Tag: are digital audio players worth it

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Sampling rate

In the separate article “Bit rate and bus width” and “Notes on reducing the bit rate”, we presented the history of the “bit rate”, which is the vertical axis of graph paper during PCM sampling.

This section considers the “sample rate” of the horizontal axis.

Maximum frequency that can be sampled (Nyquist frequency)

With the PCM method, exactly half the sample rate is the maximum recordable rate. This value is also called the “Nyquist frequency”.
For CDs recorded at a sampling frequency of 44.1 kHz, half, 22.05 kHz, is the theoretical maximum recordable frequency.

Now half the sample rate is the maximum recordable rate, and the reason is … Seeing is believing, see the figure to the right.
To represent a cycle of a sine wave, you must draw up and down round trips using at least two samples.

Alias ​​noise: frequencies higher than recordable

But what if a signal with a frequency higher than the Nyquist frequency enters the A / D converter?

See the graphic to the right.

While the input signal performs “3 round trips with 4 samples”, the resolution of the horizontal axis (measurement frequency) of the A / D converter is too low to correctly follow the movement of the signal. The sampled values ​​draw a “one round trip with four samples” waveform, which is significantly lower than the input frequency. Once again, the original signal is “manufactured” in the digital domain.

In this way, an input signal with a frequency higher than the Nyquist frequency is the source of a signal with a lower frequency than it should be, called “alias noise.”

Alias ​​noise is also called “envelope noise” because it occurs at a frequency lower than the Nyquist frequency by the amount that the original signal exceeds the Nyquist frequency, as shown in the graph to the right. The alias noise that wraps around and reaches 0Hz continues to be wrapped between 0Hz and the Nyquist frequency so that it is reflected back.

Actually, the A / D converter contains a low pass filter to prevent such alias noise, so alias noise is not a problem when recording analog signals.

However, handling alias noise can be important when using plugins. The next section explains this point.

Do you need a high sample rate?
When starting to record or compose with a DAW, how to set the sample rate of the project is a difficult place. Unlike the bitrate above, once the sample rate is set, it is difficult to change it later.
It is generally said that the higher the sample rate, the higher the quality of the master that can be produced without damaging the original sound, but the PC specs required during work will also improve dramatically.
But is there any merit in editing at a high sample rate, like when the final medium is a 44.1 kHz CD? Also, can you ignore the difference in sample rate based on the trend of the sound you are looking for, such as the genre you are dealing with?

The bottom line is that when you use harmonic effects like saturators and compressors in your DAW, the sample rate setting can have a significant effect.

Next, we’ll use an ultrasound to consider how the project’s sample rate will change.

The first is a single scan signal sonogram. You can see that the frequency of the sine wave gradually increases from 0 to 22 kHz.

Let’s add a distortion effect to this.
In each case, the signal with the effect applied is exported in projects with different sample rates, and then the file is converted to 44.1 kHz.
All effects use the same preset.
Note that the files themselves that are comparing sonograms are all 44.1 kHz.

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Sample, sample rate, sample format, and bit rate

The main function of the audio interface is to convert the signal between analog and digital formats.

As mentioned above, real sounds have endless possibilities for pitch, volume, and duration. Computers cannot process infinite information, so audio signals must be converted before they can be used.
Computer approximated waveform
Source: pcm.svg http://commons.wikimedia.org/wiki/File:Pcm.svg Retrieved from.
Figure 1.1 Computer approximated waveform

