“High resolution” basics. What is the difference between DSD, FLAC, MQA, etc.?


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“High resolution” basics. What is the difference between DSD, FLAC, MQA, etc.?

High Resolution Audio
There are several formats, even if it says “high resolution”.

High Resolution audio

If the format changes, the amount of sound information and thus the sound quality will change, the file size will also change, and whether the playback device / software will support it or not, it will also change, so choose a format is important. We will explain the main formats incorporating common technologies and unique pieces.

There are several high-resolution formats, but …
What is the sampling frequency?
Most digital sound sources are “linear PCM”. This is data obtained by digitizing (sampling) the sound waveform (analog signal) in a canned cycle, and that cycle is called the “sample rate.” If sampling is done every 1/44100 of a second it will be “44.1 kHz”, if it is 1/96000 of a second it will be “96 kHz”, if it is 1/192000 of a second it will be “192kHz”. This means that the implementation cycle of is shorter and the amount of information is greater. In other words, if you look at this number, you can see “how finely the sound was measured with respect to time.”

What is the number of quantization bits?
Value indicating the number of steps in which the amplitude of a signal is expressed when an analog signal is converted to a digital signal (AD conversion) for linear PCM generation. The higher the value, the finer the amplitude of the sound can be captured, and the closer the waveform is to the original sound (analog signal), the more accurately the sound can be reproduced with high resolution.

The large number of quantization bits is directly related to the resolution of the data. For example, when the number of quantization bits is 1 (1 bit), the width of the expression is 2 steps of “0 or 1”, but in 2 bits, there are 4 steps of “00” and “01”, ” 10 “and” 11 “. They can be expressed. Similarly, 4-bit has 16 steps, 8-bit has 256 steps, 16-bit has 65,536 steps, and 24-bit has 16,777,216 steps, allowing for detailed expression.

The “dynamic range” of the maximum / minimum sound ratio that can be handled with the sound waveform data is determined by the number of quantization bits. The dynamic range of human hearing is about 120 dB, but when the number of quantization bits is 16 bits, it reaches 96 dB, but when it is 24 bits, it reaches 144 dB, and when it is 32 bits, reaches 192 dB. (increases by 6 dB for each additional bit) If it is a high resolution sound source, it can handle tiny sounds to powerful sounds with a margin.

Original waveform data. The vertical axis is the sample rate and the horizontal axis is the number of quantization bits.

The higher the sampling frequency (finer the horizontal axis) and the greater the number of quantization bits (finer the vertical axis), the richer in information the “high resolution” sound becomes.
Definition of “high resolution”
High-resolution quality audio sources (hereinafter referred to as high-resolution sound sources) excluding DSD are distinguished by the sample rate and number of quantization bits described above. The term “more than CD specifications” is often used, which means that it is based on the sampling frequency (44.1 kHz) and the number of quantization bits (16 bits) of the CD.

As defined by the Japan Electronics and Information Technology Industries Association (JEITA), high-resolution audio must “either the sampling rate or the number of quantization bits exceed CD specifications”, and sources high resolution sound will follow this.

On the other hand, the Japan Audio Association also defines high resolution, and the recommended high resolution logo is given to audio equipment that guarantees its playability. This standard is divided into analog and digital systems, and the file format is also mentioned, such as setting the standard for high-resolution sound sources to be “96 kHz / 24-bit FLAC and WAV compliant.”


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High resolution sound source

High resolution sound source

Sample Rate

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

44100 Hz and 48000 Hz

Just add “USB-DAC” to the audio you use all the time, and you can enjoy very high quality sound! Therefore, this time, Sarah visited Onkyo Co., Ltd., which developed the most advanced “USB-DAC” that supports high-quality sound sources called “high resolution”, which has been released 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’m thinking about getting you to experience high-quality sound in a number of ways.

Sara-chan: Thank you! To enjoy music on your PC with high sound quality, “USB-DAC” is definitely important! 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 hell is that ??

Kurosawa-san: “USB-DAC” is a device that converts sound from digital signals to analog signals, right? 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 off 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 of 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-making 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 like the sound is expansive and I feel like I’m surrounded by sound! Anyway, it seems like I’ve never heard it on audio before! Impressed!

