What Is Audio Sampling Rate: A Comprehensive Explanation


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What Is Audio Sampling Rate: A Comprehensive Explanation

Sample Rate
Sample Rate

Introduction

Sample Rate
Sample Rate

Audio sampling rate is a fundamental concept in digital audio that refers to the number of samples per second used to represent an analog audio signal in digital form. In this article, we’ll explore the technical details of audio sampling rate, its importance in digital audio, and its impact on audio quality and file size.

Sampling Rate Fundamentals

The concept of audio sampling rate is based on the Nyquist-Shannon sampling theorem, which states that in order to accurately represent an analog signal in digital form, the sampling rate must be at least twice the highest frequency present in the signal. This means that a signal with a highest frequency of 20kHz (the upper limit of human hearing) must be sampled at a rate of at least 40kHz in order to be accurately represented.

Sampling rate is measured in Hertz (Hz), which refers to the number of samples per second. Common sampling rates in digital audio range from 44.1kHz (used in CDs) to 192kHz (used in some high-resolution audio formats).

Sample Rate Conversion

In some cases, it may be necessary to convert audio from one sampling rate to another. Sample rate conversion involves resampling the audio data to a different rate, which can be done using digital signal processing techniques. However, sample rate conversion can introduce artifacts and reduce audio quality, especially when downsampling from a higher rate to a lower rate.

There are various reasons why sample rate conversion may be necessary, such as when mixing audio tracks with different sampling rates, or when preparing audio for distribution on different platforms with varying requirements.

Audio Quality and Sampling Rate

The sampling rate has a significant impact on audio quality, with higher sampling rates generally resulting in better fidelity and more accurate representation of the original signal. However, the benefits of higher sampling rates are limited by the limitations of human hearing and the practical limitations of digital audio technology.

While there is debate about the benefits of “high-resolution audio” formats with sampling rates above 44.1kHz, it is generally accepted that sampling rates above 96kHz provide little additional benefit in terms of audio quality.

Bit Depth and Sampling Rate

The bit depth of an audio sample refers to the number of bits used to represent the amplitude of the signal at each sample point. Higher bit depths allow for more precise representation of the signal, but also result in larger file sizes. The bit depth and sampling rate are related, as increasing the bit depth requires more data to be stored for each sample.

There is a trade-off between sampling rate and bit depth, as higher sampling rates require more data to be stored per second, which can limit the maximum bit depth that can be used without exceeding practical file size limits. However, this trade-off can be mitigated by using efficient audio compression techniques.

Sample Rate in Practice

Common sampling rates in digital audio include 44.1kHz (used in CDs), 48kHz (used in digital video), 88.2kHz, 96kHz, 176.4kHz, and 192kHz. Streaming services such as Spotify and Apple Music typically use lower sampling rates for their audio streams, with 44.1kHz being a common choice.

The Nyquist Theorem, named after the Swedish-American physicist Harry Nyquist, states that the sampling rate should be at least twice the highest frequency component in the signal being sampled. This is why the standard CD quality sampling rate is 44.1 kHz, which is just above the upper limit of human hearing.

However, it is important to note that there are higher sampling rates available, such as 48 kHz, 96 kHz, and even 192 kHz. These higher sampling rates can provide more detail and accuracy in the digital representation of the analog signal. However, they also require more storage space and processing power.

Another important factor to consider is the bit depth, which is the number of bits used to represent each sample. The more bits used, the more accurate and detailed the representation of the analog signal. CD quality uses a bit depth of 16 bits, but higher bit depths such as 24 bits are also available.

It is worth noting that some argue that higher sampling rates and bit depths may not necessarily result in audible improvements in sound quality, especially when considering the limitations of human hearing. Additionally, some argue that the increased storage and processing requirements may not be worth the potential improvements.

In conclusion, the sampling rate is a crucial component in the digital representation of analog audio signals. A higher sampling rate can provide more detail and accuracy in the digital representation, but also requires more storage and processing power. The Nyquist Theorem provides a guideline for choosing the appropriate sampling rate based on the highest frequency component in the signal. Additionally, the bit depth is another factor to consider in the accuracy and detail of the digital representation. While higher sampling rates and bit depths are available, the potential improvements in sound quality must be balanced against the increased storage and processing requirements.


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What does the quality of an mp3 depend on? high resolution mp3

What does the quality of an mp3 depend on? high resolution mp3

high resolution mp3
high resolution mp3

Factors influencing hearing quality

high resolution mp3
high resolution mp3

High quality

Lately, very high quality audios have been promoted… are they really convenient?

We could say that if we strictly base ourselves on technical aspects, they could be considered of higher quality.

