difference between sample rate and bit rate Archives

Bitrate Part 2

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Bitrate Part 2

bitrate

The amount of information transmitted through the channel per unit of time is called the bit rate, and the unit is bits per second (bit/s), called the bit rate.

BITRATE

Bitrate is often used in communications as a synonym for connection speed, transmission speed, channel capacity, peak throughput, and digital bandwidth capacity. The higher the bit rate, the higher the data transfer. Bit rate in video refers to the sampling rate at which an analog signal is converted to a digital signal [4] . Video file quality is often measured in terms of bitrate. [4] .
Distinction of conceptedit transmission
Baud rate is also known as waveform rate or modulation rate. The code for a data unit is represented by a finite combination of numbers, each of which is a symbol (or code point). In electrical communication, an electrical waveform is often used to represent one or more symbols. Waveforms with different characteristics may represent different symbol values or symbol combination values, and the duration of the waveform corresponds to the duration of the symbol or symbol combination it represents. Obviously, the shorter the duration of an electrical waveform, the more waveforms are transmitted in a unit of time, or the more data is transmitted, that is, the higher the data rate. Therefore, we can define the baud rate as follows: In the process of data transmission, the number of waveforms transmitted per unit time on the line is the baud rate, and its unit is “baud” [5] .
“Bit rate” and “baud rate” are speed units defined in two different concepts, and it is often easy to confuse them when you are not careful. When binary waveform is used, baud rate and bit rate have the same value, but their meanings are different [5] .
Difference: Both bit rate and baud rate are units that measure the transmission rate of a modem. In data transmission, data information is represented by binary numbers “0” and “1”, and each binary number is called 1 bit. The number of bits transmitted through the channel per unit of time is called the bit rate, expressed in bits per second, usually abbreviated as bit/s. The number of symbols transmitted through the channel per unit of time is called the baud rate, also called the modulation rate. Bit rate and baud rate are consistent only when modulated with two values. For example, in quadrature modulation, every two bits of the data signal form a symbol, and there are 4 values: 00, 01, 10 and 11, which represent the phase changes of the 4 types of carrier signals respectively, for Therefore, send such a symbol. It is equivalent to transmitting two bits of data, and the baud rate is equivalent to half the bit rate. The usual transmission rates of 300, 600, 1200 and 9600, etc., refer to the baud rate, which indicates that the number of binary numbers transmitted per unit of time is 300, 600, 1200 and 9600 [6] .

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Bit rate

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

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] .

Sample rate and bit rate of MP3 Part 2

BIT RATE

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 higher 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. ]

When you understand the above two concepts, bitrate is easy to understand. Take the phone as an example, 3000 samples per second, each sample is 7 bits, then the phone’s bit rate is 21000. And the CD is 44100 samples per second, two channels, each sample is 13 bit PCM encoded, so the CD bit rate is 44100*2*13=1146600, which means the CD data volume per second is about 144KB. the capacity of a CD is 74 minutes equal to 4440 seconds, which is 639360KB=640MB.

Sound is actually a type of energy wave, so it also has the characteristics of frequency and amplitude, with frequency corresponding to the time axis and amplitude corresponding to the level axis. The wave is infinitely smooth, and the string can be considered to be made up of innumerable points. Since the storage space is relatively limited, in the process of digital encoding, the points of the string must be sampled. The sampling process consists of extracting the frequency value of a certain point. Obviously, the more points that are extracted in one second, the richer the frequency information that can be obtained. To restore the waveform, there must be two sampling points in one vibration. The highest frequency that can be felt is 20kHz, so to meet the auditory requirements of the human ear, at least 40k samples per second, expressed at 40kHz, and this 40kHz is the sample rate. Our common CD has a sample rate of 44.1 kHz. It is not enough to have only frequency information, we must also obtain and quantify the energy value of this frequency to represent the strength of the signal. The number of quantization levels is an integer power of 2, and the sample size of our common CD bit is 16 bits, that is, 2 to the power of 16. Sample size is harder to understand than bit rate. sampling, because it makes it seem abstract. For a simple example: suppose a wave is sampled 8 times, and the energy values corresponding to the sampling points are A1-A8, but we only use 2-bit sampling size, as a result we can only keep the 4 point values in A1-A8 and discard the other 4. If we use the 3bit sample size, all 8 point information is recorded. The higher the sample rate and sample size values, the closer the recorded waveform is to the original signal.

MP3 sample rate and bit rate

Bit Rate

When we listen to mp3 and watch movies, we will notice two parameters.

BIT RATE

The most common ones are 44.1 KHz sample rate and 192 Kbps bit rate. So what is the sample rate and what is the bit rate? What is the relationship between them? Explain:

The process of converting an analog audio signal to a digital audio signal is called sampling. In a nutshell, how many data points does it take to record a 1 second long sound via waveform sampling. For example: the sound sample rate of 44.1 KHz is equivalent to spending 44,000 data points to describe the sound waveform for 1 second. In principle, the higher the sample rate, the better the sound quality; sampling frequency is generally divided into three levels: 22.05KHz, 44.1KHz and 48KHz; 22.05KHz can only achieve FM radio sound quality, and 44.1KHz is the theoretical limit of CD sound quality, 48KHz has reached DVD quality.

Sampling rate refers to the sampling frequency when converting sound (analog signal) to mp3 (digital signal), i.e. how many data points are sampled per unit of time. (The data for a sample point is 8 (or even more) bits long.)

Bit rate refers to the number of bits (bits) transmitted per second. The unit is bps (bit per second). The higher the bitrate, the more data transmitted and the better the sound quality.

It can be said that the sample rate and bit rate are like the horizontal and vertical coordinates on the coordinate axis. The sampling frequency on the abscissa represents the data points sampled 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 44.1kHZ, and most people haven’t noticed the difference. ]

Quality (bit rate)

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

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

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.