What is the Nyquist Frequency?

Free Download Mp4Gain

What is the Nyquist Frequency?

Nyquist Frequency: Understanding the Basics

When it comes to digital signal processing, one of the most important concepts is the Nyquist Frequency. Simply put, the Nyquist Frequency is the highest frequency that can be accurately represented in a digital signal. But what exactly does that mean? Let’s break it down.

Imagine you are listening to a song on a CD. The CD player reads the music as a series of 0s and 1s, which are then converted into electrical signals that can be played through speakers. But how does the CD player know what the music sounds like? It uses a process called sampling, which involves taking a snapshot of the music at regular intervals.

The Nyquist Frequency comes into play because of this sampling process. According to the Nyquist-Shannon sampling theorem, in order to accurately represent a signal in digital form, you need to sample it at least twice as fast as the highest frequency you want to represent. This means that if you want to accurately represent a signal that contains frequencies up to 20kHz (which is the upper limit of human hearing), you need to sample it at least 40,000 times per second.

Nyquist Rate: What You Need to Know

The Nyquist Rate is the minimum rate at which a signal must be sampled to accurately represent it in digital form. It is calculated by multiplying the highest frequency you want to represent by two. For example, if you want to represent a signal that contains frequencies up to 10kHz, the Nyquist Rate would be 20,000 samples per second.

It’s important to note that sampling a signal at a rate that is too low can result in a phenomenon called aliasing. Aliasing occurs when a higher frequency signal is incorrectly represented as a lower frequency signal. This can cause distortion and other unwanted effects in the digital signal.

To avoid aliasing and accurately represent a signal, it’s crucial to sample at or above the Nyquist Rate. In fact, many digital audio devices sample at rates much higher than the Nyquist Rate to ensure high-quality audio reproduction.

Analog-to-Digital Conversion: The Role of the Nyquist Frequency

Analog-to-digital conversion is the process of converting an analog signal (such as an audio waveform) into a digital signal that can be processed by a computer. This process involves sampling the analog signal at regular intervals and converting each sample into a digital value.

The Nyquist Frequency plays a crucial role in analog-to-digital conversion because it determines the minimum sampling rate required to accurately represent the analog signal in digital form. If the sampling rate is too low, the resulting digital signal will be inaccurate and distorted.

To ensure high-quality analog-to-digital conversion, it’s important to sample the analog signal at or above the Nyquist Rate. This will result in a digital signal that accurately represents the original analog signal and can be processed and manipulated with high precision.

As the famous engineer and inventor, Nikola Tesla said, “The day science begins to study non-physical phenomena, it will make more progress in one decade than in all the previous centuries of its existence.” The Nyquist Frequency is a prime example of the intersection of science and engineering, and its importance cannot be overstated.

Final Words

In conclusion, the Nyquist Frequency is a fundamental concept in digital signal processing that plays a crucial role in accurately representing analog signals in digital form. By understanding the Nyquist Frequency and its relationship to sampling

Free Download Mp4Gain

Mp4Gain Main Window

Mp4Gain Features

Free Download Mp4Gain

Sampling Frequency in Digital Audio

The Role of Sampling Frequency in Digital Audio

Importance of Sampling Frequency in Digital Audio

Sampling frequency, also known as sample rate, is a crucial component of digital audio. It determines how many times per second an analog audio signal is measured and converted into a digital format. The higher the sampling frequency, the more accurately the original sound can be captured and reproduced.

As an audio engineer, I’ve had my fair share of experiences with different sampling frequencies. In my opinion, the importance of sampling frequency cannot be overstated. When working with high-quality audio, a low sampling rate can result in audible artifacts and distortion. On the other hand, using a high sampling rate can drastically improve the clarity and fidelity of the final product.

According to the book “Digital Audio Engineering” by John Watkinson, “An increase in the sampling rate produces an increase in the bandwidth and reduces the aliasing distortion.” This means that by increasing the sampling frequency, we can capture more of the original sound and reduce unwanted noise and distortion.

Digital Audio Sampling Rate

The sampling rate is measured in Hertz (Hz) and is typically represented as kHz (kilohertz). Common sampling rates for digital audio include 44.1kHz, 48kHz, and 96kHz. The standard for CD-quality audio is 44.1kHz, while higher sampling rates are often used in professional audio production.

In my experience, using a higher sampling rate can make a noticeable difference in the final sound quality. However, it’s important to note that higher sampling rates also require more storage space and processing power. For example, recording at 96kHz requires twice as much storage space as recording at 48kHz.