For example, they get to use sample rates of more than double the highest currently used.

The same happens with the bit rate, they use numbers that until now were not used at all.

Pewro first we must ask ourselves if the equipment we use to read them (the computer, a cell phone, an mp4 player) are capable of handling these qualities and if the speakers or headphones are also enabled and built to do the same.

Otherwise we will end up paying a lot for this super audio and effectively get the same.

It is worth additionally thinking about whether our ears could differentiate between one and the other.

To what extent our ear perceives the difference between 4800 and 96000 as a sample rate.

What we must avoid is falling victim to the “numbers”, which will show us that in theory they will sound better, but avoid touching reality – for example the human ear or the quality of our speakers – and therefore the theory ends up being misleading.

Bit rate

Bit rate

Bitrate

Bit rate refers to the number of bits (bit) transmitted per unit of time, in bps (bit per second).

bit rate

Bit rate is also known as “binary bit rate”, commonly known as “code rate”. Indicates the number of bits transmitted per unit of time. It is used to measure the transmission speed of digital information, often written as bit/sec. According to the number of bits occupied by each image storage frame and the transmission bit rate, the digital image information transmission speed can be calculated [1].
In modern digital communication, the transmission volume of digitized video and other information is large, so it is often measured in kilobits per second or megabits per second, which are written as kbit/sec (or kbps) and Mbit/sec. (or Mbps respectively). ). For example, the amount of information digitized from an ordinary color TV signal can reach 216 Mbit/sec. A good digital broadcast channel can transmit dozens of color TV programs, and its capacity can reach several gigabits or gigabits per second (written as Gbit/sec or Gbps) [1] .
Bitrate is often used to measure the quality of video files.
Bitrate is often used to measure the quality of video files.
flexibility edit stream
Because each network is unique and each access line has different conditions (such as length, attenuation, crosstalk environment, etc.), access lines from different telephone companies must support different data rates. For ADSL and VDSL modems, it is best to set the data rate to one of many possible data rates. For example, DMT-based ADSL and VDSL can theoretically change the tariff at fine intervals, and CAP-based RADSL (Rate Adaptive ADSL) also provides some flexibility in tariff configuration [2].
However, telephone companies may want to limit xDSL service to a small set of rates sufficient to provide a variety of services. If a limited set of tariffs can be adapted to a wide range of services, then the management of the services in this case is simpler than in the case of variable tariffs. Telephone companies want the choice of modem speed to be under the control of the network, not the user [2] .
In this mode, the selection of the transmission rate set of the xDSL network must be prudent. In this case, there is a possibility that two adjacent systems receive traffic at very different rates and the system must be able to handle such a situation. The other model, the “best match” approach using adaptive rate ADSL (similar to a voiceband modem), is more beneficial to new network operators and Internet Service Providers (ISPs) [2] .
Transmission control method
Most bit rate control schemes consist of two parts. Part of the encoded bit stream output by the encoder is fed into a buffer. For a constant bitrate channel, the data in the buffer is fetched at a constant rate, and if the buffer is large enough, the bitrate variation caused by the MPEG picture type, etc. can be smoothed out. This is necessary for both constant bit rate transmission and variable bit rate transmission in general. However, in practice, the buffer size is always limited. The buffering process will bring a delay to the system, and this delay is proportional to the size of the buffer. Latency is often a serious issue for real-time image communication, so buffers should be kept as small as possible. That is, long-term fluctuations in bitrate due to changes in scene content or changes, etc. they cannot be softened in this way, so another part is needed. This is to send some measure of the output bitrate to the encoder to control the encoding process, thus changing the output bitrate [3] .

Quality (bit rate)

Quality (bit rate)

Bit Rate

In multimedia technology, quality is often used to judge the effect of audio, and quality here is actually bitrate.

Bit Rate

1. Introduction
2 sound control
3 encoding mode
Introductionedit transmission
The term quality is widely used.
In multimedia technology, quality is often used to judge the effect of audio, and quality here is actually bitrate.
On WINDOWS it is called “bit rate” and on some players it is described as ” bit rate “.
Quality refers to the bit rate at which digital sound is converted from analog to digital format. The higher the bitrate, the better the quality of the restored sound.
sound control edit stream
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 broadcast FM 128 Kbps = tape (best setting for mobile phone stereo MP3 player, best setting for 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 quality of music is also getting higher and higher, the highest quality of MP3 is 320Kbps, but some formats can achieve higher sound quality.
For example, the emerging APE audio format can provide real audiophile level lossless sound quality and smaller volume than WAV format, and its quality is usually 550kbps-950kbps.
encoding modeedit stream
VBR (Variable Bitrate) Dynamic Bitrate means there is no fixed bitrate. The compression software immediately determines which bitrate to use based on the audio data being compressed. This is a method that takes quality as a premise and takes file size into account 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. High-frequency and insensitive frequencies use relatively low traffic, and low-frequency and large dynamic performance use high traffic, which can be used as VBR and CBR, a compromise option.
CBR (constant bitrate), constant bitrate means the file has one bitrate 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.