Mr. Kurosawa: Fufufu. Can you see 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”.

The difference between 44,100 Hz (music industry) and 48,000 Hz (video industry) part 2

The difference between 44,100 Hz (music industry) and 48,000 Hz (video industry) part 2

sample rate

2) When recording a 48 kHz music video to match the music played at 44.1 kHz on the site. In this case, it can be very difficult to match performance lips to post-production due to the different sample rates of the sound being played. It’s called sink drift.

SAMPLE RATE

3) Another point It seems that this is a problem that occurs at the time of recording, but there is a problem that the sound changes gradually when the sound recorded separately using a cheap recorder is synchronized with the sound recorded in the reference of the video. it seems that there are moments. In this case, it seems that it is necessary to manually and quickly advance the video a bit and match it with the audio file, or extract some frames at the important points of the audio and synchronize it. It seems that this has nothing to do with the sample rate, so I will describe it so as not to cause misunderstandings.

The difference between 44,100 Hz (music industry) and 48,000 Hz (video industry)

The difference between 44,100 Hz (music industry) and 48,000 Hz (video industry)

44100 Hz vs 48000

In video production, record the frame rate for shooting and the sample rate for recording. Remember this is one of the basics for shooting and recording.

44100 48000

First, about the difference in sampling frequency. Generally speaking

44,100Hz (44.1kHz) is the standard in the music industry

48,000Hz (48kHz) is the sound standard in the video industry

The difference between the two sample rates is just that. I spoke of the sample rate as
the frame rate in video in another article, “Sound Principles Required for Video Production,” Sample Rate and Bit Depth. ”
In other words, the higher the sample rate in Hz, the softer the sound will be.

There are several theories about the historical background of 44,100Hz.
I would like to introduce you to one of the most logical.

First, when sampling sound, you need a sample rate that is at least twice the highest frequency you are recording. This is the sample rate required to obtain a minimum of the waveform. This is because it is not possible to record a sound that has the character of a wave if there is only one place to take a sample. Most people say that the audible range is 50 Hz to 16,000 Hz. Double is 32 kHz, but it seems that the harmonic components that make up the tone need to be recorded in order to record the voice correctly. Only when this is taken into account does it appear that up to 44,100Hz is required. Click here for more details.

Sound Processing “I want to hear my voice clearly” (link outside Vook’s site)

What happens when the sample rate is low?
When digitizing analog information, if the sample rate is not high, the high-frequency information will be hidden in the low-frequency information.
Then the high-frequency sound will be recorded as low-frequency sound. Specifically, see the following illustration.
This is called aliasing.

See also: Wikipedia https://en.wikipedia.org/wiki/Aliasing#/media/File:AliasingSines.svg

In any case, by definition, 48,000 Hz has better sound quality than 44,100 Hz. The video industry has introduced 48,000 Hz.

One problem that sometimes occurs is that “I was recording a video at 48 kHz and the separately recorded microphone was set to 44.1 kHz.” At first I thought that different sample rates would be a big deal, but it doesn’t really seem to be the case.

Audio recorded at a small sample rate just has a small number of samples per second, but since there is almost no difference between 44.1 kHz and 48 kHz, I think the difference is barely noticeable when listening to the sound regularly. At 96 kHz, the sound quality is even higher, but the number of samples is so large that ordinary people cannot hear it at all.

In some cases, the sample rate is really important.

1) By writing the audio actually recorded with a different sample number as a video file. This is because the sample rate must be converted to a video sample rate that is different from the conventional 44.1 kHz and 96 kHz sample rates, that is, 48 ​​kHz. Software that specializes in video editing seems to have sound distortion at this point.

About the frequencies used for audio (sample rate, PCM, DSD, etc.)

About the frequencies used for audio (sample rate, PCM, DSD, etc.)

Sample Rate

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

Audio Sample Rate

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 44.1 KHz. 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 the music signal is a set of sine waves, it is 44.1 KHz that can be shaken at the maximum frequency of 20 KHz.

2. What are 16 bits and 24 bits?

As you often hear, 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.