As stated in the book “The Art of Digital Audio” by John Watkinson, “The required storage capacity increases linearly with the sampling rate.” This means that higher sampling rates can result in larger file sizes and slower processing times. It’s important to weigh the benefits of increased audio quality against the practical limitations of storage and processing power.

Impact of Sampling Rate on Audio Quality

The impact of sampling rate on audio quality can be significant, particularly when working with high-fidelity audio. In my experience, a higher sampling rate can result in a more natural and dynamic sound.

As explained in the film “Sound City,” “If you’re going to capture music with any sort of fidelity, you have to have a high sampling rate.” This sentiment is echoed by many audio professionals, who believe that a higher sampling rate is essential for capturing the nuances and subtleties of live music.

However, it’s important to note that not all audio sources require a high sampling rate. For example, speech recordings and low-quality audio files may not benefit significantly from a higher sampling rate.

Sampling Frequency and Audio Fidelity

Audio fidelity refers to the accuracy and authenticity of a sound recording. The sampling frequency plays a critical role in achieving high audio fidelity.

As stated in the book “The Science of Sound Recording” by Jay Kadis, “The higher the sampling rate, the more accurately we can represent the waveform.” This means that a higher sampling rate can result in a more accurate and faithful reproduction of the original sound.

What is the difference between 128k and 320k music? Part 3

The sampling frequency is approximately the following depending on the type of use (k is the thousand bit symbol, 1khz=1000hz):

8khz – used for phones etc, is enough to record human voices.

22.05khz: transmission use frequency.

44.1kb: Audio CD.

48khz: used in DVD and digital TV.

96khz-192khz: used for DVD-Audio, Blu-ray HD, etc.

The common range of sample precision is 8 bits to 32 bits, with 16 bits generally used on CD.

Having said that, my friends are starting to get confused. It’s not the bitrate that determines the sound quality, so why is everyone saying that 320kb sound quality is better than 128kb?

【Audio Compression】

Well, in fact, the bit rate should be said to be another dimension, it is a compression of audio files.

Nowadays, most of the audio formats we use regularly are based on the original “WAV” file of the audio CD (44.1khz sample rate, 16bit sample precision, 2ch). The original recorded sound data is stored in a matrix, which is in PCM format, while WAV format is an encoding format developed by Microsoft. Its function is to reproduce the data in PCM format through encoding.

Since the data in WAV basically completely restores the PCM data, MP3, AAC and other lossless encoding formats are basically recompressed based on the WAV files. Therefore, we can simply think that WAV is the original audio format and other audio formats are compressed formats.

When it comes to compression, storage and transmission are inseparable. The purpose of compression is to improve storage and transmission. Therefore, before we talk about compression, we need to understand the basic units of computers.

We all know that the computer is a binary number system, and the files stored by the computer are made up of two numbers, 0 and 1. Therefore, the computer’s transmission is based on each number, and each number is called 1 ” bit”. For example, for an audio piece, its basic data is “0,1,1,1,0,1, 1 ,0”, and when transmitting, these numbers are transmitted one by one. The sampling precision mentioned above is this unit.

The storage unit of the computer is “byte (Byte)”. In the computer, 1 byte consists of 8 bits, that is, 8b(bit)=1B(Byte). In computer parlance, data storage is expressed in decimal and data transmission is expressed in binary, so 1KB=1024B=1024×8b. This is also part of the reason why the hard drive capacity we see does not match the actual capacity.

Go back and talk about audio compression, the bitrate of the audio is actually the compression ratio. So the bitrate really just defines the size of the file, but because under normal conditions the larger the file, the less data you lose, so the sound quality is relatively higher. However, the bit rate itself does not directly affect the quality of the file. For example, if we take a 128kb file as the source file, even if it is converted to a 320kb file, the sound quality will not be better than 128kb. .

What is the difference between 128k and 320k music? Part 2

Bit rate, sample rate, lossless, MP3, FLAC, APE, 320kb, 192kb, 128kb, 44.1khz, CBR, VBR. Does this bunch of various names make you both familiar and unknown?

The higher the bitrate, the better the sound quality. Lossless music is the highest sound quality, right? So, let’s start with the sound collection.

【Audio composition】

Nowadays, when we talk about audio, everything is digital audio. Digital audio consists of three parts: sample rate, sample precision, and number of sound channels.