The figure in Figure 1.1, “Computer approximated waveforms” illustrates the situation. The red waveform shows the sound waves produced by the singer or acoustic instrument. The computer cannot handle the gradual change of the red waves. The computer should use the approximation shown in the gray shaded area of ​​the figure. This figure is an exaggerated example and does not represent the actual recording.
Converting between analog and digital signals reveals the difference between low-quality and high-quality audio interfaces. The sample rate and sample format control the amount of audio information stored by the computer. The more information that is stored, the better the audio interface can approximate the original microphone signal. The possible sample rates and sample formats determine only a part of the quality of the sound produced or obtained by the audio interface. For example, an audio interface built into the motherboard can support 24-bit sample formats and 192 kHz sample rates. On the other hand, a professional-grade FireWire-connected audio interface with a 16-bit sample format and 44.1 kHz sample rate may sound better.
1.3.1. Shows
A sample is a unit of audio data. The computer stores video data as a series of still images (each called a “frame”) and displays them one at a time at a specified speed (called a “frame rate”). The computer stores audio data as a series of still sound images (each called a “sample”) and plays them back one at a time at a specified frequency (called a “sample rate”).
There are not many types of frame formats and frame rates that are used to store video data. There are many types of sample formats and sample rates that are used to store audio data.

Frequency used for audio (sample rate, PCM, DSD, etc.)

On this occasion, I would like to explain the frequencies used in digital audio and their meanings.

Recently, the high-resolution sound source has increased, such as 192KHz Toka, 11.2MHz, as the frequency has been written or will, what frequency?

I would like to explain the frequency used for said audio taking as an example the Combo384 installed in the USB-DAC used in LV2.0.

1. 1. What is the sampling frequency?

Music distribution is becoming mainstream these days, but audio was first digitized on CDs, which are still on the market.

You often hear that the sample rate of a CD is 44.1KHz. Since digital signals are basically 0 or 1, to reproduce up to the 20 KHz limit that can be heard by the human ear, a resolution of twice that frequency is required. Furthermore, the frequency was decided to be 44.1 KHz taking into account the digital signal processing margin. Since a musical signal is a set of sine waves, it is 44.1 KHz that can fluctuate at a maximum frequency of 20 KHz.

2. What are 16 bits and 24 bits?

As you may hear often, CDs are sometimes described as 44.1KHz / 16bit. This 16 bit is the volume of the sound. Since 16 bits can express the size of 2 raised to 16, there are 65536 different sizes.

Converting this to dB is 20LOG (65536), which is approximately 96 dB. The dynamic range of a CD (the difference between low and loud sounds) is 96 dB.

For DVD and Hi-Res, it can be 24-bit, but in this case, it’s 16.77 million steps of 144 dB.

3. 3. PCM format

So what is the actual signal? In the case of the PCM format, the standard called I2S, which can support up to 32 bits in sample rate, is common. In the case of a CD, being stereo, the data has a frequency of 44.1 KHz with 2 channels (L, R) alternately 32 bits (although in reality 16 bits are used).

Therefore, to process this digitally, a processing capacity of 44.1KHz x 2CH x 32bit = 2.8224MHz is required.

Bitrate

In communication, it indicates how many bits of data are transferred per hour and is generally expressed in bps (bit / s) of how many bits are transferred (processed) per second.

If it is low, the size when saving as a file is small and there is space on the transmission line for communication. For example, when an audio (1 channel) is compressed to 1/3, the 3 channel audio can be sent at the same bit rate. Excuse the old story, but considering the age of analog communication (analog cell phones), digitizing + compression will support multiple calls with the same radio waves.

Finally
I often hear what is called Hi-Res Audio. The sampling frequency is said to be 96 kHz or 192 kHz, which is above 48 kHz, the number of quantization bits is 24 bits, and the limit (high range) of human hearing is about 20 kHz, but it expresses frequencies higher than that. It will be. It is the same bit rate as the image from a long time ago. .. ..
By the way, it seems that dogs can hear up to 60 kHz and cats up to about 64 kHz.

Hi-Res Example (Hi-Res Audio)
Sampling frequency Number of quantization bits Number of channels bit rate Frequency that can be expressed
192 kHz twenty-four 2 9.216 kbps 96 kHz
192 kHz 16 16 2 6,144 kbps 96 kHz
96 kHz twenty-four 2 4.608 kbps 48 kHz
96 kHz 16 16 2 3,072 kbps 48 kHz
48 kHz twenty-four 2 2,304 kbps 24 kHz
Considering the limit of human hearing (around 20 kHz), according to the sampling theorem, 48 kHz or 44.1 kHz is a sufficient frequency, but what about all of them? .. ..
In my case, I cannot distinguish the high resolution range, but it should be able to reproduce the discarded frequency at 48 kHz to 96 kHz, and when the number of quantization bits is in the 24-bit range, the sound pressure (dB) is a bit. Feels like I’m going up (?) (It’s just my ears).
I’d like to make a comparison if I get a chance, but I feel like I can’t say by ear that I don’t have a proper regenerator (like an expensive analog amp).