If this is converted to dB at 20LOG (65536), it will be approximately 96dB. 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 144 dB in 16.77 million steps.

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 32 bits of 2 channels (L, R) alternately (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.

Sample rate and bit rate Part 2

Sample rate and bit rate Part 2

sample rate and bit rate

Sampling theorem

Sample Rate & Bit Rate

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 timing) .

44.1 kHz

Why is there a poorly separated rate of 44.1? .. ..

Didn’t technicians deliberately make cumbersome clocks 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 the human ear usually cheats (it’s just my personal opinion).
However, the objective evaluation method will soon come to an end. For example, you can clearly see that you were 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 musical signal is reproduced in the range of 0 to 65535. dog. 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 from the age of analog communication (analog mobile phone), digitization + compression will be able to support multiple calls with the same radio wave.

Finally

I often hear what is called Hi-Res Audio. The sampling frequency is said to be 96 kHz or 192 kHz, which is over 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 audio example
Sampling frequency Number of quantization bits Number of channels bit rate Frequency that can be expressed
192 kHz 24 2 9,216 kbps 96 kHz
192 kHz 16 2 6,144 kbps 96 kHz
96 kHz twenty-four 2 4.608 kbps 48 kHz
96 kHz 16 2 3,072 kbps 48 kHz
48 kHz twenty-four 2 2,304 kbps 24 kHz
Considering the limit of human hearing (about 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 little. Feels like it’s going up (?) (It’s just a story from my ears).
I’d like to make a comparison if I get the chance, but I don’t think I can tell by ear without a proper regenerator (like an expensive analog amp).

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

Sample rate and bit rate

Sample rate and bit rate

Sample Rate and Bit rate

If the file size is reduced (code at a lower bit rate), the sound quality tends to deteriorate. How much should it really be? .. ..

sample rate bit rate

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 high and the file size when saved will be small, but what is the bitrate for the original sound source (PCM) without compression in the first place?

If you save it as PCM, the sound quality of the original sound will be obtained, but it can be a little inconvenient to save it without worrying about the file size. Also, depending on the application, I think the original sound size has enough memory capacity and the communication speed is correct. 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, 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 back. 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., you will naturally need to specify a bitrate less than 1,411.2 kbps. For example, when encoding at 256 kbps, the compression rate is approximately 18% and the file size is 1/5 or less when the original sound is 100%.

Encode Music CDs at 256 kbps: 256 kbps / 1,411.2 kbps = approximately 18%
Generally, the sample rates of audio devices actually connected to a PC 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 2 1,536 kbps
44.1 kHz 16 2 1,411.2 kbps Music CD
32 kHz 16 2 1,024 kbps
24 kHz 16 2 768 kbps
22.05 kHz 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 1 512 kbps Super Wide Band
24 kHz 16 1 384 kbps
16 kHz 16 1 256 kbps Broadband
8 kHz 16 1 128 kbps Narrowband

Sampling rate

If you check the web, there are explanations such as 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 can be expressed 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.

What is the fundamental difference between 44100 and 48000 Hz?

What is the fundamental difference between 44100 and 48000 Hz?

44100 vs 48000 hz
44100 vs 48000 hz

In fact, this is just a question of long-standing standards.

44100 vs 48000 hz
44100 vs 48000 hz

44100 vs 48000 hz

44100 is the CD standard.
48000 is the standard for DVD.
The difference in practice is so small that it will be impossible to notice it (I’ll tell you more: many people feel the difference between mp3 and wav, but they can’t tell which is better).
The stereotype has persisted that if you need to work with TV or movies / soundtracks, it is better to do it in 48000, suddenly some old equipment will not understand sampling.
But this is very, very unlikely these days, so there isn’t much of a difference.
It can record at 96000. There is a small chance that some plug-ins / sound effects can handle such recordings better, but it requires more CPU / RAM and much more hard disk space.
Between 16 and 24 bits, it will also be difficult to feel the difference, but at the request of the sound engineer, we wrote in 24 with the same thoughts (for plug-ins).
In general, write to 44100 if you don’t need to work with a specific television crew.

