio” that I often hear recently.

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● First of all, what is “PC audio”?

This time, let’s introduce PC audio.

Until a while ago, the only way to listen to music was with a CD, MD or radio player.

However, before I knew it, listening styles with digital music players, such as personal computers and iPods, became common.

And the idea of ​​PC audio is “I want to listen to music with good sound when I listen to music using a personal computer.”

● The point is “DAC” which performs the digital to analog conversion.

For example, when you listen to music on your computer, you usually listen to it like this.

Headphones for PC.png

Connect headphones, etc. to the audio output terminal (LINE OUT) of your computer.

Or you can play it through your computer speakers.

So how do you make this sound good?

When playing music with digital data, the DAC (duck) circuit is important in terms of sound quality.

Since MP3 or WAV is digital data, but scanning for speakers or headphones to play, you need to convert the digital data to analog signal.

DAC (short for D / A converter) performs the “digital-to-analog conversion”.

The quality of the converted analog signal depends on the accuracy of the DAC.

In other words, if the DAC is not good, it will not sound good at all.

Originally the digital data is a list of numbers like “0101001 …”, but the figure is as follows. (* Image)

DA .png conversion

If you are using your computer’s audio output terminal, you are using the DAC circuit inside your computer as shown in the figure below.

However, since the personal computer was not originally made for music playback,

The performance of the DAC installed in the personal computer is often not that high due to the balance of costs with other parts.

Also, many electronic parts run at high speed inside the personal computer, so a lot of noise is generated.

In such an environment, you cannot expect “good sound” in the audio sense.

PC Headset (with DAC) .png
Therefore, instead of using the DAC inside the personal computer, I asked them to send the digital data as it is from the USB terminal.

It is converted to analog with a music DAC and converted to an audio quality music signal.

That’s the basic idea of ​​PC audio.

Is it natural that “the sound of the computer is bad”?

Compare the performance of the audio output numerically

If you usually use an audio interface or USB DAC, many people think that the sound function installed on the PC is obsolete. The author himself thought so, but “actually, the sound quality may have improved with recent PCs.” Suddenly, I thought about that, and this time I checked the sound quality of the computer I had.

As long as you listen to a little sound with headphones, you won’t get the bouncing buzz like in ancient times. Although it is clearly different from listening to solid equipment, I don’t think there is any particular discomfort in listening to streaming audio over the Internet with a PC alone without listening to it. So, I used RMAA PRO, which I always use to check the audio interface, to compare the difference numerically. Then there was a considerable difference between the models I tested, so I summarized the results.

Compare barebones with notes and Intel NUC
I believe that an audio interface and USB DAC are a must to produce decent sound on a PC, but even professional musicians often connect to the PA from the PC’s headphone output without using the audio interface. Sometimes I see people. Well, I was wondering if I shouldn’t be particular about the sound, but I was wondering, “Is it okay?”

However, there is a chance that the onboard sound function is now decent. It is not a good idea to decide that it is not good without verifying it. So I decided to test it with three relatively new PCs that I had.

One is a basic “Shuttle SH370R6” PC that was purchased late last year and assembled early in the new year. I wanted to incorporate Intel’s Core i9, but since it only supported the 8th Gen Core processor, it was a machine with 6 cores, 12-thread Core i7-8700 installed, and 32GB of memory.

Barebone PC “Shuttle SH370R6”
After that, it seems that the 9th Gen Core processor also supports BIOS update, and I am sorry I bought it a bit later, but I will test it on such a machine.

Uses Core i7-8700 with 6 cores and 12 threads
The second is a small notebook PC that I bought last fall and the NEC “LAVIE Note Mobile NW150” that I usually carry in my bag every day. A lightweight machine weighing 904g with a battery that runs for 13 hours on an 11.6-inch screen. The CPU is a 2-core Pentium Dual-Core 4410Y that runs at 1.5 GHz, so it is not a fast machine, but it is a portable PC that is enough to surf the net with Chrome and write manuscripts with an editor. of text.