Sample Rate: Both the sample rate, which refers to the number of samples per second when recording the sound, expressed in Hertz (Hz).

Sampling Precision: Refers to the dynamic range of the recorded sound, measured in bits (Bit).

Sound channel: the number of channels (1-8).

In simple terms, we can think of a sound wave as a curve. We know that the curve is made up of points, and the sampling frequency is the number of points in the middle of the length per second (the horizontal axis of the figure above). Sampling precision is the number of points in the dynamic range (upper vertical axis). The finer the positioning of these two dimensions, the greater the true sound restoration and the better the sound quality. Of course, the larger the audio file will be. The customer mentioned by the previous colleague said that the latest Hi-Res Audio format released by SONY is a 6-channel 192kHz/24-bit recorded audio file. The size of the lossless format, of course, will be more than 200 megabytes.

What is the difference between 128k and 320k music?

I can’t fully understand music in words.show all

【Preface】

Some time ago, a colleague came across a very troubled client. The mess was said to have been caused by the client asking him to provide song files larger than 100MB-200MB in size. And my colleagues don’t know much about audio formats, so they started endlessly fumbling about FLAC, WAV and audio size. In the end, the colleague did not clearly explain to the customer what was going on.

After that, some other things happened that made me feel that in the music industry there are too many practitioners around me who have an extremely poor understanding of music and even lack some basic knowledge related to music. I don’t even have the idea to understand, which makes me very sad. It seems that music has only one merchandise attribute, and our practitioners only need to organize the shelves, encode various merchandise, and use the big data of users’ purchase records to recommend merchandise to users, no matter why to users. they like this. features that these products have, and use cold data to provide users with various services.

Therefore, I think it is necessary to write something. I don’t expect practitioners to become people who really love music. I just hope that even if you still think of “her” as a commodity, you can first figure out what you’re selling. and what is..

PS: The content of the first lesson is about media files. Since the relevant content involves a lot of technical issues, it seems a bit boring, but if you read it carefully, you will find that it is actually very easy to understand, but this basic knowledge can be very helpful.Improve your skill well. Also expect more interesting content about records, musical styles, etc. which I will post soon.

Related Audio Attribute Part 3

How samples are combined

This is mainly for two-channel or multi-channel audio. For a two-channel audio, it can be combined in the following two ways:

interleaved Taking stereo as an example, a stereo audio sample is obtained by interleaving the storage of two mono samples.
flat. The samples of each channel are stored separately.

The data after FFmpeg audio decoding is stored in the AVFrame structure.

In packed format, frame.data[0] or frame.extended_data[0] contains all the audio data.
In Planar format, frame.data[i] or frame.extended_data[i] represents the data of the i-th channel (assuming channel 0 is the first), the size of the AVFrame.data array is set to 8, if If the number of channels exceeds 8, you should get the channel data from frame.extended_data.

sample format
The sample formats in FFmpeg are mainly:

copy code
enum AVSampleFormat {
AV_SAMPLE_FMT_NONE = – 1 ,
AV_SAMPLE_FMT_U8, /// < 8 bits unsigned
AV_SAMPLE_FMT_S16, /// < 16 bits
signed AV_SAMPLE_FMT_S32, /// < 32 bits
signed AV_SAMPLE_FMT_FLT, /// < float
AV_SAMPLE_FMT_DBL, /// < double

AV_SAMPLE_FMT_U8P, /// < 8 bits unsigned, flat
AV_SAMPLE_FMT_S16P, /// < 16 bits signed, flat
AV_SAMPLE_FMT_S32P, /// < 32 bits signed, flat
AV_SAMPLE_FMT_FLTP, /// < float, flat
AV_SAMPLE_FMT_DBLP, /// < double, flat
AV_SAMPLE_FMT_S64, /// < 64 bits
signed AV_SAMPLE_FMT_S64P, /// < 64 bits signed, plain

AV_SAMPLE_FMT_NB /// < Number of sample formats DO NOT USE if dynamically linked
};
copy code
to illustrate:

1. U8 (8-bit unsigned integer), S16 (16-bit integer), S32 (32-bit integer), FLT (single-precision floating-point type), DBL (double-precision floating-point type), S64 (64-bit integer), those not ending with P are interleaved structures, and those ending with P are flat structures.
2. Flat mode is FFmpeg’s internal storage mode, and the audio files we use are in packed mode.
3. The FFmpeg audio sample format that decodes different output audio formats is not the same. The test found that the data output by AAC decoding is in floating point AV_SAMPLE_FMT_FLTP format, and the data output by MP3 decoding is in AV_SAMPLE_FMT_S16P format (the mp3 file used is 16-bit deep). For the specific sample format, you can see the format member in the decoded AVFrame or the sample_fmt member in the AVCodecContext of the decoder.