44100 vs 48000 hz
44100 vs 48000 hz

Choosing the Right Sample Rate: 44100 or 48000 hz

 

In the world of digital audio, the choice between 44,100 Hz and 48,000 Hz sample rates is a critical one. As an audio expert, I’ve spent years diving deep into this topic, examining the real-world scenarios where this choice can make or break a sound. In this article, I’ll guide you through this audio journey, shedding light on the differences and helping you make an informed choice.

44100 Hz – The Analog Heartbeat

When we talk about 44,100 Hz, it’s like stepping into a cozy vinyl record shop, where the warm crackles and pops surround you. This sample rate mirrors the heartbeats of analog audio, capturing the subtleties of your audio source much like a vintage vinyl record player.

Imagine: You’re in a dimly lit jazz club, and a saxophonist takes the stage. You close your eyes as the music begins. 44,100 Hz is akin to capturing every breath, every emotion, and every nuance of the saxophonist’s performance. It’s the sample rate that preserves the soul of analog sound.

48000 Hz – The Digital Precision

Contrastingly, 48,000 Hz feels like entering a state-of-the-art recording studio with a digital mixing console at the heart of it all. It’s the precision tool for audio in the digital age, where every sound wave is charted with utmost accuracy.

Visualize: You’re in a high-tech laboratory, and a scientist is conducting a finely tuned experiment. 48,000 Hz is like the precise instruments that measure every data point with accuracy. It’s the sample rate that excels in capturing the clarity and detail of digital audio.

The Real-World Decision

The choice between 44,100 Hz and 48,000 Hz ultimately depends on the nature of your audio project.

Subtitle: For Vintage Vibes

If you’re aiming for a warm, nostalgic sound reminiscent of classic records, 44,100 Hz is your choice. It’s like using a vintage camera to capture that old-world charm. This sample rate will maintain the character and imperfections of your audio source.

Subtitle: For Contemporary Clarity

When you require crystal-clear audio for modern projects, such as podcasts, video games, or high-quality music production, 48,000 Hz is your ally. Think of it as upgrading to a high-definition TV for the audio world. This sample rate ensures every detail is captured and reproduced faithfully.

Last words about right sample rate for your digital audio

As an audio expert, my journey has led me to understand that the choice between 44,100 Hz and 48,000 Hz is about preserving the essence of your sound in the most appropriate way. Each sample rate has its place in the vast world of audio, just as a painter chooses different brushes for different strokes on their canvas.

So, whether you’re embracing the warmth of the past or striving for the precision of the future, remember that the right choice of sample rate can be the difference between an audio masterpiece and a missed opportunity. Choose wisely, and let your sound shine in all its glory.

 

The fundamental difference between them in the coverage of the frequency range on the track (from 20Hz), the 44100 sample rate allows you to work in the range up to 22kHz, 48000 to ~ 25kHz, 96000 to ~ 35kHz, etc. 48 parameters o 96kHz are used in large studios where the reproduction of these frequencies and sound engineers strive for the slightest increase in sound quality, before and after conversion to the 44100 standard, the sound of the track objectively looks better, even though the human ear does not hear these frequencies, the psychoacoustic effect remains (the closest example: if you shoot video and plan to play back in fHD, you will prefer to shoot 4k with rear cropping for the sake of image quality, and no one will say there is no point in shooting 4k, the same is here).

It’s even more interesting in movies … because 44100Hz is the playback frequency at 24fps and 48000Hz is 25fps. If you record a video at 25 fps and the sound is separately on the recorder at 44100Hz, then the length of the tracks will not match and you will have to change the timbre of the original with a small time interval.

High-end sample rate conversion

High-end sample rate conversion

Sample Rate Conversion

The sample rate is the number of measured digital signal samples (passes) per second.

Sample Rate Conversion

High-quality conversion (change) of the sample rate is quite a complicated and resource-intensive process. Especially if the frequencies of the input and output signals are not multiples of each other (44.1 and 96 kHz). Next, we will look at the characteristics of the audio sample rate conversion process that affect sound quality.

About the DSD sample rate conversion.