NEC 「LAVIE Note Mobile NW150」
The third is an Intel NUC kit that I bought two years ago and it is a small machine called “7i7BNH”. It is a PC that I bought because it is equipped with a 7th generation Core i7-7567U and a USB Type-C type Thunderbolt 3 terminal.

Intel NUC 「7i7BNH」
Typically, if it is an audio interface, the input and output are connected directly with a cable to make an audio loop, and then the measurement is done using the RMAA PRO audio testing software. I’ve always tried 44.1 kHz, 48 kHz, 96 kHz, and 192 kHz, so I thought I’d use that method again … but if you look closely, it has input and output terminals. Only the first SH370R6. Both the LAVIE Note Mobile and the NUC have only one headphone-out jack.

Looking around now, notebook PCs are less likely to have LINE IN and mic input, and more common to have only output. Also, the purpose of the experiment here is not to check the total input / output function, but to see what the output performance of the headphones is like. In a normal audio loop, if there is a problem with the input = record function, that becomes a bottleneck, and even if the headphone output is high-performance, it will be poor. So I decided to use the same audio interface for all inputs and compare the output performance of each headphone.

This time, I used Roland’s Rubix 24 as an audio interface. It is a 2IN / 4OUT audio interface that works with USB bus power and is a device that can record and play back at a maximum of 192 kHz / 24 bits. First of all, the result of testing with RMAA PRO in the form of an audio loop to see the performance of this Rubix 24 itself is as follows. I have tested the 2IN / 2OUT Rubix 22 before, and the results show that it has roughly the same performance.

Incredible sound quality digital reverb!

Amazing sound quality digital reverb thanks to high-speed computation using high-precision 32-bit DSP!

AMBI SPACE reproduces the complex reverb mechanism created by various elements with the original FREE THE TONE algorithm, making full use of high-speed arithmetic processing using a 32-bit dual-core CPU chip and high-precision 32-bit DSP. bits. The high-quality, yet extremely natural, musical reverb produced by this small enclosure can be used in a variety of situations, not only for electric guitars and basses, but also for acoustic guitars and vocals, and even for recording mixes. Additionally, FREE THE TONE’s original new reverb sounds “CAVE” and “SERENE” emphasize the frequency components by reproducing the early reflection and late reverb that make up the reverb. By adding the complex harmonic structure produced by the above to the multi-stage reverb sound, we create a reverb sound that is fantastic, transparent and expansive like never before. It is a reverb effector that was born after many years of development combining the analog and digital technologies that Free The Tone has cultivated.

“Characteristics”
・ By using a dual-core chip with a 32-bit main CPU and a 32-bit coprocessor and performing high-speed calculations with a high-precision 32-bit DSP, sound comparable to that of a rack-type effector is achieved by being the size of a compact effector.
-Up to a total of 4 presets for all knobs can be stored as presets.
-Four presets can be recalled using MIDI program change numbers.
-Equipped with 6 reverb MODES.
-Free The Tone’s original reverb sounds “CAVE” and “SERENE” are illusions that have never been seen before by adding sounds with various overtone structures created by a complex reverb pattern design to the multi-stage reverb sound. Produces a reverb sound that is both transparent and spacious.
-The original sound (dry sound) passes from the input to the output as an analog signal, and the original sound and the reverb sound are mixed by the internal analog mixer. This will come out without compromising the sound quality of the original sound.
-Built-in HTS circuit that comprehensively manages input to output signals, which is an important feature of FREE THE TONE products, keeps the sound texture when the effect is on and off the same.