Bit rate
The transfer rate per second (bit rate, also called bitrate). Like 705.6kbps or 705600bps, where b is a bit, ps is per second (per second), which means a capacity of 705600bit per second. Compressed audio files are often represented at double speed, for example CD quality MP3 is 128kbps/44100HZ. Note that the unit here is bit instead of byte. One byte is equal to 8 bits (bits). The bit is the smallest unit. It is generally used to describe network speed and various communication speeds. The byte is used to calculate the size. hard drive and memory.

Mbps is: Millionbit per second (millions of bits per second);
Kbps is: Kilobit per second (kilobit per second);
bps is: bit per second (bit per second), the corresponding conversion ratio is:

1Millionbit=1000Kilobit=1000000bit; 1Mbps = 1000,000bps; Again, this is the unit of speed, which refers to the number of bits transmitted per second. The unit of measure for data transmission speed K is the decimal meaning, but the K for data storage is the binary meaning. E.g:

The 1M bandwidth generally described is 1 Mbps = 1,000,000 bps = 1,000,000 / 8 / 1,000 = 125; therefore, the download speed of 1M bandwidth generally does not exceed 125KB/s
. 1000 = 12.5, so the maximum download rate of 100M bandwidth can reach 12.5MB/s
. Of course, the above is only the theoretical rate. In fact, the maximum download rate may not reach that much, and it is mainly affected by various losses, generally 100MB A broadband download rate of 10MB is not bad.

Related Audio Attribute Part 2

The higher the sampling, the more realistic and natural the sound will be.

The frequency recognition range for people is 20 HZ – 20,000 HZ. If 20,000 samples per second can be sampled, it will be enough to satisfy the needs of the human ear during playback. So 22050 The sample rate is commonly used, 44100 is already CD quality, and sampling more than 48000 is no longer meaningful to the human ear. This is similar to a 24 frames per second image from a movie.

Sampling bits
After sampling the audio for a sample, two steps must be performed for the sample:

1. Quantify. The quantization bits commonly used for audio quantization are:

8 bits (that is, 1 byte) can only register 256 numbers, that is, only the amplitude can be divided into 256 levels;

16 bits (ie 2 bytes) can be as small as 65536 numbers, which is already the CD standard;

32 bits (ie 4 bytes) can subdivide the amplitude into 4294967296 levels, which is really unnecessary.

The number of quantization bits is also called the number of sampling bits, bit depth, and resolution, and refers to how many levels the continuous intensity of the sound can be divided after being digitally represented. N-bit means that the intensity of the sound is divided equally into 2^N levels. 16 bits, it is level 65535. This is a very large number and people may not be able to tell the difference in sound intensity from 1/65,535. You can also say that it is the resolution of the sound card. The higher the value, the higher the resolution and the greater the ability to produce sound. The sampling multiple here is primarily addressing the strength characteristics of the signal, and the sampling rate is addressing the time (frequency) characteristics of the signal, which are two different concepts.

2. Binary encoding. That is, the result of the quantization, ie the single channel sample, is stored in a binary keyword. There are two storage methods:

Store the result of the quantization directly in the cast, that is, the two’s complement code;

The result of quantization is stored in floating point type, ie floating point encoding code.

Most PCM sample data formats use integers to store, and for some applications that require high precision, use floating point to represent PCM sample data.

frame
After the audio is quantized to a binary codeword, it must be transformed and the transformation (MDCT) is done in block units, and a block is made up of multiple (120 or 128) samples. A frame will contain one or more blocks. Common frame sizes are 960, 1024, 2048, 4096, etc. A frame records a sound unit whose duration is the product of the sample duration and the number of channels. The nb_samples in the AVFrame structure in FFmpeg represent the number of single channel audio samples in a frame.