Is it finally time for cats and dogs to get verified in the acoustics industry? .. ..

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Sample rate and bit rate

I wrote over audio files last time, but if you reduce the file size (code at a lower bitrate), the sound quality tends to deteriorate. How much should it really be? .. ..

When compressing using audio encoding (AAC, MP3, etc.), the compression rate is determined by the bit rate at the time of encoding. Specifically, if you set a low bitrate, the compression rate will be higher and the file size when saved will be smaller, but first, what is the bitrate for the uncompressed original sound source (PCM) ?
If you save it as PCM, the sound quality will be the original sound, but it may be a little inconvenience to save it without worrying about the file size. Also, depending on the application, I think the memory capacity is sufficient even for the size of the original sound, and the communication speed is good. Therefore, I would like to write about the sample rate and bit rate that are often heard in digital audio.

The bit rate of digital audio is determined by the sampling frequency, the number of bits assigned to a sample (number of quantization bits), and the number of channels (stereo, monaural, etc.).

PCM bit rate (uncompressed) = sample rate x number of quantization bits x number of channels
As I wrote a bit last time, for example in file containers like wav and mp4 format, this information is attached as a header, so that the application can see the header and play it. The compression rate of the encoding is determined by the bit rate specified at the time of encoding for this PCM (uncompressed) bit rate.
For example, as many of you know about music CDs, with 44.1 kHz stereo, this is the next bit rate.

Music CD bit rate: 44100Hz x 16bit x 2ch (stereo) = 1411.2kbps
When encoding this with MP3, AAC, etc., it is natural to specify a bitrate lower than 1411.2 kbps. For example, when encoding at 256 kbps, the compression rate is approximately 18% when the original sound is 100% and the file size is 1/5 or less.

Encode Music CDs at 256 kbps: 256 kbps / 1,411.2 kbps = approximately 18%
Generally, the sample rates of audio devices actually connected to PCs are 48 kHz and 44.1 kHz for music, 16 kHz and 8 kHz for audio such as microphones and headphones, and 32 kHz, 24 kHz, 22.05 kHz. , etc.

The bit rate of PCM (uncompressed sound source) with 16-bit quantization bits is as follows.

Stereo (for music) PCM 16-bit bit rate (example)
Sampling frequency Number of quantization bits Number of channels Bit rate Comments
48 kHz 16 16 2 1,536 kbps
44.1 kHz 16 16 2 1,411.2 kbps Music CD
32 kHz 16 16 2 1,024 kbps
24 kHz 16 16 2 768 kbps
22.05 kHz 16 16 2 705.6 kbps
16-bit monaural PCM bit rate (for audio) (example)
Sampling frequency Number of quantization bits Number of channels Bit rate Comments
32 kHz 16 16 1 512 kbps Super Wide Band
24 kHz 16 16 1 384 kbps
16 kHz 16 16 1 256 kbps Broadband
8 kHz 16 16 1 128 kbps Narrowband

Sampling rate
If you check the web, there are explanations like the sampling required to convert analog waveforms to digital conversion. For example, it shows how many samples of an audio signal input from a microphone are taken per second and digitized. The larger the sample, the greater the range that can be recorded. When an analog waveform is digitized, the frequency that can be expressed is half the sampling frequency (sampling theorem). For example, with a sampling frequency of 48 kHz, it is possible to express up to 24 kHz. At 8 kHz (narrow band) and 16 kHz (wide band), which are often used for audio, you can only hear up to 4 kHz and 8 kHz, respectively. The higher the sample rate, the higher the bit rate.