Difference between digital and analog

Difference between digital and analog

Difference Between Analog Signal and Digital Signal

The sound is analog. And sound is the vibration of the air. How is this sound vibration transmitted?

Analog VS Digital
For example, when a stone is thrown into a calm water surface, the ripples spread around it, but if
Cut in the direction of the waves and look at the cut end, the waveform is as shown in Fig.1.

Air waves spread from the point where sound is emitted even in air. Although it is invisible to the eye, it has a
similar waveform. This is the analog waveform of sound.

Therefore, although it is digital, when such a sound waveform is recorded or communicated by phone or wireless, as
shown in Fig. 2, the change in the analog waveform is electrically replaced with a series of numerical values ​​according to a certain promise. ..

When recording or communicating, if you handle it as analog, it is easy for noise to enter and the sound quality to deteriorate, but when trying
the waveform of the sound as digital = numerical data, you can eliminate that worry and
maintain a certain quality. You can do various processing while maintaining it.

(2) What is convenient when it is digital

Digital audio signals are convenient because they can be recorded and edited using a personal computer, for example.

In addition, 74 minutes of music can be recorded on a CD with a diameter of only 12 cm, and through digital compression processing
, music of the same length can be recorded on an MD with a smaller diameter.

Since digital signals can be compressed in this way, it is also convenient for storing large amounts of information.
Not only sound, but also video signals with a higher amount of information can be recorded and communicated at high speed by using compression technology.

Especially in communication, a two-way digital multiplex communication can be realized communicating multiple pieces of information with a single wire.
In addition to electrical signals, laser light can also be used for optical communication, making communication possible at extremely high speeds.

(3) What is the sampling frequency?

Digital signals are processed at predetermined fixed time intervals.
The sample rate (sample rate) indicates how many times a second is processed and is expressed as Fs or fs.

The sampling frequency unit is Hz (Hertz), and the
44.1 kHz (kilohertz) sample rate means 44,100 pieces of data are processed per second.
(K represents 1000 times)

AD conversion converts a continuous analog signal into a digital signal,
measures the size of the signal at each moment determined by the sampling frequency (sampling) and converts
the result in a binary number (quantization).

On the other hand, DA conversion converts a digital signal into an analog signal,
It reads the digital signal in the sample rate time interval and connects it smoothly.

Since digital signals can be reproduced up to half the sampling frequency, how much
The higher the sample rate, the higher the playable frequency and the better the sound quality.
In familiar areas, 44.1 kHz is used for CD, and 48 kHz is used for DAT and B modes of satellite transmission.

In addition, recent professional equipment uses high sampling frequencies (high sampling), such as 88.2 kHz and 96 kHz,
and are designed to faithfully reproduce even higher frequency sounds to improve sound quality.

What is the audio bit rate? Relationship between “bit depth” and “sample rate” and sound quality PART 2

What is the audio bit rate? Relationship between “bit depth” and “sample rate” and sound quality PART 2

Audio bit rate

What is the bit depth that determines the sound quality?
What is the bit depth that determines the quality of the sound?
The bit rate is calculated by multiplying the two factors of bit depth and sample rate. Bit depth represents the amount of data per divided sample and is an element that affects the quality and expressiveness of the sound.

The sample rate is the finely divided horizontal axis and the bit depth is the one that overlaps the vertical axis. By providing a large amount of data (bit depth) to each sample, it is possible to create finer and more accurate audio data.

Bit depth is the precision of each image in animation production. The higher the bit depth, the more expressive the sound and effects will be, and the higher the sound quality will feel.

What is the sample rate that determines the smoothness?
Of the bit rates that determine audio quality, the sample rate is an element that represents the number of data divisions per second on the horizontal axis. It shows how many tens of thousands of data are divided per second and the higher the number of divisions, the higher the sound reproducibility and sound quality.

It is even easier to understand if you consider it a mechanism similar to animation production. The more images you use per second, the smoother your character will move, and the higher the sample rate (the number of data divisions), the smoother your sound will sound. Also, the amount of data increases according to the size of the sample rate.