Related Audio Attribute

channel, sample rate, sample bits, sample format, bit rate

Sample Rate

The PCM obtained from audio sampling contains three elements: channel, sample rate, and sample rate.

channel
When people hear the sound, they can locate the sound source. By setting the sound source to different positions, a better listening experience can be created. If the position of the audio is adjusted with the image, a better audio-visual experience will be obtained. Effect. Common channels are:

monkey monkey
Two channels, stereo, the most common type, including left and right channels
2.1 channels, adding a bass channel on the basis of two channels
5.1 channels, including one front channel, one front left channel, one front right channel, one surround left channel, one surround right channel, and one bass channel, first used in early theaters
7.1 channel, on the basis of 5.1 channel, the surround left and right channels are divided into surround left and right channels and rear left and right channels, mainly used in BD and modern theaters
Next is a two-channel audio system.

Sampling rate
Audio sampling is the conversion of sound from an analog signal to a digital signal. The sample rate is the number of times the sound is collected per second and is also the number of samples per second of the resulting digital signal. When sampling sound, common sample rates are:

8,000 Hz – telephone sampling rate, sufficient for human speech
11,025 Hz – sample rate for AM radio
22,050 Hz and 24,000 Hz – sample rate for FM radio
32,000 Hz – sampling for miniDV digital camcorder, DAT (LP mode)
44,100 Hz – Audio CD, also commonly used in MPEG-1 audio (VCD, SVCD, MP3) Sample rate 47 250
Hz – Sampling frequency
48,000 Hz for commercial PCM recorders – for miniDV, digital TV, DVD, DAT, movies, and pro audio Sampling rate 50,000 Hz for 2,000 – 96,000 or 192,000 Hz digital sound
for commercial digital sound recorders
– DVD-Audio, some LPCM DVD soundtracks, BD-ROM (Blu-ray Disc) and HD-DVD (High Definition DVD) soundtracks The sample rate used by the audio track
2.8224 MHz: The sample rate used by Direct Stream Digital’s 1-bit sigma-delta modulation process.

Definition of sampling bits, sampling rate and bit rate in audio (transfer) Part 3

1. Why do many professional standards reach 24bit/192KHz?

It is now common to use the 48kHz or 96kHz recording rate in engineering, and only convert to the 44.1kHz CD format during the final mastering process, which reduces distortion caused by multiple sample rate conversions.

In the field of computing, the AC97 specification, which is an audio hardware codec standard, only specifies 48 kHz. This causes nearly all input and output signals to be resampled (the professional term is called sample rate conversion, or SRC). SRC generally causes loss of sound quality, and the simpler (ie poorer) SRC algorithms can cause significant deterioration of sound quality. But this is already a fait accompli.

2. Since 44K is enough, why use 192KHZ to record?

First of all, 20kHz is just the hearing threshold for most people, i.e. the human ear is very insensitive to sounds above 20kHz. Insensitivity to attention does not mean a total inability to perceive. The tones of most musical instruments (especially pianos and strings) are rich in higher harmonics, known in musical terms as higher harmonics. CD audio with a cutoff frequency of 22.05 kHz gives people who are used to listening to real instruments an unnatural feel, especially in the high frequencies, because the Nyquist cutoff frequency distorts the signal from harmonics. of higher frequencies.

Second, digital recordings often require post-processing. Audio processing can introduce more distortion into the signal, including signal distortion, spectral aliasing, and more. If the original signal is only sampled at 44.1 kHz during recording, it must be upsampled before post-processing to expand the sample rate. Since this expansion is “fake”, there is really no more useful original signal, and the quality of the upsampling algorithm will also affect the distortion of the original recording signal, so this approach is undesirable. Therefore, it is common practice to sample at a higher frequency.

In today’s fully professional digital recording studios, recording, mixing and mastering are no longer compliant with the CD standard, instead the HD audio standard is preferred. which:

Use 24Bit 48KHz, 24Bit 96KHz, 24Bit 192KHz three specifications to record, of course, 24Bit 48KHz is used by some small recording studios, because their processor resources are limited. And all the big recording studios use 24bit 96KHz and 24bit 192KHz for recording.

So what are the benefits of such a recording specification?

1. Comply with HD audio standard, which is also the main standard in the future. The finished product can be directly applied to HDCD, DVD-Audio, Blu-ray disc, digital music download business and digital player business to media.

2. Fully take care of the digital video and video business, and the multi-channel film and video will adopt the HD audio specification. Including the use of portable mobile digital video equipment.

3. Fully take care of the consumer audio playback business, such as: Intel HD-Audio audio standard, AC97 audio codec, MP3 / mp4 / phone / game console portable audio highest quality audio playback.