Sample rate converter for Mac OS X, Windows

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Where are sample rate converters used?
Sample rate conversion can be: in real time (on the fly, converting the audio stream signal) or by converting files.

Sample rates are changed in real time when playing samples and mixing multiple audio tracks from the sequencer program (imported from external files with different sample rates).

In audio engineering, the 2 series of sample rates are mainly common:
1) CD: 44 100, 88 200, 176 400 Hz;
2) DVD Audio and DVD Video: 48,000, 96,000, 192,000 Hz.

Not only musicians and professional sound engineers need to bring the sample rate to the desired value, but also in the field of home audio and video. For example, when playing audio files, a media player may imperceptibly “adjust” the sample rate of the file to the sample rate set in the sound card settings.

Sample rate conversion algorithm
The algorithm for changing the sample rate (both hardware and software) consists of the following steps:
1) Increase the sampling frequency to a frequency that is a multiple of the sampling frequency of the output signal.
2) Filters out “spurious” signals (called “artifacts”) that are above half the output sample rate.
3) Multiple decimation subsampling (discarding) unnecessary samples.

Sample rate converter circuit

Up sampling is done by inserting additional samples (“virtual” – generated by the interpolator) between the existing samples in the input digital signal.

Sample interpolation: insert virtual samples between real ones

It is sometimes used to insert “virtual” samples with zero values ​​into the digital signal. This method is computationally faster. But this way of increasing the sample rate adds a significant amount of “artifacts” to those present in the interpolated signal.
Why do you need a superior sample? To complete point 3). Since it is easier to dilute the samples in multiples, simply discarding the excess ones.
The “spurious” signals (with frequencies above half the output sample rate) are then filtered. Otherwise, discarding “extra” samples will fall into the spectrum of the useful signal and distort it (add extraneous sounds).

What makes a high-end audio sample rate converter different from a medium-quality converter?
To introduce minimal distortion into the signal during conversion, we must interpolate it as accurately as possible. The interpolation precision is the maximum degree of repetition of the additional interpolator samples of the original analog signal. It should be remembered that the highest quality interpolator can accurately reconstruct the original analog signal. But not with 100% accuracy. Poor me. When the sampling frequency is increased, false signals will appear above half the sampling frequency of the output signal.

Samplerate, what is sample rate

The sampling frequency is the time that results from the time between two samples. It is given in samples per second (S / s).

Sampling Rate

The level of the sampling frequency is a criterion for the reproducibility of the frequency of the sampled signal. The closer the sampling times are, the better the signal can be reproduced.

Sampling rate

Relationship between frequency and sampling frequency

For example, if an analog signal is sampled once per millisecond (ms), the sample rate is 1 kHz and the sample rate is 1000 samples per second. If the sampled signal has a frequency of 1 kHz, the signal is sampled once per period. It cannot be played. If, on the other hand, the frequency of the signal is 100 Hz, the signal is sampled ten times with the same sample rate. Therefore, the signal is easily reproducible. Therefore, the sampling frequency must be in a certain relation to the frequency of the signal. This relationship is through the given sampling theorem. Accordingly, the reproduction of the signal requires a sampling frequency that is at least twice the frequency of the signal. This applies to sine-type signals for their 1st harmonic, but not to square wave or pulse signals.

Audio sampling frequencies

In the case of voice transmission over ISDN with a maximum frequency range of 4 kHz, the sampling frequency is 8 kHz, which corresponds to a sampling interval of 125 µs. For audio with a maximum frequency range of 20 kHz, the sampling frequency is 44.1 kHz (22.67 µs) and 48 kHz (20.83 µs). For high-quality multi-channel audio, the sample rate can be up to 192 kHz. Much higher values ​​are found for video and HDTV. For digital video, this results in a 6.5 MHz bandwidth for the luminance signal, a sampling frequency of more than 13 MHz and a sampling interval of 74 ns. The sample rate for HDTV is even higher with 74 MHz and a sample rate of 13.5 ns.
In the case of pulse-shaped signals, the sampling frequency must be many times greater than its fundamental oscillation, since otherwise important pulse parameters cannot be determined. If the sample rate is many times higher than the theoretically required sample rate, we are talking about oversampling.