Sampling theorem
It is a very simple explanation, but it can express up to half the sample rate. When sampling a signal, if the interval is small, it can be restored close to the original signal, but if it is too thick, it cannot be restored (I would like to write a little more detail when talking about signal processing or other time).

44.1 kHz
Why is the rate of 44.1 adopted, which is not well separated? .. ..
Isn’t the technician deliberately wearing a pesky watch to prevent music CDs from being easily copied? I heard something like that. When I searched, it seems this happened (?) Due to the convenience of an old PCM recorder. In this age, it is difficult to know what 44.1 kHz is in development. 44.1 kHz sampling conversion

Sampling theorem

It is a very simple explanation, but it can express up to half the sample rate.

When sampling a signal, if the interval is small, it can be restored close to the original signal, but if it is too thick, it cannot be restored (I would like to write a little more detail when talking about signal processing or other time).

44.1 kHz
Why is the rate of 44.1 adopted, which is not well separated? .. ..
Isn’t the technician deliberately wearing a pesky watch to prevent music CDs from being easily copied? I heard something like that. When I searched, it seems this happened (?) Due to the convenience of an old PCM recorder. In this age, it is difficult to know what 44.1 kHz is in development. The 44.1 kHz ↔ 48 kHz sampling conversion is a headache. For example, USB audio (USB audio device class) exchanges data at 1 ms intervals. In the case of 48kHz, the data is 48 samples, but when considering 44.1kHz, it will be 44 samples (x9) and 45 samples (x1) in 10ms. If you cheat on a sample when there are 45 samples (tentatively), it will be 44.0kHz. I think it’s more like that with voice and music, and I think the human ear is mostly misleading (it’s just my personal opinion).
However, with an objective evaluation method, it will soon come to light. For example, you can see that you were clearly fooled by a sine wave (sine wave) (maybe you are unexpectedly on the market).

Number of quantization bits
The sampling had to take a value in the direction of time (discretization), but the quantization had to take a value in the direction of amplitude. The range that it is possible to display the volume of the sound, which is often heard, “96 dB dynamic range” means that the number of quantization bits is 16 bits, and the music signal is played in the range of 0 to 65535. I can do it. The number of quantization bits is also called the bit depth or bit depth.

Bitrate
In communication, it indicates how many bits of data are transferred per hour and is generally expressed in bps (bit / s) of how many bits are transferred (processed) per second. If it is low, the size when saving as a file is small and there is space on the transmission line for communication. For example, when an audio (1 channel) is compressed to 1/3, the 3 channel audio can be sent at the same bit rate. Excuse the old story, but considering the age of analog communication (analog cell phones), digitizing + compression will support multiple calls with the same radio waves.

Finally
I often hear what is called Hi-Res Audio. The sampling frequency is said to be 96 kHz or 192 kHz, which is above 48 kHz, the number of quantization bits is 24 bits, and the limit (high range) of human hearing is about 20 kHz, but it expresses frequencies higher than that. It will be. It is the same bit rate as the image from a long time ago. .. ..
By the way, it seems that dogs can hear up to 60 kHz and cats up to about 64 kHz.

Frequency used for audio (sample rate, PCM, DSD, etc.) Part 2

LRCLK

I explained that the LRCLK that came out earlier is a signal that distinguishes between L and R. In the I2S format regulations, the signal level is specified as L channel for 0V (low) and R channel for high.

Let’s take a look at the LRCLK waveform that actually corresponds to the sample rate.

image049

LRCLK 44.1KHz CD Playback

image051

LRCLK 96KHz LPCM 96 kHz playback

image053

LRCLK 192 KHz LPCM 192 KHz playback

image055

LRCLK 384 kHz LPCM 384kHz playback

Therefore, the sample rate matches LRCLK. COMBO384 supports LPCM maximum sample rate up to 384KHz.

7. BCLK

It is sometimes referred to simply as CLK, commonly referred to as a bit clock, and is the reference clock for reading 32 bits. So 32 times the sample rate and 64 times the LR channel frequency combined.