Currently, the highest quality standard in the professional recording industry is: 24 bits deeper than a specific point, 192000 Hz sampling rate, referred to as “24 bits/192 KHz”. Of course, this standard will continue to improve in the future, and it is also possible to move towards 32Bit 384KHz.

In fact, the (genuine) products sold in the current CD market are usually HDCD discs at the lowest level, when you buy discs, you find that they are basically HDCD logos, that is, a CD contains two audio tracks: Normal CD track and HDCD track. The CD track records a 16-bit signal at 44.1 KHz (this is the compatible content on this disc, considering early CD players), and the HDCD track records a 24-bit signal at 96 KHz ( this is the main content of the disc). Ordinary CD players can only play CD audio track signals, and HDCD audio tracks require an HDCD player to play (in fact, most DVD players today can play HDCDs, and modern computers work even better).

Definition of sampling bits, sampling rate and bit rate in audio (transfer) Part 2

Bitrate values compared to real audio:

16 Kbps = phone quality
24 Kbps = increase phone quality, shortwave transmission, longwave transmission, European standard medium wave transmission
40 Kbps = American standard medium wave transmission
56Kbps=Voice
64 Kbps = boost voice (best bitrate setting for cell phone ringtones, best setting for cell phone mono MP3 players)
112 Kbps = FM stereo FM transmission
128 Kbps = tape (best setting for a mobile phone stereo MP3 player, best setting for a low-end MP3 player)
160 Kbps = HIFI high fidelity (best setting for mid to high end MP3 players)
192Kbps=CD (best setting for high-end MP3 players)
256Kbps=Studio Music Studio (for music enthusiasts)
In fact, with the advancement of technology, the bitrate is also getting higher and higher, the maximum bitrate of MP3 is 320Kbps, but some formats can reach higher bitrates and superior sound quality.
For example, the emerging APE audio format can provide true audiophile lossless sound quality and smaller volume than WAV format, and its bit rate is usually 550kbps—–950kbps.
Common coding patterns:

Dynamic bit rate VBR (Variable Bitrate), ie there is no fixed bit rate. The compression software immediately determines which bitrate to use based on the audio data during compression. This is a method that takes into account the quality of the file. and file size The recommended encoding mode;
ABR (Average Bit Rate) Average Bit Rate is an interpolation parameter of VBR. LAME created this encoding mode in response to the low file volume ratio of CBR and the variable size of files generated by VBR. Within the specified file size, ABR takes every 50 frames (about 1 second for 30 frames) as a segment. A relatively low flow rate is used for low frequency and insensitive frequencies, and a high flow rate is used for high frequencies and high dynamic performance. It can be used as VBR and CBR, a compromise option.
CBR (constant bit rate), constant bit rate, means that the file has a bit rate from start to finish. Compared to VBR and ABR, the compressed file size is very large and the sound quality will not improve significantly compared to VBR and ABR.
In simple terms:

In a nutshell, sample rate and bit rate are like horizontal and vertical coordinates on the coordinate axis.

The sampling rate on the abscissa represents the number of samples per second.

The bit rate on the ordinate represents the precision when quantizing analog quantities with digital quantities.

The sample rate is similar to the number of frames of moving images. For example, the sampling rate of movies is 24 Hz, the sampling rate of PAL format is 25 Hz, and the sampling rate of NTSC format is 30 Hz. When we play back the still images sampled at the same rate as the sampling frequency, we see a continuous image. In the same way, when a CD recorded at a sampling rate of 44.1 kHz is played back at the same rate, a continuous sound can be heard. Obviously, the higher the sample rate, the more coherent the sound will be heard and the picture will be seen. Of course, the sampling rate that human auditory and visual organs can distinguish is limited, which is basically higher than sound sampled at 44.1 kHz, and most people haven’t noticed the difference.

The number of digits in the sound is equivalent to the number of colors on the screen, indicating the amount of data per sample. Of course, the larger the amount of data, the more accurate the playback sound, so as not to confuse the sound. of the teapot with the train whistle. In the same way, it is more clear and precise for the image, so as not to confuse blood and ketchup. However, limited by the function of human organs, 16-bit sound and 24-bit image are basically the limits of ordinary humans, and the highest digits can only be distinguished by instruments. For example, the phone has 7-bit sound sampled at 3 kHz and the CD has 16-bit sound sampled at 44.1 kHz, so the CD is clearer than the phone.