In conclusion
To improve sound quality, it is important to increase the audio bit rate. The audio bit rate is determined by two factors: the bit depth, which determines the expressiveness of the sound, and the sampling rate, which determines the smoothness of the sound.

However, keep in mind that the higher the audio bit rate, the more beautiful the audio will be and will also be affected by the original sound source. If you want to create high-quality data from the moment of recording, why not ask a production company that has high-quality recording equipment?

I think there are many people who are concerned about the balance between capacity and sound quality when it comes to the audio system mounted on a computer. Professional engineers and mixers will come up with the best balance, so don’t hesitate to contact us on these points.

What is the audio bit rate? Relationship between “bit depth” and “sample rate” and sound quality

What is the audio bit rate? Relationship between “bit depth” and “sample rate” and sound quality

Digital Audio Basics

Bit rate is one of the factors that determine the quality of a video job.

BitRate

Among them, the one that determines the audio finish is the audio bit rate. Understanding the bit rate will allow you to control the sound quality and create better quality audio.

So on this occasion, I will explain bit depth and sample rate which are indispensable when talking about sound quality.

What is a bit rate?
What is a bit rate?
■ Bit rate represents high quality

Bit rate is a numerical value that represents the amount of information in the data, and the height of the bit rate is proportional to the quality of the data.

By looking at the bit rate, you can see how much data is packed in one second. Generally, the higher the bit rate, the higher the sound quality, and the sound and video are more realistic.

■ Audio bit rate and video bit rate

In the case of video, the bit rate is divided into two types, video and audio, and the overall quality of the video is determined by the height of each bit rate.

The “video data rate” shown in the video properties is the video bit rate and the bit rate assigned to “audio” is the audio bit rate. The “total bit rate” is the total of the two types of bit rates, video and audio, and is the bit rate of all video.

However, high bit rate audio is not always good. This is because if the quality of the original audio data or the equipment used for recording is poor, poor quality audio will be played. For example, if the original voice contains noise, the noise part will be reproduced realistically and the voice will be difficult to hear. Similarly, if the bit rate of the video is low, the movement will be choppy and unnatural, or the video will be uneven.

“High resolution” basics. What is the difference between DSD, FLAC, MQA, etc.? Part 7

“High resolution” basics. What is the difference between DSD, FLAC, MQA, etc.? Part 7

DSD vs. PCM

MQA encoding processing can be performed on 44.1 kHz to 768 kHz linear PCM sound sources and can be stored in existing file formats (container formats) such as FLAC, ALAC, and WAV. If you use a compatible device equipped with a dedicated decoder, you can “open origami” and demonstrate the original quality. It can be played with an unsupported device, but in that case, “Origami cannot be opened”, so normal PCM playback will be performed that does not include the original MQA information.

MQA is available for download at e-onkyo music. It has the advantage that it can be used with a low communication volume even with streaming type distribution, and abroad, the “TIDAL” music distribution is developing a high quality 96 kHz / 24 bit MQA streaming distribution ” TIDAL Masters “.

High Resolution Portable Player “DP-X1” Quickly Achieved MQA Compatibility
Current Status of Supported High-Resolution Formats and Music Distribution Services (as of July 2017)
WAV FLAC ALAC DSD MQA others
e-onkyo music 〇 〇 — 〇 〇 Dolby TrueHD
blackberry — 〇 — 〇 — —
OTOTOY 〇 〇 〇 〇 — —
Recochoku — 〇 — — — —
My sound — 〇 — — — —
slots — 〇 — 〇 — —
Is FLAC a powerful option right now?
Generally speaking, the high resolution format has been “PCM or DSD”, and for PCM, the choice has been “lossless compression” or “uncompressed”. There may be circumstances on the side of the playback device and sound preferences, but in terms of the balance between data compression effect (file capacity) and sound quality, “lossless” is a reasonable existence.

Among them, FLAC, which is easy to use on smartphones, will remain high resolution as source code is published and royalties are not incurred, it is already compatible with many compatible high resolution audio devices, and will be supported in the system level in the next iOS (iOS 11), will be the centerpiece of the format.

Looking around, there are signs of change, such as the emergence of a new face, MQA, and the spread of streaming services, but the audio format must have attractive content. Above all, I would like to hope that the record company / distribution service company is actively working in high resolution.

“High resolution” basics. What is the difference between DSD, FLAC, MQA, etc.? Part 6

“High resolution” basics. What is the difference between DSD, FLAC, MQA, etc.? Part 6

PCM DSD

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

There are several formats, even if it says “high resolution”. 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 that indicates the number of steps in which the amplitude of a signal is expressed when an analog signal is converted into a signal.