It will be the second frequency in the frequency calculation above. In the case of 44.1 KHz, it will be 2.8224 MHz.

Let’s take a look at the actual BCLK.

image035

BCLK 2.8224MHz CD 44.1 kHz

image037

BCLK 6.144MHz LPCM 96 kHz

image039

BCLK 12.288MHz LPCM 192 KHz

image041

BCLK 24.5760MHz LPCM 384kHz

This BCLK is also issued during DSD.

image045

DSD64 2.8224 MHz

image043
DSD128 5.6448 MHz

image047

DSD256 11.2896 MHz

It is often mentioned that it supports 11.2MHz in the introduction of DSD, but to be exact, it is this frequency.

The waveform is not square because the oscilloscope has a low frequency bandwidth. It is actually a square wave.

How was it?

Did you understand the frequencies used for the audio?

I think you can see audio frequencies in various situations, but if you remember that this frequency meant something like this, I think you may feel more familiar with digital audio.

In Marz, this DSD256 (11.2896MHz above) LPCM is 384KHz sells an amplifier equipped with a COMBO384 corresponding to.

Since it supports high sample rates, it can play high-resolution music.

Amplifier kit equipped with USB-DAC

・ Integrated Stereo Amplifier LV-2.0 PREMIUM Kit [LV2-KIT-PREMIUM]

USB-DAC module (with COMBO384)

・ LV-2.0 USB DAC Board Module Finished Product [LV2-USBDACM]

What are digital and analog connections?

I will explain the difference between them

Hi, I’m Yamakawa from FunLogy. Previous article [What are passive speakers and active speakers? Differences and benefits] Have you enjoyed it?

When you buy a speaker, you will notice that there is a digital connection and an analog connection to connect speakers. What is the difference between the two connections?

In addition, there are several types of terminals that connect to audio, such as speakers. In this article I will explain the role of these terminals and the points to take into account when connecting them.

Table of Contents

Types of connection terminals

1. Audio output terminal

2. Data transfer terminal

3. Network connection terminal

Types of connection terminals
Diagram of three types of connections.

There are three types of terminals that are used for the speakers.

1. Audio output terminal
2. Data transfer terminal
3. Network connection terminal

If you used to use audio, you only need to connect it to analog, so if you only understand the audio output terminal, there was no problem.

However, recent audio is connected to televisions, PCs, networks, etc., so the types of terminals are increasing and the audio connections become more complicated. I will explain each function to deepen your understanding of those terminals.

1. Audio output terminal

There are two types of audio output terminals: analog connection terminals and digital connection terminals. I will explain the characteristics and types of each terminal.

We will also introduce the advantages and disadvantages of analog connection terminals and digital connection terminals, so try the connection method that suits you best.

1-1. Characteristics and types of analog connection terminals
handset

The analog terminal is a terminal that is also used for commonly viewed headphones.
Since it is widely distributed, it has the advantage of being inexpensive and easy to obtain. Recently, audio products using many types of analog terminals are being sold in 100 yen stores and massive home electronics retailers.

On the other hand, the disadvantage of the analog terminal connection is that it is vulnerable to noise. The cause of the noise is the heat and vibration of the connected device, but when connecting to a personal computer that is prone to such noise, if you connect the analog terminal, you will be concerned about “sir” noise. end up.

Therefore, when connecting an analog terminal to a personal computer, it is desirable to take countermeasures against noise, such as using a device that once converts the analog connection to a digital connection.

When connecting audio to analog, there are two types of terminals: RCA terminal and telephone terminal. I will explain about the two terminals.

RCA terminal
Red and white wire

RCA terminals are also called pin terminals or pin connectors. Generally, it has two colored terminals, red and white. This color makes sense and each channel is assigned.
The speakers have L and R, but R is set to have the same acronym as Red. Therefore, the white becomes L. Be careful not to connect backwards.