《Main Specifications》
● Number of presets: 4
● Input impedance: INST 1 MΩ or more / LINE 300
kΩ or more ● Output load impedance: 10 kΩ or more
● Maximum input level: INST + 3 dBm / LINE + 11 dBm
● Control: PRE DELAY, DECAY, TONE, MIX, MODE, INST (-10dB) / LINE (+ 4dB) level change switch, KILL DRY switch
● Terminal: standard 1/4-inch x 4 L (mono) / R INPUT phone jack, L (mono) / R OUT, DIN 5PIN (MIDI IN) jack, DC9V input jack (for connecting AC adapter )
● Power supply: DC9V dedicated AC adapter (FA-0905D-JA)
● Current consumption: 280mA (maximum value)
● Size: 120 (W) x 102.3 (D) x 74 (H) mm (including projections such as the foot switch and connector)
● Weight: Approximately 385 g (without
accessories) ● Accessories: Warranty card, instruction manual, dedicated AC adapter (FA-0905D-JA), rubber feet x 4

High Resolution Audio Source

It can be said that “USB-DAC” is a secret weapon for playing music files on a computer with high sound quality.

Just add “USB-DAC” to the audio you use all the time, and you can enjoy much higher sound quality! Therefore, this time, Sara-chan visited Onkyo Co., Ltd., which developed the state-of-the-art “USB-DAC” that supports high-quality sound sources called “high-resolution”, which has been launched more and more in the last years. ! We also visited the audition room and asked Mr. Kurosawa, director of the high-quality music distribution site “e-onkyo music”, to teach us how to enjoy high-quality sound!
“Sample rate” and “bit rate”

Sara-chan: Hello! Wow, it’s a nice listening room! I’m excited!

Kurosawa: Hi Sarah! Today, I am trying to get you to experience high-end sound quality in various ways.

Sara-chan: Thank you! “USB-DAC” is certainly important to enjoy high quality music on your PC! But there are many “USB-DACs” and I don’t know what to choose.

Mr. Kurosawa: When choosing “USB-DAC”, it is a good idea to check the “sample rate” and the “bit rate”.

Sara-chan: Sa, Samp … Call frequency ?? What the heck is that ~ ??

Mr. Kurosawa: “USB-DAC” is a device that converts sound from digital signals to analog signals, isn’t it? The “sample rate” indicates the number of digital samples of the audio signal acquired per second during the conversion. The “sample rate” determines the frequency range of the audio file. The higher this number, the closer the digital waveform will be to the original analog waveform and the softer the sound will be. On the other hand, “bit rate” indicates the amount of information per second. They are expressed in units of “Hz” and “bit” respectively. By the way, do you know what the CD standard is?

Sara-chan: Well I’m sure it’s “44.1 kHz / 16 bit”!

Mr. Kurosawa: That’s right! It is said that the sound in the ultra high range above 20 kHz cannot be heard by the human ear, so the CD cuts out the inaudible sound. But even if you think you can’t hear it, you actually feel the vibrations in the air and it affects the sound at the frequencies you hear.
Since the high resolution sound source also records that part, it can be said that it is closer to a more realistic sound. First, at the music creation stage, work is often done at 96 kHz / 24-bit, which is why high-resolution sound sources are sometimes referred to as “studio master quality.” Recently, even more informational sound sources such as “192kHz / 24bit” have appeared. There are “44.1 kHz / 16 bit” and “96 kHz / 24 bit” sound sources here, so let’s compare them using ONKYO’s “DAC-1000 (S)”!

Sara-chan: Wow! You can feel the difference more than you imagined! It feels like a live performance is taking place right in front of you! I feel that the sound is expansive and I feel that I am surrounded by the sound! Anyway, it seems like I’ve never heard it on audio before! Impressed!

Kurosawa: Fufufu. You can feel the difference! However, even if you have a high resolution sound source such as “96 kHz / 24 bit”, there is no point in using a “USB-DAC” that does not support “96 kHz / 24 bit”. Therefore, when choosing “USB-DAC”, it is important to check the “sample rate” and “bit rate” to see if it is compatible with the sound quality you want to hear. By the way, recently there is even a “USB-DAC” that has a function to change the frequency called “upsampling”.

Sara-chan: Wow! We are in an era where high quality sound is increasingly required!

What is asynchronous forwarding?

Sara-chan: What other characteristics should I check?

Mr. Kurosawa: The data transfer method is important! When transferring a high quality sound source from a personal computer to a “USB-DAC”, some have a function called “asynchronous transfer” to avoid data corruption.