Phone jack

The telephone jack is a terminal often used for headphones and earphones. Unlike the RCA terminal, this terminal is generally a stereo phone. So you don’t have to worry about LR connections, but keep in mind that phone jacks come in different sizes.

There are standard 6.3mm phone terminals, 3.5mm mini plug terminals, and 2.5mm micro mini terminals. It is not possible to connect to terminals of different sizes, so be careful of the size of the terminals when making connections.

1-2. Characteristics and types of digital connection terminals
The digital terminal is used to receive the digital output from the player or the digital signal from the TV. The merit of the digital connection terminal is that the optical signal is transmitted and received by an optical cable, so there is less noise and deterioration when transmitting the signal, and the sound quality is superior to that of the analog connection.

On the other hand, the number of speakers that support digital connection is small, and there is the disadvantage that it tends to be more expensive than speakers that only have analog connection.
It is a drawback that even in massive home electronics retailers, there are only a few products on the shelves, or it can be difficult to compare products because you cannot see the real products unless you go to a

How different is the sound quality of Dolby TrueHD and Dolby Digital?

This changes quite a bit. A level that even people who are not interested in sound can understand.

In a way, TrueHD sounds more like a movie theater!
The comments are kind and amateurish, but the frank impression is that they really make a good sound, and I think this is very important.

In surround sound, the connection between the speakers becomes fluid.

In other words, the existence of surround sound speakers disappears.

For example, in the opening scene of Mad Max, several lines are whispered in the brain like hallucinations.

When listening to this with Dolby Digital Plus, I sometimes feel like the sound is coming from the surround right speaker in the whisper on the back right.

When this happens, the environment of the film returns to reality and you wake up.

On the other hand, in the case of Dolby TrueHD, the configuration of the 5.1ch system and the AV amplifier is exactly the same, but you can get solid surround sound.

This difference is quite large.

You can feel comfortable in the world around you.

Also, if you use other expressions, the density of the sound is different. There is a heavy sound with a core.

Gyun in the scene where Mad Max’s engine turns! It is TrueHD that the sound echoes to the internal organs.

By the way, this Gyun! When playing in Atmos, the ceiling speaker makes a screeching noise.

Ceiling speakers do not sound
Doesn’t the ceiling speaker sound too much? [The truth of Atmos]
While playing a movie in Dolby Atmos, there are surprisingly many times when there is no reaction, even if you put your ears close to the ceiling (top speaker) or touch the speaker. …
This is not because there is deep bass, but because the sound does not bleed and the freshness is high, the sound is in focus, and the original sound of the production intent is synthesized and reproduced correctly.

On the other hand, it is undeniable that Dolby Digital has a slightly fuzzy sound.

But this is a special environment. It’s hard to feel it alone when switching to TrueHD immediately after listening with Dolby Digital Plus.

It’s a difference that you can’t tell even if you do some other work and listen to it 15 minutes later.

Also, Mad Max sounds good right from the start. It is even used in manufacturers’ listening booths.

Yodobashi Camera Multimedia Akiba
[In Tokyo] I went to find a store where I could listen to the sound of a home theater!
Where is the store where you can hear the sound of the latest home theater system? What happens when you visit a store? I want to know that …
That’s why Dolby Digital Plus is pretty good, making it awkward to judge (laughs).

Eigako eigako
To be honest, I didn’t say I made a mistake …
Kurokawa Kurokawa
Actually … I also recorded a video, but I’m not sure because the recording is a smartphone, which is ridiculous quality! sorry!
Eigako eigako
This YouTube video is the worst (laughs)
It may be possible to make a slight judgment even on the level rude to the reader … It is fatal that the volume is different

However, I was able to confirm that Dolby Digital ⇒ Plus ⇒ TrueHD will gradually improve.

This conclusion

For lovers of surround sound, the difference between Dolby Digital Plus and TrueHD is quite large.

If ordinary people prefer to hear the best surround sound with a 30,000 yen system, I think a home theater that plays 5.1ch Blu-ray or 7.1ch Dolby TrueHD is the best.